Merge "Fix non-monotonic AudioTrack::getTimestamp after pause" into lmp-preview-dev
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index e862ec3..d38d976 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -53,6 +53,7 @@
kKeyESDS = 'esds', // raw data
kKeyAACProfile = 'aacp', // int32_t
kKeyAVCC = 'avcc', // raw data
+ kKeyHVCC = 'hvcc', // raw data
kKeyD263 = 'd263', // raw data
kKeyVorbisInfo = 'vinf', // raw data
kKeyVorbisBooks = 'vboo', // raw data
@@ -170,6 +171,7 @@
enum {
kTypeESDS = 'esds',
kTypeAVCC = 'avcc',
+ kTypeHVCC = 'hvcc',
kTypeD263 = 'd263',
};
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 5121c17..5590b60 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -352,6 +352,9 @@
int64_t getDecodingTimeUs();
+ status_t parseHEVCCodecSpecificData(
+ const void *data, size_t size,
+ unsigned *profile, unsigned *level);
status_t parseAVCCodecSpecificData(
const void *data, size_t size,
unsigned *profile, unsigned *level);
diff --git a/media/img_utils/src/DngUtils.cpp b/media/img_utils/src/DngUtils.cpp
index 788dfc8..14b31ec 100644
--- a/media/img_utils/src/DngUtils.cpp
+++ b/media/img_utils/src/DngUtils.cpp
@@ -19,7 +19,7 @@
namespace android {
namespace img_utils {
-OpcodeListBuilder::OpcodeListBuilder() : mOpList(), mEndianOut(&mOpList, BIG) {
+OpcodeListBuilder::OpcodeListBuilder() : mCount(0), mOpList(), mEndianOut(&mOpList, BIG) {
if(mEndianOut.open() != OK) {
ALOGE("%s: Open failed.", __FUNCTION__);
}
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 1c808d0..db61e85 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -203,23 +203,6 @@
mFrameSize = sizeof(uint8_t);
}
- // validate framecount
- size_t minFrameCount;
- status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
- sampleRate, format, channelMask);
- if (status != NO_ERROR) {
- ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
- sampleRate, format, channelMask, status);
- return status;
- }
- ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
-
- if (frameCount == 0) {
- frameCount = minFrameCount;
- } else if (frameCount < minFrameCount) {
- ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
- return BAD_VALUE;
- }
// mFrameCount is initialized in openRecord_l
mReqFrameCount = frameCount;
@@ -242,7 +225,7 @@
}
// create the IAudioRecord
- status = openRecord_l(0 /*epoch*/);
+ status_t status = openRecord_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
@@ -464,6 +447,29 @@
size_t frameCount = mReqFrameCount;
if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
+ // validate framecount
+ // If fast track was not requested, this preserves
+ // the old behavior of validating on client side.
+ // FIXME Eventually the validation should be done on server side
+ // regardless of whether it's a fast or normal track. It's debatable
+ // whether to account for the input latency to provision buffers appropriately.
+ size_t minFrameCount;
+ status = AudioRecord::getMinFrameCount(&minFrameCount,
+ mSampleRate, mFormat, mChannelMask);
+ if (status != NO_ERROR) {
+ ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; "
+ "status %d",
+ mSampleRate, mFormat, mChannelMask, status);
+ return status;
+ }
+
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ } else if (frameCount < minFrameCount) {
+ ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
+ return BAD_VALUE;
+ }
+
// Make sure that application is notified with sufficient margin before overrun
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
mNotificationFramesAct = frameCount/2;
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 219dbfd..0dbfa62 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -134,10 +134,17 @@
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
- mIsShutdown = true;
- status = NO_INIT;
- goto end;
+ if (mIsOut) {
+ ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
+ "shutting down", filled, mFrameCount);
+ mIsShutdown = true;
+ status = NO_INIT;
+ goto end;
+ }
+ // for input, sync up on overrun
+ filled = 0;
+ cblk->u.mStreaming.mFront = rear;
+ (void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
}
// don't allow filling pipe beyond the nominal size
size_t avail = mIsOut ? mFrameCount - filled : filled;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index eee72c5..77d131b 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -421,7 +421,6 @@
status = (status_t)reply.readInt32();
*num_ports = (unsigned int)reply.readInt32();
}
- ALOGI("listAudioPorts() status %d got *num_ports %d", status, *num_ports);
if (status == NO_ERROR) {
if (numPortsReq > *num_ports) {
numPortsReq = *num_ports;
@@ -840,7 +839,6 @@
status_t status = listAudioPorts(role, type, &numPorts, ports, &generation);
reply->writeInt32(status);
reply->writeInt32(numPorts);
- ALOGI("LIST_AUDIO_PORTS status %d got numPorts %d", status, numPorts);
if (status == NO_ERROR) {
if (numPortsReq > numPorts) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 469c9ca..cfbf282 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -37,6 +37,7 @@
: mNotify(notify),
mNativeWindow(nativeWindow),
mBufferGeneration(0),
+ mPaused(true),
mComponentName("decoder") {
// Every decoder has its own looper because MediaCodec operations
// are blocking, but NuPlayer needs asynchronous operations.
@@ -112,6 +113,7 @@
mOutputBuffers.size());
requestCodecNotification();
+ mPaused = false;
}
void NuPlayer::Decoder::requestCodecNotification() {
@@ -352,6 +354,11 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatFlushCompleted);
notify->post();
+ mPaused = true;
+}
+
+void NuPlayer::Decoder::onResume() {
+ mPaused = false;
}
void NuPlayer::Decoder::onShutdown() {
@@ -380,6 +387,7 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatShutdownCompleted);
notify->post();
+ mPaused = true;
}
void NuPlayer::Decoder::onMessageReceived(const sp<AMessage> &msg) {
@@ -397,7 +405,9 @@
case kWhatCodecNotify:
{
if (!isStaleReply(msg)) {
- while (handleAnInputBuffer()) {
+ if (!mPaused) {
+ while (handleAnInputBuffer()) {
+ }
}
while (handleAnOutputBuffer()) {
@@ -430,6 +440,12 @@
break;
}
+ case kWhatResume:
+ {
+ onResume();
+ break;
+ }
+
case kWhatShutdown:
{
onShutdown();
@@ -447,7 +463,7 @@
}
void NuPlayer::Decoder::signalResume() {
- // nothing to do
+ (new AMessage(kWhatResume, id()))->post();
}
void NuPlayer::Decoder::initiateShutdown() {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index 94243fc..2892584 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -87,11 +87,13 @@
void onConfigure(const sp<AMessage> &format);
void onFlush();
+ void onResume();
void onInputBufferFilled(const sp<AMessage> &msg);
void onRenderBuffer(const sp<AMessage> &msg);
void onShutdown();
int32_t mBufferGeneration;
+ bool mPaused;
AString mComponentName;
bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index e07b6aa..aabe0ec 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -95,6 +95,7 @@
uint64_t* mCurrentSampleInfoOffsets;
bool mIsAVC;
+ bool mIsHEVC;
size_t mNALLengthSize;
bool mStarted;
@@ -317,6 +318,9 @@
case FOURCC('a', 'v', 'c', '1'):
return MEDIA_MIMETYPE_VIDEO_AVC;
+ case FOURCC('h', 'v', 'c', '1'):
+ case FOURCC('h', 'e', 'v', '1'):
+ return MEDIA_MIMETYPE_VIDEO_HEVC;
default:
CHECK(!"should not be here.");
return NULL;
@@ -478,11 +482,20 @@
off64_t offset = 0;
status_t err;
while (true) {
+ off64_t orig_offset = offset;
err = parseChunk(&offset, 0);
- if (err == OK) {
- continue;
- }
+ if (offset <= orig_offset) {
+ // only continue parsing if the offset was advanced,
+ // otherwise we might end up in an infinite loop
+ ALOGE("did not advance: 0x%lld->0x%lld", orig_offset, offset);
+ err = ERROR_MALFORMED;
+ break;
+ } else if (err == OK) {
+ continue;
+ } else if (err != UNKNOWN_ERROR) {
+ break;
+ }
uint32_t hdr[2];
if (mDataSource->readAt(offset, hdr, 8) < 8) {
break;
@@ -505,8 +518,6 @@
} else {
mFileMetaData->setCString(kKeyMIMEType, "audio/mp4");
}
-
- mInitCheck = OK;
} else {
mInitCheck = err;
}
@@ -758,8 +769,25 @@
// The smallest valid chunk is 16 bytes long in this case.
return ERROR_MALFORMED;
}
+ } else if (chunk_size == 0) {
+ if (depth == 0) {
+ // atom extends to end of file
+ off64_t sourceSize;
+ if (mDataSource->getSize(&sourceSize) == OK) {
+ chunk_size = (sourceSize - *offset);
+ } else {
+ // XXX could we just pick a "sufficiently large" value here?
+ ALOGE("atom size is 0, and data source has no size");
+ return ERROR_MALFORMED;
+ }
+ } else {
+ // not allowed for non-toplevel atoms, skip it
+ *offset += 4;
+ return OK;
+ }
} else if (chunk_size < 8) {
// The smallest valid chunk is 8 bytes long.
+ ALOGE("invalid chunk size: %d", int(chunk_size));
return ERROR_MALFORMED;
}
@@ -1288,6 +1316,8 @@
case FOURCC('H', '2', '6', '3'):
case FOURCC('h', '2', '6', '3'):
case FOURCC('a', 'v', 'c', '1'):
+ case FOURCC('h', 'v', 'c', '1'):
+ case FOURCC('h', 'e', 'v', '1'):
{
mHasVideo = true;
@@ -1580,6 +1610,21 @@
break;
}
+ case FOURCC('h', 'v', 'c', 'C'):
+ {
+ sp<ABuffer> buffer = new ABuffer(chunk_data_size);
+
+ if (mDataSource->readAt(
+ data_offset, buffer->data(), chunk_data_size) < chunk_data_size) {
+ return ERROR_IO;
+ }
+
+ mLastTrack->meta->setData(
+ kKeyHVCC, kTypeHVCC, buffer->data(), chunk_data_size);
+
+ *offset += chunk_size;
+ break;
+ }
case FOURCC('d', '2', '6', '3'):
{
@@ -2452,6 +2497,11 @@
|| type != kTypeAVCC) {
return ERROR_MALFORMED;
}
+ } else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC)) {
+ if (!track->meta->findData(kKeyHVCC, &type, &data, &size)
+ || type != kTypeHVCC) {
+ return ERROR_MALFORMED;
+ }
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_MPEG4)
|| !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC)) {
if (!track->meta->findData(kKeyESDS, &type, &data, &size)
@@ -2776,6 +2826,7 @@
mCurrentSampleInfoOffsetsAllocSize(0),
mCurrentSampleInfoOffsets(NULL),
mIsAVC(false),
+ mIsHEVC(false),
mNALLengthSize(0),
mStarted(false),
mGroup(NULL),
@@ -2800,6 +2851,7 @@
CHECK(success);
mIsAVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC);
+ mIsHEVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC);
if (mIsAVC) {
uint32_t type;
@@ -2814,6 +2866,18 @@
// The number of bytes used to encode the length of a NAL unit.
mNALLengthSize = 1 + (ptr[4] & 3);
+ } else if (mIsHEVC) {
+ uint32_t type;
+ const void *data;
+ size_t size;
+ CHECK(format->findData(kKeyHVCC, &type, &data, &size));
+
+ const uint8_t *ptr = (const uint8_t *)data;
+
+ CHECK(size >= 7);
+ CHECK_EQ((unsigned)ptr[0], 1u); // configurationVersion == 1
+
+ mNALLengthSize = 1 + (ptr[14 + 7] & 3);
}
CHECK(format->findInt32(kKeyTrackID, &mTrackId));
@@ -3562,7 +3626,7 @@
}
}
- if (!mIsAVC || mWantsNALFragments) {
+ if ((!mIsAVC && !mIsHEVC) || mWantsNALFragments) {
if (newBuffer) {
ssize_t num_bytes_read =
mDataSource->readAt(offset, (uint8_t *)mBuffer->data(), size);
@@ -3594,7 +3658,7 @@
++mCurrentSampleIndex;
}
- if (!mIsAVC) {
+ if (!mIsAVC && !mIsHEVC) {
*out = mBuffer;
mBuffer = NULL;
@@ -3837,7 +3901,7 @@
bufmeta->setData(kKeyCryptoKey, 0, mCryptoKey, 16);
}
- if (!mIsAVC || mWantsNALFragments) {
+ if ((!mIsAVC && !mIsHEVC)|| mWantsNALFragments) {
if (newBuffer) {
ssize_t num_bytes_read =
mDataSource->readAt(offset, (uint8_t *)mBuffer->data(), size);
@@ -3869,7 +3933,7 @@
++mCurrentSampleIndex;
}
- if (!mIsAVC) {
+ if (!mIsAVC && !mIsHEVC) {
*out = mBuffer;
mBuffer = NULL;
@@ -4043,6 +4107,8 @@
FOURCC('i', 's', 'o', 'm'),
FOURCC('i', 's', 'o', '2'),
FOURCC('a', 'v', 'c', '1'),
+ FOURCC('h', 'v', 'c', '1'),
+ FOURCC('h', 'e', 'v', '1'),
FOURCC('3', 'g', 'p', '4'),
FOURCC('m', 'p', '4', '1'),
FOURCC('m', 'p', '4', '2'),
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
index 8a451c8..cd51582 100644
--- a/media/libstagefright/MediaCodecList.cpp
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -70,11 +70,6 @@
return;
}
- // These are currently still used by the video editing suite.
- addMediaCodec(true /* encoder */, "AACEncoder", "audio/mp4a-latm");
- addMediaCodec(
- false /* encoder */, "OMX.google.raw.decoder", "audio/raw");
-
for (size_t i = mCodecInfos.size(); i-- > 0;) {
CodecInfo *info = &mCodecInfos.editItemAt(i);
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index c028dbf..354712c 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -381,6 +381,57 @@
return NULL;
}
+status_t OMXCodec::parseHEVCCodecSpecificData(
+ const void *data, size_t size,
+ unsigned *profile, unsigned *level) {
+ const uint8_t *ptr = (const uint8_t *)data;
+
+ // verify minimum size and configurationVersion == 1.
+ if (size < 7 || ptr[0] != 1) {
+ return ERROR_MALFORMED;
+ }
+
+ *profile = (ptr[1] & 31);
+ *level = ptr[12];
+
+ ptr += 22;
+ size -= 22;
+
+ size_t numofArrays = (char)ptr[0];
+ ptr += 1;
+ size -= 1;
+ size_t j = 0, i = 0;
+ for (i = 0; i < numofArrays; i++) {
+ ptr += 1;
+ size -= 1;
+
+ // Num of nals
+ size_t numofNals = U16_AT(ptr);
+ ptr += 2;
+ size -= 2;
+
+ for (j = 0;j < numofNals;j++) {
+ if (size < 2) {
+ return ERROR_MALFORMED;
+ }
+
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ if (size < length) {
+ return ERROR_MALFORMED;
+ }
+ addCodecSpecificData(ptr, length);
+
+ ptr += length;
+ size -= length;
+ }
+ }
+ return OK;
+}
+
status_t OMXCodec::parseAVCCodecSpecificData(
const void *data, size_t size,
unsigned *profile, unsigned *level) {
@@ -493,6 +544,20 @@
CODEC_LOGI(
"AVC profile = %u (%s), level = %u",
profile, AVCProfileToString(profile), level);
+ } else if (meta->findData(kKeyHVCC, &type, &data, &size)) {
+ // Parse the HEVCDecoderConfigurationRecord
+
+ unsigned profile, level;
+ status_t err;
+ if ((err = parseHEVCCodecSpecificData(
+ data, size, &profile, &level)) != OK) {
+ ALOGE("Malformed HEVC codec specific data.");
+ return err;
+ }
+
+ CODEC_LOGI(
+ "HEVC profile = %u , level = %u",
+ profile, level);
} else if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
addCodecSpecificData(data, size);
@@ -822,6 +887,8 @@
OMX_VIDEO_CODINGTYPE compressionFormat = OMX_VIDEO_CodingUnused;
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime)) {
compressionFormat = OMX_VIDEO_CodingAVC;
+ } else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_HEVC, mime)) {
+ compressionFormat = OMX_VIDEO_CodingHEVC;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
compressionFormat = OMX_VIDEO_CodingMPEG4;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_H263, mime)) {
@@ -1217,6 +1284,8 @@
compressionFormat = OMX_VIDEO_CodingAVC;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
compressionFormat = OMX_VIDEO_CodingMPEG4;
+ } else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_HEVC, mime)) {
+ compressionFormat = OMX_VIDEO_CodingHEVC;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_H263, mime)) {
compressionFormat = OMX_VIDEO_CodingH263;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_VP8, mime)) {
@@ -1411,6 +1480,8 @@
"audio_decoder.g711alaw", "audio_encoder.g711alaw" },
{ MEDIA_MIMETYPE_VIDEO_AVC,
"video_decoder.avc", "video_encoder.avc" },
+ { MEDIA_MIMETYPE_VIDEO_HEVC,
+ "video_decoder.hevc", "video_encoder.hevc" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4,
"video_decoder.mpeg4", "video_encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_H263,
@@ -3009,7 +3080,8 @@
size_t size = specific->mSize;
- if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mMIME)
+ if ((!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mMIME) ||
+ !strcasecmp(MEDIA_MIMETYPE_VIDEO_HEVC, mMIME))
&& !(mQuirks & kWantsNALFragments)) {
static const uint8_t kNALStartCode[4] =
{ 0x00, 0x00, 0x00, 0x01 };
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 047fac7..7ff31a1 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -217,6 +217,56 @@
buffer->meta()->setInt32("csd", true);
buffer->meta()->setInt64("timeUs", 0);
msg->setBuffer("csd-1", buffer);
+ } else if (meta->findData(kKeyHVCC, &type, &data, &size)) {
+ const uint8_t *ptr = (const uint8_t *)data;
+
+ CHECK(size >= 7);
+ CHECK_EQ((unsigned)ptr[0], 1u); // configurationVersion == 1
+ uint8_t profile = ptr[1] & 31;
+ uint8_t level = ptr[12];
+ ptr += 22;
+ size -= 22;
+
+
+ size_t numofArrays = (char)ptr[0];
+ ptr += 1;
+ size -= 1;
+ size_t j = 0, i = 0;
+
+ sp<ABuffer> buffer = new ABuffer(1024);
+ buffer->setRange(0, 0);
+
+ for (i = 0; i < numofArrays; i++) {
+ ptr += 1;
+ size -= 1;
+
+ //Num of nals
+ size_t numofNals = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ for (j = 0; j < numofNals; j++) {
+ CHECK(size >= 2);
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ CHECK(size >= length);
+
+ memcpy(buffer->data() + buffer->size(), "\x00\x00\x00\x01", 4);
+ memcpy(buffer->data() + buffer->size() + 4, ptr, length);
+ buffer->setRange(0, buffer->size() + 4 + length);
+
+ ptr += length;
+ size -= length;
+ }
+ }
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-0", buffer);
+
} else if (meta->findData(kKeyESDS, &type, &data, &size)) {
ESDS esds((const char *)data, size);
CHECK_EQ(esds.InitCheck(), (status_t)OK);
diff --git a/media/libstagefright/codecs/hevcdec/Android.mk b/media/libstagefright/codecs/hevcdec/Android.mk
new file mode 100644
index 0000000..2fe347b
--- /dev/null
+++ b/media/libstagefright/codecs/hevcdec/Android.mk
@@ -0,0 +1,22 @@
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libstagefright_soft_hevcdec
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_STATIC_LIBRARIES := libhevcdec
+LOCAL_SRC_FILES := SoftHEVC.cpp
+
+LOCAL_C_INCLUDES := $(TOP)/external/libhevc/decoder
+LOCAL_C_INCLUDES += $(TOP)/external/libhevc/common
+LOCAL_C_INCLUDES += $(TOP)/frameworks/av/media/libstagefright/include
+LOCAL_C_INCLUDES += $(TOP)/frameworks/native/include/media/openmax
+
+LOCAL_SHARED_LIBRARIES := libstagefright
+LOCAL_SHARED_LIBRARIES += libstagefright_omx
+LOCAL_SHARED_LIBRARIES += libstagefright_foundation
+LOCAL_SHARED_LIBRARIES += libutils
+LOCAL_SHARED_LIBRARIES += liblog
+
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/hevcdec/SoftHEVC.cpp b/media/libstagefright/codecs/hevcdec/SoftHEVC.cpp
new file mode 100644
index 0000000..0aae5ed
--- /dev/null
+++ b/media/libstagefright/codecs/hevcdec/SoftHEVC.cpp
@@ -0,0 +1,710 @@
+/*
+ * 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 "SoftHEVC"
+#include <utils/Log.h>
+
+#include "ihevc_typedefs.h"
+#include "iv.h"
+#include "ivd.h"
+#include "ithread.h"
+#include "ihevcd_cxa.h"
+#include "SoftHEVC.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaDefs.h>
+#include <OMX_VideoExt.h>
+
+namespace android {
+
+#define componentName "video_decoder.hevc"
+#define codingType OMX_VIDEO_CodingHEVC
+#define CODEC_MIME_TYPE MEDIA_MIMETYPE_VIDEO_HEVC
+
+/** Function and structure definitions to keep code similar for each codec */
+#define ivdec_api_function ihevcd_cxa_api_function
+#define ivdext_init_ip_t ihevcd_cxa_init_ip_t
+#define ivdext_init_op_t ihevcd_cxa_init_op_t
+#define ivdext_fill_mem_rec_ip_t ihevcd_cxa_fill_mem_rec_ip_t
+#define ivdext_fill_mem_rec_op_t ihevcd_cxa_fill_mem_rec_op_t
+#define ivdext_ctl_set_num_cores_ip_t ihevcd_cxa_ctl_set_num_cores_ip_t
+#define ivdext_ctl_set_num_cores_op_t ihevcd_cxa_ctl_set_num_cores_op_t
+
+#define IVDEXT_CMD_CTL_SET_NUM_CORES \
+ (IVD_CONTROL_API_COMMAND_TYPE_T)IHEVCD_CXA_CMD_CTL_SET_NUM_CORES
+
+static const CodecProfileLevel kProfileLevels[] = {
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel1 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel2 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel21 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel3 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel31 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel4 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel41 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel5 },
+ { OMX_VIDEO_HEVCProfileMain, OMX_VIDEO_HEVCMainTierLevel51 },
+};
+
+SoftHEVC::SoftHEVC(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SoftVideoDecoderOMXComponent(name, componentName, codingType,
+ kProfileLevels, ARRAY_SIZE(kProfileLevels),
+ CODEC_MAX_WIDTH /* width */, CODEC_MAX_HEIGHT /* height */, callbacks,
+ appData, component) {
+ initPorts(kNumBuffers, INPUT_BUF_SIZE, kNumBuffers,
+ CODEC_MIME_TYPE);
+
+ mOmxColorFormat = OMX_COLOR_FormatYUV420Planar;
+ mStride = mWidth;
+
+ if (OMX_COLOR_FormatYUV420Planar == mOmxColorFormat) {
+ mIvColorFormat = IV_YUV_420P;
+ } else if (OMX_COLOR_FormatYUV420SemiPlanar == mOmxColorFormat) {
+ mIvColorFormat = IV_YUV_420SP_UV;
+ }
+
+ mInitWidth = mWidth;
+ mInitHeight = mHeight;
+
+ CHECK_EQ(initDecoder(), (status_t)OK);
+}
+
+SoftHEVC::~SoftHEVC() {
+ ALOGD("In SoftHEVC::~SoftHEVC");
+ CHECK_EQ(deInitDecoder(), (status_t)OK);
+}
+
+static size_t GetCPUCoreCount() {
+ long cpuCoreCount = 1;
+#if defined(_SC_NPROCESSORS_ONLN)
+ cpuCoreCount = sysconf(_SC_NPROCESSORS_ONLN);
+#else
+ // _SC_NPROC_ONLN must be defined...
