Merge "AudioSystem: refactor audio config cache and callbacks" into mnc-dev
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 0e3bc68..20c0094 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -17,7 +17,8 @@
$(TOP)/frameworks/native/include/media/openmax \
external/jpeg \
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -40,7 +41,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -63,7 +65,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -87,7 +90,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -110,7 +114,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -133,7 +138,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -157,7 +163,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -199,7 +206,8 @@
LOCAL_STATIC_LIBRARIES:= \
libstagefright_mediafilter
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
@@ -222,7 +230,8 @@
frameworks/av/media/libstagefright \
$(TOP)/frameworks/native/include/media/openmax
-LOCAL_CFLAGS += -Wno-multichar
+LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
+LOCAL_CLANG := true
LOCAL_MODULE_TAGS := optional
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index ac1a547..50913cd 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -21,6 +21,7 @@
#include "SimplePlayer.h"
#include <gui/Surface.h>
+
#include <media/AudioTrack.h>
#include <media/ICrypto.h>
#include <media/IMediaHTTPService.h>
@@ -29,7 +30,6 @@
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaCodec.h>
#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/NuMediaExtractor.h>
namespace android {
@@ -73,8 +73,7 @@
surface = new Surface(bufferProducer);
}
- msg->setObject(
- "native-window", new NativeWindowWrapper(surface));
+ msg->setObject("surface", surface);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
@@ -132,10 +131,8 @@
err = INVALID_OPERATION;
} else {
sp<RefBase> obj;
- CHECK(msg->findObject("native-window", &obj));
-
- mNativeWindow = static_cast<NativeWindowWrapper *>(obj.get());
-
+ CHECK(msg->findObject("surface", &obj));
+ mSurface = static_cast<Surface *>(obj.get());
err = OK;
}
@@ -324,7 +321,7 @@
err = state->mCodec->configure(
format,
- isVideo ? mNativeWindow->getSurfaceTextureClient() : NULL,
+ isVideo ? mSurface : NULL,
NULL /* crypto */,
0 /* flags */);
@@ -411,7 +408,7 @@
mStateByTrackIndex.clear();
mCodecLooper.clear();
mExtractor.clear();
- mNativeWindow.clear();
+ mSurface.clear();
mPath.clear();
return OK;
@@ -428,12 +425,12 @@
err = state->mCodec->dequeueInputBuffer(&index);
if (err == OK) {
- ALOGV("dequeued input buffer on track %d",
+ ALOGV("dequeued input buffer on track %zu",
mStateByTrackIndex.keyAt(i));
state->mAvailInputBufferIndices.push_back(index);
} else {
- ALOGV("dequeueInputBuffer on track %d returned %d",
+ ALOGV("dequeueInputBuffer on track %zu returned %d",
mStateByTrackIndex.keyAt(i), err);
}
} while (err == OK);
@@ -448,7 +445,7 @@
&info.mFlags);
if (err == OK) {
- ALOGV("dequeued output buffer on track %d",
+ ALOGV("dequeued output buffer on track %zu",
mStateByTrackIndex.keyAt(i));
state->mAvailOutputBufferInfos.push_back(info);
@@ -459,7 +456,7 @@
err = state->mCodec->getOutputBuffers(&state->mBuffers[1]);
CHECK_EQ(err, (status_t)OK);
} else {
- ALOGV("dequeueOutputBuffer on track %d returned %d",
+ ALOGV("dequeueOutputBuffer on track %zu returned %d",
mStateByTrackIndex.keyAt(i), err);
}
} while (err == OK
@@ -502,7 +499,7 @@
0);
CHECK_EQ(err, (status_t)OK);
- ALOGV("enqueued input data on track %d", trackIndex);
+ ALOGV("enqueued input data on track %zu", trackIndex);
err = mExtractor->advance();
CHECK_EQ(err, (status_t)OK);
@@ -528,8 +525,8 @@
bool release = true;
if (lateByUs > 30000ll) {
- ALOGI("track %d buffer late by %lld us, dropping.",
- mStateByTrackIndex.keyAt(i), lateByUs);
+ ALOGI("track %zu buffer late by %lld us, dropping.",
+ mStateByTrackIndex.keyAt(i), (long long)lateByUs);
state->mCodec->releaseOutputBuffer(info->mIndex);
} else {
if (state->mAudioTrack != NULL) {
@@ -558,8 +555,8 @@
break;
}
} else {
- ALOGV("track %d buffer early by %lld us.",
- mStateByTrackIndex.keyAt(i), -lateByUs);
+ ALOGV("track %zu buffer early by %lld us.",
+ mStateByTrackIndex.keyAt(i), (long long)-lateByUs);
break;
}
}
@@ -569,7 +566,7 @@
}
status_t SimplePlayer::onOutputFormatChanged(
- size_t trackIndex, CodecState *state) {
+ size_t trackIndex __unused, CodecState *state) {
sp<AMessage> format;
status_t err = state->mCodec->getOutputFormat(&format);
@@ -640,7 +637,7 @@
if (delayUs > 2000ll) {
ALOGW("AudioTrack::write took %lld us, numFramesAvailableToWrite=%u, "
"numFramesWritten=%u",
- delayUs, numFramesAvailableToWrite, numFramesWritten);
+ (long long)delayUs, numFramesAvailableToWrite, numFramesWritten);
}
info->mOffset += nbytes;
diff --git a/cmds/stagefright/SimplePlayer.h b/cmds/stagefright/SimplePlayer.h
index ce993e8..ae9dfd2 100644
--- a/cmds/stagefright/SimplePlayer.h
+++ b/cmds/stagefright/SimplePlayer.h
@@ -25,8 +25,8 @@
struct AudioTrack;
class IGraphicBufferProducer;
struct MediaCodec;
-struct NativeWindowWrapper;
struct NuMediaExtractor;
+class Surface;
struct SimplePlayer : public AHandler {
SimplePlayer();
@@ -84,7 +84,7 @@
State mState;
AString mPath;
- sp<NativeWindowWrapper> mNativeWindow;
+ sp<Surface> mSurface;
sp<NuMediaExtractor> mExtractor;
sp<ALooper> mCodecLooper;
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 7b0de24..6e9e6ec 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -18,6 +18,8 @@
#include <sys/stat.h>
#include <fcntl.h>
+#include <utils/String16.h>
+
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -34,7 +36,7 @@
static void usage(const char* name)
{
- fprintf(stderr, "Usage: %s [-d duration] [-m] [-w] [<output-file>]\n", name);
+ fprintf(stderr, "Usage: %s [-d du.ration] [-m] [-w] [<output-file>]\n", name);
fprintf(stderr, "Encodes either a sine wave or microphone input to AMR format\n");
fprintf(stderr, " -d duration in seconds, default 5 seconds\n");
fprintf(stderr, " -m use microphone for input, default sine source\n");
@@ -85,6 +87,7 @@
// talk into the appropriate microphone for the duration
source = new AudioSource(
AUDIO_SOURCE_MIC,
+ String16(),
kSampleRate,
channels);
} else {
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index d987250..dae9bbe 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -108,7 +108,7 @@
continue;
}
- ALOGV("selecting track %d", i);
+ ALOGV("selecting track %zu", i);
err = extractor->selectTrack(i);
CHECK_EQ(err, (status_t)OK);
@@ -151,7 +151,7 @@
CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
- ALOGV("got %d input and %d output buffers",
+ ALOGV("got %zu input and %zu output buffers",
state->mInBuffers.size(), state->mOutBuffers.size());
}
@@ -172,7 +172,7 @@
err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
if (err == OK) {
- ALOGV("filling input buffer %d", index);
+ ALOGV("filling input buffer %zu", index);
const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
@@ -209,7 +209,7 @@
state->mCodec->dequeueInputBuffer(&index, kTimeout);
if (err == OK) {
- ALOGV("signalling input EOS on track %d", i);
+ ALOGV("signalling input EOS on track %zu", i);
err = state->mCodec->queueInputBuffer(
index,
@@ -258,8 +258,8 @@
kTimeout);
if (err == OK) {
- ALOGV("draining output buffer %d, time = %lld us",
- index, presentationTimeUs);
+ ALOGV("draining output buffer %zu, time = %lld us",
+ index, (long long)presentationTimeUs);
++state->mNumBuffersDecoded;
state->mNumBytesDecoded += size;
@@ -293,7 +293,7 @@
CHECK_EQ((status_t)OK,
state->mCodec->getOutputBuffers(&state->mOutBuffers));
- ALOGV("got %d output buffers", state->mOutBuffers.size());
+ ALOGV("got %zu output buffers", state->mOutBuffers.size());
} else if (err == INFO_FORMAT_CHANGED) {
sp<AMessage> format;
CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
@@ -313,17 +313,17 @@
CHECK_EQ((status_t)OK, state->mCodec->release());
if (state->mIsAudio) {
- printf("track %zu: %" PRId64 " bytes received. %.2f KB/sec\n",
+ printf("track %zu: %lld bytes received. %.2f KB/sec\n",
i,
- state->mNumBytesDecoded,
+ (long long)state->mNumBytesDecoded,
state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
} else {
- printf("track %zu: %" PRId64 " frames decoded, %.2f fps. %" PRId64
+ printf("track %zu: %lld frames decoded, %.2f fps. %lld"
" bytes received. %.2f KB/sec\n",
i,
- state->mNumBuffersDecoded,
+ (long long)state->mNumBuffersDecoded,
state->mNumBuffersDecoded * 1E6 / elapsedTimeUs,
- state->mNumBytesDecoded,
+ (long long)state->mNumBytesDecoded,
state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
}
}
@@ -418,7 +418,7 @@
ssize_t displayWidth = info.w;
ssize_t displayHeight = info.h;
- ALOGV("display is %ld x %ld\n", displayWidth, displayHeight);
+ ALOGV("display is %zd x %zd\n", displayWidth, displayHeight);
control = composerClient->createSurface(
String8("A Surface"),
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
index f77b38b..1183112 100644
--- a/cmds/stagefright/mediafilter.cpp
+++ b/cmds/stagefright/mediafilter.cpp
@@ -81,7 +81,7 @@
return OK;
}
- status_t handleSetParameters(const sp<AMessage> &msg) {
+ status_t handleSetParameters(const sp<AMessage> &msg __unused) {
return OK;
}
@@ -101,7 +101,7 @@
return OK;
}
- status_t handleSetParameters(const sp<AMessage> &msg) {
+ status_t handleSetParameters(const sp<AMessage> &msg __unused) {
return OK;
}
@@ -121,7 +121,7 @@
return OK;
}
- status_t handleSetParameters(const sp<AMessage> &msg) {
+ status_t handleSetParameters(const sp<AMessage> &msg __unused) {
return OK;
}
@@ -597,7 +597,7 @@
if (err == OK) {
ALOGV("draining decoded buffer %zu, time = %lld us",
- frame.index, frame.presentationTimeUs);
+ frame.index, (long long)frame.presentationTimeUs);
++(state->mNumBuffersDecoded);
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index 461b56c..0029aec 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -53,7 +53,6 @@
using namespace android;
static int muxing(
- const android::sp<android::ALooper> &looper,
const char *path,
bool useAudio,
bool useVideo,
@@ -137,7 +136,7 @@
}
}
- ALOGV("selecting track %d", i);
+ ALOGV("selecting track %zu", i);
err = extractor->selectTrack(i);
CHECK_EQ(err, (status_t)OK);
@@ -308,7 +307,7 @@
sp<ALooper> looper = new ALooper;
looper->start();
- int result = muxing(looper, argv[0], useAudio, useVideo, outputFileName,
+ int result = muxing(argv[0], useAudio, useVideo, outputFileName,
enableTrim, trimStartTimeMs, trimEndTimeMs, rotationDegrees);
looper->stop();
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index fdc352e..594c933 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -32,13 +32,13 @@
using namespace android;
+static const int32_t kAudioBitRate = 12200;
+#if 0
static const int32_t kFramerate = 24; // fps
static const int32_t kIFramesIntervalSec = 1;
static const int32_t kVideoBitRate = 512 * 1024;
-static const int32_t kAudioBitRate = 12200;
static const int64_t kDurationUs = 10000000LL; // 10 seconds
-#if 0
class DummySource : public MediaSource {
public:
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index 172dc36..0d64d2f 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -38,10 +38,10 @@
#include <media/stagefright/MediaExtractor.h>
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
-#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/Utils.h>
#include <gui/SurfaceComposerClient.h>
+#include <gui/Surface.h>
#include "include/ESDS.h"
@@ -154,8 +154,7 @@
sp<AMessage> format = makeFormat(mSource->getFormat());
if (mSurface != NULL) {
- format->setObject(
- "native-window", new NativeWindowWrapper(mSurface));
+ format->setObject("surface", mSurface);
}
mCodec->initiateSetup(format);
@@ -328,14 +327,14 @@
CHECK(size >= 7);
CHECK_EQ((unsigned)ptr[0], 1u); // configurationVersion == 1
- uint8_t profile = ptr[1];
- uint8_t level = ptr[3];
+ uint8_t profile __unused = ptr[1];
+ uint8_t level __unused = ptr[3];
// There is decodable content out there that fails the following
// assertion, let's be lenient for now...
