resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
diff --git a/camera/CameraMetadata.cpp b/camera/CameraMetadata.cpp
index 7765914..6b726e0 100644
--- a/camera/CameraMetadata.cpp
+++ b/camera/CameraMetadata.cpp
@@ -25,6 +25,9 @@
namespace android {
+#define ALIGN_TO(val, alignment) \
+ (((uintptr_t)(val) + ((alignment) - 1)) & ~((alignment) - 1))
+
typedef Parcel::WritableBlob WritableBlob;
typedef Parcel::ReadableBlob ReadableBlob;
@@ -431,40 +434,70 @@
*out = NULL;
}
- // arg0 = metadataSize (int32)
- int32_t metadataSizeTmp = -1;
- if ((err = data.readInt32(&metadataSizeTmp)) != OK) {
+ // See CameraMetadata::writeToParcel for parcel data layout diagram and explanation.
+ // arg0 = blobSize (int32)
+ int32_t blobSizeTmp = -1;
+ if ((err = data.readInt32(&blobSizeTmp)) != OK) {
ALOGE("%s: Failed to read metadata size (error %d %s)",
__FUNCTION__, err, strerror(-err));
return err;
}
- const size_t metadataSize = static_cast<size_t>(metadataSizeTmp);
+ const size_t blobSize = static_cast<size_t>(blobSizeTmp);
+ const size_t alignment = get_camera_metadata_alignment();
- if (metadataSize == 0) {
+ // Special case: zero blob size means zero sized (NULL) metadata.
+ if (blobSize == 0) {
ALOGV("%s: Read 0-sized metadata", __FUNCTION__);
return OK;
}
- // NOTE: this doesn't make sense to me. shouldnt the blob
+ if (blobSize <= alignment) {
+ ALOGE("%s: metadata blob is malformed, blobSize(%zu) should be larger than alignment(%zu)",
+ __FUNCTION__, blobSize, alignment);
+ return BAD_VALUE;
+ }
+
+ const size_t metadataSize = blobSize - alignment;
+
+ // NOTE: this doesn't make sense to me. shouldn't the blob
// know how big it is? why do we have to specify the size
// to Parcel::readBlob ?
-
ReadableBlob blob;
// arg1 = metadata (blob)
do {
- if ((err = data.readBlob(metadataSize, &blob)) != OK) {
- ALOGE("%s: Failed to read metadata blob (sized %d). Possible "
+ if ((err = data.readBlob(blobSize, &blob)) != OK) {
+ ALOGE("%s: Failed to read metadata blob (sized %zu). Possible "
" serialization bug. Error %d %s",
- __FUNCTION__, metadataSize, err, strerror(-err));
+ __FUNCTION__, blobSize, err, strerror(-err));
break;
}
- const camera_metadata_t* tmp =
- reinterpret_cast<const camera_metadata_t*>(blob.data());
+ // arg2 = offset (blob)
+ // Must be after blob since we don't know offset until after writeBlob.
+ int32_t offsetTmp;
+ if ((err = data.readInt32(&offsetTmp)) != OK) {
+ ALOGE("%s: Failed to read metadata offsetTmp (error %d %s)",
+ __FUNCTION__, err, strerror(-err));
+ break;
+ }
+ const size_t offset = static_cast<size_t>(offsetTmp);
+ if (offset >= alignment) {
+ ALOGE("%s: metadata offset(%zu) should be less than alignment(%zu)",
+ __FUNCTION__, blobSize, alignment);
+ err = BAD_VALUE;
+ break;
+ }
+
+ const uintptr_t metadataStart = reinterpret_cast<uintptr_t>(blob.data()) + offset;
+ const camera_metadata_t* tmp =
+ reinterpret_cast<const camera_metadata_t*>(metadataStart);
+ ALOGV("%s: alignment is: %zu, metadata start: %p, offset: %zu",
+ __FUNCTION__, alignment, tmp, offset);
metadata = allocate_copy_camera_metadata_checked(tmp, metadataSize);
if (metadata == NULL) {
// We consider that allocation only fails if the validation
// also failed, therefore the readFromParcel was a failure.
+ ALOGE("%s: metadata allocation and copy failed", __FUNCTION__);
err = BAD_VALUE;
}
} while(0);
@@ -485,38 +518,79 @@
const camera_metadata_t* metadata) {
status_t res = OK;
- // arg0 = metadataSize (int32)
+ /**
+ * Below is the camera metadata parcel layout:
+ *
+ * |--------------------------------------------|
+ * | arg0: blobSize |
+ * | (length = 4) |
+ * |--------------------------------------------|<--Skip the rest if blobSize == 0.
+ * | |
+ * | |
+ * | arg1: blob |
+ * | (length = variable, see arg1 layout below) |
+ * | |
+ * | |
+ * |--------------------------------------------|
+ * | arg2: offset |
+ * | (length = 4) |
+ * |--------------------------------------------|
+ */
+ // arg0 = blobSize (int32)
if (metadata == NULL) {
+ // Write zero blobSize for null metadata.
return data.writeInt32(0);
}
+ /**
+ * Always make the blob size sufficiently larger, as we need put alignment
+ * padding and metadata into the blob. Since we don't know the alignment
+ * offset before writeBlob. Then write the metadata to aligned offset.
+ */
const size_t metadataSize = get_camera_metadata_compact_size(metadata);
- res = data.writeInt32(static_cast<int32_t>(metadataSize));
+ const size_t alignment = get_camera_metadata_alignment();
+ const size_t blobSize = metadataSize + alignment;
+ res = data.writeInt32(static_cast<int32_t>(blobSize));
if (res != OK) {
return res;
}
- // arg1 = metadata (blob)
+ size_t offset = 0;
+ /**
+ * arg1 = metadata (blob).
+ *
+ * The blob size is the sum of front padding size, metadata size and back padding
+ * size, which is equal to metadataSize + alignment.
+ *
+ * The blob layout is:
+ * |------------------------------------|<----Start address of the blob (unaligned).
+ * | front padding |
+ * | (size = offset) |
+ * |------------------------------------|<----Aligned start address of metadata.
+ * | |
+ * | |
+ * | metadata |
+ * | (size = metadataSize) |
+ * | |
+ * | |
+ * |------------------------------------|
+ * | back padding |
+ * | (size = alignment - offset) |
+ * |------------------------------------|<----End address of blob.
+ * (Blob start address + blob size).
+ */
WritableBlob blob;
do {
- res = data.writeBlob(metadataSize, &blob);
+ res = data.writeBlob(blobSize, &blob);
if (res != OK) {
break;
}
- copy_camera_metadata(blob.data(), metadataSize, metadata);
-
- IF_ALOGV() {
- if (validate_camera_metadata_structure(
- (const camera_metadata_t*)blob.data(),
- &metadataSize) != OK) {
- ALOGV("%s: Failed to validate metadata %p after writing blob",
- __FUNCTION__, blob.data());
- } else {
- ALOGV("%s: Metadata written to blob. Validation success",
- __FUNCTION__);
- }
- }
+ const uintptr_t metadataStart = ALIGN_TO(blob.data(), alignment);
+ offset = metadataStart - reinterpret_cast<uintptr_t>(blob.data());
+ ALOGV("%s: alignment is: %zu, metadata start: %p, offset: %zu",
+ __FUNCTION__, alignment, metadataStart, offset);
+ copy_camera_metadata(reinterpret_cast<void*>(metadataStart), metadataSize, metadata);
// Not too big of a problem since receiving side does hard validation
// Don't check the size since the compact size could be larger
@@ -528,6 +602,9 @@
} while(false);
blob.release();
+ // arg2 = offset (int32)
+ res = data.writeInt32(static_cast<int32_t>(offset));
+
return res;
}
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 61f83e3..b6f150c 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -45,6 +45,7 @@
#include <signal.h>
#include <getopt.h>
#include <sys/wait.h>
+#include <termios.h>
#include <assert.h>
#include "screenrecord.h"
@@ -61,6 +62,7 @@
// Command-line parameters.
static bool gVerbose = false; // chatty on stdout
static bool gRotate = false; // rotate 90 degrees
+static bool gRawOutput = false; // generate raw H.264 byte stream output
static bool gSizeSpecified = false; // was size explicitly requested?
static bool gWantInfoScreen = false; // do we want initial info screen?
static bool gWantFrameTime = false; // do we want times on each frame?
@@ -298,10 +300,12 @@
* input frames are coming from the virtual display as fast as SurfaceFlinger
* wants to send them.
*
+ * Exactly one of muxer or rawFp must be non-null.
+ *
* The muxer must *not* have been started before calling.
*/
static status_t runEncoder(const sp<MediaCodec>& encoder,
- const sp<MediaMuxer>& muxer, const sp<IBinder>& mainDpy,
+ const sp<MediaMuxer>& muxer, FILE* rawFp, const sp<IBinder>& mainDpy,
const sp<IBinder>& virtualDpy, uint8_t orientation) {
static int kTimeout = 250000; // be responsive on signal
status_t err;
@@ -311,6 +315,8 @@
int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
DisplayInfo mainDpyInfo;
+ assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
+
Vector<sp<ABuffer> > buffers;
err = encoder->getOutputBuffers(&buffers);
if (err != NO_ERROR) {
@@ -342,15 +348,16 @@
case NO_ERROR:
// got a buffer
if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) != 0) {
- // ignore this -- we passed the CSD into MediaMuxer when
- // we got the format change notification
- ALOGV("Got codec config buffer (%u bytes); ignoring", size);
- size = 0;
+ ALOGV("Got codec config buffer (%u bytes)", size);
+ if (muxer != NULL) {
+ // ignore this -- we passed the CSD into MediaMuxer when
+ // we got the format change notification
+ size = 0;
+ }
}
if (size != 0) {
ALOGV("Got data in buffer %d, size=%d, pts=%lld",
bufIndex, size, ptsUsec);
- assert(trackIdx != -1);
{ // scope
ATRACE_NAME("orientation");
@@ -379,14 +386,23 @@
ptsUsec = systemTime(SYSTEM_TIME_MONOTONIC) / 1000;
}
- // The MediaMuxer docs are unclear, but it appears that we
- // need to pass either the full set of BufferInfo flags, or
- // (flags & BUFFER_FLAG_SYNCFRAME).
- //
- // If this blocks for too long we could drop frames. We may
- // want to queue these up and do them on a different thread.
- { // scope
+ if (muxer == NULL) {
+ fwrite(buffers[bufIndex]->data(), 1, size, rawFp);
+ // Flush the data immediately in case we're streaming.
+ // We don't want to do this if all we've written is
+ // the SPS/PPS data because mplayer gets confused.
+ if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) == 0) {
+ fflush(rawFp);
+ }
+ } else {
+ // The MediaMuxer docs are unclear, but it appears that we
+ // need to pass either the full set of BufferInfo flags, or
+ // (flags & BUFFER_FLAG_SYNCFRAME).
+ //
+ // If this blocks for too long we could drop frames. We may
+ // want to queue these up and do them on a different thread.
ATRACE_NAME("write sample");
+ assert(trackIdx != -1);
err = muxer->writeSampleData(buffers[bufIndex], trackIdx,
ptsUsec, flags);
if (err != NO_ERROR) {
@@ -418,12 +434,14 @@
ALOGV("Encoder format changed");
sp<AMessage> newFormat;
encoder->getOutputFormat(&newFormat);
- trackIdx = muxer->addTrack(newFormat);
- ALOGV("Starting muxer");
- err = muxer->start();
- if (err != NO_ERROR) {
- fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
- return err;
+ if (muxer != NULL) {
+ trackIdx = muxer->addTrack(newFormat);
+ ALOGV("Starting muxer");
+ err = muxer->start();
+ if (err != NO_ERROR) {
+ fprintf(stderr, "Unable to start muxer (err=%d)\n", err);
+ return err;
+ }
}
}
break;
@@ -457,6 +475,44 @@
}
/*
+ * Raw H.264 byte stream output requested. Send the output to stdout
+ * if desired. If the output is a tty, reconfigure it to avoid the
+ * CRLF line termination that we see with "adb shell" commands.
+ */
+static FILE* prepareRawOutput(const char* fileName) {
+ FILE* rawFp = NULL;
+
+ if (strcmp(fileName, "-") == 0) {
+ if (gVerbose) {
+ fprintf(stderr, "ERROR: verbose output and '-' not compatible");
+ return NULL;
+ }
+ rawFp = stdout;
+ } else {
+ rawFp = fopen(fileName, "w");
+ if (rawFp == NULL) {
+ fprintf(stderr, "fopen raw failed: %s\n", strerror(errno));
+ return NULL;
+ }
+ }
+
+ int fd = fileno(rawFp);
+ if (isatty(fd)) {
+ // best effort -- reconfigure tty for "raw"
+ ALOGD("raw video output to tty (fd=%d)", fd);
+ struct termios term;
+ if (tcgetattr(fd, &term) == 0) {
+ cfmakeraw(&term);
+ if (tcsetattr(fd, TCSANOW, &term) == 0) {
+ ALOGD("tty successfully configured for raw");
+ }
+ }
+ }
+
+ return rawFp;
+}
+
+/*
* Main "do work" method.
*
* Configures codec, muxer, and virtual display, then starts moving bits
@@ -558,16 +614,26 @@
return err;
}
- // Configure muxer. We have to wait for the CSD blob from the encoder
- // before we can start it.
- sp<MediaMuxer> muxer = new MediaMuxer(fileName,
- MediaMuxer::OUTPUT_FORMAT_MPEG_4);
- if (gRotate) {
- muxer->setOrientationHint(90); // TODO: does this do anything?
+ sp<MediaMuxer> muxer = NULL;
+ FILE* rawFp = NULL;
+ if (gRawOutput) {
+ rawFp = prepareRawOutput(fileName);
+ if (rawFp == NULL) {
+ encoder->release();
+ return -1;
+ }
+ } else {
+ // Configure muxer. We have to wait for the CSD blob from the encoder
+ // before we can start it.
+ muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ if (gRotate) {
+ muxer->setOrientationHint(90); // TODO: does this do anything?
+ }
}
// Main encoder loop.
- err = runEncoder(encoder, muxer, mainDpy, dpy, mainDpyInfo.orientation);
+ err = runEncoder(encoder, muxer, rawFp, mainDpy, dpy,
+ mainDpyInfo.orientation);
if (err != NO_ERROR) {
fprintf(stderr, "Encoder failed (err=%d)\n", err);
// fall through to cleanup
@@ -584,9 +650,13 @@
overlay->stop();
}
encoder->stop();
- // If we don't stop muxer explicitly, i.e. let the destructor run,
- // it may hang (b/11050628).
- muxer->stop();
+ if (muxer != NULL) {
+ // If we don't stop muxer explicitly, i.e. let the destructor run,
+ // it may hang (b/11050628).
+ muxer->stop();
+ } else if (rawFp != stdout) {
+ fclose(rawFp);
+ }
encoder->release();
return err;
@@ -753,6 +823,7 @@
{ "show-frame-time", no_argument, NULL, 'f' },
{ "bugreport", no_argument, NULL, 'u' },
{ "rotate", no_argument, NULL, 'r' },
+ { "raw", no_argument, NULL, 'w' },
{ NULL, 0, NULL, 0 }
};
@@ -818,6 +889,10 @@
// experimental feature
gRotate = true;
break;
+ case 'w':
+ // experimental feature
+ gRawOutput = true;
+ break;
default:
if (ic != '?') {
fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);
@@ -831,17 +906,19 @@
return 2;
}
- // MediaMuxer tries to create the file in the constructor, but we don't
- // learn about the failure until muxer.start(), which returns a generic
- // error code without logging anything. We attempt to create the file
- // now for better diagnostics.
const char* fileName = argv[optind];
- int fd = open(fileName, O_CREAT | O_RDWR, 0644);
- if (fd < 0) {
- fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
- return 1;
+ if (!gRawOutput) {
+ // MediaMuxer tries to create the file in the constructor, but we don't
+ // learn about the failure until muxer.start(), which returns a generic
+ // error code without logging anything. We attempt to create the file
+ // now for better diagnostics.
+ int fd = open(fileName, O_CREAT | O_RDWR, 0644);
+ if (fd < 0) {
+ fprintf(stderr, "Unable to open '%s': %s\n", fileName, strerror(errno));
+ return 1;
+ }
+ close(fd);
}
- close(fd);
status_t err = recordScreen(fileName);
if (err == NO_ERROR) {
diff --git a/cmds/screenrecord/screenrecord.h b/cmds/screenrecord/screenrecord.h
index 95e8a68..9b058c2 100644
--- a/cmds/screenrecord/screenrecord.h
+++ b/cmds/screenrecord/screenrecord.h
@@ -18,6 +18,6 @@
#define SCREENRECORD_SCREENRECORD_H
#define kVersionMajor 1
-#define kVersionMinor 1
+#define kVersionMinor 2
#endif /*SCREENRECORD_SCREENRECORD_H*/
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index 5d2d721..1b2f792 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -23,6 +23,7 @@
#include <gui/Surface.h>
#include <media/AudioTrack.h>
#include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -275,7 +276,8 @@
mExtractor = new NuMediaExtractor;
- status_t err = mExtractor->setDataSource(mPath.c_str());
+ status_t err = mExtractor->setDataSource(
+ NULL /* httpService */, mPath.c_str());
if (err != OK) {
mExtractor.clear();
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index d125ad1..fd02bcc 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -24,6 +24,7 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
#include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -76,7 +77,7 @@
static int64_t kTimeout = 500ll;
sp<NuMediaExtractor> extractor = new NuMediaExtractor;
- if (extractor->setDataSource(path) != OK) {
+ if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
fprintf(stderr, "unable to instantiate extractor.\n");
return 1;
}
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index 90daea2..f4a33e8 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -20,6 +20,7 @@
#include <utils/Log.h>
#include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
@@ -59,7 +60,7 @@
int trimEndTimeMs,
int rotationDegrees) {
sp<NuMediaExtractor> extractor = new NuMediaExtractor;
- if (extractor->setDataSource(path) != OK) {
+ if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
fprintf(stderr, "unable to instantiate extractor. %s\n", path);
return 1;
}
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index b2b9ce5..3c0c7ec 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -19,8 +19,12 @@
#include <inttypes.h>
#include <utils/Log.h>
+#include <signal.h>
+
#include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
+
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -43,6 +47,18 @@
using namespace android;
+volatile static bool ctrlc = false;
+
+static sighandler_t oldhandler = NULL;
+
+static void mysighandler(int signum) {
+ if (signum == SIGINT) {
+ ctrlc = true;
+ return;
+ }
+ oldhandler(signum);
+}
+
struct Controller : public AHandler {
Controller(const char *uri, bool decodeAudio,
const sp<Surface> &surface, bool renderToSurface)
@@ -63,7 +79,30 @@
virtual ~Controller() {
}
+ virtual void printStatistics() {
+ int64_t delayUs = ALooper::GetNowUs() - mStartTimeUs;
+
+ if (mDecodeAudio) {
+ printf("%" PRId64 " bytes received. %.2f KB/sec\n",
+ mTotalBytesReceived,
+ mTotalBytesReceived * 1E6 / 1024 / delayUs);
+ } else {
+ printf("%d frames decoded, %.2f fps. %" PRId64 " bytes "
+ "received. %.2f KB/sec\n",
+ mNumOutputBuffersReceived,
+ mNumOutputBuffersReceived * 1E6 / delayUs,
+ mTotalBytesReceived,
+ mTotalBytesReceived * 1E6 / 1024 / delayUs);
+ }
+ }
+
virtual void onMessageReceived(const sp<AMessage> &msg) {
+ if (ctrlc) {
+ printf("\n");
+ printStatistics();
+ (new AMessage(kWhatStop, id()))->post();
+ ctrlc = false;
+ }
switch (msg->what()) {
case kWhatStart:
{
@@ -76,7 +115,8 @@
#endif
sp<DataSource> dataSource =
- DataSource::CreateFromURI(mURI.c_str());
+ DataSource::CreateFromURI(
+ NULL /* httpService */, mURI.c_str());
sp<MediaExtractor> extractor =
MediaExtractor::Create(dataSource);
@@ -99,7 +139,10 @@
break;
}
}
- CHECK(mSource != NULL);
+ if (mSource == NULL) {
+ printf("no %s track found\n", mDecodeAudio ? "audio" : "video");
+ exit (1);
+ }
CHECK_EQ(mSource->start(), (status_t)OK);
@@ -181,21 +224,7 @@
|| what == ACodec::kWhatError) {
printf((what == ACodec::kWhatEOS) ? "$\n" : "E\n");
- int64_t delayUs = ALooper::GetNowUs() - mStartTimeUs;
-
- if (mDecodeAudio) {
- printf("%" PRId64 " bytes received. %.2f KB/sec\n",
- mTotalBytesReceived,
- mTotalBytesReceived * 1E6 / 1024 / delayUs);
- } else {
- printf("%d frames decoded, %.2f fps. %" PRId64 " bytes "
- "received. %.2f KB/sec\n",
- mNumOutputBuffersReceived,
- mNumOutputBuffersReceived * 1E6 / delayUs,
- mTotalBytesReceived,
- mTotalBytesReceived * 1E6 / 1024 / delayUs);
- }
-
+ printStatistics();
(new AMessage(kWhatStop, id()))->post();
} else if (what == ACodec::kWhatFlushCompleted) {
mSeekState = SEEK_FLUSH_COMPLETED;
@@ -639,6 +668,8 @@
looper->registerHandler(controller);
+ signal(SIGINT, mysighandler);
+
controller->startAsync();
CHECK_EQ(looper->start(true /* runOnCallingThread */), (status_t)OK);
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 8efb39e..daaea27 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -29,6 +29,7 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ALooper.h>
#include "include/NuCachedSource2.h"
@@ -960,7 +961,8 @@
const char *filename = argv[k];
- sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+ sp<DataSource> dataSource =
+ DataSource::CreateFromURI(NULL /* httpService */, filename);
if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
fprintf(stderr, "Unable to create data source.\n");
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index dba67a9..b2abc0f 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -21,6 +21,7 @@
#include <binder/ProcessState.h>
#include <cutils/properties.h> // for property_get
+#include <media/IMediaHTTPService.h>
#include <media/IStreamSource.h>
#include <media/mediaplayer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -159,7 +160,9 @@
MyConvertingStreamSource::MyConvertingStreamSource(const char *filename)
: mCurrentBufferIndex(-1),
mCurrentBufferOffset(0) {
- sp<DataSource> dataSource = DataSource::CreateFromURI(filename);
+ sp<DataSource> dataSource =
+ DataSource::CreateFromURI(NULL /* httpService */, filename);
+
CHECK(dataSource != NULL);
sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
index 234aef2..f400732 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
@@ -316,6 +316,7 @@
if (-1 < fileDesc) {
if (FwdLockFile_attach(fileDesc) < 0) {
+ close(fileDesc);
return mimeString;
}
const char* pMimeType = FwdLockFile_GetContentType(fileDesc);
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
index ef392f0..7be449c 100644
--- a/include/media/AudioBufferProvider.h
+++ b/include/media/AudioBufferProvider.h
@@ -61,6 +61,17 @@
// buffer->frameCount 0
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
+ // Release (a portion of) the buffer previously obtained by getNextBuffer().
+ // It is permissible to call releaseBuffer() multiple times per getNextBuffer().
+ // On entry:
+ // buffer->frameCount number of frames to release, must be <= number of frames
+ // obtained but not yet released
+ // buffer->raw unused
+ // On return:
+ // buffer->frameCount 0; implementation MUST set to zero
+ // buffer->raw undefined; implementation is PERMITTED to set to any value,
+ // so if caller needs to continue using this buffer it must
+ // keep track of the pointer itself
virtual void releaseBuffer(Buffer* buffer) = 0;
};
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 05d834d..f3024b7 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -36,7 +36,7 @@
// ----------------------------------------------------------------------------
-class effect_param_cblk_t;
+struct effect_param_cblk_t;
// ----------------------------------------------------------------------------
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 052064d..7054fd4 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -26,7 +26,7 @@
// ----------------------------------------------------------------------------
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
class AudioRecordClientProxy;
// ----------------------------------------------------------------------------
@@ -39,8 +39,12 @@
* Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
*/
enum event_type {
- EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer.
- EVENT_OVERRUN = 1, // PCM buffer overrun occurred.
+ EVENT_MORE_DATA = 0, // Request to read available data from buffer.
+ // If this event is delivered but the callback handler
+ // does not want to read the available data, the handler must
+ // explicitly
+ // ignore the event by setting frameCount to zero.
+ EVENT_OVERRUN = 1, // Buffer overrun occurred.
EVENT_MARKER = 2, // Record head is at the specified marker position
// (See setMarkerPosition()).
EVENT_NEW_POS = 3, // Record head is at a new position
@@ -60,9 +64,10 @@
size_t frameCount; // number of sample frames corresponding to size;
// on input it is the number of frames available,
// on output is the number of frames actually drained
- // (currently ignored, but will make the primary field in future)
+ // (currently ignored but will make the primary field in future)
size_t size; // input/output in bytes == frameCount * frameSize
+ // on output is the number of bytes actually drained
// FIXME this is redundant with respect to frameCount,
// and TRANSFER_OBTAIN mode is broken for 8-bit data
// since we don't define the frame format
@@ -76,7 +81,7 @@
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
- * invokes the callback when a new buffer becomes ready or various conditions occur.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
/* How data is transferred from AudioRecord
*/
enum transfer_type {
- TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters
+ TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous read()
@@ -137,7 +144,7 @@
* be larger if the requested size is not compatible with current audio HAL
* latency. Zero means to use a default value.
* cbf: Callback function. If not null, this function is called periodically
- * to consume new PCM data and inform of marker, position updates, etc.
+ * to consume new data and inform of marker, position updates, etc.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames are ready in record track output buffer.
@@ -155,7 +162,7 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
@@ -171,9 +178,10 @@
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
* - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
- * - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+ * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
+ * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
*
* Parameters not listed in the AudioRecord constructors above:
*
@@ -188,11 +196,11 @@
void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
- /* Result of constructing the AudioRecord. This must be checked
+ /* Result of constructing the AudioRecord. This must be checked for successful initialization
* before using any AudioRecord API (except for set()), because using
* an uninitialized AudioRecord produces undefined results.
* See set() method above for possible return codes.
@@ -221,7 +229,7 @@
status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
- /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still
+ /* Stop a track. The callback will cease being called. Note that obtainBuffer() still
* works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
*/
void stop();
@@ -236,7 +244,7 @@
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
* To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioRecord has been opened with no callback function associated,
* the operation will fail.
*
@@ -378,8 +386,10 @@
* returning the current value by this function call. Such loss typically occurs when the
* user space process is blocked longer than the capacity of audio driver buffers.
* Units: the number of input audio frames.
+ * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+ * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
*/
- unsigned int getInputFramesLost() const;
+ uint32_t getInputFramesLost() const;
private:
/* copying audio record objects is not allowed */
@@ -412,6 +422,7 @@
bool mPaused; // whether thread is requested to pause at next loop entry
bool mPausedInt; // whether thread internally requests pause
nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored
+ bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request
};
// body of AudioRecordThread::threadLoop()
@@ -422,9 +433,10 @@
// NS_INACTIVE inactive so don't run again until re-started
// NS_NEVER never again
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
- nsecs_t processAudioBuffer(const sp<AudioRecordThread>& thread);
+ nsecs_t processAudioBuffer();
// caller must hold lock on mLock for all _l methods
+
status_t openRecord_l(size_t epoch);
// FIXME enum is faster than strcmp() for parameter 'from'
@@ -446,12 +458,13 @@
// notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
- int mObservedSequence; // last observed value of mSequence
+ uint32_t mObservedSequence; // last observed value of mSequence
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
@@ -460,9 +473,13 @@
status_t mStatus;
+ size_t mFrameCount; // corresponds to current IAudioRecord, value is
+ // reported back by AudioFlinger to the client
+ size_t mReqFrameCount; // frame count to request the first or next time
+ // a new IAudioRecord is needed, non-decreasing
+
// constant after constructor or set()
uint32_t mSampleRate;
- size_t mFrameCount;
audio_format_t mFormat;
uint32_t mChannelCount;
size_t mFrameSize; // app-level frame size == AudioFlinger frame size
@@ -473,12 +490,11 @@
int mSessionId;
transfer_type mTransfer;
- audio_io_handle_t mInput; // returned by AudioSystem::getInput()
-
- // may be changed if IAudioRecord object is re-created
+ // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 4c22412..dfc2066 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -67,20 +67,24 @@
// returns true in *state if tracks are active on the specified stream or have been active
// in the past inPastMs milliseconds
- static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
+ static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
// returns true in *state if tracks are active for what qualifies as remote playback
// on the specified stream or have been active in the past inPastMs milliseconds. Remote
// playback isn't mutually exclusive with local playback.
static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
- uint32_t inPastMs = 0);
+ uint32_t inPastMs);
// returns true in *state if a recorder is currently recording with the specified source
static status_t isSourceActive(audio_source_t source, bool *state);
// set/get audio hardware parameters. The function accepts a list of parameters
// key value pairs in the form: key1=value1;key2=value2;...
// Some keys are reserved for standard parameters (See AudioParameter class).
+ // The versions with audio_io_handle_t are intended for internal media framework use only.
static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+ // The versions without audio_io_handle_t are intended for JNI.
+ static status_t setParameters(const String8& keyValuePairs);
+ static String8 getParameters(const String8& keys);
static void setErrorCallback(audio_error_callback cb);
@@ -90,12 +94,14 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
+ // Returned samplingRate and frameCount output values are guaranteed
+ // to be non-zero if status == NO_ERROR
static status_t getOutputSamplingRate(uint32_t* samplingRate,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getOutputFrameCount(size_t* frameCount,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getOutputLatency(uint32_t* latency,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ audio_stream_type_t stream);
static status_t getSamplingRate(audio_io_handle_t output,
audio_stream_type_t streamType,
uint32_t* samplingRate);
@@ -132,7 +138,7 @@
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
// return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
- static size_t getInputFramesLost(audio_io_handle_t ioHandle);
+ static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
static int newAudioSessionId();
static void acquireAudioSessionId(int audioSession);
@@ -155,7 +161,8 @@
class OutputDescriptor {
public:
OutputDescriptor()
- : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {}
+ : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
+ {}
uint32_t samplingRate;
audio_format_t format;
@@ -193,24 +200,32 @@
static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
+ // or release it with releaseOutput().
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
+
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session = 0);
+ int session);
static status_t stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session = 0);
+ int session);
static void releaseOutput(audio_io_handle_t output);
+
+ // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+ // or release it with releaseInput().
static audio_io_handle_t getInput(audio_source_t inputSource,
- uint32_t samplingRate = 0,
- audio_format_t format = AUDIO_FORMAT_DEFAULT,
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
- int sessionId = 0);
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ int sessionId);
+
static status_t startInput(audio_io_handle_t input);
static status_t stopInput(audio_io_handle_t input);
static void releaseInput(audio_io_handle_t input);
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
index c29c7e5..99e9c3e 100644
--- a/include/media/AudioTimestamp.h
+++ b/include/media/AudioTimestamp.h
@@ -19,6 +19,8 @@
#include <time.h>
+namespace android {
+
class AudioTimestamp {
public:
AudioTimestamp() : mPosition(0) {
@@ -30,4 +32,6 @@
struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
};
+} // namespace
+
#endif // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f6646ab..5a50280 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -27,7 +27,7 @@
// ----------------------------------------------------------------------------
-class audio_track_cblk_t;
+struct audio_track_cblk_t;
class AudioTrackClientProxy;
class StaticAudioTrackClientProxy;
@@ -123,6 +123,8 @@
* - NO_ERROR: successful operation
* - NO_INIT: audio server or audio hardware not initialized
* - BAD_VALUE: unsupported configuration
+ * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+ * and is undefined otherwise.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -158,7 +160,7 @@
* sampleRate: Data source sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channelMask: Channel mask.
+ * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
* latency of the track. The actual size selected by the AudioTrack could be
@@ -185,7 +187,7 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -210,7 +212,7 @@
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -248,7 +250,7 @@
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
- int sessionId = 0,
+ int sessionId = AUDIO_SESSION_ALLOCATE,
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1);
@@ -336,7 +338,7 @@
*/
status_t setSampleRate(uint32_t sampleRate);
- /* Return current source sample rate in Hz, or 0 if unknown */
+ /* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
@@ -361,7 +363,7 @@
/* Sets marker position. When playback reaches the number of frames specified, a callback with
* event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
* notification callback. To set a marker at a position which would compute as 0,
- * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+ * a workaround is to set the marker at a nearby position such as ~0 or 1.
* If the AudioTrack has been opened with no callback function associated, the operation will
* fail.
*
@@ -452,7 +454,7 @@
* Returned value:
* handle on audio hardware output
*/
- audio_io_handle_t getOutput();
+ audio_io_handle_t getOutput() const;
/* Returns the unique session ID associated with this track.
*
@@ -566,7 +568,7 @@
uint32_t getUnderrunFrames() const;
/* Get the flags */
- audio_output_flags_t getFlags() const { return mFlags; }
+ audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
/* Set parameters - only possible when using direct output */
status_t setParameters(const String8& keyValuePairs);
@@ -626,53 +628,50 @@
// NS_INACTIVE inactive so don't run again until re-started
// NS_NEVER never again
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
- nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
- status_t processStreamEnd(int32_t waitCount);
+ nsecs_t processAudioBuffer();
+ bool isOffloaded() const;
// caller must hold lock on mLock for all _l methods
- status_t createTrack_l(audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch);
+ status_t createTrack_l(size_t epoch);
// can only be called when mState != STATE_ACTIVE
void flush_l();
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- audio_io_handle_t getOutput_l();
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
- bool isOffloaded() const
+ bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
- // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mOutput; // returned by AudioSystem::getOutput()
sp<AudioTrackThread> mAudioTrackThread;
+
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
- size_t mFrameCount; // corresponds to current IAudioTrack
- size_t mReqFrameCount; // frame count to request the next time a new
- // IAudioTrack is needed
-
+ size_t mFrameCount; // corresponds to current IAudioTrack, value is
+ // reported back by AudioFlinger to the client
+ size_t mReqFrameCount; // frame count to request the first or next time
+ // a new IAudioTrack is needed, non-decreasing
// constant after constructor or set()
audio_format_t mFormat; // as requested by client, not forced to 16-bit
audio_stream_type_t mStreamType;
uint32_t mChannelCount;
audio_channel_mask_t mChannelMask;
+ sp<IMemory> mSharedBuffer;
transfer_type mTransfer;
+ audio_offload_info_t mOffloadInfoCopy;
+ const audio_offload_info_t* mOffloadInfo;
// mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
// twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
@@ -705,21 +704,25 @@
uint32_t mNotificationFramesAct; // actual number of frames between each
// notification callback,
// at initial source sample rate
- bool mRefreshRemaining; // processAudioBuffer() should refresh next 2
+ bool mRefreshRemaining; // processAudioBuffer() should refresh
+ // mRemainingFrames and mRetryOnPartialBuffer
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
uint32_t mObservedSequence; // last observed value of mSequence
- sp<IMemory> mSharedBuffer;
uint32_t mLoopPeriod; // in frames, zero means looping is disabled
+
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
audio_output_flags_t mFlags;
+ // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD.
+ // mLock must be held to read or write those bits reliably.
+
int mSessionId;
int mAuxEffectId;
@@ -753,7 +756,6 @@
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioTrack attempt
- audio_io_handle_t mOutput; // cached output io handle
int mClientUid;
};
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 282f275..ea225ac 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -64,9 +64,12 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t output,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
@@ -78,11 +81,14 @@
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
+ // On successful return, AudioFlinger takes over the handle
+ // reference and will release it when the track is destroyed.
+ // However on failure, the client is responsible for release.
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
@@ -188,6 +194,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
+ // AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
int sessionId,
status_t *status,
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
new file mode 100644
index 0000000..e048b64
--- /dev/null
+++ b/include/media/IMediaHTTPConnection.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_CONNECTION_H_
+
+#define I_MEDIA_HTTP_CONNECTION_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPConnection.aidl */
+
+struct IMediaHTTPConnection : public IInterface {
+ DECLARE_META_INTERFACE(MediaHTTPConnection);
+
+ virtual bool connect(
+ const char *uri, const KeyedVector<String8, String8> *headers) = 0;
+
+ virtual void disconnect() = 0;
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
+ virtual off64_t getSize() = 0;
+ virtual status_t getMIMEType(String8 *mimeType) = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPConnection);
+};
+
+} // namespace android
+
+#endif // I_MEDIA_HTTP_CONNECTION_H_
diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
new file mode 100644
index 0000000..f66d6c8
--- /dev/null
+++ b/include/media/IMediaHTTPService.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef I_MEDIA_HTTP_SERVICE_H_
+
+#define I_MEDIA_HTTP_SERVICE_H_
+
+#include <binder/IInterface.h>
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+/** MUST stay in sync with IMediaHTTPService.aidl */
+
+struct IMediaHTTPService : public IInterface {
+ DECLARE_META_INTERFACE(MediaHTTPService);
+
+ virtual sp<IMediaHTTPConnection> makeHTTPConnection() = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPService);
+};
+
+} // namespace android
+
+#endif // I_MEDIA_HTTP_SERVICE_H_
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
index 6dbb2d7..2529800 100644
--- a/include/media/IMediaMetadataRetriever.h
+++ b/include/media/IMediaMetadataRetriever.h
@@ -26,6 +26,8 @@
namespace android {
+struct IMediaHTTPService;
+
class IMediaMetadataRetriever: public IInterface
{
public:
@@ -33,6 +35,7 @@
virtual void disconnect() = 0;
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *srcUrl,
const KeyedVector<String8, String8> *headers = NULL) = 0;
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index 0cbd269..db62cd5 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -33,6 +33,7 @@
class Surface;
class IStreamSource;
class IGraphicBufferProducer;
+struct IMediaHTTPService;
class IMediaPlayer: public IInterface
{
@@ -41,8 +42,11 @@
virtual void disconnect() = 0;
- virtual status_t setDataSource(const char *url,
- const KeyedVector<String8, String8>* headers) = 0;
+ virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8>* headers) = 0;
+
virtual status_t setDataSource(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setDataSource(const sp<IStreamSource>& source) = 0;
virtual status_t setVideoSurfaceTexture(
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 2998b37..5b45376 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -34,6 +34,7 @@
struct ICrypto;
struct IDrm;
struct IHDCP;
+struct IMediaHTTPService;
class IMediaRecorder;
class IOMX;
class IRemoteDisplay;
@@ -49,9 +50,14 @@
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
- virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+ virtual status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
+
virtual status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
int* pNumChannels, audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
@@ -93,9 +99,6 @@
virtual void addBatteryData(uint32_t params) = 0;
virtual status_t pullBatteryData(Parcel* reply) = 0;
-
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 9c8451c..3db2c38 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -142,6 +142,8 @@
enum InternalOptionType {
INTERNAL_OPTION_SUSPEND, // data is a bool
INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t
+ INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
+ INTERNAL_OPTION_START_TIME, // data is an int64_t
};
virtual status_t setInternalOption(
node_id node,
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
index ecc3b65..bb6b97b 100644
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ b/include/media/MediaMetadataRetrieverInterface.h
@@ -24,6 +24,8 @@
namespace android {
+struct IMediaHTTPService;
+
// Abstract base class
class MediaMetadataRetrieverBase : public RefBase
{
@@ -32,6 +34,7 @@
virtual ~MediaMetadataRetrieverBase() {}
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers = NULL) = 0;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 26d8729..87717da 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -137,6 +137,7 @@
}
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers = NULL) = 0;
@@ -213,11 +214,6 @@
return INVALID_OPERATION;
}
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- return INVALID_OPERATION;
- }
-
private:
friend class MediaPlayerService;
diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
index 0df77c1..b35cf32 100644
--- a/include/media/mediametadataretriever.h
+++ b/include/media/mediametadataretriever.h
@@ -25,6 +25,7 @@
namespace android {
+struct IMediaHTTPService;
class IMediaPlayerService;
class IMediaMetadataRetriever;
@@ -68,6 +69,7 @@
void disconnect();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *dataSourceUrl,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 4c05fc3..f8e4e3b 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -189,6 +189,8 @@
virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0;
};
+struct IMediaHTTPService;
+
class MediaPlayer : public BnMediaPlayerClient,
public virtual IMediaDeathNotifier
{
@@ -199,6 +201,7 @@
void disconnect();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers);
@@ -224,9 +227,14 @@
bool isLooping();
status_t setVolume(float leftVolume, float rightVolume);
void notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
- static status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize);
+ static status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize);
static status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
int* pNumChannels, audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize);
@@ -242,9 +250,6 @@
status_t setRetransmitEndpoint(const char* addrString, uint16_t port);
status_t setNextMediaPlayer(const sp<MediaPlayer>& player);
- status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
private:
void clear_l();
status_t seekTo_l(int msec);
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 88a42a0..142cb90 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -39,7 +39,7 @@
enum video_source {
VIDEO_SOURCE_DEFAULT = 0,
VIDEO_SOURCE_CAMERA = 1,
- VIDEO_SOURCE_GRALLOC_BUFFER = 2,
+ VIDEO_SOURCE_SURFACE = 2,
VIDEO_SOURCE_LIST_END // must be last - used to validate audio source type
};
diff --git a/include/media/mediascanner.h b/include/media/mediascanner.h
index a73403b..4537679 100644
--- a/include/media/mediascanner.h
+++ b/include/media/mediascanner.h
@@ -21,6 +21,7 @@
#include <utils/threads.h>
#include <utils/List.h>
#include <utils/Errors.h>
+#include <utils/String8.h>
#include <pthread.h>
struct dirent;
@@ -29,6 +30,7 @@
class MediaScannerClient;
class StringArray;
+class CharacterEncodingDetector;
enum MediaScanResult {
// This file or directory was scanned successfully.
@@ -94,15 +96,9 @@
virtual status_t setMimeType(const char* mimeType) = 0;
protected:
- void convertValues(uint32_t encoding);
-
-protected:
- // cached name and value strings, for native encoding support.
- StringArray* mNames;
- StringArray* mValues;
-
- // default encoding based on MediaScanner::mLocale string
- uint32_t mLocaleEncoding;
+ // default encoding from MediaScanner::mLocale
+ String8 mLocale;
+ CharacterEncodingDetector *mEncodingDetector;
};
}; // namespace android
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index 2c4aaff..b16e20a 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -27,7 +27,7 @@
class AudioBufferProviderSource : public NBAIO_Source {
public:
- AudioBufferProviderSource(AudioBufferProvider *provider, NBAIO_Format format);
+ AudioBufferProviderSource(AudioBufferProvider *provider, const NBAIO_Format& format);
virtual ~AudioBufferProviderSource();
// NBAIO_Port interface
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index d3802fe..b09b35f 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -41,7 +41,7 @@
// Note: whatever shares this object with another thread needs to do so in an SMP-safe way (like
// creating it the object before creating the other thread, or storing the object with a
// release_store). Otherwise the other thread could see a partially-constructed object.
- MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock = false);
+ MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock = false);
virtual ~MonoPipe();
// NBAIO_Port interface
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 1da0c73..56896b9 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -52,31 +52,41 @@
// the combinations that are actually needed within AudioFlinger. If the list of combinations grows
// too large, then this decision should be re-visited.
// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
-enum {
- Format_Invalid
+struct NBAIO_Format {
+//private:
+ unsigned mPacked;
};
+extern const NBAIO_Format Format_Invalid;
+
// Return the frame size of an NBAIO_Format in bytes
-size_t Format_frameSize(NBAIO_Format format);
+size_t Format_frameSize(const NBAIO_Format& format);
// Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(NBAIO_Format format);
+// or -1 if frame size is not a power of 2
+int Format_frameBitShift(const NBAIO_Format& format);
// Convert a sample rate in Hz and channel count to an NBAIO_Format
+// FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
// Return the sample rate in Hz of an NBAIO_Format
-unsigned Format_sampleRate(NBAIO_Format format);
+unsigned Format_sampleRate(const NBAIO_Format& format);
// Return the channel count of an NBAIO_Format
-unsigned Format_channelCount(NBAIO_Format format);
+unsigned Format_channelCount(const NBAIO_Format& format);
// Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
typedef ssize_t (*readVia_t)(void *user, const void *buffer,
size_t count, int64_t readPTS);
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
// Abstract class (interface) representing a data port.
class NBAIO_Port : public RefBase {
@@ -115,8 +125,8 @@
virtual NBAIO_Format format() const { return mNegotiated ? mFormat : Format_Invalid; }
protected:
- NBAIO_Port(NBAIO_Format format) : mNegotiated(false), mFormat(format),
- mBitShift(Format_frameBitShift(format)) { }
+ NBAIO_Port(const NBAIO_Format& format) : mNegotiated(false), mFormat(format),
+ mBitShift(Format_frameBitShift(format)) { }
virtual ~NBAIO_Port() { }
// Implementations are free to ignore these if they don't need them
@@ -220,7 +230,7 @@
virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
protected:
- NBAIO_Sink(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+ NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
virtual ~NBAIO_Sink() { }
// Implementations are free to ignore these if they don't need them
@@ -311,7 +321,7 @@
virtual void onTimestamp(const AudioTimestamp& timestamp) { }
protected:
- NBAIO_Source(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+ NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
virtual ~NBAIO_Source() { }
// Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
index 79a4eee..c784129 100644
--- a/include/media/nbaio/Pipe.h
+++ b/include/media/nbaio/Pipe.h
@@ -30,7 +30,7 @@
public:
// maxFrames will be rounded up to a power of 2, and all slots are available. Must be >= 2.
- Pipe(size_t maxFrames, NBAIO_Format format);
+ Pipe(size_t maxFrames, const NBAIO_Format& format);
virtual ~Pipe();
// NBAIO_Port interface
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index f1636e6..e284109 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,6 +67,8 @@
void signalRequestIDRFrame();
+ bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
+
struct PortDescription : public RefBase {
size_t countBuffers();
IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -187,6 +189,7 @@
bool mIsEncoder;
bool mUseMetadataOnEncoderOutput;
bool mShutdownInProgress;
+ bool mIsConfiguredForAdaptivePlayback;
// If "mKeepComponentAllocated" we only transition back to Loaded state
// and do not release the component instance.
@@ -202,6 +205,9 @@
int32_t mMetaDataBuffersToSubmit;
int64_t mRepeatFrameDelayUs;
+ int64_t mMaxPtsGapUs;
+
+ bool mCreateInputBuffersSuspended;
status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
status_t allocateBuffersOnPort(OMX_U32 portIndex);
diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index a829916..69cfbd0 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -185,6 +185,8 @@
virtual void dataCallbackTimestamp(int64_t timestampUs, int32_t msgType,
const sp<IMemory> &data);
+ void releaseCamera();
+
private:
friend class CameraSourceListener;
@@ -233,7 +235,6 @@
int32_t frameRate);
void stopCameraRecording();
- void releaseCamera();
status_t reset();
CameraSource(const CameraSource &);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 157b1aa..f8787dd 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -31,6 +31,7 @@
namespace android {
struct AMessage;
+struct IMediaHTTPService;
class String8;
class DataSource : public RefBase {
@@ -43,6 +44,7 @@
};
static sp<DataSource> CreateFromURI(
+ const sp<IMediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h
new file mode 100644
index 0000000..4b18a0b
--- /dev/null
+++ b/include/media/stagefright/MediaCodecSource.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MediaCodecSource_H_
+#define MediaCodecSource_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <media/stagefright/MediaSource.h>
+
+namespace android {
+
+class ALooper;
+class AMessage;
+class IGraphicBufferProducer;
+class MediaCodec;
+class MetaData;
+
+struct MediaCodecSource : public MediaSource,
+ public MediaBufferObserver {
+ enum FlagBits {
+ FLAG_USE_SURFACE_INPUT = 1,
+ FLAG_USE_METADATA_INPUT = 2,
+ };
+
+ static sp<MediaCodecSource> Create(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &format,
+ const sp<MediaSource> &source,
+ uint32_t flags = 0);
+
+ bool isVideo() const { return mIsVideo; }
+ sp<IGraphicBufferProducer> getGraphicBufferProducer();
+
+ // MediaSource
+ virtual status_t start(MetaData *params = NULL);
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual sp<MetaData> getFormat() { return mMeta; }
+ virtual status_t read(
+ MediaBuffer **buffer,
+ const ReadOptions *options = NULL);
+
+ // MediaBufferObserver
+ virtual void signalBufferReturned(MediaBuffer *buffer);
+
+ // for AHandlerReflector
+ void onMessageReceived(const sp<AMessage> &msg);
+
+protected:
+ virtual ~MediaCodecSource();
+
+private:
+ struct Puller;
+
+ enum {
+ kWhatPullerNotify,
+ kWhatEncoderActivity,
+ kWhatStart,
+ kWhatStop,
+ kWhatPause,
+ };
+
+ MediaCodecSource(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &outputFormat,
+ const sp<MediaSource> &source,
+ uint32_t flags = 0);
+
+ status_t onStart(MetaData *params);
+ status_t init();
+ status_t initEncoder();
+ void releaseEncoder();
+ status_t feedEncoderInputBuffers();
+ void scheduleDoMoreWork();
+ status_t doMoreWork();
+ void suspend();
+ void resume(int64_t skipFramesBeforeUs = -1ll);
+ void signalEOS(status_t err = ERROR_END_OF_STREAM);
+ bool reachedEOS();
+ status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+
+ sp<ALooper> mLooper;
+ sp<ALooper> mCodecLooper;
+ sp<AHandlerReflector<MediaCodecSource> > mReflector;
+ sp<AMessage> mOutputFormat;
+ sp<MetaData> mMeta;
+ sp<Puller> mPuller;
+ sp<MediaCodec> mEncoder;
+ uint32_t mFlags;
+ List<uint32_t> mStopReplyIDQueue;
+ bool mIsVideo;
+ bool mStarted;
+ bool mStopping;
+ bool mDoMoreWorkPending;
+ bool mPullerReachedEOS;
+ sp<AMessage> mEncoderActivityNotify;
+ sp<IGraphicBufferProducer> mGraphicBufferProducer;
+ Vector<sp<ABuffer> > mEncoderInputBuffers;
+ Vector<sp<ABuffer> > mEncoderOutputBuffers;
+ List<MediaBuffer *> mInputBufferQueue;
+ List<size_t> mAvailEncoderInputIndices;
+ List<int64_t> mDecodingTimeQueue; // decoding time (us) for video
+
+ // audio drift time
+ int64_t mFirstSampleTimeUs;
+ List<int64_t> mDriftTimeQueue;
+
+ // following variables are protected by mOutputBufferLock
+ Mutex mOutputBufferLock;
+ Condition mOutputBufferCond;
+ List<MediaBuffer*> mOutputBufferQueue;
+ bool mEncodedReachedEOS;
+ status_t mErrorCode;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodecSource);
+};
+
+} // namespace android
+
+#endif /* MediaCodecSource_H_ */
diff --git a/include/media/stagefright/MediaHTTP.h b/include/media/stagefright/MediaHTTP.h
new file mode 100644
index 0000000..006d8d8
--- /dev/null
+++ b/include/media/stagefright/MediaHTTP.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_HTTP_H_
+
+#define MEDIA_HTTP_H_
+
+#include <media/stagefright/foundation/AString.h>
+
+#include "include/HTTPBase.h"
+
+namespace android {
+
+struct IMediaHTTPConnection;
+
+struct MediaHTTP : public HTTPBase {
+ MediaHTTP(const sp<IMediaHTTPConnection> &conn);
+
+ virtual status_t connect(
+ const char *uri,
+ const KeyedVector<String8, String8> *headers,
+ off64_t offset);
+
+ virtual void disconnect();
+
+ virtual status_t initCheck() const;
+
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+ virtual status_t getSize(off64_t *size);
+
+ virtual uint32_t flags();
+
+ virtual status_t reconnectAtOffset(off64_t offset);
+
+protected:
+ virtual ~MediaHTTP();
+
+ virtual sp<DecryptHandle> DrmInitialization(const char* mime);
+ virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
+ virtual String8 getUri();
+ virtual String8 getMIMEType() const;
+
+private:
+ status_t mInitCheck;
+ sp<IMediaHTTPConnection> mHTTPConnection;
+
+ KeyedVector<String8, String8> mLastHeaders;
+ AString mLastURI;
+
+ bool mCachedSizeValid;
+ off64_t mCachedSize;
+
+ sp<DecryptHandle> mDecryptHandle;
+ DrmManagerClient *mDrmManagerClient;
+
+ void clearDRMState_l();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
+};
+
+} // namespace android
+
+#endif // MEDIA_HTTP_H_
diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
index 5ae6f6b..402e7f8 100644
--- a/include/media/stagefright/NuMediaExtractor.h
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -31,6 +31,7 @@
struct ABuffer;
struct AMessage;
struct DataSource;
+struct IMediaHTTPService;
struct MediaBuffer;
struct MediaExtractor;
struct MediaSource;
@@ -45,6 +46,7 @@
NuMediaExtractor();
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *path,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/include/media/stagefright/SkipCutBuffer.h b/include/media/stagefright/SkipCutBuffer.h
index 2653b53..098aa69 100644
--- a/include/media/stagefright/SkipCutBuffer.h
+++ b/include/media/stagefright/SkipCutBuffer.h
@@ -47,6 +47,7 @@
private:
void write(const char *src, size_t num);
size_t read(char *dst, size_t num);
+ int32_t mSkip;
int32_t mFrontPadding;
int32_t mBackPadding;
int32_t mWriteHead;
diff --git a/include/media/stagefright/timedtext/TimedTextDriver.h b/include/media/stagefright/timedtext/TimedTextDriver.h
index f23c337..37ef674 100644
--- a/include/media/stagefright/timedtext/TimedTextDriver.h
+++ b/include/media/stagefright/timedtext/TimedTextDriver.h
@@ -25,6 +25,7 @@
namespace android {
class ALooper;
+struct IMediaHTTPService;
class MediaPlayerBase;
class MediaSource;
class Parcel;
@@ -34,7 +35,9 @@
class TimedTextDriver {
public:
- TimedTextDriver(const wp<MediaPlayerBase> &listener);
+ TimedTextDriver(
+ const wp<MediaPlayerBase> &listener,
+ const sp<IMediaHTTPService> &httpService);
~TimedTextDriver();
@@ -77,6 +80,7 @@
sp<ALooper> mLooper;
sp<TimedTextPlayer> mPlayer;
wp<MediaPlayerBase> mListener;
+ sp<IMediaHTTPService> mHTTPService;
// Variables to be guarded by mLock.
State mState;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 7fd9379..3901e79 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -48,7 +48,7 @@
#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
-#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 // assuming up to a maximum of 20 seconds of offloaded
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
@@ -65,7 +65,9 @@
struct AudioTrackSharedStatic {
StaticAudioTrackSingleStateQueue::Shared
mSingleStateQueue;
- size_t mBufferPosition; // updated asynchronously by server,
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mBufferPosition; // updated asynchronously by server,
// "for entertainment purposes only"
};
@@ -96,11 +98,7 @@
// The value should be used "for entertainment purposes only",
// which means don't make important decisions based on it.
- size_t frameCount_; // used during creation to pass actual track buffer size
- // from AudioFlinger to client, and not referenced again
- // FIXME remove here and replace by createTrack() in/out
- // parameter
- // renamed to "_" to detect incorrect use
+ uint32_t mPad1; // unused
volatile int32_t mFutex; // event flag: down (P) by client,
// up (V) by server or binderDied() or interrupt()
@@ -108,7 +106,9 @@
private:
- size_t mMinimum; // server wakes up client if available >= mMinimum
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mMinimum; // server wakes up client if available >= mMinimum
// Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
// Left channel is in [0:15], right channel is in [16:31].
@@ -245,7 +245,11 @@
}
void setMinimum(size_t minimum) {
- mCblk->mMinimum = minimum;
+ // This can only happen on a 64-bit client
+ if (minimum > UINT32_MAX) {
+ minimum = UINT32_MAX;
+ }
+ mCblk->mMinimum = (uint32_t) minimum;
}
// Return the number of frames that would need to be obtained and released
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
index 46a5946..d483061 100644
--- a/include/private/media/StaticAudioTrackState.h
+++ b/include/private/media/StaticAudioTrackState.h
@@ -25,9 +25,13 @@
// state is wrapped by a SingleStateQueue.
struct StaticAudioTrackState {
// do not define constructors, destructors, or virtual methods
- size_t mLoopStart;
- size_t mLoopEnd;
- int mLoopCount;
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mLoopStart;
+ uint32_t mLoopEnd;
+
+ int mLoopCount;
};
} // namespace android
diff --git a/libvideoeditor/lvpp/PreviewPlayer.cpp b/libvideoeditor/lvpp/PreviewPlayer.cpp
index 2bd9f84..b36fe0a 100755
--- a/libvideoeditor/lvpp/PreviewPlayer.cpp
+++ b/libvideoeditor/lvpp/PreviewPlayer.cpp
@@ -21,6 +21,7 @@
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
@@ -1160,7 +1161,8 @@
sp<DataSource> dataSource;
sp<MediaExtractor> extractor;
- dataSource = DataSource::CreateFromURI(mUri.string(), NULL);
+ dataSource = DataSource::CreateFromURI(
+ NULL /* httpService */, mUri.string(), NULL);
if (dataSource == NULL) {
return UNKNOWN_ERROR;
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 8d656c4..f9c3879 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -57,6 +57,7 @@
status_t VideoEditorPlayer::setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url, const KeyedVector<String8, String8> *headers) {
ALOGI("setDataSource('%s')", url);
if (headers != NULL) {
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index b8c1254..781e4bc 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -98,6 +98,7 @@
virtual status_t initCheck();
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url, const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
diff --git a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
index c3cd3d0..953f35a 100755
--- a/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPreviewController.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include <gui/Surface.h>
+#include <media/IMediaHTTPService.h>
#include "VideoEditorAudioPlayer.h"
#include "PreviewRenderer.h"
@@ -967,7 +968,8 @@
ALOGV("preparePlayer: instance %d file %d", playerInstance, index);
const char* fileName = (const char*) pController->mClipList[index]->pFile;
- pController->mVePlayer[playerInstance]->setDataSource(fileName, NULL);
+ pController->mVePlayer[playerInstance]->setDataSource(
+ NULL /* httpService */, fileName, NULL);
ALOGV("preparePlayer: setDataSource instance %s",
(const char *)pController->mClipList[index]->pFile);
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 56e7787..fc4b2a5 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -25,6 +25,8 @@
AudioRecord.cpp \
AudioSystem.cpp \
mediaplayer.cpp \
+ IMediaHTTPConnection.cpp \
+ IMediaHTTPService.cpp \
IMediaLogService.cpp \
IMediaPlayerService.cpp \
IMediaPlayerClient.cpp \
@@ -44,7 +46,7 @@
IAudioPolicyService.cpp \
MediaScanner.cpp \
MediaScannerClient.cpp \
- autodetect.cpp \
+ CharacterEncodingDetector.cpp \
IMediaDeathNotifier.cpp \
MediaProfiles.cpp \
IEffect.cpp \
@@ -65,7 +67,7 @@
# Consider a separate a library for SingleStateQueueInstantiations.
LOCAL_SHARED_LIBRARIES := \
- libui liblog libcutils libutils libbinder libsonivox libicuuc libexpat \
+ libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \
libcamera_client libstagefright_foundation \
libgui libdl libaudioutils
@@ -77,6 +79,7 @@
$(call include-path-for, graphics corecg) \
$(TOP)/frameworks/native/include/media/openmax \
external/icu4c/common \
+ external/icu4c/i18n \
$(call include-path-for, audio-effects) \
$(call include-path-for, audio-utils)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 8dfffb3..35f6557 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -380,9 +380,9 @@
}
void AudioEffect::commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
+ uint32_t cmdSize __unused,
void *cmdData,
- uint32_t replySize,
+ uint32_t replySize __unused,
void *replyData)
{
if (cmdData == NULL || replyData == NULL) {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 666fafa..541cb51 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -52,7 +52,7 @@
}
if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
+ ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
@@ -71,7 +71,7 @@
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(0),
+ : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
}
@@ -87,8 +87,8 @@
int notificationFrames,
int sessionId,
transfer_type transferType,
- audio_input_flags_t flags)
- : mStatus(NO_INIT), mSessionId(0),
+ audio_input_flags_t flags __unused)
+ : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(NULL)
@@ -191,12 +191,12 @@
// validate parameters
if (!audio_is_valid_format(format)) {
- ALOGE("Invalid format %d", format);
+ ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
// Temporary restriction: AudioFlinger currently supports 16-bit PCM only
if (format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGE("Format %d is not supported", format);
+ ALOGE("Format %#x is not supported", format);
return BAD_VALUE;
}
mFormat = format;
@@ -228,12 +228,13 @@
ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
return BAD_VALUE;
}
- mFrameCount = frameCount;
+ // mFrameCount is initialized in openRecord_l
+ mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
- if (sessionId == 0 ) {
+ if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioSessionId();
} else {
mSessionId = sessionId;
@@ -244,7 +245,7 @@
// create the IAudioRecord
status = openRecord_l(0 /*epoch*/);
- if (status) {
+ if (status != NO_ERROR) {
return status;
}
@@ -255,9 +256,6 @@
mStatus = NO_ERROR;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount_;
-
mActive = false;
mCbf = cbf;
mRefreshRemaining = true;
@@ -289,6 +287,9 @@
// reset current position as seen by client to 0
mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+ // force refresh of remaining frames by processAudioBuffer() as last
+ // read before stop could be partial.
+ mRefreshRemaining = true;
mNewPosition = mProxy->getPosition() + mUpdatePeriod;
int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
@@ -352,6 +353,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
+ // The only purpose of setting marker position is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -377,6 +379,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
+ // The only purpose of setting position update period is to get a callback
if (mCbf == NULL) {
return INVALID_OPERATION;
}
@@ -412,7 +415,7 @@
return NO_ERROR;
}
-unsigned int AudioRecord::getInputFramesLost() const
+uint32_t AudioRecord::getInputFramesLost() const
{
// no need to check mActive, because if inactive this will return 0, which is what we want
return AudioSystem::getInputFramesLost(getInput());
@@ -447,11 +450,12 @@
}
mNotificationFramesAct = mNotificationFramesReq;
+ size_t frameCount = mReqFrameCount;
if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
// Make sure that application is notified with sufficient margin before overrun
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
- mNotificationFramesAct = mFrameCount/2;
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
+ mNotificationFramesAct = frameCount/2;
}
}
@@ -461,24 +465,31 @@
ALOGE("Could not get audio input for record source %d", mInputSource);
return BAD_VALUE;
}
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+ size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
mSampleRate, mFormat,
mChannelMask,
- mFrameCount,
+ &temp,
&trackFlags,
tid,
&mSessionId,
&status);
- ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+ ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
"session ID changed from %d to %d", originalSessionId, mSessionId);
if (record == 0 || status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
- AudioSystem::releaseInput(input);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = record->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -493,11 +504,24 @@
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
+
+ // We retain a copy of the I/O handle, but don't own the reference
mInput = input;
mAudioRecord = record;
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
+ // note that temp is the (possibly revised) value of mFrameCount
+ if (temp < frameCount || (frameCount == 0 && temp == 0)) {
+ ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+ }
+ frameCount = temp;
+ // If IAudioRecord is re-created, don't let the requested frameCount
+ // decrease. This can confuse clients that cache frameCount().
+ if (frameCount > mReqFrameCount) {
+ mReqFrameCount = frameCount;
+ }
+
// FIXME missing fast track frameCount logic
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
@@ -521,6 +545,8 @@
// starting address of buffers in shared memory
void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);
+ mFrameCount = frameCount;
+
// update proxy
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
mProxy->setEpoch(epoch);
@@ -530,6 +556,14 @@
mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseInput(input);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -545,13 +579,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
@@ -591,6 +625,9 @@
if (newSequence == oldSequence) {
status = restoreRecord_l("obtainBuffer");
if (status != NO_ERROR) {
+ buffer.mFrameCount = 0;
+ buffer.mRaw = NULL;
+ buffer.mNonContig = 0;
break;
}
}
@@ -692,7 +729,7 @@
// -------------------------------------------------------------------------
-nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread)
+nsecs_t AudioRecord::processAudioBuffer()
{
mLock.lock();
if (mAwaitBoost) {
@@ -767,10 +804,10 @@
mRetryOnPartialBuffer = false;
}
size_t misalignment = mProxy->getMisalignment();
- int32_t sequence = mSequence;
+ uint32_t sequence = mSequence;
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mSampleRate
+ // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
mLock.unlock();
@@ -844,8 +881,8 @@
"obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u",
- mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+ ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
break;
@@ -954,7 +991,7 @@
// =========================================================================
-void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
{
sp<AudioRecord> audioRecord = mAudioRecord.promote();
if (audioRecord != 0) {
@@ -966,7 +1003,8 @@
// =========================================================================
AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL)
+ : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
+ mIgnoreNextPausedInt(false)
{
}
@@ -983,6 +1021,10 @@
// caller will check for exitPending()
return true;
}
+ if (mIgnoreNextPausedInt) {
+ mIgnoreNextPausedInt = false;
+ mPausedInt = false;
+ }
if (mPausedInt) {
if (mPausedNs > 0) {
(void) mMyCond.waitRelative(mMyLock, mPausedNs);
@@ -993,7 +1035,7 @@
return true;
}
}
- nsecs_t ns = mReceiver.processAudioBuffer(this);
+ nsecs_t ns = mReceiver.processAudioBuffer();
switch (ns) {
case 0:
return true;
@@ -1017,12 +1059,7 @@
{
// must be in this order to avoid a race condition
Thread::requestExit();
- AutoMutex _l(mMyLock);
- if (mPaused || mPausedInt) {
- mPaused = false;
- mPausedInt = false;
- mMyCond.signal();
- }
+ resume();
}
void AudioRecord::AudioRecordThread::pause()
@@ -1034,6 +1071,7 @@
void AudioRecord::AudioRecordThread::resume()
{
AutoMutex _l(mMyLock);
+ mIgnoreNextPausedInt = true;
if (mPaused || mPausedInt) {
mPaused = false;
mPausedInt = false;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index cc5b810..87f4b4c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -40,10 +40,10 @@
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
// Cached values for recording queries, all protected by gLock
-uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
-size_t AudioSystem::gInBuffSize = 0;
+uint32_t AudioSystem::gPrevInSamplingRate;
+audio_format_t AudioSystem::gPrevInFormat;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask;
+size_t AudioSystem::gInBuffSize = 0; // zero indicates cache is invalid
// establish binder interface to AudioFlinger service
@@ -84,13 +84,15 @@
return DEAD_OBJECT;
}
-status_t AudioSystem::muteMicrophone(bool state) {
+status_t AudioSystem::muteMicrophone(bool state)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMicMute(state);
}
-status_t AudioSystem::isMicrophoneMuted(bool* state) {
+status_t AudioSystem::isMicrophoneMuted(bool* state)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*state = af->getMicMute();
@@ -175,13 +177,15 @@
return af->setMode(mode);
}
-status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
+status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setParameters(ioHandle, keyValuePairs);
}
-String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
+String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
String8 result = String8("");
if (af == 0) return result;
@@ -190,6 +194,16 @@
return result;
}
+status_t AudioSystem::setParameters(const String8& keyValuePairs)
+{
+ return setParameters((audio_io_handle_t) 0, keyValuePairs);
+}
+
+String8 AudioSystem::getParameters(const String8& keys)
+{
+ return getParameters((audio_io_handle_t) 0, keys);
+}
+
// convert volume steps to natural log scale
// change this value to change volume scaling
@@ -249,6 +263,11 @@
*samplingRate = outputDesc->samplingRate;
gLock.unlock();
}
+ if (*samplingRate == 0) {
+ ALOGE("AudioSystem::getSamplingRate failed for output %d stream type %d",
+ output, streamType);
+ return BAD_VALUE;
+ }
ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
*samplingRate);
@@ -289,6 +308,11 @@
*frameCount = outputDesc->frameCount;
gLock.unlock();
}
+ if (*frameCount == 0) {
+ ALOGE("AudioSystem::getFrameCount failed for output %d stream type %d",
+ output, streamType);
+ return BAD_VALUE;
+ }
ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output,
*frameCount);
@@ -313,7 +337,7 @@
}
status_t AudioSystem::getLatency(audio_io_handle_t output,
- audio_stream_type_t streamType,
+ audio_stream_type_t streamType __unused,
uint32_t* latency)
{
OutputDescriptor *outputDesc;
@@ -349,6 +373,12 @@
return PERMISSION_DENIED;
}
inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+ if (inBuffSize == 0) {
+ ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
+ sampleRate, format, channelMask);
+ return BAD_VALUE;
+ }
+ // A benign race is possible here: we could overwrite a fresher cache entry
gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
@@ -387,9 +417,10 @@
return af->getRenderPosition(halFrames, dspFrames, output);
}
-size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- unsigned int result = 0;
+ uint32_t result = 0;
if (af == 0) return result;
if (ioHandle == 0) return result;
@@ -397,20 +428,23 @@
return result;
}
-int AudioSystem::newAudioSessionId() {
+int AudioSystem::newAudioSessionId()
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return 0;
return af->newAudioSessionId();
}
-void AudioSystem::acquireAudioSessionId(int audioSession) {
+void AudioSystem::acquireAudioSessionId(int audioSession)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af != 0) {
af->acquireAudioSessionId(audioSession);
}
}
-void AudioSystem::releaseAudioSessionId(int audioSession) {
+void AudioSystem::releaseAudioSessionId(int audioSession)
+{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af != 0) {
af->releaseAudioSessionId(audioSession);
@@ -419,7 +453,8 @@
// ---------------------------------------------------------------------------
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
+{
Mutex::Autolock _l(AudioSystem::gLock);
AudioSystem::gAudioFlinger.clear();
@@ -455,7 +490,7 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %u, format %d channel mask %#x frameCount %u "
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
"latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
outputDesc->frameCount, outputDesc->latency);
@@ -479,7 +514,7 @@
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channel mask %#x "
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
"frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channelMask, desc->frameCount, desc->latency);
@@ -496,12 +531,14 @@
}
}
-void AudioSystem::setErrorCallback(audio_error_callback cb) {
+void AudioSystem::setErrorCallback(audio_error_callback cb)
+{
Mutex::Autolock _l(gLock);
gAudioErrorCallback = cb;
}
-bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType)
+{
switch (streamType) {
case AUDIO_STREAM_MUSIC:
case AUDIO_STREAM_VOICE_CALL:
@@ -709,7 +746,8 @@
audio_io_handle_t AudioSystem::getOutputForEffect(const effect_descriptor_t *desc)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return PERMISSION_DENIED;
+ // FIXME change return type to status_t, and return PERMISSION_DENIED here
+ if (aps == 0) return 0;
return aps->getOutputForEffect(desc);
}
@@ -804,7 +842,8 @@
// ---------------------------------------------------------------------------
-void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
+{
Mutex::Autolock _l(AudioSystem::gLock);
AudioSystem::gAudioPolicyService.clear();
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index a9d6993..63eaf1a 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -44,9 +44,6 @@
return BAD_VALUE;
}
- // default to 0 in case of error
- *frameCount = 0;
-
// FIXME merge with similar code in createTrack_l(), except we're missing
// some information here that is available in createTrack_l():
// audio_io_handle_t output
@@ -54,16 +51,26 @@
// audio_channel_mask_t channelMask
// audio_output_flags_t flags
uint32_t afSampleRate;
- if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
- return NO_INIT;
+ status_t status;
+ status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("Unable to query output sample rate for stream type %d; status %d",
+ streamType, status);
+ return status;
}
size_t afFrameCount;
- if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
- return NO_INIT;
+ status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("Unable to query output frame count for stream type %d; status %d",
+ streamType, status);
+ return status;
}
uint32_t afLatency;
- if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
- return NO_INIT;
+ status = AudioSystem::getOutputLatency(&afLatency, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("Unable to query output latency for stream type %d; status %d",
+ streamType, status);
+ return status;
}
// Ensure that buffer depth covers at least audio hardware latency
@@ -74,6 +81,13 @@
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
+ // The formula above should always produce a non-zero value, but return an error
+ // in the unlikely event that it does not, as that's part of the API contract.
+ if (*frameCount == 0) {
+ ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d",
+ streamType, sampleRate);
+ return BAD_VALUE;
+ }
ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
@@ -208,6 +222,7 @@
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
+ mSharedBuffer = sharedBuffer;
mTransfer = transferType;
// FIXME "int" here is legacy and will be replaced by size_t later
@@ -230,19 +245,24 @@
return INVALID_OPERATION;
}
- mOutput = 0;
-
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("Invalid stream type %d", streamType);
+ return BAD_VALUE;
+ }
+ mStreamType = streamType;
+ status_t status;
if (sampleRate == 0) {
- uint32_t afSampleRate;
- if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
- return NO_INIT;
+ status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+ if (status != NO_ERROR) {
+ ALOGE("Could not get output sample rate for stream type %d; status %d",
+ streamType, status);
+ return status;
}
- sampleRate = afSampleRate;
}
mSampleRate = sampleRate;
@@ -250,13 +270,16 @@
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
- if (channelMask == 0) {
- channelMask = AUDIO_CHANNEL_OUT_STEREO;
- }
// validate parameters
if (!audio_is_valid_format(format)) {
- ALOGE("Invalid format %d", format);
+ ALOGE("Invalid format %#x", format);
+ return BAD_VALUE;
+ }
+ mFormat = format;
+
+ if (!audio_is_output_channel(channelMask)) {
+ ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
@@ -282,10 +305,6 @@
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
- if (!audio_is_output_channel(channelMask)) {
- ALOGE("Invalid channel mask %#x", channelMask);
- return BAD_VALUE;
- }
mChannelMask = channelMask;
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
@@ -298,21 +317,20 @@
mFrameSizeAF = sizeof(uint8_t);
}
- audio_io_handle_t output = AudioSystem::getOutput(
- streamType,
- sampleRate, format, channelMask,
- flags,
- offloadInfo);
-
- if (output == 0) {
- ALOGE("Could not get audio output for stream type %d", streamType);
- return BAD_VALUE;
+ // Make copy of input parameter offloadInfo so that in the future:
+ // (a) createTrack_l doesn't need it as an input parameter
+ // (b) we can support re-creation of offloaded tracks
+ if (offloadInfo != NULL) {
+ mOffloadInfoCopy = *offloadInfo;
+ mOffloadInfo = &mOffloadInfoCopy;
+ } else {
+ mOffloadInfo = NULL;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0.0f;
- mFrameCount = frameCount;
+ // mFrameCount is initialized in createTrack_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
@@ -332,14 +350,7 @@
}
// create the IAudioTrack
- status_t status = createTrack_l(streamType,
- sampleRate,
- format,
- frameCount,
- flags,
- sharedBuffer,
- output,
- 0 /*epoch*/);
+ status = createTrack_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -347,17 +358,20 @@
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
//Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
return status;
}
mStatus = NO_ERROR;
- mStreamType = streamType;
- mFormat = format;
- mSharedBuffer = sharedBuffer;
mState = STATE_STOPPED;
mUserData = user;
mLoopPeriod = 0;
@@ -369,7 +383,6 @@
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
- mOutput = output;
return NO_ERROR;
}
@@ -445,12 +458,11 @@
void AudioTrack::stop()
{
AutoMutex lock(mLock);
- // FIXME pause then stop should not be a nop
- if (mState != STATE_ACTIVE) {
+ if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
return;
}
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
mState = STATE_STOPPING;
} else {
mState = STATE_STOPPED;
@@ -472,7 +484,7 @@
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
- if (!isOffloaded()) {
+ if (!isOffloaded_l()) {
t->pause();
}
} else {
@@ -510,7 +522,7 @@
mRefreshRemaining = true;
mState = STATE_FLUSHED;
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
mProxy->interrupt();
}
mProxy->flush();
@@ -543,7 +555,7 @@
mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
mAudioTrack->signal();
}
return NO_ERROR;
@@ -607,7 +619,7 @@
// sample rate can be updated during playback by the offloaded decoder so we need to
// query the HAL and update if needed.
// FIXME use Proxy return channel to update the rate from server and avoid polling here
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
if (mOutput != 0) {
uint32_t sampleRate = 0;
status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate);
@@ -691,6 +703,7 @@
AutoMutex lock(mLock);
mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
+
return NO_ERROR;
}
@@ -744,7 +757,7 @@
}
AutoMutex lock(mLock);
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
uint32_t dspFrames = 0;
if (mOutput != 0) {
@@ -793,23 +806,12 @@
return NO_ERROR;
}
-audio_io_handle_t AudioTrack::getOutput()
+audio_io_handle_t AudioTrack::getOutput() const
{
AutoMutex lock(mLock);
return mOutput;
}
-// must be called with mLock held
-audio_io_handle_t AudioTrack::getOutput_l()
-{
- if (mOutput) {
- return mOutput;
- } else {
- return AudioSystem::getOutput(mStreamType,
- mSampleRate, mFormat, mChannelMask, mFlags);
- }
-}
-
status_t AudioTrack::attachAuxEffect(int effectId)
{
AutoMutex lock(mLock);
@@ -823,15 +825,7 @@
// -------------------------------------------------------------------------
// must be called with mLock held
-status_t AudioTrack::createTrack_l(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch)
+status_t AudioTrack::createTrack_l(size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -840,50 +834,57 @@
return NO_INIT;
}
+ audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+ mChannelMask, mFlags, mOffloadInfo);
+ if (output == 0) {
+ ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
+ "channel mask %#x, flags %#x",
+ mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+ return BAD_VALUE;
+ }
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+
// Not all of these values are needed under all conditions, but it is easier to get them all
uint32_t afLatency;
- status = AudioSystem::getLatency(output, streamType, &afLatency);
+ status = AudioSystem::getLatency(output, mStreamType, &afLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
- return NO_INIT;
+ goto release;
}
size_t afFrameCount;
- status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount);
if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
uint32_t afSampleRate;
- status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+ status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate);
if (status != NO_ERROR) {
- ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
- if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
+ if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
// either of these use cases:
// use case 1: shared buffer
- (sharedBuffer != 0) ||
+ (mSharedBuffer != 0) ||
// use case 2: callback handler
- (mCbf != NULL))) {
+ (mCbf != NULL)) &&
+ // matching sample rate
+ (mSampleRate == afSampleRate))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
- if ((flags & AUDIO_OUTPUT_FLAG_FAST) && sampleRate != afSampleRate) {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client due to mismatching sample rate (%d vs %d)",
- sampleRate, afSampleRate);
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- }
-
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
// n = 1 fast track with single buffering; nBuffering is ignored
// n = 2 fast track with double buffering
@@ -891,43 +892,45 @@
// n = 3 normal track, with sample rate conversion
// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
// n > 3 very high latency or very small notification interval; nBuffering is ignored
- const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+ const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
mNotificationFramesAct = mNotificationFramesReq;
- if (!audio_is_linear_pcm(format)) {
+ size_t frameCount = mReqFrameCount;
+ if (!audio_is_linear_pcm(mFormat)) {
- if (sharedBuffer != 0) {
+ if (mSharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
- frameCount = sharedBuffer->size();
+ frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = afFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
- } else if (sharedBuffer != 0) {
+ } else if (mSharedBuffer != 0) {
// Ensure that buffer alignment matches channel count
// 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+ size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
- if (((uintptr_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+ if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
- sharedBuffer->pointer(), mChannelCount);
- return BAD_VALUE;
+ mSharedBuffer->pointer(), mChannelCount);
+ status = BAD_VALUE;
+ goto release;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
- frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
+ frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
- } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+ } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
@@ -939,10 +942,10 @@
minBufCount = nBuffering;
}
- size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+ size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
- minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
+ minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
@@ -967,26 +970,28 @@
}
pid_t tid = -1;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
- sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
- sampleRate,
+ size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
+ sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
+ mSampleRate,
// AudioFlinger only sees 16-bit PCM
- format == AUDIO_FORMAT_PCM_8_BIT ?
- AUDIO_FORMAT_PCM_16_BIT : format,
+ mFormat == AUDIO_FORMAT_PCM_8_BIT ?
+ AUDIO_FORMAT_PCM_16_BIT : mFormat,
mChannelMask,
- frameCount,
+ &temp,
&trackFlags,
- sharedBuffer,
+ mSharedBuffer,
output,
tid,
&mSessionId,
@@ -996,13 +1001,21 @@
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
return NO_INIT;
}
+ void *iMemPointer = iMem->pointer();
+ if (iMemPointer == NULL) {
+ ALOGE("Could not get control block pointer");
+ return NO_INIT;
+ }
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
@@ -1010,9 +1023,9 @@
}
mAudioTrack = track;
mCblkMemory = iMem;
- audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer());
+ audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- size_t temp = cblk->frameCount_;
+ // note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
@@ -1021,11 +1034,11 @@
}
frameCount = temp;
mAwaitBoost = false;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering.
@@ -1036,26 +1049,27 @@
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
- if (sharedBuffer == 0) {
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+ if (mSharedBuffer == 0) {
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
}
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- mFlags = flags;
- return NO_INIT;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ // FIXME This is a warning, not an error, so don't return error status
+ //return NO_INIT;
}
}
+ // We retain a copy of the I/O handle, but don't own the reference
+ mOutput = output;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
@@ -1063,15 +1077,15 @@
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
buffers = (char*)cblk + sizeof(audio_track_cblk_t);
} else {
- buffers = sharedBuffer->pointer();
+ buffers = mSharedBuffer->pointer();
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
- mLatency = afLatency + (1000*frameCount) / sampleRate;
+ mLatency = afLatency + (1000*frameCount) / mSampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
@@ -1080,7 +1094,7 @@
}
// update proxy
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
} else {
@@ -1098,6 +1112,14 @@
mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseOutput(output);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -1113,13 +1135,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
@@ -1350,7 +1372,7 @@
// -------------------------------------------------------------------------
-nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
+nsecs_t AudioTrack::processAudioBuffer()
{
// Currently the AudioTrack thread is not created if there are no callbacks.
// Would it ever make sense to run the thread, even without callbacks?