+ cpuCoreCount = sysconf(_SC_NPROC_ONLN);
+#endif
+ CHECK(cpuCoreCount >= 1);
+ ALOGD("Number of CPU cores: %ld", cpuCoreCount);
+ return (size_t)cpuCoreCount;
+}
+
+status_t SoftHEVC::getVersion() {
+ ivd_ctl_getversioninfo_ip_t s_ctl_ip;
+ ivd_ctl_getversioninfo_op_t s_ctl_op;
+ UWORD8 au1_buf[512];
+ IV_API_CALL_STATUS_T status;
+
+ s_ctl_ip.e_cmd = IVD_CMD_VIDEO_CTL;
+ s_ctl_ip.e_sub_cmd = IVD_CMD_CTL_GETVERSION;
+ s_ctl_ip.u4_size = sizeof(ivd_ctl_getversioninfo_ip_t);
+ s_ctl_op.u4_size = sizeof(ivd_ctl_getversioninfo_op_t);
+ s_ctl_ip.pv_version_buffer = au1_buf;
+ s_ctl_ip.u4_version_buffer_size = sizeof(au1_buf);
+
+ status = ivdec_api_function(mCodecCtx, (void *)&s_ctl_ip,
+ (void *)&s_ctl_op);
+
+ if (status != IV_SUCCESS) {
+ ALOGE("Error in getting version number: 0x%x",
+ s_ctl_op.u4_error_code);
+ } else {
+ ALOGD("Ittiam decoder version number: %s",
+ (char *)s_ctl_ip.pv_version_buffer);
+ }
+ return OK;
+}
+
+status_t SoftHEVC::setParams(WORD32 stride, IVD_VIDEO_DECODE_MODE_T decMode) {
+ ivd_ctl_set_config_ip_t s_ctl_ip;
+ ivd_ctl_set_config_op_t s_ctl_op;
+ IV_API_CALL_STATUS_T status;
+ s_ctl_ip.u4_disp_wd = stride;
+ s_ctl_ip.e_frm_skip_mode = IVD_SKIP_NONE;
+
+ s_ctl_ip.e_frm_out_mode = IVD_DISPLAY_FRAME_OUT;
+ s_ctl_ip.e_vid_dec_mode = decMode;
+ s_ctl_ip.e_cmd = IVD_CMD_VIDEO_CTL;
+ s_ctl_ip.e_sub_cmd = IVD_CMD_CTL_SETPARAMS;
+ s_ctl_ip.u4_size = sizeof(ivd_ctl_set_config_ip_t);
+ s_ctl_op.u4_size = sizeof(ivd_ctl_set_config_op_t);
+
+ ALOGD("Set the run-time (dynamic) parameters");
+ status = ivdec_api_function(mCodecCtx, (void *)&s_ctl_ip,
+ (void *)&s_ctl_op);
+
+ if (status != IV_SUCCESS) {
+ ALOGE("Error in setting the run-time parameters: 0x%x",
+ s_ctl_op.u4_error_code);
+
+ return UNKNOWN_ERROR;
+ }
+ return OK;
+}
+
+status_t SoftHEVC::resetPlugin() {
+ mIsInFlush = false;
+ mReceivedEOS = false;
+ memset(mTimeStamps, 0, sizeof(mTimeStamps));
+ memset(mTimeStampsValid, 0, sizeof(mTimeStampsValid));
+
+ /* Initialize both start and end times */
+ gettimeofday(&mTimeStart, NULL);
+ gettimeofday(&mTimeEnd, NULL);
+
+ return OK;
+}
+
+status_t SoftHEVC::resetDecoder() {
+ ivd_ctl_reset_ip_t s_ctl_ip;
+ ivd_ctl_reset_op_t s_ctl_op;
+ IV_API_CALL_STATUS_T status;
+
+ s_ctl_ip.e_cmd = IVD_CMD_VIDEO_CTL;
+ s_ctl_ip.e_sub_cmd = IVD_CMD_CTL_RESET;
+ s_ctl_ip.u4_size = sizeof(ivd_ctl_reset_ip_t);
+ s_ctl_op.u4_size = sizeof(ivd_ctl_reset_op_t);
+
+ status = ivdec_api_function(mCodecCtx, (void *)&s_ctl_ip,
+ (void *)&s_ctl_op);
+ if (IV_SUCCESS != status) {
+ ALOGE("Error in reset: 0x%x", s_ctl_op.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+
+ /* Set the run-time (dynamic) parameters */
+ setParams(0, IVD_DECODE_FRAME);
+
+ /* Set number of cores/threads to be used by the codec */
+ setNumCores();
+
+ return OK;
+}
+
+status_t SoftHEVC::setNumCores() {
+ ivdext_ctl_set_num_cores_ip_t s_set_cores_ip;
+ ivdext_ctl_set_num_cores_op_t s_set_cores_op;
+ IV_API_CALL_STATUS_T status;
+ s_set_cores_ip.e_cmd = IVD_CMD_VIDEO_CTL;
+ s_set_cores_ip.e_sub_cmd = IVDEXT_CMD_CTL_SET_NUM_CORES;
+ s_set_cores_ip.u4_num_cores = MIN(mNumCores, CODEC_MAX_NUM_CORES);
+ s_set_cores_ip.u4_size = sizeof(ivdext_ctl_set_num_cores_ip_t);
+ s_set_cores_op.u4_size = sizeof(ivdext_ctl_set_num_cores_op_t);
+ ALOGD("Set number of cores to %u", s_set_cores_ip.u4_num_cores);
+ status = ivdec_api_function(mCodecCtx, (void *)&s_set_cores_ip,
+ (void *)&s_set_cores_op);
+ if (IV_SUCCESS != status) {
+ ALOGE("Error in setting number of cores: 0x%x",
+ s_set_cores_op.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+ return OK;
+}
+
+status_t SoftHEVC::setFlushMode() {
+ IV_API_CALL_STATUS_T status;
+ ivd_ctl_flush_ip_t s_video_flush_ip;
+ ivd_ctl_flush_op_t s_video_flush_op;
+
+ s_video_flush_ip.e_cmd = IVD_CMD_VIDEO_CTL;
+ s_video_flush_ip.e_sub_cmd = IVD_CMD_CTL_FLUSH;
+ s_video_flush_ip.u4_size = sizeof(ivd_ctl_flush_ip_t);
+ s_video_flush_op.u4_size = sizeof(ivd_ctl_flush_op_t);
+ ALOGD("Set the decoder in flush mode ");
+
+ /* Set the decoder in Flush mode, subsequent decode() calls will flush */
+ status = ivdec_api_function(mCodecCtx, (void *)&s_video_flush_ip,
+ (void *)&s_video_flush_op);
+
+ if (status != IV_SUCCESS) {
+ ALOGE("Error in setting the decoder in flush mode: (%d) 0x%x", status,
+ s_video_flush_op.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+
+ mIsInFlush = true;
+ return OK;
+}
+
+status_t SoftHEVC::initDecoder() {
+ IV_API_CALL_STATUS_T status;
+
+ UWORD32 u4_num_reorder_frames;
+ UWORD32 u4_num_ref_frames;
+ UWORD32 u4_share_disp_buf;
+ WORD32 i4_level;
+
+ mNumCores = GetCPUCoreCount();
+
+ /* Initialize number of ref and reorder modes (for HEVC) */
+ u4_num_reorder_frames = 16;
+ u4_num_ref_frames = 16;
+ u4_share_disp_buf = 0;
+
+ if ((mWidth * mHeight) > (1920 * 1088)) {
+ i4_level = 50;
+ } else if ((mWidth * mHeight) > (1280 * 720)) {
+ i4_level = 41;
+ } else {
+ i4_level = 31;
+ }
+
+ {
+ iv_num_mem_rec_ip_t s_num_mem_rec_ip;
+ iv_num_mem_rec_op_t s_num_mem_rec_op;
+
+ s_num_mem_rec_ip.u4_size = sizeof(s_num_mem_rec_ip);
+ s_num_mem_rec_op.u4_size = sizeof(s_num_mem_rec_op);
+ s_num_mem_rec_ip.e_cmd = IV_CMD_GET_NUM_MEM_REC;
+
+ ALOGV("Get number of mem records");
+ status = ivdec_api_function(mCodecCtx, (void*)&s_num_mem_rec_ip,
+ (void*)&s_num_mem_rec_op);
+ if (IV_SUCCESS != status) {
+ ALOGE("Error in getting mem records: 0x%x",
+ s_num_mem_rec_op.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+
+ mNumMemRecords = s_num_mem_rec_op.u4_num_mem_rec;
+ }
+
+ mMemRecords = (iv_mem_rec_t*)ivd_aligned_malloc(
+ 128, mNumMemRecords * sizeof(iv_mem_rec_t));
+ if (mMemRecords == NULL) {
+ ALOGE("Allocation failure");
+ return NO_MEMORY;
+ }
+
+ {
+ size_t i;
+ ivdext_fill_mem_rec_ip_t s_fill_mem_ip;
+ ivdext_fill_mem_rec_op_t s_fill_mem_op;
+ iv_mem_rec_t *ps_mem_rec;
+
+ s_fill_mem_ip.s_ivd_fill_mem_rec_ip_t.u4_size =
+ sizeof(ivdext_fill_mem_rec_ip_t);
+ s_fill_mem_ip.i4_level = i4_level;
+ s_fill_mem_ip.u4_num_reorder_frames = u4_num_reorder_frames;
+ s_fill_mem_ip.u4_num_ref_frames = u4_num_ref_frames;
+ s_fill_mem_ip.u4_share_disp_buf = u4_share_disp_buf;
+ s_fill_mem_ip.u4_num_extra_disp_buf = 0;
+ s_fill_mem_ip.e_output_format = mIvColorFormat;
+
+ s_fill_mem_ip.s_ivd_fill_mem_rec_ip_t.e_cmd = IV_CMD_FILL_NUM_MEM_REC;
+ s_fill_mem_ip.s_ivd_fill_mem_rec_ip_t.pv_mem_rec_location = mMemRecords;
+ s_fill_mem_ip.s_ivd_fill_mem_rec_ip_t.u4_max_frm_wd = mWidth;
+ s_fill_mem_ip.s_ivd_fill_mem_rec_ip_t.u4_max_frm_ht = mHeight;
+ s_fill_mem_op.s_ivd_fill_mem_rec_op_t.u4_size =
+ sizeof(ivdext_fill_mem_rec_op_t);
+
+ ps_mem_rec = mMemRecords;
+ for (i = 0; i < mNumMemRecords; i++)
+ ps_mem_rec[i].u4_size = sizeof(iv_mem_rec_t);
+
+ status = ivdec_api_function(mCodecCtx, (void *)&s_fill_mem_ip,
+ (void *)&s_fill_mem_op);
+
+ if (IV_SUCCESS != status) {
+ ALOGE("Error in filling mem records: 0x%x",
+ s_fill_mem_op.s_ivd_fill_mem_rec_op_t.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+ mNumMemRecords =
+ s_fill_mem_op.s_ivd_fill_mem_rec_op_t.u4_num_mem_rec_filled;
+
+ ps_mem_rec = mMemRecords;
+
+ for (i = 0; i < mNumMemRecords; i++) {
+ ps_mem_rec->pv_base = ivd_aligned_malloc(
+ ps_mem_rec->u4_mem_alignment, ps_mem_rec->u4_mem_size);
+ if (ps_mem_rec->pv_base == NULL) {
+ ALOGE("Allocation failure for memory record #%zu of size %u",
+ i, ps_mem_rec->u4_mem_size);
+ status = IV_FAIL;
+ return NO_MEMORY;
+ }
+
+ ps_mem_rec++;
+ }
+ }
+
+ /* Initialize the decoder */
+ {
+ ivdext_init_ip_t s_init_ip;
+ ivdext_init_op_t s_init_op;
+
+ void *dec_fxns = (void *)ivdec_api_function;
+
+ s_init_ip.s_ivd_init_ip_t.u4_size = sizeof(ivdext_init_ip_t);
+ s_init_ip.s_ivd_init_ip_t.e_cmd = (IVD_API_COMMAND_TYPE_T)IV_CMD_INIT;
+ s_init_ip.s_ivd_init_ip_t.pv_mem_rec_location = mMemRecords;
+ s_init_ip.s_ivd_init_ip_t.u4_frm_max_wd = mWidth;
+ s_init_ip.s_ivd_init_ip_t.u4_frm_max_ht = mHeight;
+
+ s_init_ip.i4_level = i4_level;
+ s_init_ip.u4_num_reorder_frames = u4_num_reorder_frames;
+ s_init_ip.u4_num_ref_frames = u4_num_ref_frames;
+ s_init_ip.u4_share_disp_buf = u4_share_disp_buf;
+ s_init_ip.u4_num_extra_disp_buf = 0;
+
+ s_init_op.s_ivd_init_op_t.u4_size = sizeof(s_init_op);
+
+ s_init_ip.s_ivd_init_ip_t.u4_num_mem_rec = mNumMemRecords;
+ s_init_ip.s_ivd_init_ip_t.e_output_format = mIvColorFormat;
+
+ mCodecCtx = (iv_obj_t*)mMemRecords[0].pv_base;
+ mCodecCtx->pv_fxns = dec_fxns;
+ mCodecCtx->u4_size = sizeof(iv_obj_t);
+
+ ALOGD("Initializing decoder");
+ status = ivdec_api_function(mCodecCtx, (void *)&s_init_ip,
+ (void *)&s_init_op);
+ if (status != IV_SUCCESS) {
+ ALOGE("Error in init: 0x%x",
+ s_init_op.s_ivd_init_op_t.u4_error_code);
+ return UNKNOWN_ERROR;
+ }
+ }
+
+ /* Reset the plugin state */
+ resetPlugin();
+
+ /* Set the run time (dynamic) parameters */
+ setParams(0, IVD_DECODE_FRAME);
+
+ /* Set number of cores/threads to be used by the codec */
+ setNumCores();
+
+ /* Get codec version */
+ getVersion();
+
+ /* Allocate internal picture buffer */
+ mFlushOutBuffer = (uint8_t *)ivd_aligned_malloc(128, mStride * mHeight * 3 / 2);
+ if (NULL == mFlushOutBuffer) {
+ ALOGE("Could not allocate flushOutputBuffer of size %zu", mStride * mHeight * 3 / 2);
+ return NO_MEMORY;
+ }
+
+ return OK;
+}
+
+status_t SoftHEVC::deInitDecoder() {
+ size_t i;
+ iv_mem_rec_t *ps_mem_rec;
+ ps_mem_rec = mMemRecords;
+ ALOGD("Freeing codec memory");
+ for (i = 0; i < mNumMemRecords; i++) {
+ ivd_aligned_free(ps_mem_rec->pv_base);
+ ps_mem_rec++;
+ }
+
+ ivd_aligned_free(mMemRecords);
+ ivd_aligned_free(mFlushOutBuffer);
+ return OK;
+}
+
+void SoftHEVC::onReset() {
+ ALOGD("onReset called");
+ SoftVideoDecoderOMXComponent::onReset();
+
+ resetDecoder();
+ resetPlugin();
+}
+
+void SoftHEVC::onPortFlushCompleted(OMX_U32 portIndex) {
+ ALOGD("onPortFlushCompleted on port %d", portIndex);
+
+ /* Once the output buffers are flushed, ignore any buffers that are held in decoder */
+ if (kOutputPortIndex == portIndex) {
+ setFlushMode();
+
+ /* Reset the time stamp arrays */
+ memset(mTimeStamps, 0, sizeof(mTimeStamps));
+ memset(mTimeStampsValid, 0, sizeof(mTimeStampsValid));
+
+ while (true) {
+ ivd_video_decode_ip_t s_dec_ip;
+ ivd_video_decode_op_t s_dec_op;
+ IV_API_CALL_STATUS_T status;
+ size_t sizeY, sizeUV;
+
+ s_dec_ip.e_cmd = IVD_CMD_VIDEO_DECODE;
+
+ s_dec_ip.u4_ts = 0;
+ s_dec_ip.pv_stream_buffer = NULL;
+ s_dec_ip.u4_num_Bytes = 0;
+
+ s_dec_ip.u4_size = sizeof(ivd_video_decode_ip_t);
+ s_dec_op.u4_size = sizeof(ivd_video_decode_op_t);
+
+ sizeY = mStride * mHeight;
+ sizeUV = sizeY / 4;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[0] = sizeY;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[1] = sizeUV;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[2] = sizeUV;
+
+ s_dec_ip.s_out_buffer.pu1_bufs[0] = mFlushOutBuffer;
+ s_dec_ip.s_out_buffer.pu1_bufs[1] =
+ s_dec_ip.s_out_buffer.pu1_bufs[0] + sizeY;
+ s_dec_ip.s_out_buffer.pu1_bufs[2] =
+ s_dec_ip.s_out_buffer.pu1_bufs[1] + sizeUV;
+ s_dec_ip.s_out_buffer.u4_num_bufs = 3;
+
+ status = ivdec_api_function(mCodecCtx, (void *)&s_dec_ip,
+ (void *)&s_dec_op);
+ if (0 == s_dec_op.u4_output_present) {
+ resetPlugin();
+ break;
+ }
+ }
+ }
+}
+
+void SoftHEVC::onQueueFilled(OMX_U32 portIndex) {
+ IV_API_CALL_STATUS_T status;
+
+ UNUSED(portIndex);
+
+ if (mOutputPortSettingsChange != NONE) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(kInputPortIndex);
+ List<BufferInfo *> &outQueue = getPortQueue(kOutputPortIndex);
+
+ /* If input EOS is seen and decoder is not in flush mode,
+ * set the decoder in flush mode.