// CHECK((ptr[4] >> 2) == 0x3f); // reserved
- size_t lengthSize = 1 + (ptr[4] & 3);
+ size_t lengthSize __unused = 1 + (ptr[4] & 3);
// commented out check below as H264_QVGA_500_NO_AUDIO.3gp
// violates it...
@@ -491,7 +490,7 @@
if (sizeNeeded > sizeLeft) {
if (outBuffer->size() == 0) {
- ALOGE("Unable to fit even a single input buffer of size %d.",
+ ALOGE("Unable to fit even a single input buffer of size %zu.",
sizeNeeded);
}
CHECK_GT(outBuffer->size(), 0u);
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 318b56d..a9c6eda 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -965,7 +965,7 @@
OMXClient client;
status_t err = client.connect();
- for (int k = 0; k < argc; ++k) {
+ for (int k = 0; k < argc && err == OK; ++k) {
bool syncInfoPresent = true;
const char *filename = argv[k];
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 0566d14..1a40e53 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -269,7 +269,7 @@
: mEOS(false) {
}
- virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) {
+ virtual void notify(int msg, int ext1 __unused, int ext2 __unused, const Parcel *obj __unused) {
Mutex::Autolock autoLock(mLock);
if (msg == MEDIA_ERROR || msg == MEDIA_PLAYBACK_COMPLETE) {
@@ -318,7 +318,7 @@
ssize_t displayWidth = info.w;
ssize_t displayHeight = info.h;
- ALOGV("display is %d x %d\n", displayWidth, displayHeight);
+ ALOGV("display is %zd x %zd\n", displayWidth, displayHeight);
sp<SurfaceControl> control =
composerClient->createSurface(
diff --git a/drm/common/IDrmManagerService.cpp b/drm/common/IDrmManagerService.cpp
index 3f62ed7..b90da1b 100644
--- a/drm/common/IDrmManagerService.cpp
+++ b/drm/common/IDrmManagerService.cpp
@@ -34,6 +34,7 @@
#include "IDrmManagerService.h"
#define INVALID_BUFFER_LENGTH -1
+#define MAX_BINDER_TRANSACTION_SIZE ((1*1024*1024)-(4096*2))
using namespace android;
@@ -933,7 +934,12 @@
//Filling DRM info
const int infoType = data.readInt32();
- const int bufferSize = data.readInt32();
+ const uint32_t bufferSize = data.readInt32();
+
+ if (bufferSize > data.dataAvail()) {
+ return BAD_VALUE;
+ }
+
char* buffer = NULL;
if (0 < bufferSize) {
buffer = (char *)data.readInplace(bufferSize);
@@ -986,6 +992,9 @@
const int size = data.readInt32();
for (int index = 0; index < size; ++index) {
+ if (!data.dataAvail()) {
+ break;
+ }
const String8 key(data.readString8());
if (key == String8("FileDescriptorKey")) {
char buffer[16];
@@ -1035,7 +1044,12 @@
const int uniqueId = data.readInt32();
//Filling DRM Rights
- const int bufferSize = data.readInt32();
+ const uint32_t bufferSize = data.readInt32();
+ if (bufferSize > data.dataAvail()) {
+ reply->writeInt32(BAD_VALUE);
+ return DRM_NO_ERROR;
+ }
+
const DrmBuffer drmBuffer((char *)data.readInplace(bufferSize), bufferSize);
const String8 mimeType(data.readString8());
@@ -1206,10 +1220,13 @@
const int convertId = data.readInt32();
//Filling input data
- const int bufferSize = data.readInt32();
+ const uint32_t bufferSize = data.readInt32();
+ if (bufferSize > data.dataAvail()) {
+ return BAD_VALUE;
+ }
DrmBuffer* inputData = new DrmBuffer((char *)data.readInplace(bufferSize), bufferSize);
- DrmConvertedStatus* drmConvertedStatus = convertData(uniqueId, convertId, inputData);
+ DrmConvertedStatus* drmConvertedStatus = convertData(uniqueId, convertId, inputData);
if (NULL != drmConvertedStatus) {
//Filling Drm Converted Ststus
@@ -1393,7 +1410,12 @@
const int decryptUnitId = data.readInt32();
//Filling Header info
- const int bufferSize = data.readInt32();
+ const uint32_t bufferSize = data.readInt32();
+ if (bufferSize > data.dataAvail()) {
+ reply->writeInt32(BAD_VALUE);
+ clearDecryptHandle(&handle);
+ return DRM_NO_ERROR;
+ }
DrmBuffer* headerInfo = NULL;
headerInfo = new DrmBuffer((char *)data.readInplace(bufferSize), bufferSize);
@@ -1417,9 +1439,17 @@
readDecryptHandleFromParcelData(&handle, data);
const int decryptUnitId = data.readInt32();
- const int decBufferSize = data.readInt32();
+ const uint32_t decBufferSize = data.readInt32();
+ const uint32_t encBufferSize = data.readInt32();
- const int encBufferSize = data.readInt32();
+ if (encBufferSize > data.dataAvail() ||
+ decBufferSize > MAX_BINDER_TRANSACTION_SIZE) {
+ reply->writeInt32(BAD_VALUE);
+ reply->writeInt32(0);
+ clearDecryptHandle(&handle);
+ return DRM_NO_ERROR;
+ }
+
DrmBuffer* encBuffer
= new DrmBuffer((char *)data.readInplace(encBufferSize), encBufferSize);
@@ -1429,8 +1459,10 @@
DrmBuffer* IV = NULL;
if (0 != data.dataAvail()) {
- const int ivBufferlength = data.readInt32();
- IV = new DrmBuffer((char *)data.readInplace(ivBufferlength), ivBufferlength);
+ const uint32_t ivBufferlength = data.readInt32();
+ if (ivBufferlength <= data.dataAvail()) {
+ IV = new DrmBuffer((char *)data.readInplace(ivBufferlength), ivBufferlength);
+ }
}
const status_t status
@@ -1477,7 +1509,11 @@
DecryptHandle handle;
readDecryptHandleFromParcelData(&handle, data);
- const int numBytes = data.readInt32();
+ const uint32_t numBytes = data.readInt32();
+ if (numBytes > MAX_BINDER_TRANSACTION_SIZE) {
+ reply->writeInt32(BAD_VALUE);
+ return DRM_NO_ERROR;
+ }
char* buffer = new char[numBytes];
const off64_t offset = data.readInt64();
diff --git a/include/media/AVSyncSettings.h b/include/media/AVSyncSettings.h
new file mode 100644
index 0000000..10e3bcc
--- /dev/null
+++ b/include/media/AVSyncSettings.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright 2015 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_AV_SYNC_SETTINGS_H
+#define ANDROID_AV_SYNC_SETTINGS_H
+
+namespace android {
+
+enum AVSyncSource : unsigned {
+ // let the system decide the best sync source
+ AVSYNC_SOURCE_DEFAULT = 0,
+ // sync to the system clock
+ AVSYNC_SOURCE_SYSTEM_CLOCK = 1,
+ // sync to the audio track
+ AVSYNC_SOURCE_AUDIO = 2,
+ // sync to the display vsync
+ AVSYNC_SOURCE_VSYNC = 3,
+ AVSYNC_SOURCE_MAX,
+};
+
+enum AVSyncAudioAdjustMode : unsigned {
+ // let the system decide the best audio adjust mode
+ AVSYNC_AUDIO_ADJUST_MODE_DEFAULT = 0,
+ // adjust audio by time stretching
+ AVSYNC_AUDIO_ADJUST_MODE_STRETCH = 1,
+ // adjust audio by resampling
+ AVSYNC_AUDIO_ADJUST_MODE_RESAMPLE = 2,
+ AVSYNC_AUDIO_ADJUST_MODE_MAX,
+};
+
+// max tolerance when adjusting playback speed to desired playback speed
+#define AVSYNC_TOLERANCE_MAX 1.0f
+
+struct AVSyncSettings {
+ AVSyncSource mSource;
+ AVSyncAudioAdjustMode mAudioAdjustMode;
+ float mTolerance;
+ AVSyncSettings()
+ : mSource(AVSYNC_SOURCE_DEFAULT),
+ mAudioAdjustMode(AVSYNC_AUDIO_ADJUST_MODE_DEFAULT),
+ mTolerance(.044f) { }
+};
+
+} // namespace android
+
+// ---------------------------------------------------------------------------
+
+#endif // ANDROID_AV_SYNC_SETTINGS_H
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 583695d..61da4f2 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -201,8 +201,12 @@
*/
/* Simple Constructor.