@@ -1388,7 +1410,7 @@
// for offloaded tracks restoreTrack_l() will just update the sequence and clear
// AudioSystem cache. We should not exit here but after calling the callback so
// that the upper layers can recreate the track
- if (!isOffloaded() || (mSequence == mObservedSequence)) {
+ if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
status_t status = restoreTrack_l("processAudioBuffer");
mLock.unlock();
// Run again immediately, but with a new IAudioTrack
@@ -1609,7 +1631,6 @@
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
- size_t writtenFrames = writtenSize / mFrameSize;
// Sanity check on returned size
if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
@@ -1675,22 +1696,19 @@
status_t AudioTrack::restoreTrack_l(const char *from)
{
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
- isOffloaded() ? "Offloaded" : "PCM", from);
+ isOffloaded_l() ? "Offloaded" : "PCM", from);
++mSequence;
status_t result;
// refresh the audio configuration cache in this process to make sure we get new
- // output parameters in getOutput_l() and createTrack_l()
+ // output parameters in createTrack_l()
AudioSystem::clearAudioConfigCache();
- if (isOffloaded()) {
+ if (isOffloaded_l()) {
+ // FIXME re-creation of offloaded tracks is not yet implemented
return DEAD_OBJECT;
}
- // force new output query from audio policy manager;
- mOutput = 0;
- audio_io_handle_t output = getOutput_l();
-
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
@@ -1698,14 +1716,7 @@
// take the frames that will be lost by track recreation into account in saved position
size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
- result = createTrack_l(mStreamType,
- mSampleRate,
- mFormat,
- mReqFrameCount, // so that frame count never goes down
- mFlags,
- mSharedBuffer,
- output,
- position /*epoch*/);
+ result = createTrack_l(position /*epoch*/);
if (result == NO_ERROR) {
// continue playback from last known position, but
@@ -1733,10 +1744,16 @@
}
}
if (result != NO_ERROR) {
+ // Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
//Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
ALOGW("restoreTrack_l() failed status %d", result);
mState = STATE_STOPPED;
}
@@ -1769,14 +1786,21 @@
String8 AudioTrack::getParameters(const String8& keys)
{
- if (mOutput) {
- return AudioSystem::getParameters(mOutput, keys);
+ audio_io_handle_t output = getOutput();
+ if (output != 0) {
+ return AudioSystem::getParameters(output, keys);
} else {
return String8::empty();
}
}
-status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
+bool AudioTrack::isOffloaded() const
+{
+ AutoMutex lock(mLock);
+ return isOffloaded_l();
+}
+
+status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
{
const size_t SIZE = 256;
@@ -1806,7 +1830,7 @@
// =========================================================================
-void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who)
+void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
{
sp<AudioTrack> audioTrack = mAudioTrack.promote();
if (audioTrack != 0) {
@@ -1850,7 +1874,7 @@
return true;
}
}
- nsecs_t ns = mReceiver.processAudioBuffer(this);
+ nsecs_t ns = mReceiver.processAudioBuffer();
switch (ns) {
case 0:
return true;
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index caa7900..fdd1a12 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
namespace android {
audio_track_cblk_t::audio_track_cblk_t()
- : mServer(0), frameCount_(0), mFutex(0), mMinimum(0),
+ : mServer(0), mFutex(0), mMinimum(0),
mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
{
memset(&u, 0, sizeof(u));
@@ -475,9 +475,14 @@
void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
{
+ // This can only happen on a 64-bit client
+ if (loopStart > UINT32_MAX || loopEnd > UINT32_MAX) {
+ // FIXME Should return an error status
+ return;
+ }
StaticAudioTrackState newState;
- newState.mLoopStart = loopStart;
- newState.mLoopEnd = loopEnd;
+ newState.mLoopStart = (uint32_t) loopStart;
+ newState.mLoopEnd = (uint32_t) loopEnd;
newState.mLoopCount = loopCount;
mBufferPosition = loopStart;
(void) mMutator.push(newState);
@@ -487,7 +492,7 @@
{
size_t bufferPosition;
if (mMutator.ack()) {
- bufferPosition = mCblk->u.mStatic.mBufferPosition;
+ bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
if (bufferPosition > mFrameCount) {
bufferPosition = mFrameCount;
}
@@ -622,7 +627,7 @@
if (half == 0) {
half = 1;
}
- size_t minimum = cblk->mMinimum;
+ size_t minimum = (size_t) cblk->mMinimum;
if (minimum == 0) {
minimum = mIsOut ? half : 1;
} else if (minimum > half) {
@@ -760,12 +765,13 @@
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
- mCblk->u.mStatic.mBufferPosition = position;
+ // This may overflow, but client is not supposed to rely on it
+ mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
}
return (ssize_t) position;
}
-status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
+status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
{
if (mIsShutdown) {
buffer->mFrameCount = 0;
@@ -836,7 +842,8 @@
mPosition = newPosition;
cblk->mServer += stepCount;
- cblk->u.mStatic.mBufferPosition = newPosition;
+ // This may overflow, but client is not supposed to rely on it
+ cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
if (setFlags != 0) {
(void) android_atomic_or(setFlags, &cblk->mFlags);
// this would be a good place to wake a futex
@@ -847,7 +854,7 @@
buffer->mNonContig = 0;
}
-void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount)
+void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount __unused)
{
// Unlike AudioTrackServerProxy::tallyUnderrunFrames() used for streaming tracks,
// we don't have a location to count underrun frames. The underrun frame counter
diff --git a/media/libmedia/CharacterEncodingDetector.cpp b/media/libmedia/CharacterEncodingDetector.cpp
new file mode 100644
index 0000000..eb091ac
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetector.cpp
@@ -0,0 +1,364 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "CharacterEncodingDector"
+#include <utils/Log.h>
+
+#include "CharacterEncodingDetector.h"
+#include "CharacterEncodingDetectorTables.h"
+
+#include "utils/Vector.h"
+#include "StringArray.h"
+
+#include "unicode/ucnv.h"
+#include "unicode/ucsdet.h"
+#include "unicode/ustring.h"
+
+namespace android {
+
+CharacterEncodingDetector::CharacterEncodingDetector() {
+
+ UErrorCode status = U_ZERO_ERROR;
+ mUtf8Conv = ucnv_open("UTF-8", &status);
+ if (U_FAILURE(status)) {
+ ALOGE("could not create UConverter for UTF-8");
+ mUtf8Conv = NULL;
+ }
+}
+
+CharacterEncodingDetector::~CharacterEncodingDetector() {
+ ucnv_close(mUtf8Conv);
+}
+
+void CharacterEncodingDetector::addTag(const char *name, const char *value) {
+ mNames.push_back(name);
+ mValues.push_back(value);
+}
+
+size_t CharacterEncodingDetector::size() {
+ return mNames.size();
+}
+
+status_t CharacterEncodingDetector::getTag(int index, const char **name, const char**value) {
+ if (index >= mNames.size()) {
+ return BAD_VALUE;
+ }
+
+ *name = mNames.getEntry(index);
+ *value = mValues.getEntry(index);
+ return OK;
+}
+
+static bool isPrintableAscii(const char *value, size_t len) {
+ for (size_t i = 0; i < len; i++) {
+ if ((value[i] & 0x80) || value[i] < 0x20 || value[i] == 0x7f) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void CharacterEncodingDetector::detectAndConvert() {
+
+ int size = mNames.size();
+ ALOGV("%d tags before conversion", size);
+ for (int i = 0; i < size; i++) {
+ ALOGV("%s: %s", mNames.getEntry(i), mValues.getEntry(i));
+ }
+
+ if (size && mUtf8Conv) {
+
+ UErrorCode status = U_ZERO_ERROR;
+ UCharsetDetector *csd = ucsdet_open(&status);
+ const UCharsetMatch *ucm;
+
+ // try combined detection of artist/album/title etc.
+ char buf[1024];
+ buf[0] = 0;
+ int idx;
+ for (int i = 0; i < size; i++) {
+ const char *name = mNames.getEntry(i);
+ const char *value = mValues.getEntry(i);
+ if (!isPrintableAscii(value, strlen(value)) && (
+ !strcmp(name, "artist") ||
+ !strcmp(name, "albumartist") ||
+ !strcmp(name, "composer") ||
+ !strcmp(name, "genre") ||
+ !strcmp(name, "album") ||
+ !strcmp(name, "title"))) {
+ strlcat(buf, value, sizeof(buf));
+ // separate tags by space so ICU's ngram detector can do its job
+ strlcat(buf, " ", sizeof(buf));
+ }
+ }
+ ucsdet_setText(csd, buf, strlen(buf), &status);
+
+ int32_t matches;
+ const UCharsetMatch** ucma = ucsdet_detectAll(csd, &matches, &status);
+ const char *combinedenc = "???";
+
+ const UCharsetMatch* bestCombinedMatch = getPreferred(buf, strlen(buf), ucma, matches);
+
+ if (bestCombinedMatch != NULL) {
+ combinedenc = ucsdet_getName(bestCombinedMatch, &status);
+ }
+
+ for (int i = 0; i < size; i++) {
+ const char *name = mNames.getEntry(i);
+ uint8_t* src = (uint8_t *)mValues.getEntry(i);
+ int len = strlen((char *)src);
+ uint8_t* dest = src;
+
+ ALOGV("@@@ checking %s", name);
+ const char *s = mValues.getEntry(i);
+ int32_t inputLength = strlen(s);
+ const char *enc;
+
+ if (!strcmp(name, "artist") ||
+ !strcmp(name, "albumartist") ||
+ !strcmp(name, "composer") ||
+ !strcmp(name, "genre") ||
+ !strcmp(name, "album") ||
+ !strcmp(name, "title")) {
+ // use encoding determined from the combination of artist/album/title etc.
+ enc = combinedenc;
+ } else {
+ ucsdet_setText(csd, s, inputLength, &status);
+ ucm = ucsdet_detect(csd, &status);
+ if (!ucm) {
+ mValues.setEntry(i, "???");
+ continue;
+ }
+ enc = ucsdet_getName(ucm, &status);
+ ALOGV("@@@@ recognized charset: %s for %s confidence %d",
+ enc, mNames.getEntry(i), ucsdet_getConfidence(ucm, &status));
+ }
+
+ if (strcmp(enc,"UTF-8") != 0) {
+ // only convert if the source encoding isn't already UTF-8
+ ALOGV("@@@ using converter %s for %s", enc, mNames.getEntry(i));
+ UConverter *conv = ucnv_open(enc, &status);
+ if (U_FAILURE(status)) {
+ ALOGE("could not create UConverter for %s", enc);
+ continue;
+ }
+
+ // convert from native encoding to UTF-8
+ const char* source = mValues.getEntry(i);
+ int targetLength = len * 3 + 1;
+ char* buffer = new char[targetLength];
+ // don't normally check for NULL, but in this case targetLength may be large
+ if (!buffer)
+ break;
+ char* target = buffer;
+
+ ucnv_convertEx(mUtf8Conv, conv, &target, target + targetLength,
+ &source, source + strlen(source),
+ NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
+
+ if (U_FAILURE(status)) {
+ ALOGE("ucnv_convertEx failed: %d", status);
+ mValues.setEntry(i, "???");
+ } else {
+ // zero terminate
+ *target = 0;
+ mValues.setEntry(i, buffer);
+ }
+
+ delete[] buffer;
+
+ ucnv_close(conv);
+ }
+ }
+
+ for (int i = size - 1; i >= 0; --i) {
+ if (strlen(mValues.getEntry(i)) == 0) {
+ ALOGV("erasing %s because entry is empty", mNames.getEntry(i));
+ mNames.erase(i);
+ mValues.erase(i);
+ }
+ }
+
+ ucsdet_close(csd);
+ }
+}
+
+/*
+ * When ICU detects multiple encoding matches, apply additional heuristics to determine
+ * which one is the best match, since ICU can't always be trusted to make the right choice.
+ *
+ * What this method does is:
+ * - decode the input using each of the matches found
+ * - recalculate the starting confidence level for multibyte encodings using a different
+ * algorithm and larger frequent character lists than ICU
+ * - devalue encoding where the conversion contains unlikely characters (symbols, reserved, etc)
+ * - pick the highest match
+ */
+const UCharsetMatch *CharacterEncodingDetector::getPreferred(
+ const char *input, size_t len, const UCharsetMatch** ucma, size_t nummatches) {
+
+ Vector<const UCharsetMatch*> matches;
+ UErrorCode status = U_ZERO_ERROR;
+
+ ALOGV("%d matches", nummatches);
+ for (size_t i = 0; i < nummatches; i++) {
+ const char *encname = ucsdet_getName(ucma[i], &status);
+ int confidence = ucsdet_getConfidence(ucma[i], &status);
+ ALOGV("%d: %s %d", i, encname, confidence);
+ matches.push_back(ucma[i]);
+ }
+
+ size_t num = matches.size();
+ if (num == 0) {
+ return NULL;
+ }
+ if (num == 1) {
+ return matches[0];
+ }
+
+ ALOGV("considering %d matches", num);
+
+ // keep track of how many "special" characters result when converting the input using each
+ // encoding
+ Vector<int> newconfidence;
+ for (size_t i = 0; i < num; i++) {
+ const uint16_t *freqdata = NULL;
+ float freqcoverage = 0;
+ status = U_ZERO_ERROR;
+ const char *encname = ucsdet_getName(matches[i], &status);
+ int confidence = ucsdet_getConfidence(matches[i], &status);
+ if (!strcmp("GB18030", encname)) {
+ freqdata = frequent_zhCN;
+ freqcoverage = frequent_zhCN_coverage;
+ } else if (!strcmp("Big5", encname)) {
+ freqdata = frequent_zhTW;
+ freqcoverage = frequent_zhTW_coverage;
+ } else if (!strcmp("EUC-KR", encname)) {
+ freqdata = frequent_ko;
+ freqcoverage = frequent_ko_coverage;
+ } else if (!strcmp("EUC-JP", encname)) {
+ freqdata = frequent_ja;
+ freqcoverage = frequent_ja_coverage;
+ } else if (!strcmp("Shift_JIS", encname)) {
+ freqdata = frequent_ja;
+ freqcoverage = frequent_ja_coverage;
+ }
+
+ ALOGV("%d: %s %d", i, encname, confidence);
+ UConverter *conv = ucnv_open(encname, &status);
+ const char *source = input;
+ const char *sourceLimit = input + len;
+ status = U_ZERO_ERROR;
+ int demerit = 0;
+ int frequentchars = 0;
+ int totalchars = 0;
+ while (true) {
+ // demerit the current encoding for each "special" character found after conversion.
+ // The amount of demerit is somewhat arbitrarily chosen.
+ int inchar;
+ if (source != sourceLimit) {
+ inchar = (source[0] << 8) + source[1];
+ }
+ UChar32 c = ucnv_getNextUChar(conv, &source, sourceLimit, &status);
+ if (!U_SUCCESS(status)) {
+ break;
+ }
+ if (c < 0x20 || (c >= 0x7f && c <= 0x009f)) {
+ ALOGV("control character %x", c);
+ demerit += 100;
+ } else if ((c >= 0xa0 && c <= 0xbe) // symbols, superscripts
+ || (c == 0xd7) || (c == 0xf7) // multiplication and division signs
+ || (c >= 0x2000 && c <= 0x209f)) { // punctuation, superscripts
+ ALOGV("unlikely character %x", c);
+ demerit += 10;
+ } else if (c >= 0xe000 && c <= 0xf8ff) {
+ ALOGV("private use character %x", c);
+ demerit += 30;
+ } else if (c >= 0x2190 && c <= 0x2bff) {
+ // this range comprises various symbol ranges that are unlikely to appear in
+ // music file metadata.
+ ALOGV("symbol %x", c);
+ demerit += 10;
+ } else if (c == 0xfffd) {
+ ALOGV("replacement character");
+ demerit += 50;
+ } else if (c >= 0xfff0 && c <= 0xfffc) {
+ ALOGV("unicode special %x", c);
+ demerit += 50;
+ } else if (freqdata != NULL) {
+ totalchars++;
+ if (isFrequent(freqdata, c)) {
+ frequentchars++;
+ }
+ }
+ }
+ if (freqdata != NULL && totalchars != 0) {
+ int myconfidence = 10 + float((100 * frequentchars) / totalchars) / freqcoverage;
+ ALOGV("ICU confidence: %d, my confidence: %d (%d %d)", confidence, myconfidence,
+ totalchars, frequentchars);
+ if (myconfidence > 100) myconfidence = 100;
+ if (myconfidence < 0) myconfidence = 0;
+ confidence = myconfidence;
+ }
+ ALOGV("%d-%d=%d", confidence, demerit, confidence - demerit);
+ newconfidence.push_back(confidence - demerit);
+ ucnv_close(conv);
+ if (i == 0 && (confidence - demerit) == 100) {
+ // no need to check any further, we'll end up using this match anyway
+ break;
+ }
+ }
+
+ // find match with highest confidence after adjusting for unlikely characters
+ int highest = newconfidence[0];
+ size_t highestidx = 0;
+ num = newconfidence.size();
+ for (size_t i = 1; i < num; i++) {
+ if (newconfidence[i] > highest) {
+ highest = newconfidence[i];
+ highestidx = i;
+ }
+ }
+ status = U_ZERO_ERROR;
+ ALOGV("selecting '%s' w/ %d confidence", ucsdet_getName(matches[highestidx], &status), highest);
+ return matches[highestidx];
+}
+
+
+bool CharacterEncodingDetector::isFrequent(const uint16_t *values, uint32_t c) {
+
+ int start = 0;
+ int end = 511; // All the tables have 512 entries
+ int mid = (start+end)/2;
+
+ while(start <= end) {
+ if(c == values[mid]) {
+ return true;
+ } else if (c > values[mid]) {
+ start = mid + 1;
+ } else {
+ end = mid - 1;
+ }
+
+ mid = (start + end) / 2;
+ }
+
+ return false;
+}
+
+
+} // namespace android
diff --git a/media/libmedia/CharacterEncodingDetector.h b/media/libmedia/CharacterEncodingDetector.h
new file mode 100644
index 0000000..3655a91
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetector.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef _CHARACTER_ENCODING_DETECTOR_H
+#define _CHARACTER_ENCODING_DETECTOR_H
+
+#include <media/mediascanner.h>
+
+#include "StringArray.h"
+
+#include "unicode/ucnv.h"
+#include "unicode/ucsdet.h"
+#include "unicode/ustring.h"
+
+namespace android {
+
+class CharacterEncodingDetector {
+
+ public:
+ CharacterEncodingDetector();
+ ~CharacterEncodingDetector();
+
+ void addTag(const char *name, const char *value);
+ size_t size();
+
+ void detectAndConvert();
+ status_t getTag(int index, const char **name, const char**value);
+
+ private:
+ const UCharsetMatch *getPreferred(
+ const char *input, size_t len, const UCharsetMatch** ucma, size_t matches);
+
+ bool isFrequent(const uint16_t *values, uint32_t c);
+
+ // cached name and value strings, for native encoding support.
+ // TODO: replace these with byte blob arrays that don't require the data to be
+ // singlenullbyte-terminated
+ StringArray mNames;
+ StringArray mValues;
+
+ UConverter* mUtf8Conv;
+};
+
+
+
+}; // namespace android
+
+#endif
diff --git a/media/libmedia/CharacterEncodingDetectorTables.h b/media/libmedia/CharacterEncodingDetectorTables.h
new file mode 100644
index 0000000..1fe1137
--- /dev/null
+++ b/media/libmedia/CharacterEncodingDetectorTables.h
@@ -0,0 +1,2092 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// The 512 most frequently occuring characters for the zhCN language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_zhCN[] = {
+ 0x4E00, // 一, #2
+ 0x4E07, // 万, #306
+ 0x4E09, // 三, #138
+ 0x4E0A, // 上, #16
+ 0x4E0B, // 下, #25
+ 0x4E0D, // 不, #7
+ 0x4E0E, // 与, #133
+ 0x4E13, // 专, #151
+ 0x4E16, // 世, #346
+ 0x4E1A, // 业, #39
+ 0x4E1C, // 东, #197
+ 0x4E24, // 两, #376
+ 0x4E2A, // 个, #23
+ 0x4E2D, // 中, #4
+ 0x4E3A, // 为, #31
+ 0x4E3B, // 主, #95
+ 0x4E3E, // 举, #418
+ 0x4E48, // 么, #93
+ 0x4E4B, // 之, #131
+ 0x4E50, // 乐, #130
+ 0x4E5F, // 也, #145
+ 0x4E66, // 书, #283
+ 0x4E70, // 买, #483
+ 0x4E86, // 了, #13
+ 0x4E8B, // 事, #168
+ 0x4E8C, // 二, #218
+ 0x4E8E, // 于, #64
+ 0x4E94, // 五, #430
+ 0x4E9A, // 亚, #468
+ 0x4E9B, // 些, #366
+ 0x4EA4, // 交, #243
+ 0x4EA7, // 产, #86
+ 0x4EAB, // 享, #345
+ 0x4EAC, // 京, #206
+ 0x4EBA, // 人, #3
+ 0x4EC0, // 什, #287
+ 0x4ECB, // 介, #478
+ 0x4ECE, // 从, #381
+ 0x4ED6, // 他, #129
+ 0x4EE3, // 代, #241
+ 0x4EE5, // 以, #51
+ 0x4EEC, // 们, #83
+ 0x4EF6, // 件, #141
+ 0x4EF7, // 价, #140
+ 0x4EFB, // 任, #383
+ 0x4F01, // 企, #439
+ 0x4F18, // 优, #374
+ 0x4F1A, // 会, #29
+ 0x4F20, // 传, #222
+ 0x4F46, // 但, #451
+ 0x4F4D, // 位, #208
+ 0x4F53, // 体, #98
+ 0x4F55, // 何, #339
+ 0x4F5C, // 作, #44
+ 0x4F60, // 你, #76
+ 0x4F7F, // 使, #272
+ 0x4F9B, // 供, #375
+ 0x4FDD, // 保, #180
+ 0x4FE1, // 信, #84
+ 0x4FEE, // 修, #437
+ 0x503C, // 值, #450
+ 0x505A, // 做, #368
+ 0x5065, // 健, #484
+ 0x50CF, // 像, #487
+ 0x513F, // 儿, #326
+ 0x5143, // 元, #202
+ 0x5148, // 先, #485
+ 0x5149, // 光, #254
+ 0x514B, // 克, #503
+ 0x514D, // 免, #349
+ 0x5165, // 入, #156
+ 0x5168, // 全, #47
+ 0x516C, // 公, #35
+ 0x5171, // 共, #448
+ 0x5173, // 关, #49
+ 0x5176, // 其, #195
+ 0x5177, // 具, #329
+ 0x5185, // 内, #109
+ 0x518C, // 册, #225
+ 0x519B, // 军, #466
+ 0x51FA, // 出, #53
+ 0x51FB, // 击, #359
+ 0x5206, // 分, #22
+ 0x5217, // 列, #410
+ 0x521B, // 创, #399
+ 0x5229, // 利, #296
+ 0x522B, // 别, #372
+ 0x5230, // 到, #33
+ 0x5236, // 制, #192
+ 0x524D, // 前, #117
+ 0x529B, // 力, #173
+ 0x529E, // 办, #436
+ 0x529F, // 功, #455
+ 0x52A0, // 加, #97
+ 0x52A1, // 务, #100
+ 0x52A8, // 动, #46
+ 0x52A9, // 助, #365
+ 0x5305, // 包, #331
+ 0x5316, // 化, #155
+ 0x5317, // 北, #194
+ 0x533A, // 区, #105
+ 0x533B, // 医, #234
+ 0x5341, // 十, #294
+ 0x534E, // 华, #205
+ 0x5355, // 单, #259
+ 0x5357, // 南, #182
+ 0x535A, // 博, #153
+ 0x5361, // 卡, #332
+ 0x539F, // 原, #271
+ 0x53BB, // 去, #282
+ 0x53C2, // 参, #500
+ 0x53CA, // 及, #255
+ 0x53CB, // 友, #186
+ 0x53CD, // 反, #422
+ 0x53D1, // 发, #15
+ 0x53D7, // 受, #507
+ 0x53D8, // 变, #395
+ 0x53E3, // 口, #293
+ 0x53EA, // 只, #340
+ 0x53EF, // 可, #45
+ 0x53F0, // 台, #267
+ 0x53F7, // 号, #121
+ 0x53F8, // 司, #150
+ 0x5404, // 各, #491
+ 0x5408, // 合, #115
+ 0x540C, // 同, #189
+ 0x540D, // 名, #127
+ 0x540E, // 后, #75
+ 0x5411, // 向, #459
+ 0x5427, // 吧, #353
+ 0x544A, // 告, #318
+ 0x5458, // 员, #232
+ 0x5468, // 周, #347
+ 0x548C, // 和, #43
+ 0x54C1, // 品, #36
+ 0x5546, // 商, #148
+ 0x5668, // 器, #228
+ 0x56DB, // 四, #352
+ 0x56DE, // 回, #38
+ 0x56E0, // 因, #355
+ 0x56E2, // 团, #412
+ 0x56ED, // 园, #470
+ 0x56FD, // 国, #12
+ 0x56FE, // 图, #32
+ 0x5728, // 在, #10
+ 0x5730, // 地, #30
+ 0x573A, // 场, #177
+ 0x575B, // 坛, #364
+ 0x578B, // 型, #274
+ 0x57CE, // 城, #172
+ 0x57FA, // 基, #315
+ 0x58EB, // 士, #434
+ 0x58F0, // 声, #397
+ 0x5904, // 处, #416
+ 0x5907, // 备, #270
+ 0x590D, // 复, #122
+ 0x5916, // 外, #190
+ 0x591A, // 多, #40
+ 0x5927, // 大, #8
+ 0x5929, // 天, #52
+ 0x592A, // 太, #456
+ 0x5934, // 头, #258
+ 0x5973, // 女, #65
+ 0x597D, // 好, #62
+ 0x5982, // 如, #135
+ 0x5A31, // 娱, #452
+ 0x5B50, // 子, #37
+ 0x5B57, // 字, #285
+ 0x5B66, // 学, #19
+ 0x5B89, // 安, #144
+ 0x5B8C, // 完, #469
+ 0x5B9A, // 定, #179
+ 0x5B9D, // 宝, #188
+ 0x5B9E, // 实, #154
+ 0x5BA2, // 客, #174
+ 0x5BB6, // 家, #26
+ 0x5BB9, // 容, #307
+ 0x5BC6, // 密, #471
+ 0x5BF9, // 对, #90
+ 0x5BFC, // 导, #348
+ 0x5C06, // 将, #265
+ 0x5C0F, // 小, #28
+ 0x5C11, // 少, #379
+ 0x5C14, // 尔, #490
+ 0x5C31, // 就, #101
+ 0x5C55, // 展, #291
+ 0x5C71, // 山, #239
+ 0x5DDE, // 州, #227
+ 0x5DE5, // 工, #73
+ 0x5DF1, // 己, #480
+ 0x5DF2, // 已, #310
+ 0x5E02, // 市, #78
+ 0x5E03, // 布, #350
+ 0x5E08, // 师, #277
+ 0x5E16, // 帖, #396
+ 0x5E26, // 带, #449
+ 0x5E2E, // 帮, #461
+ 0x5E38, // 常, #319
+ 0x5E73, // 平, #217
+ 0x5E74, // 年, #20
+ 0x5E76, // 并, #440
+ 0x5E7F, // 广, #166
+ 0x5E93, // 库, #446
+ 0x5E94, // 应, #187
+ 0x5E97, // 店, #320
+ 0x5EA6, // 度, #114
+ 0x5EB7, // 康, #499
+ 0x5EFA, // 建, #211
+ 0x5F00, // 开, #72
+ 0x5F0F, // 式, #207
+ 0x5F15, // 引, #495
+ 0x5F20, // 张, #385
+ 0x5F3A, // 强, #404
+ 0x5F53, // 当, #233
+ 0x5F55, // 录, #146
+ 0x5F62, // 形, #494
+ 0x5F69, // 彩, #356
+ 0x5F71, // 影, #214
+ 0x5F88, // 很, #300
+ 0x5F97, // 得, #193
+ 0x5FAE, // 微, #245
+ 0x5FC3, // 心, #70
+ 0x5FEB, // 快, #324
+ 0x6001, // 态, #508
+ 0x600E, // 怎, #370
+ 0x6027, // 性, #99
+ 0x603B, // 总, #398
+ 0x606F, // 息, #176
+ 0x60A8, // 您, #251
+ 0x60C5, // 情, #87
+ 0x60F3, // 想, #290
+ 0x610F, // 意, #184
+ 0x611F, // 感, #253
+ 0x620F, // 戏, #237
+ 0x6210, // 成, #71
+ 0x6211, // 我, #11
+ 0x6216, // 或, #321
+ 0x6218, // 战, #369
+ 0x6237, // 户, #215
+ 0x623F, // 房, #236
+ 0x6240, // 所, #147
+ 0x624B, // 手, #55
+ 0x624D, // 才, #407
+ 0x6253, // 打, #281
+ 0x6280, // 技, #203
+ 0x6295, // 投, #408
+ 0x62A4, // 护, #502
+ 0x62A5, // 报, #113
+ 0x62DB, // 招, #363
+ 0x6301, // 持, #403
+ 0x6307, // 指, #414
+ 0x636E, // 据, #409
+ 0x6392, // 排, #377
+ 0x63A5, // 接, #266
+ 0x63A8, // 推, #244
+ 0x63D0, // 提, #181
+ 0x641C, // 搜, #301
+ 0x64AD, // 播, #401
+ 0x652F, // 支, #400
+ 0x6536, // 收, #158
+ 0x653E, // 放, #317
+ 0x653F, // 政, #380
+ 0x6548, // 效, #496
+ 0x6559, // 教, #170
+ 0x6570, // 数, #136
+ 0x6587, // 文, #21
+ 0x6599, // 料, #295
+ 0x65AF, // 斯, #473
+ 0x65B0, // 新, #14
+ 0x65B9, // 方, #68
+ 0x65C5, // 旅, #457
+ 0x65E0, // 无, #164
+ 0x65E5, // 日, #50
+ 0x65F6, // 时, #18
+ 0x660E, // 明, #132
+ 0x6613, // 易, #428
+ 0x661F, // 星, #240
+ 0x662F, // 是, #6
+ 0x663E, // 显, #486
+ 0x66F4, // 更, #103
+ 0x6700, // 最, #61
+ 0x6708, // 月, #80
+ 0x6709, // 有, #5
+ 0x670D, // 服, #94
+ 0x671F, // 期, #139
+ 0x672C, // 本, #56
+ 0x672F, // 术, #216
+ 0x673A, // 机, #27
+ 0x6743, // 权, #250
+ 0x6761, // 条, #309
+ 0x6765, // 来, #42
+ 0x677F, // 板, #505
+ 0x6797, // 林, #475
+ 0x679C, // 果, #212
+ 0x67E5, // 查, #165
+ 0x6807, // 标, #269
+ 0x6821, // 校, #462
+ 0x6837, // 样, #314
+ 0x683C, // 格, #238
+ 0x6848, // 案, #378
+ 0x697C, // 楼, #342
+ 0x6A21, // 模, #413
+ 0x6B21, // 次, #263
+ 0x6B22, // 欢, #443
+ 0x6B3E, // 款, #358
+ 0x6B63, // 正, #219
+ 0x6B64, // 此, #362
+ 0x6BD4, // 比, #298
+ 0x6C11, // 民, #279
+ 0x6C14, // 气, #303
+ 0x6C34, // 水, #163
+ 0x6C42, // 求, #373
+ 0x6C5F, // 江, #336
+ 0x6CA1, // 没, #229
+ 0x6CBB, // 治, #425
+ 0x6CD5, // 法, #85
+ 0x6CE8, // 注, #119
+ 0x6D3B, // 活, #231
+ 0x6D41, // 流, #280
+ 0x6D4B, // 测, #460
+ 0x6D77, // 海, #124
+ 0x6D88, // 消, #415
+ 0x6DF1, // 深, #477
+ 0x6E05, // 清, #311
+ 0x6E38, // 游, #81
+ 0x6E90, // 源, #325
+ 0x706B, // 火, #498
+ 0x70B9, // 点, #58
+ 0x70ED, // 热, #183
+ 0x7136, // 然, #308
+ 0x7167, // 照, #431
+ 0x7231, // 爱, #223
+ 0x7247, // 片, #128
+ 0x7248, // 版, #91
+ 0x724C, // 牌, #429
+ 0x7269, // 物, #169
+ 0x7279, // 特, #224
+ 0x738B, // 王, #351
+ 0x73A9, // 玩, #476
+ 0x73B0, // 现, #125
+ 0x7403, // 球, #367
+ 0x7406, // 理, #69
+ 0x751F, // 生, #24
+ 0x7528, // 用, #17
+ 0x7531, // 由, #441
+ 0x7535, // 电, #34
+ 0x7537, // 男, #275
+ 0x754C, // 界, #419
+ 0x75C5, // 病, #371
+ 0x767B, // 登, #204
+ 0x767D, // 白, #338
+ 0x767E, // 百, #157
+ 0x7684, // 的, #1
+ 0x76D8, // 盘, #493
+ 0x76EE, // 目, #261
+ 0x76F4, // 直, #391
+ 0x76F8, // 相, #143
+ 0x7701, // 省, #464
+ 0x770B, // 看, #54
+ 0x771F, // 真, #249
+ 0x7740, // 着, #302
+ 0x77E5, // 知, #142
+ 0x7801, // 码, #257
+ 0x7814, // 研, #387
+ 0x793A, // 示, #334
+ 0x793E, // 社, #343
+ 0x795E, // 神, #330
+ 0x798F, // 福, #509
+ 0x79BB, // 离, #454
+ 0x79CD, // 种, #278
+ 0x79D1, // 科, #126
+ 0x79EF, // 积, #390
+ 0x7A0B, // 程, #209
+ 0x7A76, // 究, #504
+ 0x7A7A, // 空, #312
+ 0x7ACB, // 立, #393
+ 0x7AD9, // 站, #107
+ 0x7AE0, // 章, #304
+ 0x7B2C, // 第, #96
+ 0x7B49, // 等, #210
+ 0x7B54, // 答, #256
+ 0x7B80, // 简, #474
+ 0x7BA1, // 管, #221
+ 0x7C7B, // 类, #246
+ 0x7CBE, // 精, #226
+ 0x7CFB, // 系, #89
+ 0x7D22, // 索, #354
+ 0x7EA2, // 红, #417
+ 0x7EA7, // 级, #178
+ 0x7EBF, // 线, #108
+ 0x7EC4, // 组, #389
+ 0x7EC6, // 细, #442
+ 0x7ECF, // 经, #74
+ 0x7ED3, // 结, #333
+ 0x7ED9, // 给, #384
+ 0x7EDC, // 络, #472
+ 0x7EDF, // 统, #344
+ 0x7F16, // 编, #424
+ 0x7F51, // 网, #9
+ 0x7F6E, // 置, #411
+ 0x7F8E, // 美, #60
+ 0x8001, // 老, #292
+ 0x8003, // 考, #288
+ 0x8005, // 者, #106
+ 0x800C, // 而, #297
+ 0x8054, // 联, #159
+ 0x80B2, // 育, #327
+ 0x80FD, // 能, #59
+ 0x81EA, // 自, #77
+ 0x8272, // 色, #198
+ 0x8282, // 节, #361
+ 0x82B1, // 花, #299
+ 0x82F1, // 英, #316
+ 0x8350, // 荐, #402
+ 0x836F, // 药, #481
+ 0x8425, // 营, #394
+ 0x85CF, // 藏, #337
+ 0x884C, // 行, #41
+ 0x8868, // 表, #104
+ 0x88AB, // 被, #289
+ 0x88C5, // 装, #161
+ 0x897F, // 西, #199
+ 0x8981, // 要, #48
+ 0x89C1, // 见, #360
+ 0x89C2, // 观, #423
+ 0x89C4, // 规, #453
+ 0x89C6, // 视, #120
+ 0x89E3, // 解, #264
+ 0x8A00, // 言, #433
+ 0x8BA1, // 计, #191
+ 0x8BA4, // 认, #482
+ 0x8BA9, // 让, #421
+ 0x8BAE, // 议, #427
+ 0x8BAF, // 讯, #388
+ 0x8BB0, // 记, #273
+ 0x8BBA, // 论, #66
+ 0x8BBE, // 设, #162
+ 0x8BC1, // 证, #201
+ 0x8BC4, // 评, #111
+ 0x8BC6, // 识, #463
+ 0x8BD5, // 试, #323
+ 0x8BDD, // 话, #247
+ 0x8BE2, // 询, #432
+ 0x8BE5, // 该, #447
+ 0x8BE6, // 详, #497
+ 0x8BED, // 语, #268
+ 0x8BF4, // 说, #112
+ 0x8BF7, // 请, #213
+ 0x8BFB, // 读, #341
+ 0x8C03, // 调, #438
+ 0x8D22, // 财, #488
+ 0x8D28, // 质, #386
+ 0x8D2D, // 购, #260
+ 0x8D34, // 贴, #510
+ 0x8D39, // 费, #242
+ 0x8D44, // 资, #116
+ 0x8D77, // 起, #220
+ 0x8D85, // 超, #406
+ 0x8DEF, // 路, #235
+ 0x8EAB, // 身, #262
+ 0x8F66, // 车, #82
+ 0x8F6C, // 转, #322
+ 0x8F7D, // 载, #175
+ 0x8FBE, // 达, #435
+ 0x8FC7, // 过, #118
+ 0x8FD0, // 运, #357
+ 0x8FD1, // 近, #492
+ 0x8FD8, // 还, #171
+ 0x8FD9, // 这, #57
+ 0x8FDB, // 进, #160
+ 0x8FDE, // 连, #489
+ 0x9009, // 选, #328
+ 0x901A, // 通, #137
+ 0x901F, // 速, #458
+ 0x9020, // 造, #511
+ 0x9053, // 道, #79
+ 0x90A3, // 那, #305
+ 0x90E8, // 部, #102
+ 0x90FD, // 都, #167
+ 0x914D, // 配, #479
+ 0x9152, // 酒, #444
+ 0x91CC, // 里, #196
+ 0x91CD, // 重, #230
+ 0x91CF, // 量, #248
+ 0x91D1, // 金, #134
+ 0x9500, // 销, #465
+ 0x957F, // 长, #152
+ 0x95E8, // 门, #185
+ 0x95EE, // 问, #92
+ 0x95F4, // 间, #88
+ 0x95FB, // 闻, #313
+ 0x9605, // 阅, #467
+ 0x9633, // 阳, #420
+ 0x9645, // 际, #501
+ 0x9650, // 限, #286
+ 0x9662, // 院, #276
+ 0x96C6, // 集, #284
+ 0x9700, // 需, #405
+ 0x9762, // 面, #123
+ 0x97F3, // 音, #335
+ 0x9875, // 页, #63
+ 0x9879, // 项, #506
+ 0x9891, // 频, #200
+ 0x9898, // 题, #110
+ 0x98CE, // 风, #252
+ 0x98DF, // 食, #445
+ 0x9996, // 首, #149
+ 0x9999, // 香, #512
+ 0x9A6C, // 马, #392
+ 0x9A8C, // 验, #382
+ 0x9AD8, // 高, #67
+ 0x9F99, // 龙, #426
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_zhCN_coverage=0.718950369339973;
+
+// The 512 most frequently occuring characters for the zhTW language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_zhTW[] = {
+ 0x4E00, // 一, #2
+ 0x4E09, // 三, #131
+ 0x4E0A, // 上, #12
+ 0x4E0B, // 下, #37
+ 0x4E0D, // 不, #6
+ 0x4E16, // 世, #312
+ 0x4E26, // 並, #434
+ 0x4E2D, // 中, #9
+ 0x4E3B, // 主, #97
+ 0x4E4B, // 之, #55
+ 0x4E5F, // 也, #95
+ 0x4E86, // 了, #19
+ 0x4E8B, // 事, #128
+ 0x4E8C, // 二, #187
+ 0x4E94, // 五, #339
+ 0x4E9B, // 些, #435
+ 0x4E9E, // 亞, #432
+ 0x4EA4, // 交, #264
+ 0x4EAB, // 享, #160
+ 0x4EBA, // 人, #3
+ 0x4EC0, // 什, #483
+ 0x4ECA, // 今, #380
+ 0x4ECB, // 介, #468
+ 0x4ED6, // 他, #65
+ 0x4EE3, // 代, #284
+ 0x4EE5, // 以, #26
+ 0x4EF6, // 件, #234
+ 0x4EFB, // 任, #381
+ 0x4EFD, // 份, #447
+ 0x4F46, // 但, #281
+ 0x4F4D, // 位, #202
+ 0x4F4F, // 住, #471
+ 0x4F55, // 何, #334
+ 0x4F5C, // 作, #56
+ 0x4F60, // 你, #64
+ 0x4F7F, // 使, #236
+ 0x4F86, // 來, #38
+ 0x4F9B, // 供, #397
+ 0x4FBF, // 便, #440
+ 0x4FC2, // 係, #506
+ 0x4FDD, // 保, #161
+ 0x4FE1, // 信, #268
+ 0x4FEE, // 修, #473
+ 0x500B, // 個, #27
+ 0x5011, // 們, #109
+ 0x505A, // 做, #383
+ 0x5065, // 健, #415
+ 0x5099, // 備, #461
+ 0x50B3, // 傳, #277
+ 0x50CF, // 像, #403
+ 0x50F9, // 價, #93
+ 0x512A, // 優, #396
+ 0x5143, // 元, #158
+ 0x5148, // 先, #382
+ 0x5149, // 光, #216
+ 0x514D, // 免, #321
+ 0x5152, // 兒, #374
+ 0x5165, // 入, #58
+ 0x5167, // 內, #106
+ 0x5168, // 全, #67
+ 0x5169, // 兩, #322
+ 0x516C, // 公, #53
+ 0x516D, // 六, #493
+ 0x5171, // 共, #456
+ 0x5176, // 其, #148
+ 0x5177, // 具, #328
+ 0x518A, // 冊, #360
+ 0x518D, // 再, #311
+ 0x51FA, // 出, #44
+ 0x5206, // 分, #15
+ 0x5217, // 列, #259
+ 0x5225, // 別, #361
+ 0x5229, // 利, #251
+ 0x5230, // 到, #29
+ 0x5247, // 則, #511
+ 0x524D, // 前, #82
+ 0x5275, // 創, #409
+ 0x529B, // 力, #176
+ 0x529F, // 功, #430
+ 0x52A0, // 加, #87
+ 0x52A9, // 助, #465
+ 0x52D5, // 動, #48
+ 0x52D9, // 務, #102
+ 0x5305, // 包, #248
+ 0x5316, // 化, #223
+ 0x5317, // 北, #145
+ 0x5340, // 區, #60
+ 0x5341, // 十, #242
+ 0x5357, // 南, #261
+ 0x535A, // 博, #484
+ 0x5361, // 卡, #327
+ 0x5370, // 印, #498
+ 0x5373, // 即, #351
+ 0x539F, // 原, #237
+ 0x53BB, // 去, #190
+ 0x53C3, // 參, #444
+ 0x53C8, // 又, #426
+ 0x53CA, // 及, #136
+ 0x53CB, // 友, #142
+ 0x53D6, // 取, #422
+ 0x53D7, // 受, #410
+ 0x53E3, // 口, #357
+ 0x53EA, // 只, #250
+ 0x53EF, // 可, #35
+ 0x53F0, // 台, #34
+ 0x53F8, // 司, #226
+ 0x5403, // 吃, #362
+ 0x5404, // 各, #454
+ 0x5408, // 合, #147
+ 0x540C, // 同, #173
+ 0x540D, // 名, #108
+ 0x544A, // 告, #186
+ 0x548C, // 和, #130
+ 0x54C1, // 品, #23
+ 0x54E1, // 員, #150
+ 0x5546, // 商, #75
+ 0x554F, // 問, #120
+ 0x559C, // 喜, #502
+ 0x55AE, // 單, #210
+ 0x55CE, // 嗎, #443
+ 0x5668, // 器, #305
+ 0x56DB, // 四, #318
+ 0x56DE, // 回, #59
+ 0x56E0, // 因, #253
+ 0x570B, // 國, #21
+ 0x5712, // 園, #345
+ 0x5716, // 圖, #73
+ 0x5718, // 團, #338
+ 0x5728, // 在, #11
+ 0x5730, // 地, #50
+ 0x578B, // 型, #270
+ 0x57CE, // 城, #466
+ 0x57FA, // 基, #349
+ 0x5831, // 報, #127
+ 0x5834, // 場, #165
+ 0x58EB, // 士, #372
+ 0x5916, // 外, #152
+ 0x591A, // 多, #54
+ 0x5927, // 大, #8
+ 0x5929, // 天, #43
+ 0x592A, // 太, #343
+ 0x5947, // 奇, #325
+ 0x5973, // 女, #85
+ 0x5979, // 她, #420
+ 0x597D, // 好, #22
+ 0x5982, // 如, #144
+ 0x5B50, // 子, #46
+ 0x5B57, // 字, #275
+ 0x5B78, // 學, #49
+ 0x5B89, // 安, #239
+ 0x5B8C, // 完, #320
+ 0x5B9A, // 定, #159
+ 0x5BA2, // 客, #188
+ 0x5BB6, // 家, #31
+ 0x5BB9, // 容, #244
+ 0x5BE6, // 實, #198
+ 0x5BF6, // 寶, #367
+ 0x5C07, // 將, #232
+ 0x5C08, // 專, #133
+ 0x5C0B, // 尋, #352
+ 0x5C0D, // 對, #126
+ 0x5C0E, // 導, #418
+ 0x5C0F, // 小, #20
+ 0x5C11, // 少, #368
+ 0x5C31, // 就, #63
+ 0x5C55, // 展, #341
+ 0x5C71, // 山, #273
+ 0x5DE5, // 工, #121
+ 0x5DF1, // 己, #402
+ 0x5DF2, // 已, #299
+ 0x5E02, // 市, #81
+ 0x5E2B, // 師, #262
+ 0x5E36, // 帶, #470
+ 0x5E38, // 常, #303
+ 0x5E73, // 平, #297
+ 0x5E74, // 年, #30
+ 0x5E97, // 店, #171
+ 0x5EA6, // 度, #220
+ 0x5EB7, // 康, #441
+ 0x5EE3, // 廣, #279
+ 0x5EFA, // 建, #254
+ 0x5F0F, // 式, #155
+ 0x5F15, // 引, #346
+ 0x5F35, // 張, #366
+ 0x5F37, // 強, #437
+ 0x5F71, // 影, #94
+ 0x5F88, // 很, #177
+ 0x5F8C, // 後, #66
+ 0x5F97, // 得, #113
+ 0x5F9E, // 從, #436
+ 0x5FC3, // 心, #57
+ 0x5FEB, // 快, #292
+ 0x6027, // 性, #175
+ 0x606F, // 息, #378
+ 0x60A8, // 您, #252
+ 0x60C5, // 情, #123
+ 0x60F3, // 想, #178
+ 0x610F, // 意, #168
+ 0x611B, // 愛, #125
+ 0x611F, // 感, #211
+ 0x61C9, // 應, #164
+ 0x6210, // 成, #86
+ 0x6211, // 我, #7
+ 0x6216, // 或, #199
+ 0x6230, // 戰, #438
+ 0x6232, // 戲, #309
+ 0x6236, // 戶, #497
+ 0x623F, // 房, #274
+ 0x6240, // 所, #79
+ 0x624B, // 手, #68
+ 0x624D, // 才, #400
+ 0x6253, // 打, #278
+ 0x627E, // 找, #449
+ 0x6280, // 技, #332
+ 0x6295, // 投, #425
+ 0x62C9, // 拉, #500
+ 0x62CD, // 拍, #398
+ 0x6307, // 指, #407
+ 0x6392, // 排, #458
+ 0x63A5, // 接, #326
+ 0x63A8, // 推, #153
+ 0x63D0, // 提, #235
+ 0x641C, // 搜, #314
+ 0x6469, // 摩, #472
+ 0x6536, // 收, #249
+ 0x6539, // 改, #508
+ 0x653E, // 放, #331
+ 0x653F, // 政, #295
+ 0x6559, // 教, #184
+ 0x6574, // 整, #394
+ 0x6578, // 數, #134
+ 0x6587, // 文, #16
+ 0x6599, // 料, #167
+ 0x65AF, // 斯, #476
+ 0x65B0, // 新, #10
+ 0x65B9, // 方, #96
+ 0x65BC, // 於, #70
+ 0x65C5, // 旅, #289
+ 0x65E5, // 日, #18
+ 0x660E, // 明, #118
+ 0x6613, // 易, #482
+ 0x661F, // 星, #205
+ 0x662F, // 是, #5
+ 0x6642, // 時, #13
+ 0x66F4, // 更, #149
+ 0x66F8, // 書, #209
+ 0x6700, // 最, #51
+ 0x6703, // 會, #14
+ 0x6708, // 月, #25
+ 0x6709, // 有, #4
+ 0x670D, // 服, #99
+ 0x671F, // 期, #139
+ 0x672A, // 未, #404
+ 0x672C, // 本, #45
+ 0x6771, // 東, #221
+ 0x677F, // 板, #364
+ 0x6797, // 林, #330
+ 0x679C, // 果, #179
+ 0x67E5, // 查, #283
+ 0x683C, // 格, #157
+ 0x6848, // 案, #392
+ 0x689D, // 條, #406
+ 0x696D, // 業, #103
+ 0x6A02, // 樂, #116
+ 0x6A13, // 樓, #411
+ 0x6A19, // 標, #384
+ 0x6A23, // 樣, #306
+ 0x6A5F, // 機, #40
+ 0x6AA2, // 檢, #359
+ 0x6B0A, // 權, #228
+ 0x6B21, // 次, #227
+ 0x6B3E, // 款, #276
+ 0x6B4C, // 歌, #496
+ 0x6B61, // 歡, #427
+ 0x6B63, // 正, #206
+ 0x6B64, // 此, #247
+ 0x6BCF, // 每, #391
+ 0x6BD4, // 比, #257
+ 0x6C11, // 民, #230
+ 0x6C23, // 氣, #200
+ 0x6C34, // 水, #140
+ 0x6C42, // 求, #501
+ 0x6C92, // 沒, #162
+ 0x6CD5, // 法, #89
+ 0x6D3B, // 活, #124
+ 0x6D41, // 流, #315
+ 0x6D77, // 海, #258
+ 0x6D88, // 消, #342
+ 0x6E05, // 清, #329
+ 0x6E2F, // 港, #293
+ 0x6F14, // 演, #491
+ 0x7063, // 灣, #195
+ 0x70BA, // 為, #39
+ 0x7121, // 無, #107
+ 0x7136, // 然, #215
+ 0x7167, // 照, #376
+ 0x71B1, // 熱, #245
+ 0x7247, // 片, #90
+ 0x7248, // 版, #112
+ 0x724C, // 牌, #467
+ 0x7269, // 物, #110
+ 0x7279, // 特, #183
+ 0x738B, // 王, #287
+ 0x73A9, // 玩, #354
+ 0x73FE, // 現, #143
+ 0x7403, // 球, #350
+ 0x7406, // 理, #105
+ 0x751F, // 生, #24
+ 0x7522, // 產, #201
+ 0x7528, // 用, #17
+ 0x7531, // 由, #288
+ 0x7537, // 男, #298
+ 0x754C, // 界, #399
+ 0x7559, // 留, #218
+ 0x756B, // 畫, #412
+ 0x7576, // 當, #185
+ 0x767B, // 登, #138
+ 0x767C, // 發, #28
+ 0x767D, // 白, #377
+ 0x767E, // 百, #393
+ 0x7684, // 的, #1
+ 0x76EE, // 目, #271
+ 0x76F4, // 直, #379
+ 0x76F8, // 相, #98
+ 0x770B, // 看, #52
+ 0x771F, // 真, #180
+ 0x773C, // 眼, #433
+ 0x77E5, // 知, #170
+ 0x78BC, // 碼, #481
+ 0x793A, // 示, #353
+ 0x793E, // 社, #333
+ 0x795E, // 神, #304
+ 0x7968, // 票, #477
+ 0x798F, // 福, #494
+ 0x79C1, // 私, #507
+ 0x79D1, // 科, #280
+ 0x7A0B, // 程, #272
+ 0x7A2E, // 種, #337
+ 0x7A4D, // 積, #385
+ 0x7A7A, // 空, #324
+ 0x7ACB, // 立, #286
+ 0x7AD9, // 站, #117
+ 0x7AE0, // 章, #141
+ 0x7B2C, // 第, #135
+ 0x7B49, // 等, #240
+ 0x7BA1, // 管, #340
+ 0x7BC0, // 節, #431
+ 0x7BC7, // 篇, #479
+ 0x7C21, // 簡, #499
+ 0x7CBE, // 精, #213
+ 0x7CFB, // 系, #212
+ 0x7D04, // 約, #462
+ 0x7D05, // 紅, #452
+ 0x7D1A, // 級, #267
+ 0x7D30, // 細, #486
+ 0x7D44, // 組, #335
+ 0x7D50, // 結, #243
+ 0x7D66, // 給, #355
+ 0x7D71, // 統, #375
+ 0x7D93, // 經, #111
+ 0x7DB2, // 網, #32
+ 0x7DDA, // 線, #151
+ 0x7E23, // 縣, #439
+ 0x7E3D, // 總, #370
+ 0x7F8E, // 美, #41
+ 0x7FA9, // 義, #504
+ 0x8001, // 老, #290
+ 0x8003, // 考, #428
+ 0x8005, // 者, #92
+ 0x800C, // 而, #217
+ 0x805E, // 聞, #181
+ 0x806F, // 聯, #310
+ 0x8072, // 聲, #413
+ 0x80A1, // 股, #390
+ 0x80B2, // 育, #453
+ 0x80FD, // 能, #71
+ 0x8166, // 腦, #408
+ 0x81EA, // 自, #61
+ 0x81F3, // 至, #344
+ 0x8207, // 與, #84
+ 0x8209, // 舉, #463
+ 0x8272, // 色, #192
+ 0x82B1, // 花, #255
+ 0x82F1, // 英, #348
+ 0x83EF, // 華, #196
+ 0x842C, // 萬, #316
+ 0x843D, // 落, #308
+ 0x8457, // 著, #233
+ 0x85A6, // 薦, #401
+ 0x85CF, // 藏, #503
+ 0x85DD, // 藝, #488
+ 0x8655, // 處, #419
+ 0x865F, // 號, #191
+ 0x884C, // 行, #47
+ 0x8853, // 術, #395
+ 0x8868, // 表, #77
+ 0x88AB, // 被, #291
+ 0x88DD, // 裝, #256
+ 0x88E1, // 裡, #369
+ 0x88FD, // 製, #510
+ 0x897F, // 西, #300
+ 0x8981, // 要, #36
+ 0x898B, // 見, #307
+ 0x8996, // 視, #204
+ 0x89BA, // 覺, #450
+ 0x89BD, // 覽, #387
+ 0x89C0, // 觀, #365
+ 0x89E3, // 解, #323
+ 0x8A00, // 言, #169
+ 0x8A02, // 訂, #423
+ 0x8A08, // 計, #225
+ 0x8A0A, // 訊, #156
+ 0x8A0E, // 討, #373
+ 0x8A18, // 記, #222
+ 0x8A2D, // 設, #174
+ 0x8A3B, // 註, #356
+ 0x8A55, // 評, #246
+ 0x8A66, // 試, #448
+ 0x8A71, // 話, #229
+ 0x8A72, // 該, #446
+ 0x8A8D, // 認, #464
+ 0x8A9E, // 語, #371
+ 0x8AAA, // 說, #91
+ 0x8ABF, // 調, #509
+ 0x8ACB, // 請, #119
+ 0x8AD6, // 論, #114
+ 0x8B1D, // 謝, #389
+ 0x8B49, // 證, #429
+ 0x8B58, // 識, #416
+ 0x8B70, // 議, #485
+ 0x8B77, // 護, #475
+ 0x8B80, // 讀, #386
+ 0x8B8A, // 變, #388
+ 0x8B93, // 讓, #336
+ 0x8CA8, // 貨, #313
+ 0x8CB7, // 買, #260
+ 0x8CBB, // 費, #203
+ 0x8CC7, // 資, #62
+ 0x8CE3, // 賣, #294
+ 0x8CEA, // 質, #457
+ 0x8CFC, // 購, #189
+ 0x8D77, // 起, #214
+ 0x8D85, // 超, #296
+ 0x8DDF, // 跟, #489
+ 0x8DEF, // 路, #137
+ 0x8EAB, // 身, #197
+ 0x8ECA, // 車, #76
+ 0x8F09, // 載, #301
+ 0x8F49, // 轉, #282
+ 0x8FD1, // 近, #414
+ 0x9001, // 送, #363
+ 0x9019, // 這, #42
+ 0x901A, // 通, #207
+ 0x901F, // 速, #495
+ 0x9020, // 造, #455
+ 0x9023, // 連, #285
+ 0x9032, // 進, #231
+ 0x904A, // 遊, #132
+ 0x904B, // 運, #219
+ 0x904E, // 過, #101
+ 0x9053, // 道, #146
+ 0x9054, // 達, #417
+ 0x9078, // 選, #182
+ 0x9084, // 還, #154
+ 0x908A, // 邊, #487
+ 0x90A3, // 那, #269
+ 0x90E8, // 部, #78
+ 0x90FD, // 都, #104
+ 0x914D, // 配, #421
+ 0x9152, // 酒, #512
+ 0x91AB, // 醫, #358
+ 0x91CD, // 重, #224
+ 0x91CF, // 量, #319
+ 0x91D1, // 金, #115
+ 0x9304, // 錄, #302
+ 0x9577, // 長, #172
+ 0x9580, // 門, #193
+ 0x958B, // 開, #72
+ 0x9593, // 間, #80
+ 0x95B1, // 閱, #405
+ 0x95DC, // 關, #74
+ 0x963F, // 阿, #460
+ 0x9650, // 限, #265
+ 0x9662, // 院, #474
+ 0x9664, // 除, #478
+ 0x969B, // 際, #459
+ 0x96C6, // 集, #347
+ 0x96E2, // 離, #442
+ 0x96FB, // 電, #33
+ 0x9700, // 需, #445
+ 0x975E, // 非, #451
+ 0x9762, // 面, #129
+ 0x97F3, // 音, #194
+ 0x9801, // 頁, #83
+ 0x982D, // 頭, #238
+ 0x984C, // 題, #122
+ 0x985E, // 類, #163
+ 0x98A8, // 風, #266
+ 0x98DF, // 食, #208
+ 0x9910, // 餐, #469
+ 0x9928, // 館, #424
+ 0x9996, // 首, #166
+ 0x9999, // 香, #263
+ 0x99AC, // 馬, #317
+ 0x9A57, // 驗, #492
+ 0x9AD4, // 體, #100
+ 0x9AD8, // 高, #88
+ 0x9EBC, // 麼, #241
+ 0x9EC3, // 黃, #480
+ 0x9ED1, // 黑, #490
+ 0x9EDE, // 點, #69
+ 0x9F8D, // 龍, #505
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_zhTW_coverage=0.704841200026877;
+
+// The 512 most frequently occuring characters for the ja language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_ja[] = {
+ 0x3005, // 々, #352
+ 0x3041, // ぁ, #486
+ 0x3042, // あ, #50
+ 0x3044, // い, #2
+ 0x3046, // う, #33
+ 0x3048, // え, #83
+ 0x304A, // お, #37
+ 0x304B, // か, #21
+ 0x304C, // が, #17
+ 0x304D, // き, #51
+ 0x304E, // ぎ, #324
+ 0x304F, // く, #38
+ 0x3050, // ぐ, #334
+ 0x3051, // け, #60
+ 0x3052, // げ, #296
+ 0x3053, // こ, #34
+ 0x3054, // ご, #100
+ 0x3055, // さ, #31
+ 0x3056, // ざ, #378
+ 0x3057, // し, #4
+ 0x3058, // じ, #121
+ 0x3059, // す, #12
+ 0x305A, // ず, #215
+ 0x305B, // せ, #86
+ 0x305D, // そ, #68
+ 0x305F, // た, #11
+ 0x3060, // だ, #42
+ 0x3061, // ち, #67
+ 0x3063, // っ, #23
+ 0x3064, // つ, #73
+ 0x3066, // て, #7
+ 0x3067, // で, #6
+ 0x3068, // と, #14
+ 0x3069, // ど, #75
+ 0x306A, // な, #8
+ 0x306B, // に, #5
+ 0x306D, // ね, #123
+ 0x306E, // の, #1
+ 0x306F, // は, #16
+ 0x3070, // ば, #150
+ 0x3071, // ぱ, #259
+ 0x3072, // ひ, #364
+ 0x3073, // び, #266
+ 0x3075, // ふ, #484
+ 0x3076, // ぶ, #330
+ 0x3078, // へ, #146
+ 0x3079, // べ, #207
+ 0x307B, // ほ, #254
+ 0x307E, // ま, #18
+ 0x307F, // み, #74
+ 0x3080, // む, #285
+ 0x3081, // め, #78
+ 0x3082, // も, #32
+ 0x3083, // ゃ, #111
+ 0x3084, // や, #85
+ 0x3086, // ゆ, #392
+ 0x3087, // ょ, #224
+ 0x3088, // よ, #63
+ 0x3089, // ら, #29
+ 0x308A, // り, #28
+ 0x308B, // る, #9
+ 0x308C, // れ, #35
+ 0x308D, // ろ, #127
+ 0x308F, // わ, #88
+ 0x3092, // を, #19
+ 0x3093, // ん, #22
+ 0x30A1, // ァ, #193
+ 0x30A2, // ア, #27
+ 0x30A3, // ィ, #70
+ 0x30A4, // イ, #15
+ 0x30A6, // ウ, #89
+ 0x30A7, // ェ, #134
+ 0x30A8, // エ, #81
+ 0x30A9, // ォ, #225
+ 0x30AA, // オ, #76
+ 0x30AB, // カ, #52
+ 0x30AC, // ガ, #147
+ 0x30AD, // キ, #66
+ 0x30AE, // ギ, #246
+ 0x30AF, // ク, #25
+ 0x30B0, // グ, #39
+ 0x30B1, // ケ, #137
+ 0x30B2, // ゲ, #200
+ 0x30B3, // コ, #46
+ 0x30B4, // ゴ, #183
+ 0x30B5, // サ, #64
+ 0x30B6, // ザ, #221
+ 0x30B7, // シ, #48
+ 0x30B8, // ジ, #55
+ 0x30B9, // ス, #13
+ 0x30BA, // ズ, #103
+ 0x30BB, // セ, #109
+ 0x30BC, // ゼ, #499
+ 0x30BD, // ソ, #175
+ 0x30BF, // タ, #45
+ 0x30C0, // ダ, #104
+ 0x30C1, // チ, #71
+ 0x30C3, // ッ, #20
+ 0x30C4, // ツ, #119
+ 0x30C6, // テ, #59
+ 0x30C7, // デ, #82
+ 0x30C8, // ト, #10
+ 0x30C9, // ド, #44
+ 0x30CA, // ナ, #102
+ 0x30CB, // ニ, #72
+ 0x30CD, // ネ, #117
+ 0x30CE, // ノ, #192
+ 0x30CF, // ハ, #164
+ 0x30D0, // バ, #62
+ 0x30D1, // パ, #90
+ 0x30D2, // ヒ, #398
+ 0x30D3, // ビ, #77
+ 0x30D4, // ピ, #135
+ 0x30D5, // フ, #47
+ 0x30D6, // ブ, #56
+ 0x30D7, // プ, #43
+ 0x30D8, // ヘ, #268
+ 0x30D9, // ベ, #157
+ 0x30DA, // ペ, #125
+ 0x30DB, // ホ, #155
+ 0x30DC, // ボ, #168
+ 0x30DD, // ポ, #114
+ 0x30DE, // マ, #57
+ 0x30DF, // ミ, #97
+ 0x30E0, // ム, #69
+ 0x30E1, // メ, #53
+ 0x30E2, // モ, #142
+ 0x30E3, // ャ, #93
+ 0x30E4, // ヤ, #258
+ 0x30E5, // ュ, #79
+ 0x30E6, // ユ, #405
+ 0x30E7, // ョ, #98
+ 0x30E9, // ラ, #26
+ 0x30EA, // リ, #30
+ 0x30EB, // ル, #24
+ 0x30EC, // レ, #41
+ 0x30ED, // ロ, #40
+ 0x30EF, // ワ, #144
+ 0x30F3, // ン, #3
+ 0x30F4, // ヴ, #483
+ 0x30FD, // ヽ, #501
+ 0x4E00, // 一, #84
+ 0x4E07, // 万, #337
+ 0x4E09, // 三, #323
+ 0x4E0A, // 上, #133
+ 0x4E0B, // 下, #180
+ 0x4E0D, // 不, #277
+ 0x4E16, // 世, #385
+ 0x4E2D, // 中, #87
+ 0x4E3B, // 主, #432
+ 0x4E88, // 予, #326
+ 0x4E8B, // 事, #95
+ 0x4E8C, // 二, #394
+ 0x4E95, // 井, #468
+ 0x4EA4, // 交, #410
+ 0x4EAC, // 京, #260
+ 0x4EBA, // 人, #61
+ 0x4ECA, // 今, #184
+ 0x4ECB, // 介, #358
+ 0x4ED5, // 仕, #391
+ 0x4ED6, // 他, #256
+ 0x4ED8, // 付, #243
+ 0x4EE3, // 代, #280
+ 0x4EE5, // 以, #216
+ 0x4EF6, // 件, #190
+ 0x4F1A, // 会, #105
+ 0x4F4D, // 位, #177
+ 0x4F4F, // 住, #376
+ 0x4F53, // 体, #223
+ 0x4F55, // 何, #294
+ 0x4F5C, // 作, #154
+ 0x4F7F, // 使, #233
+ 0x4F9B, // 供, #503
+ 0x4FA1, // 価, #217
+ 0x4FBF, // 便, #511
+ 0x4FDD, // 保, #279
+ 0x4FE1, // 信, #271
+ 0x500B, // 個, #415
+ 0x50CF, // 像, #178
+ 0x512A, // 優, #403
+ 0x5143, // 元, #384
+ 0x5148, // 先, #311
+ 0x5149, // 光, #488
+ 0x5165, // 入, #115
+ 0x5168, // 全, #173
+ 0x516C, // 公, #287
+ 0x5177, // 具, #447
+ 0x5185, // 内, #169
+ 0x5186, // 円, #131
+ 0x5199, // 写, #275
+ 0x51FA, // 出, #110
+ 0x5206, // 分, #130
+ 0x5207, // 切, #401
+ 0x521D, // 初, #319
+ 0x5225, // 別, #290
+ 0x5229, // 利, #226
+ 0x5236, // 制, #507
+ 0x524D, // 前, #124
+ 0x529B, // 力, #272
+ 0x52A0, // 加, #249
+ 0x52D5, // 動, #120
+ 0x52D9, // 務, #421
+ 0x52DF, // 募, #476
+ 0x5316, // 化, #308
+ 0x5317, // 北, #341
+ 0x533A, // 区, #348
+ 0x539F, // 原, #321
+ 0x53C2, // 参, #452
+ 0x53CB, // 友, #451
+ 0x53D6, // 取, #237
+ 0x53D7, // 受, #354
+ 0x53E3, // 口, #289
+ 0x53E4, // 古, #339
+ 0x53EF, // 可, #298
+ 0x53F0, // 台, #439
+ 0x53F7, // 号, #361
+ 0x5408, // 合, #118
+ 0x540C, // 同, #263
+ 0x540D, // 名, #65
+ 0x5411, // 向, #434
+ 0x544A, // 告, #386
+ 0x5468, // 周, #393
+ 0x5473, // 味, #299
+ 0x548C, // 和, #350
+ 0x54C1, // 品, #96
+ 0x54E1, // 員, #293
+ 0x5546, // 商, #198
+ 0x554F, // 問, #158
+ 0x55B6, // 営, #438
+ 0x5668, // 器, #366
+ 0x56DE, // 回, #143
+ 0x56F3, // 図, #444
+ 0x56FD, // 国, #153
+ 0x5712, // 園, #435
+ 0x571F, // 土, #239
+ 0x5728, // 在, #351
+ 0x5730, // 地, #163
+ 0x578B, // 型, #430
+ 0x5831, // 報, #112
+ 0x5834, // 場, #139
+ 0x58F2, // 売, #232
+ 0x5909, // 変, #306
+ 0x5916, // 外, #222
+ 0x591A, // 多, #336
+ 0x5927, // 大, #80
+ 0x5929, // 天, #278
+ 0x5973, // 女, #161
+ 0x597D, // 好, #349
+ 0x5A5A, // 婚, #479
+ 0x5B50, // 子, #113
+ 0x5B57, // 字, #492
+ 0x5B66, // 学, #132
+ 0x5B89, // 安, #295
+ 0x5B9A, // 定, #145
+ 0x5B9F, // 実, #220
+ 0x5BA4, // 室, #482
+ 0x5BAE, // 宮, #487
+ 0x5BB6, // 家, #211
+ 0x5BB9, // 容, #333
+ 0x5BFE, // 対, #252
+ 0x5C02, // 専, #474
+ 0x5C0F, // 小, #212
+ 0x5C11, // 少, #377
+ 0x5C4B, // 屋, #284
+ 0x5C71, // 山, #206
+ 0x5CA1, // 岡, #429
+ 0x5CF6, // 島, #297
+ 0x5DDD, // 川, #253
+ 0x5DE5, // 工, #374
+ 0x5E02, // 市, #159
+ 0x5E2F, // 帯, #416
+ 0x5E38, // 常, #437
+ 0x5E73, // 平, #390
+ 0x5E74, // 年, #54
+ 0x5E83, // 広, #367
+ 0x5E97, // 店, #149
+ 0x5EA6, // 度, #269
+ 0x5EAB, // 庫, #380
+ 0x5F0F, // 式, #265
+ 0x5F15, // 引, #345
+ 0x5F37, // 強, #446
+ 0x5F53, // 当, #240
+ 0x5F62, // 形, #502
+ 0x5F8C, // 後, #230
+ 0x5F97, // 得, #490
+ 0x5FC3, // 心, #307
+ 0x5FC5, // 必, #422
+ 0x5FDC, // 応, #356
+ 0x601D, // 思, #189
+ 0x6027, // 性, #201
+ 0x6075, // 恵, #400
+ 0x60C5, // 情, #140
+ 0x60F3, // 想, #477
+ 0x610F, // 意, #305
+ 0x611B, // 愛, #273
+ 0x611F, // 感, #257
+ 0x6210, // 成, #262
+ 0x6226, // 戦, #365
+ 0x6240, // 所, #236
+ 0x624B, // 手, #160
+ 0x6295, // 投, #129
+ 0x6301, // 持, #355
+ 0x6307, // 指, #425
+ 0x63A2, // 探, #369
+ 0x63B2, // 掲, #399
+ 0x643A, // 携, #459
+ 0x652F, // 支, #512
+ 0x653E, // 放, #469
+ 0x6559, // 教, #270
+ 0x6570, // 数, #181
+ 0x6587, // 文, #202
+ 0x6599, // 料, #106
+ 0x65B0, // 新, #99
+ 0x65B9, // 方, #126
+ 0x65C5, // 旅, #445
+ 0x65E5, // 日, #36
+ 0x660E, // 明, #300
+ 0x6620, // 映, #418
+ 0x6642, // 時, #107
+ 0x66F4, // 更, #359
+ 0x66F8, // 書, #174
+ 0x6700, // 最, #152
+ 0x6708, // 月, #49
+ 0x6709, // 有, #302
+ 0x671F, // 期, #332
+ 0x6728, // 木, #203
+ 0x672C, // 本, #92
+ 0x6750, // 材, #489
+ 0x6751, // 村, #466
+ 0x6765, // 来, #267
+ 0x6771, // 東, #191
+ 0x677F, // 板, #411
+ 0x679C, // 果, #441
+ 0x6821, // 校, #327
+ 0x682A, // 株, #412
+ 0x683C, // 格, #228
+ 0x691C, // 検, #179
+ 0x696D, // 業, #166
+ 0x697D, // 楽, #172
+ 0x69D8, // 様, #255
+ 0x6A5F, // 機, #235
+ 0x6B21, // 次, #318
+ 0x6B62, // 止, #475
+ 0x6B63, // 正, #312
+ 0x6C17, // 気, #116
+ 0x6C34, // 水, #165
+ 0x6C42, // 求, #465
+ 0x6C7A, // 決, #370
+ 0x6CBB, // 治, #505
+ 0x6CC1, // 況, #462
+ 0x6CD5, // 法, #227
+ 0x6CE8, // 注, #372
+ 0x6D3B, // 活, #303
+ 0x6D41, // 流, #480
+ 0x6D77, // 海, #274
+ 0x6E08, // 済, #417
+ 0x6F14, // 演, #504
+ 0x706B, // 火, #264
+ 0x70B9, // 点, #331
+ 0x7121, // 無, #58
+ 0x7248, // 版, #409
+ 0x7269, // 物, #170
+ 0x7279, // 特, #242
+ 0x72B6, // 状, #458
+ 0x73FE, // 現, #322
+ 0x7406, // 理, #162
+ 0x751F, // 生, #122
+ 0x7523, // 産, #320
+ 0x7528, // 用, #94
+ 0x7530, // 田, #195
+ 0x7537, // 男, #373
+ 0x753A, // 町, #314
+ 0x753B, // 画, #91
+ 0x754C, // 界, #436
+ 0x756A, // 番, #261
+ 0x75C5, // 病, #428
+ 0x767A, // 発, #194
+ 0x767B, // 登, #231
+ 0x767D, // 白, #419
+ 0x7684, // 的, #251
+ 0x76EE, // 目, #197
+ 0x76F4, // 直, #497
+ 0x76F8, // 相, #286
+ 0x770C, // 県, #199
+ 0x771F, // 真, #219
+ 0x7740, // 着, #283
+ 0x77E5, // 知, #185
+ 0x77F3, // 石, #500
+ 0x78BA, // 確, #383
+ 0x793A, // 示, #241
+ 0x793E, // 社, #167
+ 0x795E, // 神, #315
+ 0x798F, // 福, #423
+ 0x79C1, // 私, #347
+ 0x79D1, // 科, #420
+ 0x7A0E, // 税, #368
+ 0x7A2E, // 種, #455
+ 0x7A3F, // 稿, #148
+ 0x7A7A, // 空, #427
+ 0x7ACB, // 立, #309
+ 0x7B11, // 笑, #454
+ 0x7B2C, // 第, #317
+ 0x7B49, // 等, #457
+ 0x7B54, // 答, #426
+ 0x7BA1, // 管, #481
+ 0x7CFB, // 系, #408
+ 0x7D04, // 約, #276
+ 0x7D20, // 素, #407
+ 0x7D22, // 索, #214
+ 0x7D30, // 細, #381
+ 0x7D39, // 紹, #471
+ 0x7D42, // 終, #456
+ 0x7D44, // 組, #424
+ 0x7D4C, // 経, #360
+ 0x7D50, // 結, #291
+ 0x7D9A, // 続, #357
+ 0x7DCF, // 総, #467
+ 0x7DDA, // 線, #338
+ 0x7DE8, // 編, #453
+ 0x7F8E, // 美, #204
+ 0x8003, // 考, #387
+ 0x8005, // 者, #151
+ 0x805E, // 聞, #463
+ 0x8077, // 職, #363
+ 0x80B2, // 育, #433
+ 0x80FD, // 能, #250
+ 0x8179, // 腹, #396
+ 0x81EA, // 自, #156
+ 0x826F, // 良, #329
+ 0x8272, // 色, #402
+ 0x82B1, // 花, #440
+ 0x82B8, // 芸, #413
+ 0x82F1, // 英, #485
+ 0x8449, // 葉, #472
+ 0x884C, // 行, #128
+ 0x8853, // 術, #460
+ 0x8868, // 表, #209
+ 0x88FD, // 製, #431
+ 0x897F, // 西, #406
+ 0x8981, // 要, #313
+ 0x898B, // 見, #101
+ 0x898F, // 規, #375
+ 0x89A7, // 覧, #171
+ 0x89E3, // 解, #388
+ 0x8A00, // 言, #210
+ 0x8A08, // 計, #343
+ 0x8A18, // 記, #136
+ 0x8A2D, // 設, #292
+ 0x8A71, // 話, #213
+ 0x8A73, // 詳, #371
+ 0x8A8D, // 認, #404
+ 0x8A9E, // 語, #234
+ 0x8AAC, // 説, #494
+ 0x8AAD, // 読, #301
+ 0x8ABF, // 調, #443
+ 0x8AC7, // 談, #448
+ 0x8B77, // 護, #509
+ 0x8C37, // 谷, #506
+ 0x8CA9, // 販, #362
+ 0x8CB7, // 買, #346
+ 0x8CC7, // 資, #473
+ 0x8CEA, // 質, #281
+ 0x8CFC, // 購, #495
+ 0x8EAB, // 身, #470
+ 0x8ECA, // 車, #205
+ 0x8EE2, // 転, #335
+ 0x8F09, // 載, #342
+ 0x8FBC, // 込, #229
+ 0x8FD1, // 近, #304
+ 0x8FD4, // 返, #461
+ 0x8FFD, // 追, #379
+ 0x9001, // 送, #186
+ 0x901A, // 通, #182
+ 0x901F, // 速, #340
+ 0x9023, // 連, #244
+ 0x904B, // 運, #382
+ 0x904E, // 過, #498
+ 0x9053, // 道, #282
+ 0x9054, // 達, #450
+ 0x9055, // 違, #414
+ 0x9078, // 選, #288
+ 0x90E8, // 部, #208
+ 0x90FD, // 都, #344
+ 0x914D, // 配, #389
+ 0x91CD, // 重, #478
+ 0x91CE, // 野, #245
+ 0x91D1, // 金, #138
+ 0x9332, // 録, #238
+ 0x9577, // 長, #247
+ 0x9580, // 門, #508
+ 0x958B, // 開, #248
+ 0x9593, // 間, #141
+ 0x95A2, // 関, #188
+ 0x962A, // 阪, #496
+ 0x9650, // 限, #395
+ 0x9662, // 院, #449
+ 0x9664, // 除, #510
+ 0x969B, // 際, #493
+ 0x96C6, // 集, #196
+ 0x96D1, // 雑, #442
+ 0x96FB, // 電, #187
+ 0x9762, // 面, #328
+ 0x97F3, // 音, #325
+ 0x984C, // 題, #310
+ 0x985E, // 類, #491
+ 0x98A8, // 風, #353
+ 0x98DF, // 食, #218
+ 0x9928, // 館, #464
+ 0x99C5, // 駅, #316
+ 0x9A13, // 験, #397
+ 0x9AD8, // 高, #176
+ 0xFF57, // ｗ, #108
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_ja_coverage=0.880569589120162;
+
+// The 512 most frequently occuring characters for the ko language in a sample of the Internet.