+ * There can be a case where EOS is sent along with last picture data
+ * In that case, only after decoding that input data, decoder has to be
+ * put in flush. This case is handled here */
+
+ if (mReceivedEOS && !mIsInFlush) {
+ setFlushMode();
+ }
+
+ while (outQueue.size() == kNumBuffers) {
+ BufferInfo *inInfo;
+ OMX_BUFFERHEADERTYPE *inHeader;
+
+ BufferInfo *outInfo;
+ OMX_BUFFERHEADERTYPE *outHeader;
+ size_t timeStampIx;
+
+ inInfo = NULL;
+ inHeader = NULL;
+
+ if (!mIsInFlush) {
+ if (!inQueue.empty()) {
+ inInfo = *inQueue.begin();
+ inHeader = inInfo->mHeader;
+ } else {
+ break;
+ }
+ }
+
+ outInfo = *outQueue.begin();
+ outHeader = outInfo->mHeader;
+ outHeader->nFlags = 0;
+ outHeader->nTimeStamp = 0;
+ outHeader->nOffset = 0;
+
+ if (inHeader != NULL && (inHeader->nFlags & OMX_BUFFERFLAG_EOS)) {
+ ALOGD("EOS seen on input");
+ mReceivedEOS = true;
+ if (inHeader->nFilledLen == 0) {
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ setFlushMode();
+ }
+ }
+
+ /* Get a free slot in timestamp array to hold input timestamp */
+ {
+ size_t i;
+ timeStampIx = 0;
+ for (i = 0; i < MAX_TIME_STAMPS; i++) {
+ if (!mTimeStampsValid[i]) {
+ timeStampIx = i;
+ break;
+ }
+ }
+ if (inHeader != NULL) {
+ mTimeStampsValid[timeStampIx] = true;
+ mTimeStamps[timeStampIx] = inHeader->nTimeStamp;
+ }
+ }
+
+ {
+ ivd_video_decode_ip_t s_dec_ip;
+ ivd_video_decode_op_t s_dec_op;
+ WORD32 timeDelay, timeTaken;
+ size_t sizeY, sizeUV;
+
+ s_dec_ip.e_cmd = IVD_CMD_VIDEO_DECODE;
+
+ /* When in flush and after EOS with zero byte input,
+ * inHeader is set to zero. Hence check for non-null */
+ if (inHeader != NULL) {
+ s_dec_ip.u4_ts = timeStampIx;
+ s_dec_ip.pv_stream_buffer = inHeader->pBuffer
+ + inHeader->nOffset;
+ s_dec_ip.u4_num_Bytes = inHeader->nFilledLen;
+ } else {
+ s_dec_ip.u4_ts = 0;
+ s_dec_ip.pv_stream_buffer = NULL;
+ s_dec_ip.u4_num_Bytes = 0;
+ }
+
+ s_dec_ip.u4_size = sizeof(ivd_video_decode_ip_t);
+ s_dec_op.u4_size = sizeof(ivd_video_decode_op_t);
+
+ sizeY = mStride * mHeight;
+ sizeUV = sizeY / 4;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[0] = sizeY;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[1] = sizeUV;
+ s_dec_ip.s_out_buffer.u4_min_out_buf_size[2] = sizeUV;
+
+ s_dec_ip.s_out_buffer.pu1_bufs[0] = outHeader->pBuffer;
+ s_dec_ip.s_out_buffer.pu1_bufs[1] =
+ s_dec_ip.s_out_buffer.pu1_bufs[0] + sizeY;
+ s_dec_ip.s_out_buffer.pu1_bufs[2] =
+ s_dec_ip.s_out_buffer.pu1_bufs[1] + sizeUV;
+ s_dec_ip.s_out_buffer.u4_num_bufs = 3;
+
+ GETTIME(&mTimeStart, NULL);
+ /* Compute time elapsed between end of previous decode()
+ * to start of current decode() */
+ TIME_DIFF(mTimeEnd, mTimeStart, timeDelay);
+
+ status = ivdec_api_function(mCodecCtx, (void *)&s_dec_ip,
+ (void *)&s_dec_op);
+
+ GETTIME(&mTimeEnd, NULL);
+ /* Compute time taken for decode() */
+ TIME_DIFF(mTimeStart, mTimeEnd, timeTaken);
+
+ ALOGD("timeTaken=%6d delay=%6d numBytes=%6d", timeTaken, timeDelay,
+ s_dec_op.u4_num_bytes_consumed);
+
+ if ((inHeader != NULL) && (1 != s_dec_op.u4_frame_decoded_flag)) {
+ /* If the input did not contain picture data, then ignore
+ * the associated timestamp */
+ mTimeStampsValid[timeStampIx] = false;
+ }
+
+ /* If valid height and width are decoded,
+ * then look at change in resolution */
+ if ((0 < s_dec_op.u4_pic_wd) && (0 < s_dec_op.u4_pic_ht)) {
+ uint32_t width = s_dec_op.u4_pic_wd;
+ uint32_t height = s_dec_op.u4_pic_ht;
+
+ if ((width != mWidth || height != mHeight)) {
+ mWidth = width;
+ mHeight = height;
+ mStride = mWidth;
+
+ /* If width and height are greater than the
+ * the dimensions used during codec create, then
+ * delete the current instance and recreate an instance with
+ * new dimensions */
+ /* TODO: The following does not work currently, since the decoder
+ * currently returns 0 x 0 as width height when it is not supported
+ * Once the decoder is updated to return actual width and height,
+ * then this can be validated*/
+
+ if ((mWidth * mHeight) > (mInitWidth * mInitHeight)) {
+ status_t ret;
+ ALOGD("Trying reInit");
+ ret = deInitDecoder();
+ if (OK != ret) {
+ // TODO: Handle graceful exit
+ ALOGE("Create failure");
+ return;
+ }
+
+ mInitWidth = mWidth;
+ mInitHeight = mHeight;
+
+ ret = initDecoder();
+ if (OK != ret) {
+ // TODO: Handle graceful exit
+ ALOGE("Create failure");
+ return;
+ }
+ }
+ updatePortDefinitions();
+
+ notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
+ mOutputPortSettingsChange = AWAITING_DISABLED;
+ return;
+ }
+ }
+
+ if (s_dec_op.u4_output_present) {
+ outHeader->nFilledLen = (mStride * mHeight * 3) / 2;
+
+ outHeader->nTimeStamp = mTimeStamps[s_dec_op.u4_ts];
+ mTimeStampsValid[s_dec_op.u4_ts] = false;
+
+ outInfo->mOwnedByUs = false;
+ outQueue.erase(outQueue.begin());
+ outInfo = NULL;
+ notifyFillBufferDone(outHeader);
+ outHeader = NULL;
+ } else {
+ /* If in flush mode and no output is returned by the codec,
+ * then come out of flush mode */
+ mIsInFlush = false;
+
+ /* If EOS was recieved on input port and there is no output
+ * from the codec, then signal EOS on output port */
+ if (mReceivedEOS) {
+ outHeader->nFilledLen = 0;
+ outHeader->nFlags |= OMX_BUFFERFLAG_EOS;
+
+ outInfo->mOwnedByUs = false;
+ outQueue.erase(outQueue.begin());
+ outInfo = NULL;
+ notifyFillBufferDone(outHeader);
+ outHeader = NULL;
+ resetPlugin();
+ }
+ }
+ }
+
+ // TODO: Handle more than one picture data
+ if (inHeader != NULL) {
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ inInfo = NULL;
+ notifyEmptyBufferDone(inHeader);
+ inHeader = NULL;
+ }
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(const char *name,
+ const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData,
+ OMX_COMPONENTTYPE **component) {
+ return new android::SoftHEVC(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/hevcdec/SoftHEVC.h b/media/libstagefright/codecs/hevcdec/SoftHEVC.h
new file mode 100644
index 0000000..20db0e1
--- /dev/null
+++ b/media/libstagefright/codecs/hevcdec/SoftHEVC.h
@@ -0,0 +1,117 @@
+/*
+ * 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 SOFT_HEVC_H_
+
+#define SOFT_HEVC_H_
+
+#include "SoftVideoDecoderOMXComponent.h"
+#include <sys/time.h>
+
+namespace android {
+
+#define ivd_aligned_malloc(alignment, size) memalign(alignment, size)
+#define ivd_aligned_free(buf) free(buf)
+
+/** Number of entries in the time-stamp array */
+#define MAX_TIME_STAMPS 64
+
+/** Maximum number of cores supported by the codec */
+#define CODEC_MAX_NUM_CORES 4
+
+#define CODEC_MAX_WIDTH 1920
+
+#define CODEC_MAX_HEIGHT 1088
+
+/** Input buffer size */
+#define INPUT_BUF_SIZE (1024 * 1024)
+
+#define MIN(a, b) ((a) < (b)) ? (a) : (b)
+
+/** Used to remove warnings about unused parameters */
+#define UNUSED(x) ((void)(x))
+
+/** Get time */
+#define GETTIME(a, b) gettimeofday(a, b);
+
+/** Compute difference between start and end */
+#define TIME_DIFF(start, end, diff) \
+ diff = ((end.tv_sec - start.tv_sec) * 1000000) + \
+ (end.tv_usec - start.tv_usec);
+
+struct SoftHEVC: public SoftVideoDecoderOMXComponent {
+ SoftHEVC(const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftHEVC();
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+ virtual void onPortFlushCompleted(OMX_U32 portIndex);
+ virtual void onReset();
+private:
+ // Number of input and output buffers
+ enum {
+ kNumBuffers = 8
+ };
+
+ iv_obj_t *mCodecCtx; // Codec context
+ iv_mem_rec_t *mMemRecords; // Memory records requested by the codec
+ size_t mNumMemRecords; // Number of memory records requested by the codec
+
+ uint32_t mNewWidth; // New width after change in resolution
+ uint32_t mNewHeight; // New height after change in resolution
+ uint32_t mInitWidth; // Width used during codec creation
+ uint32_t mInitHeight; // Height used during codec creation
+ size_t mStride; // Stride to be used for display buffers
+
+ size_t mNumCores; // Number of cores to be uesd by the codec
+
+ struct timeval mTimeStart; // Time at the start of decode()
+ struct timeval mTimeEnd; // Time at the end of decode()
+
+ // Internal buffer to be used to flush out the buffers from decoder
+ uint8_t *mFlushOutBuffer;
+
+ // Status of entries in the timestamp array
+ bool mTimeStampsValid[MAX_TIME_STAMPS];
+
+ // Timestamp array - Since codec does not take 64 bit timestamps,
+ // they are maintained in the plugin
+ OMX_S64 mTimeStamps[MAX_TIME_STAMPS];
+
+ OMX_COLOR_FORMATTYPE mOmxColorFormat; // OMX Color format
+ IV_COLOR_FORMAT_T mIvColorFormat; // Ittiam Color format
+
+ bool mIsInFlush; // codec is flush mode
+ bool mReceivedEOS; // EOS is receieved on input port
+ bool mIsAdapting; // plugin in middle of change in resolution
+
+ status_t initDecoder();
+ status_t deInitDecoder();
+ status_t setFlushMode();
+ status_t setParams(WORD32 stride, IVD_VIDEO_DECODE_MODE_T decMode);
+ status_t getVersion();
+ status_t setNumCores();
+ status_t resetDecoder();
+ status_t resetPlugin();
+
+ DISALLOW_EVIL_CONSTRUCTORS (SoftHEVC);
+};
+
+} // namespace android
+
+#endif // SOFT_HEVC_H_
diff --git a/media/libstagefright/data/media_codecs_google_audio.xml b/media/libstagefright/data/media_codecs_google_audio.xml
index b1f93de..f6db0cc 100644
--- a/media/libstagefright/data/media_codecs_google_audio.xml
+++ b/media/libstagefright/data/media_codecs_google_audio.xml
@@ -24,6 +24,7 @@
<MediaCodec name="OMX.google.g711.mlaw.decoder" type="audio/g711-mlaw" />
<MediaCodec name="OMX.google.vorbis.decoder" type="audio/vorbis" />
<MediaCodec name="OMX.google.opus.decoder" type="audio/opus" />
+ <MediaCodec name="OMX.google.raw.decoder" type="audio/raw" />
</Decoders>
<Encoders>
diff --git a/media/libstagefright/data/media_codecs_google_video.xml b/media/libstagefright/data/media_codecs_google_video.xml
index 41e0efb..9b930bc 100644
--- a/media/libstagefright/data/media_codecs_google_video.xml
+++ b/media/libstagefright/data/media_codecs_google_video.xml
@@ -19,6 +19,7 @@
<MediaCodec name="OMX.google.mpeg4.decoder" type="video/mp4v-es" />
<MediaCodec name="OMX.google.h263.decoder" type="video/3gpp" />
<MediaCodec name="OMX.google.h264.decoder" type="video/avc" />
+ <MediaCodec name="OMX.google.hevc.decoder" type="video/hevc" />
<MediaCodec name="OMX.google.vp8.decoder" type="video/x-vnd.on2.vp8" />
<MediaCodec name="OMX.google.vp9.decoder" type="video/x-vnd.on2.vp9" />
</Decoders>
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index 65f5404..9b6958a 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -42,6 +42,7 @@
{ "OMX.google.amrwb.encoder", "amrwbenc", "audio_encoder.amrwb" },
{ "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },
{ "OMX.google.h264.encoder", "h264enc", "video_encoder.avc" },
+ { "OMX.google.hevc.decoder", "hevcdec", "video_decoder.hevc" },
{ "OMX.google.g711.alaw.decoder", "g711dec", "audio_decoder.g711alaw" },
{ "OMX.google.g711.mlaw.decoder", "g711dec", "audio_decoder.g711mlaw" },
{ "OMX.google.h263.decoder", "mpeg4dec", "video_decoder.h263" },
diff --git a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
index eb9fcf7..1c383f7 100644
--- a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
@@ -183,12 +183,12 @@
return OMX_ErrorUnsupportedIndex;
}
- if (index >= mNumProfileLevels) {
+ if (profileLevel->nProfileIndex >= mNumProfileLevels) {
return OMX_ErrorNoMore;
}
- profileLevel->eProfile = mProfileLevels[index].mProfile;
- profileLevel->eLevel = mProfileLevels[index].mLevel;
+ profileLevel->eProfile = mProfileLevels[profileLevel->nProfileIndex].mProfile;
+ profileLevel->eLevel = mProfileLevels[profileLevel->nProfileIndex].mLevel;
return OMX_ErrorNone;
}
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index f7b6f64..0bdf5a3 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -64,6 +64,7 @@
LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
+LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp
LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 457ac3d..a4cad4e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -169,7 +169,8 @@
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
- mGlobalEffectEnableTime(0)
+ mGlobalEffectEnableTime(0),
+ mPrimaryOutputSampleRate(0)
{
getpid_cached = getpid();
char value[PROPERTY_VALUE_MAX];
@@ -1265,7 +1266,7 @@
: RefBase(),
mAudioFlinger(audioFlinger),
// FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
- mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
+ mMemoryDealer(new MemoryDealer(4 * 1024*1024, "AudioFlinger::Client")),
mPid(pid),
mTimedTrackCount(0)
{
@@ -1679,6 +1680,8 @@
mHardwareStatus = AUDIO_HW_SET_MODE;
hwDevHal->set_mode(hwDevHal, mMode);
mHardwareStatus = AUDIO_HW_IDLE;
+
+ mPrimaryOutputSampleRate = config.sample_rate;
}
return id;
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 29dc6b2..6e73a14 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -50,6 +50,8 @@
#include <media/AudioBufferProvider.h>
#include <media/ExtendedAudioBufferProvider.h>
+
+#include "FastCapture.h"
#include "FastMixer.h"
#include <media/nbaio/NBAIO.h>
#include "AudioWatchdog.h"
@@ -691,6 +693,9 @@
nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled
sp<PatchPanel> mPatchPanel;
+
+ uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none
+ // protected by mHardwareLock
};
#undef INCLUDING_FROM_AUDIOFLINGER_H
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 805eaa4..d73292e 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -34,6 +34,7 @@
#include <system/audio.h>
#include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
#include <common_time/local_clock.h>
#include <common_time/cc_helper.h>
@@ -88,6 +89,103 @@
}
}
+template <typename T>
+T min(const T& a, const T& b)
+{
+ return a < b ? a : b;
+}
+
+AudioMixer::ReformatBufferProvider::ReformatBufferProvider(int32_t channels,
+ audio_format_t inputFormat, audio_format_t outputFormat) :
+ mTrackBufferProvider(NULL),
+ mChannels(channels),
+ mInputFormat(inputFormat),
+ mOutputFormat(outputFormat),
+ mInputFrameSize(channels * audio_bytes_per_sample(inputFormat)),
+ mOutputFrameSize(channels * audio_bytes_per_sample(outputFormat)),
+ mOutputData(NULL),
+ mOutputCount(0),
+ mConsumed(0)
+{
+ ALOGV("ReformatBufferProvider(%p)(%d, %#x, %#x)", this, channels, inputFormat, outputFormat);
+ if (requiresInternalBuffers()) {
+ mOutputCount = 256;
+ (void)posix_memalign(&mOutputData, 32, mOutputCount * mOutputFrameSize);
+ }
+ mBuffer.frameCount = 0;
+}
+
+AudioMixer::ReformatBufferProvider::~ReformatBufferProvider()
+{
+ ALOGV("~ReformatBufferProvider(%p)", this);
+ if (mBuffer.frameCount != 0) {
+ mTrackBufferProvider->releaseBuffer(&mBuffer);
+ }
+ free(mOutputData);
+}
+
+status_t AudioMixer::ReformatBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
+ int64_t pts) {
+ //ALOGV("ReformatBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
+ // this, pBuffer, pBuffer->frameCount, pts);
+ if (!requiresInternalBuffers()) {
+ status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+ if (res == OK) {
+ memcpy_by_audio_format(pBuffer->raw, mOutputFormat, pBuffer->raw, mInputFormat,
+ pBuffer->frameCount * mChannels);
+ }
+ return res;
+ }
+ if (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = pBuffer->frameCount;
+ status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+ // TODO: Track down a bug in the upstream provider
+ // LOG_ALWAYS_FATAL_IF(res == OK && mBuffer.frameCount == 0,
+ // "ReformatBufferProvider::getNextBuffer():"
+ // " Invalid zero framecount returned from getNextBuffer()");
+ if (res != OK || mBuffer.frameCount == 0) {
+ pBuffer->raw = NULL;
+ pBuffer->frameCount = 0;
+ return res;
+ }
+ }
+ ALOG_ASSERT(mConsumed < mBuffer.frameCount);
+ size_t count = min(mOutputCount, mBuffer.frameCount - mConsumed);
+ count = min(count, pBuffer->frameCount);
+ pBuffer->raw = mOutputData;
+ pBuffer->frameCount = count;
+ //ALOGV("reformatting %d frames from %#x to %#x, %d chan",
+ // pBuffer->frameCount, mInputFormat, mOutputFormat, mChannels);
+ memcpy_by_audio_format(pBuffer->raw, mOutputFormat,
+ (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize, mInputFormat,
+ pBuffer->frameCount * mChannels);
+ return OK;
+}
+
+void AudioMixer::ReformatBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
+ //ALOGV("ReformatBufferProvider(%p)::releaseBuffer(%p(%zu))",
+ // this, pBuffer, pBuffer->frameCount);
+ if (!requiresInternalBuffers()) {
+ mTrackBufferProvider->releaseBuffer(pBuffer);
+ return;
+ }
+ // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+ mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
+ if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
+ mConsumed = 0;
+ mTrackBufferProvider->releaseBuffer(&mBuffer);
+ // ALOG_ASSERT(mBuffer.frameCount == 0);
+ }
+ pBuffer->raw = NULL;
+ pBuffer->frameCount = 0;
+}
+
+void AudioMixer::ReformatBufferProvider::reset() {
+ if (mBuffer.frameCount != 0) {
+ mTrackBufferProvider->releaseBuffer(&mBuffer);
+ }
+ mConsumed = 0;
+}
// ----------------------------------------------------------------------------
bool AudioMixer::sIsMultichannelCapable = false;
@@ -153,8 +251,13 @@
mState.mLog = log;
}
-int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
+int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId)
{
+ if (!isValidPcmTrackFormat(format)) {
+ ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
+ return -1;
+ }
uint32_t names = (~mTrackNames) & mConfiguredNames;
if (names != 0) {
int n = __builtin_ctz(names);
@@ -162,8 +265,8 @@
// assume default parameters for the track, except where noted below
track_t* t = &mState.tracks[n];
t->needs = 0;
- t->volume[0] = UNITY_GAIN;
- t->volume[1] = UNITY_GAIN;
+ t->volume[0] = UNITY_GAIN_INT;
+ t->volume[1] = UNITY_GAIN_INT;
// no initialization needed
// t->prevVolume[0]
// t->prevVolume[1]
@@ -176,7 +279,8 @@
// t->frameCount
t->channelCount = audio_channel_count_from_out_mask(channelMask);
t->enabled = false;
- t->format = 16;
+ ALOGV_IF(channelMask != AUDIO_CHANNEL_OUT_STEREO,
+ "Non-stereo channel mask: %d\n", channelMask);
t->channelMask = channelMask;
t->sessionId = sessionId;
// setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
@@ -191,9 +295,15 @@
// setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->mInputBufferProvider = NULL;
+ t->mReformatBufferProvider = NULL;
t->downmixerBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
-
+ t->mFormat = format;
+ t->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT;
+ if (t->mFormat != t->mMixerInFormat) {
+ prepareTrackForReformat(t, n);
+ }
status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
@@ -237,9 +347,9 @@
if (pTrack->downmixerBufferProvider != NULL) {
// this track had previously been configured with a downmixer, delete it
ALOGV(" deleting old downmixer");
- pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider;
delete pTrack->downmixerBufferProvider;
pTrack->downmixerBufferProvider = NULL;
+ reconfigureBufferProviders(pTrack);
} else {
ALOGV(" nothing to do, no downmixer to delete");
}
@@ -333,21 +443,51 @@
}// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack"
// initialization successful:
- // - keep track of the real buffer provider in case it was set before
- pDbp->mTrackBufferProvider = pTrack->bufferProvider;
- // - we'll use the downmix effect integrated inside this
- // track's buffer provider, and we'll use it as the track's buffer provider
pTrack->downmixerBufferProvider = pDbp;
- pTrack->bufferProvider = pDbp;
-
+ reconfigureBufferProviders(pTrack);
return NO_ERROR;
noDownmixForActiveTrack:
delete pDbp;
pTrack->downmixerBufferProvider = NULL;
+ reconfigureBufferProviders(pTrack);
return NO_INIT;
}
+void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
+ ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
+ if (pTrack->mReformatBufferProvider != NULL) {
+ delete pTrack->mReformatBufferProvider;
+ pTrack->mReformatBufferProvider = NULL;
+ reconfigureBufferProviders(pTrack);
+ }
+}
+
+status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
+{
+ ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
+ // discard the previous reformatter if there was one
+ unprepareTrackForReformat(pTrack, trackName);
+ pTrack->mReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(pTrack->channelMask),
+ pTrack->mFormat, pTrack->mMixerInFormat);
+ reconfigureBufferProviders(pTrack);
+ return NO_ERROR;
+}
+
+void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
+{
+ pTrack->bufferProvider = pTrack->mInputBufferProvider;
+ if (pTrack->mReformatBufferProvider) {
+ pTrack->mReformatBufferProvider->mTrackBufferProvider = pTrack->bufferProvider;
+ pTrack->bufferProvider = pTrack->mReformatBufferProvider;
+ }
+ if (pTrack->downmixerBufferProvider) {
+ pTrack->downmixerBufferProvider->mTrackBufferProvider = pTrack->bufferProvider;
+ pTrack->bufferProvider = pTrack->downmixerBufferProvider;
+ }
+}
+
void AudioMixer::deleteTrackName(int name)
{
ALOGV("AudioMixer::deleteTrackName(%d)", name);
@@ -364,6 +504,8 @@
track.resampler = NULL;
// delete the downmixer
unprepareTrackForDownmix(&mState.tracks[name], name);
+ // delete the reformatter
+ unprepareTrackForReformat(&mState.tracks[name], name);
mTrackNames &= ~(1<<name);
}
@@ -394,6 +536,44 @@
}
}
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition between the previous
+ * volume to the target volume. The duration of the transition is
+ * set by ramp, which is either 0 for immediate, or typically one state
+ * framecount period.