+ *
+ * Parameters:
+ *
+ * opPackageName: The package name used for app op checks.
*/
- AudioEffect();
+ AudioEffect(const String16& opPackageName);
/* Constructor.
@@ -211,6 +215,7 @@
*
* type: type of effect created: can be null if uuid is specified. This corresponds to
* the OpenSL ES interface implemented by this effect.
+ * opPackageName: The package name used for app op checks.
* uuid: Uuid of effect created: can be null if type is specified. This uuid corresponds to
* a particular implementation of an effect type.
* priority: requested priority for effect control: the priority level corresponds to the
@@ -227,6 +232,7 @@
*/
AudioEffect(const effect_uuid_t *type,
+ const String16& opPackageName,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
effect_callback_t cbf = NULL,
@@ -239,6 +245,7 @@
* Same as above but with type and uuid specified by character strings
*/
AudioEffect(const char *typeStr,
+ const String16& opPackageName,
const char *uuidStr = NULL,
int32_t priority = 0,
effect_callback_t cbf = NULL,
@@ -406,7 +413,9 @@
void* mUserData; // client context for callback function
effect_descriptor_t mDescriptor; // effect descriptor
int32_t mId; // system wide unique effect engine instance ID
- Mutex mLock; // Mutex for mEnabled access
+ Mutex mLock; // Mutex for mEnabled access
+
+ String16 mOpPackageName; // The package name used for app op checks.
// IEffectClient
virtual void controlStatusChanged(bool controlGranted);
diff --git a/include/media/AudioPolicy.h b/include/media/AudioPolicy.h
index 800b27b..feed402 100644
--- a/include/media/AudioPolicy.h
+++ b/include/media/AudioPolicy.h
@@ -38,14 +38,17 @@
#define MIX_TYPE_PLAYERS 0
#define MIX_TYPE_RECORDERS 1
+// definition of the different events that can be reported on a dynamic policy from
+// AudioSystem's implementation of the AudioPolicyClient interface
+// keep in sync with AudioSystem.java
+#define DYNAMIC_POLICY_EVENT_MIX_STATE_UPDATE 0
+
#define MIX_STATE_DISABLED -1
#define MIX_STATE_IDLE 0
#define MIX_STATE_MIXING 1
-#define ROUTE_FLAG_RENDER 0x1
-#define ROUTE_FLAG_LOOP_BACK (0x1 << 1)
-
-#define MIX_FLAG_NOTIFY_ACTIVITY 0x1
+#define MIX_ROUTE_FLAG_RENDER 0x1
+#define MIX_ROUTE_FLAG_LOOP_BACK (0x1 << 1)
#define MAX_MIXES_PER_POLICY 10
#define MAX_CRITERIA_PER_MIX 20
@@ -67,11 +70,15 @@
class AudioMix {
public:
+ // flag on an AudioMix indicating the activity on this mix (IDLE, MIXING)
+ // must be reported through the AudioPolicyClient interface
+ static const uint32_t kCbFlagNotifyActivity = 0x1;
+
AudioMix() {}
AudioMix(Vector<AttributeMatchCriterion> criteria, uint32_t mixType, audio_config_t format,
uint32_t routeFlags, String8 registrationId, uint32_t flags) :
mCriteria(criteria), mMixType(mixType), mFormat(format),
- mRouteFlags(routeFlags), mRegistrationId(registrationId), mFlags(flags){}
+ mRouteFlags(routeFlags), mRegistrationId(registrationId), mCbFlags(flags){}
status_t readFromParcel(Parcel *parcel);
status_t writeToParcel(Parcel *parcel) const;
@@ -81,7 +88,7 @@
audio_config_t mFormat;
uint32_t mRouteFlags;
String8 mRegistrationId;
- uint32_t mFlags;
+ uint32_t mCbFlags; // flags indicating which callbacks to use, see kCbFlag*
};
}; // namespace android
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index c24a28d..b743c11 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -129,8 +129,12 @@
/* Constructs an uninitialized AudioRecord. No connection with
* AudioFlinger takes place. Use set() after this.
+ *
+ * Parameters:
+ *
+ * opPackageName: The package name used for app ops.
*/
- AudioRecord();
+ AudioRecord(const String16& opPackageName);
/* Creates an AudioRecord object and registers it with AudioFlinger.
* Once created, the track needs to be started before it can be used.
@@ -143,6 +147,7 @@
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
* channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true.
+ * opPackageName: The package name used for app ops.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
* latency of the track. The actual size selected by the AudioRecord could
@@ -165,6 +170,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& opPackageName,
size_t frameCount = 0,
callback_t cbf = NULL,
void* user = NULL,
@@ -483,7 +489,7 @@
// caller must hold lock on mLock for all _l methods
- status_t openRecord_l(size_t epoch);
+ status_t openRecord_l(size_t epoch, const String16& opPackageName);
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreRecord_l(const char *from);
@@ -520,6 +526,8 @@
status_t mStatus;
+ String16 mOpPackageName; // The package name used for app ops.
+
size_t mFrameCount; // corresponds to current IAudioRecord, value is
// reported back by AudioFlinger to the client
size_t mReqFrameCount; // frame count to request the first or next time
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 8d3396d..0cbcdb1 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -30,6 +30,7 @@
namespace android {
typedef void (*audio_error_callback)(status_t err);
+typedef void (*dynamic_policy_callback)(int event, String8 regId, int val);
class IAudioFlinger;
class IAudioPolicyService;
@@ -90,6 +91,7 @@
static String8 getParameters(const String8& keys);
static void setErrorCallback(audio_error_callback cb);
+ static void setDynPolicyCallback(dynamic_policy_callback cb);
// helper function to obtain AudioFlinger service handle
static const sp<IAudioFlinger> get_audio_flinger();
@@ -198,6 +200,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
@@ -219,6 +222,7 @@
static status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -398,6 +402,7 @@
static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient
static sp<IAudioFlinger> gAudioFlinger;
static audio_error_callback gAudioErrorCallback;
+ static dynamic_policy_callback gDynPolicyCallback;
static size_t gInBuffSize;
// previous parameters for recording buffer size queries
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e7ee0ce..51d40bb 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -360,6 +360,11 @@
/* Return current source sample rate in Hz */
uint32_t getSampleRate() const;
+ /* Return the original source sample rate in Hz. This corresponds to the sample rate
+ * if playback rate had normal speed and pitch.
+ */
+ uint32_t getOriginalSampleRate() const;
+
/* Set source playback rate for timestretch
* 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster
* 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch
@@ -749,6 +754,7 @@
float mVolume[2];
float mSendLevel;
mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it
+ uint32_t mOriginalSampleRate;
AudioPlaybackRate mPlaybackRate;
size_t mFrameCount; // corresponds to current IAudioTrack, value is
// reported back by AudioFlinger to the client
@@ -832,6 +838,9 @@
int64_t mStartUs; // the start time after flush or stop.
// only used for offloaded and direct tracks.
+ bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid
+ AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion
+
audio_output_flags_t mFlags;
// const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD.
// mLock must be held to read or write those bits reliably.
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index f927a80..046345c 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -85,6 +85,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& callingPackage,
size_t *pFrameCount,
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
@@ -198,6 +199,7 @@
// AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
int sessionId,
+ const String16& callingPackage,
status_t *status,
int *id,
int *enabled) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 56a1dc6..ee462a0 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -62,6 +62,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = 0,
@@ -80,6 +81,7 @@
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index df6130d..0fd8933 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -35,6 +35,8 @@
struct IStreamSource;
class IGraphicBufferProducer;
struct IMediaHTTPService;
+struct AudioPlaybackRate;
+struct AVSyncSettings;
class IMediaPlayer: public IInterface
{
@@ -58,7 +60,11 @@
virtual status_t stop() = 0;
virtual status_t pause() = 0;
virtual status_t isPlaying(bool* state) = 0;
- virtual status_t setPlaybackRate(float rate) = 0;
+ virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate) = 0;
+ virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) = 0;
+ virtual status_t setSyncSettings(const AVSyncSettings& sync, float videoFpsHint) = 0;
+ virtual status_t getSyncSettings(AVSyncSettings* sync /* nonnull */,
+ float* videoFps /* nonnull */) = 0;
virtual status_t seekTo(int msec) = 0;
virtual status_t getCurrentPosition(int* msec) = 0;
virtual status_t getDuration(int* msec) = 0;
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 49a3d61..a316ce2 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -47,7 +47,7 @@
public:
DECLARE_META_INTERFACE(MediaPlayerService);
- virtual sp<IMediaRecorder> createMediaRecorder() = 0;
+ virtual sp<IMediaRecorder> createMediaRecorder(const String16 &opPackageName) = 0;
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0)
= 0;
@@ -65,8 +65,8 @@
// display client when display connection, disconnection or errors occur.
// The assumption is that at most one remote display will be connected to the
// provided interface at a time.
- virtual sp<IRemoteDisplay> listenForRemoteDisplay(const sp<IRemoteDisplayClient>& client,
- const String8& iface) = 0;
+ virtual sp<IRemoteDisplay> listenForRemoteDisplay(const String16 &opPackageName,
+ const sp<IRemoteDisplayClient>& client, const String8& iface) = 0;
// codecs and audio devices usage tracking for the battery app
enum BatteryDataBits {
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 824762a..fa917f9 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -26,8 +26,10 @@
#include <utils/RefBase.h>
#include <media/mediaplayer.h>
+#include <media/AudioResamplerPublic.h>
#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
+#include <media/AVSyncSettings.h>
#include <media/Metadata.h>
// Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is
@@ -132,7 +134,8 @@
virtual void pause() = 0;
virtual void close() = 0;
- virtual status_t setPlaybackRatePermille(int32_t /* rate */) { return INVALID_OPERATION;}
+ virtual status_t setPlaybackRate(const AudioPlaybackRate& rate) = 0;
+ virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */) = 0;
virtual bool needsTrailingPadding() { return true; }
virtual status_t setParameters(const String8& /* keyValuePairs */) { return NO_ERROR; }
@@ -173,7 +176,31 @@
virtual status_t stop() = 0;
virtual status_t pause() = 0;
virtual bool isPlaying() = 0;
- virtual status_t setPlaybackRate(float /* rate */) { return INVALID_OPERATION; }
+ virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate) {
+ // by default, players only support setting rate to the default
+ if (!isAudioPlaybackRateEqual(rate, AUDIO_PLAYBACK_RATE_DEFAULT)) {
+ return BAD_VALUE;
+ }
+ return OK;
+ }
+ virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) {
+ *rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ return OK;
+ }
+ virtual status_t setSyncSettings(const AVSyncSettings& sync, float /* videoFps */) {
+ // By default, players only support setting sync source to default; all other sync
+ // settings are ignored. There is no requirement for getters to return set values.