+// Ordered by codepoint, comment shows character and ranking by frequency
+const uint16_t frequent_ko[] = {
+ 0x314B, // ㅋ, #148
+ 0x314E, // ㅎ, #390
+ 0x3160, // ㅠ, #354
+ 0x318D, // ㆍ, #439
+ 0xAC00, // 가, #6
+ 0xAC01, // 각, #231
+ 0xAC04, // 간, #106
+ 0xAC08, // 갈, #362
+ 0xAC10, // 감, #122
+ 0xAC11, // 갑, #493
+ 0xAC15, // 강, #155
+ 0xAC19, // 같, #264
+ 0xAC1C, // 개, #87
+ 0xAC1D, // 객, #198
+ 0xAC24, // 갤, #457
+ 0xAC70, // 거, #91
+ 0xAC74, // 건, #161
+ 0xAC78, // 걸, #338
+ 0xAC80, // 검, #184
+ 0xAC83, // 것, #116
+ 0xAC8C, // 게, #36
+ 0xACA0, // 겠, #233
+ 0xACA8, // 겨, #341
+ 0xACA9, // 격, #245
+ 0xACAC, // 견, #413
+ 0xACB0, // 결, #202
+ 0xACBD, // 경, #62
+ 0xACC4, // 계, #142
+ 0xACE0, // 고, #12
+ 0xACE1, // 곡, #444
+ 0xACE8, // 골, #379
+ 0xACF3, // 곳, #388
+ 0xACF5, // 공, #59
+ 0xACFC, // 과, #69
+ 0xAD00, // 관, #95
+ 0xAD11, // 광, #235
+ 0xAD50, // 교, #128
+ 0xAD6C, // 구, #52
+ 0xAD6D, // 국, #85
+ 0xAD70, // 군, #293
+ 0xAD74, // 굴, #487
+ 0xAD81, // 궁, #441
+ 0xAD8C, // 권, #192
+ 0xADC0, // 귀, #386
+ 0xADDC, // 규, #367
+ 0xADF8, // 그, #30
+ 0xADF9, // 극, #424
+ 0xADFC, // 근, #241
+ 0xAE00, // 글, #61
+ 0xAE08, // 금, #138
+ 0xAE09, // 급, #269
+ 0xAE30, // 기, #3
+ 0xAE34, // 긴, #465
+ 0xAE38, // 길, #297
+ 0xAE40, // 김, #205
+ 0xAE4C, // 까, #171
+ 0xAED8, // 께, #273
+ 0xAF43, // 꽃, #475
+ 0xB05D, // 끝, #505
+ 0xB07C, // 끼, #490
+ 0xB098, // 나, #39
+ 0xB09C, // 난, #274
+ 0xB0A0, // 날, #292
+ 0xB0A8, // 남, #139
+ 0xB0B4, // 내, #56
+ 0xB108, // 너, #272
+ 0xB110, // 널, #476
+ 0xB118, // 넘, #492
+ 0xB124, // 네, #100
+ 0xB137, // 넷, #329
+ 0xB140, // 녀, #288
+ 0xB144, // 년, #151
+ 0xB178, // 노, #149
+ 0xB17C, // 논, #491
+ 0xB180, // 놀, #464
+ 0xB18D, // 농, #442
+ 0xB204, // 누, #319
+ 0xB208, // 눈, #383
+ 0xB274, // 뉴, #173
+ 0xB290, // 느, #368
+ 0xB294, // 는, #5
+ 0xB298, // 늘, #322
+ 0xB2A5, // 능, #190
+ 0xB2C8, // 니, #16
+ 0xB2D8, // 님, #153
+ 0xB2E4, // 다, #2
+ 0xB2E8, // 단, #134
+ 0xB2EB, // 닫, #195
+ 0xB2EC, // 달, #243
+ 0xB2F4, // 담, #254
+ 0xB2F5, // 답, #287
+ 0xB2F9, // 당, #159
+ 0xB300, // 대, #33
+ 0xB313, // 댓, #303
+ 0xB354, // 더, #140
+ 0xB358, // 던, #252
+ 0xB367, // 덧, #463
+ 0xB370, // 데, #104
+ 0xB378, // 델, #429
+ 0xB3C4, // 도, #25
+ 0xB3C5, // 독, #301
+ 0xB3CC, // 돌, #309
+ 0xB3D9, // 동, #58
+ 0xB418, // 되, #82
+ 0xB41C, // 된, #189
+ 0xB420, // 될, #408
+ 0xB429, // 됩, #332
+ 0xB450, // 두, #199
+ 0xB4A4, // 뒤, #496
+ 0xB4DC, // 드, #40
+ 0xB4E0, // 든, #283
+ 0xB4E4, // 들, #54
+ 0xB4EF, // 듯, #478
+ 0xB4F1, // 등, #90
+ 0xB514, // 디, #133
+ 0xB529, // 딩, #462
+ 0xB530, // 따, #333
+ 0xB54C, // 때, #240
+ 0xB610, // 또, #313
+ 0xB77C, // 라, #42
+ 0xB77D, // 락, #355
+ 0xB780, // 란, #290
+ 0xB78C, // 람, #246
+ 0xB78D, // 랍, #420
+ 0xB791, // 랑, #270
+ 0xB798, // 래, #174
+ 0xB799, // 랙, #381
+ 0xB79C, // 랜, #357
+ 0xB7A8, // 램, #359
+ 0xB7A9, // 랩, #402
+ 0xB7C9, // 량, #346
+ 0xB7EC, // 러, #130
+ 0xB7F0, // 런, #312
+ 0xB7FC, // 럼, #327
+ 0xB7FD, // 럽, #447
+ 0xB807, // 렇, #412
+ 0xB808, // 레, #114
+ 0xB80C, // 렌, #395
+ 0xB824, // 려, #158
+ 0xB825, // 력, #194
+ 0xB828, // 련, #326
+ 0xB839, // 령, #389
+ 0xB85C, // 로, #4
+ 0xB85D, // 록, #84
+ 0xB860, // 론, #366
+ 0xB8CC, // 료, #154
+ 0xB8E8, // 루, #236
+ 0xB958, // 류, #265
+ 0xB974, // 르, #212
+ 0xB978, // 른, #250
+ 0xB97C, // 를, #35
+ 0xB984, // 름, #276
+ 0xB9AC, // 리, #19
+ 0xB9AD, // 릭, #394
+ 0xB9B0, // 린, #259
+ 0xB9B4, // 릴, #485
+ 0xB9BC, // 림, #305
+ 0xB9BD, // 립, #217
+ 0xB9C1, // 링, #351
+ 0xB9C8, // 마, #67
+ 0xB9C9, // 막, #310
+ 0xB9CC, // 만, #65
+ 0xB9CE, // 많, #257
+ 0xB9D0, // 말, #188
+ 0xB9DB, // 맛, #397
+ 0xB9DD, // 망, #370
+ 0xB9DE, // 맞, #399
+ 0xB9E4, // 매, #125
+ 0xB9E8, // 맨, #422
+ 0xBA38, // 머, #311
+ 0xBA39, // 먹, #377
+ 0xBA3C, // 먼, #469
+ 0xBA54, // 메, #147
+ 0xBA70, // 며, #191
+ 0xBA74, // 면, #72
+ 0xBA85, // 명, #131
+ 0xBAA8, // 모, #73
+ 0xBAA9, // 목, #157
+ 0xBAB0, // 몰, #401
+ 0xBAB8, // 몸, #437
+ 0xBABB, // 못, #336
+ 0xBB34, // 무, #80
+ 0xBB38, // 문, #57
+ 0xBB3C, // 물, #94
+ 0xBBA4, // 뮤, #431
+ 0xBBF8, // 미, #76
+ 0xBBFC, // 민, #200
+ 0xBC00, // 밀, #308
+ 0xBC0F, // 및, #249
+ 0xBC14, // 바, #89
+ 0xBC15, // 박, #226
+ 0xBC18, // 반, #175
+ 0xBC1B, // 받, #248
+ 0xBC1C, // 발, #164
+ 0xBC29, // 방, #92
+ 0xBC30, // 배, #162
+ 0xBC31, // 백, #256
+ 0xBC84, // 버, #111
+ 0xBC88, // 번, #167
+ 0xBC8C, // 벌, #423
+ 0xBC94, // 범, #427
+ 0xBC95, // 법, #207
+ 0xBCA0, // 베, #281
+ 0xBCA4, // 벤, #378
+ 0xBCA8, // 벨, #387
+ 0xBCC0, // 변, #253
+ 0xBCC4, // 별, #262
+ 0xBCD1, // 병, #340
+ 0xBCF4, // 보, #20
+ 0xBCF5, // 복, #204
+ 0xBCF8, // 본, #182
+ 0xBCFC, // 볼, #385
+ 0xBD09, // 봉, #405
+ 0xBD80, // 부, #46
+ 0xBD81, // 북, #261
+ 0xBD84, // 분, #105
+ 0xBD88, // 불, #225
+ 0xBDF0, // 뷰, #350
+ 0xBE0C, // 브, #214
+ 0xBE14, // 블, #99
+ 0xBE44, // 비, #55
+ 0xBE4C, // 빌, #510
+ 0xBE60, // 빠, #398
+ 0xC0AC, // 사, #14
+ 0xC0AD, // 삭, #342
+ 0xC0B0, // 산, #121
+ 0xC0B4, // 살, #279
+ 0xC0BC, // 삼, #348
+ 0xC0C1, // 상, #41
+ 0xC0C8, // 새, #282
+ 0xC0C9, // 색, #181
+ 0xC0DD, // 생, #109
+ 0xC11C, // 서, #21
+ 0xC11D, // 석, #234
+ 0xC120, // 선, #107
+ 0xC124, // 설, #170
+ 0xC131, // 성, #50
+ 0xC138, // 세, #60
+ 0xC139, // 섹, #456
+ 0xC13C, // 센, #267
+ 0xC154, // 셔, #455
+ 0xC158, // 션, #237
+ 0xC15C, // 셜, #448
+ 0xC168, // 셨, #421
+ 0xC18C, // 소, #51
+ 0xC18D, // 속, #219
+ 0xC190, // 손, #323
+ 0xC1A1, // 송, #203
+ 0xC1C4, // 쇄, #501
+ 0xC1FC, // 쇼, #364
+ 0xC218, // 수, #27
+ 0xC219, // 숙, #467
+ 0xC21C, // 순, #258
+ 0xC220, // 술, #302
+ 0xC26C, // 쉬, #511
+ 0xC288, // 슈, #384
+ 0xC2A4, // 스, #11
+ 0xC2AC, // 슬, #438
+ 0xC2B4, // 슴, #504
+ 0xC2B5, // 습, #77
+ 0xC2B9, // 승, #299
+ 0xC2DC, // 시, #13
+ 0xC2DD, // 식, #137
+ 0xC2E0, // 신, #47
+ 0xC2E4, // 실, #132
+ 0xC2EC, // 심, #196
+ 0xC2ED, // 십, #482
+ 0xC2F6, // 싶, #352
+ 0xC2F8, // 싸, #419
+ 0xC4F0, // 쓰, #278
+ 0xC528, // 씨, #360
+ 0xC544, // 아, #23
+ 0xC545, // 악, #296
+ 0xC548, // 안, #71
+ 0xC54A, // 않, #209
+ 0xC54C, // 알, #222
+ 0xC554, // 암, #460
+ 0xC558, // 았, #349
+ 0xC559, // 앙, #473
+ 0xC55E, // 앞, #434
+ 0xC560, // 애, #271
+ 0xC561, // 액, #415
+ 0xC571, // 앱, #477
+ 0xC57C, // 야, #124
+ 0xC57D, // 약, #229
+ 0xC591, // 양, #177
+ 0xC5B4, // 어, #24
+ 0xC5B5, // 억, #407
+ 0xC5B8, // 언, #294
+ 0xC5BC, // 얼, #356
+ 0xC5C4, // 엄, #426
+ 0xC5C5, // 업, #118
+ 0xC5C6, // 없, #178
+ 0xC5C8, // 었, #165
+ 0xC5D0, // 에, #9
+ 0xC5D4, // 엔, #375
+ 0xC5D8, // 엘, #506
+ 0xC5EC, // 여, #66
+ 0xC5ED, // 역, #186
+ 0xC5EE, // 엮, #488
+ 0xC5F0, // 연, #96
+ 0xC5F4, // 열, #266
+ 0xC5FC, // 염, #449
+ 0xC600, // 였, #374
+ 0xC601, // 영, #83
+ 0xC608, // 예, #168
+ 0xC624, // 오, #75
+ 0xC628, // 온, #300
+ 0xC62C, // 올, #306
+ 0xC640, // 와, #119
+ 0xC644, // 완, #361
+ 0xC654, // 왔, #489
+ 0xC655, // 왕, #418
+ 0xC678, // 외, #218
+ 0xC694, // 요, #43
+ 0xC695, // 욕, #479
+ 0xC6A9, // 용, #48
+ 0xC6B0, // 우, #64
+ 0xC6B1, // 욱, #503
+ 0xC6B4, // 운, #108
+ 0xC6B8, // 울, #223
+ 0xC6C0, // 움, #317
+ 0xC6C3, // 웃, #404
+ 0xC6CC, // 워, #280
+ 0xC6D0, // 원, #45
+ 0xC6D4, // 월, #150
+ 0xC6E8, // 웨, #446
+ 0xC6F9, // 웹, #500
+ 0xC704, // 위, #78
+ 0xC720, // 유, #81
+ 0xC721, // 육, #321
+ 0xC724, // 윤, #416
+ 0xC73C, // 으, #49
+ 0xC740, // 은, #31
+ 0xC744, // 을, #17
+ 0xC74C, // 음, #112
+ 0xC751, // 응, #461
+ 0xC758, // 의, #8
+ 0xC774, // 이, #1
+ 0xC775, // 익, #403
+ 0xC778, // 인, #18
+ 0xC77C, // 일, #28
+ 0xC784, // 임, #160
+ 0xC785, // 입, #93
+ 0xC788, // 있, #44
+ 0xC790, // 자, #22
+ 0xC791, // 작, #88
+ 0xC798, // 잘, #347
+ 0xC7A1, // 잡, #372
+ 0xC7A5, // 장, #53
+ 0xC7AC, // 재, #120
+ 0xC7C1, // 쟁, #483
+ 0xC800, // 저, #98
+ 0xC801, // 적, #97
+ 0xC804, // 전, #34
+ 0xC808, // 절, #320
+ 0xC810, // 점, #201
+ 0xC811, // 접, #331
+ 0xC815, // 정, #26
+ 0xC81C, // 제, #29
+ 0xC838, // 져, #414
+ 0xC870, // 조, #86
+ 0xC871, // 족, #373
+ 0xC874, // 존, #432
+ 0xC880, // 좀, #470
+ 0xC885, // 종, #208
+ 0xC88B, // 좋, #239
+ 0xC8E0, // 죠, #451
+ 0xC8FC, // 주, #38
+ 0xC8FD, // 죽, #471
+ 0xC900, // 준, #286
+ 0xC904, // 줄, #392
+ 0xC911, // 중, #103
+ 0xC988, // 즈, #255
+ 0xC98C, // 즌, #507
+ 0xC990, // 즐, #371
+ 0xC99D, // 증, #260
+ 0xC9C0, // 지, #10
+ 0xC9C1, // 직, #216
+ 0xC9C4, // 진, #79
+ 0xC9C8, // 질, #238
+ 0xC9D1, // 집, #206
+ 0xC9DC, // 짜, #411
+ 0xC9F8, // 째, #494
+ 0xCABD, // 쪽, #435
+ 0xCC28, // 차, #146
+ 0xCC29, // 착, #443
+ 0xCC2C, // 찬, #481
+ 0xCC30, // 찰, #440
+ 0xCC38, // 참, #343
+ 0xCC3D, // 창, #304
+ 0xCC3E, // 찾, #335
+ 0xCC44, // 채, #284
+ 0xCC45, // 책, #298
+ 0xCC98, // 처, #242
+ 0xCC9C, // 천, #143
+ 0xCCA0, // 철, #380
+ 0xCCA8, // 첨, #452
+ 0xCCAB, // 첫, #484
+ 0xCCAD, // 청, #197
+ 0xCCB4, // 체, #126
+ 0xCCD0, // 쳐, #472
+ 0xCD08, // 초, #220
+ 0xCD1D, // 총, #406
+ 0xCD5C, // 최, #179
+ 0xCD94, // 추, #136
+ 0xCD95, // 축, #337
+ 0xCD9C, // 출, #166
+ 0xCDA9, // 충, #369
+ 0xCDE8, // 취, #210
+ 0xCE20, // 츠, #215
+ 0xCE21, // 측, #468
+ 0xCE35, // 층, #512
+ 0xCE58, // 치, #102
+ 0xCE5C, // 친, #325
+ 0xCE68, // 침, #263
+ 0xCE74, // 카, #115
+ 0xCE7C, // 칼, #466
+ 0xCE90, // 캐, #454
+ 0xCEE4, // 커, #285
+ 0xCEE8, // 컨, #328
+ 0xCEF4, // 컴, #417
+ 0xCF00, // 케, #339
+ 0xCF13, // 켓, #509
+ 0xCF1C, // 켜, #508
+ 0xCF54, // 코, #193
+ 0xCF58, // 콘, #391
+ 0xCFE0, // 쿠, #393
+ 0xD035, // 퀵, #453
+ 0xD06C, // 크, #101
+ 0xD070, // 큰, #495
+ 0xD074, // 클, #289
+ 0xD0A4, // 키, #230
+ 0xD0C0, // 타, #127
+ 0xD0C1, // 탁, #314
+ 0xD0C4, // 탄, #450
+ 0xD0C8, // 탈, #436
+ 0xD0DC, // 태, #221
+ 0xD0DD, // 택, #275
+ 0xD130, // 터, #70
+ 0xD14C, // 테, #213
+ 0xD150, // 텐, #324
+ 0xD154, // 텔, #430
+ 0xD15C, // 템, #382
+ 0xD1A0, // 토, #145
+ 0xD1B5, // 통, #156
+ 0xD22C, // 투, #227
+ 0xD2B8, // 트, #37
+ 0xD2B9, // 특, #247
+ 0xD2F0, // 티, #187
+ 0xD305, // 팅, #410
+ 0xD30C, // 파, #141
+ 0xD310, // 판, #163
+ 0xD314, // 팔, #499
+ 0xD328, // 패, #307
+ 0xD32C, // 팬, #459
+ 0xD338, // 팸, #433
+ 0xD37C, // 퍼, #344
+ 0xD398, // 페, #172
+ 0xD3B8, // 편, #251
+ 0xD3C9, // 평, #291
+ 0xD3EC, // 포, #68
+ 0xD3ED, // 폭, #445
+ 0xD3F0, // 폰, #318
+ 0xD45C, // 표, #232
+ 0xD480, // 풀, #497
+ 0xD488, // 품, #113
+ 0xD48D, // 풍, #425
+ 0xD504, // 프, #110
+ 0xD508, // 픈, #498
+ 0xD50C, // 플, #211
+ 0xD53C, // 피, #169
+ 0xD544, // 필, #295
+ 0xD551, // 핑, #376
+ 0xD558, // 하, #7
+ 0xD559, // 학, #129
+ 0xD55C, // 한, #15
+ 0xD560, // 할, #144
+ 0xD568, // 함, #152
+ 0xD569, // 합, #123
+ 0xD56D, // 항, #268
+ 0xD574, // 해, #32
+ 0xD588, // 했, #180
+ 0xD589, // 행, #135
+ 0xD5A5, // 향, #345
+ 0xD5C8, // 허, #396
+ 0xD5D8, // 험, #316
+ 0xD5E4, // 헤, #474
+ 0xD604, // 현, #185
+ 0xD611, // 협, #315
+ 0xD615, // 형, #244
+ 0xD61C, // 혜, #428
+ 0xD638, // 호, #117
+ 0xD63C, // 혼, #358
+ 0xD648, // 홈, #330
+ 0xD64D, // 홍, #363
+ 0xD654, // 화, #63
+ 0xD655, // 확, #183
+ 0xD658, // 환, #224
+ 0xD65C, // 활, #277
+ 0xD669, // 황, #353
+ 0xD68C, // 회, #74
+ 0xD68D, // 획, #458
+ 0xD69F, // 횟, #409
+ 0xD6A8, // 효, #400
+ 0xD6C4, // 후, #176
+ 0xD6C8, // 훈, #486
+ 0xD734, // 휴, #365
+ 0xD754, // 흔, #480
+ 0xD76C, // 희, #334
+ 0xD788, // 히, #228
+ 0xD798, // 힘, #502
+};
+// the percentage of the sample covered by the above characters
+static const float frequent_ko_coverage=0.948157021464184;
+
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 86ff8bd..eef6a3d 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -106,6 +106,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
@@ -117,7 +118,7 @@
}
data.writeInt32((int32_t) output);
data.writeInt32((int32_t) tid);
- int lSessionId = 0;
+ int lSessionId = AUDIO_SESSION_ALLOCATE;
if (sessionId != NULL) {
lSessionId = *sessionId;
}
@@ -127,6 +128,10 @@
if (lStatus != NO_ERROR) {
ALOGE("createTrack error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -138,8 +143,19 @@
name = reply.readString8();
lStatus = reply.readInt32();
track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+ if (lStatus == NO_ERROR) {
+ if (track == 0) {
+ ALOGE("createTrack should have returned an IAudioTrack");
+ lStatus = UNKNOWN_ERROR;
+ }
+ } else {
+ if (track != 0) {
+ ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
+ track.clear();
+ }
+ }
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return track;
@@ -150,7 +166,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid,
int *sessionId,
@@ -163,11 +179,12 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
data.writeInt32((int32_t) tid);
- int lSessionId = 0;
+ int lSessionId = AUDIO_SESSION_ALLOCATE;
if (sessionId != NULL) {
lSessionId = *sessionId;
}
@@ -176,6 +193,10 @@
if (lStatus != NO_ERROR) {
ALOGE("openRecord error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -198,7 +219,7 @@
}
}
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
return record;
@@ -415,15 +436,25 @@
audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
ALOGV("openOutput() returned output, %d", output);
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
latency = reply.readInt32();
- if (pLatencyMs != NULL) *pLatencyMs = latency;
+ if (pLatencyMs != NULL) {
+ *pLatencyMs = latency;
+ }
return output;
}
@@ -487,13 +518,21 @@
remote()->transact(OPEN_INPUT, data, &reply);
audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = (audio_devices_t)reply.readInt32();
- if (pDevices != NULL) *pDevices = devices;
+ if (pDevices != NULL) {
+ *pDevices = devices;
+ }
samplingRate = reply.readInt32();
- if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pSamplingRate != NULL) {
+ *pSamplingRate = samplingRate;
+ }
format = (audio_format_t) reply.readInt32();
- if (pFormat != NULL) *pFormat = format;
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
channelMask = (audio_channel_mask_t)reply.readInt32();
- if (pChannelMask != NULL) *pChannelMask = channelMask;
+ if (pChannelMask != NULL) {
+ *pChannelMask = channelMask;
+ }
return input;
}
@@ -535,11 +574,11 @@
status_t status = reply.readInt32();
if (status == NO_ERROR) {
uint32_t tmp = reply.readInt32();
- if (halFrames) {
+ if (halFrames != NULL) {
*halFrames = tmp;
}
tmp = reply.readInt32();
- if (dspFrames) {
+ if (dspFrames != NULL) {
*dspFrames = tmp;
}
}
@@ -551,8 +590,11 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) ioHandle);
- remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
- return reply.readInt32();
+ status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
+ if (status != NO_ERROR) {
+ return 0;
+ }
+ return (uint32_t) reply.readInt32();
}
virtual int newAudioSessionId()
@@ -657,7 +699,7 @@
if (pDesc == NULL) {
return effect;
- if (status) {
+ if (status != NULL) {
*status = BAD_VALUE;
}
}
@@ -675,7 +717,7 @@
} else {
lStatus = reply.readInt32();
int tmp = reply.readInt32();
- if (id) {
+ if (id != NULL) {
*id = tmp;
}
tmp = reply.readInt32();
@@ -685,7 +727,7 @@
effect = interface_cast<IEffect>(reply.readStrongBinder());
reply.read(pDesc, sizeof(effect_descriptor_t));
}
- if (status) {
+ if (status != NULL) {
*status = lStatus;
}
@@ -775,9 +817,11 @@
} else {
track = createTrack(
(audio_stream_type_t) streamType, sampleRate, format,
- channelMask, frameCount, &flags, buffer, output, tid,
+ channelMask, &frameCount, &flags, buffer, output, tid,
&sessionId, name, clientUid, &status);
+ LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
}
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeString8(name);
@@ -797,8 +841,9 @@
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(input,
- sampleRate, format, channelMask, frameCount, &flags, tid, &sessionId, &status);
+ sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeInt32(status);
@@ -1026,7 +1071,7 @@
case GET_INPUT_FRAMES_LOST: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
- reply->writeInt32(getInputFramesLost(ioHandle));
+ reply->writeInt32((int32_t) getInputFramesLost(ioHandle));
return NO_ERROR;
} break;
case NEW_AUDIO_SESSION_ID: {
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 4a7de65..9866d70 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -50,6 +50,9 @@
status_t status = remote()->transact(GET_CBLK, data, &reply);
if (status == NO_ERROR) {
cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ if (cblk != 0 && cblk->pointer() == NULL) {
+ cblk.clear();
+ }
}
return cblk;
}
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 3cd9cfd..ffc21fc 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -60,6 +60,9 @@
status_t status = remote()->transact(GET_CBLK, data, &reply);
if (status == NO_ERROR) {
cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ if (cblk != 0 && cblk->pointer() == NULL) {
+ cblk.clear();
+ }
}
return cblk;
}
@@ -122,6 +125,9 @@
status = reply.readInt32();
if (status == NO_ERROR) {
*buffer = interface_cast<IMemory>(reply.readStrongBinder());
+ if (*buffer != 0 && (*buffer)->pointer() == NULL) {
+ (*buffer).clear();
+ }
}
}
return status;
diff --git a/media/libmedia/IEffect.cpp b/media/libmedia/IEffect.cpp
index a303a8f..b94012a 100644
--- a/media/libmedia/IEffect.cpp
+++ b/media/libmedia/IEffect.cpp
@@ -117,6 +117,9 @@
status_t status = remote()->transact(GET_CBLK, data, &reply);
if (status == NO_ERROR) {
cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ if (cblk != 0 && cblk->pointer() == NULL) {
+ cblk.clear();
+ }
}
return cblk;
}
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 9db5b1b..10b4934 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -75,7 +75,7 @@
}
void
-IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who) {
+IMediaDeathNotifier::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
ALOGW("media server died");
// Need to do this with the lock held
diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
new file mode 100644
index 0000000..622d9cf
--- /dev/null
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPConnection"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+enum {
+ CONNECT = IBinder::FIRST_CALL_TRANSACTION,
+ DISCONNECT,
+ READ_AT,
+ GET_SIZE,
+ GET_MIME_TYPE,
+};
+
+struct BpMediaHTTPConnection : public BpInterface<IMediaHTTPConnection> {
+ BpMediaHTTPConnection(const sp<IBinder> &impl)
+ : BpInterface<IMediaHTTPConnection>(impl) {
+ }
+
+ virtual bool connect(
+ const char *uri, const KeyedVector<String8, String8> *headers) {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ String16 tmp(uri);
+ data.writeString16(tmp);
+
+ tmp = String16("");
+ if (headers != NULL) {
+ for (size_t i = 0; i < headers->size(); ++i) {
+ String16 key(headers->keyAt(i).string());
+ String16 val(headers->valueAt(i).string());
+
+ tmp.append(key);
+ tmp.append(String16(": "));
+ tmp.append(val);
+ tmp.append(String16("\r\n"));
+ }
+ }
+ data.writeString16(tmp);
+
+ remote()->transact(CONNECT, data, &reply);
+
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<IBinder> binder = reply.readStrongBinder();
+ mMemory = interface_cast<IMemory>(binder);
+
+ return mMemory != NULL;
+ }
+
+ virtual void disconnect() {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ remote()->transact(DISCONNECT, data, &reply);
+ }
+
+ virtual ssize_t readAt(off64_t offset, void *buffer, size_t size) {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ data.writeInt64(offset);
+ data.writeInt32(size);
+
+ status_t err = remote()->transact(READ_AT, data, &reply);
+ CHECK_EQ(err, (status_t)OK);
+
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode) {
+ return UNKNOWN_ERROR;
+ }
+
+ int32_t len = reply.readInt32();
+
+ if (len > 0) {
+ memcpy(buffer, mMemory->pointer(), len);
+ }
+
+ return len;
+ }
+
+ virtual off64_t getSize() {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ remote()->transact(GET_SIZE, data, &reply);
+
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode) {
+ return UNKNOWN_ERROR;
+ }
+
+ return reply.readInt64();
+ }
+
+ virtual status_t getMIMEType(String8 *mimeType) {
+ *mimeType = String8("");
+
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPConnection::getInterfaceDescriptor());
+
+ remote()->transact(GET_MIME_TYPE, data, &reply);
+
+ int32_t exceptionCode = reply.readExceptionCode();
+
+ if (exceptionCode) {
+ return UNKNOWN_ERROR;
+ }
+
+ *mimeType = String8(reply.readString16());
+
+ return OK;
+ }
+
+private:
+ sp<IMemory> mMemory;
+};
+
+IMPLEMENT_META_INTERFACE(
+ MediaHTTPConnection, "android.media.IMediaHTTPConnection");
+
+} // namespace android
+
diff --git a/media/libmedia/IMediaHTTPService.cpp b/media/libmedia/IMediaHTTPService.cpp
new file mode 100644
index 0000000..1260582
--- /dev/null
+++ b/media/libmedia/IMediaHTTPService.cpp
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IMediaHTTPService"
+#include <utils/Log.h>
+
+#include <media/IMediaHTTPService.h>
+
+#include <binder/Parcel.h>
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+enum {
+ MAKE_HTTP = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+struct BpMediaHTTPService : public BpInterface<IMediaHTTPService> {
+ BpMediaHTTPService(const sp<IBinder> &impl)
+ : BpInterface<IMediaHTTPService>(impl) {
+ }
+
+ virtual sp<IMediaHTTPConnection> makeHTTPConnection() {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ IMediaHTTPService::getInterfaceDescriptor());
+
+ remote()->transact(MAKE_HTTP, data, &reply);
+
+ status_t err = reply.readInt32();
+
+ if (err != OK) {
+ return NULL;
+ }
+
+ return interface_cast<IMediaHTTPConnection>(reply.readStrongBinder());
+ }
+};
+
+IMPLEMENT_META_INTERFACE(
+ MediaHTTPService, "android.media.IMediaHTTPService");
+
+} // namespace android
+
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index bb066a0..c7d9d51 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -18,6 +18,7 @@
#include <stdint.h>
#include <sys/types.h>
#include <binder/Parcel.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaMetadataRetriever.h>
#include <utils/String8.h>
#include <utils/KeyedVector.h>
@@ -84,10 +85,16 @@
}
status_t setDataSource(
- const char *srcUrl, const KeyedVector<String8, String8> *headers)
+ const sp<IMediaHTTPService> &httpService,
+ const char *srcUrl,
+ const KeyedVector<String8, String8> *headers)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
+ data.writeInt32(httpService != NULL);
+ if (httpService != NULL) {
+ data.writeStrongBinder(httpService->asBinder());
+ }
data.writeCString(srcUrl);
if (headers == NULL) {
@@ -195,6 +202,13 @@
} break;
case SET_DATA_SOURCE_URL: {
CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
+
+ sp<IMediaHTTPService> httpService;
+ if (data.readInt32()) {
+ httpService =
+ interface_cast<IMediaHTTPService>(data.readStrongBinder());
+ }
+
const char* srcUrl = data.readCString();
KeyedVector<String8, String8> headers;
@@ -206,7 +220,8 @@
}
reply->writeInt32(
- setDataSource(srcUrl, numHeaders > 0 ? &headers : NULL));
+ setDataSource(
+ httpService, srcUrl, numHeaders > 0 ? &headers : NULL));
return NO_ERROR;
} break;
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index e79bcd2..d778d05 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -21,6 +21,7 @@
#include <binder/Parcel.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayer.h>
#include <media/IStreamSource.h>
@@ -75,11 +76,17 @@
remote()->transact(DISCONNECT, data, &reply);
}
- status_t setDataSource(const char* url,
+ status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
const KeyedVector<String8, String8>* headers)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ data.writeInt32(httpService != NULL);
+ if (httpService != NULL) {
+ data.writeStrongBinder(httpService->asBinder());
+ }
data.writeCString(url);
if (headers == NULL) {
data.writeInt32(0);
@@ -355,6 +362,13 @@
} break;
case SET_DATA_SOURCE_URL: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
+
+ sp<IMediaHTTPService> httpService;
+ if (data.readInt32()) {
+ httpService =
+ interface_cast<IMediaHTTPService>(data.readStrongBinder());
+ }
+
const char* url = data.readCString();
KeyedVector<String8, String8> headers;
int32_t numHeaders = data.readInt32();
@@ -363,7 +377,8 @@
String8 value = data.readString8();
headers.add(key, value);
}
- reply->writeInt32(setDataSource(url, numHeaders > 0 ? &headers : NULL));
+ reply->writeInt32(setDataSource(
+ httpService, url, numHeaders > 0 ? &headers : NULL));
return NO_ERROR;
} break;
case SET_DATA_SOURCE_FD: {
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 3c22b4c..d116b14 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -23,6 +23,7 @@
#include <media/ICrypto.h>
#include <media/IDrm.h>
#include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/IMediaRecorder.h>
#include <media/IOMX.h>
@@ -48,7 +49,6 @@
ADD_BATTERY_DATA,
PULL_BATTERY_DATA,
LISTEN_FOR_REMOTE_DISPLAY,
- UPDATE_PROXY_CONFIG,
};
class BpMediaPlayerService: public BpInterface<IMediaPlayerService>
@@ -86,12 +86,21 @@
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
- virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
+ virtual status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
+ data.writeInt32(httpService != NULL);
+ if (httpService != NULL) {
+ data.writeStrongBinder(httpService->asBinder());
+ }
data.writeCString(url);
data.writeStrongBinder(heap->asBinder());
status_t status = remote()->transact(DECODE_URL, data, &reply);
@@ -182,25 +191,6 @@
remote()->transact(LISTEN_FOR_REMOTE_DISPLAY, data, &reply);
return interface_cast<IRemoteDisplay>(reply.readStrongBinder());
}
-
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- Parcel data, reply;
-
- data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
- if (host == NULL) {
- data.writeInt32(0);
- } else {
- data.writeInt32(1);
- data.writeCString(host);
- data.writeInt32(port);
- data.writeCString(exclusionList);
- }
-
- remote()->transact(UPDATE_PROXY_CONFIG, data, &reply);
-
- return reply.readInt32();
- }
};
IMPLEMENT_META_INTERFACE(MediaPlayerService, "android.media.IMediaPlayerService");
@@ -222,13 +212,25 @@
} break;
case DECODE_URL: {
CHECK_INTERFACE(IMediaPlayerService, data, reply);
+ sp<IMediaHTTPService> httpService;
+ if (data.readInt32()) {
+ httpService =
+ interface_cast<IMediaHTTPService>(data.readStrongBinder());
+ }
const char* url = data.readCString();
sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
uint32_t sampleRate;
int numChannels;
audio_format_t format;
size_t size;
- status_t status = decode(url, &sampleRate, &numChannels, &format, heap, &size);
+ status_t status =
+ decode(httpService,
+ url,
+ &sampleRate,
+ &numChannels,
+ &format,
+ heap,
+ &size);
reply->writeInt32(status);
if (status == NO_ERROR) {
reply->writeInt32(sampleRate);
@@ -316,24 +318,6 @@
reply->writeStrongBinder(display->asBinder());
return NO_ERROR;
} break;
- case UPDATE_PROXY_CONFIG:
- {
- CHECK_INTERFACE(IMediaPlayerService, data, reply);
-
- const char *host = NULL;
- int32_t port = 0;
- const char *exclusionList = NULL;
-
- if (data.readInt32()) {
- host = data.readCString();
- port = data.readInt32();
- exclusionList = data.readCString();
- }
-
- reply->writeInt32(updateProxyConfig(host, port, exclusionList));
-
- return OK;
- }
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 8319cd7..1074da9 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -81,8 +81,14 @@
{"timelapseqvga", CAMCORDER_QUALITY_TIME_LAPSE_QVGA},
};
+#if LOG_NDEBUG
+#define UNUSED __unused
+#else
+#define UNUSED
+#endif
+
/*static*/ void
-MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec)
+MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec UNUSED)
{
ALOGV("video codec:");
ALOGV("codec = %d", codec.mCodec);
@@ -93,7 +99,7 @@
}
/*static*/ void
-MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec)
+MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec UNUSED)
{
ALOGV("audio codec:");
ALOGV("codec = %d", codec.mCodec);
@@ -103,7 +109,7 @@
}
/*static*/ void
-MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap)
+MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap UNUSED)
{
ALOGV("video encoder cap:");
ALOGV("codec = %d", cap.mCodec);
@@ -114,7 +120,7 @@
}
/*static*/ void
-MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap)
+MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap UNUSED)
{
ALOGV("audio encoder cap:");
ALOGV("codec = %d", cap.mCodec);
@@ -124,21 +130,21 @@
}
/*static*/ void
-MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap)
+MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap UNUSED)
{
ALOGV("video decoder cap:");
ALOGV("codec = %d", cap.mCodec);
}
/*static*/ void
-MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap)
+MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap UNUSED)
{
ALOGV("audio codec cap:");
ALOGV("codec = %d", cap.mCodec);
}
/*static*/ void
-MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap)
+MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap UNUSED)
{
ALOGV("videoeditor cap:");
ALOGV("mMaxInputFrameWidth = %d", cap.mMaxInputFrameWidth);
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 93a4a4c..1661f04 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -14,217 +14,57 @@
* limitations under the License.
*/
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaScannerClient"
+#include <utils/Log.h>
+
#include <media/mediascanner.h>
+#include "CharacterEncodingDetector.h"
#include "StringArray.h"
-#include "autodetect.h"
-#include "unicode/ucnv.h"
-#include "unicode/ustring.h"
-
namespace android {
MediaScannerClient::MediaScannerClient()
- : mNames(NULL),
- mValues(NULL),
- mLocaleEncoding(kEncodingNone)
+ : mEncodingDetector(NULL)
{
}
MediaScannerClient::~MediaScannerClient()
{
- delete mNames;
- delete mValues;
+ delete mEncodingDetector;
}
void MediaScannerClient::setLocale(const char* locale)
{
- if (!locale) return;
-
- if (!strncmp(locale, "ja", 2))
- mLocaleEncoding = kEncodingShiftJIS;
- else if (!strncmp(locale, "ko", 2))
- mLocaleEncoding = kEncodingEUCKR;
- else if (!strncmp(locale, "zh", 2)) {
- if (!strcmp(locale, "zh_CN")) {
- // simplified chinese for mainland China
- mLocaleEncoding = kEncodingGBK;
- } else {
- // assume traditional for non-mainland Chinese locales (Taiwan, Hong Kong, Singapore)
- mLocaleEncoding = kEncodingBig5;
- }
- }
+ mLocale = locale; // not currently used
}
void MediaScannerClient::beginFile()
{
- mNames = new StringArray;
- mValues = new StringArray;
+ delete mEncodingDetector;
+ mEncodingDetector = new CharacterEncodingDetector();
}
status_t MediaScannerClient::addStringTag(const char* name, const char* value)
{
- if (mLocaleEncoding != kEncodingNone) {
- // don't bother caching strings that are all ASCII.
- // call handleStringTag directly instead.
- // check to see if value (which should be utf8) has any non-ASCII characters
- bool nonAscii = false;
- const char* chp = value;
- char ch;
- while ((ch = *chp++)) {
- if (ch & 0x80) {
- nonAscii = true;
- break;
- }
- }
-
- if (nonAscii) {
- // save the strings for later so they can be used for native encoding detection
- mNames->push_back(name);
- mValues->push_back(value);
- return OK;
- }
- // else fall through
- }
-
- // autodetection is not necessary, so no need to cache the values
- // pass directly to the client instead
- return handleStringTag(name, value);
-}
-
-static uint32_t possibleEncodings(const char* s)
-{
- uint32_t result = kEncodingAll;
- // if s contains a native encoding, then it was mistakenly encoded in utf8 as if it were latin-1
- // so we need to reverse the latin-1 -> utf8 conversion to get the native chars back
- uint8_t ch1, ch2;
- uint8_t* chp = (uint8_t *)s;
-
- while ((ch1 = *chp++)) {
- if (ch1 & 0x80) {
- ch2 = *chp++;
- ch1 = ((ch1 << 6) & 0xC0) | (ch2 & 0x3F);
- // ch1 is now the first byte of the potential native char
-
- ch2 = *chp++;
- if (ch2 & 0x80)
- ch2 = ((ch2 << 6) & 0xC0) | (*chp++ & 0x3F);
- // ch2 is now the second byte of the potential native char
- int ch = (int)ch1 << 8 | (int)ch2;
- result &= findPossibleEncodings(ch);
- }
- // else ASCII character, which could be anything
- }
-
- return result;
-}
-
-void MediaScannerClient::convertValues(uint32_t encoding)
-{
- const char* enc = NULL;
- switch (encoding) {
- case kEncodingShiftJIS:
- enc = "shift-jis";
- break;
- case kEncodingGBK:
- enc = "gbk";
- break;
- case kEncodingBig5:
- enc = "Big5";
- break;
- case kEncodingEUCKR:
- enc = "EUC-KR";
- break;
- }
-
- if (enc) {
- UErrorCode status = U_ZERO_ERROR;
-
- UConverter *conv = ucnv_open(enc, &status);
- if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for %s", enc);
- return;
- }
- UConverter *utf8Conv = ucnv_open("UTF-8", &status);
- if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for UTF-8");
- ucnv_close(conv);
- return;
- }
-
- // for each value string, convert from native encoding to UTF-8
- for (int i = 0; i < mNames->size(); i++) {
- // first we need to untangle the utf8 and convert it back to the original bytes
- // since we are reducing the length of the string, we can do this in place
- uint8_t* src = (uint8_t *)mValues->getEntry(i);
- int len = strlen((char *)src);
- uint8_t* dest = src;
-
- uint8_t uch;
- while ((uch = *src++)) {
- if (uch & 0x80)
- *dest++ = ((uch << 6) & 0xC0) | (*src++ & 0x3F);
- else
- *dest++ = uch;
- }
- *dest = 0;
-
- // now convert from native encoding to UTF-8
- const char* source = mValues->getEntry(i);
- int targetLength = len * 3 + 1;
- char* buffer = new char[targetLength];
- // don't normally check for NULL, but in this case targetLength may be large
- if (!buffer)
- break;
- char* target = buffer;
-
- ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
- &source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
- if (U_FAILURE(status)) {
- ALOGE("ucnv_convertEx failed: %d", status);
- mValues->setEntry(i, "???");
- } else {
- // zero terminate
- *target = 0;
- mValues->setEntry(i, buffer);
- }
-
- delete[] buffer;
- }
-
- ucnv_close(conv);
- ucnv_close(utf8Conv);
- }
+ mEncodingDetector->addTag(name, value);
+ return OK;
}
void MediaScannerClient::endFile()
{
- if (mLocaleEncoding != kEncodingNone) {
- int size = mNames->size();
- uint32_t encoding = kEncodingAll;
+ mEncodingDetector->detectAndConvert();
- // compute a bit mask containing all possible encodings
- for (int i = 0; i < mNames->size(); i++)
- encoding &= possibleEncodings(mValues->getEntry(i));
-
- // if the locale encoding matches, then assume we have a native encoding.
- if (encoding & mLocaleEncoding)
- convertValues(mLocaleEncoding);
-
- // finally, push all name/value pairs to the client
- for (int i = 0; i < mNames->size(); i++) {
- status_t status = handleStringTag(mNames->getEntry(i), mValues->getEntry(i));
- if (status) {
- break;
- }
+ int size = mEncodingDetector->size();
+ if (size) {
+ for (int i = 0; i < size; i++) {
+ const char *name;
+ const char *value;
+ mEncodingDetector->getTag(i, &name, &value);
+ handleStringTag(name, value);
}
}
- // else addStringTag() has done all the work so we have nothing to do
-
- delete mNames;
- delete mValues;
- mNames = NULL;
- mValues = NULL;
}
} // namespace android
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 22e9fad..4885b4f 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -21,6 +21,7 @@
#define USE_SHARED_MEM_BUFFER
#include <media/AudioTrack.h>
+#include <media/IMediaHTTPService.h>
#include <media/mediaplayer.h>
#include <media/SoundPool.h>
#include "SoundPoolThread.h"
@@ -199,7 +200,7 @@
return NULL;
}
-int SoundPool::load(const char* path, int priority)
+int SoundPool::load(const char* path, int priority __unused)
{
ALOGV("load: path=%s, priority=%d", path, priority);
Mutex::Autolock lock(&mLock);
@@ -209,7 +210,7 @@
return sample->sampleID();
}
-int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
+int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
{
ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
fd, offset, length, priority);
@@ -496,7 +497,14 @@
ALOGV("Start decode");
if (mUrl) {
- status = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format, mHeap, &mSize);
+ status = MediaPlayer::decode(
+ NULL /* httpService */,
+ mUrl,
+ &sampleRate,
+ &numChannels,
+ &format,
+ mHeap,
+ &mSize);
} else {
status = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
mHeap, &mSize);
@@ -600,16 +608,15 @@
// wrong audio audio buffer size (mAudioBufferSize)
unsigned long toggle = mToggle ^ 1;
void *userData = (void *)((unsigned long)this | toggle);
- uint32_t channels = (numChannels == 2) ?
- AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO;
+ audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
// do not create a new audio track if current track is compatible with sample parameters
#ifdef USE_SHARED_MEM_BUFFER
newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channels, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
+ channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
#else
newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channels, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
+ channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
bufferFrames);
#endif
oldTrack = mAudioTrack;
@@ -730,7 +737,8 @@
count = b->size;
}
memcpy(q, p, count);
-// ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, count);
+// ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
+// count);
} else if (mPos < mAudioBufferSize) {
count = mAudioBufferSize - mPos;
if (count > b->size) {
diff --git a/media/libmedia/autodetect.cpp b/media/libmedia/autodetect.cpp
deleted file mode 100644
index be5c3b2..0000000
--- a/media/libmedia/autodetect.cpp
+++ /dev/null
@@ -1,885 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "autodetect.h"
-
-struct CharRange {
- uint16_t first;
- uint16_t last;
-};
-
-#define ARRAY_SIZE(x) (sizeof(x) / sizeof(*x))
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP932.TXT
-static const CharRange kShiftJISRanges[] = {
- { 0x8140, 0x817E },
- { 0x8180, 0x81AC },
- { 0x81B8, 0x81BF },
- { 0x81C8, 0x81CE },
- { 0x81DA, 0x81E8 },
- { 0x81F0, 0x81F7 },
- { 0x81FC, 0x81FC },
- { 0x824F, 0x8258 },
- { 0x8260, 0x8279 },
- { 0x8281, 0x829A },
- { 0x829F, 0x82F1 },
- { 0x8340, 0x837E },
- { 0x8380, 0x8396 },
- { 0x839F, 0x83B6 },
- { 0x83BF, 0x83D6 },
- { 0x8440, 0x8460 },
- { 0x8470, 0x847E },
- { 0x8480, 0x8491 },
- { 0x849F, 0x84BE },
- { 0x8740, 0x875D },
- { 0x875F, 0x8775 },
- { 0x877E, 0x877E },
- { 0x8780, 0x879C },
- { 0x889F, 0x88FC },
- { 0x8940, 0x897E },
- { 0x8980, 0x89FC },
- { 0x8A40, 0x8A7E },
- { 0x8A80, 0x8AFC },
- { 0x8B40, 0x8B7E },
- { 0x8B80, 0x8BFC },
- { 0x8C40, 0x8C7E },
- { 0x8C80, 0x8CFC },
- { 0x8D40, 0x8D7E },
- { 0x8D80, 0x8DFC },
- { 0x8E40, 0x8E7E },
- { 0x8E80, 0x8EFC },
- { 0x8F40, 0x8F7E },
- { 0x8F80, 0x8FFC },
- { 0x9040, 0x907E },
- { 0x9080, 0x90FC },
- { 0x9140, 0x917E },
- { 0x9180, 0x91FC },
- { 0x9240, 0x927E },
- { 0x9280, 0x92FC },
- { 0x9340, 0x937E },
- { 0x9380, 0x93FC },
- { 0x9440, 0x947E },
- { 0x9480, 0x94FC },
- { 0x9540, 0x957E },
- { 0x9580, 0x95FC },
- { 0x9640, 0x967E },
- { 0x9680, 0x96FC },
- { 0x9740, 0x977E },
- { 0x9780, 0x97FC },
- { 0x9840, 0x9872 },
- { 0x989F, 0x98FC },
- { 0x9940, 0x997E },
- { 0x9980, 0x99FC },
- { 0x9A40, 0x9A7E },
- { 0x9A80, 0x9AFC },
- { 0x9B40, 0x9B7E },
- { 0x9B80, 0x9BFC },
- { 0x9C40, 0x9C7E },
- { 0x9C80, 0x9CFC },
- { 0x9D40, 0x9D7E },
- { 0x9D80, 0x9DFC },
- { 0x9E40, 0x9E7E },
- { 0x9E80, 0x9EFC },
- { 0x9F40, 0x9F7E },
- { 0x9F80, 0x9FFC },
- { 0xE040, 0xE07E },
- { 0xE080, 0xE0FC },
- { 0xE140, 0xE17E },
- { 0xE180, 0xE1FC },
- { 0xE240, 0xE27E },
- { 0xE280, 0xE2FC },
- { 0xE340, 0xE37E },
- { 0xE380, 0xE3FC },
- { 0xE440, 0xE47E },
- { 0xE480, 0xE4FC },
- { 0xE540, 0xE57E },
- { 0xE580, 0xE5FC },
- { 0xE640, 0xE67E },
- { 0xE680, 0xE6FC },
- { 0xE740, 0xE77E },
- { 0xE780, 0xE7FC },
- { 0xE840, 0xE87E },
- { 0xE880, 0xE8FC },
- { 0xE940, 0xE97E },
- { 0xE980, 0xE9FC },
- { 0xEA40, 0xEA7E },
- { 0xEA80, 0xEAA4 },
- { 0xED40, 0xED7E },
- { 0xED80, 0xEDFC },
- { 0xEE40, 0xEE7E },
- { 0xEE80, 0xEEEC },
- { 0xEEEF, 0xEEFC },
- { 0xFA40, 0xFA7E },
- { 0xFA80, 0xFAFC },
- { 0xFB40, 0xFB7E },
- { 0xFB80, 0xFBFC },
- { 0xFC40, 0xFC4B },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP936.TXT
-static const CharRange kGBKRanges[] = {
- { 0x8140, 0x817E },
- { 0x8180, 0x81FE },
- { 0x8240, 0x827E },
- { 0x8280, 0x82FE },
- { 0x8340, 0x837E },
- { 0x8380, 0x83FE },
- { 0x8440, 0x847E },
- { 0x8480, 0x84FE },
- { 0x8540, 0x857E },
- { 0x8580, 0x85FE },
- { 0x8640, 0x867E },
- { 0x8680, 0x86FE },
- { 0x8740, 0x877E },
- { 0x8780, 0x87FE },
- { 0x8840, 0x887E },
- { 0x8880, 0x88FE },
- { 0x8940, 0x897E },
- { 0x8980, 0x89FE },
- { 0x8A40, 0x8A7E },
- { 0x8A80, 0x8AFE },
- { 0x8B40, 0x8B7E },
- { 0x8B80, 0x8BFE },
- { 0x8C40, 0x8C7E },
- { 0x8C80, 0x8CFE },
- { 0x8D40, 0x8D7E },
- { 0x8D80, 0x8DFE },
- { 0x8E40, 0x8E7E },
- { 0x8E80, 0x8EFE },
- { 0x8F40, 0x8F7E },
- { 0x8F80, 0x8FFE },
- { 0x9040, 0x907E },
- { 0x9080, 0x90FE },
- { 0x9140, 0x917E },
- { 0x9180, 0x91FE },
- { 0x9240, 0x927E },
- { 0x9280, 0x92FE },
- { 0x9340, 0x937E },
- { 0x9380, 0x93FE },
- { 0x9440, 0x947E },
- { 0x9480, 0x94FE },
- { 0x9540, 0x957E },
- { 0x9580, 0x95FE },
- { 0x9640, 0x967E },
- { 0x9680, 0x96FE },
- { 0x9740, 0x977E },
- { 0x9780, 0x97FE },
- { 0x9840, 0x987E },
- { 0x9880, 0x98FE },
- { 0x9940, 0x997E },
- { 0x9980, 0x99FE },
- { 0x9A40, 0x9A7E },
- { 0x9A80, 0x9AFE },
- { 0x9B40, 0x9B7E },
- { 0x9B80, 0x9BFE },
- { 0x9C40, 0x9C7E },
- { 0x9C80, 0x9CFE },
- { 0x9D40, 0x9D7E },
- { 0x9D80, 0x9DFE },
- { 0x9E40, 0x9E7E },
- { 0x9E80, 0x9EFE },
- { 0x9F40, 0x9F7E },
- { 0x9F80, 0x9FFE },
- { 0xA040, 0xA07E },
- { 0xA080, 0xA0FE },
- { 0xA1A1, 0xA1FE },
- { 0xA2A1, 0xA2AA },
- { 0xA2B1, 0xA2E2 },
- { 0xA2E5, 0xA2EE },
- { 0xA2F1, 0xA2FC },
- { 0xA3A1, 0xA3FE },
- { 0xA4A1, 0xA4F3 },
- { 0xA5A1, 0xA5F6 },
- { 0xA6A1, 0xA6B8 },
- { 0xA6C1, 0xA6D8 },
- { 0xA6E0, 0xA6EB },
- { 0xA6EE, 0xA6F2 },
- { 0xA6F4, 0xA6F5 },
- { 0xA7A1, 0xA7C1 },
- { 0xA7D1, 0xA7F1 },
- { 0xA840, 0xA87E },
- { 0xA880, 0xA895 },
- { 0xA8A1, 0xA8BB },
- { 0xA8BD, 0xA8BE },
- { 0xA8C0, 0xA8C0 },
- { 0xA8C5, 0xA8E9 },
- { 0xA940, 0xA957 },
- { 0xA959, 0xA95A },
- { 0xA95C, 0xA95C },
- { 0xA960, 0xA97E },
- { 0xA980, 0xA988 },
- { 0xA996, 0xA996 },
- { 0xA9A4, 0xA9EF },
- { 0xAA40, 0xAA7E },
- { 0xAA80, 0xAAA0 },
- { 0xAB40, 0xAB7E },
- { 0xAB80, 0xABA0 },
- { 0xAC40, 0xAC7E },
- { 0xAC80, 0xACA0 },
- { 0xAD40, 0xAD7E },
- { 0xAD80, 0xADA0 },
- { 0xAE40, 0xAE7E },
- { 0xAE80, 0xAEA0 },
- { 0xAF40, 0xAF7E },
- { 0xAF80, 0xAFA0 },
- { 0xB040, 0xB07E },
- { 0xB080, 0xB0FE },
- { 0xB140, 0xB17E },
- { 0xB180, 0xB1FE },
- { 0xB240, 0xB27E },
- { 0xB280, 0xB2FE },
- { 0xB340, 0xB37E },
- { 0xB380, 0xB3FE },
- { 0xB440, 0xB47E },
- { 0xB480, 0xB4FE },
- { 0xB540, 0xB57E },
- { 0xB580, 0xB5FE },
- { 0xB640, 0xB67E },
- { 0xB680, 0xB6FE },
- { 0xB740, 0xB77E },
- { 0xB780, 0xB7FE },
- { 0xB840, 0xB87E },
- { 0xB880, 0xB8FE },
- { 0xB940, 0xB97E },
- { 0xB980, 0xB9FE },
- { 0xBA40, 0xBA7E },
- { 0xBA80, 0xBAFE },
- { 0xBB40, 0xBB7E },
- { 0xBB80, 0xBBFE },
- { 0xBC40, 0xBC7E },
- { 0xBC80, 0xBCFE },
- { 0xBD40, 0xBD7E },
- { 0xBD80, 0xBDFE },
- { 0xBE40, 0xBE7E },
- { 0xBE80, 0xBEFE },
- { 0xBF40, 0xBF7E },
- { 0xBF80, 0xBFFE },
- { 0xC040, 0xC07E },
- { 0xC080, 0xC0FE },
- { 0xC140, 0xC17E },
- { 0xC180, 0xC1FE },
- { 0xC240, 0xC27E },
- { 0xC280, 0xC2FE },
- { 0xC340, 0xC37E },
- { 0xC380, 0xC3FE },
- { 0xC440, 0xC47E },
- { 0xC480, 0xC4FE },
- { 0xC540, 0xC57E },
- { 0xC580, 0xC5FE },
- { 0xC640, 0xC67E },
- { 0xC680, 0xC6FE },
- { 0xC740, 0xC77E },
- { 0xC780, 0xC7FE },
- { 0xC840, 0xC87E },
- { 0xC880, 0xC8FE },
- { 0xC940, 0xC97E },
- { 0xC980, 0xC9FE },
- { 0xCA40, 0xCA7E },
- { 0xCA80, 0xCAFE },
- { 0xCB40, 0xCB7E },
- { 0xCB80, 0xCBFE },
- { 0xCC40, 0xCC7E },
- { 0xCC80, 0xCCFE },
- { 0xCD40, 0xCD7E },
- { 0xCD80, 0xCDFE },
- { 0xCE40, 0xCE7E },
- { 0xCE80, 0xCEFE },
- { 0xCF40, 0xCF7E },
- { 0xCF80, 0xCFFE },
- { 0xD040, 0xD07E },
- { 0xD080, 0xD0FE },
- { 0xD140, 0xD17E },
- { 0xD180, 0xD1FE },
- { 0xD240, 0xD27E },
- { 0xD280, 0xD2FE },
- { 0xD340, 0xD37E },
- { 0xD380, 0xD3FE },
- { 0xD440, 0xD47E },
- { 0xD480, 0xD4FE },
- { 0xD540, 0xD57E },
- { 0xD580, 0xD5FE },
- { 0xD640, 0xD67E },
- { 0xD680, 0xD6FE },
- { 0xD740, 0xD77E },
- { 0xD780, 0xD7F9 },
- { 0xD840, 0xD87E },
- { 0xD880, 0xD8FE },
- { 0xD940, 0xD97E },
- { 0xD980, 0xD9FE },
- { 0xDA40, 0xDA7E },
- { 0xDA80, 0xDAFE },
- { 0xDB40, 0xDB7E },
- { 0xDB80, 0xDBFE },
- { 0xDC40, 0xDC7E },
- { 0xDC80, 0xDCFE },
- { 0xDD40, 0xDD7E },
- { 0xDD80, 0xDDFE },
- { 0xDE40, 0xDE7E },
- { 0xDE80, 0xDEFE },
- { 0xDF40, 0xDF7E },
- { 0xDF80, 0xDFFE },
- { 0xE040, 0xE07E },
- { 0xE080, 0xE0FE },
- { 0xE140, 0xE17E },
- { 0xE180, 0xE1FE },
- { 0xE240, 0xE27E },
- { 0xE280, 0xE2FE },
- { 0xE340, 0xE37E },
- { 0xE380, 0xE3FE },
- { 0xE440, 0xE47E },
- { 0xE480, 0xE4FE },
- { 0xE540, 0xE57E },
- { 0xE580, 0xE5FE },
- { 0xE640, 0xE67E },
- { 0xE680, 0xE6FE },
- { 0xE740, 0xE77E },
- { 0xE780, 0xE7FE },
- { 0xE840, 0xE87E },
- { 0xE880, 0xE8FE },
- { 0xE940, 0xE97E },
- { 0xE980, 0xE9FE },
- { 0xEA40, 0xEA7E },
- { 0xEA80, 0xEAFE },
- { 0xEB40, 0xEB7E },
- { 0xEB80, 0xEBFE },
- { 0xEC40, 0xEC7E },
- { 0xEC80, 0xECFE },
- { 0xED40, 0xED7E },
- { 0xED80, 0xEDFE },
- { 0xEE40, 0xEE7E },
- { 0xEE80, 0xEEFE },
- { 0xEF40, 0xEF7E },
- { 0xEF80, 0xEFFE },
- { 0xF040, 0xF07E },
- { 0xF080, 0xF0FE },
- { 0xF140, 0xF17E },
- { 0xF180, 0xF1FE },
- { 0xF240, 0xF27E },
- { 0xF280, 0xF2FE },
- { 0xF340, 0xF37E },
- { 0xF380, 0xF3FE },
- { 0xF440, 0xF47E },
- { 0xF480, 0xF4FE },
- { 0xF540, 0xF57E },
- { 0xF580, 0xF5FE },
- { 0xF640, 0xF67E },
- { 0xF680, 0xF6FE },
- { 0xF740, 0xF77E },
- { 0xF780, 0xF7FE },
- { 0xF840, 0xF87E },
- { 0xF880, 0xF8A0 },
- { 0xF940, 0xF97E },
- { 0xF980, 0xF9A0 },
- { 0xFA40, 0xFA7E },
- { 0xFA80, 0xFAA0 },
- { 0xFB40, 0xFB7E },
- { 0xFB80, 0xFBA0 },
- { 0xFC40, 0xFC7E },
- { 0xFC80, 0xFCA0 },
- { 0xFD40, 0xFD7E },
- { 0xFD80, 0xFDA0 },
- { 0xFE40, 0xFE4F },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP949.TXT
-static const CharRange kEUCKRRanges[] = {
- { 0x8141, 0x815A },
- { 0x8161, 0x817A },
- { 0x8181, 0x81FE },
- { 0x8241, 0x825A },
- { 0x8261, 0x827A },
- { 0x8281, 0x82FE },
- { 0x8341, 0x835A },
- { 0x8361, 0x837A },
- { 0x8381, 0x83FE },
- { 0x8441, 0x845A },
- { 0x8461, 0x847A },
- { 0x8481, 0x84FE },
- { 0x8541, 0x855A },
- { 0x8561, 0x857A },
- { 0x8581, 0x85FE },
- { 0x8641, 0x865A },
- { 0x8661, 0x867A },
- { 0x8681, 0x86FE },
- { 0x8741, 0x875A },
- { 0x8761, 0x877A },
- { 0x8781, 0x87FE },
- { 0x8841, 0x885A },
- { 0x8861, 0x887A },
- { 0x8881, 0x88FE },
- { 0x8941, 0x895A },
- { 0x8961, 0x897A },
- { 0x8981, 0x89FE },
- { 0x8A41, 0x8A5A },
- { 0x8A61, 0x8A7A },
- { 0x8A81, 0x8AFE },
- { 0x8B41, 0x8B5A },
- { 0x8B61, 0x8B7A },
- { 0x8B81, 0x8BFE },
- { 0x8C41, 0x8C5A },
- { 0x8C61, 0x8C7A },
- { 0x8C81, 0x8CFE },
- { 0x8D41, 0x8D5A },
- { 0x8D61, 0x8D7A },
- { 0x8D81, 0x8DFE },
- { 0x8E41, 0x8E5A },
- { 0x8E61, 0x8E7A },
- { 0x8E81, 0x8EFE },
- { 0x8F41, 0x8F5A },
- { 0x8F61, 0x8F7A },
- { 0x8F81, 0x8FFE },
- { 0x9041, 0x905A },
- { 0x9061, 0x907A },
- { 0x9081, 0x90FE },
- { 0x9141, 0x915A },
- { 0x9161, 0x917A },
- { 0x9181, 0x91FE },
- { 0x9241, 0x925A },
- { 0x9261, 0x927A },
- { 0x9281, 0x92FE },
- { 0x9341, 0x935A },
- { 0x9361, 0x937A },
- { 0x9381, 0x93FE },
- { 0x9441, 0x945A },
- { 0x9461, 0x947A },
- { 0x9481, 0x94FE },
- { 0x9541, 0x955A },
- { 0x9561, 0x957A },
- { 0x9581, 0x95FE },
- { 0x9641, 0x965A },
- { 0x9661, 0x967A },
- { 0x9681, 0x96FE },
- { 0x9741, 0x975A },
- { 0x9761, 0x977A },
- { 0x9781, 0x97FE },
- { 0x9841, 0x985A },
- { 0x9861, 0x987A },
- { 0x9881, 0x98FE },
- { 0x9941, 0x995A },
- { 0x9961, 0x997A },
- { 0x9981, 0x99FE },
- { 0x9A41, 0x9A5A },
- { 0x9A61, 0x9A7A },
- { 0x9A81, 0x9AFE },
- { 0x9B41, 0x9B5A },
- { 0x9B61, 0x9B7A },
- { 0x9B81, 0x9BFE },
- { 0x9C41, 0x9C5A },
- { 0x9C61, 0x9C7A },
- { 0x9C81, 0x9CFE },
- { 0x9D41, 0x9D5A },
- { 0x9D61, 0x9D7A },
- { 0x9D81, 0x9DFE },
- { 0x9E41, 0x9E5A },
- { 0x9E61, 0x9E7A },
- { 0x9E81, 0x9EFE },
- { 0x9F41, 0x9F5A },
- { 0x9F61, 0x9F7A },
- { 0x9F81, 0x9FFE },
- { 0xA041, 0xA05A },
- { 0xA061, 0xA07A },
- { 0xA081, 0xA0FE },
- { 0xA141, 0xA15A },
- { 0xA161, 0xA17A },
- { 0xA181, 0xA1FE },
- { 0xA241, 0xA25A },
- { 0xA261, 0xA27A },
- { 0xA281, 0xA2E7 },
- { 0xA341, 0xA35A },
- { 0xA361, 0xA37A },
- { 0xA381, 0xA3FE },
- { 0xA441, 0xA45A },
- { 0xA461, 0xA47A },
- { 0xA481, 0xA4FE },
- { 0xA541, 0xA55A },
- { 0xA561, 0xA57A },
- { 0xA581, 0xA5AA },
- { 0xA5B0, 0xA5B9 },
- { 0xA5C1, 0xA5D8 },
- { 0xA5E1, 0xA5F8 },
- { 0xA641, 0xA65A },
- { 0xA661, 0xA67A },
- { 0xA681, 0xA6E4 },
- { 0xA741, 0xA75A },
- { 0xA761, 0xA77A },
- { 0xA781, 0xA7EF },
- { 0xA841, 0xA85A },
- { 0xA861, 0xA87A },
- { 0xA881, 0xA8A4 },
- { 0xA8A6, 0xA8A6 },
- { 0xA8A8, 0xA8AF },
- { 0xA8B1, 0xA8FE },
- { 0xA941, 0xA95A },
- { 0xA961, 0xA97A },
- { 0xA981, 0xA9FE },
- { 0xAA41, 0xAA5A },
- { 0xAA61, 0xAA7A },
- { 0xAA81, 0xAAF3 },
- { 0xAB41, 0xAB5A },
- { 0xAB61, 0xAB7A },
- { 0xAB81, 0xABF6 },
- { 0xAC41, 0xAC5A },
- { 0xAC61, 0xAC7A },
- { 0xAC81, 0xACC1 },
- { 0xACD1, 0xACF1 },
- { 0xAD41, 0xAD5A },
- { 0xAD61, 0xAD7A },
- { 0xAD81, 0xADA0 },
- { 0xAE41, 0xAE5A },
- { 0xAE61, 0xAE7A },
- { 0xAE81, 0xAEA0 },
- { 0xAF41, 0xAF5A },
- { 0xAF61, 0xAF7A },
- { 0xAF81, 0xAFA0 },
- { 0xB041, 0xB05A },
- { 0xB061, 0xB07A },
- { 0xB081, 0xB0FE },
- { 0xB141, 0xB15A },
- { 0xB161, 0xB17A },
- { 0xB181, 0xB1FE },
- { 0xB241, 0xB25A },
- { 0xB261, 0xB27A },
- { 0xB281, 0xB2FE },
- { 0xB341, 0xB35A },
- { 0xB361, 0xB37A },
- { 0xB381, 0xB3FE },
- { 0xB441, 0xB45A },
- { 0xB461, 0xB47A },
- { 0xB481, 0xB4FE },
- { 0xB541, 0xB55A },
- { 0xB561, 0xB57A },
- { 0xB581, 0xB5FE },
- { 0xB641, 0xB65A },
- { 0xB661, 0xB67A },
- { 0xB681, 0xB6FE },
- { 0xB741, 0xB75A },
- { 0xB761, 0xB77A },
- { 0xB781, 0xB7FE },
- { 0xB841, 0xB85A },
- { 0xB861, 0xB87A },
- { 0xB881, 0xB8FE },
- { 0xB941, 0xB95A },
- { 0xB961, 0xB97A },
- { 0xB981, 0xB9FE },
- { 0xBA41, 0xBA5A },
- { 0xBA61, 0xBA7A },
- { 0xBA81, 0xBAFE },
- { 0xBB41, 0xBB5A },
- { 0xBB61, 0xBB7A },
- { 0xBB81, 0xBBFE },
- { 0xBC41, 0xBC5A },
- { 0xBC61, 0xBC7A },
- { 0xBC81, 0xBCFE },
- { 0xBD41, 0xBD5A },
- { 0xBD61, 0xBD7A },
- { 0xBD81, 0xBDFE },
- { 0xBE41, 0xBE5A },
- { 0xBE61, 0xBE7A },
- { 0xBE81, 0xBEFE },
- { 0xBF41, 0xBF5A },
- { 0xBF61, 0xBF7A },
- { 0xBF81, 0xBFFE },
- { 0xC041, 0xC05A },
- { 0xC061, 0xC07A },
- { 0xC081, 0xC0FE },
- { 0xC141, 0xC15A },
- { 0xC161, 0xC17A },
- { 0xC181, 0xC1FE },
- { 0xC241, 0xC25A },
- { 0xC261, 0xC27A },
- { 0xC281, 0xC2FE },
- { 0xC341, 0xC35A },
- { 0xC361, 0xC37A },
- { 0xC381, 0xC3FE },
- { 0xC441, 0xC45A },
- { 0xC461, 0xC47A },
- { 0xC481, 0xC4FE },
- { 0xC541, 0xC55A },
- { 0xC561, 0xC57A },
- { 0xC581, 0xC5FE },
- { 0xC641, 0xC652 },
- { 0xC6A1, 0xC6FE },
- { 0xC7A1, 0xC7FE },
- { 0xC8A1, 0xC8FE },
- { 0xCAA1, 0xCAFE },
- { 0xCBA1, 0xCBFE },
- { 0xCCA1, 0xCCFE },
- { 0xCDA1, 0xCDFE },
- { 0xCEA1, 0xCEFE },
- { 0xCFA1, 0xCFFE },
- { 0xD0A1, 0xD0FE },
- { 0xD1A1, 0xD1FE },
- { 0xD2A1, 0xD2FE },
- { 0xD3A1, 0xD3FE },
- { 0xD4A1, 0xD4FE },
- { 0xD5A1, 0xD5FE },
- { 0xD6A1, 0xD6FE },
- { 0xD7A1, 0xD7FE },
- { 0xD8A1, 0xD8FE },
- { 0xD9A1, 0xD9FE },
- { 0xDAA1, 0xDAFE },
- { 0xDBA1, 0xDBFE },
- { 0xDCA1, 0xDCFE },
- { 0xDDA1, 0xDDFE },
- { 0xDEA1, 0xDEFE },
- { 0xDFA1, 0xDFFE },
- { 0xE0A1, 0xE0FE },
- { 0xE1A1, 0xE1FE },
- { 0xE2A1, 0xE2FE },
- { 0xE3A1, 0xE3FE },
- { 0xE4A1, 0xE4FE },
- { 0xE5A1, 0xE5FE },
- { 0xE6A1, 0xE6FE },
- { 0xE7A1, 0xE7FE },
- { 0xE8A1, 0xE8FE },
- { 0xE9A1, 0xE9FE },
- { 0xEAA1, 0xEAFE },
- { 0xEBA1, 0xEBFE },
- { 0xECA1, 0xECFE },
- { 0xEDA1, 0xEDFE },
- { 0xEEA1, 0xEEFE },
- { 0xEFA1, 0xEFFE },
- { 0xF0A1, 0xF0FE },
- { 0xF1A1, 0xF1FE },
- { 0xF2A1, 0xF2FE },
- { 0xF3A1, 0xF3FE },
- { 0xF4A1, 0xF4FE },
- { 0xF5A1, 0xF5FE },
- { 0xF6A1, 0xF6FE },
- { 0xF7A1, 0xF7FE },
- { 0xF8A1, 0xF8FE },
- { 0xF9A1, 0xF9FE },
- { 0xFAA1, 0xFAFE },
- { 0xFBA1, 0xFBFE },
- { 0xFCA1, 0xFCFE },
- { 0xFDA1, 0xFDFE },
-};
-
-// generated from http://unicode.org/Public/MAPPINGS/VENDORS/MICSFT/WINDOWS/CP950.TXT
-static const CharRange kBig5Ranges[] = {
- { 0xA140, 0xA17E },
- { 0xA1A1, 0xA1FE },
- { 0xA240, 0xA27E },
- { 0xA2A1, 0xA2FE },
- { 0xA340, 0xA37E },
- { 0xA3A1, 0xA3BF },
- { 0xA3E1, 0xA3E1 },
- { 0xA440, 0xA47E },
- { 0xA4A1, 0xA4FE },
- { 0xA540, 0xA57E },
- { 0xA5A1, 0xA5FE },
- { 0xA640, 0xA67E },
- { 0xA6A1, 0xA6FE },
- { 0xA740, 0xA77E },
- { 0xA7A1, 0xA7FE },
- { 0xA840, 0xA87E },
- { 0xA8A1, 0xA8FE },
- { 0xA940, 0xA97E },
- { 0xA9A1, 0xA9FE },
- { 0xAA40, 0xAA7E },
- { 0xAAA1, 0xAAFE },
- { 0xAB40, 0xAB7E },
- { 0xABA1, 0xABFE },
- { 0xAC40, 0xAC7E },
- { 0xACA1, 0xACFE },
- { 0xAD40, 0xAD7E },
- { 0xADA1, 0xADFE },
- { 0xAE40, 0xAE7E },
- { 0xAEA1, 0xAEFE },
- { 0xAF40, 0xAF7E },
- { 0xAFA1, 0xAFFE },
- { 0xB040, 0xB07E },
- { 0xB0A1, 0xB0FE },
- { 0xB140, 0xB17E },
- { 0xB1A1, 0xB1FE },
- { 0xB240, 0xB27E },
- { 0xB2A1, 0xB2FE },
- { 0xB340, 0xB37E },
- { 0xB3A1, 0xB3FE },
- { 0xB440, 0xB47E },
- { 0xB4A1, 0xB4FE },
- { 0xB540, 0xB57E },
- { 0xB5A1, 0xB5FE },
- { 0xB640, 0xB67E },
- { 0xB6A1, 0xB6FE },
- { 0xB740, 0xB77E },
- { 0xB7A1, 0xB7FE },
- { 0xB840, 0xB87E },
- { 0xB8A1, 0xB8FE },
- { 0xB940, 0xB97E },
- { 0xB9A1, 0xB9FE },
- { 0xBA40, 0xBA7E },
- { 0xBAA1, 0xBAFE },
- { 0xBB40, 0xBB7E },
- { 0xBBA1, 0xBBFE },
- { 0xBC40, 0xBC7E },
- { 0xBCA1, 0xBCFE },
- { 0xBD40, 0xBD7E },
- { 0xBDA1, 0xBDFE },
- { 0xBE40, 0xBE7E },
- { 0xBEA1, 0xBEFE },
- { 0xBF40, 0xBF7E },
- { 0xBFA1, 0xBFFE },
- { 0xC040, 0xC07E },
- { 0xC0A1, 0xC0FE },
- { 0xC140, 0xC17E },
- { 0xC1A1, 0xC1FE },
- { 0xC240, 0xC27E },
- { 0xC2A1, 0xC2FE },
- { 0xC340, 0xC37E },
- { 0xC3A1, 0xC3FE },
- { 0xC440, 0xC47E },
- { 0xC4A1, 0xC4FE },
- { 0xC540, 0xC57E },
- { 0xC5A1, 0xC5FE },
- { 0xC640, 0xC67E },
- { 0xC940, 0xC97E },
- { 0xC9A1, 0xC9FE },
- { 0xCA40, 0xCA7E },
- { 0xCAA1, 0xCAFE },
- { 0xCB40, 0xCB7E },
- { 0xCBA1, 0xCBFE },
- { 0xCC40, 0xCC7E },
- { 0xCCA1, 0xCCFE },
- { 0xCD40, 0xCD7E },
- { 0xCDA1, 0xCDFE },
- { 0xCE40, 0xCE7E },
- { 0xCEA1, 0xCEFE },
- { 0xCF40, 0xCF7E },
- { 0xCFA1, 0xCFFE },
- { 0xD040, 0xD07E },
- { 0xD0A1, 0xD0FE },
- { 0xD140, 0xD17E },
- { 0xD1A1, 0xD1FE },
- { 0xD240, 0xD27E },
- { 0xD2A1, 0xD2FE },
- { 0xD340, 0xD37E },
- { 0xD3A1, 0xD3FE },
- { 0xD440, 0xD47E },
- { 0xD4A1, 0xD4FE },
- { 0xD540, 0xD57E },
- { 0xD5A1, 0xD5FE },
- { 0xD640, 0xD67E },
- { 0xD6A1, 0xD6FE },
- { 0xD740, 0xD77E },
- { 0xD7A1, 0xD7FE },
- { 0xD840, 0xD87E },
- { 0xD8A1, 0xD8FE },
- { 0xD940, 0xD97E },
- { 0xD9A1, 0xD9FE },
- { 0xDA40, 0xDA7E },
- { 0xDAA1, 0xDAFE },
- { 0xDB40, 0xDB7E },
- { 0xDBA1, 0xDBFE },
- { 0xDC40, 0xDC7E },
- { 0xDCA1, 0xDCFE },
- { 0xDD40, 0xDD7E },
- { 0xDDA1, 0xDDFE },
- { 0xDE40, 0xDE7E },
- { 0xDEA1, 0xDEFE },
- { 0xDF40, 0xDF7E },
- { 0xDFA1, 0xDFFE },
- { 0xE040, 0xE07E },
- { 0xE0A1, 0xE0FE },
- { 0xE140, 0xE17E },
- { 0xE1A1, 0xE1FE },
- { 0xE240, 0xE27E },
- { 0xE2A1, 0xE2FE },
- { 0xE340, 0xE37E },
- { 0xE3A1, 0xE3FE },
- { 0xE440, 0xE47E },
- { 0xE4A1, 0xE4FE },
- { 0xE540, 0xE57E },
- { 0xE5A1, 0xE5FE },
- { 0xE640, 0xE67E },
- { 0xE6A1, 0xE6FE },
- { 0xE740, 0xE77E },
- { 0xE7A1, 0xE7FE },
- { 0xE840, 0xE87E },
- { 0xE8A1, 0xE8FE },
- { 0xE940, 0xE97E },
- { 0xE9A1, 0xE9FE },
- { 0xEA40, 0xEA7E },
- { 0xEAA1, 0xEAFE },
- { 0xEB40, 0xEB7E },
- { 0xEBA1, 0xEBFE },
- { 0xEC40, 0xEC7E },
- { 0xECA1, 0xECFE },
- { 0xED40, 0xED7E },
- { 0xEDA1, 0xEDFE },
- { 0xEE40, 0xEE7E },
- { 0xEEA1, 0xEEFE },
- { 0xEF40, 0xEF7E },
- { 0xEFA1, 0xEFFE },
- { 0xF040, 0xF07E },
- { 0xF0A1, 0xF0FE },
- { 0xF140, 0xF17E },
- { 0xF1A1, 0xF1FE },
- { 0xF240, 0xF27E },
- { 0xF2A1, 0xF2FE },
- { 0xF340, 0xF37E },
- { 0xF3A1, 0xF3FE },
- { 0xF440, 0xF47E },
- { 0xF4A1, 0xF4FE },
- { 0xF540, 0xF57E },
- { 0xF5A1, 0xF5FE },
- { 0xF640, 0xF67E },
- { 0xF6A1, 0xF6FE },
- { 0xF740, 0xF77E },
- { 0xF7A1, 0xF7FE },
- { 0xF840, 0xF87E },
- { 0xF8A1, 0xF8FE },
- { 0xF940, 0xF97E },
- { 0xF9A1, 0xF9FE },
-};
-
-static bool charMatchesEncoding(int ch, const CharRange* encodingRanges, int rangeCount) {
- // Use binary search to see if the character is contained in the encoding
- int low = 0;
- int high = rangeCount;
-
- while (low < high) {
- int i = (low + high) / 2;
- const CharRange* range = &encodingRanges[i];
- if (ch >= range->first && ch <= range->last)
- return true;
- if (ch > range->last)
- low = i + 1;
- else
- high = i;
- }
-
- return false;
-}
-
-extern uint32_t findPossibleEncodings(int ch)
-{
- // ASCII matches everything
- if (ch < 256) return kEncodingAll;
-
- int result = kEncodingNone;
-
- if (charMatchesEncoding(ch, kShiftJISRanges, ARRAY_SIZE(kShiftJISRanges)))
- result |= kEncodingShiftJIS;
- if (charMatchesEncoding(ch, kGBKRanges, ARRAY_SIZE(kGBKRanges)))
- result |= kEncodingGBK;
- if (charMatchesEncoding(ch, kBig5Ranges, ARRAY_SIZE(kBig5Ranges)))
- result |= kEncodingBig5;
- if (charMatchesEncoding(ch, kEUCKRRanges, ARRAY_SIZE(kEUCKRRanges)))
- result |= kEncodingEUCKR;
-
- return result;
-}
diff --git a/media/libmedia/autodetect.h b/media/libmedia/autodetect.h
deleted file mode 100644
index 9675db3..0000000
--- a/media/libmedia/autodetect.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AUTODETECT_H
-#define AUTODETECT_H
-
-#include <inttypes.h>
-
-// flags used for native encoding detection
-enum {
- kEncodingNone = 0,
- kEncodingShiftJIS = (1 << 0),
- kEncodingGBK = (1 << 1),
- kEncodingBig5 = (1 << 2),
- kEncodingEUCKR = (1 << 3),
-
- kEncodingAll = (kEncodingShiftJIS | kEncodingGBK | kEncodingBig5 | kEncodingEUCKR),
-};
-
-
-// returns a bitfield containing the possible native encodings for the given character
-extern uint32_t findPossibleEncodings(int ch);
-
-#endif // AUTODETECT_H
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 110b94c..1d6bb6f 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -21,6 +21,7 @@
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
#include <media/mediametadataretriever.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <utils/Log.h>
#include <dlfcn.h>
@@ -93,7 +94,9 @@
}
status_t MediaMetadataRetriever::setDataSource(
- const char *srcUrl, const KeyedVector<String8, String8> *headers)
+ const sp<IMediaHTTPService> &httpService,
+ const char *srcUrl,
+ const KeyedVector<String8, String8> *headers)
{
ALOGV("setDataSource");
Mutex::Autolock _l(mLock);
@@ -106,7 +109,7 @@
return UNKNOWN_ERROR;
}
ALOGV("data source (%s)", srcUrl);
- return mRetriever->setDataSource(srcUrl, headers);
+ return mRetriever->setDataSource(httpService, srcUrl, headers);
}
status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
@@ -157,7 +160,7 @@
return mRetriever->extractAlbumArt();
}
-void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who) {
+void MediaMetadataRetriever::DeathNotifier::binderDied(const wp<IBinder>& who __unused) {
Mutex::Autolock lock(MediaMetadataRetriever::sServiceLock);
MediaMetadataRetriever::sService.clear();
ALOGW("MediaMetadataRetriever server died!");
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 0f6d897..24663ad 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -136,6 +136,7 @@
}
status_t MediaPlayer::setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url, const KeyedVector<String8, String8> *headers)
{
ALOGV("setDataSource(%s)", url);
@@ -145,7 +146,7 @@
if (service != 0) {
sp<IMediaPlayer> player(service->create(this, mAudioSessionId));
if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
- (NO_ERROR != player->setDataSource(url, headers))) {
+ (NO_ERROR != player->setDataSource(httpService, url, headers))) {
player.clear();
}
err = attachNewPlayer(player);
@@ -654,7 +655,7 @@
return BAD_VALUE;
}
- memset(&mRetransmitEndpoint, 0, sizeof(&mRetransmitEndpoint));
+ memset(&mRetransmitEndpoint, 0, sizeof(mRetransmitEndpoint));
mRetransmitEndpoint.sin_family = AF_INET;
mRetransmitEndpoint.sin_addr = saddr;
mRetransmitEndpoint.sin_port = htons(port);
@@ -776,15 +777,20 @@
}
}
-/*static*/ status_t MediaPlayer::decode(const char* url, uint32_t *pSampleRate,
- int* pNumChannels, audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
+/*static*/ status_t MediaPlayer::decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize)
{
ALOGV("decode(%s)", url);
status_t status;
const sp<IMediaPlayerService>& service = getMediaPlayerService();
if (service != 0) {
- status = service->decode(url, pSampleRate, pNumChannels, pFormat, heap, pSize);
+ status = service->decode(httpService, url, pSampleRate, pNumChannels, pFormat, heap, pSize);
} else {
ALOGE("Unable to locate media service");
status = DEAD_OBJECT;
@@ -832,15 +838,4 @@
return mPlayer->setNextPlayer(next == NULL ? NULL : next->mPlayer);
}
-status_t MediaPlayer::updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- const sp<IMediaPlayerService>& service = getMediaPlayerService();
-
- if (service != NULL) {
- return service->updateProxyConfig(host, port, exclusionList);
- }
-
- return INVALID_OPERATION;
-}
-
}; // namespace android
diff --git a/media/libmediaplayerservice/HDCP.cpp b/media/libmediaplayerservice/HDCP.cpp
index c2ac1a3..afe3936 100644
--- a/media/libmediaplayerservice/HDCP.cpp
+++ b/media/libmediaplayerservice/HDCP.cpp
@@ -107,11 +107,7 @@
return NO_INIT;
}
- // TO-DO:
- // Only support HDCP_CAPS_ENCRYPT (byte-array to byte-array) for now.
- // use mHDCPModule->getCaps() when the HDCP libraries get updated.