+ *
+ * @param newFloatValue new volume target in float [0.0, 1.0].
+ * @param ramp number of frames to increment over. ramp is 0 if the volume
+ * should be set immediately.
+ * @param volume reference to the U4.12 target volume, set on return.
+ * @param prevVolume reference to the U4.27 previous volume, set on return.
+ * @param volumeInc reference to the increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newFloatValue, int32_t ramp,
+ int16_t &volume, int32_t &prevVolume, int32_t &volumeInc) {
+ int32_t newValue = newFloatValue * AudioMixer::UNITY_GAIN_INT;
+ if (newValue > AudioMixer::UNITY_GAIN_INT) {
+ newValue = AudioMixer::UNITY_GAIN_INT;
+ } else if (newValue < 0) {
+ ALOGE("negative volume %.7g", newFloatValue);
+ newValue = 0; // should never happen, but for safety check.
+ }
+ if (newValue == volume) {
+ return false;
+ }
+ if (ramp != 0) {
+ volumeInc = ((newValue - volume) << 16) / ramp;
+ prevVolume = (volumeInc == 0 ? newValue : volume) << 16;
+ } else {
+ volumeInc = 0;
+ prevVolume = newValue << 16;
+ }
+ volume = newValue;
+ return true;
+}
+
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
name -= TRACK0;
@@ -435,9 +615,20 @@
invalidateState(1 << name);
}
break;
- case FORMAT:
- ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
- break;
+ case FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track.mFormat != format) {
+ ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+ track.mFormat = format;
+ ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+ //if (track.mFormat != track.mMixerInFormat)
+ {
+ ALOGD("Reformatting!");
+ prepareTrackForReformat(&track, name);
+ }
+ invalidateState(1 << name);
+ }
+ } break;
// FIXME do we want to support setting the downmix type from AudioFlinger?
// for a specific track? or per mixer?
/* case DOWNMIX_TYPE:
@@ -484,41 +675,23 @@
switch (param) {
case VOLUME0:
case VOLUME1:
- if (track.volume[param-VOLUME0] != valueInt) {
- ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
- track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
- track.volume[param-VOLUME0] = valueInt;
- if (target == VOLUME) {
- track.prevVolume[param-VOLUME0] = valueInt << 16;
- track.volumeInc[param-VOLUME0] = 0;
- } else {
- int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
- int32_t volInc = d / int32_t(mState.frameCount);
- track.volumeInc[param-VOLUME0] = volInc;
- if (volInc == 0) {
- track.prevVolume[param-VOLUME0] = valueInt << 16;
- }
- }
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mState.frameCount : 0,
+ track.volume[param - VOLUME0], track.prevVolume[param - VOLUME0],
+ track.volumeInc[param - VOLUME0])) {
+ ALOGV("setParameter(%s, VOLUME%d: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+ track.volume[param - VOLUME0]);
invalidateState(1 << name);
}
break;
case AUXLEVEL:
//ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt);
- if (track.auxLevel != valueInt) {
- ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
- track.prevAuxLevel = track.auxLevel << 16;
- track.auxLevel = valueInt;
- if (target == VOLUME) {
- track.prevAuxLevel = valueInt << 16;
- track.auxInc = 0;
- } else {
- int32_t d = (valueInt<<16) - track.prevAuxLevel;
- int32_t volInc = d / int32_t(mState.frameCount);
- track.auxInc = volInc;
- if (volInc == 0) {
- track.prevAuxLevel = valueInt << 16;
- }
- }
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mState.frameCount : 0,
+ track.auxLevel, track.prevAuxLevel, track.auxInc)) {
+ ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
invalidateState(1 << name);
}
break;
@@ -550,8 +723,9 @@
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
+ const int bits = mMixerInFormat == AUDIO_FORMAT_PCM_16_BIT ? 16 : /* FLOAT */ 32;
resampler = AudioResampler::create(
- format,
+ bits,
// the resampler sees the number of channels after the downmixer, if any
(int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
devSampleRate, quality);
@@ -596,21 +770,16 @@
name -= TRACK0;
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
- if (mState.tracks[name].downmixerBufferProvider != NULL) {
- // update required?
- if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) {
- ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider);
- // setting the buffer provider for a track that gets downmixed consists in:
- // 1/ setting the buffer provider to the "downmix / buffer provider" wrapper
- // so it's the one that gets called when the buffer provider is needed,
- mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider;
- // 2/ saving the buffer provider for the track so the wrapper can use it
- // when it downmixes.
- mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider;
- }
- } else {
- mState.tracks[name].bufferProvider = bufferProvider;
+ if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
+ return; // don't reset any buffer providers if identical.
}
+ if (mState.tracks[name].mReformatBufferProvider != NULL) {
+ mState.tracks[name].mReformatBufferProvider->reset();
+ } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+ }
+
+ mState.tracks[name].mInputBufferProvider = bufferProvider;
+ reconfigureBufferProviders(&mState.tracks[name]);
}
@@ -769,7 +938,7 @@
// always resample with unity gain when sending to auxiliary buffer to be able
// to apply send level after resampling
// TODO: modify each resampler to support aux channel?
- t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+ t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
@@ -779,7 +948,7 @@
}
} else {
if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
- t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+ t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
volumeRampStereo(t, out, outFrameCount, temp, aux);
@@ -1301,6 +1470,7 @@
AudioBufferProvider::Buffer& b(t.buffer);
int32_t* out = t.mainBuffer;
+ float *fout = reinterpret_cast<float*>(out);
size_t numFrames = state->frameCount;
const int16_t vl = t.volume[0];
@@ -1314,9 +1484,10 @@
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
- if (in == NULL || ((unsigned long)in & 3)) {
- memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
- ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: "
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ memset(out, 0, numFrames
+ * MAX_NUM_CHANNELS * audio_bytes_per_sample(t.mMixerFormat));
+ ALOGE_IF((((uintptr_t)in) & 3), "process stereo track: input buffer alignment pb: "
"buffer %p track %d, channels %d, needs %08x",
in, i, t.channelCount, t.needs);
return;
@@ -1324,8 +1495,7 @@
size_t outFrames = b.frameCount;
switch (t.mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT: {
- float *fout = reinterpret_cast<float*>(out);
+ case AUDIO_FORMAT_PCM_FLOAT:
do {
uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
@@ -1336,9 +1506,9 @@
// Note: In case of later int16_t sink output,
// conversion and clamping is done by memcpy_to_i16_from_float().
} while (--outFrames);
- } break;
+ break;
case AUDIO_FORMAT_PCM_16_BIT:
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 09e63a6..766ff60 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -31,7 +31,7 @@
#include <media/nbaio/NBLog.h>
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
namespace android {
@@ -58,7 +58,8 @@
// maximum number of channels supported for the content
static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
- static const uint16_t UNITY_GAIN = 0x1000;
+ static const uint16_t UNITY_GAIN_INT = 0x1000;
+ static const float UNITY_GAIN_FLOAT = 1.0f;
enum { // names
@@ -104,7 +105,10 @@
// For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
// Allocate a track name. Returns new track name if successful, -1 on failure.
- int getTrackName(audio_channel_mask_t channelMask, int sessionId);
+ // The failure could be because of an invalid channelMask or format, or that
+ // the track capacity of the mixer is exceeded.
+ int getTrackName(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId);
// Free an allocated track by name
void deleteTrackName(int name);
@@ -122,6 +126,13 @@
size_t getUnreleasedFrames(int name) const;
+ static inline bool isValidPcmTrackFormat(audio_format_t format) {
+ return format == AUDIO_FORMAT_PCM_16_BIT ||
+ format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+ format == AUDIO_FORMAT_PCM_32_BIT ||
+ format == AUDIO_FORMAT_PCM_FLOAT;
+ }
+
private:
enum {
@@ -143,6 +154,7 @@
struct state_t;
struct track_t;
class DownmixerBufferProvider;
+ class ReformatBufferProvider;
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
int32_t* aux);
@@ -170,7 +182,7 @@
uint16_t frameCount;
uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
- uint8_t format; // always 16
+ uint8_t unused_padding; // formerly format, was always 16
uint16_t enabled; // actually bool
audio_channel_mask_t channelMask;
@@ -193,14 +205,19 @@
int32_t* auxBuffer;
// 16-byte boundary
-
+ AudioBufferProvider* mInputBufferProvider; // 4 bytes
+ ReformatBufferProvider* mReformatBufferProvider; // 4 bytes
DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
int32_t sessionId;
- audio_format_t mMixerFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ // 16-byte boundary
+ audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ audio_format_t mFormat; // input track format
+ audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ // each track must be converted to this format.
- int32_t padding[1];
+ int32_t mUnused[1]; // alignment padding
// 16-byte boundary
@@ -239,6 +256,35 @@
effect_config_t mDownmixConfig;
};
+ // AudioBufferProvider wrapper that reformats track to acceptable mixer input type
+ class ReformatBufferProvider : public AudioBufferProvider {
+ public:
+ ReformatBufferProvider(int32_t channels,
+ audio_format_t inputFormat, audio_format_t outputFormat);
+ virtual ~ReformatBufferProvider();
+
+ // overrides AudioBufferProvider methods
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
+ virtual void releaseBuffer(Buffer* buffer);
+
+ void reset();
+ inline bool requiresInternalBuffers() {
+ return true; //mInputFrameSize < mOutputFrameSize;
+ }
+
+ AudioBufferProvider* mTrackBufferProvider;
+ int32_t mChannels;
+ audio_format_t mInputFormat;
+ audio_format_t mOutputFormat;
+ size_t mInputFrameSize;
+ size_t mOutputFrameSize;
+ // (only) required for reformatting to a larger size.
+ AudioBufferProvider::Buffer mBuffer;
+ void* mOutputData;
+ size_t mOutputCount;
+ size_t mConsumed;
+ };
+
// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
@@ -266,6 +312,9 @@
static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
+ static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
+ static void unprepareTrackForReformat(track_t* pTrack, int trackName);
+ static void reconfigureBufferProviders(track_t* pTrack);
static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
int32_t* aux);
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 3abe8fd..318eb57 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -455,13 +455,20 @@
const Constants& c(mConstants);
const TC* const coefs = mConstants.mFirCoefs;
TI* impulse = mInBuffer.getImpulse();
- size_t inputIndex = mInputIndex;
+ size_t inputIndex = 0;
uint32_t phaseFraction = mPhaseFraction;
const uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * 2; // stereo output
- size_t inFrameCount = getInFrameCountRequired(outFrameCount);
const uint32_t phaseWrapLimit = c.mL << c.mShift;
+ size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction)
+ / phaseWrapLimit;
+ // sanity check that inFrameCount is in signed 32 bit integer range.
+ ALOG_ASSERT(0 <= inFrameCount && inFrameCount < (1U << 31));
+
+ //ALOGV("inFrameCount:%d outFrameCount:%d"
+ // " phaseIncrement:%u phaseFraction:%u phaseWrapLimit:%u",
+ // inFrameCount, outFrameCount, phaseIncrement, phaseFraction, phaseWrapLimit);
// NOTE: be very careful when modifying the code here. register
// pressure is very high and a small change might cause the compiler
@@ -471,29 +478,39 @@
// 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) {
+ //ALOGV("LOOP: inFrameCount:%d outputIndex:%d outFrameCount:%d"
+ // " phaseFraction:%u phaseWrapLimit:%u",
+ // inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
+
+ // check inputIndex overflow
+ ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d",
+ inputIndex, mBuffer.frameCount);
+ // Buffer is empty, fetch a new one if necessary (inFrameCount > 0).
+ // We may not fetch a new buffer if the existing data is sufficient.
+ while (mBuffer.frameCount == 0 && inFrameCount > 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto resample_exit;
}
+ inFrameCount -= mBuffer.frameCount;
if (phaseFraction >= phaseWrapLimit) { // read in data
mInBuffer.template readAdvance<CHANNELS>(
impulse, c.mHalfNumCoefs,
reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
+ inputIndex++;
phaseFraction -= phaseWrapLimit;
while (phaseFraction >= phaseWrapLimit) {
- inputIndex++;
if (inputIndex >= mBuffer.frameCount) {
- inputIndex -= mBuffer.frameCount;
+ inputIndex = 0;
provider->releaseBuffer(&mBuffer);
break;
}
mInBuffer.template readAdvance<CHANNELS>(
impulse, c.mHalfNumCoefs,
reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
+ inputIndex++;
phaseFraction -= phaseWrapLimit;
}
}
@@ -504,9 +521,6 @@
const int halfNumCoefs = c.mHalfNumCoefs;
const TO* const volumeSimd = mVolumeSimd;
- // reread the last input in.
- mInBuffer.template readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
-
// main processing loop
while (CC_LIKELY(outputIndex < outputSampleCount)) {
// caution: fir() is inlined and may be large.
@@ -515,6 +529,10 @@
// from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs]
// from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs.
//
+ //ALOGV("LOOP2: inFrameCount:%d outputIndex:%d outFrameCount:%d"
+ // " phaseFraction:%u phaseWrapLimit:%u",
+ // inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
+ ALOG_ASSERT(phaseFraction < phaseWrapLimit);
fir<CHANNELS, LOCKED, STRIDE>(
&out[outputIndex],
phaseFraction, phaseWrapLimit,
@@ -524,26 +542,34 @@
phaseFraction += phaseIncrement;
while (phaseFraction >= phaseWrapLimit) {
- inputIndex++;
if (inputIndex >= frameCount) {
goto done; // need a new buffer
}
mInBuffer.template readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
+ inputIndex++;
phaseFraction -= phaseWrapLimit;
}
}
done:
- // often arrives here when input buffer runs out
- if (inputIndex >= frameCount) {
- inputIndex -= frameCount;
+ // We arrive here when we're finished or when the input buffer runs out.
+ // Regardless we need to release the input buffer if we've acquired it.
+ if (inputIndex > 0) { // we've acquired a buffer (alternatively could check frameCount)
+ ALOG_ASSERT(inputIndex == frameCount, "inputIndex(%d) != frameCount(%d)",
+ inputIndex, frameCount); // must have been fully read.
+ inputIndex = 0;
provider->releaseBuffer(&mBuffer);
- // mBuffer.frameCount MUST be zero here.
+ ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
resample_exit:
+ // inputIndex must be zero in all three cases:
+ // (1) the buffer never was been acquired; (2) the buffer was
+ // released at "done:"; or (3) getNextBuffer() failed.
+ ALOG_ASSERT(inputIndex == 0, "Releasing: inputindex:%d frameCount:%d phaseFraction:%u",
+ inputIndex, mBuffer.frameCount, phaseFraction);
+ ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
mInBuffer.setImpulse(impulse);
- mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
}
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
new file mode 100644
index 0000000..0c9b976
--- /dev/null
+++ b/services/audioflinger/FastCapture.cpp
@@ -0,0 +1,222 @@
+/*
+ * 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_TAG "FastCapture"
+//#define LOG_NDEBUG 0
+
+#define ATRACE_TAG ATRACE_TAG_AUDIO
+
+#include "Configuration.h"
+#include <linux/futex.h>
+#include <sys/syscall.h>
+#include <media/AudioBufferProvider.h>
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include "FastCapture.h"
+
+namespace android {
+
+/*static*/ const FastCaptureState FastCapture::initial;
+
+FastCapture::FastCapture() : FastThread(),
+ inputSource(NULL), inputSourceGen(0), pipeSink(NULL), pipeSinkGen(0),
+ readBuffer(NULL), readBufferState(-1), format(Format_Invalid), sampleRate(0),
+ // dummyDumpState
+ totalNativeFramesRead(0)
+{
+ previous = &initial;
+ current = &initial;
+
+ mDummyDumpState = &dummyDumpState;
+}
+
+FastCapture::~FastCapture()
+{
+}
+
+FastCaptureStateQueue* FastCapture::sq()
+{
+ return &mSQ;
+}
+
+const FastThreadState *FastCapture::poll()
+{
+ return mSQ.poll();
+}
+
+void FastCapture::setLog(NBLog::Writer *logWriter __unused)
+{
+}
+
+void FastCapture::onIdle()
+{
+ preIdle = *(const FastCaptureState *)current;
+ current = &preIdle;
+}
+
+void FastCapture::onExit()
+{
+ delete[] readBuffer;
+}
+
+bool FastCapture::isSubClassCommand(FastThreadState::Command command)
+{
+ switch ((FastCaptureState::Command) command) {
+ case FastCaptureState::READ:
+ case FastCaptureState::WRITE:
+ case FastCaptureState::READ_WRITE:
+ return true;
+ default:
+ return false;
+ }
+}
+
+void FastCapture::onStateChange()
+{
+ const FastCaptureState * const current = (const FastCaptureState *) this->current;
+ const FastCaptureState * const previous = (const FastCaptureState *) this->previous;
+ FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
+ const size_t frameCount = current->mFrameCount;
+
+ bool eitherChanged = false;
+
+ // check for change in input HAL configuration
+ NBAIO_Format previousFormat = format;
+ if (current->mInputSourceGen != inputSourceGen) {
+ inputSource = current->mInputSource;
+ inputSourceGen = current->mInputSourceGen;
+ if (inputSource == NULL) {
+ format = Format_Invalid;
+ sampleRate = 0;
+ } else {
+ format = inputSource->format();
+ sampleRate = Format_sampleRate(format);
+ unsigned channelCount = Format_channelCount(format);
+ ALOG_ASSERT(channelCount == 1 || channelCount == 2);
+ }
+ dumpState->mSampleRate = sampleRate;
+ eitherChanged = true;
+ }
+
+ // check for change in pipe
+ if (current->mPipeSinkGen != pipeSinkGen) {
+ pipeSink = current->mPipeSink;
+ pipeSinkGen = current->mPipeSinkGen;
+ eitherChanged = true;
+ }
+
+ // input source and pipe sink must be compatible
+ if (eitherChanged && inputSource != NULL && pipeSink != NULL) {
+ ALOG_ASSERT(Format_isEqual(format, pipeSink->format()));
+ }
+
+ if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
+ // FIXME to avoid priority inversion, don't delete here
+ delete[] readBuffer;
+ readBuffer = NULL;
+ if (frameCount > 0 && sampleRate > 0) {
+ // FIXME new may block for unbounded time at internal mutex of the heap
+ // implementation; it would be better to have normal capture thread allocate for
+ // us to avoid blocking here and to prevent possible priority inversion
+ unsigned channelCount = Format_channelCount(format);
+ // FIXME frameSize
+ readBuffer = new short[frameCount * channelCount];
+ periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00
+ underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75
+ overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50
+ forceNs = (frameCount * 950000000LL) / sampleRate; // 0.95
+ warmupNs = (frameCount * 500000000LL) / sampleRate; // 0.50
+ } else {
+ periodNs = 0;
+ underrunNs = 0;
+ overrunNs = 0;
+ forceNs = 0;
+ warmupNs = 0;
+ }
+ readBufferState = -1;
+ dumpState->mFrameCount = frameCount;
+ }
+
+}
+
+void FastCapture::onWork()
+{
+ const FastCaptureState * const current = (const FastCaptureState *) this->current;
+ FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
+ const FastCaptureState::Command command = this->command;
+ const size_t frameCount = current->mFrameCount;
+
+ if ((command & FastCaptureState::READ) /*&& isWarm*/) {
+ ALOG_ASSERT(inputSource != NULL);
+ ALOG_ASSERT(readBuffer != NULL);
+ dumpState->mReadSequence++;
+ ATRACE_BEGIN("read");
+ ssize_t framesRead = inputSource->read(readBuffer, frameCount,
+ AudioBufferProvider::kInvalidPTS);
+ ATRACE_END();
+ dumpState->mReadSequence++;
+ if (framesRead >= 0) {
+ LOG_ALWAYS_FATAL_IF((size_t) framesRead > frameCount);
+ totalNativeFramesRead += framesRead;
+ dumpState->mFramesRead = totalNativeFramesRead;
+ readBufferState = framesRead;
+ } else {
+ dumpState->mReadErrors++;
+ readBufferState = 0;
+ }
+ // FIXME rename to attemptedIO
+ attemptedWrite = true;
+ }
+
+ if (command & FastCaptureState::WRITE) {
+ ALOG_ASSERT(pipeSink != NULL);
+ ALOG_ASSERT(readBuffer != NULL);
+ if (readBufferState < 0) {
+ unsigned channelCount = Format_channelCount(format);
+ // FIXME frameSize
+ memset(readBuffer, 0, frameCount * channelCount * sizeof(short));
+ readBufferState = frameCount;
+ }
+ if (readBufferState > 0) {
+ ssize_t framesWritten = pipeSink->write(readBuffer, readBufferState);
+ // FIXME This supports at most one fast capture client.
+ // To handle multiple clients this could be converted to an array,
+ // or with a lot more work the control block could be shared by all clients.