+ if (sync.mSource != AVSYNC_SOURCE_DEFAULT) {
+ return BAD_VALUE;
+ }
+ return OK;
+ }
+ virtual status_t getSyncSettings(
+ AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) {
+ *sync = AVSyncSettings();
+ *videoFps = -1.f;
+ return OK;
+ }
virtual status_t seekTo(int msec) = 0;
virtual status_t getCurrentPosition(int *msec) = 0;
virtual status_t getDuration(int *msec) = 0;
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
index f55063e..f9feede 100644
--- a/include/media/MediaRecorderBase.h
+++ b/include/media/MediaRecorderBase.h
@@ -29,7 +29,8 @@
class IGraphicBufferProducer;
struct MediaRecorderBase {
- MediaRecorderBase() {}
+ MediaRecorderBase(const String16 &opPackageName)
+ : mOpPackageName(opPackageName) {}
virtual ~MediaRecorderBase() {}
virtual status_t init() = 0;
@@ -57,6 +58,10 @@
virtual status_t dump(int fd, const Vector<String16>& args) const = 0;
virtual sp<IGraphicBufferProducer> querySurfaceMediaSource() const = 0;
+
+protected:
+ String16 mOpPackageName;
+
private:
MediaRecorderBase(const MediaRecorderBase &);
MediaRecorderBase &operator=(const MediaRecorderBase &);
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index 6167dd6..b92f816 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -65,7 +65,8 @@
/* Constructor.
* See AudioEffect constructor for details on parameters.
*/
- Visualizer(int32_t priority = 0,
+ Visualizer(const String16& opPackageName,
+ int32_t priority = 0,
effect_callback_t cbf = NULL,
void* user = NULL,
int sessionId = 0);
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 256fa9a..3fe749c 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -20,6 +20,8 @@
#include <arpa/inet.h>
#include <binder/IMemory.h>
+
+#include <media/AudioResamplerPublic.h>
#include <media/IMediaPlayerClient.h>
#include <media/IMediaPlayer.h>
#include <media/IMediaDeathNotifier.h>
@@ -32,8 +34,9 @@
namespace android {
-class Surface;
+struct AVSyncSettings;
class IGraphicBufferProducer;
+class Surface;
enum media_event_type {
MEDIA_NOP = 0, // interface test message
@@ -223,7 +226,12 @@
status_t stop();
status_t pause();
bool isPlaying();
- status_t setPlaybackRate(float rate);
+ status_t setPlaybackSettings(const AudioPlaybackRate& rate);
+ status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */);
+ status_t setSyncSettings(const AVSyncSettings& sync, float videoFpsHint);
+ status_t getSyncSettings(
+ AVSyncSettings* sync /* nonnull */,
+ float* videoFps /* nonnull */);
status_t getVideoWidth(int *w);
status_t getVideoHeight(int *h);
status_t seekTo(int msec);
@@ -278,7 +286,6 @@
int mVideoWidth;
int mVideoHeight;
int mAudioSessionId;
- float mPlaybackRate;
float mSendLevel;
struct sockaddr_in mRetransmitEndpoint;
bool mRetransmitEndpointValid;
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 74a6469..8e40c5d 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -209,7 +209,7 @@
public virtual IMediaDeathNotifier
{
public:
- MediaRecorder();
+ MediaRecorder(const String16& opPackageName);
~MediaRecorder();
void died();
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 98c4fa7..e0cd965 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -25,9 +25,10 @@
namespace android {
-class MediaSource;
+struct AudioPlaybackRate;
class AudioTrack;
struct AwesomePlayer;
+class MediaSource;
class AudioPlayer : public TimeSource {
public:
@@ -73,7 +74,8 @@
bool isSeeking();
bool reachedEOS(status_t *finalStatus);
- status_t setPlaybackRatePermille(int32_t ratePermille);
+ status_t setPlaybackRate(const AudioPlaybackRate &rate);
+ status_t getPlaybackRate(AudioPlaybackRate *rate /* nonnull */);
void notifyAudioEOS();
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 4c9aaad..50cf371 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -35,6 +35,7 @@
// _not_ a bitmask of audio_channels_t constants.
AudioSource(
audio_source_t inputSource,
+ const String16 &opPackageName,
uint32_t sampleRate,
uint32_t channels = 1);
diff --git a/include/media/stagefright/MediaSync.h b/include/media/stagefright/MediaSync.h
index 8ad74a4..a349986 100644
--- a/include/media/stagefright/MediaSync.h
+++ b/include/media/stagefright/MediaSync.h
@@ -20,9 +20,12 @@
#include <gui/IConsumerListener.h>
#include <gui/IProducerListener.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/AVSyncSettings.h>
#include <media/stagefright/foundation/AHandler.h>
#include <utils/Condition.h>
+#include <utils/KeyedVector.h>
#include <utils/Mutex.h>
namespace android {
@@ -76,10 +79,7 @@
// Called when audio track is used as media clock source. It should be
// called before updateQueuedAudioData().
- // |nativeSampleRateInHz| is the sample rate of audio data fed into audio
- // track. It's the same number used to create AudioTrack.
- status_t configureAudioTrack(
- const sp<AudioTrack> &audioTrack, uint32_t nativeSampleRateInHz);
+ status_t configureAudioTrack(const sp<AudioTrack> &audioTrack);
// Create a surface for client to render video frames. This is the surface
// on which the client should render video frames. Those video frames will
@@ -98,21 +98,31 @@
// Set the consumer name of the input queue.
void setName(const AString &name);
- // Set the playback in a desired speed.
- // This method can be called any time.
- // |rate| is the ratio between desired speed and the normal one, and should
- // be non-negative. The meaning of rate values:
- // 1.0 -- normal playback
- // 0.0 -- stop or pause
- // larger than 1.0 -- faster than normal speed
- // between 0.0 and 1.0 -- slower than normal speed
- status_t setPlaybackRate(float rate);
-
// Get the media clock used by the MediaSync so that the client can obtain
// corresponding media time or real time via
// MediaClock::getMediaTime() and MediaClock::getRealTimeFor().
sp<const MediaClock> getMediaClock();
+ // Set the video frame rate hint - this is used by the video FrameScheduler
+ status_t setVideoFrameRateHint(float rate);
+
+ // Get the video frame rate measurement from the FrameScheduler
+ // returns -1 if there is no measurement
+ float getVideoFrameRate();
+
+ // Set the sync settings parameters.
+ status_t setSyncSettings(const AVSyncSettings &syncSettings);
+
+ // Gets the sync settings parameters.
+ void getSyncSettings(AVSyncSettings *syncSettings /* nonnull */);
+
+ // Sets the playback rate using playback settings.
+ // This method can be called any time.
+ status_t setPlaybackSettings(const AudioPlaybackRate &rate);
+
+ // Gets the playback rate (playback settings parameters).
+ void getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
+
// Get the play time for pending audio frames in audio sink.
status_t getPlayTimeForPendingAudioFrames(int64_t *outTimeUs);
@@ -190,9 +200,19 @@
int64_t mNextBufferItemMediaUs;
List<BufferItem> mBufferItems;
+
+ // Keep track of buffers received from |mInput|. This is needed because
+ // it's possible the consumer of |mOutput| could return a different
+ // GraphicBuffer::handle (e.g., due to passing buffers through IPC),
+ // and that could cause problem if the producer of |mInput| only
+ // supports pre-registered buffers.
+ KeyedVector<uint64_t, sp<GraphicBuffer> > mBuffersFromInput;
sp<ALooper> mLooper;
float mPlaybackRate;
+ AudioPlaybackRate mPlaybackSettings;
+ AVSyncSettings mSyncSettings;
+
sp<MediaClock> mMediaClock;
MediaSync();
@@ -231,6 +251,22 @@
// up. This must be called with mMutex locked.
void onAbandoned_l(bool isInput);
+ // Set the playback in a desired speed.
+ // This method can be called any time.
+ // |rate| is the ratio between desired speed and the normal one, and should
+ // be non-negative. The meaning of rate values:
+ // 1.0 -- normal playback
+ // 0.0 -- stop or pause
+ // larger than 1.0 -- faster than normal speed
+ // between 0.0 and 1.0 -- slower than normal speed
+ void updatePlaybackRate_l(float rate);
+
+ // apply new sync settings
+ void resync_l();
+
+ // apply playback settings only - without resyncing or updating playback rate
+ status_t setPlaybackSettings_l(const AudioPlaybackRate &rate);
+
// helper.
bool isPlaying() { return mPlaybackRate != 0.0; }
diff --git a/include/media/stagefright/NativeWindowWrapper.h b/include/media/stagefright/NativeWindowWrapper.h
deleted file mode 100644
index cfeec22..0000000
--- a/include/media/stagefright/NativeWindowWrapper.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NATIVE_WINDOW_WRAPPER_H_
-
-#define NATIVE_WINDOW_WRAPPER_H_
-
-#include <gui/Surface.h>
-
-namespace android {
-
-// Surface derives from ANativeWindow which derives from multiple
-// base classes, in order to carry it in AMessages, we'll temporarily wrap it
-// into a NativeWindowWrapper.