- //return mHDCPModule->getCaps();
- return HDCPModule::HDCP_CAPS_ENCRYPT;
+ return mHDCPModule->getCaps();
}
status_t HDCP::encrypt(
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index a392b76..1f70620 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -44,6 +44,7 @@
#include <utils/SystemClock.h>
#include <utils/Vector.h>
+#include <media/IMediaHTTPService.h>
#include <media/IRemoteDisplay.h>
#include <media/IRemoteDisplayClient.h>
#include <media/MediaPlayerInterface.h>
@@ -306,11 +307,6 @@
return new RemoteDisplay(client, iface.string());
}
-status_t MediaPlayerService::updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- return HTTPBase::UpdateProxyConfig(host, port, exclusionList);
-}
-
status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
{
const size_t SIZE = 256;
@@ -622,7 +618,9 @@
}
status_t MediaPlayerService::Client::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers)
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers)
{
ALOGV("setDataSource(%s)", url);
if (url == NULL)
@@ -657,7 +655,7 @@
return NO_INIT;
}
- setDataSource_post(p, p->setDataSource(url, headers));
+ setDataSource_post(p, p->setDataSource(httpService, url, headers));
return mStatus;
}
}
@@ -1176,9 +1174,14 @@
}
#endif
-status_t MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
+status_t MediaPlayerService::decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize)
{
ALOGV("decode(%s)", url);
sp<MediaPlayerBase> player;
@@ -1206,7 +1209,7 @@
static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
// set data source
- if (player->setDataSource(url) != NO_ERROR) goto Exit;
+ if (player->setDataSource(httpService, url) != NO_ERROR) goto Exit;
ALOGV("prepare");
player->prepareAsync();
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 9c084e1..fc355b0 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -211,12 +211,12 @@
virtual void flush() {}
virtual void pause() {}
virtual void close() {}
- void setAudioStreamType(audio_stream_type_t streamType) {}
+ void setAudioStreamType(audio_stream_type_t streamType __unused) {}
// stream type is not used for AudioCache
virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
- void setVolume(float left, float right) {}
- virtual status_t setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
+ void setVolume(float left __unused, float right __unused) {}
+ virtual status_t setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
uint32_t sampleRate() const { return mSampleRate; }
audio_format_t format() const { return mFormat; }
size_t size() const { return mSize; }
@@ -256,9 +256,15 @@
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId);
- virtual status_t decode(const char* url, uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize);
+ virtual status_t decode(
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ uint32_t *pSampleRate,
+ int* pNumChannels,
+ audio_format_t* pFormat,
+ const sp<IMemoryHeap>& heap,
+ size_t *pSize);
+
virtual status_t decode(int fd, int64_t offset, int64_t length,
uint32_t *pSampleRate, int* pNumChannels,
audio_format_t* pFormat,
@@ -272,9 +278,6 @@
const String8& iface);
virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
void removeClient(wp<Client> client);
// For battery usage tracking purpose
@@ -356,6 +359,7 @@
sp<MediaPlayerBase> createPlayer(player_type playerType);
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers);
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 348957f..c61cf89 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -31,6 +31,7 @@
#include <binder/MemoryHeapBase.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
+#include <media/IMediaHTTPService.h>
#include <media/MediaMetadataRetrieverInterface.h>
#include <media/MediaPlayerInterface.h>
#include <private/media/VideoFrame.h>
@@ -106,7 +107,9 @@
}
status_t MetadataRetrieverClient::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers)
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers)
{
ALOGV("setDataSource(%s)", url);
Mutex::Autolock lock(mLock);
@@ -127,7 +130,7 @@
ALOGV("player type = %d", playerType);
sp<MediaMetadataRetrieverBase> p = createRetriever(playerType);
if (p == NULL) return NO_INIT;
- status_t ret = p->setDataSource(url, headers);
+ status_t ret = p->setDataSource(httpService, url, headers);
if (ret == NO_ERROR) mRetriever = p;
return ret;
}
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.h b/media/libmediaplayerservice/MetadataRetrieverClient.h
index f08f933..9d3fbe9 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.h
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.h
@@ -30,6 +30,7 @@
namespace android {
+struct IMediaHTTPService;
class IMediaPlayerService;
class MemoryDealer;
@@ -43,7 +44,9 @@
virtual void disconnect();
virtual status_t setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual sp<IMemory> getFrameAtTime(int64_t timeUs, int option);
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index 0a6aa90..deeddd1 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,9 @@
}
status_t MidiFile::setDataSource(
- const char* path, const KeyedVector<String8, String8> *) {
+ const sp<IMediaHTTPService> &httpService,
+ const char* path,
+ const KeyedVector<String8, String8> *) {
ALOGV("MidiFile::setDataSource url=%s", path);
Mutex::Autolock lock(mMutex);
diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 24d59b4..12802ba 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -32,7 +32,9 @@
virtual status_t initCheck();
virtual status_t setDataSource(
- const char* path, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char* path,
+ const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual status_t setVideoSurfaceTexture(
diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.cpp b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
index 465209f..f3cf6ef 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.cpp
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.cpp
@@ -22,6 +22,8 @@
#include "MidiMetadataRetriever.h"
#include <media/mediametadataretriever.h>
+#include <media/IMediaHTTPService.h>
+
namespace android {
static status_t ERROR_NOT_OPEN = -1;
@@ -36,7 +38,9 @@
}
status_t MidiMetadataRetriever::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers)
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers)
{
ALOGV("setDataSource: %s", url? url: "NULL pointer");
Mutex::Autolock lock(mLock);
@@ -44,7 +48,7 @@
if (mMidiPlayer == 0) {
mMidiPlayer = new MidiFile();
}
- return mMidiPlayer->setDataSource(url, headers);
+ return mMidiPlayer->setDataSource(httpService, url, headers);
}
status_t MidiMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
diff --git a/media/libmediaplayerservice/MidiMetadataRetriever.h b/media/libmediaplayerservice/MidiMetadataRetriever.h
index 4cee42d..b8214ee 100644
--- a/media/libmediaplayerservice/MidiMetadataRetriever.h
+++ b/media/libmediaplayerservice/MidiMetadataRetriever.h
@@ -32,7 +32,9 @@
~MidiMetadataRetriever() {}
virtual status_t setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual const char* extractMetadata(int keyCode);
diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index de61d9b..b19e8bf 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -54,8 +54,10 @@
}
status_t StagefrightPlayer::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers) {
- return mPlayer->setDataSource(url, headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ return mPlayer->setDataSource(httpService, url, headers);
}
// Warning: The filedescriptor passed into this method will only be valid until
diff --git a/media/libmediaplayerservice/StagefrightPlayer.h b/media/libmediaplayerservice/StagefrightPlayer.h
index 600945e..e6c30ff 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.h
+++ b/media/libmediaplayerservice/StagefrightPlayer.h
@@ -34,7 +34,9 @@
virtual status_t setUID(uid_t uid);
virtual status_t setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 4da74e1..e0eae37 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -25,8 +25,10 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
-#include <media/openmax/OMX_Audio.h>
+#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AACWriter.h>
@@ -36,13 +38,12 @@
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaCodecSource.h>
#include <media/stagefright/OMXClient.h>
#include <media/stagefright/OMXCodec.h>
-#include <media/stagefright/SurfaceMediaSource.h>
#include <media/MediaProfiles.h>
#include <camera/ICamera.h>
#include <camera/CameraParameters.h>
-#include <gui/Surface.h>
#include <utils/Errors.h>
#include <sys/types.h>
@@ -72,8 +73,7 @@
mAudioSource(AUDIO_SOURCE_CNT),
mVideoSource(VIDEO_SOURCE_LIST_END),
mCaptureTimeLapse(false),
- mStarted(false),
- mSurfaceMediaSource(NULL) {
+ mStarted(false) {
ALOGV("Constructor");
reset();
@@ -82,10 +82,19 @@
StagefrightRecorder::~StagefrightRecorder() {
ALOGV("Destructor");
stop();
+
+ if (mLooper != NULL) {
+ mLooper->stop();
+ }
}
status_t StagefrightRecorder::init() {
ALOGV("init");
+
+ mLooper = new ALooper;
+ mLooper->setName("recorder_looper");
+ mLooper->start();
+
return OK;
}
@@ -94,7 +103,7 @@
// while encoding GL Frames
sp<IGraphicBufferProducer> StagefrightRecorder::querySurfaceMediaSource() const {
ALOGV("Get SurfaceMediaSource");
- return mSurfaceMediaSource->getBufferQueue();
+ return mGraphicBufferProducer;
}
status_t StagefrightRecorder::setAudioSource(audio_source_t as) {
@@ -740,15 +749,61 @@
}
status_t StagefrightRecorder::prepare() {
- return OK;
-}
-
-status_t StagefrightRecorder::start() {
- CHECK_GE(mOutputFd, 0);
+ ALOGV("prepare");
+ if (mOutputFd < 0) {
+ ALOGE("Output file descriptor is invalid");
+ return INVALID_OPERATION;
+ }
// Get UID here for permission checking
mClientUid = IPCThreadState::self()->getCallingUid();
- if (mWriter != NULL) {
+
+ status_t status = OK;
+
+ switch (mOutputFormat) {
+ case OUTPUT_FORMAT_DEFAULT:
+ case OUTPUT_FORMAT_THREE_GPP:
+ case OUTPUT_FORMAT_MPEG_4:
+ status = setupMPEG4Recording();
+ break;
+
+ case OUTPUT_FORMAT_AMR_NB:
+ case OUTPUT_FORMAT_AMR_WB:
+ status = setupAMRRecording();
+ break;
+
+ case OUTPUT_FORMAT_AAC_ADIF:
+ case OUTPUT_FORMAT_AAC_ADTS:
+ status = setupAACRecording();
+ break;
+
+ case OUTPUT_FORMAT_RTP_AVP:
+ status = setupRTPRecording();
+ break;
+
+ case OUTPUT_FORMAT_MPEG2TS:
+ status = setupMPEG2TSRecording();
+ break;
+
+ default:
+ ALOGE("Unsupported output file format: %d", mOutputFormat);
+ status = UNKNOWN_ERROR;
+ break;
+ }
+
+ return status;
+}
+
+status_t StagefrightRecorder::start() {
+ ALOGV("start");
+ if (mOutputFd < 0) {
+ ALOGE("Output file descriptor is invalid");
+ return INVALID_OPERATION;
+ }
+
+ // Get UID here for permission checking
+ mClientUid = IPCThreadState::self()->getCallingUid();
+ if (mWriter == NULL) {
ALOGE("File writer is not avaialble");
return UNKNOWN_ERROR;
}
@@ -759,31 +814,35 @@
case OUTPUT_FORMAT_DEFAULT:
case OUTPUT_FORMAT_THREE_GPP:
case OUTPUT_FORMAT_MPEG_4:
- status = startMPEG4Recording();
+ {
+ sp<MetaData> meta = new MetaData;
+ setupMPEG4MetaData(&meta);
+ status = mWriter->start(meta.get());
break;
+ }
case OUTPUT_FORMAT_AMR_NB:
case OUTPUT_FORMAT_AMR_WB:
- status = startAMRRecording();
- break;
-
case OUTPUT_FORMAT_AAC_ADIF:
case OUTPUT_FORMAT_AAC_ADTS:
- status = startAACRecording();
- break;
-
case OUTPUT_FORMAT_RTP_AVP:
- status = startRTPRecording();
- break;
-
case OUTPUT_FORMAT_MPEG2TS:
- status = startMPEG2TSRecording();
+ {
+ status = mWriter->start();
break;
+ }
default:
+ {
ALOGE("Unsupported output file format: %d", mOutputFormat);
status = UNKNOWN_ERROR;
break;
+ }
+ }
+
+ if (status != OK) {
+ mWriter.clear();
+ mWriter = NULL;
}
if ((status == OK) && (!mStarted)) {
@@ -817,58 +876,54 @@
return NULL;
}
- sp<MetaData> encMeta = new MetaData;
+ sp<AMessage> format = new AMessage;
const char *mime;
switch (mAudioEncoder) {
case AUDIO_ENCODER_AMR_NB:
case AUDIO_ENCODER_DEFAULT:
- mime = MEDIA_MIMETYPE_AUDIO_AMR_NB;
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_NB);
break;
case AUDIO_ENCODER_AMR_WB:
- mime = MEDIA_MIMETYPE_AUDIO_AMR_WB;
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_AMR_WB);
break;
case AUDIO_ENCODER_AAC:
- mime = MEDIA_MIMETYPE_AUDIO_AAC;
- encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectLC);
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+ format->setInt32("aac-profile", OMX_AUDIO_AACObjectLC);
break;
case AUDIO_ENCODER_HE_AAC:
- mime = MEDIA_MIMETYPE_AUDIO_AAC;
- encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectHE);
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+ format->setInt32("aac-profile", OMX_AUDIO_AACObjectHE);
break;
case AUDIO_ENCODER_AAC_ELD:
- mime = MEDIA_MIMETYPE_AUDIO_AAC;
- encMeta->setInt32(kKeyAACProfile, OMX_AUDIO_AACObjectELD);
+ format->setString("mime", MEDIA_MIMETYPE_AUDIO_AAC);
+ format->setInt32("aac-profile", OMX_AUDIO_AACObjectELD);
break;
default:
ALOGE("Unknown audio encoder: %d", mAudioEncoder);
return NULL;
}
- encMeta->setCString(kKeyMIMEType, mime);
int32_t maxInputSize;
CHECK(audioSource->getFormat()->findInt32(
kKeyMaxInputSize, &maxInputSize));
- encMeta->setInt32(kKeyMaxInputSize, maxInputSize);
- encMeta->setInt32(kKeyChannelCount, mAudioChannels);
- encMeta->setInt32(kKeySampleRate, mSampleRate);
- encMeta->setInt32(kKeyBitRate, mAudioBitRate);
+ format->setInt32("max-input-size", maxInputSize);
+ format->setInt32("channel-count", mAudioChannels);
+ format->setInt32("sample-rate", mSampleRate);
+ format->setInt32("bitrate", mAudioBitRate);
if (mAudioTimeScale > 0) {
- encMeta->setInt32(kKeyTimeScale, mAudioTimeScale);
+ format->setInt32("time-scale", mAudioTimeScale);
}
- OMXClient client;
- CHECK_EQ(client.connect(), (status_t)OK);
sp<MediaSource> audioEncoder =
- OMXCodec::Create(client.interface(), encMeta,
- true /* createEncoder */, audioSource);
+ MediaCodecSource::Create(mLooper, format, audioSource);
mAudioSourceNode = audioSource;
return audioEncoder;
}
-status_t StagefrightRecorder::startAACRecording() {
+status_t StagefrightRecorder::setupAACRecording() {
// FIXME:
// Add support for OUTPUT_FORMAT_AAC_ADIF
CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
@@ -879,16 +934,10 @@
CHECK(mAudioSource != AUDIO_SOURCE_CNT);
mWriter = new AACWriter(mOutputFd);
- status_t status = startRawAudioRecording();
- if (status != OK) {
- mWriter.clear();
- mWriter = NULL;
- }
-
- return status;
+ return setupRawAudioRecording();
}
-status_t StagefrightRecorder::startAMRRecording() {
+status_t StagefrightRecorder::setupAMRRecording() {
CHECK(mOutputFormat == OUTPUT_FORMAT_AMR_NB ||
mOutputFormat == OUTPUT_FORMAT_AMR_WB);
@@ -908,15 +957,10 @@
}
mWriter = new AMRWriter(mOutputFd);
- status_t status = startRawAudioRecording();
- if (status != OK) {
- mWriter.clear();
- mWriter = NULL;
- }
- return status;
+ return setupRawAudioRecording();
}
-status_t StagefrightRecorder::startRawAudioRecording() {
+status_t StagefrightRecorder::setupRawAudioRecording() {
if (mAudioSource >= AUDIO_SOURCE_CNT) {
ALOGE("Invalid audio source: %d", mAudioSource);
return BAD_VALUE;
@@ -942,12 +986,11 @@
mWriter->setMaxFileSize(mMaxFileSizeBytes);
}
mWriter->setListener(mListener);
- mWriter->start();
return OK;
}
-status_t StagefrightRecorder::startRTPRecording() {
+status_t StagefrightRecorder::setupRTPRecording() {
CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP);
if ((mAudioSource != AUDIO_SOURCE_CNT
@@ -974,7 +1017,7 @@
return err;
}
- err = setupVideoEncoder(mediaSource, mVideoBitRate, &source);
+ err = setupVideoEncoder(mediaSource, &source);
if (err != OK) {
return err;
}
@@ -984,10 +1027,10 @@
mWriter->addSource(source);
mWriter->setListener(mListener);
- return mWriter->start();
+ return OK;
}
-status_t StagefrightRecorder::startMPEG2TSRecording() {
+status_t StagefrightRecorder::setupMPEG2TSRecording() {
CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_MPEG2TS);
sp<MediaWriter> writer = new MPEG2TSWriter(mOutputFd);
@@ -1018,7 +1061,7 @@
}
sp<MediaSource> encoder;
- err = setupVideoEncoder(mediaSource, mVideoBitRate, &encoder);
+ err = setupVideoEncoder(mediaSource, &encoder);
if (err != OK) {
return err;
@@ -1037,7 +1080,7 @@
mWriter = writer;
- return mWriter->start();
+ return OK;
}
void StagefrightRecorder::clipVideoFrameRate() {
@@ -1278,49 +1321,14 @@
return err;
}
*mediaSource = cameraSource;
- } else if (mVideoSource == VIDEO_SOURCE_GRALLOC_BUFFER) {
- // If using GRAlloc buffers, setup surfacemediasource.
- // Later a handle to that will be passed
- // to the client side when queried
- status_t err = setupSurfaceMediaSource();
- if (err != OK) {
- return err;
- }
- *mediaSource = mSurfaceMediaSource;
+ } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+ *mediaSource = NULL;
} else {
return INVALID_OPERATION;
}
return OK;
}
-// setupSurfaceMediaSource creates a source with the given
-// width and height and framerate.
-// TODO: This could go in a static function inside SurfaceMediaSource
-// similar to that in CameraSource
-status_t StagefrightRecorder::setupSurfaceMediaSource() {
- status_t err = OK;
- mSurfaceMediaSource = new SurfaceMediaSource(mVideoWidth, mVideoHeight);
- if (mSurfaceMediaSource == NULL) {
- return NO_INIT;
- }
-
- if (mFrameRate == -1) {
- int32_t frameRate = 0;
- CHECK (mSurfaceMediaSource->getFormat()->findInt32(
- kKeyFrameRate, &frameRate));
- ALOGI("Frame rate is not explicitly set. Use the current frame "
- "rate (%d fps)", frameRate);
- mFrameRate = frameRate;
- } else {
- err = mSurfaceMediaSource->setFrameRate(mFrameRate);
- }
- CHECK(mFrameRate != -1);
-
- mIsMetaDataStoredInVideoBuffers =
- mSurfaceMediaSource->isMetaDataStoredInVideoBuffers();
- return err;
-}
-
status_t StagefrightRecorder::setupCameraSource(
sp<CameraSource> *cameraSource) {
status_t err = OK;
@@ -1384,25 +1392,22 @@
status_t StagefrightRecorder::setupVideoEncoder(
sp<MediaSource> cameraSource,
- int32_t videoBitRate,
sp<MediaSource> *source) {
source->clear();
- sp<MetaData> enc_meta = new MetaData;
- enc_meta->setInt32(kKeyBitRate, videoBitRate);
- enc_meta->setInt32(kKeyFrameRate, mFrameRate);
+ sp<AMessage> format = new AMessage();
switch (mVideoEncoder) {
case VIDEO_ENCODER_H263:
- enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_H263);
+ format->setString("mime", MEDIA_MIMETYPE_VIDEO_H263);
break;
case VIDEO_ENCODER_MPEG_4_SP:
- enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+ format->setString("mime", MEDIA_MIMETYPE_VIDEO_MPEG4);
break;
case VIDEO_ENCODER_H264:
- enc_meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+ format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC);
break;
default:
@@ -1410,59 +1415,69 @@
break;
}
- sp<MetaData> meta = cameraSource->getFormat();
+ if (cameraSource != NULL) {
+ sp<MetaData> meta = cameraSource->getFormat();
- int32_t width, height, stride, sliceHeight, colorFormat;
- CHECK(meta->findInt32(kKeyWidth, &width));
- CHECK(meta->findInt32(kKeyHeight, &height));
- CHECK(meta->findInt32(kKeyStride, &stride));
- CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
- CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
+ int32_t width, height, stride, sliceHeight, colorFormat;
+ CHECK(meta->findInt32(kKeyWidth, &width));
+ CHECK(meta->findInt32(kKeyHeight, &height));
+ CHECK(meta->findInt32(kKeyStride, &stride));
+ CHECK(meta->findInt32(kKeySliceHeight, &sliceHeight));
+ CHECK(meta->findInt32(kKeyColorFormat, &colorFormat));
- enc_meta->setInt32(kKeyWidth, width);
- enc_meta->setInt32(kKeyHeight, height);
- enc_meta->setInt32(kKeyIFramesInterval, mIFramesIntervalSec);
- enc_meta->setInt32(kKeyStride, stride);
- enc_meta->setInt32(kKeySliceHeight, sliceHeight);
- enc_meta->setInt32(kKeyColorFormat, colorFormat);
+ format->setInt32("width", width);
+ format->setInt32("height", height);
+ format->setInt32("stride", stride);
+ format->setInt32("slice-height", sliceHeight);
+ format->setInt32("color-format", colorFormat);
+ } else {
+ format->setInt32("width", mVideoWidth);
+ format->setInt32("height", mVideoHeight);
+ format->setInt32("stride", mVideoWidth);
+ format->setInt32("slice-height", mVideoWidth);
+ format->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
+ }
+
+ format->setInt32("bitrate", mVideoBitRate);
+ format->setInt32("frame-rate", mFrameRate);
+ format->setInt32("i-frame-interval", mIFramesIntervalSec);
+
if (mVideoTimeScale > 0) {
- enc_meta->setInt32(kKeyTimeScale, mVideoTimeScale);
+ format->setInt32("time-scale", mVideoTimeScale);
}
if (mVideoEncoderProfile != -1) {
- enc_meta->setInt32(kKeyVideoProfile, mVideoEncoderProfile);
+ format->setInt32("profile", mVideoEncoderProfile);
}
if (mVideoEncoderLevel != -1) {
- enc_meta->setInt32(kKeyVideoLevel, mVideoEncoderLevel);
+ format->setInt32("level", mVideoEncoderLevel);
}
- OMXClient client;
- CHECK_EQ(client.connect(), (status_t)OK);
-
- uint32_t encoder_flags = 0;
+ uint32_t flags = 0;
if (mIsMetaDataStoredInVideoBuffers) {
- encoder_flags |= OMXCodec::kStoreMetaDataInVideoBuffers;
+ flags |= MediaCodecSource::FLAG_USE_METADATA_INPUT;
}
- // Do not wait for all the input buffers to become available.
- // This give timelapse video recording faster response in
- // receiving output from video encoder component.
- if (mCaptureTimeLapse) {
- encoder_flags |= OMXCodec::kOnlySubmitOneInputBufferAtOneTime;
+ if (cameraSource == NULL) {
+ flags |= MediaCodecSource::FLAG_USE_SURFACE_INPUT;
}
- sp<MediaSource> encoder = OMXCodec::Create(
- client.interface(), enc_meta,
- true /* createEncoder */, cameraSource,
- NULL, encoder_flags);
+ sp<MediaCodecSource> encoder =
+ MediaCodecSource::Create(mLooper, format, cameraSource, flags);
if (encoder == NULL) {
ALOGW("Failed to create the encoder");
// When the encoder fails to be created, we need
// release the camera source due to the camera's lock
// and unlock mechanism.
- cameraSource->stop();
+ if (cameraSource != NULL) {
+ cameraSource->stop();
+ }
return UNKNOWN_ERROR;
}
+ if (cameraSource == NULL) {
+ mGraphicBufferProducer = encoder->getGraphicBufferProducer();
+ }
+
*source = encoder;
return OK;
@@ -1496,16 +1511,12 @@
return OK;
}
-status_t StagefrightRecorder::setupMPEG4Recording(
- int outputFd,
- int32_t videoWidth, int32_t videoHeight,
- int32_t videoBitRate,
- int32_t *totalBitRate,
- sp<MediaWriter> *mediaWriter) {
- mediaWriter->clear();
- *totalBitRate = 0;
+status_t StagefrightRecorder::setupMPEG4Recording() {
+ mWriter.clear();
+ mTotalBitRate = 0;
+
status_t err = OK;
- sp<MediaWriter> writer = new MPEG4Writer(outputFd);
+ sp<MediaWriter> writer = new MPEG4Writer(mOutputFd);
if (mVideoSource < VIDEO_SOURCE_LIST_END) {
@@ -1516,13 +1527,13 @@
}
sp<MediaSource> encoder;
- err = setupVideoEncoder(mediaSource, videoBitRate, &encoder);
+ err = setupVideoEncoder(mediaSource, &encoder);
if (err != OK) {
return err;
}
writer->addSource(encoder);
- *totalBitRate += videoBitRate;
+ mTotalBitRate += mVideoBitRate;
}
// Audio source is added at the end if it exists.
@@ -1531,7 +1542,7 @@
if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_CNT)) {
err = setupAudioEncoder(writer);
if (err != OK) return err;
- *totalBitRate += mAudioBitRate;
+ mTotalBitRate += mAudioBitRate;
}
if (mInterleaveDurationUs > 0) {
@@ -1549,22 +1560,28 @@
writer->setMaxFileSize(mMaxFileSizeBytes);
}
- mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+ if (mVideoSource == VIDEO_SOURCE_DEFAULT
+ || mVideoSource == VIDEO_SOURCE_CAMERA) {
+ mStartTimeOffsetMs = mEncoderProfiles->getStartTimeOffsetMs(mCameraId);
+ } else if (mVideoSource == VIDEO_SOURCE_SURFACE) {
+ // surface source doesn't need large initial delay
+ mStartTimeOffsetMs = 200;
+ }
if (mStartTimeOffsetMs > 0) {
reinterpret_cast<MPEG4Writer *>(writer.get())->
setStartTimeOffsetMs(mStartTimeOffsetMs);
}
writer->setListener(mListener);
- *mediaWriter = writer;
+ mWriter = writer;
return OK;
}
-void StagefrightRecorder::setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
- sp<MetaData> *meta) {
+void StagefrightRecorder::setupMPEG4MetaData(sp<MetaData> *meta) {
+ int64_t startTimeUs = systemTime() / 1000;
(*meta)->setInt64(kKeyTime, startTimeUs);
(*meta)->setInt32(kKeyFileType, mOutputFormat);
- (*meta)->setInt32(kKeyBitRate, totalBitRate);
+ (*meta)->setInt32(kKeyBitRate, mTotalBitRate);
(*meta)->setInt32(kKey64BitFileOffset, mUse64BitFileOffset);
if (mMovieTimeScale > 0) {
(*meta)->setInt32(kKeyTimeScale, mMovieTimeScale);
@@ -1577,27 +1594,6 @@
}
}
-status_t StagefrightRecorder::startMPEG4Recording() {
- int32_t totalBitRate;
- status_t err = setupMPEG4Recording(
- mOutputFd, mVideoWidth, mVideoHeight,
- mVideoBitRate, &totalBitRate, &mWriter);
- if (err != OK) {
- return err;
- }
-
- int64_t startTimeUs = systemTime() / 1000;
- sp<MetaData> meta = new MetaData;
- setupMPEG4MetaData(startTimeUs, totalBitRate, &meta);
-
- err = mWriter->start(meta.get());
- if (err != OK) {
- return err;
- }
-
- return OK;
-}
-
status_t StagefrightRecorder::pause() {
ALOGV("pause");
if (mWriter == NULL) {
@@ -1637,6 +1633,8 @@
mWriter.clear();
}
+ mGraphicBufferProducer.clear();
+
if (mOutputFd >= 0) {
::close(mOutputFd);
mOutputFd = -1;
@@ -1656,7 +1654,6 @@
addBatteryData(params);
}
-
return err;
}
@@ -1708,6 +1705,7 @@
mRotationDegrees = 0;
mLatitudex10000 = -3600000;
mLongitudex10000 = -3600000;
+ mTotalBitRate = 0;
mOutputFd = -1;
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 31f09e0..7d6abd3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -37,6 +37,7 @@
class MediaProfiles;
class IGraphicBufferProducer;
class SurfaceMediaSource;
+class ALooper;
struct StagefrightRecorder : public MediaRecorderBase {
StagefrightRecorder();
@@ -106,6 +107,7 @@
int32_t mLatitudex10000;
int32_t mLongitudex10000;
int32_t mStartTimeOffsetMs;
+ int32_t mTotalBitRate;
bool mCaptureTimeLapse;
int64_t mTimeBetweenTimeLapseFrameCaptureUs;
@@ -122,22 +124,16 @@
// An <IGraphicBufferProducer> pointer
// will be sent to the client side using which the
// frame buffers will be queued and dequeued
- sp<SurfaceMediaSource> mSurfaceMediaSource;
+ sp<IGraphicBufferProducer> mGraphicBufferProducer;
+ sp<ALooper> mLooper;
- status_t setupMPEG4Recording(
- int outputFd,
- int32_t videoWidth, int32_t videoHeight,
- int32_t videoBitRate,
- int32_t *totalBitRate,
- sp<MediaWriter> *mediaWriter);
- void setupMPEG4MetaData(int64_t startTimeUs, int32_t totalBitRate,
- sp<MetaData> *meta);
- status_t startMPEG4Recording();
- status_t startAMRRecording();
- status_t startAACRecording();
- status_t startRawAudioRecording();
- status_t startRTPRecording();
- status_t startMPEG2TSRecording();
+ status_t setupMPEG4Recording();
+ void setupMPEG4MetaData(sp<MetaData> *meta);
+ status_t setupAMRRecording();
+ status_t setupAACRecording();
+ status_t setupRawAudioRecording();
+ status_t setupRTPRecording();
+ status_t setupMPEG2TSRecording();
sp<MediaSource> createAudioSource();
status_t checkVideoEncoderCapabilities(
bool *supportsCameraSourceMetaDataMode);
@@ -147,14 +143,8 @@
// depending on the videosource type
status_t setupMediaSource(sp<MediaSource> *mediaSource);
status_t setupCameraSource(sp<CameraSource> *cameraSource);
- // setup the surfacemediasource for the encoder
- status_t setupSurfaceMediaSource();
-
status_t setupAudioEncoder(const sp<MediaWriter>& writer);
- status_t setupVideoEncoder(
- sp<MediaSource> cameraSource,
- int32_t videoBitRate,
- sp<MediaSource> *source);
+ status_t setupVideoEncoder(sp<MediaSource> cameraSource, sp<MediaSource> *source);
// Encoding parameter handling utilities
status_t setParameter(const String8 &key, const String8 &value);
diff --git a/media/libmediaplayerservice/TestPlayerStub.cpp b/media/libmediaplayerservice/TestPlayerStub.cpp
index 5d9728a..5795773 100644
--- a/media/libmediaplayerservice/TestPlayerStub.cpp
+++ b/media/libmediaplayerservice/TestPlayerStub.cpp
@@ -113,7 +113,9 @@
// Create the test player.
// Call setDataSource on the test player with the url in param.
status_t TestPlayerStub::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
if (!isTestUrl(url) || NULL != mHandle) {
return INVALID_OPERATION;
}
@@ -162,7 +164,7 @@
}
mPlayer = (*mNewPlayer)();
- return mPlayer->setDataSource(mContentUrl, headers);
+ return mPlayer->setDataSource(httpService, mContentUrl, headers);
}
// Internal cleanup.
diff --git a/media/libmediaplayerservice/TestPlayerStub.h b/media/libmediaplayerservice/TestPlayerStub.h
index a3802eb..55bf2c8 100644
--- a/media/libmediaplayerservice/TestPlayerStub.h
+++ b/media/libmediaplayerservice/TestPlayerStub.h
@@ -66,7 +66,9 @@
// @param url Should be a test url. See class comment.
virtual status_t setDataSource(
- const char* url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char* url,
+ const KeyedVector<String8, String8> *headers);
// Test player for a file descriptor source is not supported.
virtual status_t setDataSource(int, int64_t, int64_t) {
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index b04e7a6..06aac33 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -33,17 +33,16 @@
NuPlayer::GenericSource::GenericSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
- const KeyedVector<String8, String8> *headers,
- bool uidValid,
- uid_t uid)
+ const KeyedVector<String8, String8> *headers)
: Source(notify),
mDurationUs(0ll),
mAudioIsVorbis(false) {
DataSource::RegisterDefaultSniffers();
sp<DataSource> dataSource =
- DataSource::CreateFromURI(url, headers);
+ DataSource::CreateFromURI(httpService, url, headers);
CHECK(dataSource != NULL);
initFromDataSource(dataSource);
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index 2da680c..20d597e 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -33,10 +33,9 @@
struct NuPlayer::GenericSource : public NuPlayer::Source {
GenericSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
- const KeyedVector<String8, String8> *headers,
- bool uidValid = false,
- uid_t uid = 0);
+ const KeyedVector<String8, String8> *headers);
GenericSource(
const sp<AMessage> ¬ify,
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index f1782cc..ac2aab8 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -24,6 +24,7 @@
#include "LiveDataSource.h"
#include "LiveSession.h"
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -34,13 +35,12 @@
NuPlayer::HTTPLiveSource::HTTPLiveSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
- const KeyedVector<String8, String8> *headers,
- bool uidValid, uid_t uid)
+ const KeyedVector<String8, String8> *headers)
: Source(notify),
+ mHTTPService(httpService),
mURL(url),
- mUIDValid(uidValid),
- mUID(uid),
mFlags(0),
mFinalResult(OK),
mOffset(0),
@@ -79,8 +79,7 @@
mLiveSession = new LiveSession(
notify,
(mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0,
- mUIDValid,
- mUID);
+ mHTTPService);
mLiveLooper->registerHandler(mLiveSession);
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
index bcc3f8b..4d7251f 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
@@ -28,10 +28,9 @@
struct NuPlayer::HTTPLiveSource : public NuPlayer::Source {
HTTPLiveSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
- const KeyedVector<String8, String8> *headers,
- bool uidValid = false,
- uid_t uid = 0);
+ const KeyedVector<String8, String8> *headers);
virtual void prepareAsync();
virtual void start();
@@ -61,10 +60,9 @@
kWhatFetchSubtitleData,
};
+ sp<IMediaHTTPService> mHTTPService;
AString mURL;
KeyedVector<String8, String8> mExtraHeaders;
- bool mUIDValid;
- uid_t mUID;
uint32_t mFlags;
status_t mFinalResult;
off64_t mOffset;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 3669a5b..817395a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -212,7 +212,9 @@
}
void NuPlayer::setDataSourceAsync(
- const char *url, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
size_t len = strlen(url);
@@ -220,16 +222,18 @@
sp<Source> source;
if (IsHTTPLiveURL(url)) {
- source = new HTTPLiveSource(notify, url, headers, mUIDValid, mUID);
+ source = new HTTPLiveSource(notify, httpService, url, headers);
} else if (!strncasecmp(url, "rtsp://", 7)) {
- source = new RTSPSource(notify, url, headers, mUIDValid, mUID);
+ source = new RTSPSource(
+ notify, httpService, url, headers, mUIDValid, mUID);
} else if ((!strncasecmp(url, "http://", 7)
|| !strncasecmp(url, "https://", 8))
&& ((len >= 4 && !strcasecmp(".sdp", &url[len - 4]))
|| strstr(url, ".sdp?"))) {
- source = new RTSPSource(notify, url, headers, mUIDValid, mUID, true);
+ source = new RTSPSource(
+ notify, httpService, url, headers, mUIDValid, mUID, true);
} else {
- source = new GenericSource(notify, url, headers, mUIDValid, mUID);
+ source = new GenericSource(notify, httpService, url, headers);
}
msg->setObject("source", source);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 590e1f2..9dfe4a0 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -38,7 +38,9 @@
void setDataSourceAsync(const sp<IStreamSource> &source);
void setDataSourceAsync(
- const char *url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers);
void setDataSourceAsync(int fd, int64_t offset, int64_t length);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 239296e..9ef8dbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -71,7 +71,9 @@
}
status_t NuPlayerDriver::setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
Mutex::Autolock autoLock(mLock);
if (mState != STATE_IDLE) {
@@ -80,7 +82,7 @@
mState = STATE_SET_DATASOURCE_PENDING;
- mPlayer->setDataSourceAsync(url, headers);
+ mPlayer->setDataSourceAsync(httpService, url, headers);
while (mState == STATE_SET_DATASOURCE_PENDING) {
mCondition.wait(mLock);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index 99f72a6..0148fb1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -31,7 +31,9 @@
virtual status_t setUID(uid_t uid);
virtual status_t setDataSource(
- const char *url, const KeyedVector<String8, String8> *headers);
+ const sp<IMediaHTTPService> &httpService,
+ const char *url,
+ const KeyedVector<String8, String8> *headers);
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 18cf6d1..94800ba 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -24,6 +24,7 @@
#include "MyHandler.h"
#include "SDPLoader.h"
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
@@ -33,12 +34,14 @@
NuPlayer::RTSPSource::RTSPSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers,
bool uidValid,
uid_t uid,
bool isSDP)
: Source(notify),
+ mHTTPService(httpService),
mURL(url),
mUIDValid(uidValid),
mUID(uid),
@@ -92,7 +95,7 @@
if (mIsSDP) {
mSDPLoader = new SDPLoader(notify,
(mFlags & kFlagIncognito) ? SDPLoader::kFlagIncognito : 0,
- mUIDValid, mUID);
+ mHTTPService);
mSDPLoader->load(
mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.h b/media/libmediaplayerservice/nuplayer/RTSPSource.h
index 8cf34a0..3718bf9 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.h
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.h
@@ -34,6 +34,7 @@
struct NuPlayer::RTSPSource : public NuPlayer::Source {
RTSPSource(
const sp<AMessage> ¬ify,
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers,
bool uidValid = false,
@@ -88,6 +89,7 @@
bool mNPTMappingValid;
};
+ sp<IMediaHTTPService> mHTTPService;
AString mURL;
KeyedVector<String8, String8> mExtraHeaders;
bool mUIDValid;
diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index 74a6fdb..4a69104 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -24,11 +24,11 @@
namespace android {
AudioBufferProviderSource::AudioBufferProviderSource(AudioBufferProvider *provider,
- NBAIO_Format format) :
+ const NBAIO_Format& format) :
NBAIO_Source(format), mProvider(provider), mConsumed(0)
{
ALOG_ASSERT(provider != NULL);
- ALOG_ASSERT(format != Format_Invalid);
+ ALOG_ASSERT(Format_isValid(format));
}
AudioBufferProviderSource::~AudioBufferProviderSource()
diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index 05273f6..ae8fac8 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -40,7 +40,7 @@
ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- if (mFormat == Format_Invalid) {
+ if (!Format_isValid(mFormat)) {
mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -67,7 +67,7 @@
ssize_t AudioStreamInSource::read(void *buffer, size_t count)
{
- if (CC_UNLIKELY(mFormat == Format_Invalid)) {
+ if (CC_UNLIKELY(!Format_isValid(mFormat))) {
return NEGOTIATE;
}
ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift);
diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index e4341d7..aa9810e 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -37,7 +37,7 @@
ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- if (mFormat == Format_Invalid) {
+ if (!Format_isValid(mFormat)) {
mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -56,7 +56,7 @@
if (!mNegotiated) {
return NEGOTIATE;
}
- ALOG_ASSERT(mFormat != Format_Invalid);
+ ALOG_ASSERT(Format_isValid(mFormat));
ssize_t ret = mStream->write(mStream, buffer, count << mBitShift);
if (ret > 0) {
ret >>= mBitShift;
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 3c61b60..b23967b 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -30,7 +30,7 @@
namespace android {
-MonoPipe::MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock) :
+MonoPipe::MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock) :
NBAIO_Sink(format),
mUpdateSeq(0),
mReqFrames(reqFrames),
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index e0d2c21..51514de 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -22,17 +22,22 @@
namespace android {
-size_t Format_frameSize(NBAIO_Format format)
+size_t Format_frameSize(const NBAIO_Format& format)
{
+ // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
return Format_channelCount(format) * sizeof(short);
}
-size_t Format_frameBitShift(NBAIO_Format format)
+int Format_frameBitShift(const NBAIO_Format& format)
{
+ // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
// sizeof(short) == 2, so frame size == 1 << channels
return Format_channelCount(format);
+ // FIXME must return -1 for non-power of 2
}
+const NBAIO_Format Format_Invalid = { 0 };
+
enum {
Format_SR_8000,
Format_SR_11025,
@@ -51,12 +56,12 @@
Format_C_Mask = 0x18
};
-unsigned Format_sampleRate(NBAIO_Format format)
+unsigned Format_sampleRate(const NBAIO_Format& format)
{
- if (format == Format_Invalid) {
+ if (!Format_isValid(format)) {
return 0;
}
- switch (format & Format_SR_Mask) {
+ switch (format.mPacked & Format_SR_Mask) {
case Format_SR_8000:
return 8000;
case Format_SR_11025:
@@ -78,12 +83,12 @@
}
}
-unsigned Format_channelCount(NBAIO_Format format)
+unsigned Format_channelCount(const NBAIO_Format& format)
{
- if (format == Format_Invalid) {
+ if (!Format_isValid(format)) {
return 0;
}
- switch (format & Format_C_Mask) {
+ switch (format.mPacked & Format_C_Mask) {
case Format_C_1:
return 1;
case Format_C_2:
@@ -95,7 +100,7 @@
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
{
- NBAIO_Format format;
+ unsigned format;
switch (sampleRate) {
case 8000:
format = Format_SR_8000;
@@ -134,7 +139,9 @@
default:
return Format_Invalid;
}
- return format;
+ NBAIO_Format ret;
+ ret.mPacked = format;
+ return ret;
}
// This is a default implementation; it is expected that subclasses will optimize this.
@@ -216,9 +223,9 @@
{
ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
offers, numOffers, counterOffers, numCounterOffers);
- if (mFormat != Format_Invalid) {
+ if (Format_isValid(mFormat)) {
for (size_t i = 0; i < numOffers; ++i) {
- if (offers[i] == mFormat) {
+ if (Format_isEqual(offers[i], mFormat)) {
mNegotiated = true;
return i;
}
@@ -233,4 +240,14 @@
return (ssize_t) NEGOTIATE;
}
+bool Format_isValid(const NBAIO_Format& format)
+{
+ return format.mPacked != Format_Invalid.mPacked;
+}
+
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2)
+{
+ return format1.mPacked == format2.mPacked;
+}
+
} // namespace android
diff --git a/media/libnbaio/NBLog.cpp b/media/libnbaio/NBLog.cpp
index d74a7a6..190824d 100644
--- a/media/libnbaio/NBLog.cpp
+++ b/media/libnbaio/NBLog.cpp
@@ -441,7 +441,7 @@
bool NBLog::Reader::isIMemory(const sp<IMemory>& iMemory) const
{
- return iMemory.get() == mIMemory.get();
+ return iMemory != 0 && mIMemory != 0 && iMemory->pointer() == mIMemory->pointer();
}
} // namespace android
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 1c21f9c..115f311 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -25,7 +25,7 @@
namespace android {
-Pipe::Pipe(size_t maxFrames, NBAIO_Format format) :
+Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format) :
NBAIO_Sink(format),
mMaxFrames(roundup(maxFrames)),
mBuffer(malloc(mMaxFrames * Format_frameSize(format))),
diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp
index d786b84..24da1bd 100644
--- a/media/libnbaio/PipeReader.cpp
+++ b/media/libnbaio/PipeReader.cpp
@@ -59,7 +59,7 @@
return avail;
}
-ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS)
+ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS __unused)
{
ssize_t avail = availableToRead();
if (CC_UNLIKELY(avail <= 0)) {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index c68dfc1..f2354b4 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -365,6 +365,7 @@
mIsEncoder(false),
mUseMetadataOnEncoderOutput(false),
mShutdownInProgress(false),
+ mIsConfiguredForAdaptivePlayback(false),
mEncoderDelay(0),
mEncoderPadding(0),
mChannelMaskPresent(false),
@@ -372,7 +373,9 @@
mDequeueCounter(0),
mStoreMetaDataInOutputBuffers(false),
mMetaDataBuffersToSubmit(0),
- mRepeatFrameDelayUs(-1ll) {
+ mRepeatFrameDelayUs(-1ll),
+ mMaxPtsGapUs(-1l),
+ mCreateInputBuffersSuspended(false) {
mUninitializedState = new UninitializedState(this);
mLoadedState = new LoadedState(this);
mLoadedToIdleState = new LoadedToIdleState(this);
@@ -1114,6 +1117,16 @@
&mRepeatFrameDelayUs)) {
mRepeatFrameDelayUs = -1ll;
}
+
+ if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) {
+ mMaxPtsGapUs = -1l;
+ }
+
+ if (!msg->findInt32(
+ "create-input-buffers-suspended",
+ (int32_t*)&mCreateInputBuffersSuspended)) {
+ mCreateInputBuffersSuspended = false;
+ }
}
// Always try to enable dynamic output buffers on native surface
@@ -1121,6 +1134,7 @@
int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
obj != NULL;
mStoreMetaDataInOutputBuffers = false;
+ mIsConfiguredForAdaptivePlayback = false;
if (!encoder && video && haveNativeWindow) {
err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
if (err != OK) {
@@ -1165,12 +1179,14 @@
ALOGW_IF(err != OK,
"[%s] prepareForAdaptivePlayback failed w/ err %d",
mComponentName.c_str(), err);
+ mIsConfiguredForAdaptivePlayback = (err == OK);
}
// allow failure
err = OK;
} else {
ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
mStoreMetaDataInOutputBuffers = true;
+ mIsConfiguredForAdaptivePlayback = true;
}
int32_t push;
@@ -3318,11 +3334,11 @@
mCodec->mInputEOSResult = err;
}
break;
-
- default:
- CHECK_EQ((int)mode, (int)FREE_BUFFERS);
- break;
}
+
+ default:
+ CHECK_EQ((int)mode, (int)FREE_BUFFERS);
+ break;
}
}
@@ -3413,7 +3429,7 @@
sp<AMessage> reply =
new AMessage(kWhatOutputBufferDrained, mCodec->id());
- if (!mCodec->mSentFormat) {
+ if (!mCodec->mSentFormat && rangeLength > 0) {
mCodec->sendFormatChange(reply);
}
@@ -3773,6 +3789,7 @@
mCodec->mDequeueCounter = 0;
mCodec->mMetaDataBuffersToSubmit = 0;
mCodec->mRepeatFrameDelayUs = -1ll;
+ mCodec->mIsConfiguredForAdaptivePlayback = false;
if (mCodec->mShutdownInProgress) {
bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
@@ -3921,6 +3938,37 @@
}
}
+ if (err == OK && mCodec->mMaxPtsGapUs > 0l) {
+ err = mCodec->mOMX->setInternalOption(
+ mCodec->mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP,
+ &mCodec->mMaxPtsGapUs,
+ sizeof(mCodec->mMaxPtsGapUs));
+
+ if (err != OK) {
+ ALOGE("[%s] Unable to configure max timestamp gap (err %d)",
+ mCodec->mComponentName.c_str(),
+ err);
+ }
+ }
+
+ if (err == OK && mCodec->mCreateInputBuffersSuspended) {
+ bool suspend = true;
+ err = mCodec->mOMX->setInternalOption(
+ mCodec->mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_SUSPEND,
+ &suspend,
+ sizeof(suspend));
+
+ if (err != OK) {
+ ALOGE("[%s] Unable to configure option to suspend (err %d)",
+ mCodec->mComponentName.c_str(),
+ err);
+ }
+ }
+
if (err == OK) {
notify->setObject("input-surface",
new BufferProducerWrapper(bufferProducer));
@@ -3978,6 +4026,7 @@
bool ACodec::LoadedToIdleState::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
+ case kWhatSetParameters:
case kWhatShutdown:
{
mCodec->deferMessage(msg);
@@ -4044,6 +4093,7 @@
bool ACodec::IdleToExecutingState::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
+ case kWhatSetParameters:
case kWhatShutdown:
{
mCodec->deferMessage(msg);
@@ -4313,6 +4363,22 @@
}
}
+ int64_t skipFramesBeforeUs;
+ if (params->findInt64("skip-frames-before", &skipFramesBeforeUs)) {
+ status_t err =
+ mOMX->setInternalOption(
+ mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_START_TIME,
+ &skipFramesBeforeUs,
+ sizeof(skipFramesBeforeUs));
+
+ if (err != OK) {
+ ALOGE("Failed to set parameter 'skip-frames-before' (err %d)", err);
+ return err;
+ }
+ }
+
int32_t dropInputFrames;
if (params->findInt32("drop-input-frames", &dropInputFrames)) {
bool suspend = dropInputFrames != 0;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 6a2a696..63f9399 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -30,8 +30,10 @@
MediaBufferGroup.cpp \
MediaCodec.cpp \
MediaCodecList.cpp \
+ MediaCodecSource.cpp \
MediaDefs.cpp \
MediaExtractor.cpp \
+ http/MediaHTTP.cpp \
MediaMuxer.cpp \
MediaSource.cpp \
MetaData.cpp \
@@ -103,13 +105,6 @@
libFLAC \
libmedia_helper
-LOCAL_SRC_FILES += \
- chromium_http_stub.cpp
-LOCAL_CPPFLAGS += -DCHROMIUM_AVAILABLE=1
-
-LOCAL_SHARED_LIBRARIES += libstlport
-include external/stlport/libstlport.mk
-
LOCAL_SHARED_LIBRARIES += \
libstagefright_enc_common \
libstagefright_avc_common \
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index f0d1a14..6fdee6b 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -278,7 +278,7 @@
// Drop retrieved and previously lost audio data.
if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
- mRecord->getInputFramesLost();
+ (void) mRecord->getInputFramesLost();
ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
return OK;
}
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 29c007a..f57ee25 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -35,6 +35,8 @@
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -45,6 +47,7 @@
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaHTTP.h>
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
@@ -277,15 +280,20 @@
}
status_t AwesomePlayer::setDataSource(
- const char *uri, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *uri,
+ const KeyedVector<String8, String8> *headers) {
Mutex::Autolock autoLock(mLock);
- return setDataSource_l(uri, headers);
+ return setDataSource_l(httpService, uri, headers);
}
status_t AwesomePlayer::setDataSource_l(
- const char *uri, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *uri,
+ const KeyedVector<String8, String8> *headers) {
reset_l();
+ mHTTPService = httpService;
mUri = uri;
if (headers) {
@@ -582,6 +590,7 @@
mSeekNotificationSent = true;
mSeekTimeUs = 0;
+ mHTTPService.clear();
mUri.setTo("");
mUriHeaders.clear();
@@ -1483,7 +1492,7 @@
CHECK(source != NULL);
if (mTextDriver == NULL) {
- mTextDriver = new TimedTextDriver(mListener);
+ mTextDriver = new TimedTextDriver(mListener, mHTTPService);
}
mTextDriver->addInBandTextSource(trackIndex, source);
@@ -2193,15 +2202,14 @@
if (!strncasecmp("http://", mUri.string(), 7)
|| !strncasecmp("https://", mUri.string(), 8)
|| isWidevineStreaming) {
- mConnectingDataSource = HTTPBase::Create(
- (mFlags & INCOGNITO)
- ? HTTPBase::kFlagIncognito
- : 0);
-
- if (mUIDValid) {
- mConnectingDataSource->setUID(mUID);
+ if (mHTTPService == NULL) {
+ ALOGE("Attempt to play media from http URI without HTTP service.");
+ return UNKNOWN_ERROR;
}
+ sp<IMediaHTTPConnection> conn = mHTTPService->makeHTTPConnection();
+ mConnectingDataSource = new MediaHTTP(conn);
+
String8 cacheConfig;
bool disconnectAtHighwatermark;
NuCachedSource2::RemoveCacheSpecificHeaders(
@@ -2317,7 +2325,8 @@
}
}
} else {
- dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+ dataSource = DataSource::CreateFromURI(
+ mHTTPService, mUri.string(), &mUriHeaders);
}
if (dataSource == NULL) {
@@ -2759,7 +2768,7 @@
{
Mutex::Autolock autoLock(mLock);
if (mTextDriver == NULL) {
- mTextDriver = new TimedTextDriver(mListener);
+ mTextDriver = new TimedTextDriver(mListener, mHTTPService);
}
// String values written in Parcel are UTF-16 values.
String8 uri(request.readString16());
@@ -2771,7 +2780,7 @@
{
Mutex::Autolock autoLock(mLock);
if (mTextDriver == NULL) {
- mTextDriver = new TimedTextDriver(mListener);
+ mTextDriver = new TimedTextDriver(mListener, mHTTPService);
}
int fd = request.readFileDescriptor();
off64_t offset = request.readInt64();
@@ -2900,6 +2909,8 @@
// get current position so we can start recreated stream from here
getPosition(&mAudioTearDownPosition);
+ sp<IMediaHTTPService> savedHTTPService = mHTTPService;
+
// Reset and recreate
reset_l();
@@ -2909,7 +2920,7 @@
mFileSource = fileSource;
err = setDataSource_l(fileSource);
} else {
- err = setDataSource_l(uri, &uriHeaders);
+ err = setDataSource_l(savedHTTPService, uri, &uriHeaders);
}
mFlags |= PREPARING;
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 5772316..86844b8 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -85,7 +85,8 @@
mVideoWidth = videoSize.width;
mVideoHeight = videoSize.height;
- if (!trySettingVideoSize(videoSize.width, videoSize.height)) {
+ if (OK == mInitCheck && !trySettingVideoSize(videoSize.width, videoSize.height)) {
+ releaseCamera();
mInitCheck = NO_INIT;
}
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index 97987e2..2704b74 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -16,10 +16,6 @@
#include "include/AMRExtractor.h"
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
#include "include/AACExtractor.h"
#include "include/DRMExtractor.h"
#include "include/FLACExtractor.h"
@@ -35,10 +31,13 @@
#include "matroska/MatroskaExtractor.h"
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
#include <utils/String8.h>
#include <cutils/properties.h>
@@ -180,7 +179,9 @@
// static
sp<DataSource> DataSource::CreateFromURI(
- const char *uri, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *uri,
+ const KeyedVector<String8, String8> *headers) {
bool isWidevine = !strncasecmp("widevine://", uri, 11);
sp<DataSource> source;
@@ -189,7 +190,7 @@
} else if (!strncasecmp("http://", uri, 7)
|| !strncasecmp("https://", uri, 8)
|| isWidevine) {
- sp<HTTPBase> httpSource = HTTPBase::Create();
+ sp<HTTPBase> httpSource = new MediaHTTP(httpService->makeHTTPConnection());
String8 tmp;
if (isWidevine) {
@@ -220,11 +221,6 @@
// in the widevine:// case.
source = httpSource;
}
-
-# if CHROMIUM_AVAILABLE
- } else if (!strncasecmp("data:", uri, 5)) {
- source = createDataUriSource(uri);
-#endif
} else {
// Assume it's a filename.
source = new FileSource(uri);
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libstagefright/HTTPBase.cpp
index 5fa4b6f..ca68c3d 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libstagefright/HTTPBase.cpp
@@ -20,10 +20,6 @@
#include "include/HTTPBase.h"
-#if CHROMIUM_AVAILABLE
-#include "include/chromium_http_stub.h"
-#endif
-
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
@@ -40,34 +36,7 @@
mTotalTransferBytes(0),
mPrevBandwidthMeasureTimeUs(0),
mPrevEstimatedBandWidthKbps(0),
- mBandWidthCollectFreqMs(5000),
- mUIDValid(false),
- mUID(0) {
-}
-
-// static
-sp<HTTPBase> HTTPBase::Create(uint32_t flags) {
-#if CHROMIUM_AVAILABLE
- HTTPBase *dataSource = createChromiumHTTPDataSource(flags);
- if (dataSource) {
- return dataSource;
- }
-#endif
- {
- TRESPASS();
-
- return NULL;
- }
-}
-
-// static
-status_t HTTPBase::UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
-#if CHROMIUM_AVAILABLE
- return UpdateChromiumHTTPDataSourceProxyConfig(host, port, exclusionList);
-#else
- return INVALID_OPERATION;
-#endif
+ mBandWidthCollectFreqMs(5000) {
}
void HTTPBase::addBandwidthMeasurement(
@@ -135,21 +104,6 @@
return OK;
}
-void HTTPBase::setUID(uid_t uid) {
- mUIDValid = true;
- mUID = uid;
-}
-
-bool HTTPBase::getUID(uid_t *uid) const {
- if (!mUIDValid) {
- return false;
- }
-
- *uid = mUID;
-
- return true;
-}
-
// static
void HTTPBase::RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag) {
int res = qtaguid_tagSocket(sockfd, kTag, uid);
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 6a33ce6..e6312c1 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2441,6 +2441,58 @@
return OK;
}
+typedef enum {
+ //AOT_NONE = -1,
+ //AOT_NULL_OBJECT = 0,
+ //AOT_AAC_MAIN = 1, /**< Main profile */
+ AOT_AAC_LC = 2, /**< Low Complexity object */
+ //AOT_AAC_SSR = 3,
+ //AOT_AAC_LTP = 4,
+ AOT_SBR = 5,
+ //AOT_AAC_SCAL = 6,
+ //AOT_TWIN_VQ = 7,
+ //AOT_CELP = 8,
+ //AOT_HVXC = 9,
+ //AOT_RSVD_10 = 10, /**< (reserved) */
+ //AOT_RSVD_11 = 11, /**< (reserved) */
+ //AOT_TTSI = 12, /**< TTSI Object */
+ //AOT_MAIN_SYNTH = 13, /**< Main Synthetic object */
+ //AOT_WAV_TAB_SYNTH = 14, /**< Wavetable Synthesis object */
+ //AOT_GEN_MIDI = 15, /**< General MIDI object */
+ //AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
+ AOT_ER_AAC_LC = 17, /**< Error Resilient(ER) AAC Low Complexity */
+ //AOT_RSVD_18 = 18, /**< (reserved) */
+ //AOT_ER_AAC_LTP = 19, /**< Error Resilient(ER) AAC LTP object */
+ AOT_ER_AAC_SCAL = 20, /**< Error Resilient(ER) AAC Scalable object */
+ //AOT_ER_TWIN_VQ = 21, /**< Error Resilient(ER) TwinVQ object */
+ AOT_ER_BSAC = 22, /**< Error Resilient(ER) BSAC object */
+ AOT_ER_AAC_LD = 23, /**< Error Resilient(ER) AAC LowDelay object */
+ //AOT_ER_CELP = 24, /**< Error Resilient(ER) CELP object */
+ //AOT_ER_HVXC = 25, /**< Error Resilient(ER) HVXC object */
+ //AOT_ER_HILN = 26, /**< Error Resilient(ER) HILN object */
+ //AOT_ER_PARA = 27, /**< Error Resilient(ER) Parametric object */
+ //AOT_RSVD_28 = 28, /**< might become SSC */
+ AOT_PS = 29, /**< PS, Parametric Stereo (includes SBR) */
+ //AOT_MPEGS = 30, /**< MPEG Surround */
+
+ AOT_ESCAPE = 31, /**< Signal AOT uses more than 5 bits */
+
+ //AOT_MP3ONMP4_L1 = 32, /**< MPEG-Layer1 in mp4 */
+ //AOT_MP3ONMP4_L2 = 33, /**< MPEG-Layer2 in mp4 */
+ //AOT_MP3ONMP4_L3 = 34, /**< MPEG-Layer3 in mp4 */
+ //AOT_RSVD_35 = 35, /**< might become DST */
+ //AOT_RSVD_36 = 36, /**< might become ALS */
+ //AOT_AAC_SLS = 37, /**< AAC + SLS */
+ //AOT_SLS = 38, /**< SLS */
+ //AOT_ER_AAC_ELD = 39, /**< AAC Enhanced Low Delay */
+
+ //AOT_USAC = 42, /**< USAC */
+ //AOT_SAOC = 43, /**< SAOC */
+ //AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */
+
+ //AOT_RSVD50 = 50, /**< Interim AOT for Rsvd50 */
+} AUDIO_OBJECT_TYPE;
+
status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
const void *esds_data, size_t esds_size) {
ESDS esds(esds_data, esds_size);
@@ -2523,7 +2575,7 @@
sampleRate = kSamplingRate[freqIndex];
}
- if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+ if (objectType == AOT_SBR || objectType == AOT_PS) {//SBR specific config per 14496-3 table 1.13
uint32_t extFreqIndex = br.getBits(4);
int32_t extSampleRate;
if (extFreqIndex == 15) {
@@ -2541,6 +2593,111 @@
// mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
}
+ switch (numChannels) {
+ // values defined in 14496-3_2009 amendment-4 Table 1.19 - Channel Configuration
+ case 0:
+ case 1:// FC
+ case 2:// FL FR
+ case 3:// FC, FL FR
+ case 4:// FC, FL FR, RC
+ case 5:// FC, FL FR, SL SR
+ case 6:// FC, FL FR, SL SR, LFE
+ //numChannels already contains the right value
+ break;
+ case 11:// FC, FL FR, SL SR, RC, LFE
+ numChannels = 7;
+ break;
+ case 7: // FC, FCL FCR, FL FR, SL SR, LFE
+ case 12:// FC, FL FR, SL SR, RL RR, LFE
+ case 14:// FC, FL FR, SL SR, LFE, FHL FHR
+ numChannels = 8;
+ break;
+ default:
+ return ERROR_UNSUPPORTED;
+ }
+
+ {
+ if (objectType == AOT_SBR || objectType == AOT_PS) {
+ const int32_t extensionSamplingFrequency = br.getBits(4);
+ objectType = br.getBits(5);
+
+ if (objectType == AOT_ESCAPE) {
+ objectType = 32 + br.getBits(6);
+ }
+ }
+ if (objectType == AOT_AAC_LC || objectType == AOT_ER_AAC_LC ||
+ objectType == AOT_ER_AAC_LD || objectType == AOT_ER_AAC_SCAL ||
+ objectType == AOT_ER_BSAC) {
+ const int32_t frameLengthFlag = br.getBits(1);
+
+ const int32_t dependsOnCoreCoder = br.getBits(1);
+
+ if (dependsOnCoreCoder ) {
+ const int32_t coreCoderDelay = br.getBits(14);
+ }
+
+ const int32_t extensionFlag = br.getBits(1);
+
+ if (numChannels == 0 ) {
+ int32_t channelsEffectiveNum = 0;
+ int32_t channelsNum = 0;
+ const int32_t ElementInstanceTag = br.getBits(4);
+ const int32_t Profile = br.getBits(2);
+ const int32_t SamplingFrequencyIndex = br.getBits(4);
+ const int32_t NumFrontChannelElements = br.getBits(4);
+ const int32_t NumSideChannelElements = br.getBits(4);
+ const int32_t NumBackChannelElements = br.getBits(4);
+ const int32_t NumLfeChannelElements = br.getBits(2);
+ const int32_t NumAssocDataElements = br.getBits(3);
+ const int32_t NumValidCcElements = br.getBits(4);
+
+ const int32_t MonoMixdownPresent = br.getBits(1);
+ if (MonoMixdownPresent != 0) {
+ const int32_t MonoMixdownElementNumber = br.getBits(4);
+ }
+
+ const int32_t StereoMixdownPresent = br.getBits(1);
+ if (StereoMixdownPresent != 0) {
+ const int32_t StereoMixdownElementNumber = br.getBits(4);
+ }
+
+ const int32_t MatrixMixdownIndexPresent = br.getBits(1);
+ if (MatrixMixdownIndexPresent != 0) {
+ const int32_t MatrixMixdownIndex = br.getBits(2);
+ const int32_t PseudoSurroundEnable = br.getBits(1);
+ }
+
+ int i;
+ for (i=0; i < NumFrontChannelElements; i++) {
+ const int32_t FrontElementIsCpe = br.getBits(1);
+ const int32_t FrontElementTagSelect = br.getBits(4);
+ channelsNum += FrontElementIsCpe ? 2 : 1;
+ }
+
+ for (i=0; i < NumSideChannelElements; i++) {
+ const int32_t SideElementIsCpe = br.getBits(1);
+ const int32_t SideElementTagSelect = br.getBits(4);
+ channelsNum += SideElementIsCpe ? 2 : 1;
+ }
+
+ for (i=0; i < NumBackChannelElements; i++) {
+ const int32_t BackElementIsCpe = br.getBits(1);
+ const int32_t BackElementTagSelect = br.getBits(4);
+ channelsNum += BackElementIsCpe ? 2 : 1;
+ }
+ channelsEffectiveNum = channelsNum;
+
+ for (i=0; i < NumLfeChannelElements; i++) {
+ const int32_t LfeElementTagSelect = br.getBits(4);
+ channelsNum += 1;
+ }
+ ALOGV("mpeg4 audio channelsNum = %d", channelsNum);
+ ALOGV("mpeg4 audio channelsEffectiveNum = %d", channelsEffectiveNum);
+ numChannels = channelsNum;
+ }
+ }
+ }
+
if (numChannels == 0) {
return ERROR_UNSUPPORTED;
}
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index e7d3cc2..ff6feb9 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -973,13 +973,16 @@
if (param && param->findInt32(kKeyFileType, &fileType) &&
fileType != OUTPUT_FORMAT_MPEG_4) {
writeFourcc("3gp4");
- } else {
+ writeInt32(0);
writeFourcc("isom");
+ writeFourcc("3gp4");
+ } else {
+ writeFourcc("mp42");
+ writeInt32(0);
+ writeFourcc("isom");
+ writeFourcc("mp42");
}
- writeInt32(0);
- writeFourcc("isom");
- writeFourcc("3gp4");
endBox();
}
@@ -1761,7 +1764,7 @@
}
status_t MPEG4Writer::Track::stop() {
- ALOGD("Stopping %s track", mIsAudio? "Audio": "Video");
+ ALOGD("%s track stopping", mIsAudio? "Audio": "Video");
if (!mStarted) {
ALOGE("Stop() called but track is not started");
return ERROR_END_OF_STREAM;
@@ -1772,19 +1775,14 @@
}
mDone = true;
+ ALOGD("%s track source stopping", mIsAudio? "Audio": "Video");
+ mSource->stop();
+ ALOGD("%s track source stopped", mIsAudio? "Audio": "Video");
+
void *dummy;
pthread_join(mThread, &dummy);
-
status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
- ALOGD("Stopping %s track source", mIsAudio? "Audio": "Video");
- {
- status_t status = mSource->stop();
- if (err == OK && status != OK && status != ERROR_END_OF_STREAM) {
- err = status;
- }
- }
-
ALOGD("%s track stopped", mIsAudio? "Audio": "Video");
return err;
}
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
new file mode 100644
index 0000000..2c20d62
--- /dev/null
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -0,0 +1,881 @@
+/*
+ * Copyright 2014, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodecSource"
+#define DEBUG_DRIFT_TIME 0
+#include <gui/IGraphicBufferProducer.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MediaCodecSource.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+static void ReleaseMediaBufferReference(const sp<ABuffer> &accessUnit) {
+ void *mbuf;
+ if (accessUnit->meta()->findPointer("mediaBuffer", &mbuf)
+ && mbuf != NULL) {
+ ALOGV("releasing mbuf %p", mbuf);
+
+ accessUnit->meta()->setPointer("mediaBuffer", NULL);
+
+ static_cast<MediaBuffer *>(mbuf)->release();
+ mbuf = NULL;
+ }
+}
+
+struct MediaCodecSource::Puller : public AHandler {
+ Puller(const sp<MediaSource> &source);
+
+ status_t start(const sp<MetaData> &meta, const sp<AMessage> ¬ify);
+ void stopAsync();
+
+ void pause();
+ void resume();
+
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+ virtual ~Puller();
+
+private:
+ enum {
+ kWhatStart = 'msta',
+ kWhatStop,
+ kWhatPull,
+ kWhatPause,
+ kWhatResume,
+ };
+
+ sp<MediaSource> mSource;
+ sp<AMessage> mNotify;
+ sp<ALooper> mLooper;
+ int32_t mPullGeneration;
+ bool mIsAudio;
+ bool mPaused;
+ bool mReachedEOS;
+
+ status_t postSynchronouslyAndReturnError(const sp<AMessage> &msg);
+ void schedulePull();
+ void handleEOS();
+
+ DISALLOW_EVIL_CONSTRUCTORS(Puller);
+};
+
+MediaCodecSource::Puller::Puller(const sp<MediaSource> &source)
+ : mSource(source),
+ mLooper(new ALooper()),
+ mPullGeneration(0),
+ mIsAudio(false),
+ mPaused(false),
+ mReachedEOS(false) {
+ sp<MetaData> meta = source->getFormat();
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ mIsAudio = !strncasecmp(mime, "audio/", 6);
+
+ mLooper->setName("pull_looper");
+}
+
+MediaCodecSource::Puller::~Puller() {
+ mLooper->unregisterHandler(id());
+ mLooper->stop();
+}
+
+status_t MediaCodecSource::Puller::postSynchronouslyAndReturnError(
+ const sp<AMessage> &msg) {
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!response->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+
+status_t MediaCodecSource::Puller::start(const sp<MetaData> &meta,
+ const sp<AMessage> ¬ify) {
+ ALOGV("puller (%s) start", mIsAudio ? "audio" : "video");
+ mLooper->start(
+ false /* runOnCallingThread */,
+ false /* canCallJava */,
+ PRIORITY_AUDIO);
+ mLooper->registerHandler(this);
+ mNotify = notify;
+
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+ msg->setObject("meta", meta);
+ return postSynchronouslyAndReturnError(msg);
+}
+
+void MediaCodecSource::Puller::stopAsync() {
+ ALOGV("puller (%s) stopAsync", mIsAudio ? "audio" : "video");
+ (new AMessage(kWhatStop, id()))->post();
+}
+
+void MediaCodecSource::Puller::pause() {
+ (new AMessage(kWhatPause, id()))->post();
+}
+
+void MediaCodecSource::Puller::resume() {
+ (new AMessage(kWhatResume, id()))->post();
+}
+
+void MediaCodecSource::Puller::schedulePull() {
+ sp<AMessage> msg = new AMessage(kWhatPull, id());
+ msg->setInt32("generation", mPullGeneration);
+ msg->post();
+}
+
+void MediaCodecSource::Puller::handleEOS() {
+ if (!mReachedEOS) {
+ ALOGV("puller (%s) posting EOS", mIsAudio ? "audio" : "video");
+ mReachedEOS = true;
+ sp<AMessage> notify = mNotify->dup();
+ notify->setPointer("accessUnit", NULL);
+ notify->post();
+ }
+}
+
+void MediaCodecSource::Puller::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatStart:
+ {
+ sp<RefBase> obj;
+ CHECK(msg->findObject("meta", &obj));
+
+ mReachedEOS = false;
+
+ status_t err = mSource->start(static_cast<MetaData *>(obj.get()));
+
+ if (err == OK) {
+ schedulePull();
+ }
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatStop:
+ {
+ ALOGV("source (%s) stopping", mIsAudio ? "audio" : "video");
+ mSource->stop();
+ ALOGV("source (%s) stopped", mIsAudio ? "audio" : "video");
+ ++mPullGeneration;
+
+ handleEOS();
+ break;
+ }
+
+ case kWhatPull:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mPullGeneration) {
+ break;
+ }
+
+ MediaBuffer *mbuf;
+ status_t err = mSource->read(&mbuf);
+
+ if (mPaused) {
+ if (err == OK) {
+ mbuf->release();
+ mbuf = NULL;
+ }
+
+ msg->post();
+ break;
+ }
+
+ if (err != OK) {
+ if (err == ERROR_END_OF_STREAM) {
+ ALOGV("stream ended, mbuf %p", mbuf);
+ } else {
+ ALOGE("error %d reading stream.", err);
+ }
+ handleEOS();
+ } else {
+ sp<AMessage> notify = mNotify->dup();
+
+ notify->setPointer("accessUnit", mbuf);
+ notify->post();
+
+ msg->post();
+ }
+ break;
+ }
+
+ case kWhatPause:
+ {
+ mPaused = true;
+ break;
+ }
+
+ case kWhatResume:
+ {
+ mPaused = false;
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+// static
+sp<MediaCodecSource> MediaCodecSource::Create(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &format,
+ const sp<MediaSource> &source,
+ uint32_t flags) {
+ sp<MediaCodecSource> mediaSource =
+ new MediaCodecSource(looper, format, source, flags);
+
+ if (mediaSource->init() == OK) {
+ return mediaSource;
+ }
+ return NULL;
+}
+
+status_t MediaCodecSource::start(MetaData* params) {
+ sp<AMessage> msg = new AMessage(kWhatStart, mReflector->id());
+ msg->setObject("meta", params);
+ return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::stop() {
+ sp<AMessage> msg = new AMessage(kWhatStop, mReflector->id());
+ return postSynchronouslyAndReturnError(msg);
+}
+
+status_t MediaCodecSource::pause() {
+ (new AMessage(kWhatPause, mReflector->id()))->post();
+ return OK;
+}
+
+sp<IGraphicBufferProducer> MediaCodecSource::getGraphicBufferProducer() {
+ CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+ return mGraphicBufferProducer;
+}
+
+status_t MediaCodecSource::read(MediaBuffer** buffer,
+ const ReadOptions* options) {
+ Mutex::Autolock autolock(mOutputBufferLock);
+
+ *buffer = NULL;
+ while (mOutputBufferQueue.size() == 0 && !mEncodedReachedEOS) {
+ mOutputBufferCond.wait(mOutputBufferLock);
+ }
+ if (!mEncodedReachedEOS) {
+ *buffer = *mOutputBufferQueue.begin();
+ mOutputBufferQueue.erase(mOutputBufferQueue.begin());
+ return OK;
+ }
+ return mErrorCode;
+}
+
+void MediaCodecSource::signalBufferReturned(MediaBuffer *buffer) {
+ buffer->setObserver(0);
+ buffer->release();
+}
+
+MediaCodecSource::MediaCodecSource(
+ const sp<ALooper> &looper,
+ const sp<AMessage> &outputFormat,
+ const sp<MediaSource> &source,
+ uint32_t flags)
+ : mLooper(looper),
+ mOutputFormat(outputFormat),
+ mMeta(new MetaData),
+ mFlags(flags),
+ mIsVideo(false),
+ mStarted(false),
+ mStopping(false),
+ mDoMoreWorkPending(false),
+ mPullerReachedEOS(false),
+ mFirstSampleTimeUs(-1ll),
+ mEncodedReachedEOS(false),
+ mErrorCode(OK) {
+ CHECK(mLooper != NULL);
+
+ AString mime;
+ CHECK(mOutputFormat->findString("mime", &mime));
+
+ if (!strncasecmp("video/", mime.c_str(), 6)) {
+ mIsVideo = true;
+ }
+
+ if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+ mPuller = new Puller(source);
+ }
+}
+
+MediaCodecSource::~MediaCodecSource() {
+ releaseEncoder();
+
+ mCodecLooper->stop();
+ mLooper->unregisterHandler(mReflector->id());
+}
+
+status_t MediaCodecSource::init() {
+ status_t err = initEncoder();
+
+ if (err != OK) {
+ releaseEncoder();
+ }
+
+ return err;
+}
+
+status_t MediaCodecSource::initEncoder() {
+ mReflector = new AHandlerReflector<MediaCodecSource>(this);
+ mLooper->registerHandler(mReflector);
+
+ mCodecLooper = new ALooper;
+ mCodecLooper->setName("codec_looper");
+ mCodecLooper->start();
+
+ if (mFlags & FLAG_USE_METADATA_INPUT) {
+ mOutputFormat->setInt32("store-metadata-in-buffers", 1);
+ }
+
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ mOutputFormat->setInt32("create-input-buffers-suspended", 1);
+ }
+
+ AString outputMIME;
+ CHECK(mOutputFormat->findString("mime", &outputMIME));
+
+ mEncoder = MediaCodec::CreateByType(
+ mCodecLooper, outputMIME.c_str(), true /* encoder */);
+
+ if (mEncoder == NULL) {
+ return NO_INIT;
+ }
+
+ ALOGV("output format is '%s'", mOutputFormat->debugString(0).c_str());
+
+ status_t err = mEncoder->configure(
+ mOutputFormat,
+ NULL /* nativeWindow */,
+ NULL /* crypto */,
+ MediaCodec::CONFIGURE_FLAG_ENCODE);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mEncoder->getOutputFormat(&mOutputFormat);
+ convertMessageToMetaData(mOutputFormat, mMeta);
+
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ CHECK(mIsVideo);
+
+ err = mEncoder->createInputSurface(&mGraphicBufferProducer);
+
+ if (err != OK) {
+ return err;
+ }
+ }
+
+ err = mEncoder->start();
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = mEncoder->getInputBuffers(&mEncoderInputBuffers);
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mEncodedReachedEOS = false;
+ mErrorCode = OK;
+
+ return OK;
+}
+
+void MediaCodecSource::releaseEncoder() {
+ if (mEncoder == NULL) {
+ return;
+ }
+
+ mEncoder->release();
+ mEncoder.clear();
+
+ while (!mInputBufferQueue.empty()) {
+ MediaBuffer *mbuf = *mInputBufferQueue.begin();
+ mInputBufferQueue.erase(mInputBufferQueue.begin());
+ if (mbuf != NULL) {
+ mbuf->release();
+ }
+ }
+
+ for (size_t i = 0; i < mEncoderInputBuffers.size(); ++i) {
+ sp<ABuffer> accessUnit = mEncoderInputBuffers.itemAt(i);
+ ReleaseMediaBufferReference(accessUnit);
+ }
+
+ mEncoderInputBuffers.clear();
+ mEncoderOutputBuffers.clear();
+}
+
+bool MediaCodecSource::reachedEOS() {
+ return mEncodedReachedEOS && ((mPuller == NULL) || mPullerReachedEOS);
+}
+
+status_t MediaCodecSource::postSynchronouslyAndReturnError(
+ const sp<AMessage> &msg) {
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!response->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+
+void MediaCodecSource::signalEOS(status_t err) {
+ if (!mEncodedReachedEOS) {
+ ALOGI("encoder (%s) reached EOS", mIsVideo ? "video" : "audio");
+ {
+ Mutex::Autolock autoLock(mOutputBufferLock);
+ // release all unread media buffers
+ for (List<MediaBuffer*>::iterator it = mOutputBufferQueue.begin();
+ it != mOutputBufferQueue.end(); it++) {
+ (*it)->release();
+ }
+ mOutputBufferQueue.clear();
+ mEncodedReachedEOS = true;
+ mErrorCode = err;
+ mOutputBufferCond.signal();
+ }
+
+ releaseEncoder();
+ }
+ if (mStopping && reachedEOS()) {
+ ALOGI("MediaCodecSource (%s) fully stopped",
+ mIsVideo ? "video" : "audio");
+ // posting reply to everyone that's waiting
+ List<uint32_t>::iterator it;
+ for (it = mStopReplyIDQueue.begin();
+ it != mStopReplyIDQueue.end(); it++) {
+ (new AMessage)->postReply(*it);
+ }
+ mStopReplyIDQueue.clear();
+ mStopping = false;
+ }
+}
+
+void MediaCodecSource::suspend() {
+ CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+ if (mEncoder != NULL) {
+ sp<AMessage> params = new AMessage;
+ params->setInt32("drop-input-frames", true);
+ mEncoder->setParameters(params);
+ }
+}
+
+void MediaCodecSource::resume(int64_t skipFramesBeforeUs) {
+ CHECK(mFlags & FLAG_USE_SURFACE_INPUT);
+ if (mEncoder != NULL) {
+ sp<AMessage> params = new AMessage;
+ params->setInt32("drop-input-frames", false);
+ if (skipFramesBeforeUs > 0) {
+ params->setInt64("skip-frames-before", skipFramesBeforeUs);
+ }
+ mEncoder->setParameters(params);
+ }
+}
+
+void MediaCodecSource::scheduleDoMoreWork() {
+ if (mDoMoreWorkPending) {
+ return;
+ }
+
+ mDoMoreWorkPending = true;
+
+ if (mEncoderActivityNotify == NULL) {
+ mEncoderActivityNotify = new AMessage(
+ kWhatEncoderActivity, mReflector->id());
+ }
+ mEncoder->requestActivityNotification(mEncoderActivityNotify);
+}
+
+status_t MediaCodecSource::feedEncoderInputBuffers() {
+ while (!mInputBufferQueue.empty()
+ && !mAvailEncoderInputIndices.empty()) {
+ MediaBuffer* mbuf = *mInputBufferQueue.begin();
+ mInputBufferQueue.erase(mInputBufferQueue.begin());
+
+ size_t bufferIndex = *mAvailEncoderInputIndices.begin();
+ mAvailEncoderInputIndices.erase(mAvailEncoderInputIndices.begin());
+
+ int64_t timeUs = 0ll;
+ uint32_t flags = 0;
+ size_t size = 0;
+
+ if (mbuf != NULL) {
+ CHECK(mbuf->meta_data()->findInt64(kKeyTime, &timeUs));
+
+ // push decoding time for video, or drift time for audio
+ if (mIsVideo) {
+ mDecodingTimeQueue.push_back(timeUs);
+ } else {
+#if DEBUG_DRIFT_TIME
+ if (mFirstSampleTimeUs < 0ll) {
+ mFirstSampleTimeUs = timeUs;
+ }
+
+ int64_t driftTimeUs = 0;
+ if (mbuf->meta_data()->findInt64(kKeyDriftTime, &driftTimeUs)
+ && driftTimeUs) {
+ driftTimeUs = timeUs - mFirstSampleTimeUs - driftTimeUs;
+ }
+ mDriftTimeQueue.push_back(driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+ }
+
+ size = mbuf->size();
+
+ memcpy(mEncoderInputBuffers.itemAt(bufferIndex)->data(),
+ mbuf->data(), size);
+
+ if (mIsVideo) {
+ // video encoder will release MediaBuffer when done
+ // with underlying data.
+ mEncoderInputBuffers.itemAt(bufferIndex)->meta()
+ ->setPointer("mediaBuffer", mbuf);
+ } else {
+ mbuf->release();
+ }
+ } else {
+ flags = MediaCodec::BUFFER_FLAG_EOS;
+ }
+
+ status_t err = mEncoder->queueInputBuffer(
+ bufferIndex, 0, size, timeUs, flags);
+
+ if (err != OK) {
+ return err;
+ }
+ }
+
+ return OK;
+}
+
+status_t MediaCodecSource::doMoreWork() {
+ status_t err;
+
+ if (!(mFlags & FLAG_USE_SURFACE_INPUT)) {
+ for (;;) {
+ size_t bufferIndex;
+ err = mEncoder->dequeueInputBuffer(&bufferIndex);
+
+ if (err != OK) {
+ break;
+ }
+
+ mAvailEncoderInputIndices.push_back(bufferIndex);
+ }
+
+ feedEncoderInputBuffers();
+ }
+
+ for (;;) {
+ size_t bufferIndex;
+ size_t offset;
+ size_t size;
+ int64_t timeUs;
+ uint32_t flags;
+ native_handle_t* handle = NULL;
+ err = mEncoder->dequeueOutputBuffer(
+ &bufferIndex, &offset, &size, &timeUs, &flags);
+
+ if (err != OK) {
+ if (err == INFO_FORMAT_CHANGED) {
+ continue;
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ mEncoder->getOutputBuffers(&mEncoderOutputBuffers);
+ continue;
+ }
+
+ if (err == -EAGAIN) {
+ err = OK;
+ }
+ break;
+ }
+ if (!(flags & MediaCodec::BUFFER_FLAG_EOS)) {
+ sp<ABuffer> outbuf = mEncoderOutputBuffers.itemAt(bufferIndex);
+
+ MediaBuffer *mbuf = new MediaBuffer(outbuf->size());
+ memcpy(mbuf->data(), outbuf->data(), outbuf->size());
+
+ if (!(flags & MediaCodec::BUFFER_FLAG_CODECCONFIG)) {
+ if (mIsVideo) {
+ int64_t decodingTimeUs;
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ // GraphicBufferSource is supposed to discard samples
+ // queued before start, and offset timeUs by start time
+ CHECK_GE(timeUs, 0ll);
+ // TODO:
+ // Decoding time for surface source is unavailable,
+ // use presentation time for now. May need to move
+ // this logic into MediaCodec.
+ decodingTimeUs = timeUs;
+ } else {
+ CHECK(!mDecodingTimeQueue.empty());
+ decodingTimeUs = *(mDecodingTimeQueue.begin());
+ mDecodingTimeQueue.erase(mDecodingTimeQueue.begin());
+ }
+ mbuf->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
+
+ ALOGV("[video] time %lld us (%.2f secs), dts/pts diff %lld",
+ timeUs, timeUs / 1E6, decodingTimeUs - timeUs);
+ } else {
+ int64_t driftTimeUs = 0;
+#if DEBUG_DRIFT_TIME
+ CHECK(!mDriftTimeQueue.empty());
+ driftTimeUs = *(mDriftTimeQueue.begin());
+ mDriftTimeQueue.erase(mDriftTimeQueue.begin());
+ mbuf->meta_data()->setInt64(kKeyDriftTime, driftTimeUs);
+#endif // DEBUG_DRIFT_TIME
+ ALOGV("[audio] time %lld us (%.2f secs), drift %lld",
+ timeUs, timeUs / 1E6, driftTimeUs);
+ }
+ mbuf->meta_data()->setInt64(kKeyTime, timeUs);
+ } else {
+ mbuf->meta_data()->setInt32(kKeyIsCodecConfig, true);
+ }
+ if (flags & MediaCodec::BUFFER_FLAG_SYNCFRAME) {
+ mbuf->meta_data()->setInt32(kKeyIsSyncFrame, true);
+ }
+ mbuf->setObserver(this);
+ mbuf->add_ref();
+
+ {
+ Mutex::Autolock autoLock(mOutputBufferLock);
+ mOutputBufferQueue.push_back(mbuf);
+ mOutputBufferCond.signal();
+ }
+ }
+
+ mEncoder->releaseOutputBuffer(bufferIndex);
+
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ err = ERROR_END_OF_STREAM;
+ break;
+ }
+ }
+
+ return err;
+}
+
+status_t MediaCodecSource::onStart(MetaData *params) {
+ if (mStopping) {
+ ALOGE("Failed to start while we're stopping");
+ return INVALID_OPERATION;
+ }
+
+ if (mStarted) {
+ ALOGI("MediaCodecSource (%s) resuming", mIsVideo ? "video" : "audio");
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ resume();
+ } else {
+ CHECK(mPuller != NULL);
+ mPuller->resume();
+ }
+ return OK;
+ }
+
+ ALOGI("MediaCodecSource (%s) starting", mIsVideo ? "video" : "audio");
+
+ status_t err = OK;
+
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ int64_t startTimeUs;
+ if (!params || !params->findInt64(kKeyTime, &startTimeUs)) {
+ startTimeUs = -1ll;
+ }
+ resume(startTimeUs);
+ scheduleDoMoreWork();
+ } else {
+ CHECK(mPuller != NULL);
+ sp<AMessage> notify = new AMessage(
+ kWhatPullerNotify, mReflector->id());
+ err = mPuller->start(params, notify);
+ if (err != OK) {
+ mPullerReachedEOS = true;
+ return err;
+ }
+ }
+
+ ALOGI("MediaCodecSource (%s) started", mIsVideo ? "video" : "audio");
+
+ mStarted = true;
+ return OK;
+}
+
+void MediaCodecSource::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatPullerNotify:
+ {
+ MediaBuffer *mbuf;
+ CHECK(msg->findPointer("accessUnit", (void**)&mbuf));
+
+ if (mbuf == NULL) {
+ ALOGI("puller (%s) reached EOS",
+ mIsVideo ? "video" : "audio");
+ mPullerReachedEOS = true;
+ }
+
+ if (mEncoder == NULL) {
+ ALOGV("got msg '%s' after encoder shutdown.",
+ msg->debugString().c_str());
+
+ if (mbuf != NULL) {
+ mbuf->release();
+ } else {
+ signalEOS();
+ }
+ break;
+ }
+
+ mInputBufferQueue.push_back(mbuf);
+
+ feedEncoderInputBuffers();
+ scheduleDoMoreWork();
+
+ break;
+ }
+ case kWhatEncoderActivity:
+ {
+ mDoMoreWorkPending = false;
+
+ if (mEncoder == NULL) {
+ break;
+ }
+
+ status_t err = doMoreWork();
+
+ if (err == OK) {
+ scheduleDoMoreWork();
+ } else {
+ // reached EOS, or error
+ signalEOS(err);
+ }
+
+ break;
+ }
+ case kWhatStart:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<RefBase> obj;
+ CHECK(msg->findObject("meta", &obj));
+ MetaData *params = static_cast<MetaData *>(obj.get());
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", onStart(params));
+ response->postReply(replyID);
+ break;
+ }
+ case kWhatStop:
+ {
+ ALOGI("MediaCodecSource (%s) stopping", mIsVideo ? "video" : "audio");
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (reachedEOS()) {
+ // if we already reached EOS, reply and return now
+ ALOGI("MediaCodecSource (%s) already stopped",
+ mIsVideo ? "video" : "audio");
+ (new AMessage)->postReply(replyID);
+ break;
+ }
+
+ mStopReplyIDQueue.push_back(replyID);
+ if (mStopping) {
+ // nothing to do if we're already stopping, reply will be posted
+ // to all when we're stopped.
+ break;
+ }
+
+ mStopping = true;
+
+ // if using surface, signal source EOS and wait for EOS to come back.
+ // otherwise, release encoder and post EOS if haven't done already
+ if (mFlags & FLAG_USE_SURFACE_INPUT) {
+ mEncoder->signalEndOfInputStream();
+ } else {
+ CHECK(mPuller != NULL);
+ mPuller->stopAsync();
+ signalEOS();
+ }
+ break;
+ }
+ case kWhatPause:
+ {
+ if (mFlags && FLAG_USE_SURFACE_INPUT) {
+ suspend();
+ } else {
+ CHECK(mPuller != NULL);
+ mPuller->pause();
+ }
+ break;
+ }
+ default:
+ TRESPASS();
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 05e599b..1287fb1 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -213,7 +213,14 @@
mLooper->setName("NuCachedSource2");
mLooper->registerHandler(mReflector);
- mLooper->start();
+
+ // Since it may not be obvious why our looper thread needs to be
+ // able to call into java since it doesn't appear to do so at all...