+ audio_track_cblk_t* cblk = current->mCblk;
+ if (cblk != NULL && framesWritten > 0) {
+ int32_t rear = cblk->u.mStreaming.mRear;
+ android_atomic_release_store(framesWritten + rear, &cblk->u.mStreaming.mRear);
+ cblk->mServer += framesWritten;
+ int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
+ if (!(old & CBLK_FUTEX_WAKE)) {
+ // client is never in server process, so don't use FUTEX_WAKE_PRIVATE
+ (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, 1);
+ }
+ }
+ }
+ }
+}
+
+FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
+ mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
+{
+}
+
+FastCaptureDumpState::~FastCaptureDumpState()
+{
+}
+
+} // namespace android
diff --git a/services/audioflinger/FastCapture.h b/services/audioflinger/FastCapture.h
new file mode 100644
index 0000000..e535b9d
--- /dev/null
+++ b/services/audioflinger/FastCapture.h
@@ -0,0 +1,78 @@
+/*
+ * 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 ANDROID_AUDIO_FAST_CAPTURE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_H
+
+#include "FastThread.h"
+#include "StateQueue.h"
+#include "FastCaptureState.h"
+
+namespace android {
+
+typedef StateQueue<FastCaptureState> FastCaptureStateQueue;
+
+struct FastCaptureDumpState : FastThreadDumpState {
+ FastCaptureDumpState();
+ /*virtual*/ ~FastCaptureDumpState();
+
+ // FIXME by renaming, could pull up many of these to FastThreadDumpState
+ uint32_t mReadSequence; // incremented before and after each read()
+ uint32_t mFramesRead; // total number of frames read successfully
+ uint32_t mReadErrors; // total number of read() errors
+ uint32_t mSampleRate;
+ size_t mFrameCount;
+};
+
+class FastCapture : public FastThread {
+
+public:
+ FastCapture();
+ virtual ~FastCapture();
+
+ FastCaptureStateQueue* sq();
+
+private:
+ FastCaptureStateQueue mSQ;
+
+ // callouts
+ virtual const FastThreadState *poll();
+ virtual void setLog(NBLog::Writer *logWriter);
+ virtual void onIdle();
+ virtual void onExit();
+ virtual bool isSubClassCommand(FastThreadState::Command command);
+ virtual void onStateChange();
+ virtual void onWork();
+
+ static const FastCaptureState initial;
+ FastCaptureState preIdle; // copy of state before we went into idle
+ // FIXME by renaming, could pull up many of these to FastThread
+ NBAIO_Source *inputSource;
+ int inputSourceGen;
+ NBAIO_Sink *pipeSink;
+ int pipeSinkGen;
+ short *readBuffer;
+ ssize_t readBufferState; // number of initialized frames in readBuffer, or -1 to clear
+ NBAIO_Format format;
+ unsigned sampleRate;
+ FastCaptureDumpState dummyDumpState;
+ uint32_t totalNativeFramesRead; // copied to dumpState->mFramesRead
+
+}; // class FastCapture
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_FAST_CAPTURE_H
diff --git a/services/audioflinger/FastCaptureState.cpp b/services/audioflinger/FastCaptureState.cpp
new file mode 100644
index 0000000..1d029b7
--- /dev/null
+++ b/services/audioflinger/FastCaptureState.cpp
@@ -0,0 +1,30 @@
+/*
+ * 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.
+ */
+
+#include "FastCaptureState.h"
+
+namespace android {
+
+FastCaptureState::FastCaptureState() : FastThreadState(),
+ mInputSource(NULL), mInputSourceGen(0), mPipeSink(NULL), mPipeSinkGen(0), mFrameCount(0)
+{
+}
+
+FastCaptureState::~FastCaptureState()
+{
+}
+
+} // android
diff --git a/services/audioflinger/FastCaptureState.h b/services/audioflinger/FastCaptureState.h
new file mode 100644
index 0000000..29c865a
--- /dev/null
+++ b/services/audioflinger/FastCaptureState.h
@@ -0,0 +1,51 @@
+/*
+ * 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 ANDROID_AUDIO_FAST_CAPTURE_STATE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_STATE_H
+
+#include <media/nbaio/NBAIO.h>
+#include "FastThreadState.h"
+#include <private/media/AudioTrackShared.h>
+
+namespace android {
+
+// Represent a single state of the fast capture
+struct FastCaptureState : FastThreadState {
+ FastCaptureState();
+ /*virtual*/ ~FastCaptureState();
+
+ // all pointer fields use raw pointers; objects are owned and ref-counted by RecordThread
+ NBAIO_Source *mInputSource; // HAL input device, must already be negotiated
+ // FIXME by renaming, could pull up these fields to FastThreadState
+ int mInputSourceGen; // increment when mInputSource is assigned
+ NBAIO_Sink *mPipeSink; // after reading from input source, write to this pipe sink
+ int mPipeSinkGen; // increment when mPipeSink is assigned
+ size_t mFrameCount; // number of frames per fast capture buffer
+ audio_track_cblk_t *mCblk; // control block for the single fast client, or NULL
+
+ // Extends FastThreadState::Command
+ static const Command
+ // The following commands also process configuration changes, and can be "or"ed:
+ READ = 0x8, // read from input source
+ WRITE = 0x10, // write to pipe sink
+ READ_WRITE = 0x18; // read from input source and write to pipe sink
+
+}; // struct FastCaptureState
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_FAST_CAPTURE_STATE_H
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 1caed11..c9a3f10 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -37,6 +37,7 @@
#include <cpustats/ThreadCpuUsage.h>
#endif
#endif
+#include <audio_utils/format.h>
#include "AudioMixer.h"
#include "FastMixer.h"
@@ -53,8 +54,12 @@
outputSink(NULL),
outputSinkGen(0),
mixer(NULL),
- mixBuffer(NULL),
- mixBufferState(UNDEFINED),
+ mSinkBuffer(NULL),
+ mSinkBufferSize(0),
+ mMixerBuffer(NULL),
+ mMixerBufferSize(0),
+ mMixerBufferFormat(AUDIO_FORMAT_PCM_16_BIT),
+ mMixerBufferState(UNDEFINED),
format(Format_Invalid),
sampleRate(0),
fastTracksGen(0),
@@ -109,7 +114,8 @@
void FastMixer::onExit()
{
delete mixer;
- delete[] mixBuffer;
+ free(mMixerBuffer);
+ free(mSinkBuffer);
}
bool FastMixer::isSubClassCommand(FastThreadState::Command command)
@@ -155,14 +161,23 @@
// FIXME to avoid priority inversion, don't delete here
delete mixer;
mixer = NULL;
- delete[] mixBuffer;
- mixBuffer = NULL;
+ free(mMixerBuffer);
+ mMixerBuffer = NULL;
+ free(mSinkBuffer);
+ mSinkBuffer = NULL;
if (frameCount > 0 && sampleRate > 0) {
// FIXME new may block for unbounded time at internal mutex of the heap
// implementation; it would be better to have normal mixer allocate for us
// to avoid blocking here and to prevent possible priority inversion
mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks);
- mixBuffer = new short[frameCount * FCC_2];
+ const size_t mixerFrameSize = FCC_2 * audio_bytes_per_sample(mMixerBufferFormat);
+ mMixerBufferSize = mixerFrameSize * frameCount;
+ (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
+ const size_t sinkFrameSize = FCC_2 * audio_bytes_per_sample(format.mFormat);
+ if (sinkFrameSize > mixerFrameSize) { // need a sink buffer
+ mSinkBufferSize = sinkFrameSize * frameCount;
+ (void)posix_memalign(&mSinkBuffer, 32, mSinkBufferSize);
+ }
periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00
underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75
overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50
@@ -175,7 +190,7 @@
forceNs = 0;
warmupNs = 0;
}
- mixBufferState = UNDEFINED;
+ mMixerBufferState = UNDEFINED;
#if !LOG_NDEBUG
for (unsigned i = 0; i < FastMixerState::kMaxFastTracks; ++i) {
fastTrackNames[i] = -1;
@@ -193,7 +208,7 @@
const unsigned currentTrackMask = current->mTrackMask;
dumpState->mTrackMask = currentTrackMask;
if (current->mFastTracksGen != fastTracksGen) {
- ALOG_ASSERT(mixBuffer != NULL);
+ ALOG_ASSERT(mMixerBuffer != NULL);
int name;
// process removed tracks first to avoid running out of track names
@@ -224,13 +239,20 @@
AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
ALOG_ASSERT(bufferProvider != NULL && fastTrackNames[i] == -1);
if (mixer != NULL) {
- name = mixer->getTrackName(fastTrack->mChannelMask, AUDIO_SESSION_OUTPUT_MIX);
+ name = mixer->getTrackName(fastTrack->mChannelMask,
+ fastTrack->mFormat, AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
fastTrackNames[i] = name;
mixer->setBufferProvider(name, bufferProvider);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
- (void *) mixBuffer);
+ (void *) mMixerBuffer);
// newly allocated track names default to full scale volume
+ mixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ (void *)(uintptr_t)fastTrack->mFormat);
mixer->enable(name);
}
generations[i] = fastTrack->mGeneration;
@@ -252,13 +274,18 @@
ALOG_ASSERT(name >= 0);
mixer->setBufferProvider(name, bufferProvider);
if (fastTrack->mVolumeProvider == NULL) {
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *) MAX_GAIN_INT);
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *) MAX_GAIN_INT);
+ float f = AudioMixer::UNITY_GAIN_FLOAT;
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
}
mixer->setParameter(name, AudioMixer::RESAMPLE,
AudioMixer::REMOVE, NULL);
+ mixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ (void *)(uintptr_t)fastTrack->mFormat);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
(void *)(uintptr_t) fastTrack->mChannelMask);
// already enabled
@@ -281,7 +308,7 @@
const size_t frameCount = current->mFrameCount;
if ((command & FastMixerState::MIX) && (mixer != NULL) && isWarm) {
- ALOG_ASSERT(mixBuffer != NULL);
+ ALOG_ASSERT(mMixerBuffer != NULL);
// for each track, update volume and check for underrun
unsigned currentTrackMask = current->mTrackMask;
while (currentTrackMask != 0) {
@@ -309,12 +336,11 @@
ALOG_ASSERT(name >= 0);
if (fastTrack->mVolumeProvider != NULL) {
gain_minifloat_packed_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *) (uintptr_t)
- (float_from_gain(gain_minifloat_unpack_left(vlr)) * MAX_GAIN_INT));
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *) (uintptr_t)
- (float_from_gain(gain_minifloat_unpack_right(vlr)) * MAX_GAIN_INT));
+ float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+ float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
+
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &vlf);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &vrf);
}
// FIXME The current implementation of framesReady() for fast tracks
// takes a tryLock, which can block
@@ -358,26 +384,31 @@
// process() is CPU-bound
mixer->process(pts);
- mixBufferState = MIXED;
- } else if (mixBufferState == MIXED) {
- mixBufferState = UNDEFINED;
+ mMixerBufferState = MIXED;
+ } else if (mMixerBufferState == MIXED) {
+ mMixerBufferState = UNDEFINED;
}
//bool didFullWrite = false; // dumpsys could display a count of partial writes
- if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mixBuffer != NULL)) {
- if (mixBufferState == UNDEFINED) {
- memset(mixBuffer, 0, frameCount * FCC_2 * sizeof(short));
- mixBufferState = ZEROED;
+ if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mMixerBuffer != NULL)) {
+ if (mMixerBufferState == UNDEFINED) {
+ memset(mMixerBuffer, 0, mMixerBufferSize);
+ mMixerBufferState = ZEROED;
+ }
+ void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;
+ if (format.mFormat != mMixerBufferFormat) { // sink format not the same as mixer format
+ memcpy_by_audio_format(buffer, format.mFormat, mMixerBuffer, mMixerBufferFormat,
+ frameCount * Format_channelCount(format));
}
// if non-NULL, then duplicate write() to this non-blocking sink
NBAIO_Sink* teeSink;
if ((teeSink = current->mTeeSink) != NULL) {
- (void) teeSink->write(mixBuffer, frameCount);
+ (void) teeSink->write(mMixerBuffer, frameCount);
}
// FIXME write() is non-blocking and lock-free for a properly implemented NBAIO sink,
// but this code should be modified to handle both non-blocking and blocking sinks
dumpState->mWriteSequence++;
ATRACE_BEGIN("write");
- ssize_t framesWritten = outputSink->write(mixBuffer, frameCount);
+ ssize_t framesWritten = outputSink->write(buffer, frameCount);
ATRACE_END();
dumpState->mWriteSequence++;
if (framesWritten >= 0) {
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index db89ef4..4671670 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -61,8 +61,16 @@
NBAIO_Sink *outputSink;
int outputSinkGen;
AudioMixer* mixer;
- short *mixBuffer;
- enum {UNDEFINED, MIXED, ZEROED} mixBufferState;
+
+ // mSinkBuffer audio format is stored in format.mFormat.
+ void* mSinkBuffer; // used for mixer output format translation
+ // if sink format is different than mixer output.
+ size_t mSinkBufferSize;
+ void* mMixerBuffer; // mixer output buffer.
+ size_t mMixerBufferSize;
+ audio_format_t mMixerBufferFormat; // mixer output format: AUDIO_FORMAT_PCM_(16_BIT|FLOAT).
+
+ enum {UNDEFINED, MIXED, ZEROED} mMixerBufferState;
NBAIO_Format format;
unsigned sampleRate;
int fastTracksGen;
diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 8e6d0d4..3aa8dad 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -20,7 +20,7 @@
FastTrack::FastTrack() :
mBufferProvider(NULL), mVolumeProvider(NULL),
- mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0)
+ mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mFormat(AUDIO_FORMAT_INVALID), mGeneration(0)
{
}
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index e388fb3..661c9ca 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -45,6 +45,7 @@
ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active
VolumeProvider* mVolumeProvider; // optional; if NULL then full-scale
audio_channel_mask_t mChannelMask; // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO
+ audio_format_t mFormat; // track format
int mGeneration; // increment when any field is assigned
};
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 02b5094..6d84296 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -235,7 +235,7 @@
param.addInt(String8(AudioParameter::keyInputSource),
(int)patch->sinks[0].ext.mix.usecase.source);
- ALOGW("createAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
+ ALOGV("createAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
param.toString().string());
status = thread->setParameters(param.toString());
}
@@ -354,7 +354,7 @@
}
AudioParameter param;
param.addInt(String8(AudioParameter::keyRouting), 0);
- ALOGW("releaseAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
+ ALOGV("releaseAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
param.toString().string());
status = thread->setParameters(param.toString());
}
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
index 152455d..8246fef 100644
--- a/services/audioflinger/ServiceUtilities.cpp
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -59,6 +59,13 @@
return ok;
}
+bool modifyAudioRoutingAllowed() {
+ static const String16 sModifyAudioRoutingAllowed("android.permission.MODIFY_AUDIO_ROUTING");
+ bool ok = checkCallingPermission(sModifyAudioRoutingAllowed);
+ if (!ok) ALOGE("android.permission.MODIFY_AUDIO_ROUTING");
+ return ok;
+}
+
bool dumpAllowed() {
// don't optimize for same pid, since mediaserver never dumps itself
static const String16 sDump("android.permission.DUMP");
diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h
index 531bc56..df6f6f4 100644
--- a/services/audioflinger/ServiceUtilities.h
+++ b/services/audioflinger/ServiceUtilities.h
@@ -24,6 +24,7 @@
bool captureAudioOutputAllowed();
bool captureHotwordAllowed();
bool settingsAllowed();
+bool modifyAudioRoutingAllowed();
bool dumpAllowed();
}
diff --git a/services/audioflinger/StateQueueInstantiations.cpp b/services/audioflinger/StateQueueInstantiations.cpp
index 0d5cd0c..6f4505e 100644
--- a/services/audioflinger/StateQueueInstantiations.cpp
+++ b/services/audioflinger/StateQueueInstantiations.cpp
@@ -16,12 +16,14 @@
#include "Configuration.h"
#include "FastMixerState.h"
+#include "FastCaptureState.h"
#include "StateQueue.h"
// FIXME hack for gcc
namespace android {
-template class StateQueue<FastMixerState>; // typedef FastMixerStateQueue
+template class StateQueue<FastMixerState>; // typedef FastMixerStateQueue
+template class StateQueue<FastCaptureState>; // typedef FastCaptureStateQueue
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 4972c7a..d6333be 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -38,6 +38,7 @@
#include <audio_utils/minifloat.h>
// NBAIO implementations
+#include <media/nbaio/AudioStreamInSource.h>
#include <media/nbaio/AudioStreamOutSink.h>
#include <media/nbaio/MonoPipe.h>
#include <media/nbaio/MonoPipeReader.h>
@@ -53,6 +54,7 @@
#include "AudioFlinger.h"
#include "AudioMixer.h"
#include "FastMixer.h"
+#include "FastCapture.h"
#include "ServiceUtilities.h"
#include "SchedulingPolicyService.h"
@@ -131,9 +133,17 @@
// up large writes into smaller ones, and the wrapper would need to deal with scheduler.
} kUseFastMixer = FastMixer_Static;
+// Whether to use fast capture
+static const enum {
+ FastCapture_Never, // never initialize or use: for debugging only
+ FastCapture_Always, // always initialize and use, even if not needed: for debugging only
+ FastCapture_Static, // initialize if needed, then use all the time if initialized
+} kUseFastCapture = FastCapture_Static;
+
// Priorities for requestPriority
static const int kPriorityAudioApp = 2;
static const int kPriorityFastMixer = 3;
+static const int kPriorityFastCapture = 3;
// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
// for the track. The client then sub-divides this into smaller buffers for its use.
@@ -142,8 +152,17 @@
// FIXME It would be better for client to tell AudioFlinger the value of N,
// so AudioFlinger could allocate the right amount of memory.
// See the client's minBufCount and mNotificationFramesAct calculations for details.
+
+// This is the default value, if not specified by property.
static const int kFastTrackMultiplier = 2;
+// The minimum and maximum allowed values
+static const int kFastTrackMultiplierMin = 1;
+static const int kFastTrackMultiplierMax = 2;
+
+// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
+static int sFastTrackMultiplier = kFastTrackMultiplier;
+
// See Thread::readOnlyHeap().
// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
@@ -152,6 +171,22 @@
// ----------------------------------------------------------------------------
+static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
+
+static void sFastTrackMultiplierInit()
+{
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
+ char *endptr;
+ unsigned long ul = strtoul(value, &endptr, 0);
+ if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
+ sFastTrackMultiplier = (int) ul;
+ }
+ }
+}
+
+// ----------------------------------------------------------------------------
+
#ifdef ADD_BATTERY_DATA
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
@@ -1356,7 +1391,12 @@
) {
// if frameCount not specified, then it defaults to fast mixer (HAL) frame count
if (frameCount == 0) {
- frameCount = mFrameCount * kFastTrackMultiplier;
+ // read the fast track multiplier property the first time it is needed
+ int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
+ if (ok != 0) {
+ ALOGE("%s pthread_once failed: %d", __func__, ok);
+ }
+ frameCount = mFrameCount * sFastTrackMultiplier;
}
ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
frameCount, mFrameCount);
@@ -2715,9 +2755,27 @@
break;
}
if (initFastMixer) {
+ audio_format_t fastMixerFormat;
+ if (mMixerBufferEnabled && mEffectBufferEnabled) {
+ fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
+ } else {
+ fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+ }
+ if (mFormat != fastMixerFormat) {
+ // change our Sink format to accept our intermediate precision
+ mFormat = fastMixerFormat;
+ free(mSinkBuffer);
+ mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+ const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
+ (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
+ }
// create a MonoPipe to connect our submix to FastMixer
NBAIO_Format format = mOutputSink->format();
+ // adjust format to match that of the Fast Mixer
+ format.mFormat = fastMixerFormat;
+ format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
+
// This pipe depth compensates for scheduling latency of the normal mixer thread.
// When it wakes up after a maximum latency, it runs a few cycles quickly before
// finally blocking. Note the pipe implementation rounds up the request to a power of 2.