-
-struct NativeWindowWrapper : RefBase {
- NativeWindowWrapper(
- const sp<Surface> &surfaceTextureClient) :
- mSurfaceTextureClient(surfaceTextureClient) { }
-
- sp<ANativeWindow> getNativeWindow() const {
- return mSurfaceTextureClient;
- }
-
- sp<Surface> getSurfaceTextureClient() const {
- return mSurfaceTextureClient;
- }
-
-private:
- const sp<Surface> mSurfaceTextureClient;
-
- DISALLOW_EVIL_CONSTRUCTORS(NativeWindowWrapper);
-};
-
-} // namespace android
-
-#endif // NATIVE_WINDOW_WRAPPER_H_
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index 0ce1603..5e9d7d4 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -76,6 +76,15 @@
bool operator <(const HLSTime &t0, const HLSTime &t1);
+// read and write various object to/from AMessage
+
+void writeToAMessage(sp<AMessage> msg, const AudioPlaybackRate &rate);
+void readFromAMessage(const sp<AMessage> &msg, AudioPlaybackRate *rate /* nonnull */);
+
+void writeToAMessage(sp<AMessage> msg, const AVSyncSettings &sync, float videoFpsHint);
+void readFromAMessage(
+ const sp<AMessage> &msg, AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
+
} // namespace android
#endif // UTILS_H_
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 7d8222f..bbeb854 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -35,13 +35,14 @@
// ---------------------------------------------------------------------------
-AudioEffect::AudioEffect()
- : mStatus(NO_INIT)
+AudioEffect::AudioEffect(const String16& opPackageName)
+ : mStatus(NO_INIT), mOpPackageName(opPackageName)
{
}
AudioEffect::AudioEffect(const effect_uuid_t *type,
+ const String16& opPackageName,
const effect_uuid_t *uuid,
int32_t priority,
effect_callback_t cbf,
@@ -49,12 +50,13 @@
int sessionId,
audio_io_handle_t io
)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT), mOpPackageName(opPackageName)
{
mStatus = set(type, uuid, priority, cbf, user, sessionId, io);
}
AudioEffect::AudioEffect(const char *typeStr,
+ const String16& opPackageName,
const char *uuidStr,
int32_t priority,
effect_callback_t cbf,
@@ -62,7 +64,7 @@
int sessionId,
audio_io_handle_t io
)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT), mOpPackageName(opPackageName)
{
effect_uuid_t type;
effect_uuid_t *pType = NULL;
@@ -128,7 +130,7 @@
mIEffectClient = new EffectClient(this);
iEffect = audioFlinger->createEffect((effect_descriptor_t *)&mDescriptor,
- mIEffectClient, priority, io, mSessionId, &mStatus, &mId, &enabled);
+ mIEffectClient, priority, io, mSessionId, mOpPackageName, &mStatus, &mId, &enabled);
if (iEffect == 0 || (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS)) {
ALOGE("set(): AudioFlinger could not create effect, status: %d", mStatus);
diff --git a/media/libmedia/AudioPolicy.cpp b/media/libmedia/AudioPolicy.cpp
index 786eb63..9d07011 100644
--- a/media/libmedia/AudioPolicy.cpp
+++ b/media/libmedia/AudioPolicy.cpp
@@ -68,7 +68,7 @@
mFormat.format = (audio_format_t)parcel->readInt32();
mRouteFlags = parcel->readInt32();
mRegistrationId = parcel->readString8();
- mFlags = (uint32_t)parcel->readInt32();
+ mCbFlags = (uint32_t)parcel->readInt32();
size_t size = (size_t)parcel->readInt32();
if (size > MAX_CRITERIA_PER_MIX) {
size = MAX_CRITERIA_PER_MIX;
@@ -90,7 +90,7 @@
parcel->writeInt32(mFormat.format);
parcel->writeInt32(mRouteFlags);
parcel->writeString8(mRegistrationId);
- parcel->writeInt32(mFlags);
+ parcel->writeInt32(mCbFlags);
size_t size = mCriteria.size();
if (size > MAX_CRITERIA_PER_MIX) {
size = MAX_CRITERIA_PER_MIX;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 9f5c4c5..23015c0 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -65,8 +65,8 @@
// ---------------------------------------------------------------------------
-AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
+AudioRecord::AudioRecord(const String16 &opPackageName)
+ : mStatus(NO_INIT), mOpPackageName(opPackageName), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
@@ -77,6 +77,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& opPackageName,
size_t frameCount,
callback_t cbf,
void* user,
@@ -85,7 +86,9 @@
transfer_type transferType,
audio_input_flags_t flags,
const audio_attributes_t* pAttributes)
- : mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
+ : mStatus(NO_INIT),
+ mOpPackageName(opPackageName),
+ mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(NULL),
@@ -136,9 +139,9 @@
const audio_attributes_t* pAttributes)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
- "notificationFrames %u, sessionId %d, transferType %d, flags %#x",
+ "notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s",
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
- sessionId, transferType, flags);
+ sessionId, transferType, flags, String8(mOpPackageName).string());
switch (transferType) {
case TRANSFER_DEFAULT:
@@ -235,7 +238,7 @@
}
// create the IAudioRecord
- status_t status = openRecord_l(0 /*epoch*/);
+ status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
@@ -435,7 +438,7 @@
// -------------------------------------------------------------------------
// must be called with mLock held
-status_t AudioRecord::openRecord_l(size_t epoch)
+status_t AudioRecord::openRecord_l(size_t epoch, const String16& opPackageName)
{
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
@@ -478,6 +481,7 @@
audio_io_handle_t input;
status_t status = AudioSystem::getInputForAttr(&mAttributes, &input,
(audio_session_t)mSessionId,
+ IPCThreadState::self()->getCallingUid(),
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId);
@@ -502,8 +506,10 @@
sp<IMemory> iMem; // for cblk
sp<IMemory> bufferMem;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
- mSampleRate, mFormat,
+ mSampleRate,
+ mFormat,
mChannelMask,
+ opPackageName,
&temp,
&trackFlags,
tid,
@@ -1032,7 +1038,7 @@
// It will also delete the strong references on previous IAudioRecord and IMemory
size_t position = mProxy->getPosition();
mNewPosition = position + mUpdatePeriod;
- status_t result = openRecord_l(position);
+ status_t result = openRecord_l(position, mOpPackageName);
if (result == NO_ERROR) {
if (mActive) {
// callback thread or sync event hasn't changed
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 01e6d71..85ed2b1 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -36,6 +36,7 @@
sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
+dynamic_policy_callback AudioSystem::gDynPolicyCallback = NULL;
// establish binder interface to AudioFlinger service
@@ -531,12 +532,18 @@
return desc;
}
-void AudioSystem::setErrorCallback(audio_error_callback cb)
+/*static*/ void AudioSystem::setErrorCallback(audio_error_callback cb)
{
Mutex::Autolock _l(gLock);
gAudioErrorCallback = cb;
}
+/*static*/ void AudioSystem::setDynPolicyCallback(dynamic_policy_callback cb)
+{
+ Mutex::Autolock _l(gLock);
+ gDynPolicyCallback = cb;
+}
+
// client singleton for AudioPolicyService binder interface
// protected by gLockAPS
sp<IAudioPolicyService> AudioSystem::gAudioPolicyService;
@@ -648,6 +655,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -657,7 +665,7 @@
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return NO_INIT;
- return aps->getOutputForAttr(attr, output, session, stream,
+ return aps->getOutputForAttr(attr, output, session, stream, uid,
samplingRate, format, channelMask,
flags, selectedDeviceId, offloadInfo);
}
@@ -692,6 +700,7 @@
status_t AudioSystem::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -701,7 +710,7 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return NO_INIT;
return aps->getInputForAttr(
- attr, input, session, samplingRate, format, channelMask, flags, selectedDeviceId);
+ attr, input, session, uid, samplingRate, format, channelMask, flags, selectedDeviceId);
}
status_t AudioSystem::startInput(audio_io_handle_t input,
@@ -937,6 +946,7 @@
return gAudioPolicyServiceClient->addAudioPortCallback(callBack);
}
+/*static*/
status_t AudioSystem::removeAudioPortCallback(const sp<AudioPortCallback>& callBack)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -949,7 +959,6 @@
return gAudioPolicyServiceClient->removeAudioPortCallback(callBack);
}
-
status_t AudioSystem::acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
audio_devices_t *device)
@@ -1047,7 +1056,16 @@
void AudioSystem::AudioPolicyServiceClient::onDynamicPolicyMixStateUpdate(
String8 regId, int32_t state)
{
- ALOGV("TODO propagate onDynamicPolicyMixStateUpdate(%s, %d)", regId.string(), state);
+ ALOGV("AudioPolicyServiceClient::onDynamicPolicyMixStateUpdate(%s, %d)", regId.string(), state);
+ dynamic_policy_callback cb = NULL;
+ {
+ Mutex::Autolock _l(AudioSystem::gLock);
+ cb = gDynPolicyCallback;
+ }
+
+ if (cb != NULL) {
+ cb(DYNAMIC_POLICY_EVENT_MIX_STATE_UPDATE, regId, state);
+ }
}
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 8555983..76d9169 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -393,6 +393,7 @@
return BAD_VALUE;
}
mSampleRate = sampleRate;
+ mOriginalSampleRate = sampleRate;
mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
// Make copy of input parameter offloadInfo so that in the future:
@@ -470,6 +471,7 @@
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
+ mPreviousTimestampValid = false;
return NO_ERROR;
}
@@ -496,6 +498,8 @@
if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
// reset current position as seen by client to 0
mPosition = 0;
+ mPreviousTimestampValid = false;
+
// For offloaded tracks, we don't know if the hardware counters are really zero here,
// since the flush is asynchronous and stop may not fully drain.
// We save the time when the track is started to later verify whether
@@ -756,6 +760,15 @@
return mSampleRate;
}
+uint32_t AudioTrack::getOriginalSampleRate() const
+{
+ if (mIsTimed) {
+ return 0;
+ }
+
+ return mOriginalSampleRate;
+}
+
status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
AutoMutex lock(mLock);
@@ -995,6 +1008,7 @@
mNewPosition = mUpdatePeriod;
(void) updateAndGetPosition_l();
mPosition = 0;
+ mPreviousTimestampValid = false;
#if 0
// The documentation is not clear on the behavior of reload() and the restoration
// of loop count. Historically we have not restored loop count, start, end,
@@ -1063,7 +1077,7 @@
status_t status;
status = AudioSystem::getOutputForAttr(attr, &output,
- (audio_session_t)mSessionId, &streamType,
+ (audio_session_t)mSessionId, &streamType, mClientUid,
mSampleRate, mFormat, mChannelMask,
mFlags, mSelectedDeviceId, mOffloadInfo);
@@ -1102,6 +1116,7 @@
}
if (mSampleRate == 0) {
mSampleRate = afSampleRate;
+ mOriginalSampleRate = afSampleRate;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
@@ -2089,6 +2104,11 @@
status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
{
AutoMutex lock(mLock);
+
+ bool previousTimestampValid = mPreviousTimestampValid;
+ // Set false here to cover all the error return cases.
+ mPreviousTimestampValid = false;
+
// FIXME not implemented for fast tracks; should use proxy and SSQ
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
return INVALID_OPERATION;
@@ -2187,6 +2207,39 @@
// IAudioTrack. And timestamp.mPosition is initially in server's
// point of view, so we need to apply the same fudge factor to it.
}
+
+ // Prevent retrograde motion in timestamp.
+ // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
+ if (status == NO_ERROR) {
+ if (previousTimestampValid) {
+#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * 1000000000 + time.tv_nsec)
+ const uint64_t previousTimeNanos = TIME_TO_NANOS(mPreviousTimestamp.mTime);
+ const uint64_t currentTimeNanos = TIME_TO_NANOS(timestamp.mTime);
+#undef TIME_TO_NANOS
+ if (currentTimeNanos < previousTimeNanos) {
+ ALOGW("retrograde timestamp time");
+ // FIXME Consider blocking this from propagating upwards.
+ }
+
+ // Looking at signed delta will work even when the timestamps
+ // are wrapping around.
+ int32_t deltaPosition = static_cast<int32_t>(timestamp.mPosition
+ - mPreviousTimestamp.mPosition);
+ // position can bobble slightly as an artifact; this hides the bobble
+ static const int32_t MINIMUM_POSITION_DELTA = 8;
+ ALOGW_IF(deltaPosition < 0,
+ "retrograde timestamp position corrected, %d = %u - %u",
+ deltaPosition,
+ timestamp.mPosition,
+ mPreviousTimestamp.mPosition);
+ if (deltaPosition < MINIMUM_POSITION_DELTA) {
+ timestamp = mPreviousTimestamp; // Use last valid timestamp.
+ }
+ }
+ mPreviousTimestamp = timestamp;
+ mPreviousTimestampValid = true;
+ }
+
return status;
}
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 38055f9..d48532e 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -174,6 +174,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& opPackageName,
size_t *pFrameCount,
track_flags_t *flags,
pid_t tid,
@@ -190,6 +191,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ data.writeString16(opPackageName);
size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt64(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
@@ -702,6 +704,7 @@
int32_t priority,
audio_io_handle_t output,
int sessionId,
+ const String16& opPackageName,
status_t *status,
int *id,
int *enabled)
@@ -722,6 +725,7 @@
data.writeInt32(priority);
data.writeInt32((int32_t) output);
data.writeInt32(sessionId);
+ data.writeString16(opPackageName);
status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
if (lStatus != NO_ERROR) {
@@ -950,6 +954,7 @@
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
+ const String16& opPackageName = data.readString16();
size_t frameCount = data.readInt64();
track_flags_t flags = (track_flags_t) data.readInt32();
pid_t tid = (pid_t) data.readInt32();
@@ -959,9 +964,8 @@
sp<IMemory> buffers;
status_t status;
sp<IAudioRecord> record = openRecord(input,
- sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId,
- ¬ificationFrames,
- cblk, buffers, &status);
+ sampleRate, format, channelMask, opPackageName, &frameCount, &flags, tid,
+ &sessionId, ¬ificationFrames, cblk, buffers, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
reply->writeInt64(frameCount);
reply->writeInt32(flags);
@@ -1247,12 +1251,13 @@
int32_t priority = data.readInt32();
audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
int sessionId = data.readInt32();
+ const String16 opPackageName = data.readString16();
status_t status;
int id;
int enabled;
sp<IEffect> effect = createEffect(&desc, client, priority, output, sessionId,
- &status, &id, &enabled);
+ opPackageName, &status, &id, &enabled);
reply->writeInt32(status);
reply->writeInt32(id);
reply->writeInt32(enabled);
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index fc36a7f..fd18f17 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -171,6 +171,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -207,6 +208,7 @@
data.writeInt32(1);
data.writeInt32(*stream);
}
+ data.writeInt32(uid);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channelMask);
@@ -275,6 +277,7 @@
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -293,6 +296,7 @@
}
data.write(attr, sizeof(audio_attributes_t));
data.writeInt32(session);
+ data.writeInt32(uid);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channelMask);
@@ -852,6 +856,7 @@
if (hasStream) {
stream = (audio_stream_type_t)data.readInt32();
}
+ uid_t uid = (uid_t)data.readInt32();
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
@@ -865,7 +870,7 @@
}
audio_io_handle_t output;
status_t status = getOutputForAttr(hasAttributes ? &attr : NULL,
- &output, session, &stream,
+ &output, session, &stream, uid,
samplingRate, format, channelMask,
flags, selectedDeviceId, hasOffloadInfo ? &offloadInfo : NULL);
reply->writeInt32(status);
@@ -912,13 +917,14 @@
audio_attributes_t attr;
data.read(&attr, sizeof(audio_attributes_t));
audio_session_t session = (audio_session_t)data.readInt32();
+ uid_t uid = (uid_t)data.readInt32();
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
audio_input_flags_t flags = (audio_input_flags_t) data.readInt32();
audio_port_handle_t selectedDeviceId = (audio_port_handle_t) data.readInt32();
audio_io_handle_t input;
- status_t status = getInputForAttr(&attr, &input, session,
+ status_t status = getInputForAttr(&attr, &input, session, uid,
samplingRate, format, channelMask,
flags, selectedDeviceId);
reply->writeInt32(status);
diff --git a/media/libmedia/IHDCP.cpp b/media/libmedia/IHDCP.cpp
index 79944ee..f3a8902 100644
--- a/media/libmedia/IHDCP.cpp
+++ b/media/libmedia/IHDCP.cpp
@@ -284,11 +284,17 @@
size_t offset = data.readInt32();
size_t size = data.readInt32();
uint32_t streamCTR = data.readInt32();
- void *outData = malloc(size);
+ void *outData = NULL;
uint64_t inputCTR;
- status_t err = encryptNative(graphicBuffer, offset, size,
- streamCTR, &inputCTR, outData);
+ status_t err = ERROR_OUT_OF_RANGE;
+
+ outData = malloc(size);
+
+ if (outData != NULL) {
+ err = encryptNative(graphicBuffer, offset, size,
+ streamCTR, &inputCTR, outData);
+ }
reply->writeInt32(err);
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index 0091078..bde35f2 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -21,6 +21,9 @@
#include <binder/Parcel.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/AVSyncSettings.h>
+
#include <media/IDataSource.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayer.h>
@@ -41,7 +44,10 @@
START,
STOP,
IS_PLAYING,
- SET_PLAYBACK_RATE,
+ SET_PLAYBACK_SETTINGS,
+ GET_PLAYBACK_SETTINGS,
+ SET_SYNC_SETTINGS,
+ GET_SYNC_SETTINGS,
PAUSE,
SEEK_TO,
GET_CURRENT_POSITION,
@@ -175,15 +181,63 @@
return reply.readInt32();
}
- status_t setPlaybackRate(float rate)
+ status_t setPlaybackSettings(const AudioPlaybackRate& rate)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
- data.writeFloat(rate);
- remote()->transact(SET_PLAYBACK_RATE, data, &reply);
+ data.writeFloat(rate.mSpeed);
+ data.writeFloat(rate.mPitch);
+ data.writeInt32((int32_t)rate.mFallbackMode);
+ data.writeInt32((int32_t)rate.mStretchMode);
+ remote()->transact(SET_PLAYBACK_SETTINGS, data, &reply);
return reply.readInt32();
}
+ status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ remote()->transact(GET_PLAYBACK_SETTINGS, data, &reply);
+ status_t err = reply.readInt32();
+ if (err == OK) {
+ *rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ rate->mSpeed = reply.readFloat();
+ rate->mPitch = reply.readFloat();
+ rate->mFallbackMode = (AudioTimestretchFallbackMode)reply.readInt32();
+ rate->mStretchMode = (AudioTimestretchStretchMode)reply.readInt32();
+ }
+ return err;
+ }
+
+ status_t setSyncSettings(const AVSyncSettings& sync, float videoFpsHint)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ data.writeInt32((int32_t)sync.mSource);
+ data.writeInt32((int32_t)sync.mAudioAdjustMode);
+ data.writeFloat(sync.mTolerance);
+ data.writeFloat(videoFpsHint);
+ remote()->transact(SET_SYNC_SETTINGS, data, &reply);
+ return reply.readInt32();
+ }
+
+ status_t getSyncSettings(AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ remote()->transact(GET_SYNC_SETTINGS, data, &reply);
+ status_t err = reply.readInt32();
+ if (err == OK) {
+ AVSyncSettings settings;
+ settings.mSource = (AVSyncSource)reply.readInt32();
+ settings.mAudioAdjustMode = (AVSyncAudioAdjustMode)reply.readInt32();
+ settings.mTolerance = reply.readFloat();
+ *sync = settings;
+ *videoFps = reply.readFloat();
+ }
+ return err;
+ }
+
status_t pause()
{
Parcel data, reply;
@@ -453,9 +507,51 @@
reply->writeInt32(ret);
return NO_ERROR;
} break;
- case SET_PLAYBACK_RATE: {
+ case SET_PLAYBACK_SETTINGS: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
- reply->writeInt32(setPlaybackRate(data.readFloat()));
+ AudioPlaybackRate rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ rate.mSpeed = data.readFloat();
+ rate.mPitch = data.readFloat();
+ rate.mFallbackMode = (AudioTimestretchFallbackMode)data.readInt32();
+ rate.mStretchMode = (AudioTimestretchStretchMode)data.readInt32();
+ reply->writeInt32(setPlaybackSettings(rate));
+ return NO_ERROR;
+ } break;
+ case GET_PLAYBACK_SETTINGS: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+ AudioPlaybackRate rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ status_t err = getPlaybackSettings(&rate);
+ reply->writeInt32(err);
+ if (err == OK) {
+ reply->writeFloat(rate.mSpeed);
+ reply->writeFloat(rate.mPitch);
+ reply->writeInt32((int32_t)rate.mFallbackMode);
+ reply->writeInt32((int32_t)rate.mStretchMode);
+ }
+ return NO_ERROR;
+ } break;
+ case SET_SYNC_SETTINGS: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+ AVSyncSettings sync;
+ sync.mSource = (AVSyncSource)data.readInt32();
+ sync.mAudioAdjustMode = (AVSyncAudioAdjustMode)data.readInt32();
+ sync.mTolerance = data.readFloat();
+ float videoFpsHint = data.readFloat();
+ reply->writeInt32(setSyncSettings(sync, videoFpsHint));
+ return NO_ERROR;
+ } break;
+ case GET_SYNC_SETTINGS: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+ AVSyncSettings sync;
+ float videoFps;
+ status_t err = getSyncSettings(&sync, &videoFps);
+ reply->writeInt32(err);