+ // IMediaHTTPConnection may be (and most likely is) implemented in JAVA
+ // and a local JAVA IBinder will call directly into JNI methods.
+ // So whenever we call DataSource::readAt it may end up in a call to
+ // IMediaHTTPConnection::readAt and therefore call back into JAVA.
+ mLooper->start(false /* runOnCallingThread */, true /* canCallJava */);
Mutex::Autolock autoLock(mLock);
(new AMessage(kWhatFetchMore, mReflector->id()))->post();
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 7bc7da2..64f56e9 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -58,7 +58,9 @@
}
status_t NuMediaExtractor::setDataSource(
- const char *path, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *path,
+ const KeyedVector<String8, String8> *headers) {
Mutex::Autolock autoLock(mLock);
if (mImpl != NULL) {
@@ -66,7 +68,7 @@
}
sp<DataSource> dataSource =
- DataSource::CreateFromURI(path, headers);
+ DataSource::CreateFromURI(httpService, path, headers);
if (dataSource == NULL) {
return -ENOENT;
diff --git a/media/libstagefright/SkipCutBuffer.cpp b/media/libstagefright/SkipCutBuffer.cpp
index 773854f..e2e6d79 100644
--- a/media/libstagefright/SkipCutBuffer.cpp
+++ b/media/libstagefright/SkipCutBuffer.cpp
@@ -25,7 +25,7 @@
namespace android {
SkipCutBuffer::SkipCutBuffer(int32_t skip, int32_t cut) {
- mFrontPadding = skip;
+ mFrontPadding = mSkip = skip;
mBackPadding = cut;
mWriteHead = 0;
mReadHead = 0;
@@ -94,6 +94,7 @@
void SkipCutBuffer::clear() {
mWriteHead = mReadHead = 0;
+ mFrontPadding = mSkip;
}
void SkipCutBuffer::write(const char *src, size_t num) {
diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index af8186c..edd12fc 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -24,6 +24,7 @@
#include <media/stagefright/StagefrightMediaScanner.h>
+#include <media/IMediaHTTPService.h>
#include <media/mediametadataretriever.h>
#include <private/media/VideoFrame.h>
@@ -147,7 +148,7 @@
status_t status;
if (fd < 0) {
// couldn't open it locally, maybe the media server can?
- status = mRetriever->setDataSource(path);
+ status = mRetriever->setDataSource(NULL /* httpService */, path);
} else {
status = mRetriever->setDataSource(fd, 0, 0x7ffffffffffffffL);
close(fd);
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index fcd9a85..9475d05 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -21,6 +21,7 @@
#include "include/StagefrightMetadataRetriever.h"
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
#include <media/stagefright/DataSource.h>
@@ -51,7 +52,9 @@
}
status_t StagefrightMetadataRetriever::setDataSource(
- const char *uri, const KeyedVector<String8, String8> *headers) {
+ const sp<IMediaHTTPService> &httpService,
+ const char *uri,
+ const KeyedVector<String8, String8> *headers) {
ALOGV("setDataSource(%s)", uri);
mParsedMetaData = false;
@@ -59,7 +62,7 @@
delete mAlbumArt;
mAlbumArt = NULL;
- mSource = DataSource::CreateFromURI(uri, headers);
+ mSource = DataSource::CreateFromURI(httpService, uri, headers);
if (mSource == NULL) {
ALOGE("Unable to create data source for '%s'.", uri);
diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
deleted file mode 100644
index f26f386..0000000
--- a/media/libstagefright/chromium_http/Android.mk
+++ /dev/null
@@ -1,37 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-ifneq ($(TARGET_BUILD_PDK), true)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- DataUriSource.cpp \
- ChromiumHTTPDataSource.cpp \
- support.cpp \
- chromium_http_stub.cpp
-
-LOCAL_C_INCLUDES:= \
- $(TOP)/frameworks/av/media/libstagefright \
- $(TOP)/frameworks/native/include/media/openmax \
- external/chromium \
- external/chromium/android
-
-LOCAL_CFLAGS += -Wno-multichar
-
-LOCAL_SHARED_LIBRARIES += \
- libstlport \
- libchromium_net \
- libutils \
- libcutils \
- liblog \
- libstagefright_foundation \
- libstagefright \
- libdrmframework
-
-include external/stlport/libstlport.mk
-
-LOCAL_MODULE:= libstagefright_chromium_http
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-endif
diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
deleted file mode 100644
index 7e5c280..0000000
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ /dev/null
@@ -1,355 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSource"
-#include <media/stagefright/foundation/ADebug.h>
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/MediaErrors.h>
-
-#include "support.h"
-
-#include <cutils/properties.h> // for property_get
-
-namespace android {
-
-ChromiumHTTPDataSource::ChromiumHTTPDataSource(uint32_t flags)
- : mFlags(flags),
- mState(DISCONNECTED),
- mDelegate(new SfDelegate),
- mCurrentOffset(0),
- mIOResult(OK),
- mContentSize(-1),
- mDecryptHandle(NULL),
- mDrmManagerClient(NULL) {
- mDelegate->setOwner(this);
-}
-
-ChromiumHTTPDataSource::~ChromiumHTTPDataSource() {
- disconnect();
-
- delete mDelegate;
- mDelegate = NULL;
-
- clearDRMState_l();
-
- if (mDrmManagerClient != NULL) {
- delete mDrmManagerClient;
- mDrmManagerClient = NULL;
- }
-}
-
-status_t ChromiumHTTPDataSource::connect(
- const char *uri,
- const KeyedVector<String8, String8> *headers,
- off64_t offset) {
- Mutex::Autolock autoLock(mLock);
-
- uid_t uid;
- if (getUID(&uid)) {
- mDelegate->setUID(uid);
- }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
- LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "connect on behalf of uid %d", uid);
-#endif
-
- return connect_l(uri, headers, offset);
-}
-
-status_t ChromiumHTTPDataSource::connect_l(
- const char *uri,
- const KeyedVector<String8, String8> *headers,
- off64_t offset) {
- if (mState != DISCONNECTED) {
- disconnect_l();
- }
-
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
- LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
- "connect to <URL suppressed> @%lld", offset);
-#endif
-
- mURI = uri;
- mContentType = String8("application/octet-stream");
-
- if (headers != NULL) {
- mHeaders = *headers;
- } else {
- mHeaders.clear();
- }
-
- mState = CONNECTING;
- mContentSize = -1;
- mCurrentOffset = offset;
-
- mDelegate->initiateConnection(mURI.c_str(), &mHeaders, offset);
-
- while (mState == CONNECTING || mState == DISCONNECTING) {
- mCondition.wait(mLock);
- }
-
- return mState == CONNECTED ? OK : mIOResult;
-}
-
-void ChromiumHTTPDataSource::onRedirect(const char *url) {
- Mutex::Autolock autoLock(mLock);
- mURI = url;
-}
-
-void ChromiumHTTPDataSource::onConnectionEstablished(
- int64_t contentSize, const char *contentType) {
- Mutex::Autolock autoLock(mLock);
-
- if (mState != CONNECTING) {
- // We may have initiated disconnection.
- CHECK_EQ(mState, DISCONNECTING);
- return;
- }
-
- mState = CONNECTED;
- mContentSize = (contentSize < 0) ? -1 : contentSize + mCurrentOffset;
- mContentType = String8(contentType);
- mCondition.broadcast();
-}
-
-void ChromiumHTTPDataSource::onConnectionFailed(status_t err) {
- Mutex::Autolock autoLock(mLock);
- mState = DISCONNECTED;
- mCondition.broadcast();
-
- // mURI.clear();
-
- mIOResult = err;
-}
-
-void ChromiumHTTPDataSource::disconnect() {
- Mutex::Autolock autoLock(mLock);
- disconnect_l();
-}
-
-void ChromiumHTTPDataSource::disconnect_l() {
- if (mState == DISCONNECTED) {
- return;
- }
-
- mState = DISCONNECTING;
- mIOResult = -EINTR;
-
- mDelegate->initiateDisconnect();
-
- while (mState == DISCONNECTING) {
- mCondition.wait(mLock);
- }
-
- CHECK_EQ((int)mState, (int)DISCONNECTED);
-}
-
-status_t ChromiumHTTPDataSource::initCheck() const {
- Mutex::Autolock autoLock(mLock);
-
- return mState == CONNECTED ? OK : NO_INIT;
-}
-
-ssize_t ChromiumHTTPDataSource::readAt(off64_t offset, void *data, size_t size) {
- Mutex::Autolock autoLock(mLock);
-
- if (mState != CONNECTED) {
- return INVALID_OPERATION;
- }
-
-#if 0
- char value[PROPERTY_VALUE_MAX];
- if (property_get("media.stagefright.disable-net", value, 0)
- && (!strcasecmp(value, "true") || !strcmp(value, "1"))) {
- LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Simulating that the network is down.");
- disconnect_l();
- return ERROR_IO;
- }
-#endif
-
- if (offset != mCurrentOffset) {
- AString tmp = mURI;
- KeyedVector<String8, String8> tmpHeaders = mHeaders;
-
- disconnect_l();
-
- status_t err = connect_l(tmp.c_str(), &tmpHeaders, offset);
-
- if (err != OK) {
- return err;
- }
- }
-
- mState = READING;
-
- int64_t startTimeUs = ALooper::GetNowUs();
-
- mDelegate->initiateRead(data, size);
-
- while (mState == READING) {
- mCondition.wait(mLock);
- }
-
- if (mIOResult < OK) {
- return mIOResult;
- }
-
- if (mState == CONNECTED) {
- int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
-
- // The read operation was successful, mIOResult contains
- // the number of bytes read.
- addBandwidthMeasurement(mIOResult, delayUs);
-
- mCurrentOffset += mIOResult;
- return mIOResult;
- }
-
- return ERROR_IO;
-}
-
-void ChromiumHTTPDataSource::onReadCompleted(ssize_t size) {
- Mutex::Autolock autoLock(mLock);
-
- mIOResult = size;
-
- if (mState == READING) {
- mState = CONNECTED;
- mCondition.broadcast();
- }
-}
-
-status_t ChromiumHTTPDataSource::getSize(off64_t *size) {
- Mutex::Autolock autoLock(mLock);
-
- if (mContentSize < 0) {
- return ERROR_UNSUPPORTED;
- }
-
- *size = mContentSize;
-
- return OK;
-}
-
-uint32_t ChromiumHTTPDataSource::flags() {
- return kWantsPrefetching | kIsHTTPBasedSource;
-}
-
-// static
-void ChromiumHTTPDataSource::InitiateRead(
- ChromiumHTTPDataSource *me, void *data, size_t size) {
- me->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::initiateRead(void *data, size_t size) {
- mDelegate->initiateRead(data, size);
-}
-
-void ChromiumHTTPDataSource::onDisconnectComplete() {
- Mutex::Autolock autoLock(mLock);
- CHECK_EQ((int)mState, (int)DISCONNECTING);
-
- mState = DISCONNECTED;
- // mURI.clear();
- mIOResult = -ENOTCONN;
-
- mCondition.broadcast();
-}
-
-sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization(const char* mime) {
- Mutex::Autolock autoLock(mLock);
-
- if (mDrmManagerClient == NULL) {
- mDrmManagerClient = new DrmManagerClient();
- }
-
- if (mDrmManagerClient == NULL) {
- return NULL;
- }
-
- if (mDecryptHandle == NULL) {
- /* Note if redirect occurs, mUri is the redirect uri instead of the
- * original one
- */
- mDecryptHandle = mDrmManagerClient->openDecryptSession(
- String8(mURI.c_str()), mime);
- }
-
- if (mDecryptHandle == NULL) {
- delete mDrmManagerClient;
- mDrmManagerClient = NULL;
- }
-
- return mDecryptHandle;
-}
-
-void ChromiumHTTPDataSource::getDrmInfo(
- sp<DecryptHandle> &handle, DrmManagerClient **client) {
- Mutex::Autolock autoLock(mLock);
-
- handle = mDecryptHandle;
- *client = mDrmManagerClient;
-}
-
-String8 ChromiumHTTPDataSource::getUri() {
- Mutex::Autolock autoLock(mLock);
-
- return String8(mURI.c_str());
-}
-
-String8 ChromiumHTTPDataSource::getMIMEType() const {
- Mutex::Autolock autoLock(mLock);
-
- return mContentType;
-}
-
-void ChromiumHTTPDataSource::clearDRMState_l() {
- if (mDecryptHandle != NULL) {
- // To release mDecryptHandle
- CHECK(mDrmManagerClient);
- mDrmManagerClient->closeDecryptSession(mDecryptHandle);
- mDecryptHandle = NULL;
- }
-}
-
-status_t ChromiumHTTPDataSource::reconnectAtOffset(off64_t offset) {
- Mutex::Autolock autoLock(mLock);
-
- if (mURI.empty()) {
- return INVALID_OPERATION;
- }
-
- LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnecting...");
- status_t err = connect_l(mURI.c_str(), &mHeaders, offset);
- if (err != OK) {
- LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "Reconnect failed w/ err 0x%08x", err);
- }
-
- return err;
-}
-
-// static
-status_t ChromiumHTTPDataSource::UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- return SfDelegate::UpdateProxyConfig(host, port, exclusionList);
-}
-
-} // namespace android
-
diff --git a/media/libstagefright/chromium_http/DataUriSource.cpp b/media/libstagefright/chromium_http/DataUriSource.cpp
deleted file mode 100644
index ecf3fa1..0000000
--- a/media/libstagefright/chromium_http/DataUriSource.cpp
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <include/DataUriSource.h>
-
-#include <net/base/data_url.h>
-#include <googleurl/src/gurl.h>
-
-
-namespace android {
-
-DataUriSource::DataUriSource(const char *uri) :
- mDataUri(uri),
- mInited(NO_INIT) {
-
- // Copy1: const char *uri -> String8 mDataUri.
- std::string mimeTypeStr, unusedCharsetStr, dataStr;
- // Copy2: String8 mDataUri -> std::string
- const bool ret = net::DataURL::Parse(
- GURL(std::string(mDataUri.string())),
- &mimeTypeStr, &unusedCharsetStr, &dataStr);
- // Copy3: std::string dataStr -> AString mData
- mData.setTo(dataStr.data(), dataStr.length());
- mInited = ret ? OK : UNKNOWN_ERROR;
-
- // The chromium data url implementation defaults to using "text/plain"
- // if no mime type is specified. We prefer to leave this unspecified
- // instead, since the mime type is sniffed in most cases.
- if (mimeTypeStr != "text/plain") {
- mMimeType = mimeTypeStr.c_str();
- }
-}
-
-ssize_t DataUriSource::readAt(off64_t offset, void *out, size_t size) {
- if (mInited != OK) {
- return mInited;
- }
-
- const off64_t length = mData.size();
- if (offset >= length) {
- return UNKNOWN_ERROR;
- }
-
- const char *dataBuf = mData.c_str();
- const size_t bytesToCopy =
- offset + size >= length ? (length - offset) : size;
-
- if (bytesToCopy > 0) {
- memcpy(out, dataBuf + offset, bytesToCopy);
- }
-
- return bytesToCopy;
-}
-
-} // namespace android
diff --git a/media/libstagefright/chromium_http/chromium_http_stub.cpp b/media/libstagefright/chromium_http/chromium_http_stub.cpp
deleted file mode 100644
index 289f6de..0000000
--- a/media/libstagefright/chromium_http/chromium_http_stub.cpp
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <dlfcn.h>
-
-#include <include/chromium_http_stub.h>
-#include <include/ChromiumHTTPDataSource.h>
-#include <include/DataUriSource.h>
-
-namespace android {
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
- return new ChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- return ChromiumHTTPDataSource::UpdateProxyConfig(host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
- return new DataUriSource(uri);
-}
-
-}
diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
deleted file mode 100644
index 3b33212..0000000
--- a/media/libstagefright/chromium_http/support.cpp
+++ /dev/null
@@ -1,559 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ChromiumHTTPDataSourceSupport"
-#include <utils/Log.h>
-
-#include <media/stagefright/foundation/AString.h>
-
-#include "support.h"
-
-#include "android/net/android_network_library_impl.h"
-#include "base/logging.h"
-#include "base/threading/thread.h"
-#include "net/base/cert_verifier.h"
-#include "net/base/cookie_monster.h"
-#include "net/base/host_resolver.h"
-#include "net/base/ssl_config_service.h"
-#include "net/http/http_auth_handler_factory.h"
-#include "net/http/http_cache.h"
-#include "net/proxy/proxy_config_service_android.h"
-
-#include "include/ChromiumHTTPDataSource.h"
-
-#include <cutils/log.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <string>
-
-namespace android {
-
-static Mutex gNetworkThreadLock;
-static base::Thread *gNetworkThread = NULL;
-static scoped_refptr<SfRequestContext> gReqContext;
-static scoped_ptr<net::NetworkChangeNotifier> gNetworkChangeNotifier;
-
-bool logMessageHandler(
- int severity,
- const char* file,
- int line,
- size_t message_start,
- const std::string& str) {
- int androidSeverity = ANDROID_LOG_VERBOSE;
- switch(severity) {
- case logging::LOG_FATAL:
- androidSeverity = ANDROID_LOG_FATAL;
- break;
- case logging::LOG_ERROR_REPORT:
- case logging::LOG_ERROR:
- androidSeverity = ANDROID_LOG_ERROR;
- break;
- case logging::LOG_WARNING:
- androidSeverity = ANDROID_LOG_WARN;
- break;
- default:
- androidSeverity = ANDROID_LOG_VERBOSE;
- break;
- }
- android_printLog(androidSeverity, "chromium-libstagefright",
- "%s:%d: %s", file, line, str.c_str());
- return false;
-}
-
-struct AutoPrioritySaver {
- AutoPrioritySaver()
- : mTID(androidGetTid()),
- mPrevPriority(androidGetThreadPriority(mTID)) {
- androidSetThreadPriority(mTID, ANDROID_PRIORITY_NORMAL);
- }
-
- ~AutoPrioritySaver() {
- androidSetThreadPriority(mTID, mPrevPriority);
- }
-
-private:
- pid_t mTID;
- int mPrevPriority;
-
- DISALLOW_EVIL_CONSTRUCTORS(AutoPrioritySaver);
-};
-
-static void InitializeNetworkThreadIfNecessary() {
- Mutex::Autolock autoLock(gNetworkThreadLock);
-
- if (gNetworkThread == NULL) {
- // Make sure any threads spawned by the chromium framework are
- // running at normal priority instead of inheriting this thread's.
- AutoPrioritySaver saver;
-
- gNetworkThread = new base::Thread("network");
- base::Thread::Options options;
- options.message_loop_type = MessageLoop::TYPE_IO;
- CHECK(gNetworkThread->StartWithOptions(options));
-
- gReqContext = new SfRequestContext;
-
- gNetworkChangeNotifier.reset(net::NetworkChangeNotifier::Create());
-
- net::AndroidNetworkLibrary::RegisterSharedInstance(
- new SfNetworkLibrary);
- logging::SetLogMessageHandler(logMessageHandler);
- }
-}
-
-static void MY_LOGI(const char *s) {
- LOG_PRI(ANDROID_LOG_INFO, LOG_TAG, "%s", s);
-}
-
-static void MY_LOGV(const char *s) {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
- LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG, "%s", s);
-#endif
-}
-
-SfNetLog::SfNetLog()
- : mNextID(1) {
-}
-
-void SfNetLog::AddEntry(
- EventType type,
- const base::TimeTicks &time,
- const Source &source,
- EventPhase phase,
- EventParameters *params) {
-#if 0
- MY_LOGI(StringPrintf(
- "AddEntry time=%s type=%s source=%s phase=%s\n",
- TickCountToString(time).c_str(),
- EventTypeToString(type),
- SourceTypeToString(source.type),
- EventPhaseToString(phase)).c_str());
-#endif
-}
-
-uint32 SfNetLog::NextID() {
- return mNextID++;
-}
-
-net::NetLog::LogLevel SfNetLog::GetLogLevel() const {
- return LOG_BASIC;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfRequestContext::SfRequestContext() {
- mUserAgent = MakeUserAgent().c_str();
-
- set_net_log(new SfNetLog());
-
- set_host_resolver(
- net::CreateSystemHostResolver(
- net::HostResolver::kDefaultParallelism,
- NULL /* resolver_proc */,
- net_log()));
-
- set_ssl_config_service(
- net::SSLConfigService::CreateSystemSSLConfigService());
-
- mProxyConfigService = new net::ProxyConfigServiceAndroid;
-
- set_proxy_service(net::ProxyService::CreateWithoutProxyResolver(
- mProxyConfigService, net_log()));
-
- set_http_transaction_factory(new net::HttpCache(
- host_resolver(),
- new net::CertVerifier(),
- dnsrr_resolver(),
- dns_cert_checker(),
- proxy_service(),
- ssl_config_service(),
- net::HttpAuthHandlerFactory::CreateDefault(host_resolver()),
- network_delegate(),
- net_log(),
- NULL)); // backend_factory
-
- set_cookie_store(new net::CookieMonster(NULL, NULL));
-}
-
-const std::string &SfRequestContext::GetUserAgent(const GURL &url) const {
- return mUserAgent;
-}
-
-status_t SfRequestContext::updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- Mutex::Autolock autoLock(mProxyConfigLock);
-
- if (host == NULL || *host == '\0') {
- MY_LOGV("updateProxyConfig NULL");
-
- std::string proxy;
- std::string exList;
- mProxyConfigService->UpdateProxySettings(proxy, exList);
- } else {
-#if !defined(LOG_NDEBUG) || LOG_NDEBUG == 0
- LOG_PRI(ANDROID_LOG_VERBOSE, LOG_TAG,
- "updateProxyConfig %s:%d, exclude '%s'",
- host, port, exclusionList);
-#endif
-
- std::string proxy = StringPrintf("%s:%d", host, port).c_str();
- std::string exList = exclusionList;
- mProxyConfigService->UpdateProxySettings(proxy, exList);
- }
-
- return OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfNetworkLibrary::SfNetworkLibrary() {}
-
-SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain(
- const std::vector<std::string>& cert_chain,
- const std::string& hostname,
- const std::string& auth_type) {
- return VERIFY_OK;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-
-SfDelegate::SfDelegate()
- : mOwner(NULL),
- mURLRequest(NULL),
- mReadBuffer(new net::IOBufferWithSize(8192)),
- mNumBytesRead(0),
- mNumBytesTotal(0),
- mDataDestination(NULL),
- mAtEOS(false) {
- InitializeNetworkThreadIfNecessary();
-}
-
-SfDelegate::~SfDelegate() {
- CHECK(mURLRequest == NULL);
-}
-
-// static
-status_t SfDelegate::UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- InitializeNetworkThreadIfNecessary();
-
- return gReqContext->updateProxyConfig(host, port, exclusionList);
-}
-
-void SfDelegate::setOwner(ChromiumHTTPDataSource *owner) {
- mOwner = owner;
-}
-
-void SfDelegate::setUID(uid_t uid) {
- gReqContext->setUID(uid);
-}
-
-bool SfDelegate::getUID(uid_t *uid) const {
- return gReqContext->getUID(uid);
-}
-
-void SfDelegate::OnReceivedRedirect(
- net::URLRequest *request, const GURL &new_url, bool *defer_redirect) {
- MY_LOGV("OnReceivedRedirect");
- mOwner->onRedirect(new_url.spec().c_str());
-}
-
-void SfDelegate::OnAuthRequired(
- net::URLRequest *request, net::AuthChallengeInfo *auth_info) {
- MY_LOGV("OnAuthRequired");
-
- inherited::OnAuthRequired(request, auth_info);
-}
-
-void SfDelegate::OnCertificateRequested(
- net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info) {
- MY_LOGV("OnCertificateRequested");
-
- inherited::OnCertificateRequested(request, cert_request_info);
-}
-
-void SfDelegate::OnSSLCertificateError(
- net::URLRequest *request, int cert_error, net::X509Certificate *cert) {
- fprintf(stderr, "OnSSLCertificateError cert_error=%d\n", cert_error);
-
- inherited::OnSSLCertificateError(request, cert_error, cert);
-}
-
-void SfDelegate::OnGetCookies(net::URLRequest *request, bool blocked_by_policy) {
- MY_LOGV("OnGetCookies");
-}
-
-void SfDelegate::OnSetCookie(
- net::URLRequest *request,
- const std::string &cookie_line,
- const net::CookieOptions &options,
- bool blocked_by_policy) {
- MY_LOGV("OnSetCookie");
-}
-
-void SfDelegate::OnResponseStarted(net::URLRequest *request) {
- if (request->status().status() != net::URLRequestStatus::SUCCESS) {
- MY_LOGI(StringPrintf(
- "Request failed with status %d and os_error %d",
- request->status().status(),
- request->status().os_error()).c_str());
-
- delete mURLRequest;
- mURLRequest = NULL;
-
- mOwner->onConnectionFailed(ERROR_IO);
- return;
- } else if (mRangeRequested && request->GetResponseCode() != 206) {
- MY_LOGI(StringPrintf(
- "We requested a content range, but server didn't "
- "support that. (responded with %d)",
- request->GetResponseCode()).c_str());
-
- delete mURLRequest;
- mURLRequest = NULL;
-
- mOwner->onConnectionFailed(-EPIPE);
- return;
- } else if ((request->GetResponseCode() / 100) != 2) {
- MY_LOGI(StringPrintf(
- "Server responded with http status %d",
- request->GetResponseCode()).c_str());
-
- delete mURLRequest;
- mURLRequest = NULL;
-
- mOwner->onConnectionFailed(ERROR_IO);
- return;
- }
-
- MY_LOGV("OnResponseStarted");
-
- std::string headers;
- request->GetAllResponseHeaders(&headers);
-
- MY_LOGV(StringPrintf("response headers: %s", headers.c_str()).c_str());
-
- std::string contentType;
- request->GetResponseHeaderByName("Content-Type", &contentType);
-
- mOwner->onConnectionEstablished(
- request->GetExpectedContentSize(), contentType.c_str());
-}
-
-void SfDelegate::OnReadCompleted(net::URLRequest *request, int bytes_read) {
- if (bytes_read == -1) {
- MY_LOGI(StringPrintf(
- "OnReadCompleted, read failed, status %d",
- request->status().status()).c_str());
-
- mOwner->onReadCompleted(ERROR_IO);
- return;
- }
-
- MY_LOGV(StringPrintf("OnReadCompleted, read %d bytes", bytes_read).c_str());
-
- if (bytes_read < 0) {
- MY_LOGI(StringPrintf(
- "Read failed w/ status %d\n",
- request->status().status()).c_str());
-
- mOwner->onReadCompleted(ERROR_IO);
- return;
- } else if (bytes_read == 0) {
- mAtEOS = true;
- mOwner->onReadCompleted(mNumBytesRead);
- return;
- }
-
- CHECK_GT(bytes_read, 0);
- CHECK_LE(mNumBytesRead + bytes_read, mNumBytesTotal);
-
- memcpy((uint8_t *)mDataDestination + mNumBytesRead,
- mReadBuffer->data(),
- bytes_read);
-
- mNumBytesRead += bytes_read;
-
- readMore(request);
-}
-
-void SfDelegate::readMore(net::URLRequest *request) {
- while (mNumBytesRead < mNumBytesTotal) {
- size_t copy = mNumBytesTotal - mNumBytesRead;
- if (copy > mReadBuffer->size()) {
- copy = mReadBuffer->size();
- }
-
- int n;
- if (request->Read(mReadBuffer, copy, &n)) {
- MY_LOGV(StringPrintf("Read %d bytes directly.", n).c_str());
-
- CHECK_LE((size_t)n, copy);
-
- memcpy((uint8_t *)mDataDestination + mNumBytesRead,
- mReadBuffer->data(),
- n);
-
- mNumBytesRead += n;
-
- if (n == 0) {
- mAtEOS = true;
- break;
- }
- } else {
- MY_LOGV("readMore pending read");
-
- if (request->status().status() != net::URLRequestStatus::IO_PENDING) {
- MY_LOGI(StringPrintf(
- "Direct read failed w/ status %d\n",
- request->status().status()).c_str());
-
- mOwner->onReadCompleted(ERROR_IO);
- return;
- }
-
- return;
- }
- }
-
- mOwner->onReadCompleted(mNumBytesRead);
-}
-
-void SfDelegate::initiateConnection(
- const char *uri,
- const KeyedVector<String8, String8> *headers,
- off64_t offset) {
- GURL url(uri);
-
- MessageLoop *loop = gNetworkThread->message_loop();
- loop->PostTask(
- FROM_HERE,
- NewRunnableFunction(
- &SfDelegate::OnInitiateConnectionWrapper,
- this,
- url,
- headers,
- offset));
-
-}
-
-// static
-void SfDelegate::OnInitiateConnectionWrapper(
- SfDelegate *me, GURL url,
- const KeyedVector<String8, String8> *headers,
- off64_t offset) {
- me->onInitiateConnection(url, headers, offset);
-}
-
-void SfDelegate::onInitiateConnection(
- const GURL &url,
- const KeyedVector<String8, String8> *extra,
- off64_t offset) {
- CHECK(mURLRequest == NULL);
-
- mURLRequest = new net::URLRequest(url, this);
- mAtEOS = false;
-
- mRangeRequested = false;
-
- if (offset != 0 || extra != NULL) {
- net::HttpRequestHeaders headers =
- mURLRequest->extra_request_headers();
-
- if (offset != 0) {
- headers.AddHeaderFromString(
- StringPrintf("Range: bytes=%lld-", offset).c_str());
-
- mRangeRequested = true;
- }
-
- if (extra != NULL) {
- for (size_t i = 0; i < extra->size(); ++i) {
- AString s;
- s.append(extra->keyAt(i).string());
- s.append(": ");
- s.append(extra->valueAt(i).string());
-
- headers.AddHeaderFromString(s.c_str());
- }
- }
-
- mURLRequest->SetExtraRequestHeaders(headers);
- }
-
- mURLRequest->set_context(gReqContext);
-
- mURLRequest->Start();
-}
-
-void SfDelegate::initiateDisconnect() {
- MessageLoop *loop = gNetworkThread->message_loop();
- loop->PostTask(
- FROM_HERE,
- NewRunnableFunction(
- &SfDelegate::OnInitiateDisconnectWrapper, this));
-}
-
-// static
-void SfDelegate::OnInitiateDisconnectWrapper(SfDelegate *me) {
- me->onInitiateDisconnect();
-}
-
-void SfDelegate::onInitiateDisconnect() {
- if (mURLRequest == NULL) {
- return;
- }
-
- mURLRequest->Cancel();
-
- delete mURLRequest;
- mURLRequest = NULL;
-
- mOwner->onDisconnectComplete();
-}
-
-void SfDelegate::initiateRead(void *data, size_t size) {
- MessageLoop *loop = gNetworkThread->message_loop();
- loop->PostTask(
- FROM_HERE,
- NewRunnableFunction(
- &SfDelegate::OnInitiateReadWrapper, this, data, size));
-}
-
-// static
-void SfDelegate::OnInitiateReadWrapper(
- SfDelegate *me, void *data, size_t size) {
- me->onInitiateRead(data, size);
-}
-
-void SfDelegate::onInitiateRead(void *data, size_t size) {
- CHECK(mURLRequest != NULL);
-
- mNumBytesRead = 0;
- mNumBytesTotal = size;
- mDataDestination = data;
-
- if (mAtEOS) {
- mOwner->onReadCompleted(0);
- return;
- }
-
- readMore(mURLRequest);
-}
-
-} // namespace android
-
diff --git a/media/libstagefright/chromium_http/support.h b/media/libstagefright/chromium_http/support.h
deleted file mode 100644
index 975a1d3..0000000
--- a/media/libstagefright/chromium_http/support.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef SUPPORT_H_
-
-#define SUPPORT_H_
-
-#include <assert.h>
-
-#include "net/base/net_log.h"
-#include "net/url_request/url_request.h"
-#include "net/url_request/url_request_context.h"
-#include "net/base/android_network_library.h"
-#include "net/base/io_buffer.h"
-
-#include <utils/KeyedVector.h>
-#include <utils/Mutex.h>
-#include <utils/String8.h>
-
-namespace net {
- struct ProxyConfigServiceAndroid;
-};
-
-namespace android {
-
-struct SfNetLog : public net::NetLog {
- SfNetLog();
-
- virtual void AddEntry(
- EventType type,
- const base::TimeTicks &time,
- const Source &source,
- EventPhase phase,
- EventParameters *params);
-
- virtual uint32 NextID();
- virtual LogLevel GetLogLevel() const;
-
-private:
- uint32 mNextID;
-
- DISALLOW_EVIL_CONSTRUCTORS(SfNetLog);
-};
-
-struct SfRequestContext : public net::URLRequestContext {
- SfRequestContext();
-
- virtual const std::string &GetUserAgent(const GURL &url) const;
-
- status_t updateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
-private:
- Mutex mProxyConfigLock;
-
- std::string mUserAgent;
- net::ProxyConfigServiceAndroid *mProxyConfigService;
-
- DISALLOW_EVIL_CONSTRUCTORS(SfRequestContext);
-};
-
-// This is required for https support, we don't really verify certificates,
-// we accept anything...
-struct SfNetworkLibrary : public net::AndroidNetworkLibrary {
- SfNetworkLibrary();
-
- virtual VerifyResult VerifyX509CertChain(
- const std::vector<std::string>& cert_chain,
- const std::string& hostname,
- const std::string& auth_type);
-
-private:
- DISALLOW_EVIL_CONSTRUCTORS(SfNetworkLibrary);
-};
-
-struct ChromiumHTTPDataSource;
-
-struct SfDelegate : public net::URLRequest::Delegate {
- SfDelegate();
- virtual ~SfDelegate();
-
- void initiateConnection(
- const char *uri,
- const KeyedVector<String8, String8> *headers,
- off64_t offset);
-
- void initiateDisconnect();
- void initiateRead(void *data, size_t size);
-
- void setOwner(ChromiumHTTPDataSource *mOwner);
-
- // Gets the UID of the calling process
- bool getUID(uid_t *uid) const;
-
- void setUID(uid_t uid);
-
- virtual void OnReceivedRedirect(
- net::URLRequest *request, const GURL &new_url, bool *defer_redirect);
-
- virtual void OnAuthRequired(
- net::URLRequest *request, net::AuthChallengeInfo *auth_info);
-
- virtual void OnCertificateRequested(
- net::URLRequest *request, net::SSLCertRequestInfo *cert_request_info);
-
- virtual void OnSSLCertificateError(
- net::URLRequest *request, int cert_error, net::X509Certificate *cert);
-
- virtual void OnGetCookies(net::URLRequest *request, bool blocked_by_policy);
-
- virtual void OnSetCookie(
- net::URLRequest *request,
- const std::string &cookie_line,
- const net::CookieOptions &options,
- bool blocked_by_policy);
-
- virtual void OnResponseStarted(net::URLRequest *request);
-
- virtual void OnReadCompleted(net::URLRequest *request, int bytes_read);
-
- static status_t UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
-private:
- typedef Delegate inherited;
-
- ChromiumHTTPDataSource *mOwner;
-
- net::URLRequest *mURLRequest;
- scoped_refptr<net::IOBufferWithSize> mReadBuffer;
-
- size_t mNumBytesRead;
- size_t mNumBytesTotal;
- void *mDataDestination;
-
- bool mRangeRequested;
- bool mAtEOS;
-
- void readMore(net::URLRequest *request);
-
- static void OnInitiateConnectionWrapper(
- SfDelegate *me,
- GURL url,
- const KeyedVector<String8, String8> *headers,
- off64_t offset);
-
- static void OnInitiateDisconnectWrapper(SfDelegate *me);
-
- static void OnInitiateReadWrapper(
- SfDelegate *me, void *data, size_t size);
-
- void onInitiateConnection(
- const GURL &url,
- const KeyedVector<String8, String8> *headers,
- off64_t offset);
-
- void onInitiateDisconnect();
- void onInitiateRead(void *data, size_t size);
-
- DISALLOW_EVIL_CONSTRUCTORS(SfDelegate);
-};
-
-} // namespace android
-
-#endif // SUPPORT_H_
diff --git a/media/libstagefright/chromium_http_stub.cpp b/media/libstagefright/chromium_http_stub.cpp
deleted file mode 100644
index ed8a878..0000000
--- a/media/libstagefright/chromium_http_stub.cpp
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <dlfcn.h>
-
-#include <media/stagefright/DataSource.h>
-
-#include "include/chromium_http_stub.h"
-#include "include/HTTPBase.h"
-
-namespace android {
-
-static bool gFirst = true;
-static void *gHandle;
-static Mutex gLibMutex;
-
-HTTPBase *(*gLib_createChromiumHTTPDataSource)(uint32_t flags);
-DataSource *(*gLib_createDataUriSource)(const char *uri);
-
-status_t (*gLib_UpdateChromiumHTTPDataSourceProxyConfig)(
- const char *host, int32_t port, const char *exclusionList);
-
-static bool load_libstagefright_chromium_http() {
- Mutex::Autolock autoLock(gLibMutex);
- void *sym;
-
- if (!gFirst) {
- return (gHandle != NULL);
- }
-
- gFirst = false;
-
- gHandle = dlopen("libstagefright_chromium_http.so", RTLD_NOW);
- if (gHandle == NULL) {
- return false;
- }
-
- sym = dlsym(gHandle, "createChromiumHTTPDataSource");
- if (sym == NULL) {
- gHandle = NULL;
- return false;
- }
- gLib_createChromiumHTTPDataSource = (HTTPBase *(*)(uint32_t))sym;
-
- sym = dlsym(gHandle, "createDataUriSource");
- if (sym == NULL) {
- gHandle = NULL;
- return false;
- }
- gLib_createDataUriSource = (DataSource *(*)(const char *))sym;
-
- sym = dlsym(gHandle, "UpdateChromiumHTTPDataSourceProxyConfig");
- if (sym == NULL) {
- gHandle = NULL;
- return false;
- }
- gLib_UpdateChromiumHTTPDataSourceProxyConfig =
- (status_t (*)(const char *, int32_t, const char *))sym;
-
- return true;
-}
-
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags) {
- if (!load_libstagefright_chromium_http()) {
- return NULL;
- }
-
- return gLib_createChromiumHTTPDataSource(flags);
-}
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
- const char *host, int32_t port, const char *exclusionList) {
- if (!load_libstagefright_chromium_http()) {
- return INVALID_OPERATION;
- }
-
- return gLib_UpdateChromiumHTTPDataSourceProxyConfig(
- host, port, exclusionList);
-}
-
-DataSource *createDataUriSource(const char *uri) {
- if (!load_libstagefright_chromium_http()) {
- return NULL;
- }
-
- return gLib_createDataUriSource(uri);
-}
-
-}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 1b20cbb..4ac8999 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -30,7 +30,7 @@
#define DRC_DEFAULT_MOBILE_REF_LEVEL 64 /* 64*-0.25dB = -16 dB below full scale for mobile conf */
#define DRC_DEFAULT_MOBILE_DRC_CUT 127 /* maximum compression of dynamic range for mobile conf */
#define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
-#define MAX_CHANNEL_COUNT 6 /* maximum number of audio channels that can be decoded */
+#define MAX_CHANNEL_COUNT 8 /* maximum number of audio channels that can be decoded */
// names of properties that can be used to override the default DRC settings
#define PROP_DRC_OVERRIDE_REF_LEVEL "aac_drc_reference_level"
#define PROP_DRC_OVERRIDE_CUT "aac_drc_cut"
@@ -58,6 +58,8 @@
mIsADTS(false),
mInputBufferCount(0),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mAnchorTimeUs(0),
mNumSamplesOutput(0),
mOutputPortSettingsChange(NONE) {
@@ -294,8 +296,11 @@
if (!(property_get("media.aac_51_output_enabled", value, NULL) &&
(!strcmp(value, "1") || !strcasecmp(value, "true")))) {
ALOGI("Downmixing multichannel AAC to stereo");
- aacDecoder_SetParam(mAACDecoder, AAC_PCM_OUTPUT_CHANNELS, 2);
+ aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
mStreamInfo->numChannels = 2;
+ // By default, the decoder creates a 5.1 channel downmix signal
+ // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
+ // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
}
}
}
@@ -350,115 +355,83 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- if (mDecoderHasData) {
- // flush out the decoder's delayed data by calling DecodeFrame
- // one more time, with the AACDEC_FLUSH flag set
- INT_PCM *outBuffer =
- reinterpret_cast<INT_PCM *>(
- outHeader->pBuffer + outHeader->nOffset);
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumSamplesOutput = 0;
+ }
- AAC_DECODER_ERROR decoderErr =
- aacDecoder_DecodeFrame(mAACDecoder,
- outBuffer,
- outHeader->nAllocLen,
- AACDEC_FLUSH);
- mDecoderHasData = false;
+ if (mIsADTS && inHeader->nFilledLen) {
+ size_t adtsHeaderSize = 0;
+ // skip 30 bits, aac_frame_length follows.
+ // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
- if (decoderErr != AAC_DEC_OK) {
+ const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
+
+ bool signalError = false;
+ if (inHeader->nFilledLen < 7) {
+ ALOGE("Audio data too short to contain even the ADTS header. "
+ "Got %ld bytes.", inHeader->nFilledLen);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ bool protectionAbsent = (adtsHeader[1] & 1);
+
+ unsigned aac_frame_length =
+ ((adtsHeader[3] & 3) << 11)
+ | (adtsHeader[4] << 3)
+ | (adtsHeader[5] >> 5);
+
+ if (inHeader->nFilledLen < aac_frame_length) {
+ ALOGE("Not enough audio data for the complete frame. "
+ "Got %ld bytes, frame size according to the ADTS "
+ "header is %u bytes.",
+ inHeader->nFilledLen, aac_frame_length);
+ hexdump(adtsHeader, inHeader->nFilledLen);
+ signalError = true;
+ } else {
+ adtsHeaderSize = (protectionAbsent ? 7 : 9);
+
+ inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
+ inBufferLength[0] = aac_frame_length - adtsHeaderSize;
+
+ inHeader->nOffset += adtsHeaderSize;
+ inHeader->nFilledLen -= adtsHeaderSize;
+ }
+ }
+
+ if (signalError) {
mSignalledError = true;
- notify(OMX_EventError, OMX_ErrorUndefined, decoderErr,
+ notify(OMX_EventError,
+ OMX_ErrorStreamCorrupt,
+ ERROR_MALFORMED,
NULL);
return;
}
-
- outHeader->nFilledLen =
- mStreamInfo->frameSize
- * sizeof(int16_t)
- * mStreamInfo->numChannels;
} else {
- // we never submitted any data to the decoder, so there's nothing to flush out
- outHeader->nFilledLen = 0;
- }
-
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
- }
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumSamplesOutput = 0;
- }
-
- size_t adtsHeaderSize = 0;
- if (mIsADTS) {
- // skip 30 bits, aac_frame_length follows.
- // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
-
- const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
-
- bool signalError = false;
- if (inHeader->nFilledLen < 7) {
- ALOGE("Audio data too short to contain even the ADTS header. "
- "Got %ld bytes.", inHeader->nFilledLen);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- bool protectionAbsent = (adtsHeader[1] & 1);
-
- unsigned aac_frame_length =
- ((adtsHeader[3] & 3) << 11)
- | (adtsHeader[4] << 3)
- | (adtsHeader[5] >> 5);
-
- if (inHeader->nFilledLen < aac_frame_length) {
- ALOGE("Not enough audio data for the complete frame. "
- "Got %ld bytes, frame size according to the ADTS "
- "header is %u bytes.",
- inHeader->nFilledLen, aac_frame_length);
- hexdump(adtsHeader, inHeader->nFilledLen);
- signalError = true;
- } else {
- adtsHeaderSize = (protectionAbsent ? 7 : 9);
-
- inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
- inBufferLength[0] = aac_frame_length - adtsHeaderSize;
-
- inHeader->nOffset += adtsHeaderSize;
- inHeader->nFilledLen -= adtsHeaderSize;
- }
- }
-
- if (signalError) {
- mSignalledError = true;
-
- notify(OMX_EventError,
- OMX_ErrorStreamCorrupt,
- ERROR_MALFORMED,
- NULL);
-
- return;
+ inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+ inBufferLength[0] = inHeader->nFilledLen;
}
} else {
- inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
- inBufferLength[0] = inHeader->nFilledLen;
+ inBufferLength[0] = 0;
}
// Fill and decode
@@ -471,50 +444,66 @@
int prevNumChannels = mStreamInfo->numChannels;
AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
- while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+ mDecoderHasData |= (bytesValid[0] > 0);
aacDecoder_Fill(mAACDecoder,
inBuffer,
inBufferLength,
bytesValid);
- mDecoderHasData = true;
decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
outBuffer,
outHeader->nAllocLen,
0 /* flags */);
-
if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
- ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ if (mSawInputEos && bytesValid[0] <= 0) {
+ if (mDecoderHasData) {
+ // flush out the decoder's delayed data by calling DecodeFrame
+ // one more time, with the AACDEC_FLUSH flag set
+ decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
+ outBuffer,
+ outHeader->nAllocLen,
+ AACDEC_FLUSH);
+ mDecoderHasData = false;
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ break;
+ } else {
+ ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+ }
}
}
size_t numOutBytes =
mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
- if (decoderErr == AAC_DEC_OK) {
- UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
- inHeader->nFilledLen -= inBufferUsedLength;
- inHeader->nOffset += inBufferUsedLength;
- } else {
- ALOGW("AAC decoder returned error %d, substituting silence",
- decoderErr);
+ if (inHeader) {
+ if (decoderErr == AAC_DEC_OK) {
+ UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
+ inHeader->nFilledLen -= inBufferUsedLength;
+ inHeader->nOffset += inBufferUsedLength;
+ } else {
+ ALOGW("AAC decoder returned error %d, substituting silence",
+ decoderErr);
- memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+ memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
- // Discard input buffer.
- inHeader->nFilledLen = 0;
+ // Discard input buffer.
+ inHeader->nFilledLen = 0;
- aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
+ aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
- // fall through
- }
+ // fall through
+ }
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
}
/*
@@ -555,7 +544,6 @@
// we've previously decoded valid data, in the latter case
// (decode failed) we'll output a silent frame.
outHeader->nFilledLen = numOutBytes;
- outHeader->nFlags = 0;
outHeader->nTimeStamp =
mAnchorTimeUs
@@ -582,6 +570,12 @@
// depend on fragments from the last one decoded.
// drain all existing data
drainDecoder();
+ // force decoder loop to drop the first decoded buffer by resetting these state variables,
+ // but only if initialization has already happened.
+ if (mInputBufferCount != 0) {
+ mInputBufferCount = 1;
+ mStreamInfo->sampleRate = 0;
+ }
}
}
@@ -606,6 +600,8 @@
mStreamInfo->sampleRate = 0;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 2d960ab..a7ea1e2 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -55,6 +55,8 @@
bool mDecoderHasData;
size_t mInputBufferCount;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
int64_t mAnchorTimeUs;
int64_t mNumSamplesOutput;
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 7c382fb..61df65e 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -49,6 +49,8 @@
mNumChannels(2),
mSamplingRate(44100),
mSignalledError(false),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
initDecoder();
@@ -194,48 +196,36 @@
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
-
- if (!mIsFirst) {
- // pad the end of the stream with 529 samples, since that many samples
- // were trimmed off the beginning when decoding started
- outHeader->nFilledLen =
- kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
-
- memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
- } else {
- // Since we never discarded frames from the start, we won't have
- // to add any padding at the end either.
- outHeader->nFilledLen = 0;
+ if (inHeader) {
+ if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
}
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ mConfig->pInputBuffer =
+ inHeader->pBuffer + inHeader->nOffset;
+
+ mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
+ } else {
+ mConfig->pInputBuffer = NULL;
+ mConfig->inputBufferCurrentLength = 0;
}
-
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- mConfig->pInputBuffer =
- inHeader->pBuffer + inHeader->nOffset;
-
- mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
mConfig->inputBufferMaxLength = 0;
mConfig->inputBufferUsedLength = 0;
@@ -262,13 +252,28 @@
mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
}
- // This is recoverable, just ignore the current frame and
- // play silence instead.
- memset(outHeader->pBuffer,
- 0,
- mConfig->outputFrameSize * sizeof(int16_t));
+ if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) {
+ if (!mIsFirst) {
+ // pad the end of the stream with 529 samples, since that many samples
+ // were trimmed off the beginning when decoding started
+ outHeader->nOffset = 0;
+ outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
- mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
+ }
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ } else {
+ // This is recoverable, just ignore the current frame and
+ // play silence instead.
+ memset(outHeader->pBuffer,
+ 0,
+ mConfig->outputFrameSize * sizeof(int16_t));
+
+ if (inHeader) {
+ mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+ }
+ }
} else if (mConfig->samplingRate != mSamplingRate
|| mConfig->num_channels != mNumChannels) {
mSamplingRate = mConfig->samplingRate;
@@ -289,7 +294,7 @@
outHeader->nFilledLen =
mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset;
- } else {
+ } else if (!mSignalledOutputEos) {
outHeader->nOffset = 0;
outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t);
}
@@ -298,23 +303,24 @@
mAnchorTimeUs
+ (mNumFramesOutput * 1000000ll) / mConfig->samplingRate;
- outHeader->nFlags = 0;
+ if (inHeader) {
+ CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
- CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
+ inHeader->nOffset += mConfig->inputBufferUsedLength;
+ inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
- inHeader->nOffset += mConfig->inputBufferUsedLength;
- inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
+
+ if (inHeader->nFilledLen == 0) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
+ }
mNumFramesOutput += mConfig->outputFrameSize / mNumChannels;
- if (inHeader->nFilledLen == 0) {
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
- }
-
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
outInfo = NULL;
@@ -329,6 +335,9 @@
// depend on fragments from the last one decoded.
pvmp3_InitDecoder(mConfig, mDecoderBuf);
mIsFirst = true;
+ mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
}
}
@@ -362,6 +371,8 @@
pvmp3_InitDecoder(mConfig, mDecoderBuf);
mIsFirst = true;
mSignalledError = false;
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h
index 4af91ea..f9e7b53 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.h
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h
@@ -61,6 +61,8 @@
bool mIsFirst;
bool mSignalledError;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 51bb958..515e4d3 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -54,6 +54,8 @@
mAnchorTimeUs(0),
mNumFramesOutput(0),
mNumFramesLeftOnPage(-1),
+ mSawInputEos(false),
+ mSignalledOutputEos(false),
mOutputPortSettingsChange(NONE) {
initPorts();
CHECK_EQ(initDecoder(), (status_t)OK);
@@ -290,48 +292,47 @@
return;
}
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+ BufferInfo *inInfo = NULL;
+ OMX_BUFFERHEADERTYPE *inHeader = NULL;
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ }
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
+ int32_t numPageSamples = 0;
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ if (inHeader) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mSawInputEos = true;
+ }
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
+ if (inHeader->nFilledLen || !mSawInputEos) {
+ CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
+ memcpy(&numPageSamples,
+ inHeader->pBuffer
+ + inHeader->nOffset + inHeader->nFilledLen - 4,
+ sizeof(numPageSamples));
+
+ if (inHeader->nOffset == 0) {
+ mAnchorTimeUs = inHeader->nTimeStamp;
+ mNumFramesOutput = 0;
+ }
+
+ inHeader->nFilledLen -= sizeof(numPageSamples);;
+ }
}
- int32_t numPageSamples;
- CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
- memcpy(&numPageSamples,
- inHeader->pBuffer
- + inHeader->nOffset + inHeader->nFilledLen - 4,
- sizeof(numPageSamples));
-
if (numPageSamples >= 0) {
mNumFramesLeftOnPage = numPageSamples;
}
- if (inHeader->nOffset == 0) {
- mAnchorTimeUs = inHeader->nTimeStamp;
- mNumFramesOutput = 0;
- }
-
- inHeader->nFilledLen -= sizeof(numPageSamples);;
-
ogg_buffer buf;
- buf.data = inHeader->pBuffer + inHeader->nOffset;
- buf.size = inHeader->nFilledLen;
+ buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL;
+ buf.size = inHeader ? inHeader->nFilledLen : 0;
buf.refcount = 1;
buf.ptr.owner = NULL;
@@ -351,6 +352,7 @@
int numFrames = 0;
+ outHeader->nFlags = 0;
int err = vorbis_dsp_synthesis(mState, &pack, 1);
if (err != 0) {
ALOGW("vorbis_dsp_synthesis returned %d", err);
@@ -370,13 +372,16 @@
ALOGV("discarding %d frames at end of page",
numFrames - mNumFramesLeftOnPage);
numFrames = mNumFramesLeftOnPage;
+ if (mSawInputEos) {
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ mSignalledOutputEos = true;
+ }
}
mNumFramesLeftOnPage -= numFrames;
}
outHeader->nFilledLen = numFrames * sizeof(int16_t) * mVi->channels;
outHeader->nOffset = 0;
- outHeader->nFlags = 0;
outHeader->nTimeStamp =
mAnchorTimeUs
@@ -384,11 +389,13 @@
mNumFramesOutput += numFrames;
- inInfo->mOwnedByUs = false;
- inQueue.erase(inQueue.begin());
- inInfo = NULL;
- notifyEmptyBufferDone(inHeader);
- inHeader = NULL;
+ if (inHeader) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
@@ -425,6 +432,8 @@
mVi = NULL;
}
+ mSawInputEos = false;
+ mSignalledOutputEos = false;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
index cb628a0..1d00816 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
@@ -59,6 +59,8 @@
int64_t mAnchorTimeUs;
int64_t mNumFramesOutput;
int32_t mNumFramesLeftOnPage;
+ bool mSawInputEos;
+ bool mSignalledOutputEos;
enum {
NONE,
diff --git a/media/libstagefright/http/Android.mk b/media/libstagefright/http/Android.mk
new file mode 100644
index 0000000..a6b481f
--- /dev/null
+++ b/media/libstagefright/http/Android.mk
@@ -0,0 +1,26 @@
+LOCAL_PATH:= $(call my-dir)
+
+ifneq ($(TARGET_BUILD_PDK), true)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ HTTPHelper.cpp \
+
+LOCAL_C_INCLUDES:= \
+ $(TOP)/frameworks/av/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/frameworks/base/core/jni \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright liblog libutils libbinder libstagefright_foundation \
+ libandroid_runtime \
+ libmedia
+
+LOCAL_MODULE:= libstagefright_http_support
+
+LOCAL_CFLAGS += -Wno-multichar
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
diff --git a/media/libstagefright/http/HTTPHelper.cpp b/media/libstagefright/http/HTTPHelper.cpp
new file mode 100644
index 0000000..77845e2
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "HTTPHelper"
+#include <utils/Log.h>
+
+#include "HTTPHelper.h"
+
+#include "android_runtime/AndroidRuntime.h"
+#include "android_util_Binder.h"
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <nativehelper/ScopedLocalRef.h>
+#include "jni.h"
+
+namespace android {
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext() {
+ if (AndroidRuntime::getJavaVM() == NULL) {
+ ALOGE("CreateHTTPServiceInCurrentJavaContext called outside "
+ "JAVA environment.");
+ return NULL;
+ }
+
+ JNIEnv *env = AndroidRuntime::getJNIEnv();
+
+ ScopedLocalRef<jclass> clazz(
+ env, env->FindClass("android/media/MediaHTTPService"));
+ CHECK(clazz.get() != NULL);
+
+ jmethodID constructID = env->GetMethodID(clazz.get(), "<init>", "()V");
+ CHECK(constructID != NULL);
+
+ ScopedLocalRef<jobject> httpServiceObj(
+ env, env->NewObject(clazz.get(), constructID));
+
+ sp<IMediaHTTPService> httpService;
+ if (httpServiceObj.get() != NULL) {
+ jmethodID asBinderID =
+ env->GetMethodID(clazz.get(), "asBinder", "()Landroid/os/IBinder;");
+ CHECK(asBinderID != NULL);
+
+ ScopedLocalRef<jobject> httpServiceBinderObj(
+ env, env->CallObjectMethod(httpServiceObj.get(), asBinderID));
+ CHECK(httpServiceBinderObj.get() != NULL);
+
+ sp<IBinder> binder =
+ ibinderForJavaObject(env, httpServiceBinderObj.get());
+
+ httpService = interface_cast<IMediaHTTPService>(binder);
+ }
+
+ return httpService;
+}
+
+} // namespace android
diff --git a/media/libstagefright/http/HTTPHelper.h b/media/libstagefright/http/HTTPHelper.h
new file mode 100644
index 0000000..8aef115
--- /dev/null
+++ b/media/libstagefright/http/HTTPHelper.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef HTTP_HELPER_H_
+
+#define HTTP_HELPER_H_
+
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct IMediaHTTPService;
+
+sp<IMediaHTTPService> CreateHTTPServiceInCurrentJavaContext();
+
+} // namespace android
+
+#endif // HTTP_HELPER_H_
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
new file mode 100644
index 0000000..157d967
--- /dev/null
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -0,0 +1,201 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaHTTP"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaHTTP.h>
+
+#include <binder/IServiceManager.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/Utils.h>
+
+#include <media/IMediaHTTPConnection.h>
+
+namespace android {
+
+MediaHTTP::MediaHTTP(const sp<IMediaHTTPConnection> &conn)
+ : mInitCheck(NO_INIT),
+ mHTTPConnection(conn),
+ mCachedSizeValid(false),
+ mCachedSize(0ll),
+ mDrmManagerClient(NULL) {
+ mInitCheck = OK;
+}
+
+MediaHTTP::~MediaHTTP() {
+ clearDRMState_l();
+}
+
+status_t MediaHTTP::connect(
+ const char *uri,
+ const KeyedVector<String8, String8> *headers,
+ off64_t /* offset */) {
+ if (mInitCheck != OK) {
+ return mInitCheck;
+ }
+
+ KeyedVector<String8, String8> extHeaders;
+ if (headers != NULL) {
+ extHeaders = *headers;
+ }
+ extHeaders.add(String8("User-Agent"), String8(MakeUserAgent().c_str()));
+
+ bool success = mHTTPConnection->connect(uri, &extHeaders);
+
+ mLastHeaders = extHeaders;
+ mLastURI = uri;
+
+ mCachedSizeValid = false;
+
+ return success ? OK : UNKNOWN_ERROR;
+}
+
+void MediaHTTP::disconnect() {
+ if (mInitCheck != OK) {
+ return;
+ }
+
+ mHTTPConnection->disconnect();
+}
+
+status_t MediaHTTP::initCheck() const {
+ return mInitCheck;
+}
+
+ssize_t MediaHTTP::readAt(off64_t offset, void *data, size_t size) {
+ if (mInitCheck != OK) {
+ return mInitCheck;
+ }
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+
+ size_t numBytesRead = 0;
+ while (numBytesRead < size) {
+ size_t copy = size - numBytesRead;
+
+ if (copy > 64 * 1024) {
+ // limit the buffer sizes transferred across binder boundaries
+ // to avoid spurious transaction failures.
+ copy = 64 * 1024;
+ }
+
+ ssize_t n = mHTTPConnection->readAt(
+ offset + numBytesRead, (uint8_t *)data + numBytesRead, copy);
+
+ if (n < 0) {
+ return n;
+ } else if (n == 0) {
+ break;
+ }
+
+ numBytesRead += n;
+ }
+
+ int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
+
+ addBandwidthMeasurement(numBytesRead, delayUs);
+
+ return numBytesRead;
+}
+
+status_t MediaHTTP::getSize(off64_t *size) {
+ if (mInitCheck != OK) {
+ return mInitCheck;
+ }
+
+ // Caching the returned size so that it stays valid even after a
+ // disconnect. NuCachedSource2 relies on this.
+
+ if (!mCachedSizeValid) {
+ mCachedSize = mHTTPConnection->getSize();
+ mCachedSizeValid = true;
+ }
+
+ *size = mCachedSize;
+
+ return *size < 0 ? *size : OK;
+}
+
+uint32_t MediaHTTP::flags() {
+ return kWantsPrefetching | kIsHTTPBasedSource;
+}
+
+status_t MediaHTTP::reconnectAtOffset(off64_t offset) {
+ return connect(mLastURI.c_str(), &mLastHeaders, offset);
+}
+
+// DRM...
+
+sp<DecryptHandle> MediaHTTP::DrmInitialization(const char* mime) {
+ if (mDrmManagerClient == NULL) {
+ mDrmManagerClient = new DrmManagerClient();
+ }
+
+ if (mDrmManagerClient == NULL) {
+ return NULL;
+ }
+
+ if (mDecryptHandle == NULL) {
+ mDecryptHandle = mDrmManagerClient->openDecryptSession(
+ String8(mLastURI.c_str()), mime);
+ }
+
+ if (mDecryptHandle == NULL) {
+ delete mDrmManagerClient;
+ mDrmManagerClient = NULL;
+ }
+
+ return mDecryptHandle;
+}
+
+void MediaHTTP::getDrmInfo(
+ sp<DecryptHandle> &handle, DrmManagerClient **client) {
+ handle = mDecryptHandle;
+ *client = mDrmManagerClient;
+}
+
+String8 MediaHTTP::getUri() {
+ return String8(mLastURI.c_str());
+}
+
+String8 MediaHTTP::getMIMEType() const {
+ if (mInitCheck != OK) {
+ return String8("application/octet-stream");
+ }
+
+ String8 mimeType;
+ status_t err = mHTTPConnection->getMIMEType(&mimeType);
+
+ if (err != OK) {
+ return String8("application/octet-stream");
+ }
+
+ return mimeType;
+}
+
+void MediaHTTP::clearDRMState_l() {
+ if (mDecryptHandle != NULL) {
+ // To release mDecryptHandle
+ CHECK(mDrmManagerClient);
+ mDrmManagerClient->closeDecryptSession(mDecryptHandle);
+ mDecryptHandle = NULL;
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 233db44..149a817 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -27,6 +27,8 @@
#include "mpeg2ts/AnotherPacketSource.h"
#include <cutils/properties.h>
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -34,6 +36,7 @@
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaHTTP.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
@@ -44,17 +47,13 @@
namespace android {
LiveSession::LiveSession(
- const sp<AMessage> ¬ify, uint32_t flags, bool uidValid, uid_t uid)
+ const sp<AMessage> ¬ify, uint32_t flags,
+ const sp<IMediaHTTPService> &httpService)
: mNotify(notify),
mFlags(flags),
- mUIDValid(uidValid),
- mUID(uid),
+ mHTTPService(httpService),
mInPreparationPhase(true),
- mHTTPDataSource(
- HTTPBase::Create(
- (mFlags & kFlagIncognito)
- ? HTTPBase::kFlagIncognito
- : 0)),
+ mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())),
mPrevBandwidthIndex(-1),
mStreamMask(0),
mCheckBandwidthGeneration(0),
@@ -62,10 +61,6 @@
mRealTimeBaseUs(0ll),
mReconfigurationInProgress(false),
mDisconnectReplyID(0) {
- if (mUIDValid) {
- mHTTPDataSource->setUID(mUID);
- }
-
mPacketSources.add(
STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */));
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 99b480a8..ee10e70 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -28,6 +28,7 @@
struct AnotherPacketSource;
struct DataSource;
struct HTTPBase;
+struct IMediaHTTPService;
struct LiveDataSource;
struct M3UParser;
struct PlaylistFetcher;
@@ -40,7 +41,8 @@
};
LiveSession(
const sp<AMessage> ¬ify,
- uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+ uint32_t flags,
+ const sp<IMediaHTTPService> &httpService);
enum StreamType {
STREAMTYPE_AUDIO = 1,
@@ -107,8 +109,7 @@
sp<AMessage> mNotify;
uint32_t mFlags;
- bool mUIDValid;
- uid_t mUID;
+ sp<IMediaHTTPService> mHTTPService;
bool mInPreparationPhase;
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 973b779..f095987 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -47,6 +47,7 @@
// static
const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
+const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
PlaylistFetcher::PlaylistFetcher(
const sp<AMessage> ¬ify,
@@ -61,6 +62,7 @@
mSeqNumber(-1),
mNumRetries(0),
mStartup(true),
+ mPrepared(false),
mNextPTSTimeUs(-1ll),
mMonitorQueueGeneration(0),
mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
@@ -103,10 +105,16 @@
return segmentStartUs;
}
-bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const {
- if (mPlaylist == NULL) {
+int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
+ int64_t nowUs = ALooper::GetNowUs();
+
+ if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) {
CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
- return true;
+ return 0ll;
+ }
+
+ if (mPlaylist->isComplete()) {
+ return (~0llu >> 1);
}
int32_t targetDurationSecs;
@@ -157,7 +165,8 @@
break;
}
- return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs;
+ int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs;
+ return delayUs > 0ll ? delayUs : 0ll;
}
status_t PlaylistFetcher::decryptBuffer(
@@ -274,7 +283,15 @@
return OK;
}
-void PlaylistFetcher::postMonitorQueue(int64_t delayUs) {
+void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) {
+ int64_t maxDelayUs = delayUsToRefreshPlaylist();
+ if (maxDelayUs < minDelayUs) {
+ maxDelayUs = minDelayUs;
+ }
+ if (delayUs > maxDelayUs) {
+ ALOGV("Need to refresh playlist in %lld", maxDelayUs);
+ delayUs = maxDelayUs;
+ }
sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
msg->setInt32("generation", mMonitorQueueGeneration);
msg->post(delayUs);
@@ -415,6 +432,7 @@
if (mStartTimeUs >= 0ll) {
mSeqNumber = -1;
mStartup = true;
+ mPrepared = false;
}
postMonitorQueue();
@@ -456,40 +474,62 @@
void PlaylistFetcher::onMonitorQueue() {
bool downloadMore = false;
+ refreshPlaylist();
- status_t finalResult;
+ int32_t targetDurationSecs;
+ int64_t targetDurationUs = kMinBufferedDurationUs;
+ if (mPlaylist != NULL) {
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+ targetDurationUs = targetDurationSecs * 1000000ll;
+ }
+
+ // buffer at least 3 times the target duration, or up to 10 seconds
+ int64_t durationToBufferUs = targetDurationUs * 3;
+ if (durationToBufferUs > kMinBufferedDurationUs) {
+ durationToBufferUs = kMinBufferedDurationUs;
+ }
+
+ int64_t bufferedDurationUs = 0ll;
+ status_t finalResult = NOT_ENOUGH_DATA;
if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
sp<AnotherPacketSource> packetSource =
mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
- int64_t bufferedDurationUs =
+ bufferedDurationUs =
packetSource->getBufferedDurationUs(&finalResult);
-
- downloadMore = (bufferedDurationUs < kMinBufferedDurationUs);
finalResult = OK;
} else {
bool first = true;
- int64_t minBufferedDurationUs = 0ll;
for (size_t i = 0; i < mPacketSources.size(); ++i) {
if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
continue;
}
- int64_t bufferedDurationUs =
+ int64_t bufferedStreamDurationUs =
mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
-
- if (first || bufferedDurationUs < minBufferedDurationUs) {
- minBufferedDurationUs = bufferedDurationUs;
+ if (first || bufferedStreamDurationUs < bufferedDurationUs) {
+ bufferedDurationUs = bufferedStreamDurationUs;
first = false;
}
}
+ }
+ downloadMore = (bufferedDurationUs < durationToBufferUs);
- downloadMore =
- !first && (minBufferedDurationUs < kMinBufferedDurationUs);
+ // signal start if buffered up at least the target size
+ if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
+ mPrepared = true;
+
+ ALOGV("prepared, buffered=%lld > %lld",
+ bufferedDurationUs, targetDurationUs);
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatTemporarilyDoneFetching);
+ msg->post();
}
if (finalResult == OK && downloadMore) {
+ ALOGV("monitoring, buffered=%lld < %lld",
+ bufferedDurationUs, durationToBufferUs);
onDownloadNext();
} else {
// Nothing to do yet, try again in a second.
@@ -498,15 +538,17 @@
msg->setInt32("what", kWhatTemporarilyDoneFetching);
msg->post();
- postMonitorQueue(1000000ll);
+ int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
+ ALOGV("pausing for %lld, buffered=%lld > %lld",
+ delayUs, bufferedDurationUs, durationToBufferUs);
+ // :TRICKY: need to enforce minimum delay because the delay to
+ // refresh the playlist will become 0
+ postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0);
}
}
-void PlaylistFetcher::onDownloadNext() {
- int64_t nowUs = ALooper::GetNowUs();
-
- if (mLastPlaylistFetchTimeUs < 0ll
- || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) {
+status_t PlaylistFetcher::refreshPlaylist() {
+ if (delayUsToRefreshPlaylist() <= 0) {
bool unchanged;
sp<M3UParser> playlist = mSession->fetchPlaylist(
mURI.c_str(), mPlaylistHash, &unchanged);
@@ -522,7 +564,7 @@
} else {
ALOGE("failed to load playlist at url '%s'", mURI.c_str());
notifyError(ERROR_IO);
- return;
+ return ERROR_IO;
}
} else {
mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
@@ -535,6 +577,13 @@
mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
}
+ return OK;
+}
+
+void PlaylistFetcher::onDownloadNext() {
+ if (refreshPlaylist() != OK) {
+ return;
+ }
int32_t firstSeqNumberInPlaylist;
if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
@@ -553,12 +602,18 @@
if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
mSeqNumber = getSeqNumberForTime(mStartTimeUs);
+ ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)",
+ mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist);
} else {
// If this is a live session, start 3 segments from the end.
mSeqNumber = lastSeqNumberInPlaylist - 3;
if (mSeqNumber < firstSeqNumberInPlaylist) {
mSeqNumber = firstSeqNumberInPlaylist;
}
+ ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)",
+ mSeqNumber, firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist);
}
mStartTimeUs = -1ll;
@@ -570,16 +625,34 @@
++mNumRetries;
if (mSeqNumber > lastSeqNumberInPlaylist) {
- mLastPlaylistFetchTimeUs = -1;
- postMonitorQueue(3000000ll);
+ // refresh in increasing fraction (1/2, 1/3, ...) of the
+ // playlist's target duration or 3 seconds, whichever is less
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32(
+ "target-duration", &targetDurationSecs));
+ int64_t delayUs = mPlaylist->size() * targetDurationSecs *
+ 1000000ll / (1 + mNumRetries);
+ if (delayUs > kMaxMonitorDelayUs) {
+ delayUs = kMaxMonitorDelayUs;
+ }
+ ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)",
+ mSeqNumber, firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist, delayUs, mNumRetries);
+ postMonitorQueue(delayUs);
return;
}
// we've missed the boat, let's start from the lowest sequence
// number available and signal a discontinuity.
- ALOGI("We've missed the boat, restarting playback.");
- mSeqNumber = lastSeqNumberInPlaylist;
+ ALOGI("We've missed the boat, restarting playback."
+ " mStartup=%d, was looking for %d in %d-%d",
+ mStartup, mSeqNumber, firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist);
+ mSeqNumber = lastSeqNumberInPlaylist - 3;
+ if (mSeqNumber < firstSeqNumberInPlaylist) {
+ mSeqNumber = firstSeqNumberInPlaylist;
+ }
explicitDiscontinuity = true;
// fall through
@@ -788,12 +861,13 @@
&& source->dequeueAccessUnit(&accessUnit) == OK) {
// Note that we do NOT dequeue any discontinuities.
+ // for simplicity, store a reference to the format in each unit
+ sp<MetaData> format = source->getFormat();
+ if (format != NULL) {
+ accessUnit->meta()->setObject("format", format);
+ }
packetSource->queueAccessUnit(accessUnit);
}
-
- if (packetSource->getFormat() == NULL) {
- packetSource->setFormat(source->getFormat());
- }
}
return OK;
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 1648e02..78dea20 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -79,6 +79,7 @@
};
static const int64_t kMinBufferedDurationUs;
+ static const int64_t kMaxMonitorDelayUs;
sp<AMessage> mNotify;
sp<LiveSession> mSession;
@@ -97,6 +98,7 @@
int32_t mSeqNumber;
int32_t mNumRetries;
bool mStartup;
+ bool mPrepared;
int64_t mNextPTSTimeUs;
int32_t mMonitorQueueGeneration;
@@ -120,10 +122,11 @@
status_t decryptBuffer(
size_t playlistIndex, const sp<ABuffer> &buffer);
- void postMonitorQueue(int64_t delayUs = 0);
+ void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0);
void cancelMonitorQueue();
- bool timeToRefreshPlaylist(int64_t nowUs) const;
+ int64_t delayUsToRefreshPlaylist() const;
+ status_t refreshPlaylist();
// Returns the media time in us of the segment specified by seqNumber.
// This is computed by summing the durations of all segments before it.
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 1ec4a40..f0f203c 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -468,49 +468,6 @@
}
}
-static void convertISO8859ToString8(
- const uint8_t *data, size_t size,
- String8 *s) {
- size_t utf8len = 0;
- for (size_t i = 0; i < size; ++i) {
- if (data[i] == '\0') {
- size = i;
- break;
- } else if (data[i] < 0x80) {
- ++utf8len;
- } else {
- utf8len += 2;
- }
- }
-
- if (utf8len == size) {
- // Only ASCII characters present.
-
- s->setTo((const char *)data, size);
- return;
- }
-
- char *tmp = new char[utf8len];
- char *ptr = tmp;
- for (size_t i = 0; i < size; ++i) {
- if (data[i] == '\0') {
- break;
- } else if (data[i] < 0x80) {
- *ptr++ = data[i];
- } else if (data[i] < 0xc0) {
- *ptr++ = 0xc2;
- *ptr++ = data[i];
- } else {
- *ptr++ = 0xc3;
- *ptr++ = data[i] - 64;
- }
- }
-
- s->setTo(tmp, utf8len);
-
- delete[] tmp;
- tmp = NULL;
-}
// the 2nd argument is used to get the data following the \0 in a comment field
void ID3::Iterator::getString(String8 *id, String8 *comment) const {
@@ -543,7 +500,9 @@
return;
}
- convertISO8859ToString8(frameData, mFrameSize, id);
+ // this is supposed to be ISO-8859-1, but pass it up as-is to the caller, who will figure
+ // out the real encoding
+ id->setTo((const char*)frameData, mFrameSize);
return;
}
@@ -561,13 +520,13 @@
}
if (encoding == 0x00) {
- // ISO 8859-1
- convertISO8859ToString8(frameData + 1, n, id);
+ // supposedly ISO 8859-1
+ id->setTo((const char*)frameData + 1, n);
} else if (encoding == 0x03) {
- // UTF-8
+ // supposedly UTF-8
id->setTo((const char *)(frameData + 1), n);
} else if (encoding == 0x02) {
- // UTF-16 BE, no byte order mark.
+ // supposedly UTF-16 BE, no byte order mark.
// API wants number of characters, not number of bytes...
int len = n / 2;
const char16_t *framedata = (const char16_t *) (frameData + 1);
@@ -583,7 +542,7 @@
if (framedatacopy != NULL) {
delete[] framedatacopy;
}
- } else {
+ } else if (encoding == 0x01) {
// UCS-2
// API wants number of characters, not number of bytes...
int len = n / 2;
@@ -602,7 +561,27 @@
framedata++;
len--;
}
- id->setTo(framedata, len);
+
+ // check if the resulting data consists entirely of 8-bit values
+ bool eightBit = true;
+ for (int i = 0; i < len; i++) {
+ if (framedata[i] > 0xff) {
+ eightBit = false;
+ break;
+ }
+ }
+ if (eightBit) {
+ // collapse to 8 bit, then let the media scanner client figure out the real encoding
+ char *frame8 = new char[len];
+ for (int i = 0; i < len; i++) {
+ frame8[i] = framedata[i];
+ }
+ id->setTo(frame8, len);
+ delete [] frame8;
+ } else {
+ id->setTo(framedata, len);
+ }
+
if (framedatacopy != NULL) {
delete[] framedatacopy;
}
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 271df8e..a81bbba 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -63,6 +63,7 @@
void setUID(uid_t uid);
status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL);
@@ -159,6 +160,7 @@
SystemTimeSource mSystemTimeSource;
TimeSource *mTimeSource;
+ sp<IMediaHTTPService> mHTTPService;
String8 mUri;
KeyedVector<String8, String8> mUriHeaders;
@@ -247,6 +249,7 @@
sp<MediaExtractor> mExtractor;
status_t setDataSource_l(
+ const sp<IMediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
deleted file mode 100644
index da188dd..0000000
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CHROME_HTTP_DATA_SOURCE_H_
-
-#define CHROME_HTTP_DATA_SOURCE_H_
-
-#include <media/stagefright/foundation/AString.h>
-#include <utils/threads.h>
-
-#include "HTTPBase.h"
-
-namespace android {
-
-struct SfDelegate;
-
-struct ChromiumHTTPDataSource : public HTTPBase {
- ChromiumHTTPDataSource(uint32_t flags = 0);
-
- virtual status_t connect(
- const char *uri,
- const KeyedVector<String8, String8> *headers = NULL,
- off64_t offset = 0);
-
- virtual void disconnect();
-
- virtual status_t initCheck() const;
-
- virtual ssize_t readAt(off64_t offset, void *data, size_t size);
- virtual status_t getSize(off64_t *size);
- virtual uint32_t flags();
-
- virtual sp<DecryptHandle> DrmInitialization(const char *mime);
-
- virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
-
- virtual String8 getUri();
-
- virtual String8 getMIMEType() const;
-
- virtual status_t reconnectAtOffset(off64_t offset);
-
- static status_t UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
-protected:
- virtual ~ChromiumHTTPDataSource();
-
-private:
- friend struct SfDelegate;
-
- enum State {
- DISCONNECTED,
- CONNECTING,
- CONNECTED,
- READING,
- DISCONNECTING
- };
-
- const uint32_t mFlags;
-
- mutable Mutex mLock;
- Condition mCondition;
-
- State mState;
-
- SfDelegate *mDelegate;
-
- AString mURI;
- KeyedVector<String8, String8> mHeaders;
-
- off64_t mCurrentOffset;
-
- // Any connection error or the result of a read operation
- // (for the lattter this is the number of bytes read, if successful).
- ssize_t mIOResult;
-
- int64_t mContentSize;
-
- String8 mContentType;
-
- sp<DecryptHandle> mDecryptHandle;
- DrmManagerClient *mDrmManagerClient;
-
- void disconnect_l();
-
- status_t connect_l(
- const char *uri,
- const KeyedVector<String8, String8> *headers,
- off64_t offset);
-
- static void InitiateRead(
- ChromiumHTTPDataSource *me, void *data, size_t size);
-
- void initiateRead(void *data, size_t size);
-
- void onConnectionEstablished(
- int64_t contentSize, const char *contentType);
-
- void onConnectionFailed(status_t err);
- void onReadCompleted(ssize_t size);
- void onDisconnectComplete();
- void onRedirect(const char *url);
-
- void clearDRMState_l();
-
- DISALLOW_EVIL_CONSTRUCTORS(ChromiumHTTPDataSource);
-};
-
-} // namespace android
-
-#endif // CHROME_HTTP_DATA_SOURCE_H_
diff --git a/media/libstagefright/include/HTTPBase.h b/media/libstagefright/include/HTTPBase.h
index d4b7f9f..1c3cd5e 100644
--- a/media/libstagefright/include/HTTPBase.h
+++ b/media/libstagefright/include/HTTPBase.h
@@ -48,14 +48,6 @@
virtual status_t setBandwidthStatCollectFreq(int32_t freqMs);
- static status_t UpdateProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
- void setUID(uid_t uid);
- bool getUID(uid_t *uid) const;
-
- static sp<HTTPBase> Create(uint32_t flags = 0);
-
static void RegisterSocketUserTag(int sockfd, uid_t uid, uint32_t kTag);
static void UnRegisterSocketUserTag(int sockfd);
@@ -87,9 +79,6 @@
int32_t mPrevEstimatedBandWidthKbps;
int32_t mBandWidthCollectFreqMs;
- bool mUIDValid;
- uid_t mUID;
-
DISALLOW_EVIL_CONSTRUCTORS(HTTPBase);
};
diff --git a/media/libstagefright/include/SDPLoader.h b/media/libstagefright/include/SDPLoader.h
index ca59dc0..2c4f543 100644
--- a/media/libstagefright/include/SDPLoader.h
+++ b/media/libstagefright/include/SDPLoader.h
@@ -25,6 +25,7 @@
namespace android {
struct HTTPBase;
+struct IMediaHTTPService;
struct SDPLoader : public AHandler {
enum Flags {
@@ -34,7 +35,10 @@
enum {
kWhatSDPLoaded = 'sdpl'
};
- SDPLoader(const sp<AMessage> ¬ify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
+ SDPLoader(
+ const sp<AMessage> ¬ify,
+ uint32_t flags,
+ const sp<IMediaHTTPService> &httpService);
void load(const char* url, const KeyedVector<String8, String8> *headers);
@@ -55,8 +59,6 @@
sp<AMessage> mNotify;
const char* mUrl;
uint32_t mFlags;
- bool mUIDValid;
- uid_t mUID;
sp<ALooper> mNetLooper;
bool mCancelled;
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index b02ed0e..6632c27 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -33,6 +33,7 @@
virtual ~StagefrightMetadataRetriever();
virtual status_t setDataSource(
+ const sp<IMediaHTTPService> &httpService,
const char *url,
const KeyedVector<String8, String8> *headers);
diff --git a/media/libstagefright/include/chromium_http_stub.h b/media/libstagefright/include/chromium_http_stub.h
deleted file mode 100644
index e0651a4..0000000
--- a/media/libstagefright/include/chromium_http_stub.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CHROMIUM_HTTP_STUB_H_
-#define CHROMIUM_HTTP_STUB_H_
-
-#include <include/HTTPBase.h>
-#include <media/stagefright/DataSource.h>
-
-namespace android {
-extern "C" {
-HTTPBase *createChromiumHTTPDataSource(uint32_t flags);
-
-status_t UpdateChromiumHTTPDataSourceProxyConfig(
- const char *host, int32_t port, const char *exclusionList);
-
-DataSource *createDataUriSource(const char *uri);
-}
-}
-
-#endif
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d260d0f..dcb1cda 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -716,41 +716,61 @@
return mIsLiveStreaming;
}
+static int bytesForSize(size_t size) {
+ // use at most 28 bits (4 times 7)
+ CHECK(size <= 0xfffffff);
+
+ if (size > 0x1fffff) {
+ return 4;
+ } else if (size > 0x3fff) {
+ return 3;
+ } else if (size > 0x7f) {
+ return 2;
+ }
+ return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+ int numBytes = bytesForSize(size);
+ idx += numBytes;
+
+ data += idx;
+ size_t next = 0;
+ while (numBytes--) {
+ *--data = (size & 0x7f) | next;
+ size >>= 7;
+ next = 0x80;
+ }
+}
+
static void addESDSFromCodecPrivate(
const sp<MetaData> &meta,
bool isAudio, const void *priv, size_t privSize) {
- static const uint8_t kStaticESDS[] = {
- 0x03, 22,
- 0x00, 0x00, // ES_ID
- 0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag
- 0x04, 17,
- 0x40, // ObjectTypeIndication
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
-
- 0x05,
- // CodecSpecificInfo (with size prefix) follows
- };
-
- // Make sure all sizes can be coded in a single byte.
- CHECK(privSize + 22 - 2 < 128);
- size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
+ int privSizeBytesRequired = bytesForSize(privSize);
+ int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+ int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+ int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+ int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+ size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
uint8_t *esds = new uint8_t[esdsSize];
- memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
- uint8_t *ptr = esds + sizeof(kStaticESDS);
- *ptr++ = privSize;
- memcpy(ptr, priv, privSize);
- // Increment by codecPrivateSize less 2 bytes that are accounted for
- // already in lengths of 22/17
- esds[1] += privSize - 2;
- esds[6] += privSize - 2;
-
- // Set ObjectTypeIndication.
- esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
- : 0x20; // Visual ISO/IEC 14496-2
+ size_t idx = 0;
+ esds[idx++] = 0x03;
+ storeSize(esds, idx, esdsSize1);
+ esds[idx++] = 0x00; // ES_ID
+ esds[idx++] = 0x00; // ES_ID
+ esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+ esds[idx++] = 0x04;
+ storeSize(esds, idx, esdsSize2);
+ esds[idx++] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
+ : 0x20; // Visual ISO/IEC 14496-2
+ for (int i = 0; i < 12; i++) {
+ esds[idx++] = 0x00;
+ }
+ esds[idx++] = 0x05;
+ storeSize(esds, idx, privSize);
+ memcpy(esds + idx, priv, privSize);
meta->setData(kKeyESDS, 0, esds, esdsSize);
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 3153c8b..52fb2a5 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -70,7 +70,27 @@
}
sp<MetaData> AnotherPacketSource::getFormat() {
- return mFormat;
+ Mutex::Autolock autoLock(mLock);
+ if (mFormat != NULL) {
+ return mFormat;
+ }
+
+ List<sp<ABuffer> >::iterator it = mBuffers.begin();
+ while (it != mBuffers.end()) {
+ sp<ABuffer> buffer = *it;
+ int32_t discontinuity;
+ if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ break;
+ }
+
+ sp<RefBase> object;
+ if (buffer->meta()->findObject("format", &object)) {
+ return static_cast<MetaData*>(object.get());
+ }
+
+ ++it;
+ }
+ return NULL;
}
status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) {
@@ -94,6 +114,11 @@
return INFO_DISCONTINUITY;
}
+ sp<RefBase> object;
+ if ((*buffer)->meta()->findObject("format", &object)) {
+ mFormat = static_cast<MetaData*>(object.get());
+ }
+
return OK;
}
@@ -120,17 +145,22 @@
}
return INFO_DISCONTINUITY;
- } else {
- int64_t timeUs;
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
-
- MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
-
- mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
-
- *out = mediaBuffer;
- return OK;
}
+
+ sp<RefBase> object;
+ if (buffer->meta()->findObject("format", &object)) {
+ mFormat = static_cast<MetaData*>(object.get());
+ }
+
+ int64_t timeUs;
+ CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+
+ MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+
+ mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
+
+ *out = mediaBuffer;
+ return OK;
}
return mEOSResult;
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index b8970ad..1e6de55 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -42,7 +42,12 @@
mEndOfStream(false),
mEndOfStreamSent(false),
mRepeatAfterUs(-1ll),
+ mMaxTimestampGapUs(-1ll),
+ mPrevOriginalTimeUs(-1ll),
+ mPrevModifiedTimeUs(-1ll),
+ mSkipFramesBeforeNs(-1ll),
mRepeatLastFrameGeneration(0),
+ mRepeatLastFrameTimestamp(-1ll),
mLatestSubmittedBufferId(-1),
mLatestSubmittedBufferFrameNum(0),
mLatestSubmittedBufferUseCount(0),
@@ -299,6 +304,32 @@
return;
}
+void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mMaxTimestampGapUs > 0ll
+ && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+ ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
+ if (index >= 0) {
+ ALOGV("OUT timestamp: %lld -> %lld",
+ header->nTimeStamp, mOriginalTimeUs[index]);
+ header->nTimeStamp = mOriginalTimeUs[index];
+ mOriginalTimeUs.removeItemsAt(index);
+ } else {
+ // giving up the effort as encoder doesn't appear to preserve pts
+ ALOGW("giving up limiting timestamp gap (pts = %lld)",
+ header->nTimeStamp);
+ mMaxTimestampGapUs = -1ll;
+ }
+ if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
+ // something terribly wrong must have happened, giving up...
+ ALOGE("mOriginalTimeUs has too many entries (%d)",
+ mOriginalTimeUs.size());
+ mMaxTimestampGapUs = -1ll;
+ }
+ }
+}
+
void GraphicBufferSource::suspend(bool suspend) {
Mutex::Autolock autoLock(mMutex);
@@ -384,7 +415,18 @@
mBufferSlot[item.mBuf] = item.mGraphicBuffer;
}
- err = submitBuffer_l(item, cbi);
+ err = UNKNOWN_ERROR;
+
+ // only submit sample if start time is unspecified, or sample
+ // is queued after the specified start time
+ if (mSkipFramesBeforeNs < 0ll || item.mTimestamp >= mSkipFramesBeforeNs) {
+ // if start time is set, offset time stamp by start time
+ if (mSkipFramesBeforeNs > 0) {
+ item.mTimestamp -= mSkipFramesBeforeNs;
+ }
+ err = submitBuffer_l(item, cbi);
+ }
+
if (err != OK) {
ALOGV("submitBuffer_l failed, releasing bq buf %d", item.mBuf);
mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
@@ -431,6 +473,7 @@
BufferQueue::BufferItem item;
item.mBuf = mLatestSubmittedBufferId;
item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+ item.mTimestamp = mRepeatLastFrameTimestamp;
status_t err = submitBuffer_l(item, cbi);
@@ -440,6 +483,20 @@
++mLatestSubmittedBufferUseCount;
+ /* repeat last frame up to kRepeatLastFrameCount times.
+ * in case of static scene, a single repeat might not get rid of encoder
+ * ghosting completely, refresh a couple more times to get better quality
+ */
+ if (--mRepeatLastFrameCount > 0) {
+ mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
+ if (mReflector != NULL) {
+ sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+ msg->setInt32("generation", ++mRepeatLastFrameGeneration);
+ msg->post(mRepeatAfterUs);
+ }
+ }
+
return true;
}
@@ -460,8 +517,11 @@
mLatestSubmittedBufferId = item.mBuf;
mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+ mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
mLatestSubmittedBufferUseCount = 1;
mRepeatBufferDeferred = false;
+ mRepeatLastFrameCount = kRepeatLastFrameCount;
if (mReflector != NULL) {
sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -497,6 +557,39 @@
return OK;
}
+int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
+ int64_t timeUs = item.mTimestamp / 1000;
+
+ if (mMaxTimestampGapUs > 0ll) {
+ /* Cap timestamp gap between adjacent frames to specified max
+ *
+ * In the scenario of cast mirroring, encoding could be suspended for
+ * prolonged periods. Limiting the pts gap to workaround the problem
+ * where encoder's rate control logic produces huge frames after a
+ * long period of suspension.
+ */
+
+ int64_t originalTimeUs = timeUs;
+ if (mPrevOriginalTimeUs >= 0ll) {
+ if (originalTimeUs < mPrevOriginalTimeUs) {
+ // Drop the frame if it's going backward in time. Bad timestamp
+ // could disrupt encoder's rate control completely.
+ ALOGV("Dropping frame that's going backward in time");
+ return -1;
+ }
+ int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
+ timeUs = (timestampGapUs < mMaxTimestampGapUs ?
+ timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs;
+ }
+ mPrevOriginalTimeUs = originalTimeUs;
+ mPrevModifiedTimeUs = timeUs;
+ mOriginalTimeUs.add(timeUs, originalTimeUs);
+ ALOGV("IN timestamp: %lld -> %lld", originalTimeUs, timeUs);
+ }
+
+ return timeUs;
+}
+
status_t GraphicBufferSource::submitBuffer_l(
const BufferQueue::BufferItem &item, int cbi) {
ALOGV("submitBuffer_l cbi=%d", cbi);
@@ -513,9 +606,15 @@
memcpy(data, &type, 4);
memcpy(data + 4, &handle, sizeof(buffer_handle_t));
+ int64_t timeUs = getTimestamp(item);
+ if (timeUs < 0ll) {
+ ALOGE("Dropping frame with bad timestamp");
+ return UNKNOWN_ERROR;
+ }
+
status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
- item.mTimestamp / 1000);
+ timeUs);
if (err != OK) {
ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
codecBuffer.mGraphicBuffer = NULL;
@@ -609,6 +708,12 @@
BufferQueue::BufferItem item;
status_t err = mBufferQueue->acquireBuffer(&item, 0);
if (err == OK) {
+ // If this is the first time we're seeing this buffer, add it to our
+ // slot table.
+ if (item.mGraphicBuffer != NULL) {
+ ALOGV("fillCodecBuffer_l: setting mBufferSlot %d", item.mBuf);
+ mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+ }
mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence);
}
@@ -658,6 +763,25 @@
return OK;
}
+status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mExecuting || maxGapUs <= 0ll) {
+ return INVALID_OPERATION;
+ }
+
+ mMaxTimestampGapUs = maxGapUs;
+
+ return OK;
+}
+
+void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
+ Mutex::Autolock autoLock(mMutex);
+
+ mSkipFramesBeforeNs =
+ (skipFramesBeforeUs > 0) ? (skipFramesBeforeUs * 1000) : -1ll;
+}
+
void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
case kWhatRepeatLastFrame:
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 9e5eee6..153f2a0 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -87,6 +87,10 @@
// fill it with a new frame of data; otherwise, just mark it as available.
void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header);
+ // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the
+ // buffer source will fix timestamp in the header if needed.)
+ void codecBufferFilled(OMX_BUFFERHEADERTYPE* header);
+
// This is called after the last input frame has been submitted. We
// need to submit an empty buffer with the EOS flag set. If we don't
// have a codec buffer ready, we just set the mEndOfStream flag.
@@ -105,6 +109,19 @@
// state and once this behaviour is specified it cannot be reset.
status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs);
+ // When set, the timestamp fed to the encoder will be modified such that
+ // the gap between two adjacent frames is capped at maxGapUs. Timestamp
+ // will be restored to the original when the encoded frame is returned to
+ // the client.
+ // This is to solve a problem in certain real-time streaming case, where
+ // encoder's rate control logic produces huge frames after a long period
+ // of suspension on input.
+ status_t setMaxTimestampGapUs(int64_t maxGapUs);
+
+ // Sets the start time us (in system time), samples before which should
+ // be dropped and not submitted to encoder
+ void setSkipFramesBeforeUs(int64_t startTimeUs);
+
protected:
// BufferQueue::ConsumerListener interface, called when a new frame of
// data is available. If we're executing and a codec buffer is
@@ -165,6 +182,7 @@
void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
bool repeatLatestSubmittedBuffer_l();
+ int64_t getTimestamp(const BufferQueue::BufferItem &item);
// Lock, covers all member variables.
mutable Mutex mMutex;
@@ -206,13 +224,23 @@
enum {
kWhatRepeatLastFrame,
};
+ enum {
+ kRepeatLastFrameCount = 10,
+ };
- int64_t mRepeatAfterUs;
+ KeyedVector<int64_t, int64_t> mOriginalTimeUs;
+ int64_t mMaxTimestampGapUs;
+ int64_t mPrevOriginalTimeUs;
+ int64_t mPrevModifiedTimeUs;
+ int64_t mSkipFramesBeforeNs;
sp<ALooper> mLooper;
sp<AHandlerReflector<GraphicBufferSource> > mReflector;
+ int64_t mRepeatAfterUs;
int32_t mRepeatLastFrameGeneration;
+ int64_t mRepeatLastFrameTimestamp;
+ int32_t mRepeatLastFrameCount;
int mLatestSubmittedBufferId;
uint64_t mLatestSubmittedBufferFrameNum;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 5f104fc..dba522f 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -849,6 +849,8 @@
switch (type) {
case IOMX::INTERNAL_OPTION_SUSPEND:
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
+ case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+ case IOMX::INTERNAL_OPTION_START_TIME:
{
const sp<GraphicBufferSource> &bufferSource =
getGraphicBufferSource();
@@ -864,7 +866,8 @@
bool suspend = *(bool *)data;
bufferSource->suspend(suspend);
- } else {
+ } else if (type ==
+ IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){
if (size != sizeof(int64_t)) {
return INVALID_OPERATION;
}
@@ -872,6 +875,23 @@
int64_t delayUs = *(int64_t *)data;
return bufferSource->setRepeatPreviousFrameDelayUs(delayUs);
+ } else if (type ==
+ IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP){
+ if (size != sizeof(int64_t)) {
+ return INVALID_OPERATION;
+ }
+
+ int64_t maxGapUs = *(int64_t *)data;
+
+ return bufferSource->setMaxTimestampGapUs(maxGapUs);
+ } else { // IOMX::INTERNAL_OPTION_START_TIME
+ if (size != sizeof(int64_t)) {
+ return INVALID_OPERATION;
+ }
+
+ int64_t skipFramesBeforeUs = *(int64_t *)data;
+
+ bufferSource->setSkipFramesBeforeUs(skipFramesBeforeUs);
}
return OK;
@@ -883,6 +903,8 @@
}
void OMXNodeInstance::onMessage(const omx_message &msg) {
+ const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
+
if (msg.type == omx_message::FILL_BUFFER_DONE) {
OMX_BUFFERHEADERTYPE *buffer =
static_cast<OMX_BUFFERHEADERTYPE *>(
@@ -892,10 +914,18 @@
static_cast<BufferMeta *>(buffer->pAppPrivate);
buffer_meta->CopyFromOMX(buffer);
- } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
- const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
if (bufferSource != NULL) {
+ // fix up the buffer info (especially timestamp) if needed
+ bufferSource->codecBufferFilled(buffer);
+
+ omx_message newMsg = msg;
+ newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp;
+ mObserver->onMessage(newMsg);
+ return;
+ }
+ } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
+ if (bufferSource != NULL) {
// This is one of the buffers used exclusively by
// GraphicBufferSource.
// Don't dispatch a message back to ACodec, since it doesn't
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 44e4f9d..03725df 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -26,6 +26,7 @@
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
+#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
@@ -242,7 +243,8 @@
};
static sp<MediaExtractor> CreateExtractorFromURI(const char *uri) {
- sp<DataSource> source = DataSource::CreateFromURI(uri);
+ sp<DataSource> source =
+ DataSource::CreateFromURI(NULL /* httpService */, uri);
if (source == NULL) {
return NULL;
diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
index 462c384..09f52bc 100644
--- a/media/libstagefright/rtsp/APacketSource.cpp
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -23,7 +23,7 @@
#include "ARawAudioAssembler.h"
#include "ASessionDescription.h"
-#include "avc_utils.h"
+#include "include/avc_utils.h"
#include <ctype.h>
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index efde7a9..4054da6 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -33,7 +33,7 @@
#include <openssl/md5.h>
#include <sys/socket.h>
-#include "HTTPBase.h"
+#include "include/HTTPBase.h"
namespace android {
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index e77c69c..02e44f4 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -20,7 +20,7 @@
SDPLoader.cpp \
LOCAL_C_INCLUDES:= \
- $(TOP)/frameworks/av/media/libstagefright/include \
+ $(TOP)/frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax \
$(TOP)/external/openssl/include
diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index ed3fa7e..ce1e89d 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -18,11 +18,13 @@
#define LOG_TAG "SDPLoader"
#include <utils/Log.h>
-#include "SDPLoader.h"
+#include "include/SDPLoader.h"
#include "ASessionDescription.h"
-#include "HTTPBase.h"
+#include <media/IMediaHTTPConnection.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/MediaHTTP.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -30,22 +32,15 @@
namespace android {
-SDPLoader::SDPLoader(const sp<AMessage> ¬ify, uint32_t flags, bool uidValid, uid_t uid)
+SDPLoader::SDPLoader(
+ const sp<AMessage> ¬ify,
+ uint32_t flags,
+ const sp<IMediaHTTPService> &httpService)
: mNotify(notify),
mFlags(flags),
- mUIDValid(uidValid),
- mUID(uid),
mNetLooper(new ALooper),
mCancelled(false),
- mHTTPDataSource(
- HTTPBase::Create(
- (mFlags & kFlagIncognito)
- ? HTTPBase::kFlagIncognito
- : 0)) {
- if (mUIDValid) {
- mHTTPDataSource->setUID(mUID);
- }
-
+ mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())) {
mNetLooper->setName("sdp net");
mNetLooper->start(false /* runOnCallingThread */,
false /* canCallJava */,
@@ -130,7 +125,7 @@
ssize_t readSize = mHTTPDataSource->readAt(0, buffer->data(), sdpSize);
if (readSize < 0) {
- ALOGE("Failed to read SDP, error code = %ld", readSize);
+ ALOGE("Failed to read SDP, error code = %d", readSize);
err = UNKNOWN_ERROR;
} else {
desc = new ASessionDescription;
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 49ffcd6..ddedad7 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -746,9 +746,8 @@
CHECK(fd >= 0);
sp<MediaRecorder> mr = SurfaceMediaSourceGLTest::setUpMediaRecorder(fd,
- VIDEO_SOURCE_GRALLOC_BUFFER,
- OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth,
- mYuvTexHeight, 30);
+ VIDEO_SOURCE_SURFACE, OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264,
+ mYuvTexWidth, mYuvTexHeight, 30);
// get the reference to the surfacemediasource living in
// mediaserver that is created by stagefrightrecorder
sp<IGraphicBufferProducer> iST = mr->querySurfaceMediaSourceFromMediaServer();
@@ -880,7 +879,7 @@
}
CHECK(fd >= 0);
- sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+ sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
// get the reference to the surfacemediasource living in
@@ -923,7 +922,7 @@
}
CHECK(fd >= 0);
- sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_GRALLOC_BUFFER,
+ sp<MediaRecorder> mr = setUpMediaRecorder(fd, VIDEO_SOURCE_SURFACE,
OUTPUT_FORMAT_MPEG_4, VIDEO_ENCODER_H264, mYuvTexWidth, mYuvTexHeight, 30);
// get the reference to the surfacemediasource living in
diff --git a/media/libstagefright/timedtext/TimedTextDriver.cpp b/media/libstagefright/timedtext/TimedTextDriver.cpp
index 12fd7f4..05d6d02 100644
--- a/media/libstagefright/timedtext/TimedTextDriver.cpp
+++ b/media/libstagefright/timedtext/TimedTextDriver.cpp
@@ -20,6 +20,7 @@
#include <binder/IPCThreadState.h>
+#include <media/IMediaHTTPService.h>
#include <media/mediaplayer.h>
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/DataSource.h>
@@ -40,7 +41,8 @@
namespace android {
TimedTextDriver::TimedTextDriver(
- const wp<MediaPlayerBase> &listener)
+ const wp<MediaPlayerBase> &listener,
+ const sp<IMediaHTTPService> &httpService)
: mLooper(new ALooper),
mListener(listener),
mState(UNINITIALIZED),
@@ -207,7 +209,7 @@
}
sp<DataSource> dataSource =
- DataSource::CreateFromURI(uri);
+ DataSource::CreateFromURI(mHTTPService, uri);
return createOutOfBandTextSource(trackIndex, mimeType, dataSource);
}
diff --git a/media/libstagefright/timedtext/test/Android.mk b/media/libstagefright/timedtext/test/Android.mk
index a5e7ba2..9a9fde2 100644
--- a/media/libstagefright/timedtext/test/Android.mk
+++ b/media/libstagefright/timedtext/test/Android.mk
@@ -2,7 +2,6 @@
# ================================================================
# Unit tests for libstagefright_timedtext
-# See also /development/testrunner/test_defs.xml
# ================================================================
# ================================================================
@@ -18,10 +17,13 @@
LOCAL_C_INCLUDES := \
$(TOP)/external/expat/lib \
- $(TOP)/frameworks/base/media/libstagefright/timedtext
+ $(TOP)/frameworks/av/media/libstagefright/timedtext
LOCAL_SHARED_LIBRARIES := \
+ libbinder \
libexpat \
- libstagefright
+ libstagefright \
+ libstagefright_foundation \
+ libutils
include $(BUILD_NATIVE_TEST)
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 286ea13..1a5acba 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -29,6 +29,7 @@
#include <binder/IServiceManager.h>
#include <cutils/properties.h>
#include <media/IHDCP.h>
+#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -749,7 +750,8 @@
mExtractor = new NuMediaExtractor;
- status_t err = mExtractor->setDataSource(mMediaPath.c_str());
+ status_t err = mExtractor->setDataSource(
+ NULL /* httpService */, mMediaPath.c_str());
if (err != OK) {
return err;
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index edcc087..50d317a 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@
uint8_t *ptr = dup->data();
*ptr++ = 0xff;
- *ptr++ = 0xf1; // b11110001, ID=0, layer=0, protection_absent=1
+ *ptr++ = 0xf9; // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
*ptr++ =
profile << 6
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 1ac647a..f848054 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -15,6 +15,7 @@
LOCAL_SHARED_LIBRARIES := \
libaudioflinger \
+ libcamera_metadata\
libcameraservice \
libmedialogservice \
libcutils \
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index d5207d5..a347951 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -37,7 +37,7 @@
using namespace android;
-int main(int argc, char** argv)
+int main(int argc __unused, char** argv)
{
signal(SIGPIPE, SIG_IGN);
char value[PROPERTY_VALUE_MAX];
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 54377f1..4524d3c 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -23,7 +23,8 @@
AudioPolicyService.cpp \
ServiceUtilities.cpp \
AudioResamplerCubic.cpp.arm \
- AudioResamplerSinc.cpp.arm
+ AudioResamplerSinc.cpp.arm \
+ AudioResamplerDyn.cpp.arm
LOCAL_SRC_FILES += StateQueue.cpp
@@ -74,12 +75,20 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- test-resample.cpp \
+ test-resample.cpp \
AudioResampler.cpp.arm \
- AudioResamplerCubic.cpp.arm \
- AudioResamplerSinc.cpp.arm
+ AudioResamplerCubic.cpp.arm \
+ AudioResamplerSinc.cpp.arm \
+ AudioResamplerDyn.cpp.arm
+
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils)
+
+LOCAL_STATIC_LIBRARIES := \
+ libsndfile
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
libdl \
libcutils \
libutils \
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index e9c38e3..21bd2f4 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -104,6 +104,27 @@
// ----------------------------------------------------------------------------
+const char *formatToString(audio_format_t format) {
+ switch(format) {
+ case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
+ case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
+ case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
+ case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
+ case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
+ case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
+ case AUDIO_FORMAT_MP3: return "mp3";
+ case AUDIO_FORMAT_AMR_NB: return "amr-nb";
+ case AUDIO_FORMAT_AMR_WB: return "amr-wb";
+ case AUDIO_FORMAT_AAC: return "aac";
+ case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
+ case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
+ case AUDIO_FORMAT_VORBIS: return "vorbis";
+ default:
+ break;
+ }
+ return "unknown";
+}
+
static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
@@ -162,12 +183,15 @@
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
- if (teeEnabled & 1)
+ if (teeEnabled & 1) {
mTeeSinkInputEnabled = true;
- if (teeEnabled & 2)
+ }
+ if (teeEnabled & 2) {
mTeeSinkOutputEnabled = true;
- if (teeEnabled & 4)
+ }
+ if (teeEnabled & 4) {
mTeeSinkTrackEnabled = true;
+ }
#endif
}
@@ -210,6 +234,18 @@
audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
delete mAudioHwDevs.valueAt(i);
}
+
+ // Tell media.log service about any old writers that still need to be unregistered
+ sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
+ if (binder != 0) {
+ sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
+ for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+ sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
+ mUnregisteredWriters.pop();
+ mediaLogService->unregisterWriter(iMemory);
+ }
+ }
+
}
static const char * const audio_interfaces[] = {
@@ -249,7 +285,7 @@
return NULL;
}
-void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -271,17 +307,17 @@
}
result.append("Global session refs:\n");
- result.append(" session pid count\n");
+ result.append(" session pid count\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
- snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
+ snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
result.append(buffer);
}
write(fd, result.string(), result.size());
}
-void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -296,7 +332,7 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -403,16 +439,44 @@
sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
{
+ // If there is no memory allocated for logs, return a dummy writer that does nothing
if (mLogMemoryDealer == 0) {
return new NBLog::Writer();
}
- sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
- sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
+ // Similarly if we can't contact the media.log service, also return a dummy writer
+ if (binder == 0) {
+ return new NBLog::Writer();
}
- return writer;
+ sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
+ sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
+ // If allocation fails, consult the vector of previously unregistered writers
+ // and garbage-collect one or more them until an allocation succeeds
+ if (shared == 0) {
+ Mutex::Autolock _l(mUnregisteredWritersLock);
+ for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
+ {
+ // Pick the oldest stale writer to garbage-collect
+ sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
+ mUnregisteredWriters.removeAt(0);
+ mediaLogService->unregisterWriter(iMemory);
+ // Now the media.log remote reference to IMemory is gone. When our last local
+ // reference to IMemory also drops to zero at end of this block,
+ // the IMemory destructor will deallocate the region from mLogMemoryDealer.
+ }
+ // Re-attempt the allocation
+ shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
+ if (shared != 0) {
+ goto success;
+ }
+ }
+ // Even after garbage-collecting all old writers, there is still not enough memory,
+ // so return a dummy writer
+ return new NBLog::Writer();
+ }
+success:
+ mediaLogService->registerWriter(shared, size, name);
+ return new NBLog::Writer(size, shared);
}
void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
@@ -424,13 +488,10 @@
if (iMemory == 0) {
return;
}
- sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
- if (binder != 0) {
- interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
- // Now the media.log remote reference to IMemory is gone.
- // When our last local reference to IMemory also drops to zero,
- // the IMemory destructor will deallocate the region from mMemoryDealer.
- }
+ // Rather than removing the writer immediately, append it to a queue of old writers to
+ // be garbage-collected later. This allows us to continue to view old logs for a while.
+ Mutex::Autolock _l(mUnregisteredWritersLock);
+ mUnregisteredWriters.push(writer);
}
// IAudioFlinger interface
@@ -441,7 +502,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -468,7 +529,13 @@
// client is responsible for conversion of 8-bit PCM to 16-bit PCM,
// and we don't yet support 8.24 or 32-bit PCM
if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGE("createTrack() invalid format %d", format);
+ ALOGE("createTrack() invalid format %#x", format);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
+ ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
lStatus = BAD_VALUE;
goto Exit;
}
@@ -488,7 +555,7 @@
client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
- if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+ if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
@@ -513,10 +580,13 @@
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
+ // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
+ // no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
moveEffectChain_l(lSessionId, effectThread, thread, true);
@@ -536,7 +606,9 @@
}
}
}
+
}
+
if (lStatus == NO_ERROR) {
// s for server's pid, n for normal mixer name, f for fast index
name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
@@ -550,9 +622,7 @@
}
Exit:
- if (status != NULL) {
- *status = lStatus;
- }
+ *status = lStatus;
return trackHandle;
}
@@ -1210,7 +1280,7 @@
{
}
-void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
{
sp<NotificationClient> keep(this);
mAudioFlinger->removeNotificationClient(mPid);
@@ -1228,7 +1298,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
@@ -1250,7 +1320,7 @@
}
if (format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGE("openRecord() invalid format %d", format);
+ ALOGE("openRecord() invalid format %#x", format);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -1276,7 +1346,7 @@
client = registerPid_l(pid);
// If no audio session id is provided, create one here
- if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
+ if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId();
@@ -1291,8 +1361,9 @@
frameCount, lSessionId,
IPCThreadState::self()->getCallingUid(),
flags, tid, &lStatus);
- LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
}
+
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
// Client destructor is called by the TrackBase destructor with mLock held
@@ -1301,14 +1372,11 @@
goto Exit;
}
- // return to handle to client
+ // return handle to client
recordHandle = new RecordHandle(recordTrack);
- lStatus = NO_ERROR;
Exit:
- if (status) {
- *status = lStatus;
- }
+ *status = lStatus;
return recordHandle;
}
@@ -1449,18 +1517,15 @@
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
- PlaybackThread *thread = NULL;
struct audio_config config;
+ memset(&config, 0, sizeof(config));
config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
- if (offloadInfo) {
+ if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
- audio_stream_out_t *outStream = NULL;
- AudioHwDevice *outHwDev;
-
ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
module,
(pDevices != NULL) ? *pDevices : 0,
@@ -1469,7 +1534,7 @@
config.channel_mask,
flags);
ALOGV("openOutput(), offloadInfo %p version 0x%04x",
- offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
+ offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
if (pDevices == NULL || *pDevices == 0) {
return 0;
@@ -1477,15 +1542,17 @@
Mutex::Autolock _l(mLock);
- outHwDev = findSuitableHwDev_l(module, *pDevices);
- if (outHwDev == NULL)
+ AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
+ if (outHwDev == NULL) {
return 0;
+ }
audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
audio_io_handle_t id = nextUniqueId();
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ audio_stream_out_t *outStream = NULL;
status_t status = hwDevHal->open_output_stream(hwDevHal,
id,
*pDevices,
@@ -1505,6 +1572,7 @@
if (status == NO_ERROR && outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
+ PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, output, id, *pDevices);
ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
@@ -1672,18 +1740,15 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask)
{
- status_t status;
- RecordThread *thread = NULL;
struct audio_config config;
+ memset(&config, 0, sizeof(config));
config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t reqSamplingRate = config.sample_rate;
audio_format_t reqFormat = config.format;
- audio_channel_mask_t reqChannels = config.channel_mask;
- audio_stream_in_t *inStream = NULL;
- AudioHwDevice *inHwDev;
+ audio_channel_mask_t reqChannelMask = config.channel_mask;
if (pDevices == NULL || *pDevices == 0) {
return 0;
@@ -1691,16 +1756,18 @@
Mutex::Autolock _l(mLock);
- inHwDev = findSuitableHwDev_l(module, *pDevices);
- if (inHwDev == NULL)
+ AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
+ if (inHwDev == NULL) {
return 0;
+ }
audio_hw_device_t *inHwHal = inHwDev->hwDevice();
audio_io_handle_t id = nextUniqueId();
- status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
+ audio_stream_in_t *inStream = NULL;
+ status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
&inStream);
- ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
+ ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
"status %d",
inStream,
config.sample_rate,
@@ -1714,10 +1781,12 @@
if (status == BAD_VALUE &&
reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
(config.sample_rate <= 2 * reqSamplingRate) &&
- (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
+ (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
+ // FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput() reopening with proposed sampling rate and channel mask");
inStream = NULL;
status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
+ // FIXME log this new status; HAL should not propose any further changes
}
if (status == NO_ERROR && inStream != NULL) {
@@ -1735,7 +1804,7 @@
popcount(inStream->common.get_channels(&inStream->common)));
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
- } else if (format == Format_Invalid) {
+ } else if (!Format_isValid(format)) {
kind = TEE_SINK_NO;
} else if (mRecordTeeSink == 0) {
kind = TEE_SINK_NEW;
@@ -1776,10 +1845,10 @@
// Start record thread
// RecordThread requires both input and output device indication to forward to audio
// pre processing modules
- thread = new RecordThread(this,
+ RecordThread *thread = new RecordThread(this,
input,
reqSamplingRate,
- reqChannels,
+ reqChannelMask,
id,
primaryOutputDevice_l(),
*pDevices
@@ -1796,7 +1865,7 @@
*pFormat = config.format;
}
if (pChannelMask != NULL) {
- *pChannelMask = reqChannels;
+ *pChannelMask = reqChannelMask;
}
// notify client processes of the new input creation
@@ -1954,7 +2023,7 @@
}
}
if (!found) {
- Mutex::Autolock _l (t->mLock);
+ Mutex::Autolock _l(t->mLock);
// remove all effects from the chain
while (ec->mEffects.size()) {
sp<EffectModule> effect = ec->mEffects[0];
@@ -2249,9 +2318,7 @@
}
Exit:
- if (status != NULL) {
- *status = lStatus;
- }
+ *status = lStatus;
return handle;
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 7320144..459d2ec 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -60,8 +60,8 @@
namespace android {
-class audio_track_cblk_t;
-class effect_param_cblk_t;
+struct audio_track_cblk_t;
+struct effect_param_cblk_t;
class AudioMixer;
class AudioBuffer;
class AudioResampler;
@@ -102,7 +102,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -110,18 +110,18 @@
int *sessionId,
String8& name,
int clientUid,
- status_t *status);
+ status_t *status /*non-NULL*/);
virtual sp<IAudioRecord> openRecord(
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
- status_t *status);
+ status_t *status /*non-NULL*/);
virtual uint32_t sampleRate(audio_io_handle_t output) const;
virtual int channelCount(audio_io_handle_t output) const;
@@ -210,7 +210,7 @@
int32_t priority,
audio_io_handle_t io,
int sessionId,
- status_t *status,
+ status_t *status /*non-NULL*/,
int *id,
int *enabled);
@@ -235,8 +235,12 @@
sp<NBLog::Writer> newWriter_l(size_t size, const char *name);
void unregisterWriter(const sp<NBLog::Writer>& writer);
private:
- static const size_t kLogMemorySize = 10 * 1024;
+ static const size_t kLogMemorySize = 40 * 1024;
sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled
+ // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
+ // for as long as possible. The memory is only freed when it is needed for another log writer.
+ Vector< sp<NBLog::Writer> > mUnregisteredWriters;
+ Mutex mUnregisteredWritersLock;
public:
class SyncEvent;
@@ -499,7 +503,7 @@
private:
const char * const mModuleName;
audio_hw_device_t * const mHwDevice;
- Flags mFlags;
+ const Flags mFlags;
};
// AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
@@ -509,7 +513,7 @@
struct AudioStreamOut {
AudioHwDevice* const audioHwDev;
audio_stream_out_t* const stream;
- audio_output_flags_t flags;
+ const audio_output_flags_t flags;
audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
@@ -651,6 +655,8 @@
#undef INCLUDING_FROM_AUDIOFLINGER_H
+const char *formatToString(audio_format_t format);
+
// ----------------------------------------------------------------------------
}; // namespace android
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index f92421e..67d83b1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -58,7 +58,7 @@
status_t AudioMixer::DownmixerBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
int64_t pts) {
//ALOGV("DownmixerBufferProvider::getNextBuffer()");
- if (this->mTrackBufferProvider != NULL) {
+ if (mTrackBufferProvider != NULL) {
status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
if (res == OK) {
mDownmixConfig.inputCfg.buffer.frameCount = pBuffer->frameCount;
@@ -81,7 +81,7 @@
void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
//ALOGV("DownmixerBufferProvider::releaseBuffer()");
- if (this->mTrackBufferProvider != NULL) {
+ if (mTrackBufferProvider != NULL) {
mTrackBufferProvider->releaseBuffer(pBuffer);
} else {
ALOGE("DownmixerBufferProvider::releaseBuffer() error: NULL track buffer provider");
@@ -90,9 +90,9 @@
// ----------------------------------------------------------------------------
-bool AudioMixer::isMultichannelCapable = false;
+bool AudioMixer::sIsMultichannelCapable = false;
-effect_descriptor_t AudioMixer::dwnmFxDesc;
+effect_descriptor_t AudioMixer::sDwnmFxDesc;
// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
// The value of 1 << x is undefined in C when x >= 32.
@@ -113,8 +113,6 @@
// AudioMixer is not yet capable of multi-channel output beyond stereo
ALOG_ASSERT(2 == MAX_NUM_CHANNELS, "bad MAX_NUM_CHANNELS %d", MAX_NUM_CHANNELS);
- LocalClock lc;
-
pthread_once(&sOnceControl, &sInitRoutine);
mState.enabledTracks= 0;
@@ -136,27 +134,6 @@
t++;
}
- // find multichannel downmix effect if we have to play multichannel content
- uint32_t numEffects = 0;
- int ret = EffectQueryNumberEffects(&numEffects);
- if (ret != 0) {
- ALOGE("AudioMixer() error %d querying number of effects", ret);
- return;
- }
- ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
-
- for (uint32_t i = 0 ; i < numEffects ; i++) {
- if (EffectQueryEffect(i, &dwnmFxDesc) == 0) {
- ALOGV("effect %d is called %s", i, dwnmFxDesc.name);
- if (memcmp(&dwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
- ALOGI("found effect \"%s\" from %s",
- dwnmFxDesc.name, dwnmFxDesc.implementor);
- isMultichannelCapable = true;
- break;
- }
- }
- }
- ALOGE_IF(!isMultichannelCapable, "unable to find downmix effect");
}
AudioMixer::~AudioMixer()
@@ -229,7 +206,7 @@
void AudioMixer::invalidateState(uint32_t mask)
{
- if (mask) {
+ if (mask != 0) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
@@ -252,7 +229,7 @@
return status;
}
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName) {
+void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
if (pTrack->downmixerBufferProvider != NULL) {
@@ -276,13 +253,13 @@
DownmixerBufferProvider* pDbp = new DownmixerBufferProvider();
int32_t status;
- if (!isMultichannelCapable) {
+ if (!sIsMultichannelCapable) {
ALOGE("prepareTrackForDownmix(%d) fails: mixer doesn't support multichannel content",
trackName);
goto noDownmixForActiveTrack;
}
- if (EffectCreate(&dwnmFxDesc.uuid,
+ if (EffectCreate(&sDwnmFxDesc.uuid,
pTrack->sessionId /*sessionId*/, -2 /*ioId not relevant here, using random value*/,
&pDbp->mDownmixHandle/*pHandle*/) != 0) {
ALOGE("prepareTrackForDownmix(%d) fails: error creating downmixer effect", trackName);
@@ -560,14 +537,14 @@
// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
if (!((value == 44100 && devSampleRate == 48000) ||
(value == 48000 && devSampleRate == 44100))) {
- quality = AudioResampler::LOW_QUALITY;
+ quality = AudioResampler::DYN_LOW_QUALITY;
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
resampler = AudioResampler::create(
format,
// the resampler sees the number of channels after the downmixer, if any
- downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
+ (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
devSampleRate, quality);
resampler->setLocalTimeFreq(sLocalTimeFreq);
}
@@ -668,27 +645,29 @@
countActiveTracks++;
track_t& t = state->tracks[i];
uint32_t n = 0;
+ // FIXME can overflow (mask is only 3 bits)
n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
- n |= NEEDS_FORMAT_16;
- n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
+ if (t.doesResample()) {
+ n |= NEEDS_RESAMPLE;
+ }
if (t.auxLevel != 0 && t.auxBuffer != NULL) {
- n |= NEEDS_AUX_ENABLED;
+ n |= NEEDS_AUX;
}
if (t.volumeInc[0]|t.volumeInc[1]) {
volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
- n |= NEEDS_MUTE_ENABLED;
+ n |= NEEDS_MUTE;
}
t.needs = n;
- if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
+ if (n & NEEDS_MUTE) {
t.hook = track__nop;
} else {
- if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (n & NEEDS_AUX) {
all16BitsStereoNoResample = false;
}
- if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+ if (n & NEEDS_RESAMPLE) {
all16BitsStereoNoResample = false;
resampling = true;
t.hook = track__genericResample;
@@ -710,7 +689,7 @@
// select the processing hooks
state->hook = process__nop;
- if (countActiveTracks) {
+ if (countActiveTracks > 0) {
if (resampling) {
if (!state->outputTemp) {
state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -746,16 +725,15 @@
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
- if (countActiveTracks) {
+ if (countActiveTracks > 0) {
bool allMuted = true;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
track_t& t = state->tracks[i];
- if (!t.doesResample() && t.volumeRL == 0)
- {
- t.needs |= NEEDS_MUTE_ENABLED;
+ if (!t.doesResample() && t.volumeRL == 0) {
+ t.needs |= NEEDS_MUTE;
t.hook = track__nop;
} else {
allMuted = false;
@@ -806,8 +784,8 @@
}
}
-void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp,
- int32_t* aux)
+void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+ size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
{
}
@@ -883,8 +861,8 @@
}
}
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
- int32_t* aux)
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
+ int32_t* temp __unused, int32_t* aux)
{
const int16_t *in = static_cast<const int16_t *>(t->in);
@@ -974,8 +952,8 @@
t->in = in;
}
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
- int32_t* aux)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
+ int32_t* temp __unused, int32_t* aux)
{
const int16_t *in = static_cast<int16_t const *>(t->in);
@@ -1154,7 +1132,7 @@
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
- if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
+ if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
@@ -1166,7 +1144,7 @@
break;
}
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
- if (inFrames) {
+ if (inFrames > 0) {
t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
state->resampleTemp, aux);
t.frameCount -= inFrames;
@@ -1242,14 +1220,14 @@
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
- if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
+ if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
aux = t.auxBuffer;
}
// this is a little goofy, on the resampling case we don't
// acquire/release the buffers because it's done by
// the resampler.
- if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+ if (t.needs & NEEDS_RESAMPLE) {
t.resampler->setPTS(pts);
t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
@@ -1449,8 +1427,9 @@
int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex)
{
- if (AudioBufferProvider::kInvalidPTS == basePTS)
+ if (AudioBufferProvider::kInvalidPTS == basePTS) {
return AudioBufferProvider::kInvalidPTS;
+ }
return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
}
@@ -1462,6 +1441,28 @@
{
LocalClock lc;
sLocalTimeFreq = lc.getLocalFreq();
+
+ // find multichannel downmix effect if we have to play multichannel content
+ uint32_t numEffects = 0;
+ int ret = EffectQueryNumberEffects(&numEffects);
+ if (ret != 0) {
+ ALOGE("AudioMixer() error %d querying number of effects", ret);
+ return;
+ }
+ ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
+
+ for (uint32_t i = 0 ; i < numEffects ; i++) {
+ if (EffectQueryEffect(i, &sDwnmFxDesc) == 0) {
+ ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
+ if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
+ ALOGI("found effect \"%s\" from %s",
+ sDwnmFxDesc.name, sDwnmFxDesc.implementor);
+ sIsMultichannelCapable = true;
+ break;
+ }
+ }
+ }
+ ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
}
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 43aeb86..d286986 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -120,27 +120,19 @@
private:
enum {
+ // FIXME this representation permits up to 8 channels
NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
- NEEDS_FORMAT__MASK = 0x000000F0,
- NEEDS_MUTE__MASK = 0x00000100,
- NEEDS_RESAMPLE__MASK = 0x00001000,
- NEEDS_AUX__MASK = 0x00010000,
};
enum {
- NEEDS_CHANNEL_1 = 0x00000000,
- NEEDS_CHANNEL_2 = 0x00000001,
+ NEEDS_CHANNEL_1 = 0x00000000, // mono
+ NEEDS_CHANNEL_2 = 0x00000001, // stereo
- NEEDS_FORMAT_16 = 0x00000010,
+ // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
- NEEDS_MUTE_DISABLED = 0x00000000,
- NEEDS_MUTE_ENABLED = 0x00000100,
-
- NEEDS_RESAMPLE_DISABLED = 0x00000000,
- NEEDS_RESAMPLE_ENABLED = 0x00001000,
-
- NEEDS_AUX_DISABLED = 0x00000000,
- NEEDS_AUX_ENABLED = 0x00010000,
+ NEEDS_MUTE = 0x00000100,
+ NEEDS_RESAMPLE = 0x00001000,
+ NEEDS_AUX = 0x00010000,
};
struct state_t;
@@ -224,7 +216,7 @@
NBLog::Writer* mLog;
int32_t reserved[1];
// FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
- track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
+ track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
};
// AudioBufferProvider that wraps a track AudioBufferProvider by a call to a downmix effect
@@ -256,9 +248,9 @@
state_t mState __attribute__((aligned(32)));
// effect descriptor for the downmixer used by the mixer
- static effect_descriptor_t dwnmFxDesc;
+ static effect_descriptor_t sDwnmFxDesc;
// indicates whether a downmix effect has been found and is usable by this mixer
- static bool isMultichannelCapable;
+ static bool sIsMultichannelCapable;
// Call after changing either the enabled status of a track, or parameters of an enabled track.
// OK to call more often than that, but unnecessary.
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 646a317..9980344 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -60,7 +60,7 @@
// ----------------------------------------------------------------------------
AudioPolicyService::AudioPolicyService()
- : BnAudioPolicyService() , mpAudioPolicyDev(NULL) , mpAudioPolicy(NULL)
+ : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL)
{
char value[PROPERTY_VALUE_MAX];
const struct hw_module_t *module;
@@ -77,24 +77,28 @@
mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
/* instantiate the audio policy manager */
rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
- if (rc)
+ if (rc) {
return;
+ }
rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc));
- if (rc)
+ if (rc) {
return;
+ }
rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this,
&mpAudioPolicy);
ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc));
- if (rc)
+ if (rc) {
return;
+ }
rc = mpAudioPolicy->init_check(mpAudioPolicy);
ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc));
- if (rc)
+ if (rc) {
return;
+ }
ALOGI("Loaded audio policy from %s (%s)", module->name, module->id);
@@ -126,10 +130,12 @@
}
mInputs.clear();
- if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL)
+ if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) {
mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
- if (mpAudioPolicyDev != NULL)
+ }
+ if (mpAudioPolicyDev != NULL) {
audio_policy_dev_close(mpAudioPolicyDev);
+ }
}
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
@@ -469,8 +475,9 @@
audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc)
{
+ // FIXME change return type to status_t, and return NO_INIT here
if (mpAudioPolicy == NULL) {
- return NO_INIT;
+ return 0;
}
Mutex::Autolock _l(mLock);
return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
@@ -606,7 +613,7 @@
return NO_ERROR;
}
-status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args __unused)
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd);
@@ -1114,11 +1121,13 @@
int AudioPolicyService::startTone(audio_policy_tone_t tone,
audio_stream_type_t stream)
{
- if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION)
+ if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) {
ALOGE("startTone: illegal tone requested (%d)", tone);
- if (stream != AUDIO_STREAM_VOICE_CALL)
+ }
+ if (stream != AUDIO_STREAM_VOICE_CALL) {
ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
tone);
+ }
mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
AUDIO_STREAM_VOICE_CALL);
return 0;
@@ -1452,7 +1461,7 @@
extern "C" {
-static audio_module_handle_t aps_load_hw_module(void *service,
+static audio_module_handle_t aps_load_hw_module(void *service __unused,
const char *name)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1465,7 +1474,7 @@
}
// deprecated: replaced by aps_open_output_on_module()
-static audio_io_handle_t aps_open_output(void *service,
+static audio_io_handle_t aps_open_output(void *service __unused,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
@@ -1483,7 +1492,7 @@
pLatencyMs, flags);
}
-static audio_io_handle_t aps_open_output_on_module(void *service,
+static audio_io_handle_t aps_open_output_on_module(void *service __unused,
audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
@@ -1502,7 +1511,7 @@
pLatencyMs, flags, offloadInfo);
}
-static audio_io_handle_t aps_open_dup_output(void *service,
+static audio_io_handle_t aps_open_dup_output(void *service __unused,
audio_io_handle_t output1,
audio_io_handle_t output2)
{
@@ -1514,16 +1523,17 @@
return af->openDuplicateOutput(output1, output2);
}
-static int aps_close_output(void *service, audio_io_handle_t output)
+static int aps_close_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == 0)
+ if (af == 0) {
return PERMISSION_DENIED;
+ }
return af->closeOutput(output);
}
-static int aps_suspend_output(void *service, audio_io_handle_t output)
+static int aps_suspend_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1534,7 +1544,7 @@
return af->suspendOutput(output);
}
-static int aps_restore_output(void *service, audio_io_handle_t output)
+static int aps_restore_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1546,12 +1556,12 @@
}
// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
-static audio_io_handle_t aps_open_input(void *service,
+static audio_io_handle_t aps_open_input(void *service __unused,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
- audio_in_acoustics_t acoustics)
+ audio_in_acoustics_t acoustics __unused)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1562,7 +1572,7 @@
return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
}
-static audio_io_handle_t aps_open_input_on_module(void *service,
+static audio_io_handle_t aps_open_input_on_module(void *service __unused,
audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
@@ -1578,37 +1588,40 @@
return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
}
-static int aps_close_input(void *service, audio_io_handle_t input)
+static int aps_close_input(void *service __unused, audio_io_handle_t input)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == 0)
+ if (af == 0) {
return PERMISSION_DENIED;
+ }
return af->closeInput(input);
}
-static int aps_set_stream_output(void *service, audio_stream_type_t stream,
+static int aps_set_stream_output(void *service __unused, audio_stream_type_t stream,
audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == 0)
+ if (af == 0) {
return PERMISSION_DENIED;
+ }
return af->setStreamOutput(stream, output);
}
-static int aps_move_effects(void *service, int session,
+static int aps_move_effects(void *service __unused, int session,
audio_io_handle_t src_output,
audio_io_handle_t dst_output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == 0)
+ if (af == 0) {
return PERMISSION_DENIED;
+ }
return af->moveEffects(session, src_output, dst_output);
}
-static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
+static char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
const char *keys)
{
String8 result = AudioSystem::getParameters(io_handle, String8(keys));
@@ -1659,24 +1672,24 @@
namespace {
struct audio_policy_service_ops aps_ops = {
- open_output : aps_open_output,
- open_duplicate_output : aps_open_dup_output,
- close_output : aps_close_output,
- suspend_output : aps_suspend_output,
- restore_output : aps_restore_output,
- open_input : aps_open_input,
- close_input : aps_close_input,
- set_stream_volume : aps_set_stream_volume,
- set_stream_output : aps_set_stream_output,
- set_parameters : aps_set_parameters,
- get_parameters : aps_get_parameters,
- start_tone : aps_start_tone,
- stop_tone : aps_stop_tone,
- set_voice_volume : aps_set_voice_volume,
- move_effects : aps_move_effects,
- load_hw_module : aps_load_hw_module,
- open_output_on_module : aps_open_output_on_module,
- open_input_on_module : aps_open_input_on_module,
+ .open_output = aps_open_output,
+ .open_duplicate_output = aps_open_dup_output,
+ .close_output = aps_close_output,
+ .suspend_output = aps_suspend_output,
+ .restore_output = aps_restore_output,
+ .open_input = aps_open_input,
+ .close_input = aps_close_input,
+ .set_stream_volume = aps_set_stream_volume,
+ .set_stream_output = aps_set_stream_output,
+ .set_parameters = aps_set_parameters,
+ .get_parameters = aps_get_parameters,
+ .start_tone = aps_start_tone,
+ .stop_tone = aps_stop_tone,
+ .set_voice_volume = aps_set_voice_volume,
+ .move_effects = aps_move_effects,
+ .load_hw_module = aps_load_hw_module,
+ .open_output_on_module = aps_open_output_on_module,
+ .open_input_on_module = aps_open_input_on_module,
};
}; // namespace <unnamed>
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index e5cceb1..b206116 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -25,6 +25,7 @@
#include "AudioResampler.h"
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
+#include "AudioResamplerDyn.h"
#ifdef __arm__
#include <machine/cpu-features.h>
@@ -77,6 +78,9 @@
int mX0R;
};
+/*static*/
+const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
+
bool AudioResampler::qualityIsSupported(src_quality quality)
{
switch (quality) {
@@ -85,6 +89,9 @@
case MED_QUALITY:
case HIGH_QUALITY:
case VERY_HIGH_QUALITY:
+ case DYN_LOW_QUALITY:
+ case DYN_MED_QUALITY:
+ case DYN_HIGH_QUALITY:
return true;
default:
return false;
@@ -105,7 +112,7 @@
if (*endptr == '\0') {
defaultQuality = (src_quality) l;
ALOGD("forcing AudioResampler quality to %d", defaultQuality);
- if (defaultQuality < DEFAULT_QUALITY || defaultQuality > VERY_HIGH_QUALITY) {
+ if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
defaultQuality = DEFAULT_QUALITY;
}
}
@@ -125,6 +132,12 @@
return 20;
case VERY_HIGH_QUALITY:
return 34;
+ case DYN_LOW_QUALITY:
+ return 4;
+ case DYN_MED_QUALITY:
+ return 6;
+ case DYN_HIGH_QUALITY:
+ return 12;
}
}
@@ -148,6 +161,16 @@
atFinalQuality = true;
}
+ /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+ * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+ * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+ * due to estimated CPU load of having too many active resamplers
+ * (the code below the if).
+ */
+ if (quality == DEFAULT_QUALITY) {
+ quality = DYN_MED_QUALITY;
+ }
+
// naive implementation of CPU load throttling doesn't account for whether resampler is active
pthread_mutex_lock(&mutex);
for (;;) {
@@ -162,7 +185,6 @@
// not enough CPU available for proposed quality level, so try next lowest level
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
atFinalQuality = true;
break;
@@ -175,6 +197,15 @@
case VERY_HIGH_QUALITY:
quality = HIGH_QUALITY;
break;
+ case DYN_LOW_QUALITY:
+ atFinalQuality = true;
+ break;
+ case DYN_MED_QUALITY:
+ quality = DYN_LOW_QUALITY;
+ break;
+ case DYN_HIGH_QUALITY:
+ quality = DYN_MED_QUALITY;
+ break;
}
}
pthread_mutex_unlock(&mutex);
@@ -183,7 +214,6 @@
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
@@ -200,6 +230,12 @@
ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
break;
+ case DYN_LOW_QUALITY:
+ case DYN_MED_QUALITY:
+ case DYN_HIGH_QUALITY:
+ ALOGV("Create dynamic Resampler = %d", quality);
+ resampler = new AudioResamplerDyn(bitDepth, inChannelCount, sampleRate, quality);
+ break;
}
// initialize resampler
@@ -339,8 +375,9 @@
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (outputIndex == outputSampleCount)
+ if (outputIndex == outputSampleCount) {
break;
+ }
}
// process input samples
@@ -434,8 +471,9 @@
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (outputIndex == outputSampleCount)
+ if (outputIndex == outputSampleCount) {
break;
+ }
}
// process input samples
@@ -514,6 +552,16 @@
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
+ (void)maxOutPt; // remove unused parameter warnings
+ (void)maxInIdx;
+ (void)outputIndex;
+ (void)out;
+ (void)inputIndex;
+ (void)vl;
+ (void)vr;
+ (void)phaseFraction;
+ (void)phaseIncrement;
+ (void)in;
#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
asm(
@@ -625,6 +673,16 @@
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
+ (void)maxOutPt; // remove unused parameter warnings
+ (void)maxInIdx;
+ (void)outputIndex;
+ (void)out;
+ (void)inputIndex;
+ (void)vl;
+ (void)vr;
+ (void)phaseFraction;
+ (void)phaseIncrement;
+ (void)in;
#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex)
asm(
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 33e64ce..dc33f29 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -41,6 +41,9 @@
MED_QUALITY=2,
HIGH_QUALITY=3,
VERY_HIGH_QUALITY=4,
+ DYN_LOW_QUALITY=5,
+ DYN_MED_QUALITY=6,
+ DYN_HIGH_QUALITY=7,
};
static AudioResampler* create(int bitDepth, int inChannelCount,
@@ -81,7 +84,7 @@
static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
// multiplier to calculate fixed point phase increment
- static const double kPhaseMultiplier = 1L << kNumPhaseBits;
+ static const double kPhaseMultiplier;
AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 18e59e9..1f9714b 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -66,8 +66,9 @@
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
- if (mBuffer.raw == NULL)
+ if (mBuffer.raw == NULL) {
return;
+ }
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
@@ -97,8 +98,9 @@
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
- if (mBuffer.raw == NULL)
+ if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
+ }
in = mBuffer.i16;
// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
@@ -132,8 +134,9 @@
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer, mPTS);
- if (mBuffer.raw == NULL)
+ if (mBuffer.raw == NULL) {
return;
+ }
// ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
@@ -163,8 +166,9 @@
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
- if (mBuffer.raw == NULL)
+ if (mBuffer.raw == NULL) {
goto save_state; // ugly, but efficient
+ }
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
in = mBuffer.i16;
}
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
new file mode 100644
index 0000000..cd67df5
--- /dev/null
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -0,0 +1,547 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioResamplerDyn"
+//#define LOG_NDEBUG 0
+
+#include <malloc.h>
+#include <string.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <math.h>
+
+#include <cutils/compiler.h>
+#include <cutils/properties.h>
+#include <utils/Log.h>
+
+#include "AudioResamplerFirOps.h" // USE_NEON and USE_INLINE_ASSEMBLY defined here
+#include "AudioResamplerFirProcess.h"
+#include "AudioResamplerFirProcessNeon.h"
+#include "AudioResamplerFirGen.h" // requires math.h
+#include "AudioResamplerDyn.h"
+
+//#define DEBUG_RESAMPLER
+
+namespace android {
+
+// generate a unique resample type compile-time constant (constexpr)
+#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE, COEFTYPE) \
+ ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 | (COEFTYPE)<<2 \
+ | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<3)
+
+/*
+ * InBuffer is a type agnostic input buffer.
+ *
+ * Layout of the state buffer for halfNumCoefs=8.
+ *
+ * [rrrrrrppppppppnnnnnnnnrrrrrrrrrrrrrrrrrrr.... rrrrrrr]
+ * S I R
+ *
+ * S = mState
+ * I = mImpulse
+ * R = mRingFull
+ * p = past samples, convoluted with the (p)ositive side of sinc()
+ * n = future samples, convoluted with the (n)egative side of sinc()
+ * r = extra space for implementing the ring buffer
+ */
+
+template<typename TI>
+AudioResamplerDyn::InBuffer<TI>::InBuffer()
+ : mState(NULL), mImpulse(NULL), mRingFull(NULL), mStateSize(0) {
+}
+
+template<typename TI>
+AudioResamplerDyn::InBuffer<TI>::~InBuffer() {
+ init();
+}
+
+template<typename TI>
+void AudioResamplerDyn::InBuffer<TI>::init() {
+ free(mState);
+ mState = NULL;
+ mImpulse = NULL;
+ mRingFull = NULL;
+ mStateSize = 0;
+}
+
+// resizes the state buffer to accommodate the appropriate filter length
+template<typename TI>
+void AudioResamplerDyn::InBuffer<TI>::resize(int CHANNELS, int halfNumCoefs) {
+ // calculate desired state size
+ int stateSize = halfNumCoefs * CHANNELS * 2
+ * kStateSizeMultipleOfFilterLength;
+
+ // check if buffer needs resizing
+ if (mState
+ && stateSize == mStateSize
+ && mRingFull-mState == mStateSize-halfNumCoefs*CHANNELS) {
+ return;
+ }
+
+ // create new buffer
+ TI* state = (int16_t*)memalign(32, stateSize*sizeof(*state));
+ memset(state, 0, stateSize*sizeof(*state));
+
+ // attempt to preserve state
+ if (mState) {
+ TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
+ TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
+ TI* dst = state;
+
+ if (srcLo < mState) {
+ dst += mState-srcLo;
+ srcLo = mState;
+ }
+ if (srcHi > mState + mStateSize) {
+ srcHi = mState + mStateSize;
+ }
+ memcpy(dst, srcLo, (srcHi - srcLo) * sizeof(*srcLo));
+ free(mState);
+ }
+
+ // set class member vars
+ mState = state;
+ mStateSize = stateSize;
+ mImpulse = mState + halfNumCoefs*CHANNELS; // actually one sample greater than needed
+ mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS;
+}
+
+// copy in the input data into the head (impulse+halfNumCoefs) of the buffer.
+template<typename TI>
+template<int CHANNELS>
+void AudioResamplerDyn::InBuffer<TI>::readAgain(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex) {
+ int16_t* head = impulse + halfNumCoefs*CHANNELS;
+ for (size_t i=0 ; i<CHANNELS ; i++) {
+ head[i] = in[inputIndex*CHANNELS + i];
+ }
+}
+
+// advance the impulse pointer, and load in data into the head (impulse+halfNumCoefs)
+template<typename TI>
+template<int CHANNELS>
+void AudioResamplerDyn::InBuffer<TI>::readAdvance(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex) {
+ impulse += CHANNELS;
+
+ if (CC_UNLIKELY(impulse >= mRingFull)) {
+ const size_t shiftDown = mRingFull - mState - halfNumCoefs*CHANNELS;
+ memcpy(mState, mState+shiftDown, halfNumCoefs*CHANNELS*2*sizeof(TI));
+ impulse -= shiftDown;
+ }
+ readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+}
+
+void AudioResamplerDyn::Constants::set(
+ int L, int halfNumCoefs, int inSampleRate, int outSampleRate)
+{
+ int bits = 0;
+ int lscale = inSampleRate/outSampleRate < 2 ? L - 1 :
+ static_cast<int>(static_cast<uint64_t>(L)*inSampleRate/outSampleRate);
+ for (int i=lscale; i; ++bits, i>>=1)
+ ;
+ mL = L;
+ mShift = kNumPhaseBits - bits;
+ mHalfNumCoefs = halfNumCoefs;
+}
+
+AudioResamplerDyn::AudioResamplerDyn(int bitDepth,
+ int inChannelCount, int32_t sampleRate, src_quality quality)
+ : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+ mResampleType(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
+ mCoefBuffer(NULL)
+{
+ mVolumeSimd[0] = mVolumeSimd[1] = 0;
+ mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
+}
+
+AudioResamplerDyn::~AudioResamplerDyn() {
+ free(mCoefBuffer);
+}
+
+void AudioResamplerDyn::init() {
+ mFilterSampleRate = 0; // always trigger new filter generation
+ mInBuffer.init();
+}
+
+void AudioResamplerDyn::setVolume(int16_t left, int16_t right) {
+ AudioResampler::setVolume(left, right);
+ mVolumeSimd[0] = static_cast<int32_t>(left)<<16;
+ mVolumeSimd[1] = static_cast<int32_t>(right)<<16;
+}
+
+template <typename T> T max(T a, T b) {return a > b ? a : b;}
+
+template <typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;}
+
+template<typename T>
+void AudioResamplerDyn::createKaiserFir(Constants &c, double stopBandAtten,
+ int inSampleRate, int outSampleRate, double tbwCheat) {
+ T* buf = reinterpret_cast<T*>(memalign(32, (c.mL+1)*c.mHalfNumCoefs*sizeof(T)));
+ static const double atten = 0.9998; // to avoid ripple overflow
+ double fcr;
+ double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
+
+ if (inSampleRate < outSampleRate) { // upsample
+ fcr = max(0.5*tbwCheat - tbw/2, tbw/2);
+ } else { // downsample
+ fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2);
+ }
+ // create and set filter
+ firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten);
+ c.setBuf(buf);
+ if (mCoefBuffer) {
+ free(mCoefBuffer);
+ }
+ mCoefBuffer = buf;
+#ifdef DEBUG_RESAMPLER
+ // print basic filter stats
+ printf("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n",
+ c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw);
+ // test the filter and report results
+ double fp = (fcr - tbw/2)/c.mL;
+ double fs = (fcr + tbw/2)/c.mL;
+ double passMin, passMax, passRipple;
+ double stopMax, stopRipple;
+ testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
+ passMin, passMax, passRipple, stopMax, stopRipple);
+ printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
+ printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
+#endif
+}
+
+// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop.
+static int gcd(int n, int m) {
+ if (m == 0) {
+ return n;
+ }
+ return gcd(m, n % m);
+}
+
+static bool isClose(int32_t newSampleRate, int32_t prevSampleRate,
+ int32_t filterSampleRate, int32_t outSampleRate) {
+
+ // different upsampling ratios do not need a filter change.
+ if (filterSampleRate != 0
+ && filterSampleRate < outSampleRate
+ && newSampleRate < outSampleRate)
+ return true;
+
+ // check design criteria again if downsampling is detected.
+ int pdiff = absdiff(newSampleRate, prevSampleRate);
+ int adiff = absdiff(newSampleRate, filterSampleRate);
+
+ // allow up to 6% relative change increments.
+ // allow up to 12% absolute change increments (from filter design)
+ return pdiff < prevSampleRate>>4 && adiff < filterSampleRate>>3;
+}
+
+void AudioResamplerDyn::setSampleRate(int32_t inSampleRate) {
+ if (mInSampleRate == inSampleRate) {
+ return;
+ }
+ int32_t oldSampleRate = mInSampleRate;
+ int32_t oldHalfNumCoefs = mConstants.mHalfNumCoefs;
+ uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift;
+ bool useS32 = false;
+
+ mInSampleRate = inSampleRate;
+
+ // TODO: Add precalculated Equiripple filters
+
+ if (mFilterQuality != getQuality() ||
+ !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) {
+ mFilterSampleRate = inSampleRate;
+ mFilterQuality = getQuality();
+
+ // Begin Kaiser Filter computation
+ //
+ // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB.
+ // Keep the stop band attenuation no greater than 84-85dB for 32 length S16 filters
+ //
+ // For s32 we keep the stop band attenuation at the same as 16b resolution, about
+ // 96-98dB
+ //
+
+ double stopBandAtten;
+ double tbwCheat = 1.; // how much we "cheat" into aliasing
+ int halfLength;
+ if (mFilterQuality == DYN_HIGH_QUALITY) {
+ // 32b coefficients, 64 length
+ useS32 = true;
+ stopBandAtten = 98.;
+ halfLength = 32;
+ } else if (mFilterQuality == DYN_LOW_QUALITY) {
+ // 16b coefficients, 16-32 length
+ useS32 = false;
+ stopBandAtten = 80.;
+ if (mSampleRate >= inSampleRate * 2) {
+ halfLength = 16;
+ } else {
+ halfLength = 8;
+ }
+ if (mSampleRate >= inSampleRate) {
+ tbwCheat = 1.05;
+ } else {
+ tbwCheat = 1.03;
+ }
+ } else { // DYN_MED_QUALITY
+ // 16b coefficients, 32-64 length
+ // note: > 64 length filters with 16b coefs can have quantization noise problems
+ useS32 = false;
+ stopBandAtten = 84.;
+ if (mSampleRate >= inSampleRate * 4) {
+ halfLength = 32;
+ } else if (mSampleRate >= inSampleRate * 2) {
+ halfLength = 24;
+ } else {
+ halfLength = 16;
+ }
+ if (mSampleRate >= inSampleRate) {
+ tbwCheat = 1.03;
+ } else {
+ tbwCheat = 1.01;
+ }
+ }
+
+ // determine the number of polyphases in the filterbank.
+ // for 16b, it is desirable to have 2^(16/2) = 256 phases.
+ // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html
+ //
+ // We are a bit more lax on this.
+
+ int phases = mSampleRate / gcd(mSampleRate, inSampleRate);
+
+ // TODO: Once dynamic sample rate change is an option, the code below
+ // should be modified to execute only when dynamic sample rate change is enabled.
+ //
+ // as above, #phases less than 63 is too few phases for accurate linear interpolation.
+ // we increase the phases to compensate, but more phases means more memory per
+ // filter and more time to compute the filter.
+ //
+ // if we know that the filter will be used for dynamic sample rate changes,
+ // that would allow us skip this part for fixed sample rate resamplers.
+ //
+ while (phases<63) {
+ phases *= 2; // this code only needed to support dynamic rate changes
+ }
+
+ if (phases>=256) { // too many phases, always interpolate
+ phases = 127;
+ }
+
+ // create the filter
+ mConstants.set(phases, halfLength, inSampleRate, mSampleRate);
+ if (useS32) {
+ createKaiserFir<int32_t>(mConstants, stopBandAtten,
+ inSampleRate, mSampleRate, tbwCheat);
+ } else {
+ createKaiserFir<int16_t>(mConstants, stopBandAtten,
+ inSampleRate, mSampleRate, tbwCheat);
+ }
+ } // End Kaiser filter
+
+ // update phase and state based on the new filter.
+ const Constants& c(mConstants);
+ mInBuffer.resize(mChannelCount, c.mHalfNumCoefs);
+ const uint32_t phaseWrapLimit = c.mL << c.mShift;
+ // try to preserve as much of the phase fraction as possible for on-the-fly changes
+ mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)
+ * phaseWrapLimit / oldPhaseWrapLimit;
+ mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.
+ mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit)
+ * inSampleRate / mSampleRate);
+
+ // determine which resampler to use
+ // check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
+ int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
+ int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2;
+ if (locked) {
+ mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
+ }
+
+ mResampleType = RESAMPLETYPE(mChannelCount, locked, stride, !!useS32);
+#ifdef DEBUG_RESAMPLER
+ printf("channels:%d %s stride:%d %s coef:%d shift:%d\n",
+ mChannelCount, locked ? "locked" : "interpolated",
+ stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift);
+#endif
+}
+
+void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider)
+{
+ // TODO:
+ // 24 cases - this perhaps can be reduced later, as testing might take too long
+ switch (mResampleType) {
+
+ // stride 16 (falls back to stride 2 for machines that do not support NEON)
+ case RESAMPLETYPE(1, true, 16, 0):
+ return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, true, 16, 0):
+ return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, false, 16, 0):
+ return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, false, 16, 0):
+ return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, true, 16, 1):
+ return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, true, 16, 1):
+ return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(1, false, 16, 1):
+ return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, false, 16, 1):
+ return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+#if 0
+ // TODO: Remove these?
+ // stride 8
+ case RESAMPLETYPE(1, true, 8, 0):
+ return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, true, 8, 0):
+ return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, false, 8, 0):
+ return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, false, 8, 0):
+ return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, true, 8, 1):
+ return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, true, 8, 1):
+ return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(1, false, 8, 1):
+ return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, false, 8, 1):
+ return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ // stride 2 (can handle any filter length)
+ case RESAMPLETYPE(1, true, 2, 0):
+ return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, true, 2, 0):
+ return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, false, 2, 0):
+ return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(2, false, 2, 0):
+ return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
+ case RESAMPLETYPE(1, true, 2, 1):
+ return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, true, 2, 1):
+ return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(1, false, 2, 1):
+ return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+ case RESAMPLETYPE(2, false, 2, 1):
+ return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
+#endif
+ default:
+ ; // error
+ }
+}
+
+template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
+ const TC* const coefs, AudioBufferProvider* provider)
+{
+ const Constants& c(mConstants);
+ int16_t* impulse = mInBuffer.getImpulse();
+ size_t inputIndex = mInputIndex;
+ uint32_t phaseFraction = mPhaseFraction;
+ const uint32_t phaseIncrement = mPhaseIncrement;
+ size_t outputIndex = 0;
+ size_t outputSampleCount = outFrameCount * 2; // stereo output
+ size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+ const uint32_t phaseWrapLimit = c.mL << c.mShift;
+
+ // NOTE: be very careful when modifying the code here. register
+ // pressure is very high and a small change might cause the compiler
+ // to generate far less efficient code.
+ // Always sanity check the result with objdump or test-resample.
+
+ // the following logic is a bit convoluted to keep the main processing loop
+ // as tight as possible with register allocation.
+ while (outputIndex < outputSampleCount) {
+ // buffer is empty, fetch a new one
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
+ if (mBuffer.raw == NULL) {
+ goto resample_exit;
+ }
+ if (phaseFraction >= phaseWrapLimit) { // read in data
+ mInBuffer.readAdvance<CHANNELS>(
+ impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
+ phaseFraction -= phaseWrapLimit;
+ while (phaseFraction >= phaseWrapLimit) {
+ inputIndex++;
+ if (inputIndex >= mBuffer.frameCount) {
+ inputIndex -= mBuffer.frameCount;
+ provider->releaseBuffer(&mBuffer);
+ break;
+ }
+ mInBuffer.readAdvance<CHANNELS>(
+ impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
+ phaseFraction -= phaseWrapLimit;
+ }
+ }
+ }
+ const int16_t* const in = mBuffer.i16;
+ const size_t frameCount = mBuffer.frameCount;
+ const int coefShift = c.mShift;
+ const int halfNumCoefs = c.mHalfNumCoefs;
+ const int32_t* const volumeSimd = mVolumeSimd;
+
+ // reread the last input in.
+ mInBuffer.readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+
+ // main processing loop
+ while (CC_LIKELY(outputIndex < outputSampleCount)) {
+ // caution: fir() is inlined and may be large.
+ // output will be loaded with the appropriate values
+ //
+ // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs]
+ // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs.
+ //
+ fir<CHANNELS, LOCKED, STRIDE>(
+ &out[outputIndex],
+ phaseFraction, phaseWrapLimit,
+ coefShift, halfNumCoefs, coefs,
+ impulse, volumeSimd);
+ outputIndex += 2;
+
+ phaseFraction += phaseIncrement;
+ while (phaseFraction >= phaseWrapLimit) {
+ inputIndex++;
+ if (inputIndex >= frameCount) {
+ goto done; // need a new buffer
+ }
+ mInBuffer.readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+ phaseFraction -= phaseWrapLimit;
+ }
+ }
+done:
+ // often arrives here when input buffer runs out
+ if (inputIndex >= frameCount) {
+ inputIndex -= frameCount;
+ provider->releaseBuffer(&mBuffer);
+ // mBuffer.frameCount MUST be zero here.
+ }
+ }
+
+resample_exit:
+ mInBuffer.setImpulse(impulse);
+ mInputIndex = inputIndex;
+ mPhaseFraction = phaseFraction;
+}
+
+// ----------------------------------------------------------------------------
+}; // namespace android
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
new file mode 100644
index 0000000..df1fdbe
--- /dev/null
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -0,0 +1,124 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_DYN_H
+#define ANDROID_AUDIO_RESAMPLER_DYN_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+
+namespace android {
+
+class AudioResamplerDyn: public AudioResampler {
+public:
+ AudioResamplerDyn(int bitDepth, int inChannelCount, int32_t sampleRate,
+ src_quality quality);
+
+ virtual ~AudioResamplerDyn();
+
+ virtual void init();
+
+ virtual void setSampleRate(int32_t inSampleRate);
+
+ virtual void setVolume(int16_t left, int16_t right);
+
+ virtual void resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider);
+
+private:
+
+ class Constants { // stores the filter constants.
+ public:
+ Constants() :
+ mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefsS16(NULL)
+ {}
+ void set(int L, int halfNumCoefs,
+ int inSampleRate, int outSampleRate);
+ inline void setBuf(int16_t* buf) {
+ mFirCoefsS16 = buf;
+ }
+ inline void setBuf(int32_t* buf) {
+ mFirCoefsS32 = buf;
+ }
+
+ int mL; // interpolation phases in the filter.
+ int mShift; // right shift to get polyphase index
+ unsigned int mHalfNumCoefs; // filter half #coefs
+ union { // polyphase filter bank
+ const int16_t* mFirCoefsS16;
+ const int32_t* mFirCoefsS32;
+ };
+ };
+
+ // Input buffer management for a given input type TI, now (int16_t)
+ // Is agnostic of the actual type, can work with int32_t and float.
+ template<typename TI>
+ class InBuffer {
+ public:
+ InBuffer();
+ ~InBuffer();
+ void init();
+ void resize(int CHANNELS, int halfNumCoefs);
+
+ // used for direct management of the mImpulse pointer
+ inline TI* getImpulse() {
+ return mImpulse;
+ }
+ inline void setImpulse(TI *impulse) {
+ mImpulse = impulse;
+ }
+ template<int CHANNELS>
+ inline void readAgain(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex);
+ template<int CHANNELS>
+ inline void readAdvance(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex);
+
+ private:
+ // tuning parameter guidelines: 2 <= multiple <= 8
+ static const int kStateSizeMultipleOfFilterLength = 4;
+
+ TI* mState; // base pointer for the input buffer storage
+ TI* mImpulse; // current location of the impulse response (centered)
+ TI* mRingFull; // mState <= mImpulse < mRingFull
+ // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
+ size_t mStateSize; // in units of TI.
+ };
+
+ template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+ void resample(int32_t* out, size_t outFrameCount,
+ const TC* const coefs, AudioBufferProvider* provider);
+
+ template<typename T>
+ void createKaiserFir(Constants &c, double stopBandAtten,
+ int inSampleRate, int outSampleRate, double tbwCheat);
+
+ InBuffer<int16_t> mInBuffer;
+ Constants mConstants; // current set of coefficient parameters
+ int32_t __attribute__ ((aligned (8))) mVolumeSimd[2];
+ int32_t mResampleType; // contains the resample type.
+ int32_t mFilterSampleRate; // designed filter sample rate.
+ src_quality mFilterQuality; // designed filter quality.
+ void* mCoefBuffer; // if a filter is created, this is not null
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
new file mode 100644
index 0000000..fac3001
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -0,0 +1,684 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_GEN_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_GEN_H
+
+namespace android {
+
+/*
+ * generates a sine wave at equal steps.
+ *
+ * As most of our functions use sine or cosine at equal steps,
+ * it is very efficient to compute them that way (single multiply and subtract),
+ * rather than invoking the math library sin() or cos() each time.
+ *
+ * SineGen uses Goertzel's Algorithm (as a generator not a filter)
+ * to calculate sine(wstart + n * wstep) or cosine(wstart + n * wstep)
+ * by stepping through 0, 1, ... n.
+ *
+ * e^i(wstart+wstep) = 2cos(wstep) * e^i(wstart) - e^i(wstart-wstep)
+ *
+ * or looking at just the imaginary sine term, as the cosine follows identically:
+ *
+ * sin(wstart+wstep) = 2cos(wstep) * sin(wstart) - sin(wstart-wstep)
+ *
+ * Goertzel's algorithm is more efficient than the angle addition formula,
+ * e^i(wstart+wstep) = e^i(wstart) * e^i(wstep), which takes up to
+ * 4 multiplies and 2 adds (or 3* and 3+) and requires both sine and
+ * cosine generation due to the complex * complex multiply (full rotation).
+ *
+ * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
+ *
+ */
+
+class SineGen {
+public:
+ SineGen(double wstart, double wstep, bool cosine = false) {
+ if (cosine) {
+ mCurrent = cos(wstart);
+ mPrevious = cos(wstart - wstep);
+ } else {
+ mCurrent = sin(wstart);
+ mPrevious = sin(wstart - wstep);
+ }
+ mTwoCos = 2.*cos(wstep);
+ }
+ SineGen(double expNow, double expPrev, double twoCosStep) {
+ mCurrent = expNow;
+ mPrevious = expPrev;
+ mTwoCos = twoCosStep;
+ }
+ inline double value() const {
+ return mCurrent;
+ }
+ inline void advance() {
+ double tmp = mCurrent;
+ mCurrent = mCurrent*mTwoCos - mPrevious;
+ mPrevious = tmp;
+ }
+ inline double valueAdvance() {
+ double tmp = mCurrent;
+ mCurrent = mCurrent*mTwoCos - mPrevious;
+ mPrevious = tmp;
+ return tmp;
+ }
+
+private:
+ double mCurrent; // current value of sine/cosine
+ double mPrevious; // previous value of sine/cosine
+ double mTwoCos; // stepping factor
+};
+
+/*
+ * generates a series of sine generators, phase offset by fixed steps.
+ *
+ * This is used to generate polyphase sine generators, one per polyphase
+ * in the filter code below.
+ *
+ * The SineGen returned by value() starts at innerStart = outerStart + n*outerStep;
+ * increments by innerStep.
+ *
+ */
+
+class SineGenGen {
+public:
+ SineGenGen(double outerStart, double outerStep, double innerStep, bool cosine = false)
+ : mSineInnerCur(outerStart, outerStep, cosine),
+ mSineInnerPrev(outerStart-innerStep, outerStep, cosine)
+ {
+ mTwoCos = 2.*cos(innerStep);
+ }
+ inline SineGen value() {
+ return SineGen(mSineInnerCur.value(), mSineInnerPrev.value(), mTwoCos);
+ }
+ inline void advance() {
+ mSineInnerCur.advance();
+ mSineInnerPrev.advance();
+ }
+ inline SineGen valueAdvance() {
+ return SineGen(mSineInnerCur.valueAdvance(), mSineInnerPrev.valueAdvance(), mTwoCos);
+ }
+
+private:
+ SineGen mSineInnerCur; // generate the inner sine values (stepped by outerStep).
+ SineGen mSineInnerPrev; // generate the inner sine previous values
+ // (behind by innerStep, stepped by outerStep).
+ double mTwoCos; // the inner stepping factor for the returned SineGen.
+};
+
+static inline double sqr(double x) {
+ return x * x;
+}
+
+/*
+ * rounds a double to the nearest integer for FIR coefficients.
+ *
+ * One variant uses noise shaping, which must keep error history
+ * to work (the err parameter, initialized to 0).
+ * The other variant is a non-noise shaped version for
+ * S32 coefficients (noise shaping doesn't gain much).
+ *
+ * Caution: No bounds saturation is applied, but isn't needed in this case.
+ *
+ * @param x is the value to round.
+ *
+ * @param maxval is the maximum integer scale factor expressed as an int64 (for headroom).
+ * Typically this may be the maximum positive integer+1 (using the fact that double precision
+ * FIR coefficients generated here are never that close to 1.0 to pose an overflow condition).
+ *
+ * @param err is the previous error (actual - rounded) for the previous rounding op.
+ * For 16b coefficients this can improve stopband dB performance by up to 2dB.
+ *
+ * Many variants exist for the noise shaping: http://en.wikipedia.org/wiki/Noise_shaping
+ *
+ */
+
+static inline int64_t toint(double x, int64_t maxval, double& err) {
+ double val = x * maxval;
+ double ival = floor(val + 0.5 + err*0.2);
+ err = val - ival;
+ return static_cast<int64_t>(ival);
+}
+
+static inline int64_t toint(double x, int64_t maxval) {
+ return static_cast<int64_t>(floor(x * maxval + 0.5));
+}
+
+/*
+ * Modified Bessel function of the first kind
+ * http://en.wikipedia.org/wiki/Bessel_function
+ *
+ * The formulas are taken from Abramowitz and Stegun,
+ * _Handbook of Mathematical Functions_ (links below):
+ *
+ * http://people.math.sfu.ca/~cbm/aands/page_375.htm
+ * http://people.math.sfu.ca/~cbm/aands/page_378.htm
+ *
+ * http://dlmf.nist.gov/10.25
+ * http://dlmf.nist.gov/10.40
+ *
+ * Note we assume x is nonnegative (the function is symmetric,
+ * pass in the absolute value as needed).
+ *
+ * Constants are compile time derived with templates I0Term<> and
+ * I0ATerm<> to the precision of the compiler. The series can be expanded
+ * to any precision needed, but currently set around 24b precision.
+ *
+ * We use a bit of template math here, constexpr would probably be
+ * more appropriate for a C++11 compiler.
+ *
+ * For the intermediate range 3.75 < x < 15, we use minimax polynomial fit.
+ *
+ */
+
+template <int N>
+struct I0Term {
+ static const double value = I0Term<N-1>::value / (4. * N * N);
+};
+
+template <>
+struct I0Term<0> {
+ static const double value = 1.;
+};
+
+template <int N>
+struct I0ATerm {
+ static const double value = I0ATerm<N-1>::value * (2.*N-1.) * (2.*N-1.) / (8. * N);
+};
+
+template <>
+struct I0ATerm<0> { // 1/sqrt(2*PI);
+ static const double value = 0.398942280401432677939946059934381868475858631164934657665925;
+};
+
+#if USE_HORNERS_METHOD
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Horner's Method: http://en.wikipedia.org/wiki/Horner's_method
+ *
+ * This has fewer multiplications than Estrin's method below, but has back to back
+ * floating point dependencies.
+ *
+ * On ARM this appears to work slower, so USE_HORNERS_METHOD is not default enabled.
+ */
+
+inline double Poly2(double A, double B, double x) {
+ return A + x * B;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+ return A + x * (B + x * (C + x * (D)));
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+ double x) {
+ return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G))))));
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+ double H, double I, double x) {
+ return A + x * (B + x * (C + x * (D + x * (E + x * (F + x * (G + x * (H + x * (I))))))));
+}
+
+#else
+/* Polynomial evaluation of A + Bx + Cx^2 + Dx^3 + ...
+ * using Estrin's Method: http://en.wikipedia.org/wiki/Estrin's_scheme
+ *
+ * This is typically faster, perhaps gains about 5-10% overall on ARM processors
+ * over Horner's method above.
+ */
+
+inline double Poly2(double A, double B, double x) {
+ return A + B * x;
+}
+
+inline double Poly3(double A, double B, double C, double x, double x2) {
+ return Poly2(A, B, x) + C * x2;
+}
+
+inline double Poly3(double A, double B, double C, double x) {
+ return Poly2(A, B, x) + C * x * x;
+}
+
+inline double Poly4(double A, double B, double C, double D, double x, double x2) {
+ return Poly2(A, B, x) + Poly2(C, D, x) * x2; // same as poly2(poly2, poly2, x2);
+}
+
+inline double Poly4(double A, double B, double C, double D, double x) {
+ return Poly4(A, B, C, D, x, x * x);
+}
+
+inline double Poly7(double A, double B, double C, double D, double E, double F, double G,
+ double x) {
+ double x2 = x * x;
+ return Poly4(A, B, C, D, x, x2) + Poly3(E, F, G, x, x2) * (x2 * x2);
+}
+
+inline double Poly8(double A, double B, double C, double D, double E, double F, double G,
+ double H, double x, double x2, double x4) {
+ return Poly4(A, B, C, D, x, x2) + Poly4(E, F, G, H, x, x2) * x4;
+}
+
+inline double Poly9(double A, double B, double C, double D, double E, double F, double G,
+ double H, double I, double x) {
+ double x2 = x * x;
+#if 1
+ // It does not seem faster to explicitly decompose Poly8 into Poly4, but
+ // could depend on compiler floating point scheduling.
+ double x4 = x2 * x2;
+ return Poly8(A, B, C, D, E, F, G, H, x, x2, x4) + I * (x4 * x4);
+#else
+ double val = Poly4(A, B, C, D, x, x2);
+ double x4 = x2 * x2;
+ return val + Poly4(E, F, G, H, x, x2) * x4 + I * (x4 * x4);
+#endif
+}
+#endif
+
+static inline double I0(double x) {
+ if (x < 3.75) {
+ x *= x;
+ return Poly7(I0Term<0>::value, I0Term<1>::value,
+ I0Term<2>::value, I0Term<3>::value,
+ I0Term<4>::value, I0Term<5>::value,
+ I0Term<6>::value, x); // e < 1.6e-7
+ }
+ if (1) {
+ /*
+ * Series expansion coefs are easy to calculate, but are expanded around 0,
+ * so error is unequal over the interval 0 < x < 3.75, the error being
+ * significantly better near 0.