@@ -2758,6 +2816,8 @@
// wrap the source side of the MonoPipe to make it an AudioBufferProvider
fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
fastTrack->mVolumeProvider = NULL;
+ fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
+ fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
fastTrack->mGeneration++;
state->mFastTracksGen++;
state->mTrackMask = 1;
@@ -3210,6 +3270,7 @@
fastTrack->mBufferProvider = eabp;
fastTrack->mVolumeProvider = vp;
fastTrack->mChannelMask = track->mChannelMask;
+ fastTrack->mFormat = track->mFormat;
fastTrack->mGeneration++;
state->mTrackMask |= 1 << j;
didModify = true;
@@ -3319,9 +3380,11 @@
}
// compute volume for this track
- uint32_t vl, vr, va;
+ uint32_t vl, vr; // in U8.24 integer format
+ float vlf, vrf, vaf; // in [0.0, 1.0] float format
if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
- vl = vr = va = 0;
+ vl = vr = 0;
+ vlf = vrf = vaf = 0.;
if (track->isPausing()) {
track->setPaused();
}
@@ -3332,8 +3395,8 @@
float v = masterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
- float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
- float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
+ vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+ vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vlf > GAIN_FLOAT_UNITY) {
ALOGV("Track left volume out of range: %.3g", vlf);
@@ -3344,20 +3407,22 @@
vrf = GAIN_FLOAT_UNITY;
}
// now apply the master volume and stream type volume
- // FIXME we're losing the wonderful dynamic range in the minifloat representation
- float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT);
- vl = (uint32_t) (v8_24 * vlf);
- vr = (uint32_t) (v8_24 * vrf);
+ vlf *= v;
+ vrf *= v;
// assuming master volume and stream type volume each go up to 1.0,
- // vl and vr are now in 8.24 format
-
+ // then derive vl and vr as U8.24 versions for the effect chain
+ const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
+ vl = (uint32_t) (scaleto8_24 * vlf);
+ vr = (uint32_t) (scaleto8_24 * vrf);
+ // vl and vr are now in U8.24 format
uint16_t sendLevel = proxy->getSendLevel_U4_12();
// send level comes from shared memory and so may be corrupt
if (sendLevel > MAX_GAIN_INT) {
ALOGV("Track send level out of range: %04X", sendLevel);
sendLevel = MAX_GAIN_INT;
}
- va = (uint32_t)(v * sendLevel);
+ // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
+ vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
// Delegate volume control to effect in track effect chain if needed
@@ -3374,29 +3439,13 @@
track->mHasVolumeController = false;
}
- // FIXME Use float
- // Convert volumes from 8.24 to 4.12 format
- // This additional clamping is needed in case chain->setVolume_l() overshot
- vl = (vl + (1 << 11)) >> 12;
- if (vl > MAX_GAIN_INT) {
- vl = MAX_GAIN_INT;
- }
- vr = (vr + (1 << 11)) >> 12;
- if (vr > MAX_GAIN_INT) {
- vr = MAX_GAIN_INT;
- }
-
- if (va > MAX_GAIN_INT) {
- va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
- }
-
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(name, track);
mAudioMixer->enable(name);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr);
- mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
@@ -3601,9 +3650,10 @@
}
// getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
+int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId)
{
- return mAudioMixer->getTrackName(channelMask, sessionId);
+ return mAudioMixer->getTrackName(channelMask, format, sessionId);
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
@@ -3716,7 +3766,8 @@
delete mAudioMixer;
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
- int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
+ int name = getTrackName_l(mTracks[i]->mChannelMask,
+ mTracks[i]->mFormat, mTracks[i]->mSessionId);
if (name < 0) {
break;
}
@@ -4007,7 +4058,7 @@
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
- int sessionId __unused)
+ audio_format_t format __unused, int sessionId __unused)
{
return 0;
}
@@ -4708,16 +4759,151 @@
#endif
, mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
"RecordThreadRO", MemoryHeapBase::READ_ONLY))
+ // mFastCapture below
+ , mFastCaptureFutex(0)
+ // mInputSource
+ // mPipeSink
+ // mPipeSource
+ , mPipeFramesP2(0)
+ // mPipeMemory
+ // mFastCaptureNBLogWriter
+ , mFastTrackAvail(true)
{
snprintf(mName, kNameLength, "AudioIn_%X", id);
mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
readInputParameters_l();
+
+ // create an NBAIO source for the HAL input stream, and negotiate
+ mInputSource = new AudioStreamInSource(input->stream);
+ size_t numCounterOffers = 0;
+ const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
+ ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+
+ // initialize fast capture depending on configuration
+ bool initFastCapture;
+ switch (kUseFastCapture) {
+ case FastCapture_Never:
+ initFastCapture = false;
+ break;
+ case FastCapture_Always:
+ initFastCapture = true;
+ break;
+ case FastCapture_Static:
+ uint32_t primaryOutputSampleRate;
+ {
+ AutoMutex _l(audioFlinger->mHardwareLock);
+ primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate;
+ }
+ initFastCapture =
+ // either capture sample rate is same as (a reasonable) primary output sample rate
+ (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) &&
+ (mSampleRate == primaryOutputSampleRate)) ||
+ // or primary output sample rate is unknown, and capture sample rate is reasonable
+ ((primaryOutputSampleRate == 0) &&
+ ((mSampleRate == 44100 || mSampleRate == 48000)))) &&
+ // and the buffer size is < 10 ms
+ (mFrameCount * 1000) / mSampleRate < 10;
+ break;
+ // case FastCapture_Dynamic:
+ }
+
+ if (initFastCapture) {
+ // create a Pipe for FastMixer to write to, and for us and fast tracks to read from
+ NBAIO_Format format = mInputSource->format();
+ size_t pipeFramesP2 = roundup(mFrameCount * 8);
+ size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
+ void *pipeBuffer;
+ const sp<MemoryDealer> roHeap(readOnlyHeap());
+ sp<IMemory> pipeMemory;
+ if ((roHeap == 0) ||
+ (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
+ (pipeBuffer = pipeMemory->pointer()) == NULL) {
+ ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
+ goto failed;
+ }
+ // pipe will be shared directly with fast clients, so clear to avoid leaking old information
+ memset(pipeBuffer, 0, pipeSize);
+ Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
+ const NBAIO_Format offers[1] = {format};
+ size_t numCounterOffers = 0;
+ ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ mPipeSink = pipe;
+ PipeReader *pipeReader = new PipeReader(*pipe);
+ numCounterOffers = 0;
+ index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ mPipeSource = pipeReader;
+ mPipeFramesP2 = pipeFramesP2;
+ mPipeMemory = pipeMemory;
+
+ // create fast capture
+ mFastCapture = new FastCapture();
+ FastCaptureStateQueue *sq = mFastCapture->sq();
+#ifdef STATE_QUEUE_DUMP
+ // FIXME
+#endif
+ FastCaptureState *state = sq->begin();
+ state->mCblk = NULL;
+ state->mInputSource = mInputSource.get();
+ state->mInputSourceGen++;
+ state->mPipeSink = pipe;
+ state->mPipeSinkGen++;
+ state->mFrameCount = mFrameCount;
+ state->mCommand = FastCaptureState::COLD_IDLE;
+ // already done in constructor initialization list
+ //mFastCaptureFutex = 0;
+ state->mColdFutexAddr = &mFastCaptureFutex;
+ state->mColdGen++;
+ state->mDumpState = &mFastCaptureDumpState;
+#ifdef TEE_SINK
+ // FIXME
+#endif
+ mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
+ state->mNBLogWriter = mFastCaptureNBLogWriter.get();
+ sq->end();
+ sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+
+ // start the fast capture
+ mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
+ pid_t tid = mFastCapture->getTid();
+ int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+ if (err != 0) {
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+ kPriorityFastCapture, getpid_cached, tid, err);
+ }
+
+#ifdef AUDIO_WATCHDOG
+ // FIXME
+#endif
+
+ }
+failed: ;
+
+ // FIXME mNormalSource
}
AudioFlinger::RecordThread::~RecordThread()
{
+ if (mFastCapture != 0) {
+ FastCaptureStateQueue *sq = mFastCapture->sq();
+ FastCaptureState *state = sq->begin();
+ if (state->mCommand == FastCaptureState::COLD_IDLE) {
+ int32_t old = android_atomic_inc(&mFastCaptureFutex);
+ if (old == -1) {
+ (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
+ }
+ }
+ state->mCommand = FastCaptureState::EXIT;
+ sq->end();
+ sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+ mFastCapture->join();
+ mFastCapture.clear();
+ }
+ mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
mAudioFlinger->unregisterWriter(mNBLogWriter);
delete[] mRsmpInBuffer;
}
@@ -4772,6 +4958,8 @@
// activeTracks accumulates a copy of a subset of mActiveTracks
Vector< sp<RecordTrack> > activeTracks;
+ // reference to the (first and only) fast track
+ sp<RecordTrack> fastTrack;
{ // scope for mLock
Mutex::Autolock _l(mLock);
@@ -4853,6 +5041,11 @@
activeTracks.add(activeTrack);
i++;
+ if (activeTrack->isFastTrack()) {
+ ALOG_ASSERT(!mFastTrackAvail);
+ ALOG_ASSERT(fastTrack == 0);
+ fastTrack = activeTrack;
+ }
}
if (doBroadcast) {
mStartStopCond.broadcast();
@@ -4878,6 +5071,36 @@
effectChains[i]->process_l();
}
+ // Start the fast capture if it's not already running
+ if (mFastCapture != 0) {
+ FastCaptureStateQueue *sq = mFastCapture->sq();
+ FastCaptureState *state = sq->begin();
+ if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
+ (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
+ if (state->mCommand == FastCaptureState::COLD_IDLE) {
+ int32_t old = android_atomic_inc(&mFastCaptureFutex);
+ if (old == -1) {
+ (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
+ }
+ }
+ state->mCommand = FastCaptureState::READ_WRITE;
+#if 0 // FIXME
+ mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
+ FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+#endif
+ state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL;
+ sq->end();
+ sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+#if 0
+ if (kUseFastCapture == FastCapture_Dynamic) {
+ mNormalSource = mPipeSource;
+ }
+#endif
+ } else {
+ sq->end(false /*didModify*/);
+ }
+ }
+
// Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
// Only the client(s) that are too slow will overrun. But if even the fastest client is too
// slow, then this RecordThread will overrun by not calling HAL read often enough.
@@ -4885,24 +5108,45 @@
// copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
- ssize_t bytesRead = mInput->stream->read(mInput->stream,
- &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
- if (bytesRead <= 0) {
- ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize);
+ ssize_t framesRead;
+
+ // If an NBAIO source is present, use it to read the normal capture's data
+ if (mPipeSource != 0) {
+ size_t framesToRead = mBufferSize / mFrameSize;
+ framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount],
+ framesToRead, AudioBufferProvider::kInvalidPTS);
+ if (framesRead == 0) {
+ // since pipe is non-blocking, simulate blocking input
+ sleepUs = (framesToRead * 1000000LL) / mSampleRate;
+ }
+ // otherwise use the HAL / AudioStreamIn directly
+ } else {
+ ssize_t bytesRead = mInput->stream->read(mInput->stream,
+ &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
+ if (bytesRead < 0) {
+ framesRead = bytesRead;
+ } else {
+ framesRead = bytesRead / mFrameSize;
+ }
+ }
+
+ if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
+ ALOGE("read failed: framesRead=%d", framesRead);
// Force input into standby so that it tries to recover at next read attempt
inputStandBy();
sleepUs = kRecordThreadSleepUs;
+ }
+ if (framesRead <= 0) {
continue;
}
- ALOG_ASSERT((size_t) bytesRead <= mBufferSize);
- size_t framesRead = bytesRead / mFrameSize;
ALOG_ASSERT(framesRead > 0);
+
if (mTeeSink != 0) {
(void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead);
}
// If destination is non-contiguous, we now correct for reading past end of buffer.
size_t part1 = mRsmpInFramesP2 - rear;
- if (framesRead > part1) {
+ if ((size_t) framesRead > part1) {
memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
(framesRead - part1) * mFrameSize);
}
@@ -4913,6 +5157,11 @@
for (size_t i = 0; i < size; i++) {
activeTrack = activeTracks[i];
+ // skip fast tracks, as those are handled directly by FastCapture
+ if (activeTrack->isFastTrack()) {
+ continue;
+ }
+
enum {
OVERRUN_UNKNOWN,
OVERRUN_TRUE,
@@ -5141,6 +5390,30 @@
void AudioFlinger::RecordThread::inputStandBy()
{
+ // Idle the fast capture if it's currently running
+ if (mFastCapture != 0) {
+ FastCaptureStateQueue *sq = mFastCapture->sq();
+ FastCaptureState *state = sq->begin();
+ if (!(state->mCommand & FastCaptureState::IDLE)) {
+ state->mCommand = FastCaptureState::COLD_IDLE;
+ state->mColdFutexAddr = &mFastCaptureFutex;
+ state->mColdGen++;
+ mFastCaptureFutex = 0;
+ sq->end();
+ // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
+ sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
+#if 0
+ if (kUseFastCapture == FastCapture_Dynamic) {
+ // FIXME
+ }
+#endif
+#ifdef AUDIO_WATCHDOG
+ // FIXME
+#endif
+ } else {
+ sq->end(false /*didModify*/);
+ }
+ }
mInput->stream->common.standby(&mInput->stream->common);
}
@@ -5167,42 +5440,47 @@
// use case: callback handler and frame count is default or at least as large as HAL
(
(tid != -1) &&
- ((frameCount == 0) ||
+ ((frameCount == 0) /*||
+ // FIXME must be equal to pipe depth, so don't allow it to be specified by client
// FIXME not necessarily true, should be native frame count for native SR!
- (frameCount >= mFrameCount))
+ (frameCount >= mFrameCount)*/)
) &&
// PCM data
audio_is_linear_pcm(format) &&
+ // native format
+ (format == mFormat) &&
// mono or stereo
( (channelMask == AUDIO_CHANNEL_IN_MONO) ||
(channelMask == AUDIO_CHANNEL_IN_STEREO) ) &&
- // hardware sample rate
- // FIXME actually the native hardware sample rate
+ // native channel mask
+ (channelMask == mChannelMask) &&
+ // native hardware sample rate
(sampleRate == mSampleRate) &&
// record thread has an associated fast capture
- hasFastCapture()
- // fast capture does not require slots
+ hasFastCapture() &&
+ // there are sufficient fast track slots available
+ mFastTrackAvail
) {
- // if frameCount not specified, then it defaults to fast capture (HAL) frame count
+ // if frameCount not specified, then it defaults to pipe frame count
if (frameCount == 0) {
- // FIXME wrong mFrameCount
- frameCount = mFrameCount * kFastTrackMultiplier;
+ frameCount = mPipeFramesP2;
}
ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d "
"mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
- "hasFastCapture=%d tid=%d",
+ "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
frameCount, mFrameCount, format,
audio_is_linear_pcm(format),
- channelMask, sampleRate, mSampleRate, hasFastCapture(), tid);
+ channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail);
*flags &= ~IAudioFlinger::TRACK_FAST;
// FIXME It's not clear that we need to enforce this any more, since we have a pipe.
// For compatibility with AudioRecord calculation, buffer depth is forced
// to be at least 2 x the record thread frame count and cover audio hardware latency.
// This is probably too conservative, but legacy application code may depend on it.
// If you change this calculation, also review the start threshold which is related.
+ // FIXME It's not clear how input latency actually matters. Perhaps this should be 0.
uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
size_t mNormalFrameCount = 2048; // FIXME
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
@@ -5424,6 +5702,10 @@
{
mTracks.remove(track);
// need anything related to effects here?
+ if (track->isFastTrack()) {
+ ALOG_ASSERT(!mFastTrackAvail);
+ mFastTrackAvail = true;
+ }
}
void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
@@ -5442,6 +5724,7 @@
} else {
fdprintf(fd, " No active record clients\n");
}
+ dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
dumpBase(fd, args);
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index f8037c6..07887fb 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -356,6 +356,8 @@
// If a thread does not have such a heap, this method returns 0.
virtual sp<MemoryDealer> readOnlyHeap() const { return 0; }
+ virtual sp<IMemory> pipeMemory() const { return 0; }
+
mutable Mutex mLock;
protected:
@@ -674,7 +676,8 @@
// Allocate a track name for a given channel mask.
// Returns name >= 0 if successful, -1 on failure.
- virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
+ virtual int getTrackName_l(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId) = 0;
virtual void deleteTrackName_l(int name) = 0;
// Time to sleep between cycles when:
@@ -831,7 +834,8 @@
protected:
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
- virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+ virtual int getTrackName_l(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
@@ -884,7 +888,8 @@
status_t& status);
protected:
- virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+ virtual int getTrackName_l(audio_channel_mask_t channelMask,
+ audio_format_t format, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
@@ -1059,6 +1064,8 @@
virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; }
+ virtual sp<IMemory> pipeMemory() const { return mPipeMemory; }
+
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
@@ -1110,7 +1117,7 @@
static void syncStartEventCallback(const wp<SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
- bool hasFastCapture() const { return false; }
+ bool hasFastCapture() const { return mFastCapture != 0; }
private:
// Enter standby if not already in standby, and set mStandby flag
@@ -1140,4 +1147,40 @@
const sp<NBAIO_Sink> mTeeSink;
const sp<MemoryDealer> mReadOnlyHeap;
+
+ // one-time initialization, no locks required
+ sp<FastCapture> mFastCapture; // non-0 if there is also a fast capture
+ // FIXME audio watchdog thread
+
+ // contents are not guaranteed to be consistent, no locks required
+ FastCaptureDumpState mFastCaptureDumpState;
+#ifdef STATE_QUEUE_DUMP
+ // FIXME StateQueue observer and mutator dump fields
+#endif
+ // FIXME audio watchdog dump
+
+ // accessible only within the threadLoop(), no locks required
+ // mFastCapture->sq() // for mutating and pushing state
+ int32_t mFastCaptureFutex; // for cold idle
+
+ // The HAL input source is treated as non-blocking,
+ // but current implementation is blocking
+ sp<NBAIO_Source> mInputSource;
+ // The source for the normal capture thread to read from: mInputSource or mPipeSource
+ sp<NBAIO_Source> mNormalSource;
+ // If a fast capture is present, the non-blocking pipe sink written to by fast capture,
+ // otherwise clear
+ sp<NBAIO_Sink> mPipeSink;
+ // If a fast capture is present, the non-blocking pipe source read by normal thread,
+ // otherwise clear
+ sp<NBAIO_Source> mPipeSource;
+ // Depth of pipe from fast capture to normal thread and fast clients, always power of 2
+ size_t mPipeFramesP2;
+ // If a fast capture is present, the Pipe as IMemory, otherwise clear
+ sp<IMemory> mPipeMemory;
+
+ static const size_t kFastCaptureLogSize = 4 * 1024;
+ sp<NBLog::Writer> mFastCaptureNBLogWriter;
+
+ bool mFastTrackAvail; // true if fast track available
};
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 5f13be3..4cba3fd 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -39,6 +39,13 @@
STARTING_2, // for RecordTrack only
};
+ // where to allocate the data buffer
+ enum alloc_type {
+ ALLOC_CBLK, // allocate immediately after control block
+ ALLOC_READONLY, // allocate from a separate read-only heap per thread
+ ALLOC_PIPE, // do not allocate; use the pipe buffer
+ };
+
TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
@@ -50,7 +57,7 @@
int uid,
IAudioFlinger::track_flags_t flags,
bool isOut,
- bool useReadOnlyHeap = false);
+ alloc_type alloc = ALLOC_CBLK);
virtual ~TrackBase();
virtual status_t initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index cf71d65..4e8a058 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -73,7 +73,7 @@
int clientUid,
IAudioFlinger::track_flags_t flags,
bool isOut,
- bool useReadOnlyHeap)
+ alloc_type alloc)
: RefBase(),
mThread(thread),
mClient(client),
@@ -117,7 +117,7 @@
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
- if (sharedBuffer == 0 && !useReadOnlyHeap) {
+ if (sharedBuffer == 0 && alloc == ALLOC_CBLK) {
size += bufferSize;
}
@@ -139,7 +139,8 @@
// construct the shared structure in-place.
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
- if (useReadOnlyHeap) {
+ switch (alloc) {
+ case ALLOC_READONLY: {
const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
if (roHeap == 0 ||
(mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
@@ -153,7 +154,17 @@
return;
}
memset(mBuffer, 0, bufferSize);
- } else {
+ } break;
+ case ALLOC_PIPE:
+ mBufferMemory = thread->pipeMemory();
+ // mBuffer is the virtual address as seen from current process (mediaserver),
+ // and should normally be coming from mBufferMemory->pointer().
+ // However in this case the TrackBase does not reference the buffer directly.
+ // It should references the buffer via the pipe.
+ // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
+ mBuffer = NULL;
+ break;
+ case ALLOC_CBLK:
// clear all buffers
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
@@ -164,6 +175,7 @@
mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
#endif
}
+ break;
}
#ifdef TEE_SINK
@@ -388,7 +400,7 @@
}
mServerProxy = mAudioTrackServerProxy;
- mName = thread->getTrackName_l(channelMask, sessionId);
+ mName = thread->getTrackName_l(channelMask, format, sessionId);
if (mName < 0) {
ALOGE("no more track names available");
return;
@@ -1869,7 +1881,7 @@
: TrackBase(thread, client, sampleRate, format,
channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
flags, false /*isOut*/,
- (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
+ flags & IAudioFlinger::TRACK_FAST ? ALLOC_PIPE : ALLOC_CBLK),
mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
// See real initialization of mRsmpInFront at RecordThread::start()
mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
@@ -1888,9 +1900,14 @@
mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
// source SR
mResampler->setSampleRate(thread->mSampleRate);
- mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT);
mResamplerBufferProvider = new ResamplerBufferProvider(this);
}
+
+ if (flags & IAudioFlinger::TRACK_FAST) {
+ ALOG_ASSERT(thread->mFastTrackAvail);
+ thread->mFastTrackAvail = false;
+ }
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
index 2b33703..8cc386a 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
@@ -470,6 +470,9 @@
unsigned int *generation)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
@@ -480,6 +483,9 @@
status_t AudioPolicyService::getAudioPort(struct audio_port *port)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
@@ -491,6 +497,9 @@
audio_patch_handle_t *handle)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
@@ -501,6 +510,9 @@
status_t AudioPolicyService::releaseAudioPatch(audio_patch_handle_t handle)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
@@ -514,6 +526,9 @@
unsigned int *generation)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
@@ -524,6 +539,9 @@
status_t AudioPolicyService::setAudioPortConfig(const struct audio_port_config *config)
{
Mutex::Autolock _l(mLock);
+ if(!modifyAudioRoutingAllowed()) {
+ return PERMISSION_DENIED;
+ }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index 8c3af5b..61edac2 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -223,7 +223,7 @@
index = mAvailableOutputDevices.add(devDesc);
if (index >= 0) {
mAvailableOutputDevices[index]->mId = nextUniqueId();
- HwModule *module = getModuleForDevice(device);
+ sp<HwModule> module = getModuleForDevice(device);
ALOG_ASSERT(module != NULL, "setDeviceConnectionState():"
"could not find HW module for device %08x", device);
mAvailableOutputDevices[index]->mModule = module;
@@ -260,7 +260,7 @@
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
// close unused outputs after device disconnection or direct outputs that have been
// opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
@@ -311,7 +311,7 @@
ALOGW("setDeviceConnectionState() device already connected: %d", device);
return INVALID_OPERATION;
}
- HwModule *module = getModuleForDevice(device);
+ sp<HwModule> module = getModuleForDevice(device);
if (module == NULL) {
ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
device);
@@ -442,7 +442,7 @@
checkOutputForAllStrategies();
updateDevicesAndOutputs();
- AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
// force routing command to audio hardware when ending call
// even if no device change is needed
@@ -454,7 +454,7 @@
if (isStateInCall(state)) {
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
// mute media and sonification strategies and delay device switch by the largest
// latency of any output where either strategy is active.
// This avoid sending the ring tone or music tail into the earpiece or headset.