+ if (err == OK) {
+ reply->writeInt32((int32_t)sync.mSource);
+ reply->writeInt32((int32_t)sync.mAudioAdjustMode);
+ reply->writeFloat(sync.mTolerance);
+ reply->writeFloat(videoFps);
+ }
return NO_ERROR;
} break;
case PAUSE: {
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index aa7b2e1..05f8670 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -78,10 +78,11 @@
return interface_cast<IMediaPlayer>(reply.readStrongBinder());
}
- virtual sp<IMediaRecorder> createMediaRecorder()
+ virtual sp<IMediaRecorder> createMediaRecorder(const String16 &opPackageName)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
+ data.writeString16(opPackageName);
remote()->transact(CREATE_MEDIA_RECORDER, data, &reply);
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
@@ -128,11 +129,12 @@
return remote()->transact(PULL_BATTERY_DATA, data, reply);
}
- virtual sp<IRemoteDisplay> listenForRemoteDisplay(const sp<IRemoteDisplayClient>& client,
- const String8& iface)
+ virtual sp<IRemoteDisplay> listenForRemoteDisplay(const String16 &opPackageName,
+ const sp<IRemoteDisplayClient>& client, const String8& iface)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
+ data.writeString16(opPackageName);
data.writeStrongBinder(IInterface::asBinder(client));
data.writeString8(iface);
remote()->transact(LISTEN_FOR_REMOTE_DISPLAY, data, &reply);
@@ -166,7 +168,8 @@
} break;
case CREATE_MEDIA_RECORDER: {
CHECK_INTERFACE(IMediaPlayerService, data, reply);
- sp<IMediaRecorder> recorder = createMediaRecorder();
+ const String16 opPackageName = data.readString16();
+ sp<IMediaRecorder> recorder = createMediaRecorder(opPackageName);
reply->writeStrongBinder(IInterface::asBinder(recorder));
return NO_ERROR;
} break;
@@ -214,10 +217,11 @@
} break;
case LISTEN_FOR_REMOTE_DISPLAY: {
CHECK_INTERFACE(IMediaPlayerService, data, reply);
+ const String16 opPackageName = data.readString16();
sp<IRemoteDisplayClient> client(
interface_cast<IRemoteDisplayClient>(data.readStrongBinder()));
String8 iface(data.readString8());
- sp<IRemoteDisplay> display(listenForRemoteDisplay(client, iface));
+ sp<IRemoteDisplay> display(listenForRemoteDisplay(opPackageName, client, iface));
reply->writeStrongBinder(IInterface::asBinder(display));
return NO_ERROR;
} break;
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index 9d69b6a..dc46038 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -34,11 +34,12 @@
// ---------------------------------------------------------------------------
-Visualizer::Visualizer (int32_t priority,
+Visualizer::Visualizer (const String16& opPackageName,
+ int32_t priority,
effect_callback_t cbf,
void* user,
int sessionId)
- : AudioEffect(SL_IID_VISUALIZATION, NULL, priority, cbf, user, sessionId),
+ : AudioEffect(SL_IID_VISUALIZATION, opPackageName, NULL, priority, cbf, user, sessionId),
mCaptureRate(CAPTURE_RATE_DEF),
mCaptureSize(CAPTURE_SIZE_DEF),
mSampleRate(44100000),
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 9a276ae..81a5e8c 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -32,7 +32,9 @@
#include <gui/Surface.h>
#include <media/mediaplayer.h>
+#include <media/AudioResamplerPublic.h>
#include <media/AudioSystem.h>
+#include <media/AVSyncSettings.h>
#include <media/IDataSource.h>
#include <binder/MemoryBase.h>
@@ -60,7 +62,6 @@
mLoop = false;
mLeftVolume = mRightVolume = 1.0;
mVideoWidth = mVideoHeight = 0;
- mPlaybackRate = 1.0;
mLockThreadId = 0;
mAudioSessionId = AudioSystem::newAudioUniqueId();
AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);
@@ -389,6 +390,9 @@
if ((mCurrentState & MEDIA_PLAYER_STARTED) && ! temp) {
ALOGE("internal/external state mismatch corrected");
mCurrentState = MEDIA_PLAYER_PAUSED;
+ } else if ((mCurrentState & MEDIA_PLAYER_PAUSED) && temp) {
+ ALOGE("internal/external state mismatch corrected");
+ mCurrentState = MEDIA_PLAYER_STARTED;
}
return temp;
}
@@ -396,22 +400,50 @@
return false;
}
-status_t MediaPlayer::setPlaybackRate(float rate)
+status_t MediaPlayer::setPlaybackSettings(const AudioPlaybackRate& rate)
{
- ALOGV("setPlaybackRate: %f", rate);
- if (rate <= 0.0) {
+ ALOGV("setPlaybackSettings: %f %f %d %d",
+ rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
+ // Negative speed and pitch does not make sense. Further validation will
+ // be done by the respective mediaplayers.
+ if (rate.mSpeed < 0.f || rate.mPitch < 0.f) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
- if (mPlayer != 0) {
- if (mPlaybackRate == rate) {
- return NO_ERROR;
+ if (mPlayer == 0) return INVALID_OPERATION;
+ status_t err = mPlayer->setPlaybackSettings(rate);
+ if (err == OK) {
+ if (rate.mSpeed == 0.f && mCurrentState == MEDIA_PLAYER_STARTED) {
+ mCurrentState = MEDIA_PLAYER_PAUSED;
+ } else if (rate.mSpeed != 0.f && mCurrentState == MEDIA_PLAYER_PAUSED) {
+ mCurrentState = MEDIA_PLAYER_STARTED;
}
- mPlaybackRate = rate;
- return mPlayer->setPlaybackRate(rate);
}
- ALOGV("setPlaybackRate: no active player");
- return INVALID_OPERATION;
+ return err;
+}
+
+status_t MediaPlayer::getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */)
+{
+ Mutex::Autolock _l(mLock);
+ if (mPlayer == 0) return INVALID_OPERATION;
+ return mPlayer->getPlaybackSettings(rate);
+}
+
+status_t MediaPlayer::setSyncSettings(const AVSyncSettings& sync, float videoFpsHint)
+{
+ ALOGV("setSyncSettings: %u %u %f %f",
+ sync.mSource, sync.mAudioAdjustMode, sync.mTolerance, videoFpsHint);
+ Mutex::Autolock _l(mLock);
+ if (mPlayer == 0) return INVALID_OPERATION;
+ return mPlayer->setSyncSettings(sync, videoFpsHint);
+}
+
+status_t MediaPlayer::getSyncSettings(
+ AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */)
+{
+ Mutex::Autolock _l(mLock);
+ if (mPlayer == 0) return INVALID_OPERATION;
+ return mPlayer->getSyncSettings(sync, videoFps);
}
status_t MediaPlayer::getVideoWidth(int *w)
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index a2d6e53..9470936 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -594,13 +594,13 @@
return INVALID_OPERATION;
}
-MediaRecorder::MediaRecorder() : mSurfaceMediaSource(NULL)
+MediaRecorder::MediaRecorder(const String16& opPackageName) : mSurfaceMediaSource(NULL)
{
ALOGV("constructor");
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != NULL) {
- mMediaRecorder = service->createMediaRecorder();
+ mMediaRecorder = service->createMediaRecorder(opPackageName);
}
if (mMediaRecorder != NULL) {
mCurrentState = MEDIA_RECORDER_IDLE;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 3bc763f..891a9e9 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -307,10 +307,10 @@
ALOGV("MediaPlayerService destroyed");
}
-sp<IMediaRecorder> MediaPlayerService::createMediaRecorder()
+sp<IMediaRecorder> MediaPlayerService::createMediaRecorder(const String16 &opPackageName)
{
pid_t pid = IPCThreadState::self()->getCallingPid();
- sp<MediaRecorderClient> recorder = new MediaRecorderClient(this, pid);
+ sp<MediaRecorderClient> recorder = new MediaRecorderClient(this, pid, opPackageName);
wp<MediaRecorderClient> w = recorder;
Mutex::Autolock lock(mLock);
mMediaRecorderClients.add(w);
@@ -381,12 +381,13 @@
}
sp<IRemoteDisplay> MediaPlayerService::listenForRemoteDisplay(
+ const String16 &opPackageName,
const sp<IRemoteDisplayClient>& client, const String8& iface) {
if (!checkPermission("android.permission.CONTROL_WIFI_DISPLAY")) {
return NULL;
}
- return new RemoteDisplay(client, iface.string());
+ return new RemoteDisplay(opPackageName, client, iface.string());
}
status_t MediaPlayerService::AudioOutput::dump(int fd, const Vector<String16>& args) const
@@ -980,12 +981,52 @@
return NO_ERROR;
}
-status_t MediaPlayerService::Client::setPlaybackRate(float rate)
+status_t MediaPlayerService::Client::setPlaybackSettings(const AudioPlaybackRate& rate)
{
- ALOGV("[%d] setPlaybackRate(%f)", mConnId, rate);
+ ALOGV("[%d] setPlaybackSettings(%f, %f, %d, %d)",
+ mConnId, rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
sp<MediaPlayerBase> p = getPlayer();
if (p == 0) return UNKNOWN_ERROR;
- return p->setPlaybackRate(rate);
+ return p->setPlaybackSettings(rate);
+}
+
+status_t MediaPlayerService::Client::getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */)
+{
+ sp<MediaPlayerBase> p = getPlayer();
+ if (p == 0) return UNKNOWN_ERROR;
+ status_t ret = p->getPlaybackSettings(rate);
+ if (ret == NO_ERROR) {
+ ALOGV("[%d] getPlaybackSettings(%f, %f, %d, %d)",
+ mConnId, rate->mSpeed, rate->mPitch, rate->mFallbackMode, rate->mStretchMode);
+ } else {
+ ALOGV("[%d] getPlaybackSettings returned %d", mConnId, ret);
+ }
+ return ret;
+}
+
+status_t MediaPlayerService::Client::setSyncSettings(
+ const AVSyncSettings& sync, float videoFpsHint)
+{
+ ALOGV("[%d] setSyncSettings(%u, %u, %f, %f)",
+ mConnId, sync.mSource, sync.mAudioAdjustMode, sync.mTolerance, videoFpsHint);
+ sp<MediaPlayerBase> p = getPlayer();
+ if (p == 0) return UNKNOWN_ERROR;
+ return p->setSyncSettings(sync, videoFpsHint);
+}
+
+status_t MediaPlayerService::Client::getSyncSettings(
+ AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */)
+{
+ sp<MediaPlayerBase> p = getPlayer();
+ if (p == 0) return UNKNOWN_ERROR;
+ status_t ret = p->getSyncSettings(sync, videoFps);
+ if (ret == NO_ERROR) {
+ ALOGV("[%d] getSyncSettings(%u, %u, %f, %f)",
+ mConnId, sync->mSource, sync->mAudioAdjustMode, sync->mTolerance, *videoFps);
+ } else {
+ ALOGV("[%d] getSyncSettings returned %d", mConnId, ret);
+ }
+ return ret;
}
status_t MediaPlayerService::Client::getCurrentPosition(int *msec)
@@ -1310,7 +1351,7 @@
mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
- mPlaybackRatePermille = 1000;
+ mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
mSampleRateHz = 0;
mMsecsPerFrame = 0;