+ *
+ * A better solution is to use precise minimax polynomial fits.
+ *
+ * We use a slightly more complicated solution for 3.75 < x < 15, based on
+ * the tables in Blair and Edwards, "Stable Rational Minimax Approximations
+ * to the Modified Bessel Functions I0(x) and I1(x)", Chalk Hill Nuclear Laboratory,
+ * AECL-4928.
+ *
+ * http://www.iaea.org/inis/collection/NCLCollectionStore/_Public/06/178/6178667.pdf
+ *
+ * See Table 11 for 0 < x < 15; e < 10^(-7.13).
+ *
+ * Note: Beta cannot exceed 15 (hence Stopband cannot exceed 144dB = 24b).
+ *
+ * This speeds up overall computation by about 40% over using the else clause below,
+ * which requires sqrt and exp.
+ *
+ */
+
+ x *= x;
+ double num = Poly9(-0.13544938430e9, -0.33153754512e8,
+ -0.19406631946e7, -0.48058318783e5,
+ -0.63269783360e3, -0.49520779070e1,
+ -0.24970910370e-1, -0.74741159550e-4,
+ -0.18257612460e-6, x);
+ double y = x - 225.; // reflection around 15 (squared)
+ double den = Poly4(-0.34598737196e8, 0.23852643181e6,
+ -0.70699387620e3, 0.10000000000e1, y);
+ return num / den;
+
+#if IO_EXTENDED_BETA
+ /* Table 42 for x > 15; e < 10^(-8.11).
+ * This is used for Beta>15, but is disabled here as
+ * we never use Beta that high.
+ *
+ * NOTE: This should be enabled only for x > 15.
+ */
+
+ double y = 1./x;
+ double z = y - (1./15);
+ double num = Poly2(0.415079861746e1, -0.5149092496e1, z);
+ double den = Poly3(0.103150763823e2, -0.14181687413e2,
+ 0.1000000000e1, z);
+ return exp(x) * sqrt(y) * num / den;
+#endif
+ } else {
+ /*
+ * NOT USED, but reference for large Beta.
+ *
+ * Abramowitz and Stegun asymptotic formula.
+ * works for x > 3.75.
+ */
+ double y = 1./x;
+ return exp(x) * sqrt(y) *
+ // note: reciprocal squareroot may be easier!
+ // http://en.wikipedia.org/wiki/Fast_inverse_square_root
+ Poly9(I0ATerm<0>::value, I0ATerm<1>::value,
+ I0ATerm<2>::value, I0ATerm<3>::value,
+ I0ATerm<4>::value, I0ATerm<5>::value,
+ I0ATerm<6>::value, I0ATerm<7>::value,
+ I0ATerm<8>::value, y); // (... e) < 1.9e-7
+ }
+}
+
+/*
+ * calculates the transition bandwidth for a Kaiser filter
+ *
+ * Formula 3.2.8, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+ * Formula 7.76, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
+ *
+ * @param halfNumCoef is half the number of coefficients per filter phase.
+ *
+ * @param stopBandAtten is the stop band attenuation desired.
+ *
+ * @return the transition bandwidth in normalized frequency (0 <= f <= 0.5)
+ */
+static inline double firKaiserTbw(int halfNumCoef, double stopBandAtten) {
+ return (stopBandAtten - 7.95)/((2.*14.36)*halfNumCoef);
+}
+
+/*
+ * calculates the fir transfer response of the overall polyphase filter at w.
+ *
+ * Calculates the DTFT transfer coefficient H(w) for 0 <= w <= PI, utilizing the
+ * fact that h[n] is symmetric (cosines only, no complex arithmetic).
+ *
+ * We use Goertzel's algorithm to accelerate the computation to essentially
+ * a single multiply and 2 adds per filter coefficient h[].
+ *
+ * Be careful be careful to consider that h[n] is the overall polyphase filter,
+ * with L phases, so rescaling H(w)/L is probably what you expect for "unity gain",
+ * as you only use one of the polyphases at a time.
+ */
+template <typename T>
+static inline double firTransfer(const T* coef, int L, int halfNumCoef, double w) {
+ double accum = static_cast<double>(coef[0])*0.5; // "center coefficient" from first bank
+ coef += halfNumCoef; // skip first filterbank (picked up by the last filterbank).
+#if SLOW_FIRTRANSFER
+ /* Original code for reference. This is equivalent to the code below, but slower. */
+ for (int i=1 ; i<=L ; ++i) {
+ for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
+ accum += cos(ix*w)*static_cast<double>(*coef++);
+ }
+ }
+#else
+ /*
+ * Our overall filter is stored striped by polyphases, not a contiguous h[n].
+ * We could fetch coefficients in a non-contiguous fashion
+ * but that will not scale to vector processing.
+ *
+ * We apply Goertzel's algorithm directly to each polyphase filter bank instead of
+ * using cosine generation/multiplication, thereby saving one multiply per inner loop.
+ *
+ * See: http://en.wikipedia.org/wiki/Goertzel_algorithm
+ * Also: Oppenheim and Schafer, _Discrete Time Signal Processing, 3e_, p. 720.
+ *
+ * We use the basic recursion to incorporate the cosine steps into real sequence x[n]:
+ * s[n] = x[n] + (2cosw)*s[n-1] + s[n-2]
+ *
+ * y[n] = s[n] - e^(iw)s[n-1]
+ * = sum_{k=-\infty}^{n} x[k]e^(-iw(n-k))
+ * = e^(-iwn) sum_{k=0}^{n} x[k]e^(iwk)
+ *
+ * The summation contains the frequency steps we want multiplied by the source
+ * (similar to a DTFT).
+ *
+ * Using symmetry, and just the real part (be careful, this must happen
+ * after any internal complex multiplications), the polyphase filterbank
+ * transfer function is:
+ *
+ * Hpp[n, w, w_0] = sum_{k=0}^{n} x[k] * cos(wk + w_0)
+ * = Re{ e^(iwn + iw_0) y[n]}
+ * = cos(wn+w_0) * s[n] - cos(w(n+1)+w_0) * s[n-1]
+ *
+ * using the fact that s[n] of real x[n] is real.
+ *
+ */
+ double dcos = 2. * cos(L*w);
+ int start = ((halfNumCoef)*L + 1);
+ SineGen cc((start - L) * w, w, true); // cosine
+ SineGen cp(start * w, w, true); // cosine
+ for (int i=1 ; i<=L ; ++i) {
+ double sc = 0;
+ double sp = 0;
+ for (int j=0 ; j<halfNumCoef ; ++j) {
+ double tmp = sc;
+ sc = static_cast<double>(*coef++) + dcos*sc - sp;
+ sp = tmp;
+ }
+ // If we are awfully clever, we can apply Goertzel's algorithm
+ // again on the sc and sp sequences returned here.
+ accum += cc.valueAdvance() * sc - cp.valueAdvance() * sp;
+ }
+#endif
+ return accum*2.;
+}
+
+/*
+ * evaluates the minimum and maximum |H(f)| bound in a band region.
+ *
+ * This is usually done with equally spaced increments in the target band in question.
+ * The passband is often very small, and sampled that way. The stopband is often much
+ * larger.
+ *
+ * We use the fact that the overall polyphase filter has an additional bank at the end
+ * for interpolation; hence it is overspecified for the H(f) computation. Thus the
+ * first polyphase is never actually checked, excepting its first term.
+ *
+ * In this code we use the firTransfer() evaluator above, which uses Goertzel's
+ * algorithm to calculate the transfer function at each point.
+ *
+ * TODO: An alternative with equal spacing is the FFT/DFT. An alternative with unequal
+ * spacing is a chirp transform.
+ *
+ * @param coef is the designed polyphase filter banks
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param fstart is the normalized frequency start.
+ *
+ * @param fend is the normalized frequency end.
+ *
+ * @param steps is the number of steps to take (sampling) between frequency start and end
+ *
+ * @param firMin returns the minimum transfer |H(f)| found
+ *
+ * @param firMax returns the maximum transfer |H(f)| found
+ *
+ * 0 <= f <= 0.5.
+ * This is used to test passband and stopband performance.
+ */
+template <typename T>
+static void testFir(const T* coef, int L, int halfNumCoef,
+ double fstart, double fend, int steps, double &firMin, double &firMax) {
+ double wstart = fstart*(2.*M_PI);
+ double wend = fend*(2.*M_PI);
+ double wstep = (wend - wstart)/steps;
+ double fmax, fmin;
+ double trf = firTransfer(coef, L, halfNumCoef, wstart);
+ if (trf<0) {
+ trf = -trf;
+ }
+ fmin = fmax = trf;
+ wstart += wstep;
+ for (int i=1; i<steps; ++i) {
+ trf = firTransfer(coef, L, halfNumCoef, wstart);
+ if (trf<0) {
+ trf = -trf;
+ }
+ if (trf>fmax) {
+ fmax = trf;
+ }
+ else if (trf<fmin) {
+ fmin = trf;
+ }
+ wstart += wstep;
+ }
+ // renormalize - this is only needed for integer filter types
+ double norm = 1./((1ULL<<(sizeof(T)*8-1))*L);
+
+ firMin = fmin * norm;
+ firMax = fmax * norm;
+}
+
+/*
+ * evaluates the |H(f)| lowpass band characteristics.
+ *
+ * This function tests the lowpass characteristics for the overall polyphase filter,
+ * and is used to verify the design. For this case, fp should be set to the
+ * passband normalized frequency from 0 to 0.5 for the overall filter (thus it
+ * is the designed polyphase bank value / L). Likewise for fs.
+ *
+ * @param coef is the designed polyphase filter banks
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param fp is the passband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param fs is the stopband normalized frequency, 0 < fp < fs < 0.5.
+ *
+ * @param passSteps is the number of passband sampling steps.
+ *
+ * @param stopSteps is the number of stopband sampling steps.
+ *
+ * @param passMin is the minimum value in the passband
+ *
+ * @param passMax is the maximum value in the passband (useful for scaling). This should
+ * be less than 1., to avoid sine wave test overflow.
+ *
+ * @param passRipple is the passband ripple. Typically this should be less than 0.1 for
+ * an audio filter. Generally speaker/headphone device characteristics will dominate
+ * the passband term.
+ *
+ * @param stopMax is the maximum value in the stopband.
+ *
+ * @param stopRipple is the stopband ripple, also known as stopband attenuation.
+ * Typically this should be greater than ~80dB for low quality, and greater than
+ * ~100dB for full 16b quality, otherwise aliasing may become noticeable.
+ *
+ */
+template <typename T>
+static void testFir(const T* coef, int L, int halfNumCoef,
+ double fp, double fs, int passSteps, int stopSteps,
+ double &passMin, double &passMax, double &passRipple,
+ double &stopMax, double &stopRipple) {
+ double fmin, fmax;
+ testFir(coef, L, halfNumCoef, 0., fp, passSteps, fmin, fmax);
+ double d1 = (fmax - fmin)/2.;
+ passMin = fmin;
+ passMax = fmax;
+ passRipple = -20.*log10(1. - d1); // passband ripple
+ testFir(coef, L, halfNumCoef, fs, 0.5, stopSteps, fmin, fmax);
+ // fmin is really not important for the stopband.
+ stopMax = fmax;
+ stopRipple = -20.*log10(fmax); // stopband ripple/attenuation
+}
+
+/*
+ * Calculates the overall polyphase filter based on a windowed sinc function.
+ *
+ * The windowed sinc is an odd length symmetric filter of exactly L*halfNumCoef*2+1
+ * taps for the entire kernel. This is then decomposed into L+1 polyphase filterbanks.
+ * The last filterbank is used for interpolation purposes (and is mostly composed
+ * of the first bank shifted by one sample), and is unnecessary if one does
+ * not do interpolation.
+ *
+ * We use the last filterbank for some transfer function calculation purposes,
+ * so it needs to be generated anyways.
+ *
+ * @param coef is the caller allocated space for coefficients. This should be
+ * exactly (L+1)*halfNumCoef in size.
+ *
+ * @param L is the number of phases (for interpolation)
+ *
+ * @param halfNumCoef should be half the number of coefficients for a single
+ * polyphase.
+ *
+ * @param stopBandAtten is the stopband value, should be >50dB.
+ *
+ * @param fcr is cutoff frequency/sampling rate (<0.5). At this point, the energy
+ * should be 6dB less. (fcr is where the amplitude drops by half). Use the
+ * firKaiserTbw() to calculate the transition bandwidth. fcr is the midpoint
+ * between the stop band and the pass band (fstop+fpass)/2.
+ *
+ * @param atten is the attenuation (generally slightly less than 1).
+ */
+
+template <typename T>
+static inline void firKaiserGen(T* coef, int L, int halfNumCoef,
+ double stopBandAtten, double fcr, double atten) {
+ //
+ // Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+ // Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
+ //
+ // See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf
+ //
+ // Kaiser window and beta parameter
+ //
+ // | 0.1102*(A - 8.7) A > 50
+ // beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50
+ // | 0. A < 21
+ //
+ // with A is the desired stop-band attenuation in dBFS
+ //
+ // 30 dB 2.210
+ // 40 dB 3.384
+ // 50 dB 4.538
+ // 60 dB 5.658
+ // 70 dB 6.764
+ // 80 dB 7.865
+ // 90 dB 8.960
+ // 100 dB 10.056
+
+ const int N = L * halfNumCoef; // non-negative half
+ const double beta = 0.1102 * (stopBandAtten - 8.7); // >= 50dB always
+ const double xstep = (2. * M_PI) * fcr / L;
+ const double xfrac = 1. / N;
+ const double yscale = atten * L / (I0(beta) * M_PI);
+
+ // We use sine generators, which computes sines on regular step intervals.
+ // This speeds up overall computation about 40% from computing the sine directly.
+
+ SineGenGen sgg(0., xstep, L*xstep); // generates sine generators (one per polyphase)
+
+ for (int i=0 ; i<=L ; ++i) { // generate an extra set of coefs for interpolation
+
+ // computation for a single polyphase of the overall filter.
+ SineGen sg = sgg.valueAdvance(); // current sine generator for "j" inner loop.
+ double err = 0; // for noise shaping on int16_t coefficients (over each polyphase)
+
+ for (int j=0, ix=i ; j<halfNumCoef ; ++j, ix+=L) {
+ double y;
+ if (CC_LIKELY(ix)) {
+ double x = static_cast<double>(ix);
+
+ // sine generator: sg.valueAdvance() returns sin(ix*xstep);
+ y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x;
+ } else {
+ y = 2. * atten * fcr; // center of filter, sinc(0) = 1.
+ sg.advance();
+ }
+
+ // (caution!) float version does not need rounding
+ if (is_same<T, int16_t>::value) { // int16_t needs noise shaping
+ *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1), err));
+ } else {
+ *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1)));
+ }
+ }
+ }
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_GEN_H*/
diff --git a/services/audioflinger/AudioResamplerFirOps.h b/services/audioflinger/AudioResamplerFirOps.h
new file mode 100644
index 0000000..bf2163f
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirOps.h
@@ -0,0 +1,163 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_OPS_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_OPS_H
+
+namespace android {
+
+#if defined(__arm__) && !defined(__thumb__)
+#define USE_INLINE_ASSEMBLY (true)
+#else
+#define USE_INLINE_ASSEMBLY (false)
+#endif
+
+#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
+#define USE_NEON (true)
+#include <arm_neon.h>
+#else
+#define USE_NEON (false)
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+ static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T> // partial specialization
+{
+ static const bool value = true;
+};
+
+static inline
+int32_t mulRL(int left, int32_t in, uint32_t vRL)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ if (left) {
+ asm( "smultb %[out], %[in], %[vRL] \n"
+ : [out]"=r"(out)
+ : [in]"%r"(in), [vRL]"r"(vRL)
+ : );
+ } else {
+ asm( "smultt %[out], %[in], %[vRL] \n"
+ : [out]"=r"(out)
+ : [in]"%r"(in), [vRL]"r"(vRL)
+ : );
+ }
+ return out;
+#else
+ int16_t v = left ? static_cast<int16_t>(vRL) : static_cast<int16_t>(vRL>>16);
+ return static_cast<int32_t>((static_cast<int64_t>(in) * v) >> 16);
+#endif
+}
+
+static inline
+int32_t mulAdd(int16_t in, int16_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ asm( "smlabb %[out], %[v], %[in], %[a] \n"
+ : [out]"=r"(out)
+ : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+ : );
+ return out;
+#else
+ return a + v * in;
+#endif
+}
+
+static inline
+int32_t mulAdd(int16_t in, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ asm( "smlawb %[out], %[v], %[in], %[a] \n"
+ : [out]"=r"(out)
+ : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+ : );
+ return out;
+#else
+ return a + static_cast<int32_t>((static_cast<int64_t>(v) * in) >> 16);
+#endif
+}
+
+static inline
+int32_t mulAdd(int32_t in, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ asm( "smmla %[out], %[v], %[in], %[a] \n"
+ : [out]"=r"(out)
+ : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
+ : );
+ return out;
+#else
+ return a + static_cast<int32_t>((static_cast<int64_t>(v) * in) >> 32);
+#endif
+}
+
+static inline
+int32_t mulAddRL(int left, uint32_t inRL, int16_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ if (left) {
+ asm( "smlabb %[out], %[v], %[inRL], %[a] \n"
+ : [out]"=r"(out)
+ : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+ : );
+ } else {
+ asm( "smlabt %[out], %[v], %[inRL], %[a] \n"
+ : [out]"=r"(out)
+ : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+ : );
+ }
+ return out;
+#else
+ int16_t s = left ? static_cast<int16_t>(inRL) : static_cast<int16_t>(inRL>>16);
+ return a + v * s;
+#endif
+}
+
+static inline
+int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a)
+{
+#if USE_INLINE_ASSEMBLY
+ int32_t out;
+ if (left) {
+ asm( "smlawb %[out], %[v], %[inRL], %[a] \n"
+ : [out]"=r"(out)
+ : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+ : );
+ } else {
+ asm( "smlawt %[out], %[v], %[inRL], %[a] \n"
+ : [out]"=r"(out)
+ : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a)
+ : );
+ }
+ return out;
+#else
+ int16_t s = left ? static_cast<int16_t>(inRL) : static_cast<int16_t>(inRL>>16);
+ return a + static_cast<int32_t>((static_cast<int64_t>(v) * s) >> 16);
+#endif
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_OPS_H*/
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
new file mode 100644
index 0000000..38e387c
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -0,0 +1,256 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H
+
+namespace android {
+
+// depends on AudioResamplerFirOps.h
+
+template<int CHANNELS, typename TC>
+static inline
+void mac(
+ int32_t& l, int32_t& r,
+ const TC coef,
+ const int16_t* samples)
+{
+ if (CHANNELS == 2) {
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+ l = mulAddRL(1, rl, coef, l);
+ r = mulAddRL(0, rl, coef, r);
+ } else {
+ r = l = mulAdd(samples[0], coef, l);
+ }
+}
+
+template<int CHANNELS, typename TC>
+static inline
+void interpolate(
+ int32_t& l, int32_t& r,
+ const TC coef_0, const TC coef_1,
+ const int16_t lerp, const int16_t* samples)
+{
+ TC sinc;
+
+ if (is_same<TC, int16_t>::value) {
+ sinc = (lerp * ((coef_1-coef_0)<<1)>>16) + coef_0;
+ } else {
+ sinc = mulAdd(lerp, (coef_1-coef_0)<<1, coef_0);
+ }
+ if (CHANNELS == 2) {
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+ l = mulAddRL(1, rl, sinc, l);
+ r = mulAddRL(0, rl, sinc, r);
+ } else {
+ r = l = mulAdd(samples[0], sinc, l);
+ }
+}
+
+/*
+ * Calculates a single output sample (two stereo frames).
+ *
+ * This function computes both the positive half FIR dot product and
+ * the negative half FIR dot product, accumulates, and then applies the volume.
+ *
+ * This is a locked phase filter (it does not compute the interpolation).
+ *
+ * Use fir() to compute the proper coefficient pointers for a polyphase
+ * filter bank.
+ */
+
+template <int CHANNELS, int STRIDE, typename TC>
+static inline
+void ProcessL(int32_t* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ int32_t l = 0;
+ int32_t r = 0;
+ do {
+ mac<CHANNELS>(l, r, *coefsP++, sP);
+ sP -= CHANNELS;
+ mac<CHANNELS>(l, r, *coefsN++, sN);
+ sN += CHANNELS;
+ } while (--count > 0);
+ out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+ out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) interpolating phase.
+ *
+ * This function computes both the positive half FIR dot product and
+ * the negative half FIR dot product, accumulates, and then applies the volume.
+ *
+ * This is an interpolated phase filter.
+ *
+ * Use fir() to compute the proper coefficient pointers for a polyphase
+ * filter bank.
+ */
+
+template <int CHANNELS, int STRIDE, typename TC>
+static inline
+void Process(int32_t* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const TC* coefsP1,
+ const TC* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ (void) coefsP1; // suppress unused parameter warning
+ (void) coefsN1;
+ if (sizeof(*coefsP)==4) {
+ lerpP >>= 16; // ensure lerpP is 16b
+ }
+ int32_t l = 0;
+ int32_t r = 0;
+ for (size_t i = 0; i < count; ++i) {
+ interpolate<CHANNELS>(l, r, coefsP[0], coefsP[count], lerpP, sP);
+ coefsP++;
+ sP -= CHANNELS;
+ interpolate<CHANNELS>(l, r, coefsN[count], coefsN[0], lerpP, sN);
+ coefsN++;
+ sN += CHANNELS;
+ }
+ out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+ out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) from input sample pointer.
+ *
+ * This sets up the params for the accelerated Process() and ProcessL()
+ * functions to do the appropriate dot products.
+ *
+ * @param out should point to the output buffer with at least enough space for 2 output frames.
+ *
+ * @param phase is the fractional distance between input samples for interpolation:
+ * phase >= 0 && phase < phaseWrapLimit. It can be thought of as a rational fraction
+ * of phase/phaseWrapLimit.
+ *
+ * @param phaseWrapLimit is #polyphases<<coefShift, where #polyphases is the number of polyphases
+ * in the polyphase filter. Likewise, #polyphases can be obtained as (phaseWrapLimit>>coefShift).
+ *
+ * @param coefShift gives the bit alignment of the polyphase index in the phase parameter.
+ *
+ * @param halfNumCoefs is the half the number of coefficients per polyphase filter. Since the
+ * overall filterbank is odd-length symmetric, only halfNumCoefs need be stored.
+ *
+ * @param coefs is the polyphase filter bank, starting at from polyphase index 0, and ranging to
+ * and including the #polyphases. Each polyphase of the filter has half-length halfNumCoefs
+ * (due to symmetry). The total size of the filter bank in coefficients is
+ * (#polyphases+1)*halfNumCoefs.
+ *
+ * The filter bank coefs should be aligned to a minimum of 16 bytes (preferrably to cache line).
+ *
+ * The coefs should be attenuated (to compensate for passband ripple)
+ * if storing back into the native format.
+ *
+ * @param samples are unaligned input samples. The position is in the "middle" of the
+ * sample array with respect to the FIR filter:
+ * the negative half of the filter is dot product from samples+1 to samples+halfNumCoefs;
+ * the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
+ *
+ * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
+ * expressed as a S32 integer. A negative value inverts the channel 180 degrees.
+ * The pointer volumeLR should be aligned to a minimum of 8 bytes.
+ * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
+ *
+ * In between calls to filterCoefficient, the phase is incremented by phaseIncrement, where
+ * phaseIncrement is calculated as inputSampling * phaseWrapLimit / outputSampling.
+ *
+ * The filter polyphase index is given by indexP = phase >> coefShift. Due to
+ * odd length symmetric filter, the polyphase index of the negative half depends on
+ * whether interpolation is used.
+ *
+ * The fractional siting between the polyphase indices is given by the bits below coefShift:
+ *
+ * lerpP = phase << 32 - coefShift >> 1; // for 32 bit unsigned phase multiply
+ * lerpP = phase << 32 - coefShift >> 17; // for 16 bit unsigned phase multiply
+ *
+ * For integer types, this is expressed as:
+ *
+ * lerpP = phase << sizeof(phase)*8 - coefShift
+ * >> (sizeof(phase)-sizeof(*coefs))*8 + 1;
+ *
+ */
+
+template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+static inline
+void fir(int32_t* const out,
+ const uint32_t phase, const uint32_t phaseWrapLimit,
+ const int coefShift, const int halfNumCoefs, const TC* const coefs,
+ const int16_t* const samples, const int32_t* const volumeLR)
+{
+ // NOTE: be very careful when modifying the code here. register
+ // pressure is very high and a small change might cause the compiler
+ // to generate far less efficient code.
+ // Always sanity check the result with objdump or test-resample.
+
+ if (LOCKED) {
+ // locked polyphase (no interpolation)
+ // Compute the polyphase filter index on the positive and negative side.
+ uint32_t indexP = phase >> coefShift;
+ uint32_t indexN = (phaseWrapLimit - phase) >> coefShift;
+ const TC* coefsP = coefs + indexP*halfNumCoefs;
+ const TC* coefsN = coefs + indexN*halfNumCoefs;
+ const int16_t* sP = samples;
+ const int16_t* sN = samples + CHANNELS;
+
+ // dot product filter.
+ ProcessL<CHANNELS, STRIDE>(out,
+ halfNumCoefs, coefsP, coefsN, sP, sN, volumeLR);
+ } else {
+ // interpolated polyphase
+ // Compute the polyphase filter index on the positive and negative side.
+ uint32_t indexP = phase >> coefShift;
+ uint32_t indexN = (phaseWrapLimit - phase - 1) >> coefShift; // one's complement.
+ const TC* coefsP = coefs + indexP*halfNumCoefs;
+ const TC* coefsN = coefs + indexN*halfNumCoefs;
+ const TC* coefsP1 = coefsP + halfNumCoefs;
+ const TC* coefsN1 = coefsN + halfNumCoefs;
+ const int16_t* sP = samples;
+ const int16_t* sN = samples + CHANNELS;
+
+ // Interpolation fraction lerpP derived by shifting all the way up and down
+ // to clear the appropriate bits and align to the appropriate level
+ // for the integer multiply. The constants should resolve in compile time.
+ //
+ // The interpolated filter coefficient is derived as follows for the pos/neg half:
+ //
+ // interpolated[P] = index[P]*lerpP + index[P+1]*(1-lerpP)
+ // interpolated[N] = index[N+1]*lerpP + index[N]*(1-lerpP)
+ uint32_t lerpP = phase << (sizeof(phase)*8 - coefShift)
+ >> ((sizeof(phase)-sizeof(*coefs))*8 + 1);
+
+ // on-the-fly interpolated dot product filter
+ Process<CHANNELS, STRIDE>(out,
+ halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR);
+ }
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H*/
diff --git a/services/audioflinger/AudioResamplerFirProcessNeon.h b/services/audioflinger/AudioResamplerFirProcessNeon.h
new file mode 100644
index 0000000..f311cef
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirProcessNeon.h
@@ -0,0 +1,1149 @@
+/*
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H
+#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H
+
+namespace android {
+
+// depends on AudioResamplerFirOps.h, AudioResamplerFirProcess.h
+
+#if USE_NEON
+//
+// NEON specializations are enabled for Process() and ProcessL()
+//
+// TODO: Stride 16 and Stride 8 can be combined with one pass stride 8 (if necessary)
+// and looping stride 16 (or vice versa). This has some polyphase coef data alignment
+// issues with S16 coefs. Consider this later.
+
+// Macros to save a mono/stereo accumulator sample in q0 (and q4) as stereo out.
+#define ASSEMBLY_ACCUMULATE_MONO \
+ "vld1.s32 {d2}, [%[vLR]:64] \n"/* (1) load volumes */\
+ "vld1.s32 {d3}, %[out] \n"/* (2) unaligned load the output */\
+ "vpadd.s32 d0, d0, d1 \n"/* (1) add all 4 partial sums */\
+ "vpadd.s32 d0, d0, d0 \n"/* (1+4d) and replicate L/R */\
+ "vqrdmulh.s32 d0, d0, d2 \n"/* (2+3d) apply volume */\
+ "vqadd.s32 d3, d3, d0 \n"/* (1+4d) accumulate result (saturating) */\
+ "vst1.s32 {d3}, %[out] \n"/* (2+2d) store result */
+
+#define ASSEMBLY_ACCUMULATE_STEREO \
+ "vld1.s32 {d2}, [%[vLR]:64] \n"/* (1) load volumes*/\
+ "vld1.s32 {d3}, %[out] \n"/* (2) unaligned load the output*/\
+ "vpadd.s32 d0, d0, d1 \n"/* (1) add all 4 partial sums from q0*/\
+ "vpadd.s32 d8, d8, d9 \n"/* (1) add all 4 partial sums from q4*/\
+ "vpadd.s32 d0, d0, d8 \n"/* (1+4d) combine into L/R*/\
+ "vqrdmulh.s32 d0, d0, d2 \n"/* (2+3d) apply volume*/\
+ "vqadd.s32 d3, d3, d0 \n"/* (1+4d) accumulate result (saturating)*/\
+ "vst1.s32 {d3}, %[out] \n"/* (2+2d)store result*/
+
+template <>
+inline void ProcessL<1, 16>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
+
+ "1: \n"
+
+ "vld1.16 {q2}, [%[sP]] \n"// (2+0d) load 8 16-bits mono samples
+ "vld1.16 {q3}, [%[sN]]! \n"// (2) load 8 16-bits mono samples
+ "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+ // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+ "vmlal.s16 q0, d4, d17 \n"// (1+0d) multiply (reversed)samples by coef
+ "vmlal.s16 q0, d5, d16 \n"// (1) multiply (reversed)samples by coef
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
+ "vmlal.s16 q0, d7, d21 \n"// (1) multiply neg samples
+
+ // moving these ARM instructions before neon above seems to be slower
+ "subs %[count], %[count], #8 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #16 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q10"
+ );
+}
+
+template <>
+inline void ProcessL<2, 16>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// (1) acc_L = 0
+ "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
+
+ "1: \n"
+
+ "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
+ "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
+ "vrev64.16 q3, q3 \n"// (0 combines+) reverse right positive
+
+ "vmlal.s16 q0, d4, d17 \n"// (1) multiply (reversed) samples left
+ "vmlal.s16 q0, d5, d16 \n"// (1) multiply (reversed) samples left
+ "vmlal.s16 q4, d6, d17 \n"// (1) multiply (reversed) samples right
+ "vmlal.s16 q4, d7, d16 \n"// (1) multiply (reversed) samples right
+ "vmlal.s16 q0, d10, d20 \n"// (1) multiply samples left
+ "vmlal.s16 q0, d11, d21 \n"// (1) multiply samples left
+ "vmlal.s16 q4, d12, d20 \n"// (1) multiply samples right
+ "vmlal.s16 q4, d13, d21 \n"// (1) multiply samples right
+
+ // moving these ARM before neon seems to be slower
+ "subs %[count], %[count], #8 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #32 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q10"
+ );
+}
+
+template <>
+inline void Process<1, 16>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* coefsP1,
+ const int16_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase S32 Q15
+ "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
+
+ "1: \n"
+
+ "vld1.16 {q2}, [%[sP]] \n"// (2+0d) load 8 16-bits mono samples
+ "vld1.16 {q3}, [%[sN]]! \n"// (2) load 8 16-bits mono samples
+ "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q9}, [%[coefsP1]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+ "vld1.16 {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q11}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+
+ "vsub.s16 q9, q9, q8 \n"// (1) interpolate (step1) 1st set of coefs
+ "vsub.s16 q11, q11, q10 \n"// (1) interpolate (step1) 2nd set of coets
+
+ "vqrdmulh.s16 q9, q9, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
+ "vqrdmulh.s16 q11, q11, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+ "vadd.s16 q8, q8, q9 \n"// (1+2d) interpolate (step3) 1st set
+ "vadd.s16 q10, q10, q11 \n"// (1+1d) interpolate (step3) 2nd set
+
+ // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+ "vmlal.s16 q0, d4, d17 \n"// (1+0d) multiply reversed samples by coef
+ "vmlal.s16 q0, d5, d16 \n"// (1) multiply reversed samples by coef
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
+ "vmlal.s16 q0, d7, d21 \n"// (1) multiply neg samples
+
+ // moving these ARM instructions before neon above seems to be slower
+ "subs %[count], %[count], #8 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #16 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11"
+ );
+}
+
+template <>
+inline void Process<2, 16>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* coefsP1,
+ const int16_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// (1) acc_L = 0
+ "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
+
+ "1: \n"
+
+ "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
+ "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q9}, [%[coefsP1]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+ "vld1.16 {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {q11}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs for interpolation
+
+ "vsub.s16 q9, q9, q8 \n"// (1) interpolate (step1) 1st set of coefs
+ "vsub.s16 q11, q11, q10 \n"// (1) interpolate (step1) 2nd set of coets
+
+ "vqrdmulh.s16 q9, q9, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
+ "vqrdmulh.s16 q11, q11, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
+ "vrev64.16 q3, q3 \n"// (1) reverse 8 frames of the right positive
+
+ "vadd.s16 q8, q8, q9 \n"// (1+1d) interpolate (step3) 1st set
+ "vadd.s16 q10, q10, q11 \n"// (1+1d) interpolate (step3) 2nd set
+
+ "vmlal.s16 q0, d4, d17 \n"// (1) multiply reversed samples left
+ "vmlal.s16 q0, d5, d16 \n"// (1) multiply reversed samples left
+ "vmlal.s16 q4, d6, d17 \n"// (1) multiply reversed samples right
+ "vmlal.s16 q4, d7, d16 \n"// (1) multiply reversed samples right
+ "vmlal.s16 q0, d10, d20 \n"// (1) multiply samples left
+ "vmlal.s16 q0, d11, d21 \n"// (1) multiply samples left
+ "vmlal.s16 q4, d12, d20 \n"// (1) multiply samples right
+ "vmlal.s16 q4, d13, d21 \n"// (1) multiply samples right
+
+ // moving these ARM before neon seems to be slower
+ "subs %[count], %[count], #8 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #32 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q9", "q10", "q11"
+ );
+}
+
+template <>
+inline void ProcessL<1, 16>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld1.16 {q2}, [%[sP]] \n"// load 8 16-bits mono samples
+ "vld1.16 {q3}, [%[sN]]! \n"// load 8 16-bits mono samples
+ "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+ "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q0, q0, q15 \n"// accumulate result
+ "vadd.s32 q0, q0, q13 \n"// accumulate result
+
+ "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
+ "subs %[count], %[count], #8 \n"// update loop counter
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+template <>
+inline void ProcessL<2, 16>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// result, initialize to 0
+ "veor q4, q4, q4 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples
+ "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+ "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d10, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d11, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result
+ "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result
+
+ "vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d12, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d13, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q4, q4, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result
+ "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result
+
+ "subs %[count], %[count], #8 \n"// update loop counter
+ "sub %[sP], %[sP], #32 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+template <>
+inline void Process<1, 16>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int32_t* coefsP1,
+ const int32_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld1.16 {q2}, [%[sP]] \n"// load 8 16-bits mono samples
+ "vld1.16 {q3}, [%[sN]]! \n"// load 8 16-bits mono samples
+ "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q14, q15}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+ "vsub.s32 q12, q12, q8 \n"// interpolate (step1)
+ "vsub.s32 q13, q13, q9 \n"// interpolate (step1)
+ "vsub.s32 q14, q14, q10 \n"// interpolate (step1)
+ "vsub.s32 q15, q15, q11 \n"// interpolate (step1)
+
+ "vqrdmulh.s32 q12, q12, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q13, q13, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q14, q14, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q15, q15, d2[0] \n"// interpolate (step2)
+
+ "vadd.s32 q8, q8, q12 \n"// interpolate (step3)
+ "vadd.s32 q9, q9, q13 \n"// interpolate (step3)
+ "vadd.s32 q10, q10, q14 \n"// interpolate (step3)
+ "vadd.s32 q11, q11, q15 \n"// interpolate (step3)
+
+ "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q0, q0, q15 \n"// accumulate result
+ "vadd.s32 q0, q0, q13 \n"// accumulate result
+
+ "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
+ "subs %[count], %[count], #8 \n"// update loop counter
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+template <>
+inline void Process<2, 16>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int32_t* coefsP1,
+ const int32_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 16;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// result, initialize to 0
+ "veor q4, q4, q4 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples
+ "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q14, q15}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
+
+ "vsub.s32 q12, q12, q8 \n"// interpolate (step1)
+ "vsub.s32 q13, q13, q9 \n"// interpolate (step1)
+ "vsub.s32 q14, q14, q10 \n"// interpolate (step1)
+ "vsub.s32 q15, q15, q11 \n"// interpolate (step1)
+
+ "vqrdmulh.s32 q12, q12, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q13, q13, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q14, q14, d2[0] \n"// interpolate (step2)
+ "vqrdmulh.s32 q15, q15, d2[0] \n"// interpolate (step2)
+
+ "vadd.s32 q8, q8, q12 \n"// interpolate (step3)
+ "vadd.s32 q9, q9, q13 \n"// interpolate (step3)
+ "vadd.s32 q10, q10, q14 \n"// interpolate (step3)
+ "vadd.s32 q11, q11, q15 \n"// interpolate (step3)
+
+ "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d10, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d11, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result
+ "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result
+
+ "vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d12, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d13, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q4, q4, q12 \n"// accumulate result
+ "vadd.s32 q13, q13, q14 \n"// accumulate result
+ "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result
+ "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result
+
+ "subs %[count], %[count], #8 \n"// update loop counter
+ "sub %[sP], %[sP], #32 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+template <>
+inline void ProcessL<1, 8>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
+
+ "1: \n"
+
+ "vld1.16 {d4}, [%[sP]] \n"// (2+0d) load 4 16-bits mono samples
+ "vld1.16 {d6}, [%[sN]]! \n"// (2) load 4 16-bits mono samples
+ "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 4 16-bits coefs
+ "vld1.16 {d20}, [%[coefsN0]:64]! \n"// (1) load 4 16-bits coefs
+
+ "vrev64.16 d4, d4 \n"// (1) reversed s3, s2, s1, s0, s7, s6, s5, s4
+
+ // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+ "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed)samples by coef
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
+
+ // moving these ARM instructions before neon above seems to be slower
+ "subs %[count], %[count], #4 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #8 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q10"
+ );
+}
+
+template <>
+inline void ProcessL<2, 8>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// (1) acc_L = 0
+ "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
+
+ "1: \n"
+
+ "vld2.16 {d4, d5}, [%[sP]] \n"// (2+0d) load 8 16-bits stereo samples
+ "vld2.16 {d6, d7}, [%[sN]]! \n"// (2) load 8 16-bits stereo samples
+ "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {d20}, [%[coefsN0]:64]! \n"// (1) load 8 16-bits coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
+
+ "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed) samples left
+ "vmlal.s16 q4, d5, d16 \n"// (1) multiply (reversed) samples right
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply samples left
+ "vmlal.s16 q4, d7, d20 \n"// (1) multiply samples right
+
+ // moving these ARM before neon seems to be slower
+ "subs %[count], %[count], #4 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #16 \n"// (0) move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q10"
+ );
+}
+
+template <>
+inline void Process<1, 8>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* coefsP1,
+ const int16_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase S32 Q15
+ "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
+
+ "1: \n"
+
+ "vld1.16 {d4}, [%[sP]] \n"// (2+0d) load 4 16-bits mono samples
+ "vld1.16 {d6}, [%[sN]]! \n"// (2) load 4 16-bits mono samples
+ "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 4 16-bits coefs
+ "vld1.16 {d17}, [%[coefsP1]:64]! \n"// (1) load 4 16-bits coefs for interpolation
+ "vld1.16 {d20}, [%[coefsN1]:64]! \n"// (1) load 4 16-bits coefs
+ "vld1.16 {d21}, [%[coefsN0]:64]! \n"// (1) load 4 16-bits coefs for interpolation
+
+ "vsub.s16 d17, d17, d16 \n"// (1) interpolate (step1) 1st set of coefs
+ "vsub.s16 d21, d21, d20 \n"// (1) interpolate (step1) 2nd set of coets
+
+ "vqrdmulh.s16 d17, d17, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
+ "vqrdmulh.s16 d21, d21, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
+
+ "vrev64.16 d4, d4 \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
+
+ "vadd.s16 d16, d16, d17 \n"// (1+2d) interpolate (step3) 1st set
+ "vadd.s16 d20, d20, d21 \n"// (1+1d) interpolate (step3) 2nd set
+
+ // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
+ "vmlal.s16 q0, d4, d16 \n"// (1+0d) multiply (reversed)by coef
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
+
+ // moving these ARM instructions before neon above seems to be slower
+ "subs %[count], %[count], #4 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11"
+ );
+}
+
+template <>
+inline void Process<2, 8>(int32_t* const out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* coefsP1,
+ const int16_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// (1) acc_L = 0
+ "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
+
+ "1: \n"
+
+ "vld2.16 {d4, d5}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
+ "vld2.16 {d6, d7}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
+ "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {d17}, [%[coefsP1]:64]! \n"// (1) load 8 16-bits coefs for interpolation
+ "vld1.16 {d20}, [%[coefsN1]:64]! \n"// (1) load 8 16-bits coefs
+ "vld1.16 {d21}, [%[coefsN0]:64]! \n"// (1) load 8 16-bits coefs for interpolation
+
+ "vsub.s16 d17, d17, d16 \n"// (1) interpolate (step1) 1st set of coefs
+ "vsub.s16 d21, d21, d20 \n"// (1) interpolate (step1) 2nd set of coets
+
+ "vqrdmulh.s16 d17, d17, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
+ "vqrdmulh.s16 d21, d21, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
+
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
+
+ "vadd.s16 d16, d16, d17 \n"// (1+1d) interpolate (step3) 1st set
+ "vadd.s16 d20, d20, d21 \n"// (1+1d) interpolate (step3) 2nd set
+
+ "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed) samples left
+ "vmlal.s16 q4, d5, d16 \n"// (1) multiply (reversed) samples right
+ "vmlal.s16 q0, d6, d20 \n"// (1) multiply samples left
+ "vmlal.s16 q4, d7, d20 \n"// (1) multiply samples right
+
+ // moving these ARM before neon seems to be slower
+ "subs %[count], %[count], #4 \n"// (1) update loop counter
+ "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
+
+ // sP used after branch (warning)
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q4", "q5", "q6",
+ "q8", "q9", "q10", "q11"
+ );
+}
+
+template <>
+inline void ProcessL<1, 8>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld1.16 {d4}, [%[sP]] \n"// load 4 16-bits mono samples
+ "vld1.16 {d6}, [%[sN]]! \n"// load 4 16-bits mono samples
+ "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+ "vrev64.16 d4, d4 \n"// reverse 2 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// (stall) extend samples to 31 bits
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q0, q0, q14 \n"// (stall) accumulate result
+
+ "subs %[count], %[count], #4 \n"// update loop counter
+ "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11",
+ "q12", "q14"
+ );
+}
+
+template <>
+inline void ProcessL<2, 8>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "veor q0, q0, q0 \n"// result, initialize to 0
+ "veor q4, q4, q4 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld2.16 {d4, d5}, [%[sP]] \n"// load 4 16-bits stereo samples
+ "vld2.16 {d6, d7}, [%[sN]]! \n"// load 4 16-bits stereo samples
+ "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+
+ "vrev64.16 q2, q2 \n"// reverse 2 frames of the positive side
+
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef
+ "vqrdmulh.s32 q15, q15, q10 \n"// multiply samples by coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q4, q4, q13 \n"// accumulate result
+ "vadd.s32 q0, q0, q14 \n"// accumulate result
+ "vadd.s32 q4, q4, q15 \n"// accumulate result
+
+ "subs %[count], %[count], #4 \n"// update loop counter
+ "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsN0] "+r" (coefsN),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3", "q4",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+template <>
+inline void Process<1, 8>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int32_t* coefsP1,
+ const int32_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 1; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// result, initialize to 0
+
+ "1: \n"
+
+ "vld1.16 {d4}, [%[sP]] \n"// load 4 16-bits mono samples
+ "vld1.16 {d6}, [%[sN]]! \n"// load 4 16-bits mono samples
+ "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q9}, [%[coefsP1]:128]! \n"// load 4 32-bits coefs for interpolation
+ "vld1.32 {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
+
+ "vrev64.16 d4, d4 \n"// reverse 2 frames of the positive side
+
+ "vsub.s32 q9, q9, q8 \n"// interpolate (step1) 1st set of coefs
+ "vsub.s32 q11, q11, q10 \n"// interpolate (step1) 2nd set of coets
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q9, q9, d2[0] \n"// interpolate (step2) 1st set of coefs
+ "vqrdmulh.s32 q11, q11, d2[0] \n"// interpolate (step2) 2nd set of coefs
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+
+ "vadd.s32 q8, q8, q9 \n"// interpolate (step3) 1st set
+ "vadd.s32 q10, q10, q11 \n"// interpolate (step4) 2nd set
+
+ "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q0, q0, q14 \n"// accumulate result
+
+ "subs %[count], %[count], #4 \n"// update loop counter
+ "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_MONO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN0] "+r" (coefsN),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3",
+ "q8", "q9", "q10", "q11",
+ "q12", "q14"
+ );
+}
+
+template <>
+inline
+void Process<2, 8>(int32_t* const out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int32_t* coefsP1,
+ const int32_t* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ const int CHANNELS = 2; // template specialization does not preserve params
+ const int STRIDE = 8;
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ asm (
+ "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
+ "veor q0, q0, q0 \n"// result, initialize to 0
+ "veor q4, q4, q4 \n"// result, initialize to 0
+
+ "1: \n"
+ "vld2.16 {d4, d5}, [%[sP]] \n"// load 4 16-bits stereo samples
+ "vld2.16 {d6, d7}, [%[sN]]! \n"// load 4 16-bits stereo samples
+ "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q9}, [%[coefsP1]:128]! \n"// load 4 32-bits coefs for interpolation
+ "vld1.32 {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
+ "vld1.32 {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
+
+ "vrev64.16 q2, q2 \n"// (reversed) 2 frames of the positive side
+
+ "vsub.s32 q9, q9, q8 \n"// interpolate (step1) 1st set of coefs
+ "vsub.s32 q11, q11, q10 \n"// interpolate (step1) 2nd set of coets
+ "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
+
+ "vqrdmulh.s32 q9, q9, d2[0] \n"// interpolate (step2) 1st set of coefs
+ "vqrdmulh.s32 q11, q11, d2[1] \n"// interpolate (step3) 2nd set of coefs
+ "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
+ "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
+
+ "vadd.s32 q8, q8, q9 \n"// interpolate (step3) 1st set
+ "vadd.s32 q10, q10, q11 \n"// interpolate (step4) 2nd set
+
+ "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q15, q15, q10 \n"// multiply samples by interpolated coef
+
+ "vadd.s32 q0, q0, q12 \n"// accumulate result
+ "vadd.s32 q4, q4, q13 \n"// accumulate result
+ "vadd.s32 q0, q0, q14 \n"// accumulate result
+ "vadd.s32 q4, q4, q15 \n"// accumulate result
+
+ "subs %[count], %[count], #4 \n"// update loop counter
+ "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
+
+ "bne 1b \n"// loop
+
+ ASSEMBLY_ACCUMULATE_STEREO
+
+ : [out] "=Uv" (out[0]),
+ [count] "+r" (count),
+ [coefsP0] "+r" (coefsP),
+ [coefsP1] "+r" (coefsP1),
+ [coefsN0] "+r" (coefsN),
+ [coefsN1] "+r" (coefsN1),
+ [sP] "+r" (sP),
+ [sN] "+r" (sN)
+ : [lerpP] "r" (lerpP),
+ [vLR] "r" (volumeLR)
+ : "cc", "memory",
+ "q0", "q1", "q2", "q3", "q4",
+ "q8", "q9", "q10", "q11",
+ "q12", "q13", "q14", "q15"
+ );
+}
+
+#endif //USE_NEON
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H*/
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 010e233..29b56db 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -116,8 +116,9 @@
continue;
}
// first non destroyed handle is considered in control
- if (controlHandle == NULL)
+ if (controlHandle == NULL) {
controlHandle = h;
+ }
if (h->priority() <= priority) {
break;
}
@@ -804,7 +805,112 @@
return mOffloaded;
}
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+String8 effectFlagsToString(uint32_t flags) {
+ String8 s;
+
+ s.append("conn. mode: ");
+ switch (flags & EFFECT_FLAG_TYPE_MASK) {
+ case EFFECT_FLAG_TYPE_INSERT: s.append("insert"); break;
+ case EFFECT_FLAG_TYPE_AUXILIARY: s.append("auxiliary"); break;
+ case EFFECT_FLAG_TYPE_REPLACE: s.append("replace"); break;
+ case EFFECT_FLAG_TYPE_PRE_PROC: s.append("preproc"); break;
+ case EFFECT_FLAG_TYPE_POST_PROC: s.append("postproc"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ s.append("insert pref: ");
+ switch (flags & EFFECT_FLAG_INSERT_MASK) {
+ case EFFECT_FLAG_INSERT_ANY: s.append("any"); break;
+ case EFFECT_FLAG_INSERT_FIRST: s.append("first"); break;
+ case EFFECT_FLAG_INSERT_LAST: s.append("last"); break;
+ case EFFECT_FLAG_INSERT_EXCLUSIVE: s.append("exclusive"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ s.append("volume mgmt: ");
+ switch (flags & EFFECT_FLAG_VOLUME_MASK) {
+ case EFFECT_FLAG_VOLUME_NONE: s.append("none"); break;
+ case EFFECT_FLAG_VOLUME_CTRL: s.append("implements control"); break;
+ case EFFECT_FLAG_VOLUME_IND: s.append("requires indication"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ uint32_t devind = flags & EFFECT_FLAG_DEVICE_MASK;
+ if (devind) {
+ s.append("device indication: ");
+ switch (devind) {
+ case EFFECT_FLAG_DEVICE_IND: s.append("requires updates"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ s.append("input mode: ");
+ switch (flags & EFFECT_FLAG_INPUT_MASK) {
+ case EFFECT_FLAG_INPUT_DIRECT: s.append("direct"); break;
+ case EFFECT_FLAG_INPUT_PROVIDER: s.append("provider"); break;
+ case EFFECT_FLAG_INPUT_BOTH: s.append("direct+provider"); break;
+ default: s.append("not set"); break;
+ }
+ s.append(", ");
+
+ s.append("output mode: ");
+ switch (flags & EFFECT_FLAG_OUTPUT_MASK) {
+ case EFFECT_FLAG_OUTPUT_DIRECT: s.append("direct"); break;
+ case EFFECT_FLAG_OUTPUT_PROVIDER: s.append("provider"); break;
+ case EFFECT_FLAG_OUTPUT_BOTH: s.append("direct+provider"); break;
+ default: s.append("not set"); break;
+ }
+ s.append(", ");
+
+ uint32_t accel = flags & EFFECT_FLAG_HW_ACC_MASK;
+ if (accel) {
+ s.append("hardware acceleration: ");
+ switch (accel) {
+ case EFFECT_FLAG_HW_ACC_SIMPLE: s.append("non-tunneled"); break;
+ case EFFECT_FLAG_HW_ACC_TUNNEL: s.append("tunneled"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ uint32_t modeind = flags & EFFECT_FLAG_AUDIO_MODE_MASK;
+ if (modeind) {
+ s.append("mode indication: ");
+ switch (modeind) {
+ case EFFECT_FLAG_AUDIO_MODE_IND: s.append("required"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ uint32_t srcind = flags & EFFECT_FLAG_AUDIO_SOURCE_MASK;
+ if (srcind) {
+ s.append("source indication: ");
+ switch (srcind) {
+ case EFFECT_FLAG_AUDIO_SOURCE_IND: s.append("required"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ if (flags & EFFECT_FLAG_OFFLOAD_MASK) {
+ s.append("offloadable, ");
+ }
+
+ int len = s.length();
+ if (s.length() > 2) {
+ char *str = s.lockBuffer(len);
+ s.unlockBuffer(len - 2);
+ }
+ return s;
+}
+
+
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -838,9 +944,10 @@
mDescriptor.type.node[2],
mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
result.append(buffer);
- snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
+ snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X (%s)\n",
mDescriptor.apiVersion,
- mDescriptor.flags);
+ mDescriptor.flags,
+ effectFlagsToString(mDescriptor.flags).string());
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- name: %s\n",
mDescriptor.name);
@@ -851,37 +958,37 @@
result.append("\t\t- Input configuration:\n");
result.append("\t\t\tFrames Smp rate Channels Format Buffer\n");
- snprintf(buffer, SIZE, "\t\t\t%05zu %05d %08x %6d %p\n",
+ snprintf(buffer, SIZE, "\t\t\t%05zu %05d %08x %6d (%s) %p\n",
mConfig.inputCfg.buffer.frameCount,
mConfig.inputCfg.samplingRate,
mConfig.inputCfg.channels,
mConfig.inputCfg.format,
+ formatToString((audio_format_t)mConfig.inputCfg.format),
mConfig.inputCfg.buffer.raw);
result.append(buffer);
result.append("\t\t- Output configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
- snprintf(buffer, SIZE, "\t\t\t%p %05zu %05d %08x %d\n",
+ snprintf(buffer, SIZE, "\t\t\t%p %05zu %05d %08x %d (%s)\n",
mConfig.outputCfg.buffer.raw,
mConfig.outputCfg.buffer.frameCount,
mConfig.outputCfg.samplingRate,
mConfig.outputCfg.channels,
- mConfig.outputCfg.format);
+ mConfig.outputCfg.format,
+ formatToString((audio_format_t)mConfig.outputCfg.format));
result.append(buffer);
snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size());
result.append(buffer);
- result.append("\t\t\tPid Priority Ctrl Locked client server\n");
+ result.append("\t\t\t Pid Priority Ctrl Locked client server\n");
for (size_t i = 0; i < mHandles.size(); ++i) {
EffectHandle *handle = mHandles[i];
if (handle != NULL && !handle->destroyed_l()) {
- handle->dump(buffer, SIZE);
+ handle->dumpToBuffer(buffer, SIZE);
result.append(buffer);
}
}
- result.append("\n");
-
write(fd, result.string(), result.length());
if (locked) {
@@ -911,18 +1018,15 @@
}
int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
- if (mCblkMemory != 0) {
- mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
-
- if (mCblk != NULL) {
- new(mCblk) effect_param_cblk_t();
- mBuffer = (uint8_t *)mCblk + bufOffset;
- }
- } else {
+ if (mCblkMemory == 0 ||
+ (mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer())) == NULL) {
ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
sizeof(effect_param_cblk_t));
+ mCblkMemory.clear();
return;
}
+ new(mCblk) effect_param_cblk_t();
+ mBuffer = (uint8_t *)mCblk + bufOffset;
}
AudioFlinger::EffectHandle::~EffectHandle()
@@ -939,6 +1043,11 @@
disconnect(false);
}
+status_t AudioFlinger::EffectHandle::initCheck()
+{
+ return mClient == 0 || mCblkMemory != 0 ? OK : NO_MEMORY;
+}
+
status_t AudioFlinger::EffectHandle::enable()
{
ALOGV("enable %p", this);
@@ -1179,15 +1288,15 @@
}
-void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
+void AudioFlinger::EffectHandle::dumpToBuffer(char* buffer, size_t size)
{
bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
- snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
+ snprintf(buffer, size, "\t\t\t%5d %5d %3s %3s %5u %5u\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mPriority,
- mHasControl,
- !locked,
+ mHasControl ? "yes" : "no",
+ locked ? "yes" : "no",
mCblk ? mCblk->clientIndex : 0,
mCblk ? mCblk->serverIndex : 0
);
@@ -1568,33 +1677,35 @@
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
+ size_t numEffects = mEffects.size();
+ snprintf(buffer, SIZE, " %d effects for session %d\n", numEffects, mSessionId);
result.append(buffer);
- bool locked = AudioFlinger::dumpTryLock(mLock);
- // failed to lock - AudioFlinger is probably deadlocked
- if (!locked) {
- result.append("\tCould not lock mutex:\n");
- }
-
- result.append("\tNum fx In buffer Out buffer Active tracks:\n");
- snprintf(buffer, SIZE, "\t%02zu %p %p %d\n",
- mEffects.size(),
- mInBuffer,
- mOutBuffer,
- mActiveTrackCnt);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- for (size_t i = 0; i < mEffects.size(); ++i) {
- sp<EffectModule> effect = mEffects[i];
- if (effect != 0) {
- effect->dump(fd, args);
+ if (numEffects) {
+ bool locked = AudioFlinger::dumpTryLock(mLock);
+ // failed to lock - AudioFlinger is probably deadlocked
+ if (!locked) {
+ result.append("\tCould not lock mutex:\n");
}
- }
- if (locked) {
- mLock.unlock();
+ result.append("\tIn buffer Out buffer Active tracks:\n");
+ snprintf(buffer, SIZE, "\t%p %p %d\n",
+ mInBuffer,
+ mOutBuffer,
+ mActiveTrackCnt);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ for (size_t i = 0; i < numEffects; ++i) {
+ sp<EffectModule> effect = mEffects[i];
+ if (effect != 0) {
+ effect->dump(fd, args);
+ }
+ }
+
+ if (locked) {
+ mLock.unlock();
+ }
}
}
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index b717857..ccc4825 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -169,6 +169,7 @@
const sp<IEffectClient>& effectClient,
int32_t priority);
virtual ~EffectHandle();
+ virtual status_t initCheck();
// IEffect
virtual status_t enable();
@@ -208,7 +209,7 @@
// destroyed_l() must be called with the associated EffectModule mLock held
bool destroyed_l() const { return mDestroyed; }
- void dump(char* buffer, size_t size);
+ void dumpToBuffer(char* buffer, size_t size);
protected:
friend class AudioFlinger; // for mEffect, mHasControl, mEnabled
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 85d637e..90122e0 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -238,7 +238,7 @@
}
}
- if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
+ if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
// FIXME to avoid priority inversion, don't delete here
delete mixer;
mixer = NULL;
@@ -440,8 +440,9 @@
}
int64_t pts;
- if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts)))
+ if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) {
pts = AudioBufferProvider::kInvalidPTS;
+ }
// process() is CPU-bound
mixer->process(pts);
@@ -695,7 +696,7 @@
void FastMixerDumpState::dump(int fd) const
{
if (mCommand == FastMixerState::INITIAL) {
- fdprintf(fd, "FastMixer not initialized\n");
+ fdprintf(fd, " FastMixer not initialized\n");
return;
}
#define COMMAND_MAX 32
@@ -729,10 +730,10 @@
double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
(mMeasuredWarmupTs.tv_nsec / 1000000.0);
double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
- fdprintf(fd, "FastMixer command=%s writeSequence=%u framesWritten=%u\n"
- " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
- " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
- " mixPeriod=%.2f ms\n",
+ fdprintf(fd, " FastMixer command=%s writeSequence=%u framesWritten=%u\n"
+ " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
+ " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
+ " mixPeriod=%.2f ms\n",
string, mWriteSequence, mFramesWritten,
mNumTracks, mWriteErrors, mUnderruns, mOverruns,
mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
@@ -783,14 +784,20 @@
previousCpukHz = sampleCpukHz;
#endif
}
- fdprintf(fd, "Simple moving statistics over last %.1f seconds:\n", wall.n() * mixPeriodSec);
- fdprintf(fd, " wall clock time in ms per mix cycle:\n"
- " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
- wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6, wall.stddev()*1e-6);
- fdprintf(fd, " raw CPU load in us per mix cycle:\n"
- " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
- loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
- loadNs.stddev()*1e-3);
+ if (n) {
+ fdprintf(fd, " Simple moving statistics over last %.1f seconds:\n",
+ wall.n() * mixPeriodSec);
+ fdprintf(fd, " wall clock time in ms per mix cycle:\n"
+ " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
+ wall.stddev()*1e-6);
+ fdprintf(fd, " raw CPU load in us per mix cycle:\n"
+ " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+ loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
+ loadNs.stddev()*1e-3);
+ } else {
+ fdprintf(fd, " No FastMixer statistics available currently\n");
+ }
#ifdef CPU_FREQUENCY_STATISTICS
fdprintf(fd, " CPU clock frequency in MHz:\n"
" mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
@@ -808,9 +815,9 @@
left.sample(tail[i]);
right.sample(tail[n - (i + 1)]);
}
- fdprintf(fd, "Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
- " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
- " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ fdprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
+ " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
+ " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
right.stddev()*1e-6);
@@ -823,9 +830,9 @@
// Instead we always display all tracks, with an indication
// of whether we think the track is active.
uint32_t trackMask = mTrackMask;
- fdprintf(fd, "Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+ fdprintf(fd, " Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
FastMixerState::kMaxFastTracks, trackMask);
- fdprintf(fd, "Index Active Full Partial Empty Recent Ready\n");
+ fdprintf(fd, " Index Active Full Partial Empty Recent Ready\n");
for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
bool isActive = trackMask & 1;
const FastTrackDump *ftDump = &mTracks[i];
@@ -845,7 +852,7 @@
mostRecent = "?";
break;
}
- fdprintf(fd, "%5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
+ fdprintf(fd, " %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
(underruns.mBitFields.mFull) & UNDERRUN_MASK,
(underruns.mBitFields.mPartial) & UNDERRUN_MASK,
(underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 43b77f3..b5e763d 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -34,9 +34,10 @@
int uid,
IAudioFlinger::track_flags_t flags);
virtual ~Track();
+ virtual status_t initCheck() const;
static void appendDumpHeader(String8& result);
- void dump(char* buffer, size_t size);
+ void dump(char* buffer, size_t size, bool active);
virtual status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
@@ -93,6 +94,8 @@
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
+ bool isFlushPending() const { return mFlushHwPending; }
+ void flushAck();
bool isOutputTrack() const {
return (mStreamType == AUDIO_STREAM_CNT);
@@ -154,6 +157,7 @@
bool mIsInvalid; // non-resettable latch, set by invalidate()
AudioTrackServerProxy* mAudioTrackServerProxy;
bool mResumeToStopping; // track was paused in stopping state.
+ bool mFlushHwPending; // track requests for thread flush
}; // end of Track
class TimedTrack : public Track {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 57de568..fc3171f 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -45,7 +45,7 @@
return tmp; }
static void appendDumpHeader(String8& result);
- void dump(char* buffer, size_t size);
+ void dump(char* buffer, size_t size, bool active);
private:
friend class AudioFlinger; // for mState
@@ -59,5 +59,4 @@
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
- AudioRecordServerProxy* mAudioRecordServerProxy;
};
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 498ddb6..b064e89 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -185,7 +185,11 @@
{
}
-void CpuStats::sample(const String8 &title) {
+void CpuStats::sample(const String8 &title
+#ifndef DEBUG_CPU_USAGE
+ __unused
+#endif
+ ) {
#ifdef DEBUG_CPU_USAGE
// get current thread's delta CPU time in wall clock ns
double wcNs;
@@ -269,8 +273,8 @@
: Thread(false /*canCallJava*/),
mType(type),
mAudioFlinger(audioFlinger),
- // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
- // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
+ // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
+ // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
mParamStatus(NO_ERROR),
//FIXME: mStandby should be true here. Is this some kind of hack?
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
@@ -297,6 +301,17 @@
}
}
+status_t AudioFlinger::ThreadBase::readyToRun()
+{
+ status_t status = initCheck();
+ if (status == NO_ERROR) {
+ ALOGI("AudioFlinger's thread %p ready to run", this);
+ } else {
+ ALOGE("No working audio driver found.");
+ }
+ return status;
+}
+
void AudioFlinger::ThreadBase::exit()
{
ALOGV("ThreadBase::exit");
@@ -369,7 +384,13 @@
void AudioFlinger::ThreadBase::processConfigEvents()
{
- mLock.lock();
+ Mutex::Autolock _l(mLock);
+ processConfigEvents_l();
+}
+
+// post condition: mConfigEvents.isEmpty()
+void AudioFlinger::ThreadBase::processConfigEvents_l()
+{
while (!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent *event = mConfigEvents[0];
@@ -377,35 +398,81 @@
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
- switch(event->type()) {
- case CFG_EVENT_PRIO: {
- PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
- // FIXME Need to understand why this has be done asynchronously
- int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(),
- true /*asynchronous*/);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
- "error %d",
- prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
- }
- } break;
- case CFG_EVENT_IO: {
- IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
- mAudioFlinger->mLock.lock();
+ switch (event->type()) {
+ case CFG_EVENT_PRIO: {
+ PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
+ // FIXME Need to understand why this has be done asynchronously
+ int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(),
+ true /*asynchronous*/);
+ if (err != 0) {
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+ prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
+ }
+ } break;
+ case CFG_EVENT_IO: {
+ IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
+ {
+ Mutex::Autolock _l(mAudioFlinger->mLock);
audioConfigChanged_l(ioEvent->event(), ioEvent->param());
- mAudioFlinger->mLock.unlock();
- } break;
- default:
- ALOGE("processConfigEvents() unknown event type %d", event->type());
- break;
+ }
+ } break;
+ default:
+ ALOGE("processConfigEvents() unknown event type %d", event->type());
+ break;
}
delete event;
mLock.lock();
}
- mLock.unlock();
}
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
+ String8 s;
+ if (output) {
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
+ if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
+ } else {
+ if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
+ if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
+ if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
+ if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
+ if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
+ if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
+ if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
+ if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
+ if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
+ if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
+ if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
+ }
+ int len = s.length();
+ if (s.length() > 2) {
+ char *str = s.lockBuffer(len);
+ s.unlockBuffer(len - 2);
+ }
+ return s;
+}
+
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -413,47 +480,43 @@
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
- write(fd, buffer, strlen(buffer));
+ fdprintf(fd, "thread %p maybe dead locked\n", this);
}
- snprintf(buffer, SIZE, "io handle: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, "TID: %d\n", getTid());
- result.append(buffer);
- snprintf(buffer, SIZE, "standby: %d\n", mStandby);
- result.append(buffer);
- snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
- result.append(buffer);
- snprintf(buffer, SIZE, "HAL frame count: %zu\n", mFrameCount);
- result.append(buffer);
- snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
- result.append(buffer);
- snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, "Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, "Frame size: %zu\n", mFrameSize);
- result.append(buffer);
-
- snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
- result.append(buffer);
- result.append(" Index Command");
- for (size_t i = 0; i < mNewParameters.size(); ++i) {
- snprintf(buffer, SIZE, "\n %02zu ", i);
- result.append(buffer);
- result.append(mNewParameters[i]);
+ fdprintf(fd, " I/O handle: %d\n", mId);
+ fdprintf(fd, " TID: %d\n", getTid());
+ fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
+ fdprintf(fd, " Sample rate: %u\n", mSampleRate);
+ fdprintf(fd, " HAL frame count: %zu\n", mFrameCount);
+ fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
+ fdprintf(fd, " Channel Count: %u\n", mChannelCount);
+ fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
+ channelMaskToString(mChannelMask, mType != RECORD).string());
+ fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+ fdprintf(fd, " Frame size: %zu\n", mFrameSize);
+ fdprintf(fd, " Pending setParameters commands:");
+ size_t numParams = mNewParameters.size();
+ if (numParams) {
+ fdprintf(fd, "\n Index Command");
+ for (size_t i = 0; i < numParams; ++i) {
+ fdprintf(fd, "\n %02zu ", i);
+ fdprintf(fd, mNewParameters[i]);
+ }
+ fdprintf(fd, "\n");
+ } else {
+ fdprintf(fd, " none\n");
}
-
- snprintf(buffer, SIZE, "\n\nPending config events: \n");
- result.append(buffer);
- for (size_t i = 0; i < mConfigEvents.size(); i++) {
- mConfigEvents[i]->dump(buffer, SIZE);
- result.append(buffer);
+ fdprintf(fd, " Pending config events:");
+ size_t numConfig = mConfigEvents.size();
+ if (numConfig) {
+ for (size_t i = 0; i < numConfig; i++) {
+ mConfigEvents[i]->dump(buffer, SIZE);
+ fdprintf(fd, "\n %s", buffer);
+ }
+ fdprintf(fd, "\n");
+ } else {
+ fdprintf(fd, " none\n");
}
- result.append("\n");
-
- write(fd, result.string(), result.size());
if (locked) {
mLock.unlock();
@@ -466,10 +529,11 @@
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "\n- %zu Effect Chains:\n", mEffectChains.size());
+ size_t numEffectChains = mEffectChains.size();
+ snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mEffectChains.size(); ++i) {
+ for (size_t i = 0; i < numEffectChains; ++i) {
sp<EffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
@@ -586,7 +650,7 @@
mPowerManager.clear();
}
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
@@ -739,8 +803,7 @@
int sessionId,
effect_descriptor_t *desc,
int *enabled,
- status_t *status
- )
+ status_t *status)
{
sp<EffectModule> effect;
sp<EffectHandle> handle;
@@ -829,7 +892,10 @@
}
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
- lStatus = effect->addHandle(handle.get());
+ lStatus = handle->initCheck();
+ if (lStatus == OK) {
+ lStatus = effect->addHandle(handle.get());
+ }
if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
}
@@ -850,9 +916,7 @@
handle.clear();
}
- if (status != NULL) {
- *status = lStatus;
- }
+ *status = lStatus;
return handle;
}
@@ -1002,7 +1066,7 @@
type_t type)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
mNormalFrameCount(0), mMixBuffer(NULL),
- mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ mSuspended(0), mBytesWritten(0),
mActiveTracksGeneration(0),
// mStreamTypes[] initialized in constructor body
mOutput(output),
@@ -1060,7 +1124,7 @@
AudioFlinger::PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete [] mAllocMixBuffer;
+ delete[] mMixBuffer;
}
void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1070,13 +1134,13 @@
dumpEffectChains(fd, args);
}
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- result.appendFormat("Output thread %p stream volumes in dB:\n ", this);
+ result.appendFormat(" Stream volumes in dB: ");
for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
const stream_type_t *st = &mStreamTypes[i];
if (i > 0) {
@@ -1091,75 +1155,67 @@
write(fd, result.string(), result.length());
result.clear();
- snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
- result.append(buffer);
- Track::appendDumpHeader(result);
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
- }
-
- snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
- result.append(buffer);
- Track::appendDumpHeader(result);
- for (size_t i = 0; i < mActiveTracks.size(); ++i) {
- sp<Track> track = mActiveTracks[i].promote();
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
- }
- write(fd, result.string(), result.size());
-
// These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
FastTrackUnderruns underruns = getFastTrackUnderruns(0);
- fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
+ fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
+
+ size_t numtracks = mTracks.size();
+ size_t numactive = mActiveTracks.size();
+ fdprintf(fd, " %d Tracks", numtracks);
+ size_t numactiveseen = 0;
+ if (numtracks) {
+ fdprintf(fd, " of which %d are active\n", numactive);
+ Track::appendDumpHeader(result);
+ for (size_t i = 0; i < numtracks; ++i) {
+ sp<Track> track = mTracks[i];
+ if (track != 0) {
+ bool active = mActiveTracks.indexOf(track) >= 0;
+ if (active) {
+ numactiveseen++;
+ }
+ track->dump(buffer, SIZE, active);
+ result.append(buffer);
+ }
+ }
+ } else {
+ result.append("\n");
+ }
+ if (numactiveseen != numactive) {
+ // some tracks in the active list were not in the tracks list
+ snprintf(buffer, SIZE, " The following tracks are in the active list but"
+ " not in the track list\n");
+ result.append(buffer);
+ Track::appendDumpHeader(result);
+ for (size_t i = 0; i < numactive; ++i) {
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0 && mTracks.indexOf(track) < 0) {
+ track->dump(buffer, SIZE, true);
+ result.append(buffer);
+ }
+ }
+ }
+
+ write(fd, result.string(), result.size());
+
}
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
- result.append(buffer);
- snprintf(buffer, SIZE, "Normal frame count: %zu\n", mNormalFrameCount);
- result.append(buffer);
- snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
- ns2ms(systemTime() - mLastWriteTime));
- result.append(buffer);
- snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
- result.append(buffer);
- snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
- result.append(buffer);
- snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
- result.append(buffer);
- snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
- result.append(buffer);
- snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
- result.append(buffer);
- write(fd, result.string(), result.size());
- fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
+ fdprintf(fd, "\nOutput thread %p:\n", this);
+ fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
+ fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+ fdprintf(fd, " Total writes: %d\n", mNumWrites);
+ fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
+ fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
+ fdprintf(fd, " Suspend count: %d\n", mSuspended);
+ fdprintf(fd, " Mix buffer : %p\n", mMixBuffer);
+ fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
dumpBase(fd, args);
}
// Thread virtuals
-status_t AudioFlinger::PlaybackThread::readyToRun()
-{
- status_t status = initCheck();
- if (status == NO_ERROR) {
- ALOGI("AudioFlinger's thread %p ready to run", this);
- } else {
- ALOGE("No working audio driver found.");
- }
- return status;
-}
void AudioFlinger::PlaybackThread::onFirstRef()
{
@@ -1182,7 +1238,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
@@ -1190,6 +1246,7 @@
int uid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<Track> track;
status_t lStatus;
@@ -1256,12 +1313,13 @@
}
}
}
+ *pFrameCount = frameCount;
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
- "for output %p with format %d",
+ ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
+ "for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
@@ -1269,16 +1327,16 @@
}
} else if (mType == OFFLOAD) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
- "for output %p with format %d",
+ ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
+ "for output %p with format %#x",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
} else {
if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
- ALOGE("createTrack_l() Bad parameter: format %d \""
- "for output %p with format %d",
+ ALOGE("createTrack_l() Bad parameter: format %#x \""
+ "for output %p with format %#x",
format, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
@@ -1324,8 +1382,13 @@
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId, uid);
}
- if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
- lStatus = NO_MEMORY;
+
+ // new Track always returns non-NULL,
+ // but TimedTrack::create() is a factory that could fail by returning NULL
+ lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
+ if (lStatus != NO_ERROR) {
+ ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
@@ -1350,9 +1413,7 @@
lStatus = NO_ERROR;
Exit:
- if (status) {
- *status = lStatus;
- }
+ *status = lStatus;
return track;
}
@@ -1471,9 +1532,7 @@
status = NO_ERROR;
}
- ALOGV("signal playback thread");
- broadcast_l();
-
+ onAddNewTrack_l();
return status;
}
@@ -1599,7 +1658,7 @@
// static
int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
- void *param,
+ void *param __unused,
void *cookie)
{
AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
@@ -1633,14 +1692,15 @@
mChannelCount = popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
if (!audio_is_valid_format(mFormat)) {
- LOG_FATAL("HAL format %d not valid for output", mFormat);
+ LOG_FATAL("HAL format %#x not valid for output", mFormat);
}
if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
+ LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT",
mFormat);
}
mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
- mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+ mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
+ mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
mFrameCount);
@@ -1697,11 +1757,11 @@
ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
mNormalFrameCount);
- delete[] mAllocMixBuffer;
- size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize;
- mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1];
- mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align);
- memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize);
+ delete[] mMixBuffer;
+ size_t normalBufferSize = mNormalFrameCount * mFrameSize;
+ // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)
+ mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];
+ memset(mMixBuffer, 0, normalBufferSize);
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
@@ -1839,7 +1899,7 @@
const Vector< sp<Track> >& tracksToRemove)
{
size_t count = tracksToRemove.size();
- if (count) {
+ if (count > 0) {
for (size_t i = 0 ; i < count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
if (!track->isOutputTrack()) {
@@ -1915,7 +1975,7 @@
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
- size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t);
+ size_t offset = (mCurrentWriteLength - mBytesRemaining);
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
@@ -1926,7 +1986,7 @@
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->stream->write(mOutput->stream,
- mMixBuffer + offset, mBytesRemaining);
+ (char *)mMixBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -2346,20 +2406,20 @@
(mMixerStatus == MIXER_DRAIN_ALL)) {
threadLoop_drain();
}
-if (mType == MIXER) {
- // write blocked detection
- nsecs_t now = systemTime();
- nsecs_t delta = now - mLastWriteTime;
- if (!mStandby && delta > maxPeriod) {
- mNumDelayedWrites++;
- if ((now - lastWarning) > kWarningThrottleNs) {
- ATRACE_NAME("underrun");
- ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
- ns2ms(delta), mNumDelayedWrites, this);
- lastWarning = now;
+ if (mType == MIXER) {
+ // write blocked detection
+ nsecs_t now = systemTime();
+ nsecs_t delta = now - mLastWriteTime;
+ if (!mStandby && delta > maxPeriod) {
+ mNumDelayedWrites++;
+ if ((now - lastWarning) > kWarningThrottleNs) {
+ ATRACE_NAME("underrun");
+ ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+ ns2ms(delta), mNumDelayedWrites, this);
+ lastWarning = now;
+ }
}
}
-}
} else {
usleep(sleepTime);
@@ -2407,7 +2467,7 @@
void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
{
size_t count = tracksToRemove.size();
- if (count) {
+ if (count > 0) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
mActiveTracks.remove(track);
@@ -2711,12 +2771,6 @@
PlaybackThread::threadLoop_standby();
}
-// Empty implementation for standard mixer
-// Overridden for offloaded playback
-void AudioFlinger::PlaybackThread::flushOutput_l()
-{
-}
-
bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
{
return false;
@@ -2748,6 +2802,12 @@
}
}
+void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+{
+ ALOGV("signal playback thread");
+ broadcast_l();
+}
+
void AudioFlinger::MixerThread::threadLoop_mix()
{
// obtain the presentation timestamp of the next output buffer
@@ -2800,7 +2860,7 @@
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
- memset (mMixBuffer, 0, mixBufferSize);
+ memset(mMixBuffer, 0, mixBufferSize);
sleepTime = 0;
ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
"anticipated start");
@@ -3025,12 +3085,14 @@
// +1 for rounding and +1 for additional sample needed for interpolation
desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
- // because cblk->framesReady() will include these frames
+ // because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+#if 0
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+#endif
}
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
@@ -3356,6 +3418,7 @@
if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
+ // no need to save value, since it's constant
reconfig = true;
}
}
@@ -3363,6 +3426,7 @@
if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
+ // no need to save value, since it's constant
reconfig = true;
}
}
@@ -3466,9 +3530,7 @@
PlaybackThread::dumpInternals(fd, args);
- snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
- result.append(buffer);
- write(fd, result.string(), result.size());
+ fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
// Make a non-atomic copy of fast mixer dump state so it won't change underneath us
const FastMixerDumpState copy(mFastMixerDumpState);
@@ -3722,14 +3784,14 @@
}
// getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
- int sessionId)
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
+ int sessionId __unused)
{
return 0;
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
{
}
@@ -3982,6 +4044,17 @@
sp<Track> l = mLatestActiveTrack.promote();
bool last = l.get() == track;
+ if (track->isInvalid()) {
+ ALOGW("An invalidated track shouldn't be in active list");
+ tracksToRemove->add(track);
+ continue;
+ }
+
+ if (track->mState == TrackBase::IDLE) {
+ ALOGW("An idle track shouldn't be in active list");
+ continue;
+ }
+
if (track->isPausing()) {
track->setPaused();
if (last) {
@@ -4000,6 +4073,11 @@
mBytesRemaining = 0; // stop writing
}
tracksToRemove->add(track);
+ } else if (track->isFlushPending()) {
+ track->flushAck();
+ if (last) {
+ mFlushPending = true;
+ }
} else if (track->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -4049,7 +4127,6 @@
// seek when resuming.
if (previousTrack->sessionId() != track->sessionId()) {
previousTrack->invalidate();
- mFlushPending = true;
}
}
}
@@ -4125,9 +4202,6 @@
// if resume is received before pause is executed.
if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
mOutput->stream->pause(mOutput->stream);
- if (!doHwPause) {
- doHwResume = true;
- }
}
if (mFlushPending) {
flushHw_l();
@@ -4143,11 +4217,6 @@
return mixerStatus;
}
-void AudioFlinger::OffloadThread::flushOutput_l()
-{
- mFlushPending = true;
-}
-
// must be called with thread mutex locked
bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
{
@@ -4162,15 +4231,15 @@
// must be called with thread mutex locked
bool AudioFlinger::OffloadThread::shouldStandby_l()
{
- bool TrackPaused = false;
+ bool trackPaused = false;
// do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
// after a timeout and we will enter standby then.
if (mTracks.size() > 0) {
- TrackPaused = mTracks[mTracks.size() - 1]->isPaused();
+ trackPaused = mTracks[mTracks.size() - 1]->isPaused();
}
- return !mStandby && !TrackPaused;
+ return !mStandby && !trackPaused;
}
@@ -4188,6 +4257,8 @@
mBytesRemaining = 0;
mPausedWriteLength = 0;
mPausedBytesRemaining = 0;
+ mHwPaused = false;
+
if (mUseAsyncWrite) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -4198,6 +4269,18 @@
}
}
+void AudioFlinger::OffloadThread::onAddNewTrack_l()
+{
+ sp<Track> previousTrack = mPreviousTrack.promote();
+ sp<Track> latestTrack = mLatestActiveTrack.promote();
+
+ if (previousTrack != 0 && latestTrack != 0 &&
+ (previousTrack->sessionId() != latestTrack->sessionId())) {
+ mFlushPending = true;
+ }
+ PlaybackThread::onAddNewTrack_l();
+}
+
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
@@ -4377,8 +4460,10 @@
#endif
) :
ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
- mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
- // mRsmpInIndex and mBufferSize set by readInputParameters()
+ mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear
+ // are set by readInputParameters()
+ // mRsmpInIndex LEGACY
mReqChannelCount(popcount(channelMask)),
mReqSampleRate(sampleRate)
// mBytesRead is only meaningful while active, and so is cleared in start()
@@ -4388,6 +4473,7 @@
#endif
{
snprintf(mName, kNameLength, "AudioIn_%X", id);
+ mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
readInputParameters();
}
@@ -4395,6 +4481,7 @@
AudioFlinger::RecordThread::~RecordThread()
{
+ mAudioFlinger->unregisterWriter(mNBLogWriter);
delete[] mRsmpInBuffer;
delete mResampler;
delete[] mRsmpOutBuffer;
@@ -4405,230 +4492,323 @@
run(mName, PRIORITY_URGENT_AUDIO);
}
-status_t AudioFlinger::RecordThread::readyToRun()
-{
- status_t status = initCheck();
- ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
- return status;
-}
-
bool AudioFlinger::RecordThread::threadLoop()
{
- AudioBufferProvider::Buffer buffer;
- sp<RecordTrack> activeTrack;
- Vector< sp<EffectChain> > effectChains;
-
nsecs_t lastWarning = 0;
inputStandBy();
- {
- Mutex::Autolock _l(mLock);
- activeTrack = mActiveTrack;
- acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1);
- }
// used to verify we've read at least once before evaluating how many bytes were read
bool readOnce = false;
- // start recording
- while (!exitPending()) {
+ // used to request a deferred sleep, to be executed later while mutex is unlocked
+ bool doSleep = false;
- processConfigEvents();
+reacquire_wakelock:
+ sp<RecordTrack> activeTrack;
+ int activeTracksGen;
+ {
+ Mutex::Autolock _l(mLock);
+ size_t size = mActiveTracks.size();
+ activeTracksGen = mActiveTracksGen;
+ if (size > 0) {
+ // FIXME an arbitrary choice
+ activeTrack = mActiveTracks[0];
+ acquireWakeLock_l(activeTrack->uid());
+ if (size > 1) {
+ SortedVector<int> tmp;
+ for (size_t i = 0; i < size; i++) {
+ tmp.add(mActiveTracks[i]->uid());
+ }
+ updateWakeLockUids_l(tmp);
+ }
+ } else {
+ acquireWakeLock_l(-1);
+ }
+ }
+
+ // start recording
+ for (;;) {
+ TrackBase::track_state activeTrackState;
+ Vector< sp<EffectChain> > effectChains;
+
+ // sleep with mutex unlocked
+ if (doSleep) {
+ doSleep = false;
+ usleep(kRecordThreadSleepUs);
+ }
{ // scope for mLock
Mutex::Autolock _l(mLock);
- checkForNewParameters_l();
- if (mActiveTrack != 0 && activeTrack != mActiveTrack) {
- SortedVector<int> tmp;
- tmp.add(mActiveTrack->uid());
- updateWakeLockUids_l(tmp);
+
+ processConfigEvents_l();
+ // return value 'reconfig' is currently unused
+ bool reconfig = checkForNewParameters_l();
+
+ // check exitPending here because checkForNewParameters_l() and
+ // checkForNewParameters_l() can temporarily release mLock
+ if (exitPending()) {
+ break;
}
- activeTrack = mActiveTrack;
- if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
- standby();
- if (exitPending()) {
- break;
- }
-
+ // if no active track(s), then standby and release wakelock
+ size_t size = mActiveTracks.size();
+ if (size == 0) {
+ standbyIfNotAlreadyInStandby();
+ // exitPending() can't become true here
releaseWakeLock_l();
ALOGV("RecordThread: loop stopping");
// go to sleep
mWaitWorkCV.wait(mLock);
ALOGV("RecordThread: loop starting");
- acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1);
+ goto reacquire_wakelock;
+ }
+
+ if (mActiveTracksGen != activeTracksGen) {
+ activeTracksGen = mActiveTracksGen;
+ SortedVector<int> tmp;
+ for (size_t i = 0; i < size; i++) {
+ tmp.add(mActiveTracks[i]->uid());
+ }
+ updateWakeLockUids_l(tmp);
+ // FIXME an arbitrary choice
+ activeTrack = mActiveTracks[0];
+ }
+
+ if (activeTrack->isTerminated()) {
+ removeTrack_l(activeTrack);
+ mActiveTracks.remove(activeTrack);
+ mActiveTracksGen++;
continue;
}
- if (mActiveTrack != 0) {
- if (mActiveTrack->isTerminated()) {
- removeTrack_l(mActiveTrack);
- mActiveTrack.clear();
- } else if (mActiveTrack->mState == TrackBase::PAUSING) {
- standby();
- mActiveTrack.clear();
+
+ activeTrackState = activeTrack->mState;
+ switch (activeTrackState) {
+ case TrackBase::PAUSING:
+ standbyIfNotAlreadyInStandby();
+ mActiveTracks.remove(activeTrack);
+ mActiveTracksGen++;
+ mStartStopCond.broadcast();
+ doSleep = true;
+ continue;
+
+ case TrackBase::RESUMING:
+ mStandby = false;
+ if (mReqChannelCount != activeTrack->channelCount()) {
+ mActiveTracks.remove(activeTrack);
+ mActiveTracksGen++;
mStartStopCond.broadcast();
- } else if (mActiveTrack->mState == TrackBase::RESUMING) {
- if (mReqChannelCount != mActiveTrack->channelCount()) {
- mActiveTrack.clear();
- mStartStopCond.broadcast();
- } else if (readOnce) {
- // record start succeeds only if first read from audio input
- // succeeds
- if (mBytesRead >= 0) {
- mActiveTrack->mState = TrackBase::ACTIVE;
- } else {
- mActiveTrack.clear();
- }
- mStartStopCond.broadcast();
- }
- mStandby = false;
+ continue;
}
+ if (readOnce) {
+ mStartStopCond.broadcast();
+ // record start succeeds only if first read from audio input succeeds
+ if (mBytesRead < 0) {
+ mActiveTracks.remove(activeTrack);
+ mActiveTracksGen++;
+ continue;
+ }
+ activeTrack->mState = TrackBase::ACTIVE;
+ }
+ break;
+
+ case TrackBase::ACTIVE:
+ break;
+
+ case TrackBase::IDLE:
+ doSleep = true;
+ continue;
+
+ default:
+ LOG_FATAL("Unexpected activeTrackState %d", activeTrackState);
}
lockEffectChains_l(effectChains);
}
- if (mActiveTrack != 0) {
- if (mActiveTrack->mState != TrackBase::ACTIVE &&
- mActiveTrack->mState != TrackBase::RESUMING) {
- unlockEffectChains(effectChains);
- usleep(kRecordThreadSleepUs);
- continue;
- }
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
+ // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable
+ // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING
- buffer.frameCount = mFrameCount;
- status_t status = mActiveTrack->getNextBuffer(&buffer);
- if (status == NO_ERROR) {
- readOnce = true;
- size_t framesOut = buffer.frameCount;
- if (mResampler == NULL) {
- // no resampling
- while (framesOut) {
- size_t framesIn = mFrameCount - mRsmpInIndex;
- if (framesIn) {
- int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
- int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
- mActiveTrack->mFrameSize;
- if (framesIn > framesOut)
- framesIn = framesOut;
- mRsmpInIndex += framesIn;
- framesOut -= framesIn;
- if (mChannelCount == mReqChannelCount) {
- memcpy(dst, src, framesIn * mFrameSize);
- } else {
- if (mChannelCount == 1) {
- upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
- (int16_t *)src, framesIn);
- } else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
- (int16_t *)src, framesIn);
- }
- }
- }
- if (framesOut && mFrameCount == mRsmpInIndex) {
- void *readInto;
- if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) {
- readInto = buffer.raw;
- framesOut = 0;
- } else {
- readInto = mRsmpInBuffer;
- mRsmpInIndex = 0;
- }
- mBytesRead = mInput->stream->read(mInput->stream, readInto,
- mBufferSize);
- if (mBytesRead <= 0) {
- if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
- {
- ALOGE("Error reading audio input");
- // Force input into standby so that it tries to
- // recover at next read attempt
- inputStandBy();
- usleep(kRecordThreadSleepUs);
- }
- mRsmpInIndex = mFrameCount;
- framesOut = 0;
- buffer.frameCount = 0;
- }
-#ifdef TEE_SINK
- else if (mTeeSink != 0) {
- (void) mTeeSink->write(readInto,
- mBytesRead >> Format_frameBitShift(mTeeSink->format()));
- }
-#endif
- }
- }
- } else {
- // resampling
-
- // resampler accumulates, but we only have one source track
- memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
- // alter output frame count as if we were expecting stereo samples
- if (mChannelCount == 1 && mReqChannelCount == 1) {
- framesOut >>= 1;
- }
- mResampler->resample(mRsmpOutBuffer, framesOut,
- this /* AudioBufferProvider* */);
- // ditherAndClamp() works as long as all buffers returned by
- // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
- if (mChannelCount == 2 && mReqChannelCount == 1) {
- // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
- ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
- // the resampler always outputs stereo samples:
- // do post stereo to mono conversion
- downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
- framesOut);
- } else {
- ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
- }
- // now done with mRsmpOutBuffer
-
- }
- if (mFramestoDrop == 0) {
- mActiveTrack->releaseBuffer(&buffer);
- } else {
- if (mFramestoDrop > 0) {
- mFramestoDrop -= buffer.frameCount;
- if (mFramestoDrop <= 0) {
- clearSyncStartEvent();
- }
- } else {
- mFramestoDrop += buffer.frameCount;
- if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
- mSyncStartEvent->isCancelled()) {
- ALOGW("Synced record %s, session %d, trigger session %d",
- (mFramestoDrop >= 0) ? "timed out" : "cancelled",
- mActiveTrack->sessionId(),
- (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
- clearSyncStartEvent();
- }
- }
- }
- mActiveTrack->clearOverflow();
- }
- // client isn't retrieving buffers fast enough
- else {
- if (!mActiveTrack->setOverflow()) {
- nsecs_t now = systemTime();
- if ((now - lastWarning) > kWarningThrottleNs) {
- ALOGW("RecordThread: buffer overflow");
- lastWarning = now;
- }
- }
- // Release the processor for a while before asking for a new buffer.