@@ -644,7 +644,7 @@
if (mTestOutputs[mCurOutput] == 0) {
ALOGV("getOutput() opening test output");
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
outputDesc->mDevice = mTestDevice;
outputDesc->mSamplingRate = mTestSamplingRate;
outputDesc->mFormat = mTestFormat;
@@ -696,10 +696,10 @@
}
if (profile != 0) {
- AudioOutputDescriptor *outputDesc = NULL;
+ sp<AudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
outputDesc = desc;
// reuse direct output if currently open and configured with same parameters
@@ -747,7 +747,6 @@
if (output != 0) {
mpClientInterface->closeOutput(output);
}
- delete outputDesc;
return 0;
}
audio_io_handle_t srcOutput = getOutputForEffect();
@@ -804,7 +803,7 @@
audio_io_handle_t outputPrimary = 0;
for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (!outputDesc->isDuplicated()) {
int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
if (commonFlags > maxCommonFlags) {
@@ -839,7 +838,7 @@
return BAD_VALUE;
}
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// increment usage count for this stream on the requested output:
// NOTE that the usage count is the same for duplicated output and hardware output which is
@@ -854,7 +853,7 @@
uint32_t waitMs = 0;
bool force = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
// force a device change if any other output is managed by the same hw
// module and has a current device selection that differs from selected device.
@@ -907,7 +906,7 @@
return BAD_VALUE;
}
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// handle special case for sonification while in call
if (isInCall()) {
@@ -932,7 +931,7 @@
// one being selected for this output
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t curOutput = mOutputs.keyAt(i);
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (curOutput != output &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
@@ -965,10 +964,9 @@
#ifdef AUDIO_POLICY_TEST
int testIndex = testOutputIndex(output);
if (testIndex != 0) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
if (outputDesc->isActive()) {
mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
mOutputs.removeItem(output);
mTestOutputs[testIndex] = 0;
}
@@ -976,7 +974,7 @@
}
#endif //AUDIO_POLICY_TEST
- AudioOutputDescriptor *desc = mOutputs.valueAt(index);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (desc->mDirectOpenCount <= 0) {
ALOGW("releaseOutput() invalid open count %d for output %d",
@@ -1045,7 +1043,7 @@
return 0;
}
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
inputDesc->mInputSource = inputSource;
inputDesc->mDevice = device;
@@ -1069,7 +1067,6 @@
if (input != 0) {
mpClientInterface->closeInput(input);
}
- delete inputDesc;
return 0;
}
addInput(input, inputDesc);
@@ -1085,7 +1082,7 @@
ALOGW("startInput() unknown input %d", input);
return BAD_VALUE;
}
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
#ifdef AUDIO_POLICY_TEST
if (mTestInput == 0)
@@ -1095,7 +1092,7 @@
// uses AUDIO_SOURCE_HOTWORD in which case it is closed.
audio_io_handle_t activeInput = getActiveInput();
if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
- AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
ALOGW("startInput() preempting already started low-priority input %d", activeInput);
stopInput(activeInput);
@@ -1129,7 +1126,7 @@
ALOGW("stopInput() unknown input %d", input);
return BAD_VALUE;
}
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
if (inputDesc->mRefCount == 0) {
ALOGW("stopInput() input %d already stopped", input);
@@ -1156,7 +1153,6 @@
return;
}
mpClientInterface->closeInput(input);
- delete mInputs.valueAt(index);
mInputs.removeItem(input);
nextAudioPortGeneration();
mpClientInterface->onAudioPortListUpdate();
@@ -1265,7 +1261,7 @@
audio_io_handle_t outputDeepBuffer = 0;
for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
outputOffloaded = outputs[i];
@@ -1327,14 +1323,14 @@
desc->name, io, strategy, session, id);
ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
- EffectDescriptor *pDesc = new EffectDescriptor();
- memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
- pDesc->mIo = io;
- pDesc->mStrategy = (routing_strategy)strategy;
- pDesc->mSession = session;
- pDesc->mEnabled = false;
+ sp<EffectDescriptor> effectDesc = new EffectDescriptor();
+ memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
+ effectDesc->mIo = io;
+ effectDesc->mStrategy = (routing_strategy)strategy;
+ effectDesc->mSession = session;
+ effectDesc->mEnabled = false;
- mEffects.add(id, pDesc);
+ mEffects.add(id, effectDesc);
return NO_ERROR;
}
@@ -1347,21 +1343,20 @@
return INVALID_OPERATION;
}
- EffectDescriptor *pDesc = mEffects.valueAt(index);
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(index);
- setEffectEnabled(pDesc, false);
+ setEffectEnabled(effectDesc, false);
- if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
+ if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
ALOGW("unregisterEffect() memory %d too big for total %d",
- pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
- pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+ effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
}
- mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
+ mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
- pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
mEffects.removeItem(id);
- delete pDesc;
return NO_ERROR;
}
@@ -1377,43 +1372,43 @@
return setEffectEnabled(mEffects.valueAt(index), enabled);
}
-status_t AudioPolicyManager::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
+status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled)
{
- if (enabled == pDesc->mEnabled) {
+ if (enabled == effectDesc->mEnabled) {
ALOGV("setEffectEnabled(%s) effect already %s",
enabled?"true":"false", enabled?"enabled":"disabled");
return INVALID_OPERATION;
}
if (enabled) {
- if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+ if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
- pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
+ effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
return INVALID_OPERATION;
}
- mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
+ mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
} else {
- if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
+ if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
- pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
- pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+ effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+ effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
}
- mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
+ mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
}
- pDesc->mEnabled = enabled;
+ effectDesc->mEnabled = enabled;
return NO_ERROR;
}
bool AudioPolicyManager::isNonOffloadableEffectEnabled()
{
for (size_t i = 0; i < mEffects.size(); i++) {
- const EffectDescriptor * const pDesc = mEffects.valueAt(i);
- if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
- ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
+ ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
- pDesc->mDesc.name, pDesc->mSession);
+ effectDesc->mDesc.name, effectDesc->mSession);
return true;
}
}
@@ -1424,7 +1419,7 @@
{
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
return true;
}
@@ -1437,7 +1432,7 @@
{
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
return true;
@@ -1449,7 +1444,7 @@
bool AudioPolicyManager::isSourceActive(audio_source_t source) const
{
for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if ((inputDescriptor->mInputSource == (int)source ||
(source == AUDIO_SOURCE_VOICE_RECOGNITION &&
inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
@@ -1674,10 +1669,10 @@
return NO_ERROR;
}
-AudioPolicyManager::AudioOutputDescriptor *AudioPolicyManager::getOutputFromId(
+sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
audio_port_handle_t id) const
{
- AudioOutputDescriptor *outputDesc = NULL;
+ sp<AudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
outputDesc = mOutputs.valueAt(i);
if (outputDesc->mId == id) {
@@ -1687,10 +1682,10 @@
return outputDesc;
}
-AudioPolicyManager::AudioInputDescriptor *AudioPolicyManager::getInputFromId(
+sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
audio_port_handle_t id) const
{
- AudioInputDescriptor *inputDesc = NULL;
+ sp<AudioInputDescriptor> inputDesc = NULL;
for (size_t i = 0; i < mInputs.size(); i++) {
inputDesc = mInputs.valueAt(i);
if (inputDesc->mId == id) {
@@ -1700,8 +1695,11 @@
return inputDesc;
}
-AudioPolicyManager::HwModule *AudioPolicyManager::getModuleForDevice(audio_devices_t device) const
+sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
+ audio_devices_t device) const
{
+ sp <HwModule> module;
+
for (size_t i = 0; i < mHwModules.size(); i++) {
if (mHwModules[i]->mHandle == 0) {
continue;
@@ -1722,18 +1720,20 @@
}
}
}
- return NULL;
+ return module;
}
-AudioPolicyManager::HwModule *AudioPolicyManager::getModuleFromName(const char *name) const
+sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
{
+ sp <HwModule> module;
+
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (strcmp(mHwModules[i]->mName, name) == 0) {
return mHwModules[i];
}
}
- return NULL;
+ return module;
}
@@ -1783,7 +1783,7 @@
return BAD_VALUE;
}
// output mix to output device connection
- AudioOutputDescriptor *outputDesc = getOutputFromId(patch->sources[0].id);
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
@@ -1832,7 +1832,7 @@
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
// input device to input mix connection
- AudioInputDescriptor *inputDesc = getInputFromId(patch->sinks[0].id);
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
return BAD_VALUE;
}
@@ -1957,7 +1957,7 @@
struct audio_patch *patch = &patchDesc->mPatch;
patchDesc->mUid = mUidCached;
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
- AudioOutputDescriptor *outputDesc = getOutputFromId(patch->sources[0].id);
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
@@ -1970,7 +1970,7 @@
NULL);
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- AudioInputDescriptor *inputDesc = getInputFromId(patch->sinks[0].id);
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
return BAD_VALUE;
@@ -2035,27 +2035,24 @@
}
ALOGV("setAudioPortConfig() on port handle %d", config->id);
// Only support gain configuration for now
- if (config->config_mask != AUDIO_PORT_CONFIG_GAIN || config->gain.index < 0) {
- return BAD_VALUE;
+ if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
+ return INVALID_OPERATION;
}
- sp<AudioPort> portDesc;
- struct audio_port_config portConfig;
+ sp<AudioPortConfig> audioPortConfig;
if (config->type == AUDIO_PORT_TYPE_MIX) {
if (config->role == AUDIO_PORT_ROLE_SOURCE) {
- AudioOutputDescriptor *outputDesc = getOutputFromId(config->id);
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(config->id);
if (outputDesc == NULL) {
return BAD_VALUE;
}
- portDesc = outputDesc->mProfile;
- outputDesc->toAudioPortConfig(&portConfig);
+ audioPortConfig = outputDesc;
} else if (config->role == AUDIO_PORT_ROLE_SINK) {
- AudioInputDescriptor *inputDesc = getInputFromId(config->id);
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id);
if (inputDesc == NULL) {
return BAD_VALUE;
}
- portDesc = inputDesc->mProfile;
- inputDesc->toAudioPortConfig(&portConfig);
+ audioPortConfig = inputDesc;
} else {
return BAD_VALUE;
}
@@ -2071,46 +2068,21 @@
if (deviceDesc == NULL) {
return BAD_VALUE;
}
- portDesc = deviceDesc;
- deviceDesc->toAudioPortConfig(&portConfig);
+ audioPortConfig = deviceDesc;
} else {
return BAD_VALUE;
}
- if ((size_t)config->gain.index >= portDesc->mGains.size()) {
- return INVALID_OPERATION;
+ struct audio_port_config backupConfig;
+ status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
+ if (status == NO_ERROR) {
+ struct audio_port_config newConfig;
+ audioPortConfig->toAudioPortConfig(&newConfig, config);
+ status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
}
- const struct audio_gain *gain = &portDesc->mGains[config->gain.index]->mGain;
- if ((config->gain.mode & ~gain->mode) != 0) {
- return BAD_VALUE;
+ if (status != NO_ERROR) {
+ audioPortConfig->applyAudioPortConfig(&backupConfig);
}
- if ((config->gain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->gain.values[0] < gain->min_value) ||
- (config->gain.values[0] > gain->max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->gain.channel_mask & ~gain->channel_mask) != 0) {
- return BAD_VALUE;
- }
- size_t numValues = popcount(config->gain.channel_mask);
- for (size_t i = 0; i < numValues; i++) {
- if ((config->gain.values[i] < gain->min_value) ||
- (config->gain.values[i] > gain->max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->gain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->gain.ramp_duration_ms < gain->min_ramp_ms) ||
- (config->gain.ramp_duration_ms > gain->max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
-
- portConfig.gain = config->gain;
-
- status_t status = mpClientInterface->setAudioPortConfig(&portConfig, 0);
return status;
}
@@ -2230,7 +2202,7 @@
audio_devices_t profileTypes = outProfile->mSupportedDevices.types();
if ((profileTypes & outputDeviceTypes) &&
((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice->mDeviceType & profileTypes);
audio_io_handle_t output = mpClientInterface->openOutput(
@@ -2245,7 +2217,6 @@
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
mHwModules[i]->mName);
- delete outputDesc;
} else {
for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
@@ -2282,7 +2253,7 @@
audio_devices_t profileTypes = inProfile->mSupportedDevices.types();
if (profileTypes & inputDeviceTypes) {
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(inProfile);
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
inputDesc->mInputSource = AUDIO_SOURCE_MIC;
inputDesc->mDevice = inProfile->mSupportedDevices[0]->mDeviceType;
@@ -2310,7 +2281,6 @@
inputDesc->mDevice,
mHwModules[i]->mName);
}
- delete inputDesc;
}
}
}
@@ -2372,17 +2342,15 @@
#endif //AUDIO_POLICY_TEST
for (size_t i = 0; i < mOutputs.size(); i++) {
mpClientInterface->closeOutput(mOutputs.keyAt(i));
- delete mOutputs.valueAt(i);
}
for (size_t i = 0; i < mInputs.size(); i++) {
mpClientInterface->closeInput(mInputs.keyAt(i));
- delete mInputs.valueAt(i);
- }
- for (size_t i = 0; i < mHwModules.size(); i++) {
- delete mHwModules[i];
}
mAvailableOutputDevices.clear();
mAvailableInputDevices.clear();
+ mOutputs.clear();
+ mInputs.clear();
+ mHwModules.clear();
}
status_t AudioPolicyManager::initCheck()
@@ -2486,15 +2454,14 @@
if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_reopen"));
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
mpClientInterface->closeOutput(mPrimaryOutput);
audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
- delete mOutputs.valueFor(mPrimaryOutput);
mOutputs.removeItem(mPrimaryOutput);
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
&outputDesc->mDevice,
@@ -2542,7 +2509,7 @@
// ---
-void AudioPolicyManager::addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc)
+void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
{
outputDesc->mIoHandle = output;
outputDesc->mId = nextUniqueId();
@@ -2550,7 +2517,7 @@
nextAudioPortGeneration();
}
-void AudioPolicyManager::addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc)
+void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
{
inputDesc->mIoHandle = input;
inputDesc->mId = nextUniqueId();
@@ -2571,7 +2538,7 @@
SortedVector<audio_io_handle_t>& outputs,
const String8 address)
{
- AudioOutputDescriptor *desc;
+ sp<AudioOutputDescriptor> desc;
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open outputs that can be routed to this device
@@ -2714,7 +2681,7 @@
mPrimaryOutput);
if (duplicatedOutput != 0) {
// add duplicated output descriptor
- AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
+ sp<AudioOutputDescriptor> dupOutputDesc = new AudioOutputDescriptor(NULL);
dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
dupOutputDesc->mSamplingRate = desc->mSamplingRate;
@@ -2736,7 +2703,6 @@
}
if (output == 0) {
ALOGW("checkOutputsForDevice() could not open output for device %x", device);
- delete desc;
profiles.removeAt(profile_index);
profile_index--;
} else {
@@ -2796,7 +2762,7 @@
SortedVector<audio_io_handle_t>& inputs,
const String8 address)
{
- AudioInputDescriptor *desc;
+ sp<AudioInputDescriptor> desc;
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open inputs that can be routed to this device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
@@ -2911,7 +2877,6 @@
if (input == 0) {
ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
- delete desc;
profiles.removeAt(profile_index);
profile_index--;
} else {
@@ -2972,7 +2937,7 @@
{
ALOGV("closeOutput(%d)", output);
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc == NULL) {
ALOGW("closeOutput() unknown output %d", output);
return;
@@ -2980,11 +2945,11 @@
// look for duplicated outputs connected to the output being removed.
for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
if (dupOutputDesc->isDuplicated() &&
(dupOutputDesc->mOutput1 == outputDesc ||
dupOutputDesc->mOutput2 == outputDesc)) {
- AudioOutputDescriptor *outputDesc2;
+ sp<AudioOutputDescriptor> outputDesc2;
if (dupOutputDesc->mOutput1 == outputDesc) {
outputDesc2 = dupOutputDesc->mOutput2;
} else {
@@ -3002,7 +2967,6 @@
ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
mpClientInterface->closeOutput(duplicatedOutput);
- delete mOutputs.valueFor(duplicatedOutput);
mOutputs.removeItem(duplicatedOutput);
}
}
@@ -3012,14 +2976,13 @@
mpClientInterface->setParameters(output, param.toString());
mpClientInterface->closeOutput(output);
- delete outputDesc;
mOutputs.removeItem(output);
mPreviousOutputs = mOutputs;
nextAudioPortGeneration();
}
SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs)
{
SortedVector<audio_io_handle_t> outputs;
@@ -3061,7 +3024,7 @@
strategy, srcOutputs[0], dstOutputs[0]);
// mute strategy while moving tracks from one output to another
for (size_t i = 0; i < srcOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
if (desc->isStrategyActive(strategy)) {
setStrategyMute(strategy, true, srcOutputs[i]);
setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
@@ -3073,17 +3036,17 @@
audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
SortedVector<audio_io_handle_t> moved;
for (size_t i = 0; i < mEffects.size(); i++) {
- EffectDescriptor *desc = mEffects.valueAt(i);
- if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
- desc->mIo != fxOutput) {
- if (moved.indexOf(desc->mIo) < 0) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+ effectDesc->mIo != fxOutput) {
+ if (moved.indexOf(effectDesc->mIo) < 0) {
ALOGV("checkOutputForStrategy() moving effect %d to output %d",
mEffects.keyAt(i), fxOutput);
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
fxOutput);
- moved.add(desc->mIo);
+ moved.add(effectDesc->mIo);
}
- desc->mIo = fxOutput;
+ effectDesc->mIo = fxOutput;
}
}
}
@@ -3109,7 +3072,7 @@
audio_io_handle_t AudioPolicyManager::getA2dpOutput()
{
for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
return mOutputs.keyAt(i);
}
@@ -3167,7 +3130,7 @@
{
audio_devices_t device = AUDIO_DEVICE_NONE;
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
if (index >= 0) {
@@ -3213,7 +3176,7 @@
audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
{
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
if (index >= 0) {
@@ -3247,7 +3210,7 @@
devices = getDeviceForStrategy(strategy, true /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (outputDesc->isStrategyActive(strategy)) {
devices = outputDesc->device();
break;
@@ -3514,7 +3477,7 @@
mPreviousOutputs = mOutputs;
}
-uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
audio_devices_t prevDevice,
uint32_t delayMs)
{
@@ -3543,7 +3506,7 @@
}
if (doMute) {
for (size_t j = 0; j < mOutputs.size(); j++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(j);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
// skip output if it does not share any device with current output
if ((desc->supportedDevices() & outputDesc->supportedDevices())
== AUDIO_DEVICE_NONE) {
@@ -3601,7 +3564,7 @@
audio_patch_handle_t *patchHandle)
{
ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
AudioParameter param;
uint32_t muteWaitMs;
@@ -3703,7 +3666,7 @@
int delayMs,
audio_patch_handle_t *patchHandle)
{
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
@@ -3730,7 +3693,7 @@
{
status_t status = NO_ERROR;
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
inputDesc->mDevice = device;
@@ -3785,7 +3748,7 @@
status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
audio_patch_handle_t *patchHandle)
{
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
@@ -3899,7 +3862,7 @@
audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
{
for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
+ const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i);
if ((input_descriptor->mRefCount > 0)
&& (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
return mInputs.keyAt(i);
@@ -4144,7 +4107,7 @@
audio_devices_t device)
{
float volume = 1.0;
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
StreamDescriptor &streamDesc = mStreams[stream];
if (device == AUDIO_DEVICE_NONE) {
@@ -4300,7 +4263,7 @@
audio_devices_t device)
{
StreamDescriptor &streamDesc = mStreams[stream];
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->device();
}
@@ -4345,7 +4308,7 @@
const routing_strategy stream_strategy = getStrategy(stream);
if ((stream_strategy == STRATEGY_SONIFICATION) ||
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
stream, starting, outputDesc->mDevice, stateChange);
if (outputDesc->mRefCount[stream]) {
@@ -4403,8 +4366,7 @@
AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
- mChannelMask(0), mLatency(0),
+ : mId(0), mIoHandle(0), mLatency(0),
mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0),
mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
{
@@ -4419,9 +4381,13 @@
mStrategyMutedByDevice[i] = false;
}
if (profile != NULL) {
+ mAudioPort = profile;
mSamplingRate = profile->mSamplingRates[0];
mFormat = profile->mFormats[0];
mChannelMask = profile->mChannelMasks[0];
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
mFlags = profile->mFlags;
}
}
@@ -4445,7 +4411,7 @@
}
bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
- const AudioOutputDescriptor *outputDesc)
+ const sp<AudioOutputDescriptor> outputDesc)
{
if (isDuplicated()) {
return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
@@ -4528,31 +4494,16 @@
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask &= srcConfig->config_mask;
+ }
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->sample_rate = mSamplingRate;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->format = mFormat;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = srcConfig->format;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
dstConfig->ext.mix.handle = mIoHandle;
dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
@@ -4603,15 +4554,22 @@
// --- AudioInputDescriptor class implementation
AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mSamplingRate(0),
- mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
+ : mId(0), mIoHandle(0),
mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0), mRefCount(0),
mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile)
{
if (profile != NULL) {
+ mAudioPort = profile;
mSamplingRate = profile->mSamplingRates[0];
mFormat = profile->mFormats[0];
mChannelMask = profile->mChannelMasks[0];
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ } else {
+ mSamplingRate = 0;
+ mFormat = AUDIO_FORMAT_DEFAULT;
+ mChannelMask = 0;
}
}
@@ -4619,31 +4577,17 @@
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask &= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SINK;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->sample_rate = mSamplingRate;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->format = mFormat;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = srcConfig->format;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
dstConfig->ext.mix.handle = mIoHandle;
dstConfig->ext.mix.usecase.source = mInputSource;
@@ -4745,7 +4689,8 @@
// --- HwModule class implementation
AudioPolicyManager::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+ mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
{
}
@@ -4904,6 +4849,8 @@
result.append(buffer);
snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
result.append(buffer);
+ snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+ result.append(buffer);
write(fd, result.string(), result.size());
if (mOutputProfiles.size()) {
write(fd, " - outputs:\n", strlen(" - outputs:\n"));
@@ -4931,6 +4878,15 @@
// --- AudioPort class implementation
+
+AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module) :
+ mName(name), mType(type), mRole(role), mModule(module)
+{
+ mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
{
port->role = mRole;
@@ -5066,18 +5022,17 @@
return mode;
}
-void AudioPolicyManager::AudioPort::loadGain(cnode *root)
+void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
{
cnode *node = root->first_child;
- sp<AudioGain> gain = new AudioGain();
+ sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
while (node) {
if (strcmp(node->name, GAIN_MODE) == 0) {
gain->mGain.mode = loadGainMode((char *)node->value);
} else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if ((mType == AUDIO_PORT_TYPE_DEVICE && mRole == AUDIO_PORT_ROLE_SOURCE) ||
- (mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK)) {
+ if (mUseInChannelMask) {
gain->mGain.channel_mask =
(audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
ARRAY_SIZE(sInChannelsNameToEnumTable),
@@ -5116,13 +5071,53 @@
void AudioPolicyManager::AudioPort::loadGains(cnode *root)
{
cnode *node = root->first_child;
+ int index = 0;
while (node) {
ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node);
+ loadGain(node, index++);
node = node->next;
}
}
+status_t AudioPolicyManager::AudioPort::checkSamplingRate(uint32_t samplingRate) const
+{
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if (mSamplingRates[i] == samplingRate) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPolicyManager::AudioPort::checkChannelMask(audio_channel_mask_t channelMask) const
+{
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ if (mChannelMasks[i] == channelMask) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
+{
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if (mFormats[i] == format) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+ int index) const
+{
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+}
+
void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
{
const size_t SIZE = 256;
@@ -5181,11 +5176,72 @@
// --- AudioGain class implementation
-AudioPolicyManager::AudioGain::AudioGain()
+AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
memset(&mGain, 0, sizeof(struct audio_gain));
}
+void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
{
const size_t SIZE = 256;
@@ -5214,10 +5270,116 @@
write(fd, result.string(), result.size());
}
+// --- AudioPortConfig class implementation
+
+AudioPolicyManager::AudioPortConfig::AudioPortConfig()
+{
+ mSamplingRate = 0;
+ mChannelMask = AUDIO_CHANNEL_NONE;
+ mFormat = AUDIO_FORMAT_INVALID;
+ mGain.index = -1;
+}
+
+status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig;
+ status_t status = NO_ERROR;
+
+ localBackupConfig.config_mask = config->config_mask;
+ toAudioPortConfig(&localBackupConfig);
+
+ if (mAudioPort == 0) {
+ status = NO_INIT;
+ goto exit;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ status = mAudioPort->checkSamplingRate(config->sample_rate);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ status = mAudioPort->checkChannelMask(config->channel_mask);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ status = mAudioPort->checkFormat(config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ status = mAudioPort->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mGain = config->gain;
+ }
+
+exit:
+ if (status != NO_ERROR) {
+ applyAudioPortConfig(&localBackupConfig);
+ }
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ dstConfig->sample_rate = mSamplingRate;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+ dstConfig->sample_rate = srcConfig->sample_rate;
+ }
+ } else {
+ dstConfig->sample_rate = 0;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ dstConfig->channel_mask = mChannelMask;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+ dstConfig->channel_mask = srcConfig->channel_mask;
+ }
+ } else {
+ dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ dstConfig->format = mFormat;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+ dstConfig->format = srcConfig->format;
+ }
+ } else {
+ dstConfig->format = AUDIO_FORMAT_INVALID;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
// --- IOProfile class implementation
AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- HwModule *module)
+ const sp<HwModule>& module)
: AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module), mFlags((audio_output_flags_t)0)
{
}
@@ -5245,32 +5407,13 @@
if ((mFlags & flags) != flags) {
return false;
}
- size_t i;
- for (i = 0; i < mSamplingRates.size(); i++)
- {
- if (mSamplingRates[i] == samplingRate) {
- break;
- }
- }
- if (i == mSamplingRates.size()) {
+ if (checkSamplingRate(samplingRate) != NO_ERROR) {
return false;
}
- for (i = 0; i < mFormats.size(); i++)
- {
- if (mFormats[i] == format) {
- break;
- }
- }
- if (i == mFormats.size()) {
+ if (checkChannelMask(channelMask) != NO_ERROR) {
return false;
}
- for (i = 0; i < mChannelMasks.size(); i++)
- {
- if (mChannelMasks[i] == channelMask) {
- break;
- }
- }
- if (i == mChannelMasks.size()) {
+ if (checkFormat(format) != NO_ERROR) {
return false;
}
return true;
@@ -5322,6 +5465,21 @@
// --- DeviceDescriptor implementation
+
+AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mDeviceType(type), mAddress(""),
+ mChannelMask(AUDIO_CHANNEL_NONE), mId(0)
+{
+ mAudioPort = this;
+ if (mGains.size() > 0) {
+ mGains[0]->getDefaultConfig(&mGain);
+ }
+}
+
bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
{
// Devices are considered equal if they:
@@ -5486,23 +5644,17 @@
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask &= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
dstConfig->id = mId;
dstConfig->role = audio_is_output_device(mDeviceType) ?
AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
dstConfig->ext.device.type = mDeviceType;
dstConfig->ext.device.hw_module = mModule->mHandle;
strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
@@ -5598,7 +5750,7 @@
{
status_t status = NAME_NOT_FOUND;
cnode *node;
- HwModule *module = new HwModule(root->name);
+ sp<HwModule> module = new HwModule(root->name);
node = config_find(root, DEVICES_TAG);
if (node != NULL) {
@@ -5640,8 +5792,6 @@
if (status == NO_ERROR) {
mHwModules.add(module);
- } else {
- delete module;
}
}
@@ -5660,9 +5810,10 @@
}
}
-void AudioPolicyManager::loadGlobalConfig(cnode *root, HwModule *module)
+void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
{
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+
if (node == NULL) {
return;
}
@@ -5695,6 +5846,12 @@
} else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
mSpeakerDrcEnabled = stringToBool((char *)node->value);
ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+ } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
+ uint32_t major, minor;
+ sscanf((char *)node->value, "%u.%u", &major, &minor);
+ module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
+ ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
+ module->mHalVersion, major, minor);
}
node = node->next;
}
@@ -5726,9 +5883,10 @@
void AudioPolicyManager::defaultAudioPolicyConfig(void)
{
- HwModule *module;
+ sp<HwModule> module;
sp<IOProfile> profile;
- sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_IN_BUILTIN_MIC);
+ sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""),
+ AUDIO_DEVICE_IN_BUILTIN_MIC);
mAvailableOutputDevices.add(mDefaultOutputDevice);
mAvailableInputDevices.add(defaultInputDevice);
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index e012d63..adf1b33 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -195,20 +195,23 @@
class AudioGain: public RefBase
{
public:
- AudioGain();
+ AudioGain(int index, bool useInChannelMask);
virtual ~AudioGain() {}
void dump(int fd, int spaces, int index) const;
+ void getDefaultConfig(struct audio_gain_config *config);
+ status_t checkConfig(const struct audio_gain_config *config);
+ int mIndex;
struct audio_gain mGain;
+ bool mUseInChannelMask;
};
- class AudioPort: public RefBase
+ class AudioPort: public virtual RefBase
{
public:
AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, HwModule *module) :
- mName(name), mType(type), mRole(role), mModule(module) {}
+ audio_port_role_t role, const sp<HwModule>& module);
virtual ~AudioPort() {}
virtual void toAudioPort(struct audio_port *port) const;
@@ -219,14 +222,20 @@
void loadInChannels(char *name);
audio_gain_mode_t loadGainMode(char *name);
- void loadGain(cnode *root);
+ void loadGain(cnode *root, int index);
void loadGains(cnode *root);
+ status_t checkSamplingRate(uint32_t samplingRate) const;
+ status_t checkChannelMask(audio_channel_mask_t channelMask) const;
+ status_t checkFormat(audio_format_t format) const;
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
void dump(int fd, int spaces) const;
String8 mName;
audio_port_type_t mType;
audio_port_role_t mRole;
+ bool mUseInChannelMask;
// by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
// indicates the supported parameters should be read from the output stream
// after it is opened for the first time
@@ -234,9 +243,27 @@
Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
Vector <audio_format_t> mFormats; // supported audio formats
Vector < sp<AudioGain> > mGains; // gain controllers
- HwModule *mModule; // audio HW module exposing this I/O stream
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
};
+ class AudioPortConfig: public virtual RefBase
+ {
+ public:
+ AudioPortConfig();
+ virtual ~AudioPortConfig() {}
+
+ status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const = 0;
+ sp<AudioPort> mAudioPort;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ struct audio_gain_config mGain;
+ };
+
+
class AudioPatch: public RefBase
{
public:
@@ -250,29 +277,15 @@
audio_patch_handle_t mAfPatchHandle;
};
- class DeviceDescriptor: public AudioPort
+ class DeviceDescriptor: public AudioPort, public AudioPortConfig
{
public:
- DeviceDescriptor(const String8& name, audio_devices_t type, String8 address,
- audio_channel_mask_t channelMask) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(address),
- mChannelMask(channelMask), mId(0) {}
+ DeviceDescriptor(const String8& name, audio_devices_t type);
- DeviceDescriptor(String8 name, audio_devices_t type) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(""),
- mChannelMask(0), mId(0) {}
virtual ~DeviceDescriptor() {}
bool equals(const sp<DeviceDescriptor>& other) const;
- void toAudioPortConfig(struct audio_port_config *dstConfig,
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
virtual void toAudioPort(struct audio_port *port) const;
@@ -317,7 +330,7 @@
class IOProfile : public AudioPort
{
public:
- IOProfile(const String8& name, audio_port_role_t role, HwModule *module);
+ IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
virtual ~IOProfile();
bool isCompatibleProfile(audio_devices_t device,
@@ -335,7 +348,7 @@
// direct output...). For outputs only.
};
- class HwModule {
+ class HwModule : public RefBase{
public:
HwModule(const char *name);
~HwModule();
@@ -346,8 +359,9 @@
void dump(int fd);
- const char *const mName; // base name of the audio HW module (primary, a2dp ...)
- audio_module_handle_t mHandle;
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ uint32_t mHalVersion; // audio HAL API version
+ audio_module_handle_t mHandle;
Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
@@ -373,7 +387,7 @@
// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
// and keep track of the usage of this output by each audio stream type.
- class AudioOutputDescriptor
+ class AudioOutputDescriptor: public AudioPortConfig
{
public:
AudioOutputDescriptor(const sp<IOProfile>& profile);
@@ -386,7 +400,7 @@
bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
audio_devices_t supportedDevices();
uint32_t latency();
- bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+ bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
bool isActive(uint32_t inPastMs = 0) const;
bool isStreamActive(audio_stream_type_t stream,
uint32_t inPastMs = 0,
@@ -395,23 +409,20 @@
uint32_t inPastMs = 0,
nsecs_t sysTime = 0) const;
- void toAudioPortConfig(struct audio_port_config *dstConfig,
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
void toAudioPort(struct audio_port *port) const;
audio_port_handle_t mId;
audio_io_handle_t mIoHandle; // output handle
- uint32_t mSamplingRate; //
- audio_format_t mFormat; //
- audio_channel_mask_t mChannelMask; // output configuration
uint32_t mLatency; //
audio_output_flags_t mFlags; //
audio_devices_t mDevice; // current device this output is routed to
audio_patch_handle_t mPatchHandle;
uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
nsecs_t mStopTime[AUDIO_STREAM_CNT];
- AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output
- AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output
+ sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
+ sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
const sp<IOProfile> mProfile; // I/O profile this output derives from
@@ -422,7 +433,7 @@
// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
// and keep track of the usage of this input.
- class AudioInputDescriptor
+ class AudioInputDescriptor: public AudioPortConfig
{
public:
AudioInputDescriptor(const sp<IOProfile>& profile);
@@ -431,16 +442,13 @@
audio_port_handle_t mId;
audio_io_handle_t mIoHandle; // input handle
- uint32_t mSamplingRate; //
- audio_format_t mFormat; // input configuration
- audio_channel_mask_t mChannelMask; //
audio_devices_t mDevice; // current device this input is routed to
audio_patch_handle_t mPatchHandle;
uint32_t mRefCount; // number of AudioRecord clients using this output
audio_source_t mInputSource; // input source selected by application (mediarecorder.h)
const sp<IOProfile> mProfile; // I/O profile this output derives from
- void toAudioPortConfig(struct audio_port_config *dstConfig,
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
void toAudioPort(struct audio_port *port) const;
};
@@ -463,7 +471,7 @@
};
// stream descriptor used for volume control
- class EffectDescriptor
+ class EffectDescriptor : public RefBase
{
public:
@@ -476,8 +484,8 @@
bool mEnabled; // enabled state: CPU load being used or not
};
- void addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc);
- void addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc);
+ void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+ void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
// return the strategy corresponding to a given stream type
static routing_strategy getStrategy(audio_stream_type_t stream);
@@ -618,7 +626,7 @@
int testOutputIndex(audio_io_handle_t output);
#endif //AUDIO_POLICY_TEST
- status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+ status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
// returns the category the device belongs to with regard to volume curve management
static device_category getDeviceCategory(audio_devices_t device);
@@ -627,7 +635,7 @@
static audio_devices_t getDeviceForVolume(audio_devices_t device);
SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
SortedVector<audio_io_handle_t>& outputs2);
@@ -635,7 +643,7 @@
// if muting, wait for the audio in pcm buffer to be drained before proceeding
// if unmuting, unmute only after the specified delay
// Returns the number of ms waited
- uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+ uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
audio_devices_t prevDevice,
uint32_t delayMs);
@@ -659,10 +667,10 @@
const sp<AudioPatch>& patch);
status_t removeAudioPatch(audio_patch_handle_t handle);
- AudioOutputDescriptor *getOutputFromId(audio_port_handle_t id) const;
- AudioInputDescriptor *getInputFromId(audio_port_handle_t id) const;
- HwModule *getModuleForDevice(audio_devices_t device) const;
- HwModule *getModuleFromName(const char *name) const;
+ sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+ sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+ sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+ sp<HwModule> getModuleFromName(const char *name) const;
//
// Audio policy configuration file parsing (audio_policy.conf)
//
@@ -677,7 +685,7 @@
static audio_devices_t parseDeviceNames(char *name);
void loadHwModule(cnode *root);
void loadHwModules(cnode *root);
- void loadGlobalConfig(cnode *root, HwModule *module);
+ void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
status_t loadAudioPolicyConfig(const char *path);
void defaultAudioPolicyConfig(void);
@@ -686,11 +694,11 @@
AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
audio_io_handle_t mPrimaryOutput; // primary output handle
// list of descriptors for outputs currently opened
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
// copy of mOutputs before setDeviceConnectionState() opens new outputs
// reset to mOutputs when updateDevicesAndOutputs() is called.
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
- DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors
DeviceVector mAvailableOutputDevices; // all available output devices
DeviceVector mAvailableInputDevices; // all available input devices
int mPhoneState; // current phone state
@@ -707,13 +715,13 @@
static const uint32_t MAX_EFFECTS_MEMORY = 512;
uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
uint32_t mTotalEffectsMemory; // current memory used by effects
- KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects
+ KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects
bool mA2dpSuspended; // true if A2DP output is suspended
sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
// to boost soft sounds, used to adjust volume curves accordingly
- Vector <HwModule *> mHwModules;
+ Vector < sp<HwModule> > mHwModules;
volatile int32_t mNextUniqueId;
volatile int32_t mAudioPortGeneration;
diff --git a/services/audiopolicy/audio_policy.conf b/services/audiopolicy/audio_policy.conf
new file mode 100644
index 0000000..9b83fef
--- /dev/null
+++ b/services/audiopolicy/audio_policy.conf
@@ -0,0 +1,145 @@
+#
+# Template audio policy configuration file
+#
+
+# Global configuration section:
+# - before audio HAL version 3.0:
+# lists input and output devices always present on the device
+# as well as the output device selected by default.
+# Devices are designated by a string that corresponds to the enum in audio.h
+#
+# global_configuration {
+# attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
+# default_output_device AUDIO_DEVICE_OUT_SPEAKER
+# attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
+# }
+#
+# - after and including audio HAL 3.0 the global_configuration section is included in each
+# hardware module section.
+# it also includes the audio HAL version of this hw module:
+# global_configuration {
+# ...
+# audio_hal_version <major.minor> # audio HAL version in e.g. 3.0
+# }
+# other attributes (attached devices, default device) have to be included in the
+# global_configuration section of each hardware module
+
+
+# audio hardware module section: contains descriptors for all audio hw modules present on the
+# device. Each hw module node is named after the corresponding hw module library base name.
+# For instance, "primary" corresponds to audio.primary.<device>.so.
+# The "primary" module is mandatory and must include at least one output with
+# AUDIO_OUTPUT_FLAG_PRIMARY flag.
+# Each module descriptor contains one or more output profile descriptors and zero or more
+# input profile descriptors. Each profile lists all the parameters supported by a given output
+# or input stream category.
+# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding
+# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "\n".
+#
+# For audio HAL version posterior to 3.0 the following sections or sub sections can be present in
+# a hw module section:
+# - A "global_configuration" section: see above
+# - Optionally a "devices" section:
+# This section contains descriptors for audio devices with attributes like an address or a
+# gain controller. The syntax for the devices section and device descriptor is as follows:
+# devices {
+# <device name> { # <device name>: any string without space
+# type <device type> # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
+# address <address> # optional: device address, char string less than 64 in length
+# }
+# }
+# - one or more "gains" sections can be present in a device descriptor section.
+# If present, they describe the capabilities of gain controllers attached to this input or
+# output device. e.g. :
+# <device name> { # <device name>: any string without space
+# type <device type> # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
+# address <address> # optional: device address, char string less than 64 in length
+# gains {
+# <gain name> {
+# mode <gain modes supported> # e.g. AUDIO_GAIN_MODE_CHANNELS
+# channel_mask <controlled channels> # needed if mode AUDIO_GAIN_MODE_CHANNELS
+# min_value_mB <min value in millibel>
+# max_value_mB <max value in millibel>
+# default_value_mB <default value in millibel>
+# step_value_mB <step value in millibel>
+# min_ramp_ms <min duration in ms> # needed if mode AUDIO_GAIN_MODE_RAMP
+# max_ramp_ms <max duration ms> # needed if mode AUDIO_GAIN_MODE_RAMP
+# }
+# }
+# }
+# - when a device descriptor is present, output and input profiles can refer to this device by
+# its name in their "devices" section instead of specifying a device type. e.g. :
+# outputs {
+# primary {
+# sampling_rates 44100
+# channel_masks AUDIO_CHANNEL_OUT_STEREO
+# formats AUDIO_FORMAT_PCM_16_BIT
+# devices <device name>
+# flags AUDIO_OUTPUT_FLAG_PRIMARY
+# }
+# }
+# sample audio_policy.conf file below
+
+audio_hw_modules {
+ primary {
+ global_configuration {
+ attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
+ default_output_device AUDIO_DEVICE_OUT_SPEAKER
+ attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC
+ audio_hal_version 3.0
+ }
+ devices {
+ speaker {
+ type AUDIO_DEVICE_OUT_SPEAKER
+ gains {
+ gain_1 {
+ mode AUDIO_GAIN_MODE_JOINT
+ min_value_mB -8400
+ max_value_mB 4000
+ default_value_mB 0
+ step_value_mB 100
+ }
+ }
+ }
+ }
+ outputs {
+ primary {
+ sampling_rates 48000
+ channel_masks AUDIO_CHANNEL_OUT_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices speaker
+ flags AUDIO_OUTPUT_FLAG_PRIMARY
+ }
+ }
+ inputs {
+ primary {
+ sampling_rates 8000|16000
+ channel_masks AUDIO_CHANNEL_IN_MONO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_IN_BUILTIN_MIC
+ }
+ }
+ }
+ r_submix {
+ global_configuration {
+ attached_input_devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
+ audio_hal_version 2.0
+ }
+ outputs {
+ submix {
+ sampling_rates 48000
+ channel_masks AUDIO_CHANNEL_OUT_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+ }
+ }
+ inputs {
+ submix {
+ sampling_rates 48000
+ channel_masks AUDIO_CHANNEL_IN_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
+ }
+ }
+ }
+}
diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/audio_policy_conf.h
index 79f20f1..2535a67 100644
--- a/services/audiopolicy/audio_policy_conf.h
+++ b/services/audiopolicy/audio_policy_conf.h
@@ -35,6 +35,7 @@
#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
+#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
// hw modules descriptions
#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 9fd35e1..638b871 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -852,13 +852,12 @@
switch(deviceVersion) {
case CAMERA_DEVICE_API_VERSION_1_0:
- ALOGW("Camera using old HAL version: %d", deviceVersion);
- return -EOPNOTSUPP;
- // TODO: don't allow 2.0 Only allow 2.1 and higher
case CAMERA_DEVICE_API_VERSION_2_0:
case CAMERA_DEVICE_API_VERSION_2_1:
case CAMERA_DEVICE_API_VERSION_3_0:
case CAMERA_DEVICE_API_VERSION_3_1:
+ ALOGW("Camera using old HAL version: %d", deviceVersion);
+ return -EOPNOTSUPP;
case CAMERA_DEVICE_API_VERSION_3_2:
client = new CameraDeviceClient(this, cameraCb, String16(),
cameraId, facing, callingPid, USE_CALLING_UID, getpid());
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 4fce1b3..8154e59 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -310,6 +310,10 @@
Mutex::Autolock icl(mBinderSerializationLock);
+ if (bufferProducer == NULL) {
+ ALOGE("%s: bufferProducer must not be null", __FUNCTION__);
+ return BAD_VALUE;
+ }
if (!mDevice.get()) return DEAD_OBJECT;
// Don't create multiple streams for the same target surface