- // This will give the application more chance to read from the buffer and
- // clear the overflow.
- usleep(kRecordThreadSleepUs);
- }
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ // thread mutex is not locked, but effect chain is locked
+ effectChains[i]->process_l();
}
+
+ AudioBufferProvider::Buffer buffer;
+ buffer.frameCount = mFrameCount;
+ status_t status = activeTrack->getNextBuffer(&buffer);
+ if (status == NO_ERROR) {
+ readOnce = true;
+ size_t framesOut = buffer.frameCount;
+ if (mResampler == NULL) {
+ // no resampling
+ while (framesOut) {
+ size_t framesIn = mFrameCount - mRsmpInIndex;
+ if (framesIn > 0) {
+ int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+ int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+ activeTrack->mFrameSize;
+ if (framesIn > framesOut) {
+ framesIn = framesOut;
+ }
+ mRsmpInIndex += framesIn;
+ framesOut -= framesIn;
+ if (mChannelCount == mReqChannelCount) {
+ memcpy(dst, src, framesIn * mFrameSize);
+ } else {
+ if (mChannelCount == 1) {
+ upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
+ (int16_t *)src, framesIn);
+ } else {
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+ (int16_t *)src, framesIn);
+ }
+ }
+ }
+ if (framesOut > 0 && mFrameCount == mRsmpInIndex) {
+ void *readInto;
+ if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) {
+ readInto = buffer.raw;
+ framesOut = 0;
+ } else {
+ readInto = mRsmpInBuffer;
+ mRsmpInIndex = 0;
+ }
+ mBytesRead = mInput->stream->read(mInput->stream, readInto, mBufferSize);
+ if (mBytesRead <= 0) {
+ // TODO: verify that it's benign to use a stale track state
+ if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE))
+ {
+ ALOGE("Error reading audio input");
+ // Force input into standby so that it tries to
+ // recover at next read attempt
+ inputStandBy();
+ doSleep = true;
+ }
+ mRsmpInIndex = mFrameCount;
+ framesOut = 0;
+ buffer.frameCount = 0;
+ }
+#ifdef TEE_SINK
+ else if (mTeeSink != 0) {
+ (void) mTeeSink->write(readInto,
+ mBytesRead >> Format_frameBitShift(mTeeSink->format()));
+ }
+#endif
+ }
+ }
+ } else {
+ // resampling
+
+ // avoid busy-waiting if client doesn't keep up
+ bool madeProgress = false;
+
+ // keep mRsmpInBuffer full so resampler always has sufficient input
+ for (;;) {
+ int32_t rear = mRsmpInRear;
+ ssize_t filled = rear - mRsmpInFront;
+ ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2);
+ // exit once there is enough data in buffer for resampler
+ if ((size_t) filled >= mRsmpInFrames) {
+ break;
+ }
+ size_t avail = mRsmpInFramesP2 - filled;
+ // Only try to read full HAL buffers.
+ // But if the HAL read returns a partial buffer, use it.
+ if (avail < mFrameCount) {
+ ALOGE("insufficient space to read: avail %d < mFrameCount %d",
+ avail, mFrameCount);
+ break;
+ }
+ // If 'avail' is non-contiguous, first read past the nominal end of buffer, then
+ // copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
+ rear &= mRsmpInFramesP2 - 1;
+ mBytesRead = mInput->stream->read(mInput->stream,
+ &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
+ if (mBytesRead <= 0) {
+ ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize);
+ break;
+ }
+ ALOG_ASSERT((size_t) mBytesRead <= mBufferSize);
+ size_t framesRead = mBytesRead / mFrameSize;
+ ALOG_ASSERT(framesRead > 0);
+ madeProgress = true;
+ // If 'avail' was non-contiguous, we now correct for reading past end of buffer.
+ size_t part1 = mRsmpInFramesP2 - rear;
+ if (framesRead > part1) {
+ memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
+ (framesRead - part1) * mFrameSize);
+ }
+ mRsmpInRear += framesRead;
+ }
+
+ if (!madeProgress) {
+ ALOGV("Did not make progress");
+ usleep(((mFrameCount * 1000) / mSampleRate) * 1000);
+ }
+
+ // resampler accumulates, but we only have one source track
+ memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
+ mResampler->resample(mRsmpOutBuffer, framesOut,
+ this /* AudioBufferProvider* */);
+ // ditherAndClamp() works as long as all buffers returned by
+ // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
+ if (mReqChannelCount == 1) {
+ // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
+ ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ // the resampler always outputs stereo samples:
+ // do post stereo to mono conversion
+ downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
+ framesOut);
+ } else {
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+ // now done with mRsmpOutBuffer
+
+ }
+ if (mFramestoDrop == 0) {
+ activeTrack->releaseBuffer(&buffer);
+ } else {
+ if (mFramestoDrop > 0) {
+ mFramestoDrop -= buffer.frameCount;
+ if (mFramestoDrop <= 0) {
+ clearSyncStartEvent();
+ }
+ } else {
+ mFramestoDrop += buffer.frameCount;
+ if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
+ mSyncStartEvent->isCancelled()) {
+ ALOGW("Synced record %s, session %d, trigger session %d",
+ (mFramestoDrop >= 0) ? "timed out" : "cancelled",
+ activeTrack->sessionId(),
+ (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
+ clearSyncStartEvent();
+ }
+ }
+ }
+ activeTrack->clearOverflow();
+ }
+ // client isn't retrieving buffers fast enough
+ else {
+ if (!activeTrack->setOverflow()) {
+ nsecs_t now = systemTime();
+ if ((now - lastWarning) > kWarningThrottleNs) {
+ ALOGW("RecordThread: buffer overflow");
+ lastWarning = now;
+ }
+ }
+ // Release the processor for a while before asking for a new buffer.
+ // This will give the application more chance to read from the buffer and
+ // clear the overflow.
+ doSleep = true;
+ }
+
// enable changes in effect chain
unlockEffectChains(effectChains);
- effectChains.clear();
+ // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
}
- standby();
+ standbyIfNotAlreadyInStandby();
{
Mutex::Autolock _l(mLock);
@@ -4636,7 +4816,8 @@
sp<RecordTrack> track = mTracks[i];
track->invalidate();
}
- mActiveTrack.clear();
+ mActiveTracks.clear();
+ mActiveTracksGen++;
mStartStopCond.broadcast();
}
@@ -4646,7 +4827,7 @@
return false;
}
-void AudioFlinger::RecordThread::standby()
+void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
{
if (!mStandby) {
inputStandBy();
@@ -4659,18 +4840,19 @@
mInput->stream->common.standby(&mInput->stream->common);
}
-sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
+sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<RecordTrack> track;
status_t lStatus;
@@ -4679,6 +4861,7 @@
ALOGE("createRecordTrack_l() audio driver not initialized");
goto Exit;
}
+
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
@@ -4729,6 +4912,7 @@
}
}
}
+ *pFrameCount = frameCount;
// FIXME use flags and tid similar to createTrack_l()
@@ -4738,10 +4922,10 @@
track = new RecordTrack(this, client, sampleRate,
format, channelMask, frameCount, sessionId, uid);
- if (track->getCblk() == 0) {
- ALOGE("createRecordTrack_l() no control block");
- lStatus = NO_MEMORY;
- track.clear();
+ lStatus = track->initCheck();
+ if (lStatus != NO_ERROR) {
+ ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
@@ -4762,9 +4946,7 @@
lStatus = NO_ERROR;
Exit:
- if (status) {
- *status = lStatus;
- }
+ *status = lStatus;
return track;
}
@@ -4795,43 +4977,57 @@
}
{
+ // This section is a rendezvous between binder thread executing start() and RecordThread
AutoMutex lock(mLock);
- if (mActiveTrack != 0) {
- if (recordTrack != mActiveTrack.get()) {
+ if (mActiveTracks.size() > 0) {
+ // FIXME does not work for multiple active tracks
+ if (mActiveTracks.indexOf(recordTrack) != 0) {
status = -EBUSY;
- } else if (mActiveTrack->mState == TrackBase::PAUSING) {
- mActiveTrack->mState = TrackBase::ACTIVE;
+ } else if (recordTrack->mState == TrackBase::PAUSING) {
+ recordTrack->mState = TrackBase::ACTIVE;
}
return status;
}
+ // FIXME why? already set in constructor, 'STARTING_1' would be more accurate
recordTrack->mState = TrackBase::IDLE;
- mActiveTrack = recordTrack;
+ mActiveTracks.add(recordTrack);
+ mActiveTracksGen++;
mLock.unlock();
status_t status = AudioSystem::startInput(mId);
mLock.lock();
+ // FIXME should verify that mActiveTrack is still == recordTrack
if (status != NO_ERROR) {
- mActiveTrack.clear();
+ mActiveTracks.remove(recordTrack);
+ mActiveTracksGen++;
clearSyncStartEvent();
return status;
}
+ // FIXME LEGACY
mRsmpInIndex = mFrameCount;
+ mRsmpInFront = 0;
+ mRsmpInRear = 0;
+ mRsmpInUnrel = 0;
mBytesRead = 0;
if (mResampler != NULL) {
mResampler->reset();
}
- mActiveTrack->mState = TrackBase::RESUMING;
+ // FIXME hijacking a playback track state name which was intended for start after pause;
+ // here 'STARTING_2' would be more accurate
+ recordTrack->mState = TrackBase::RESUMING;
// signal thread to start
ALOGV("Signal record thread");
mWaitWorkCV.broadcast();
// do not wait for mStartStopCond if exiting
if (exitPending()) {
- mActiveTrack.clear();
+ mActiveTracks.remove(recordTrack);
+ mActiveTracksGen++;
status = INVALID_OPERATION;
goto startError;
}
+ // FIXME incorrect usage of wait: no explicit predicate or loop
mStartStopCond.wait(mLock);
- if (mActiveTrack == 0) {
+ if (mActiveTracks.indexOf(recordTrack) < 0) {
ALOGV("Record failed to start");
status = BAD_VALUE;
goto startError;
@@ -4877,29 +5073,31 @@
bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
AutoMutex _l(mLock);
- if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
+ if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
return false;
}
+ // note that threadLoop may still be processing the track at this point [without lock]
recordTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
if (exitPending()) {
return true;
}
+ // FIXME incorrect usage of wait: no explicit predicate or loop
mStartStopCond.wait(mLock);
- // if we have been restarted, recordTrack == mActiveTrack.get() here
- if (exitPending() || recordTrack != mActiveTrack.get()) {
+ // if we have been restarted, recordTrack is in mActiveTracks here
+ if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
ALOGV("Record stopped OK");
return true;
}
return false;
}
-bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
{
return false;
}
-status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
{
#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
if (!isValidSyncEvent(event)) {
@@ -4930,7 +5128,7 @@
track->terminate();
track->mState = TrackBase::STOPPED;
// active tracks are removed by threadLoop()
- if (mActiveTrack != track) {
+ if (mActiveTracks.indexOf(track) < 0) {
removeTrack_l(track);
}
}
@@ -4950,104 +5148,110 @@
void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
+ fdprintf(fd, "\nInput thread %p:\n", this);
- snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
- result.append(buffer);
-
- if (mActiveTrack != 0) {
- snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex);
- result.append(buffer);
- snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize);
- result.append(buffer);
- snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
- result.append(buffer);
- snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
- result.append(buffer);
- snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
- result.append(buffer);
+ if (mActiveTracks.size() > 0) {
+ fdprintf(fd, " In index: %zu\n", mRsmpInIndex);
+ fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
+ fdprintf(fd, " Resampling: %d\n", (mResampler != NULL));
+ fdprintf(fd, " Out channel count: %u\n", mReqChannelCount);
+ fdprintf(fd, " Out sample rate: %u\n", mReqSampleRate);
} else {
- result.append("No active record client\n");
+ fdprintf(fd, " No active record client\n");
}
- write(fd, result.string(), result.size());
-
dumpBase(fd, args);
}
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
- result.append(buffer);
- RecordTrack::appendDumpHeader(result);
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<RecordTrack> track = mTracks[i];
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
+ size_t numtracks = mTracks.size();
+ size_t numactive = mActiveTracks.size();
+ size_t numactiveseen = 0;
+ fdprintf(fd, " %d Tracks", numtracks);
+ if (numtracks) {
+ fdprintf(fd, " of which %d are active\n", numactive);
+ RecordTrack::appendDumpHeader(result);
+ for (size_t i = 0; i < numtracks ; ++i) {
+ sp<RecordTrack> track = mTracks[i];
+ if (track != 0) {
+ bool active = mActiveTracks.indexOf(track) >= 0;
+ if (active) {
+ numactiveseen++;
+ }
+ track->dump(buffer, SIZE, active);
+ result.append(buffer);
+ }
}
+ } else {
+ fdprintf(fd, "\n");
}
- if (mActiveTrack != 0) {
- snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
+ if (numactiveseen != numactive) {
+ snprintf(buffer, SIZE, " The following tracks are in the active list but"
+ " not in the track list\n");
result.append(buffer);
RecordTrack::appendDumpHeader(result);
- mActiveTrack->dump(buffer, SIZE);
- result.append(buffer);
+ for (size_t i = 0; i < numactive; ++i) {
+ sp<RecordTrack> track = mActiveTracks[i];
+ if (mTracks.indexOf(track) < 0) {
+ track->dump(buffer, SIZE, true);
+ result.append(buffer);
+ }
+ }
}
write(fd, result.string(), result.size());
}
// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
- size_t framesReq = buffer->frameCount;
- size_t framesReady = mFrameCount - mRsmpInIndex;
- int channelCount;
-
- if (framesReady == 0) {
- mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize);
- if (mBytesRead <= 0) {
- if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
- ALOGE("RecordThread::getNextBuffer() Error reading audio input");
- // Force input into standby so that it tries to
- // recover at next read attempt
- inputStandBy();
- usleep(kRecordThreadSleepUs);
- }
- buffer->raw = NULL;
- buffer->frameCount = 0;
- return NOT_ENOUGH_DATA;
- }
- mRsmpInIndex = 0;
- framesReady = mFrameCount;
+ int32_t rear = mRsmpInRear;
+ int32_t front = mRsmpInFront;
+ ssize_t filled = rear - front;
+ ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2);
+ // 'filled' may be non-contiguous, so return only the first contiguous chunk
+ front &= mRsmpInFramesP2 - 1;
+ size_t part1 = mRsmpInFramesP2 - front;
+ if (part1 > (size_t) filled) {
+ part1 = filled;
+ }
+ size_t ask = buffer->frameCount;
+ ALOG_ASSERT(ask > 0);
+ if (part1 > ask) {
+ part1 = ask;
+ }
+ if (part1 == 0) {
+ // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
+ ALOGE("RecordThread::getNextBuffer() starved");
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ mRsmpInUnrel = 0;
+ return NOT_ENOUGH_DATA;
}
- if (framesReq > framesReady) {
- framesReq = framesReady;
- }
-
- if (mChannelCount == 1 && mReqChannelCount == 2) {
- channelCount = 1;
- } else {
- channelCount = 2;
- }
- buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
- buffer->frameCount = framesReq;
+ buffer->raw = mRsmpInBuffer + front * mChannelCount;
+ buffer->frameCount = part1;
+ mRsmpInUnrel = part1;
return NO_ERROR;
}
// AudioBufferProvider interface
void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
- mRsmpInIndex += buffer->frameCount;
+ size_t stepCount = buffer->frameCount;
+ if (stepCount == 0) {
+ return;
+ }
+ ALOG_ASSERT(stepCount <= mRsmpInUnrel);
+ mRsmpInUnrel -= stepCount;
+ mRsmpInFront += stepCount;
+ buffer->raw = NULL;
buffer->frameCount = 0;
}
@@ -5062,7 +5266,7 @@
int value;
audio_format_t reqFormat = mFormat;
uint32_t reqSamplingRate = mReqSampleRate;
- uint32_t reqChannelCount = mReqChannelCount;
+ audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount);
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reqSamplingRate = value;
@@ -5077,14 +5281,19 @@
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- reqChannelCount = popcount(value);
- reconfig = true;
+ audio_channel_mask_t mask = (audio_channel_mask_t) value;
+ if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
+ status = BAD_VALUE;
+ } else {
+ reqChannelMask = mask;
+ reconfig = true;
+ }
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
- if (mActiveTrack != 0) {
+ if (mActiveTracks.size() > 0) {
status = INVALID_OPERATION;
} else {
reconfig = true;
@@ -5127,6 +5336,7 @@
}
mAudioSource = (audio_source_t)value;
}
+
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common,
keyValuePair.string());
@@ -5143,7 +5353,8 @@
<= (2 * reqSamplingRate)) &&
popcount(mInput->stream->common.get_channels(&mInput->stream->common))
<= FCC_2 &&
- (reqChannelCount <= FCC_2)) {
+ (reqChannelMask == AUDIO_CHANNEL_IN_MONO ||
+ reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
@@ -5177,9 +5388,9 @@
return out_s8;
}
-void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) {
AudioSystem::OutputDescriptor desc;
- void *param2 = NULL;
+ const void *param2 = NULL;
switch (event) {
case AudioSystem::INPUT_OPENED:
@@ -5213,39 +5424,32 @@
mChannelCount = popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+ ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
}
mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
- mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+ // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to
+ // 1 full output buffer, regardless of the alignment of the available input.
+ mRsmpInFrames = mFrameCount * 3;
+ mRsmpInFramesP2 = roundup(mRsmpInFrames);
+ // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
+ mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
+ mRsmpInFront = 0;
+ mRsmpInRear = 0;
+ mRsmpInUnrel = 0;
- if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
- {
- int channelCount;
- // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
- // stereo to mono post process as the resampler always outputs stereo.
- if (mChannelCount == 1 && mReqChannelCount == 2) {
- channelCount = 1;
- } else {
- channelCount = 2;
- }
- mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) {
+ mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+ // resampler always outputs stereo
mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];
-
- // optmization: if mono to mono, alter input frame count as if we were inputing
- // stereo samples
- if (mChannelCount == 1 && mReqChannelCount == 1) {
- mFrameCount >>= 1;
- }
-
}
mRsmpInIndex = mFrameCount;
}
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t AudioFlinger::RecordThread::getInputFramesLost()
{
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index a2fb874..999fea3 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -36,6 +36,8 @@
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
+ virtual status_t readyToRun();
+
void dumpBase(int fd, const Vector<String16>& args);
void dumpEffectChains(int fd, const Vector<String16>& args);
@@ -63,7 +65,7 @@
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(int event, int param) :
- ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+ ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {}
virtual ~IoConfigEvent() {}
int event() const { return mEvent; }
@@ -141,6 +143,7 @@
void sendIoConfigEvent_l(int event, int param = 0);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
void processConfigEvents();
+ void processConfigEvents_l();
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
@@ -156,7 +159,7 @@
int sessionId,
effect_descriptor_t *desc,
int *enabled,
- status_t *status);
+ status_t *status /*non-NULL*/);
void disconnectEffect(const sp< EffectModule>& effect,
EffectHandle *handle,
bool unpinIfLast);
@@ -198,13 +201,13 @@
// effect
void removeEffect_l(const sp< EffectModule>& effect);
// detach all tracks connected to an auxiliary effect
- virtual void detachAuxEffect_l(int effectId) {}
+ virtual void detachAuxEffect_l(int effectId __unused) {}
// returns either EFFECT_SESSION if effects on this audio session exist in one
// chain, or TRACK_SESSION if tracks on this audio session exist, or both
virtual uint32_t hasAudioSession(int sessionId) const = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
- virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+ virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
// suspend or restore effect according to the type of effect passed. a NULL
// type pointer means suspend all effects in the session
@@ -275,6 +278,7 @@
uint32_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
+ size_t mBufferSize; // HAL buffer size for read() or write()
// Parameter sequence by client: binder thread calling setParameters():
// 1. Lock mLock
@@ -303,12 +307,12 @@
Vector<ConfigEvent *> mConfigEvents;
// These fields are written and read by thread itself without lock or barrier,
- // and read by other threads without lock or barrier via standby() , outDevice()
+ // and read by other threads without lock or barrier via standby(), outDevice()
// and inDevice().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
- bool mStandby; // Whether thread is currently in standby.
+ bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
audio_source_t mAudioSource; // (see audio.h, audio_source_t)
@@ -358,7 +362,6 @@
void dump(int fd, const Vector<String16>& args);
// Thread virtuals
- virtual status_t readyToRun();
virtual bool threadLoop();
// RefBase
@@ -391,7 +394,7 @@
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
-
+ virtual void onAddNewTrack_l();
// ThreadBase virtuals
virtual void preExit();
@@ -419,13 +422,13 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int uid,
- status_t *status);
+ status_t *status /*non-NULL*/);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
@@ -479,7 +482,6 @@
size_t mNormalFrameCount; // normal mixer and effects
int16_t* mMixBuffer; // frame size aligned mix buffer
- int8_t* mAllocMixBuffer; // mixer buffer allocation address
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
@@ -623,13 +625,12 @@
sp<NBLog::Writer> mFastMixerNBLogWriter;
public:
virtual bool hasFastMixer() const = 0;
- virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+ virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
- virtual void flushOutput_l();
private:
// timestamp latch:
@@ -748,11 +749,11 @@
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_exit();
- virtual void flushOutput_l();
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
+ virtual void onAddNewTrack_l();
private:
void flushHw_l();
@@ -867,23 +868,23 @@
// Thread virtuals
virtual bool threadLoop();
- virtual status_t readyToRun();
// RefBase
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
+
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
- status_t *status);
+ status_t *status /*non-NULL*/);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
@@ -905,7 +906,7 @@
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
- virtual unsigned int getInputFramesLost();
+ virtual uint32_t getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
@@ -926,30 +927,43 @@
bool hasFastRecorder() const { return false; }
private:
- void clearSyncStartEvent();
+ void clearSyncStartEvent();
// Enter standby if not already in standby, and set mStandby flag
- void standby();
+ void standbyIfNotAlreadyInStandby();
// Call the HAL standby method unconditionally, and don't change mStandby flag
- void inputStandBy();
+ void inputStandBy();
AudioStreamIn *mInput;
SortedVector < sp<RecordTrack> > mTracks;
- // mActiveTrack has dual roles: it indicates the current active track, and
+ // mActiveTracks has dual roles: it indicates the current active track(s), and
// is used together with mStartStopCond to indicate start()/stop() progress
- sp<RecordTrack> mActiveTrack;
+ SortedVector< sp<RecordTrack> > mActiveTracks;
+ // generation counter for mActiveTracks
+ int mActiveTracksGen;
Condition mStartStopCond;
// updated by RecordThread::readInputParameters()
AudioResampler *mResampler;
// interleaved stereo pairs of fixed-point signed Q19.12
int32_t *mRsmpOutBuffer;
- int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount]
- size_t mRsmpInIndex;
- size_t mBufferSize; // stream buffer size for read()
+
+ // resampler converts input at HAL Hz to output at AudioRecord client Hz
+ int16_t *mRsmpInBuffer; // see new[] for details on the size
+ size_t mRsmpInFrames; // size of resampler input in frames
+ size_t mRsmpInFramesP2;// size rounded up to a power-of-2
+ size_t mRsmpInUnrel; // unreleased frames remaining from
+ // most recent getNextBuffer
+ // these are rolling counters that are never cleared
+ int32_t mRsmpInFront; // next available frame
+ int32_t mRsmpInRear; // last filled frame + 1
+ size_t mRsmpInIndex; // FIXME legacy
+
+ // client's requested configuration, which may differ from the HAL configuration
const uint32_t mReqChannelCount;
const uint32_t mReqSampleRate;
+
ssize_t mBytesRead;
// sync event triggering actual audio capture. Frames read before this event will
// be dropped and therefore not read by the application.
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index cd201d9..05fde7c 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -48,6 +48,7 @@
int uid,
bool isOut);
virtual ~TrackBase();
+ virtual status_t initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
virtual status_t start(AudioSystem::sync_event_t event,
int triggerSession) = 0;
@@ -78,15 +79,6 @@
virtual uint32_t sampleRate() const { return mSampleRate; }
- // Return a pointer to the start of a contiguous slice of the track buffer.
- // Parameter 'offset' is the requested start position, expressed in
- // monotonically increasing frame units relative to the track epoch.
- // Parameter 'frames' is the requested length, also in frame units.
- // Always returns non-NULL. It is the caller's responsibility to
- // verify that this will be successful; the result of calling this
- // function with invalid 'offset' or 'frames' is undefined.
- void* getBuffer(uint32_t offset, uint32_t frames) const;
-
bool isStopped() const {
return (mState == STOPPED || mState == FLUSHED);
}
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index fccc7b8..e5152b8 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -116,12 +116,11 @@
if (client != 0) {
mCblkMemory = client->heap()->allocate(size);
- if (mCblkMemory != 0) {
- mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
- // can't assume mCblk != NULL
- } else {
+ if (mCblkMemory == 0 ||
+ (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
ALOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
+ mCblkMemory.clear();
return;
}
} else {
@@ -134,7 +133,6 @@
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
// clear all buffers
- mCblk->frameCount_ = frameCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
@@ -148,7 +146,7 @@
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
- if (pipeFormat != Format_Invalid) {
+ if (Format_isValid(pipeFormat)) {
Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {pipeFormat};
@@ -275,6 +273,11 @@
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
+ if (buffer == 0 || buffer->pointer() == NULL) {
+ ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()");
+ return BAD_VALUE;
+ }
+
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->queueTimedBuffer(buffer, pts);
@@ -344,7 +347,8 @@
mCachedVolume(1.0),
mIsInvalid(false),
mAudioTrackServerProxy(NULL),
- mResumeToStopping(false)
+ mResumeToStopping(false),
+ mFlushHwPending(false)
{
if (mCblk != NULL) {
if (sharedBuffer == 0) {
@@ -396,6 +400,15 @@
}
}
+status_t AudioFlinger::PlaybackThread::Track::initCheck() const
+{
+ status_t status = TrackBase::initCheck();
+ if (status == NO_ERROR && mName < 0) {
+ status = NO_MEMORY;
+ }
+ return status;
+}
+
void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
@@ -422,17 +435,19 @@
/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
- result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate "
+ result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate "
"L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n");
}
-void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
{
uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
if (isFastTrack()) {
- sprintf(buffer, " F %2d", mFastIndex);
+ sprintf(buffer, " F %2d", mFastIndex);
+ } else if (mName >= AudioMixer::TRACK0) {
+ sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
} else {
- sprintf(buffer, " %4d", mName - AudioMixer::TRACK0);
+ sprintf(buffer, " none");
}
track_state state = mState;
char stateChar;
@@ -487,8 +502,9 @@
nowInUnderrun = '?';
break;
}
- snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g "
+ snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g "
"%08X %p %p 0x%03X %9u%c\n",
+ active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
@@ -514,7 +530,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
+ AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
@@ -551,7 +567,7 @@
// Don't call for fast tracks; the framesReady() could result in priority inversion
bool AudioFlinger::PlaybackThread::Track::isReady() const {
- if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
+ if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing() || isStopping()) {
return true;
}
@@ -564,8 +580,8 @@
return false;
}
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
- int triggerSession)
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
+ int triggerSession __unused)
{
status_t status = NO_ERROR;
ALOGV("start(%d), calling pid %d session %d",
@@ -719,6 +735,7 @@
mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
}
+ mFlushHwPending = true;
mResumeToStopping = false;
} else {
if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
@@ -739,11 +756,19 @@
// Prevent flush being lost if the track is flushed and then resumed
// before mixer thread can run. This is important when offloading
// because the hardware buffer could hold a large amount of audio
- playbackThread->flushOutput_l();
playbackThread->broadcast_l();
}
}
+// must be called with thread lock held
+void AudioFlinger::PlaybackThread::Track::flushAck()
+{
+ if (!isOffloaded())
+ return;
+
+ mFlushHwPending = false;
+}
+
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
@@ -979,7 +1004,8 @@
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
- int uid) {
+ int uid)
+{
if (!client->reserveTimedTrack())
return 0;
@@ -1045,15 +1071,14 @@
mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
"AudioFlingerTimed");
- if (mTimedMemoryDealer == NULL)
+ if (mTimedMemoryDealer == NULL) {
return NO_MEMORY;
+ }
}
sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
- if (newBuffer == NULL) {
- newBuffer = mTimedMemoryDealer->allocate(size);
- if (newBuffer == NULL)
- return NO_MEMORY;
+ if (newBuffer == 0 || newBuffer->pointer() == NULL) {
+ return NO_MEMORY;
}
*buffer = newBuffer;
@@ -1152,7 +1177,7 @@
void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
const TimedBuffer& buf,
- const char* logTag) {
+ const char* logTag __unused) {
uint32_t bufBytes = buf.buffer()->size();
uint32_t consumedAlready = buf.position();
@@ -1504,9 +1529,9 @@
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
- "mCblk->frameCount_ %u, mChannelMask 0x%08x",
+ "frameCount %u, mChannelMask 0x%08x",
mCblk, mBuffer,
- mCblk->frameCount_, mChannelMask);
+ frameCount, mChannelMask);
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
@@ -1764,9 +1789,7 @@
{
ALOGV("RecordTrack constructor");
if (mCblk != NULL) {
- mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
- mFrameSize);
- mServerProxy = mAudioRecordServerProxy;
+ mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
}
}
@@ -1777,7 +1800,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts)
+ int64_t pts __unused)
{
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
@@ -1845,12 +1868,13 @@
/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
- result.append("Client Fmt Chn mask Session S Server fCount\n");
+ result.append(" Active Client Fmt Chn mask Session S Server fCount\n");
}
-void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
{
- snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6zu\n",
+ snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu\n",
+ active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 7a314cf..66fcd90 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -26,54 +26,30 @@
#include <errno.h>
#include <time.h>
#include <math.h>
+#include <audio_utils/sndfile.h>
using namespace android;
-struct HeaderWav {
- HeaderWav(size_t size, int nc, int sr, int bits) {
- strncpy(RIFF, "RIFF", 4);
- chunkSize = size + sizeof(HeaderWav);
- strncpy(WAVE, "WAVE", 4);
- strncpy(fmt, "fmt ", 4);
- fmtSize = 16;
- audioFormat = 1;
- numChannels = nc;
- samplesRate = sr;
- byteRate = sr * numChannels * (bits/8);
- align = nc*(bits/8);
- bitsPerSample = bits;
- strncpy(data, "data", 4);
- dataSize = size;
- }
-
- char RIFF[4]; // RIFF
- uint32_t chunkSize; // File size
- char WAVE[4]; // WAVE
- char fmt[4]; // fmt\0
- uint32_t fmtSize; // fmt size
- uint16_t audioFormat; // 1=PCM
- uint16_t numChannels; // num channels
- uint32_t samplesRate; // sample rate in hz
- uint32_t byteRate; // Bps
- uint16_t align; // 2=16-bit mono, 4=16-bit stereo
- uint16_t bitsPerSample; // bits per sample
- char data[4]; // "data"
- uint32_t dataSize; // size
-};
+bool gVerbose = false;
static int usage(const char* name) {
- fprintf(stderr,"Usage: %s [-p] [-h] [-s] [-q {dq|lq|mq|hq|vhq}] [-i input-sample-rate] "
- "[-o output-sample-rate] [<input-file>] <output-file>\n", name);
+ fprintf(stderr,"Usage: %s [-p] [-h] [-v] [-s] [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+ " [-i input-sample-rate] [-o output-sample-rate] [<input-file>]"
+ " <output-file>\n", name);
fprintf(stderr," -p enable profiling\n");
fprintf(stderr," -h create wav file\n");
- fprintf(stderr," -s stereo\n");
+ fprintf(stderr," -v verbose : log buffer provider calls\n");
+ fprintf(stderr," -s stereo (ignored if input file is specified)\n");
fprintf(stderr," -q resampler quality\n");
fprintf(stderr," dq : default quality\n");
fprintf(stderr," lq : low quality\n");
fprintf(stderr," mq : medium quality\n");
fprintf(stderr," hq : high quality\n");
fprintf(stderr," vhq : very high quality\n");
- fprintf(stderr," -i input file sample rate\n");
+ fprintf(stderr," dlq : dynamic low quality\n");
+ fprintf(stderr," dmq : dynamic medium quality\n");
+ fprintf(stderr," dhq : dynamic high quality\n");
+ fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n");
fprintf(stderr," -o output file sample rate\n");
return -1;
}
@@ -81,7 +57,8 @@
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
- bool profiling = false;
+ bool profileResample = false;
+ bool profileFilter = false;
bool writeHeader = false;
int channels = 1;
int input_freq = 0;
@@ -89,14 +66,20 @@
AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
int ch;
- while ((ch = getopt(argc, argv, "phsq:i:o:")) != -1) {
+ while ((ch = getopt(argc, argv, "pfhvsq:i:o:")) != -1) {
switch (ch) {
case 'p':
- profiling = true;
+ profileResample = true;
+ break;
+ case 'f':
+ profileFilter = true;
break;
case 'h':
writeHeader = true;
break;
+ case 'v':
+ gVerbose = true;
+ break;
case 's':
channels = 2;
break;
@@ -111,6 +94,12 @@
quality = AudioResampler::HIGH_QUALITY;
else if (!strcmp(optarg, "vhq"))
quality = AudioResampler::VERY_HIGH_QUALITY;
+ else if (!strcmp(optarg, "dlq"))
+ quality = AudioResampler::DYN_LOW_QUALITY;
+ else if (!strcmp(optarg, "dmq"))
+ quality = AudioResampler::DYN_MED_QUALITY;
+ else if (!strcmp(optarg, "dhq"))
+ quality = AudioResampler::DYN_HIGH_QUALITY;
else {
usage(progname);
return -1;
@@ -148,25 +137,22 @@
size_t input_size;
void* input_vaddr;
if (argc == 2) {
- struct stat st;
- if (stat(file_in, &st) < 0) {
- fprintf(stderr, "stat: %s\n", strerror(errno));
- return -1;
+ SF_INFO info;
+ info.format = 0;
+ SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+ if (sf == NULL) {
+ perror(file_in);
+ return EXIT_FAILURE;
}
-
- int input_fd = open(file_in, O_RDONLY);
- if (input_fd < 0) {
- fprintf(stderr, "open: %s\n", strerror(errno));
- return -1;
- }
-
- input_size = st.st_size;
- input_vaddr = mmap(0, input_size, PROT_READ, MAP_PRIVATE, input_fd, 0);
- if (input_vaddr == MAP_FAILED ) {
- fprintf(stderr, "mmap: %s\n", strerror(errno));
- return -1;
- }
+ input_size = info.frames * info.channels * sizeof(short);
+ input_vaddr = malloc(input_size);
+ (void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
+ sf_close(sf);
+ channels = info.channels;
+ input_freq = info.samplerate;
} else {
+ // data for testing is exactly (input sampling rate/1000)/2 seconds
+ // so 44.1khz input is 22.05 seconds
double k = 1000; // Hz / s
double time = (input_freq / 2) / k;
size_t input_frames = size_t(input_freq * time);
@@ -178,7 +164,7 @@
double y = sin(M_PI * k * t * t);
int16_t yi = floor(y * 32767.0 + 0.5);
for (size_t j=0 ; j<(size_t)channels ; j++) {
- in[i*channels + j] = yi / (1+j);
+ in[i*channels + j] = yi / (1+j); // right ch. 1/2 left ch.
}
}
}
@@ -186,89 +172,238 @@
// ----------------------------------------------------------
class Provider: public AudioBufferProvider {
- int16_t* mAddr;
- size_t mNumFrames;
+ int16_t* const mAddr; // base address
+ const size_t mNumFrames; // total frames
+ const int mChannels;
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
public:
- Provider(const void* addr, size_t size, int channels) {
- mAddr = (int16_t*) addr;
- mNumFrames = size / (channels*sizeof(int16_t));
+ Provider(const void* addr, size_t size, int channels)
+ : mAddr((int16_t*) addr),
+ mNumFrames(size / (channels*sizeof(int16_t))),
+ mChannels(channels),
+ mNextFrame(0), mUnrel(0) {
}
virtual status_t getNextBuffer(Buffer* buffer,
int64_t pts = kInvalidPTS) {
- buffer->frameCount = mNumFrames;
- buffer->i16 = mAddr;
- return NO_ERROR;
+ (void)pts; // suppress warning
+ size_t requestedFrames = buffer->frameCount;
+ if (requestedFrames > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ if (gVerbose) {
+ printf("getNextBuffer() requested %u frames out of %u frames available,"
+ " and returned %u frames\n",
+ requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+ }
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->i16 = &mAddr[mChannels * mNextFrame];
+ return NO_ERROR;
+ } else {
+ buffer->i16 = NULL;
+ return NOT_ENOUGH_DATA;
+ }
}
virtual void releaseBuffer(Buffer* buffer) {
+ if (buffer->frameCount > mUnrel) {
+ fprintf(stderr, "ERROR releaseBuffer() released %u frames but only %u available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+ if (gVerbose) {
+ printf("releaseBuffer() released %u frames out of %u frames available "
+ "to release\n", buffer->frameCount, mUnrel);
+ }
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->i16 = NULL;
+ }
+ void reset() {
+ mNextFrame = 0;
}
} provider(input_vaddr, input_size, channels);
size_t input_frames = input_size / (channels * sizeof(int16_t));
+ if (gVerbose) {
+ printf("%u input frames\n", input_frames);
+ }
size_t output_size = 2 * 4 * ((int64_t) input_frames * output_freq) / input_freq;
output_size &= ~7; // always stereo, 32-bits
- void* output_vaddr = malloc(output_size);
-
- if (profiling) {
+ if (profileFilter) {
+ // Check how fast sample rate changes are that require filter changes.
+ // The delta sample rate changes must indicate a downsampling ratio,
+ // and must be larger than 10% changes.
+ //
+ // On fast devices, filters should be generated between 0.1ms - 1ms.
+ // (single threaded).
AudioResampler* resampler = AudioResampler::create(16, channels,
- output_freq, quality);
-
- size_t out_frames = output_size/8;
- resampler->setSampleRate(input_freq);
- resampler->setVolume(0x1000, 0x1000);
-
- memset(output_vaddr, 0, output_size);
+ 8000, quality);
+ int looplimit = 100;
timespec start, end;
clock_gettime(CLOCK_MONOTONIC, &start);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
- resampler->resample((int*) output_vaddr, out_frames, &provider);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->setSampleRate(9000);
+ resampler->setSampleRate(12000);
+ resampler->setSampleRate(20000);
+ resampler->setSampleRate(30000);
+ }
clock_gettime(CLOCK_MONOTONIC, &end);
int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
- int64_t time = (end_ns - start_ns)/4;
- printf("%f Mspl/s\n", out_frames/(time/1e9)/1e6);
+ int64_t time = end_ns - start_ns;
+ printf("%.2f sample rate changes with filter calculation/sec\n",
+ looplimit * 4 / (time / 1e9));
+ // Check how fast sample rate changes are without filter changes.
+ // This should be very fast, probably 0.1us - 1us per sample rate
+ // change.
+ resampler->setSampleRate(1000);
+ looplimit = 1000;
+ clock_gettime(CLOCK_MONOTONIC, &start);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->setSampleRate(1000+i);
+ }
+ clock_gettime(CLOCK_MONOTONIC, &end);
+ start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+ end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+ time = end_ns - start_ns;
+ printf("%.2f sample rate changes without filter calculation/sec\n",
+ looplimit / (time / 1e9));
+ resampler->reset();
delete resampler;
}
+ void* output_vaddr = malloc(output_size);
AudioResampler* resampler = AudioResampler::create(16, channels,
output_freq, quality);
size_t out_frames = output_size/8;
+
+ /* set volume precision to 12 bits, so the volume scale is 1<<12.
+ * This means the "integer" part fits in the Q19.12 precision
+ * representation of output int32_t.
+ *
+ * Generally 0 < volumePrecision <= 14 (due to the limits of
+ * int16_t values for Volume). volumePrecision cannot be 0 due
+ * to rounding and shifts.
+ */
+ const int volumePrecision = 12; // in bits
+
resampler->setSampleRate(input_freq);
- resampler->setVolume(0x1000, 0x1000);
+ resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
+
+ if (profileResample) {
+ /*
+ * For profiling on mobile devices, upon experimentation
+ * it is better to run a few trials with a shorter loop limit,
+ * and take the minimum time.
+ *
+ * Long tests can cause CPU temperature to build up and thermal throttling
+ * to reduce CPU frequency.
+ *
+ * For frequency checks (index=0, or 1, etc.):
+ * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+ *
+ * For temperature checks (index=0, or 1, etc.):
+ * "cat /sys/class/thermal/thermal_zone${index}/temp"
+ *
+ * Another way to avoid thermal throttling is to fix the CPU frequency
+ * at a lower level which prevents excessive temperatures.
+ */
+ const int trials = 4;
+ const int looplimit = 4;
+ timespec start, end;
+ int64_t time;
+
+ for (int n = 0; n < trials; ++n) {
+ clock_gettime(CLOCK_MONOTONIC, &start);
+ for (int i = 0; i < looplimit; ++i) {
+ resampler->resample((int*) output_vaddr, out_frames, &provider);
+ provider.reset(); // during benchmarking reset only the provider
+ }
+ clock_gettime(CLOCK_MONOTONIC, &end);
+ int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+ int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+ int64_t diff_ns = end_ns - start_ns;
+ if (n == 0 || diff_ns < time) {
+ time = diff_ns; // save the best out of our trials.
+ }
+ }
+ // Mfrms/s is "Millions of output frames per second".
+ printf("quality: %d channels: %d msec: %lld Mfrms/s: %.2lf\n",
+ quality, channels, time/1000000, out_frames * looplimit / (time / 1e9) / 1e6);
+ resampler->reset();
+ }
memset(output_vaddr, 0, output_size);
+ if (gVerbose) {
+ printf("resample() %u output frames\n", out_frames);
+ }
resampler->resample((int*) output_vaddr, out_frames, &provider);
+ if (gVerbose) {
+ printf("resample() complete\n");
+ }
+ resampler->reset();
+ if (gVerbose) {
+ printf("reset() complete\n");
+ }
+ delete resampler;
+ resampler = NULL;
- // down-mix (we just truncate and keep the left channel)
+ // mono takes left channel only
+ // stereo right channel is half amplitude of stereo left channel (due to input creation)
int32_t* out = (int32_t*) output_vaddr;
int16_t* convert = (int16_t*) malloc(out_frames * channels * sizeof(int16_t));
+
+ // round to half towards zero and saturate at int16 (non-dithered)
+ const int roundVal = (1<<(volumePrecision-1)) - 1; // volumePrecision > 0
+
for (size_t i = 0; i < out_frames; i++) {
- for (int j=0 ; j<channels ; j++) {
- int32_t s = out[i * 2 + j] >> 12;
- if (s > 32767) s = 32767;
- else if (s < -32768) s = -32768;
+ for (int j = 0; j < channels; j++) {
+ int32_t s = out[i * 2 + j] + roundVal; // add offset here
+ if (s < 0) {
+ s = (s + 1) >> volumePrecision; // round to 0
+ if (s < -32768) {
+ s = -32768;
+ }
+ } else {
+ s = s >> volumePrecision;
+ if (s > 32767) {
+ s = 32767;
+ }
+ }
convert[i * channels + j] = int16_t(s);
}
}
// write output to disk
- int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
- S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
- if (output_fd < 0) {
- fprintf(stderr, "open: %s\n", strerror(errno));
- return -1;
- }
-
if (writeHeader) {
- HeaderWav wav(out_frames * channels * sizeof(int16_t), channels, output_freq, 16);
- write(output_fd, &wav, sizeof(wav));
+ SF_INFO info;
+ info.frames = 0;
+ info.samplerate = output_freq;
+ info.channels = channels;
+ info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+ SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
+ if (sf == NULL) {
+ perror(file_out);
+ return EXIT_FAILURE;
+ }
+ (void) sf_writef_short(sf, convert, out_frames);
+ sf_close(sf);
+ } else {
+ int output_fd = open(file_out, O_WRONLY | O_CREAT | O_TRUNC,
+ S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH);
+ if (output_fd < 0) {
+ perror(file_out);
+ return EXIT_FAILURE;
+ }
+ write(output_fd, convert, out_frames * channels * sizeof(int16_t));
+ close(output_fd);
}
- write(output_fd, convert, out_frames * channels * sizeof(int16_t));
- close(output_fd);
-
- return 0;
+ return EXIT_SUCCESS;
}
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 87027f7..79fbf76 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -32,6 +32,7 @@
#include <gui/Surface.h>
#include <hardware/hardware.h>
#include <media/AudioSystem.h>
+#include <media/IMediaHTTPService.h>
#include <media/mediaplayer.h>
#include <utils/Errors.h>
#include <utils/Log.h>
@@ -866,7 +867,7 @@
MediaPlayer* CameraService::newMediaPlayer(const char *file) {
MediaPlayer* mp = new MediaPlayer();
- if (mp->setDataSource(file, NULL) == NO_ERROR) {
+ if (mp->setDataSource(NULL /* httpService */, file, NULL) == NO_ERROR) {
mp->setAudioStreamType(AUDIO_STREAM_ENFORCED_AUDIBLE);
mp->prepare();
} else {
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index ba1e772..0a88a75 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -238,7 +238,7 @@
result.append(" Scene mode: ");
switch (p.sceneMode) {
- case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+ case ANDROID_CONTROL_SCENE_MODE_DISABLED:
result.append("AUTO\n"); break;
CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_ACTION)
CASE_APPEND_ENUM(ANDROID_CONTROL_SCENE_MODE_PORTRAIT)
@@ -732,6 +732,7 @@
return OK;
}
params.state = Parameters::STOPPED;
+ int lastPreviewStreamId = mStreamingProcessor->getPreviewStreamId();
res = mStreamingProcessor->updatePreviewStream(params);
if (res != OK) {
@@ -740,6 +741,8 @@
return res;
}
+ bool previewStreamChanged = mStreamingProcessor->getPreviewStreamId() != lastPreviewStreamId;
+
// We could wait to create the JPEG output stream until first actual use
// (first takePicture call). However, this would substantially increase the
// first capture latency on HAL3 devices, and potentially on some HAL2
@@ -789,6 +792,19 @@
return res;
}
outputStreams.push(getCallbackStreamId());
+ } else if (previewStreamChanged && mCallbackProcessor->getStreamId() != NO_STREAM) {
+ /**
+ * Delete the unused callback stream when preview stream is changed and
+ * preview is not enabled. Don't need stop preview stream as preview is in
+ * STOPPED state now.
+ */
+ ALOGV("%s: Camera %d: Delete unused preview callback stream.", __FUNCTION__, mCameraId);
+ res = mCallbackProcessor->deleteStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete callback stream %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
+ return res;
+ }
}
if (params.zslMode && !params.recordingHint) {
res = updateProcessorStream(mZslProcessor, params);
@@ -798,6 +814,8 @@
return res;
}
outputStreams.push(getZslStreamId());
+ } else {
+ mZslProcessor->deleteStream();
}
outputStreams.push(getPreviewStreamId());
@@ -1144,7 +1162,7 @@
* Handle quirk mode for AF in scene modes
*/
if (l.mParameters.quirks.triggerAfWithAuto &&
- l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED &&
+ l.mParameters.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED &&
l.mParameters.focusMode != Parameters::FOCUS_MODE_AUTO &&
!l.mParameters.focusingAreas[0].isEmpty()) {
ALOGV("%s: Quirk: Switching from focusMode %d to AUTO",
diff --git a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
index 77d5c8a..ec81456 100644
--- a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
@@ -200,51 +200,60 @@
ATRACE_CALL();
status_t res;
sp<Camera2Heap> captureHeap;
+ sp<MemoryBase> captureBuffer;
CpuConsumer::LockedBuffer imgBuffer;
- res = mCaptureConsumer->lockNextBuffer(&imgBuffer);
- if (res != OK) {
- if (res != BAD_VALUE) {
- ALOGE("%s: Camera %d: Error receiving still image buffer: "
- "%s (%d)", __FUNCTION__,
- mId, strerror(-res), res);
+ {
+ Mutex::Autolock l(mInputMutex);
+ if (mCaptureStreamId == NO_STREAM) {
+ ALOGW("%s: Camera %d: No stream is available", __FUNCTION__, mId);
+ return INVALID_OPERATION;
}
- return res;
- }
- ALOGV("%s: Camera %d: Still capture available", __FUNCTION__,
- mId);
+ res = mCaptureConsumer->lockNextBuffer(&imgBuffer);
+ if (res != OK) {
+ if (res != BAD_VALUE) {
+ ALOGE("%s: Camera %d: Error receiving still image buffer: "
+ "%s (%d)", __FUNCTION__,
+ mId, strerror(-res), res);
+ }
+ return res;
+ }
- if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) {
- ALOGE("%s: Camera %d: Unexpected format for still image: "
- "%x, expected %x", __FUNCTION__, mId,
- imgBuffer.format,
- HAL_PIXEL_FORMAT_BLOB);
+ ALOGV("%s: Camera %d: Still capture available", __FUNCTION__,
+ mId);
+
+ if (imgBuffer.format != HAL_PIXEL_FORMAT_BLOB) {
+ ALOGE("%s: Camera %d: Unexpected format for still image: "
+ "%x, expected %x", __FUNCTION__, mId,
+ imgBuffer.format,
+ HAL_PIXEL_FORMAT_BLOB);
+ mCaptureConsumer->unlockBuffer(imgBuffer);
+ return OK;
+ }
+
+ // Find size of JPEG image
+ size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width);
+ if (jpegSize == 0) { // failed to find size, default to whole buffer
+ jpegSize = imgBuffer.width;
+ }
+ size_t heapSize = mCaptureHeap->getSize();
+ if (jpegSize > heapSize) {
+ ALOGW("%s: JPEG image is larger than expected, truncating "
+ "(got %d, expected at most %d bytes)",
+ __FUNCTION__, jpegSize, heapSize);
+ jpegSize = heapSize;
+ }
+
+ // TODO: Optimize this to avoid memcopy
+ captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize);
+ void* captureMemory = mCaptureHeap->getBase();
+ memcpy(captureMemory, imgBuffer.data, jpegSize);
+
mCaptureConsumer->unlockBuffer(imgBuffer);
- return OK;
}
- // Find size of JPEG image
- size_t jpegSize = findJpegSize(imgBuffer.data, imgBuffer.width);
- if (jpegSize == 0) { // failed to find size, default to whole buffer
- jpegSize = imgBuffer.width;
- }
- size_t heapSize = mCaptureHeap->getSize();
- if (jpegSize > heapSize) {
- ALOGW("%s: JPEG image is larger than expected, truncating "
- "(got %d, expected at most %d bytes)",
- __FUNCTION__, jpegSize, heapSize);
- jpegSize = heapSize;
- }
-
- // TODO: Optimize this to avoid memcopy
- sp<MemoryBase> captureBuffer = new MemoryBase(mCaptureHeap, 0, jpegSize);
- void* captureMemory = mCaptureHeap->getBase();
- memcpy(captureMemory, imgBuffer.data, jpegSize);
-
- mCaptureConsumer->unlockBuffer(imgBuffer);
-
sp<CaptureSequencer> sequencer = mSequencer.promote();
if (sequencer != 0) {
sequencer->onCaptureAvailable(imgBuffer.timestamp, captureBuffer);
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 08af566..2cf0d29 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -466,7 +466,7 @@
supportedAntibanding);
}
- sceneMode = ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+ sceneMode = ANDROID_CONTROL_SCENE_MODE_DISABLED;
params.set(CameraParameters::KEY_SCENE_MODE,
CameraParameters::SCENE_MODE_AUTO);
@@ -482,7 +482,7 @@
if (addComma) supportedSceneModes += ",";
addComma = true;
switch (availableSceneModes.data.u8[i]) {
- case ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED:
+ case ANDROID_CONTROL_SCENE_MODE_DISABLED:
noSceneModes = true;
break;
case ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY:
@@ -664,13 +664,13 @@
focusState = ANDROID_CONTROL_AF_STATE_INACTIVE;
shadowFocusMode = FOCUS_MODE_INVALID;
- camera_metadata_ro_entry_t max3aRegions =
- staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1);
- if (!max3aRegions.count) return NO_INIT;
+ camera_metadata_ro_entry_t max3aRegions = staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION);
+ if (max3aRegions.count != Parameters::NUM_REGION) return NO_INIT;
int32_t maxNumFocusAreas = 0;
if (focusMode != Parameters::FOCUS_MODE_FIXED) {
- maxNumFocusAreas = max3aRegions.data.i32[0];
+ maxNumFocusAreas = max3aRegions.data.i32[Parameters::REGION_AF];
}
params.set(CameraParameters::KEY_MAX_NUM_FOCUS_AREAS, maxNumFocusAreas);
params.set(CameraParameters::KEY_FOCUS_AREAS,
@@ -730,7 +730,7 @@
meteringAreas.add(Parameters::Area(0, 0, 0, 0, 0));
params.set(CameraParameters::KEY_MAX_NUM_METERING_AREAS,
- max3aRegions.data.i32[0]);
+ max3aRegions.data.i32[Parameters::REGION_AE]);
params.set(CameraParameters::KEY_METERING_AREAS,
"(0,0,0,0,0)");
@@ -1446,7 +1446,7 @@
newParams.get(CameraParameters::KEY_SCENE_MODE) );
if (validatedParams.sceneMode != sceneMode &&
validatedParams.sceneMode !=
- ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED) {
+ ANDROID_CONTROL_SCENE_MODE_DISABLED) {
camera_metadata_ro_entry_t availableSceneModes =
staticInfo(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
for (i = 0; i < availableSceneModes.count; i++) {
@@ -1461,7 +1461,7 @@
}
}
bool sceneModeSet =
- validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+ validatedParams.sceneMode != ANDROID_CONTROL_SCENE_MODE_DISABLED;
// FLASH_MODE
if (sceneModeSet) {
@@ -1591,10 +1591,11 @@
// FOCUS_AREAS
res = parseAreas(newParams.get(CameraParameters::KEY_FOCUS_AREAS),
&validatedParams.focusingAreas);
- size_t max3aRegions =
- (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1).data.i32[0];
+ size_t maxAfRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AF];
if (res == OK) res = validateAreas(validatedParams.focusingAreas,
- max3aRegions, AREA_KIND_FOCUS);
+ maxAfRegions, AREA_KIND_FOCUS);
if (res != OK) {
ALOGE("%s: Requested focus areas are malformed: %s",
__FUNCTION__, newParams.get(CameraParameters::KEY_FOCUS_AREAS));
@@ -1624,10 +1625,13 @@
newParams.get(CameraParameters::KEY_AUTO_WHITEBALANCE_LOCK));
// METERING_AREAS
+ size_t maxAeRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AE];
res = parseAreas(newParams.get(CameraParameters::KEY_METERING_AREAS),
&validatedParams.meteringAreas);
if (res == OK) {
- res = validateAreas(validatedParams.meteringAreas, max3aRegions,
+ res = validateAreas(validatedParams.meteringAreas, maxAeRegions,
AREA_KIND_METERING);
}
if (res != OK) {
@@ -1776,7 +1780,7 @@
// (face detection statistics and face priority scene mode). Map from other
// to the other.
bool sceneModeActive =
- sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+ sceneMode != (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
uint8_t reqControlMode = ANDROID_CONTROL_MODE_AUTO;
if (enableFaceDetect || sceneModeActive) {
reqControlMode = ANDROID_CONTROL_MODE_USE_SCENE_MODE;
@@ -1788,7 +1792,7 @@
uint8_t reqSceneMode =
sceneModeActive ? sceneMode :
enableFaceDetect ? (uint8_t)ANDROID_CONTROL_SCENE_MODE_FACE_PRIORITY :
- (uint8_t)ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED;
+ (uint8_t)ANDROID_CONTROL_SCENE_MODE_DISABLED;
res = request->update(ANDROID_CONTROL_SCENE_MODE,
&reqSceneMode, 1);
if (res != OK) return res;
@@ -1858,23 +1862,23 @@
size_t reqFocusingAreasSize = focusingAreas.size() * 5;
int32_t *reqFocusingAreas = new int32_t[reqFocusingAreasSize];
- for (size_t i = 0; i < reqFocusingAreasSize; i += 5) {
- if (focusingAreas[i].weight != 0) {
+ for (size_t i = 0, j = 0; i < reqFocusingAreasSize; i += 5, j++) {
+ if (focusingAreas[j].weight != 0) {
reqFocusingAreas[i + 0] =
- normalizedXToArray(focusingAreas[i].left);
+ normalizedXToArray(focusingAreas[j].left);
reqFocusingAreas[i + 1] =
- normalizedYToArray(focusingAreas[i].top);
+ normalizedYToArray(focusingAreas[j].top);
reqFocusingAreas[i + 2] =
- normalizedXToArray(focusingAreas[i].right);
+ normalizedXToArray(focusingAreas[j].right);
reqFocusingAreas[i + 3] =
- normalizedYToArray(focusingAreas[i].bottom);
+ normalizedYToArray(focusingAreas[j].bottom);
} else {
reqFocusingAreas[i + 0] = 0;
reqFocusingAreas[i + 1] = 0;
reqFocusingAreas[i + 2] = 0;
reqFocusingAreas[i + 3] = 0;
}
- reqFocusingAreas[i + 4] = focusingAreas[i].weight;
+ reqFocusingAreas[i + 4] = focusingAreas[j].weight;
}
res = request->update(ANDROID_CONTROL_AF_REGIONS,
reqFocusingAreas, reqFocusingAreasSize);
@@ -1887,28 +1891,45 @@
size_t reqMeteringAreasSize = meteringAreas.size() * 5;
int32_t *reqMeteringAreas = new int32_t[reqMeteringAreasSize];
- for (size_t i = 0; i < reqMeteringAreasSize; i += 5) {
- if (meteringAreas[i].weight != 0) {
+ for (size_t i = 0, j = 0; i < reqMeteringAreasSize; i += 5, j++) {
+ if (meteringAreas[j].weight != 0) {
reqMeteringAreas[i + 0] =
- normalizedXToArray(meteringAreas[i].left);
+ normalizedXToArray(meteringAreas[j].left);
reqMeteringAreas[i + 1] =
- normalizedYToArray(meteringAreas[i].top);
+ normalizedYToArray(meteringAreas[j].top);
reqMeteringAreas[i + 2] =
- normalizedXToArray(meteringAreas[i].right);
+ normalizedXToArray(meteringAreas[j].right);
reqMeteringAreas[i + 3] =
- normalizedYToArray(meteringAreas[i].bottom);
+ normalizedYToArray(meteringAreas[j].bottom);
} else {
reqMeteringAreas[i + 0] = 0;
reqMeteringAreas[i + 1] = 0;
reqMeteringAreas[i + 2] = 0;
reqMeteringAreas[i + 3] = 0;
}
- reqMeteringAreas[i + 4] = meteringAreas[i].weight;
+ reqMeteringAreas[i + 4] = meteringAreas[j].weight;
}
res = request->update(ANDROID_CONTROL_AE_REGIONS,
reqMeteringAreas, reqMeteringAreasSize);
if (res != OK) return res;
+ // Set awb regions to be the same as the metering regions if allowed
+ size_t maxAwbRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AWB];
+ if (maxAwbRegions > 0) {
+ if (maxAwbRegions >= meteringAreas.size()) {
+ res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+ reqMeteringAreas, reqMeteringAreasSize);
+ } else {
+ // Ensure the awb regions are zeroed if the region count is too high.
+ int32_t zeroedAwbAreas[5] = {0, 0, 0, 0, 0};
+ res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+ zeroedAwbAreas, sizeof(zeroedAwbAreas)/sizeof(int32_t));
+ }
+ if (res != OK) return res;
+ }
+
delete[] reqMeteringAreas;
/* don't include jpeg thumbnail size - it's valid for
@@ -2149,9 +2170,9 @@
int Parameters::sceneModeStringToEnum(const char *sceneMode) {
return
!sceneMode ?
- ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+ ANDROID_CONTROL_SCENE_MODE_DISABLED :
!strcmp(sceneMode, CameraParameters::SCENE_MODE_AUTO) ?
- ANDROID_CONTROL_SCENE_MODE_UNSUPPORTED :
+ ANDROID_CONTROL_SCENE_MODE_DISABLED :
!strcmp(sceneMode, CameraParameters::SCENE_MODE_ACTION) ?
ANDROID_CONTROL_SCENE_MODE_ACTION :
!strcmp(sceneMode, CameraParameters::SCENE_MODE_PORTRAIT) ?
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 32dbd42..60c4687 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -113,6 +113,14 @@
bool autoExposureLock;
bool autoWhiteBalanceLock;
+ // 3A region types, for use with ANDROID_CONTROL_MAX_REGIONS
+ enum region_t {
+ REGION_AE = 0,
+ REGION_AWB,
+ REGION_AF,
+ NUM_REGION // Number of region types
+ } region;
+
Vector<Area> meteringAreas;
int zoom;
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
index 6b4e57a..c1d0496 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
@@ -273,6 +273,15 @@
return INVALID_OPERATION;
}
+ // Flush device to clear out all in-flight requests pending in HAL.
+ res = client->getCameraDevice()->flush();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Failed to flush device: "
+ "%s (%d)",
+ __FUNCTION__, client->getCameraId(), strerror(-res), res);
+ return res;
+ }
+
// Update JPEG settings
{
SharedParameters::Lock l(client->getParameters());
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 1cdf8dc..187220e 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -635,26 +635,56 @@
return INVALID_OPERATION;
}
+ camera_metadata_ro_entry_t entryFacing = staticInfo.find(ANDROID_LENS_FACING);
+ if (entry.count == 0) {
+ ALOGE("%s: Camera %d: Can't find android.lens.facing in "
+ "static metadata!", __FUNCTION__, mCameraId);
+ return INVALID_OPERATION;
+ }
+
int32_t& flags = *transform;
+ bool mirror = (entryFacing.data.u8[0] == ANDROID_LENS_FACING_FRONT);
int orientation = entry.data.i32[0];
- switch (orientation) {
- case 0:
- flags = 0;
- break;
- case 90:
- flags = NATIVE_WINDOW_TRANSFORM_ROT_90;
- break;
- case 180:
- flags = NATIVE_WINDOW_TRANSFORM_ROT_180;
- break;
- case 270:
- flags = NATIVE_WINDOW_TRANSFORM_ROT_270;
- break;
- default:
- ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
- __FUNCTION__, orientation);
- return INVALID_OPERATION;
+ if (!mirror) {
+ switch (orientation) {
+ case 0:
+ flags = 0;
+ break;
+ case 90:
+ flags = NATIVE_WINDOW_TRANSFORM_ROT_90;
+ break;
+ case 180:
+ flags = NATIVE_WINDOW_TRANSFORM_ROT_180;
+ break;
+ case 270:
+ flags = NATIVE_WINDOW_TRANSFORM_ROT_270;
+ break;
+ default:
+ ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
+ __FUNCTION__, orientation);
+ return INVALID_OPERATION;
+ }
+ } else {
+ switch (orientation) {
+ case 0:
+ flags = HAL_TRANSFORM_FLIP_H;
+ break;
+ case 90:
+ flags = HAL_TRANSFORM_FLIP_H | HAL_TRANSFORM_ROT_90;
+ break;
+ case 180:
+ flags = HAL_TRANSFORM_FLIP_V;
+ break;
+ case 270:
+ flags = HAL_TRANSFORM_FLIP_V | HAL_TRANSFORM_ROT_90;
+ break;
+ default:
+ ALOGE("%s: Invalid HAL android.sensor.orientation value: %d",
+ __FUNCTION__, orientation);
+ return INVALID_OPERATION;
+ }
+
}
/**
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 3dbc1b0..da3e121 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -100,8 +100,10 @@
camera3_device_t *device;
+ ATRACE_BEGIN("camera3->open");
res = module->common.methods->open(&module->common, deviceName.string(),
reinterpret_cast<hw_device_t**>(&device));
+ ATRACE_END();
if (res != OK) {
SET_ERR_L("Could not open camera: %s (%d)", strerror(-res), res);
@@ -269,7 +271,9 @@
mStatusTracker.clear();
if (mHal3Device != NULL) {
+ ATRACE_BEGIN("camera3->close");
mHal3Device->common.close(&mHal3Device->common);
+ ATRACE_END();
mHal3Device = NULL;
}
@@ -836,16 +840,20 @@
}
sp<Camera3StreamInterface> deletedStream;
+ ssize_t outputStreamIdx = mOutputStreams.indexOfKey(id);
if (mInputStream != NULL && id == mInputStream->getId()) {
deletedStream = mInputStream;
mInputStream.clear();
} else {
- ssize_t idx = mOutputStreams.indexOfKey(id);
- if (idx == NAME_NOT_FOUND) {
+ if (outputStreamIdx == NAME_NOT_FOUND) {
CLOGE("Stream %d does not exist", id);
return BAD_VALUE;
}
- deletedStream = mOutputStreams.editValueAt(idx);
+ }
+
+ // Delete output stream or the output part of a bi-directional stream.
+ if (outputStreamIdx != NAME_NOT_FOUND) {
+ deletedStream = mOutputStreams.editValueAt(outputStreamIdx);
mOutputStreams.removeItem(id);
}
@@ -1120,7 +1128,14 @@
Mutex::Autolock l(mLock);
mRequestThread->clear();
- return mHal3Device->ops->flush(mHal3Device);
+ status_t res;
+ if (mHal3Device->common.version >= CAMERA_DEVICE_API_VERSION_3_1) {
+ res = mHal3Device->ops->flush(mHal3Device);
+ } else {
+ res = waitUntilDrained();
+ }
+
+ return res;
}
/**
@@ -1664,8 +1679,10 @@
return;
}
- // Check if everything has arrived for this result (buffers and metadata)
- if (request.haveResultMetadata && request.numBuffersLeft == 0) {
+ // Check if everything has arrived for this result (buffers and metadata), remove it from
+ // InFlightMap if both arrived or HAL reports error for this request (i.e. during flush).
+ if ((request.requestStatus != OK) ||
+ (request.haveResultMetadata && request.numBuffersLeft == 0)) {
ATRACE_ASYNC_END("frame capture", frameNumber);
mInFlightMap.removeItemsAt(idx, 1);
}
diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.cpp b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
index 5aa9a3e..e1c492b 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
@@ -199,13 +199,33 @@
assert(mMaxSize == 0);
assert(camera3_stream::format != HAL_PIXEL_FORMAT_BLOB);
- mTotalBufferCount = BufferQueue::MIN_UNDEQUEUED_BUFFERS +
- camera3_stream::max_buffers;
mDequeuedBufferCount = 0;
mFrameCount = 0;
if (mConsumer.get() == 0) {
sp<BufferQueue> bq = new BufferQueue();
+
+ int minUndequeuedBuffers = 0;
+ res = bq->query(NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, &minUndequeuedBuffers);
+ if (res != OK || minUndequeuedBuffers < 0) {
+ ALOGE("%s: Stream %d: Could not query min undequeued buffers (error %d, bufCount %d)",
+ __FUNCTION__, mId, res, minUndequeuedBuffers);
+ return res;
+ }
+ size_t minBufs = static_cast<size_t>(minUndequeuedBuffers);
+ /*
+ * We promise never to 'acquire' more than camera3_stream::max_buffers
+ * at any one time.
+ *
+ * Boost the number up to meet the minimum required buffer count.
+ *
+ * (Note that this sets consumer-side buffer count only,
+ * and not the sum of producer+consumer side as in other camera streams).
+ */
+ mTotalBufferCount = camera3_stream::max_buffers > minBufs ?
+ camera3_stream::max_buffers : minBufs;
+ // TODO: somehow set the total buffer count when producer connects?
+
mConsumer = new BufferItemConsumer(bq, camera3_stream::usage,
mTotalBufferCount);
mConsumer->setName(String8::format("Camera3-InputStream-%d", mId));
diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.h b/services/camera/libcameraservice/device3/Camera3InputStream.h
index 681d684..ae49467 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.h
@@ -44,6 +44,8 @@
virtual void dump(int fd, const Vector<String16> &args) const;
+ // TODO: expose an interface to get the IGraphicBufferProducer
+
private:
typedef BufferItemConsumer::BufferItem BufferItem;
diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
index 1a54923..e509350 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
@@ -109,14 +109,14 @@
} // namespace anonymous
Camera3ZslStream::Camera3ZslStream(int id, uint32_t width, uint32_t height,
- int depth) :
+ int bufferCount) :
Camera3OutputStream(id, CAMERA3_STREAM_BIDIRECTIONAL,
width, height,
HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED),
- mDepth(depth) {
+ mDepth(bufferCount) {
sp<BufferQueue> bq = new BufferQueue();
- mProducer = new RingBufferConsumer(bq, GRALLOC_USAGE_HW_CAMERA_ZSL, depth);
+ mProducer = new RingBufferConsumer(bq, GRALLOC_USAGE_HW_CAMERA_ZSL, bufferCount);
mConsumer = new Surface(bq);
}
diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.h b/services/camera/libcameraservice/device3/Camera3ZslStream.h
index c7f4490..6721832 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.h
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.h
@@ -37,10 +37,10 @@
public Camera3OutputStream {
public:
/**
- * Set up a ZSL stream of a given resolution. Depth is the number of buffers
+ * Set up a ZSL stream of a given resolution. bufferCount is the number of buffers
* cached within the stream that can be retrieved for input.
*/
- Camera3ZslStream(int id, uint32_t width, uint32_t height, int depth);
+ Camera3ZslStream(int id, uint32_t width, uint32_t height, int bufferCount);
~Camera3ZslStream();
virtual void dump(int fd, const Vector<String16> &args) const;
diff --git a/services/camera/libcameraservice/gui/RingBufferConsumer.h b/services/camera/libcameraservice/gui/RingBufferConsumer.h
index b4ad824..a03736d 100644
--- a/services/camera/libcameraservice/gui/RingBufferConsumer.h
+++ b/services/camera/libcameraservice/gui/RingBufferConsumer.h
@@ -64,7 +64,7 @@
// bufferCount parameter specifies how many buffers can be pinned for user
// access at the same time.
RingBufferConsumer(const sp<IGraphicBufferConsumer>& consumer, uint32_t consumerUsage,
- int bufferCount = BufferQueue::MIN_UNDEQUEUED_BUFFERS);
+ int bufferCount);
virtual ~RingBufferConsumer();
diff --git a/services/medialog/MediaLogService.cpp b/services/medialog/MediaLogService.cpp
index 683fdf3..0c7fbbd 100644
--- a/services/medialog/MediaLogService.cpp
+++ b/services/medialog/MediaLogService.cpp
@@ -54,7 +54,7 @@
}
}
-status_t MediaLogService::dump(int fd, const Vector<String16>& args)
+status_t MediaLogService::dump(int fd, const Vector<String16>& args __unused)
{
// FIXME merge with similar but not identical code at services/audioflinger/ServiceUtilities.cpp
static const String16 sDump("android.permission.DUMP");
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
index cc3d509..3d6a74d 100644
--- a/tools/resampler_tools/fir.cpp
+++ b/tools/resampler_tools/fir.cpp
@@ -20,15 +20,25 @@
#include <stdlib.h>
#include <string.h>
-static double sinc(double x) {
+static inline double sinc(double x) {
if (fabs(x) == 0.0f) return 1.0f;
return sin(x) / x;
}
-static double sqr(double x) {
+static inline double sqr(double x) {
return x*x;
}
+static inline int64_t toint(double x, int64_t maxval) {
+ int64_t v;
+
+ v = static_cast<int64_t>(floor(x * maxval + 0.5));
+ if (v >= maxval) {
+ return maxval - 1; // error!
+ }
+ return v;
+}
+
static double I0(double x) {
// from the Numerical Recipes in C p. 237
double ax,ans,y;
@@ -54,11 +64,12 @@
return I0(beta * sqrt(1.0 - sqr((2.0*k)/N - 1.0))) / I0(beta);
}
-
static void usage(char* name) {
fprintf(stderr,
- "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] [-l lerp]\n"
- " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings] [-f {float|fixed}] [-b beta] [-v dBFS] -p M/N\n"
+ "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+ " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] [-l lerp]\n"
+ " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+ " [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] -p M/N\n"
" -h this help message\n"
" -d debug, print comma-separated coefficient table\n"
" -p generate poly-phase filter coefficients, with sample increment M/N\n"
@@ -66,6 +77,7 @@
" -c cut-off frequency (20478)\n"
" -n number of zero-crossings on one side (8)\n"
" -l number of lerping bits (4)\n"
+ " -m number of polyphases (related to -l, default 16)\n"
" -f output format, can be fixed-point or floating-point (fixed)\n"
" -b kaiser window parameter beta (7.865 [-80dB])\n"
" -v attenuation in dBFS (0)\n",
@@ -77,8 +89,7 @@
int main(int argc, char** argv)
{
// nc is the number of bits to store the coefficients
- const int nc = 32;
-
+ int nc = 32;
bool polyphase = false;
unsigned int polyM = 160;
unsigned int polyN = 147;
@@ -88,7 +99,6 @@
double atten = 1;
int format = 0;
-
// in order to keep the errors associated with the linear
// interpolation of the coefficients below the quantization error
// we must satisfy:
@@ -104,7 +114,6 @@
// Smith, J.O. Digital Audio Resampling Home Page
// https://ccrma.stanford.edu/~jos/resample/, 2011-03-29
//
- int nz = 4;
// | 0.1102*(A - 8.7) A > 50
// beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21) 21 <= A <= 50
@@ -123,7 +132,6 @@
// 100 dB 10.056
double beta = 7.865;
-
// 2*nzc = (A - 8) / (2.285 * dw)
// with dw the transition width = 2*pi*dF/Fs
//
@@ -148,8 +156,9 @@
// nzc = 20
//
+ int M = 1 << 4; // number of phases for interpolation
int ch;
- while ((ch = getopt(argc, argv, ":hds:c:n:f:l:b:p:v:")) != -1) {
+ while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:")) != -1) {
switch (ch) {
case 'd':
debug = true;
@@ -169,13 +178,26 @@
case 'n':
nzc = atoi(optarg);
break;
+ case 'm':
+ M = atoi(optarg);
+ break;
case 'l':
- nz = atoi(optarg);
+ M = 1 << atoi(optarg);
break;
case 'f':
- if (!strcmp(optarg,"fixed")) format = 0;
- else if (!strcmp(optarg,"float")) format = 1;
- else usage(argv[0]);
+ if (!strcmp(optarg, "fixed")) {
+ format = 0;
+ }
+ else if (!strcmp(optarg, "fixed16")) {
+ format = 0;
+ nc = 16;
+ }
+ else if (!strcmp(optarg, "float")) {
+ format = 1;
+ }
+ else {
+ usage(argv[0]);
+ }
break;
case 'b':
beta = atof(optarg);
@@ -193,11 +215,14 @@
// cut off frequency ratio Fc/Fs
double Fcr = Fc / Fs;
-
// total number of coefficients (one side)
- const int M = (1 << nz);
+
const int N = M * nzc;
+ // lerp (which is most useful if M is a power of 2)
+
+ int nz = 0; // recalculate nz as the bits needed to represent M
+ for (int i = M-1 ; i; i>>=1, nz++);
// generate the right half of the filter
if (!debug) {
printf("// cmd-line: ");
@@ -207,7 +232,7 @@
printf("\n");
if (!polyphase) {
printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N);
- printf("const int32_t RESAMPLE_FIR_LERP_INT_BITS = %d;\n", nz);
+ printf("const int32_t RESAMPLE_FIR_INT_PHASES = %d;\n", M);
printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc);
} else {
printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN);
@@ -224,7 +249,7 @@
for (int i=0 ; i<=M ; i++) { // an extra set of coefs for interpolation
for (int j=0 ; j<nzc ; j++) {
int ix = j*M + i;
- double x = (2.0 * M_PI * ix * Fcr) / (1 << nz);
+ double x = (2.0 * M_PI * ix * Fcr) / M;
double y = kaiser(ix+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;
y *= atten;
@@ -232,11 +257,13 @@
if (j == 0)
printf("\n ");
}
-
if (!format) {
- int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
- if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
- printf("0x%08x, ", int32_t(yi));
+ int64_t yi = toint(y, 1ULL<<(nc-1));
+ if (nc > 16) {
+ printf("0x%08x, ", int32_t(yi));
+ } else {
+ printf("0x%04x, ", int32_t(yi)&0xffff);
+ }
} else {
printf("%.9g%s ", y, debug ? "," : "f,");
}
@@ -254,9 +281,12 @@
double y = kaiser(i+N, 2*N, beta) * sinc(x) * 2.0 * Fcr;;
y *= atten;
if (!format) {
- int64_t yi = floor(y * ((1ULL<<(nc-1))) + 0.5);
- if (yi >= (1LL<<(nc-1))) yi = (1LL<<(nc-1))-1;
- printf("0x%08x", int32_t(yi));
+ int64_t yi = toint(y, 1ULL<<(nc-1));
+ if (nc > 16) {
+ printf("0x%08x, ", int32_t(yi));
+ } else {
+ printf("0x%04x, ", int32_t(yi)&0xffff);
+ }
} else {
printf("%.9g%s", y, debug ? "" : "f");
}
@@ -277,5 +307,3 @@
}
// http://www.csee.umbc.edu/help/sound/AFsp-V2R1/html/audio/ResampAudio.html
-
-