mAuxEffectId = 0;
@@ -1633,7 +1674,7 @@
mSampleRateHz = sampleRate;
mFlags = flags;
- mMsecsPerFrame = mPlaybackRatePermille / (float) sampleRate;
+ mMsecsPerFrame = 1E3f / (mPlaybackRate.mSpeed * sampleRate);
uint32_t pos;
if (t->getPosition(&pos) == OK) {
mBytesWritten = uint64_t(pos) * t->frameSize();
@@ -1642,7 +1683,7 @@
status_t res = NO_ERROR;
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) {
- res = t->setSampleRate(mPlaybackRatePermille * mSampleRateHz / 1000);
+ res = t->setPlaybackRate(mPlaybackRate);
if (res == NO_ERROR) {
t->setAuxEffectSendLevel(mSendLevel);
res = t->attachAuxEffect(mAuxEffectId);
@@ -1738,22 +1779,38 @@
}
}
-status_t MediaPlayerService::AudioOutput::setPlaybackRatePermille(int32_t ratePermille)
+status_t MediaPlayerService::AudioOutput::setPlaybackRate(const AudioPlaybackRate &rate)
{
- ALOGV("setPlaybackRatePermille(%d)", ratePermille);
- status_t res = NO_ERROR;
- if (mTrack != 0) {
- res = mTrack->setSampleRate(ratePermille * mSampleRateHz / 1000);
- } else {
- res = NO_INIT;
+ ALOGV("setPlaybackRate(%f %f %d %d)",
+ rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
+ if (mTrack == 0) {
+ // remember rate so that we can set it when the track is opened
+ mPlaybackRate = rate;
+ return OK;
}
- mPlaybackRatePermille = ratePermille;
+ status_t res = mTrack->setPlaybackRate(rate);
+ if (res != NO_ERROR) {
+ return res;
+ }
+ // rate.mSpeed is always greater than 0 if setPlaybackRate succeeded
+ CHECK_GT(rate.mSpeed, 0.f);
+ mPlaybackRate = rate;
if (mSampleRateHz != 0) {
- mMsecsPerFrame = mPlaybackRatePermille / (float) mSampleRateHz;
+ mMsecsPerFrame = 1E3f / (rate.mSpeed * mSampleRateHz);
}
return res;
}
+status_t MediaPlayerService::AudioOutput::getPlaybackRate(AudioPlaybackRate *rate)
+{
+ ALOGV("setPlaybackRate");
+ if (mTrack == 0) {
+ return NO_INIT;
+ }
+ *rate = mTrack->getPlaybackRate();
+ return NO_ERROR;
+}
+
status_t MediaPlayerService::AudioOutput::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 6ddfe14..5103841 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -34,7 +34,9 @@
namespace android {
+struct AudioPlaybackRate;
class AudioTrack;
+struct AVSyncSettings;
class IDataSource;
class IMediaRecorder;
class IMediaMetadataRetriever;
@@ -109,7 +111,9 @@
void setAudioAttributes(const audio_attributes_t * attributes);
void setVolume(float left, float right);
- virtual status_t setPlaybackRatePermille(int32_t ratePermille);
+ virtual status_t setPlaybackRate(const AudioPlaybackRate& rate);
+ virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */);
+
status_t setAuxEffectSendLevel(float level);
status_t attachAuxEffect(int effectId);
virtual status_t dump(int fd, const Vector<String16>& args) const;
@@ -139,7 +143,7 @@
const audio_attributes_t *mAttributes;
float mLeftVolume;
float mRightVolume;
- int32_t mPlaybackRatePermille;
+ AudioPlaybackRate mPlaybackRate;
uint32_t mSampleRateHz; // sample rate of the content, as set in open()
float mMsecsPerFrame;
int mSessionId;
@@ -188,7 +192,7 @@
static void instantiate();
// IMediaPlayerService interface
- virtual sp<IMediaRecorder> createMediaRecorder();
+ virtual sp<IMediaRecorder> createMediaRecorder(const String16 &opPackageName);
void removeMediaRecorderClient(wp<MediaRecorderClient> client);
virtual sp<IMediaMetadataRetriever> createMetadataRetriever();
@@ -200,8 +204,8 @@
virtual sp<IDrm> makeDrm();
virtual sp<IHDCP> makeHDCP(bool createEncryptionModule);
- virtual sp<IRemoteDisplay> listenForRemoteDisplay(const sp<IRemoteDisplayClient>& client,
- const String8& iface);
+ virtual sp<IRemoteDisplay> listenForRemoteDisplay(const String16 &opPackageName,
+ const sp<IRemoteDisplayClient>& client, const String8& iface);
virtual status_t dump(int fd, const Vector<String16>& args);
void removeClient(wp<Client> client);
@@ -262,7 +266,11 @@
virtual status_t stop();
virtual status_t pause();
virtual status_t isPlaying(bool* state);
- virtual status_t setPlaybackRate(float rate);
+ virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate);
+ virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */);
+ virtual status_t setSyncSettings(const AVSyncSettings& rate, float videoFpsHint);
+ virtual status_t getSyncSettings(AVSyncSettings* rate /* nonnull */,
+ float* videoFps /* nonnull */);
virtual status_t seekTo(int msec);
virtual status_t getCurrentPosition(int* msec);
virtual status_t getDuration(int* msec);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index 319ebb0..40e9d1c 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -290,11 +290,12 @@
return NO_ERROR;
}
-MediaRecorderClient::MediaRecorderClient(const sp<MediaPlayerService>& service, pid_t pid)
+MediaRecorderClient::MediaRecorderClient(const sp<MediaPlayerService>& service, pid_t pid,
+ const String16& opPackageName)
{
ALOGV("Client constructor");
mPid = pid;
- mRecorder = new StagefrightRecorder;
+ mRecorder = new StagefrightRecorder(opPackageName);
mMediaPlayerService = service;
}
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index b45344b..e03ec3f 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -62,7 +62,8 @@
MediaRecorderClient(
const sp<MediaPlayerService>& service,
- pid_t pid);
+ pid_t pid,
+ const String16& opPackageName);
virtual ~MediaRecorderClient();
pid_t mPid;
diff --git a/media/libmediaplayerservice/RemoteDisplay.cpp b/media/libmediaplayerservice/RemoteDisplay.cpp
index eb959b4..0eb4b5d 100644
--- a/media/libmediaplayerservice/RemoteDisplay.cpp
+++ b/media/libmediaplayerservice/RemoteDisplay.cpp
@@ -26,13 +26,14 @@
namespace android {
RemoteDisplay::RemoteDisplay(
+ const String16 &opPackageName,
const sp<IRemoteDisplayClient> &client,
const char *iface)
: mLooper(new ALooper),
mNetSession(new ANetworkSession) {
mLooper->setName("wfd_looper");
- mSource = new WifiDisplaySource(mNetSession, client);
+ mSource = new WifiDisplaySource(opPackageName, mNetSession, client);
mLooper->registerHandler(mSource);
mNetSession->start();
diff --git a/media/libmediaplayerservice/RemoteDisplay.h b/media/libmediaplayerservice/RemoteDisplay.h
index 1a48981..d4573e9 100644
--- a/media/libmediaplayerservice/RemoteDisplay.h
+++ b/media/libmediaplayerservice/RemoteDisplay.h
@@ -33,6 +33,7 @@
struct RemoteDisplay : public BnRemoteDisplay {
RemoteDisplay(
+ const String16 &opPackageName,
const sp<IRemoteDisplayClient> &client,
const char *iface);
diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index b37aee3..3fedd9b 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -188,6 +188,14 @@
return mPlayer->getParameter(key, reply);
}
+status_t StagefrightPlayer::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ return mPlayer->setPlaybackSettings(rate);
+}
+
+status_t StagefrightPlayer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
+ return mPlayer->getPlaybackSettings(rate);
+}
+
status_t StagefrightPlayer::getMetadata(
const media::Metadata::Filter& /* ids */, Parcel *records) {
using media::Metadata;
diff --git a/media/libmediaplayerservice/StagefrightPlayer.h b/media/libmediaplayerservice/StagefrightPlayer.h
index e6c30ff..96013df 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.h
+++ b/media/libmediaplayerservice/StagefrightPlayer.h
@@ -60,6 +60,8 @@
virtual void setAudioSink(const sp<AudioSink> &audioSink);
virtual status_t setParameter(int key, const Parcel &request);
virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t setPlaybackSettings(const AudioPlaybackRate &rate);
+ virtual status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
virtual status_t getMetadata(
const media::Metadata::Filter& ids, Parcel *records);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 8a0b060d..aa19a25 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -69,8 +69,9 @@
}
-StagefrightRecorder::StagefrightRecorder()
- : mWriter(NULL),
+StagefrightRecorder::StagefrightRecorder(const String16 &opPackageName)
+ : MediaRecorderBase(opPackageName),
+ mWriter(NULL),
mOutputFd(-1),
mAudioSource(AUDIO_SOURCE_CNT),
mVideoSource(VIDEO_SOURCE_LIST_END),
@@ -905,6 +906,7 @@
sp<AudioSource> audioSource =
new AudioSource(
mAudioSource,
+ mOpPackageName,
mSampleRate,
mAudioChannels);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 8fa5bfa..1425f59 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -40,7 +40,7 @@
struct ALooper;
struct StagefrightRecorder : public MediaRecorderBase {
- StagefrightRecorder();
+ StagefrightRecorder(const String16 &opPackageName);
virtual ~StagefrightRecorder();
virtual status_t init();
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index 7eaa0e0..5e7b644 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -1065,6 +1065,7 @@
const char *mime;
CHECK(meta->findCString(kKeyMIMEType, &mime));
+ format->setString("mime", mime);
int32_t trackType;
if (!strncasecmp(mime, "video/", 6)) {
@@ -1085,8 +1086,6 @@
format->setString("language", lang);
if (trackType == MEDIA_TRACK_TYPE_SUBTITLE) {
- format->setString("mime", mime);
-
int32_t isAutoselect = 1, isDefault = 0, isForced = 0;
meta->findInt32(kKeyTrackIsAutoselect, &isAutoselect);
meta->findInt32(kKeyTrackIsDefault, &isDefault);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index b670d68..9963353 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -37,6 +37,9 @@
#include <cutils/properties.h>
+#include <media/AudioResamplerPublic.h>
+#include <media/AVSyncSettings.h>
+
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -45,7 +48,9 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
+
#include <gui/IGraphicBufferProducer.h>
+#include <gui/Surface.h>
#include "avc_utils.h"
@@ -96,16 +101,16 @@
};
struct NuPlayer::SetSurfaceAction : public Action {
- SetSurfaceAction(const sp<NativeWindowWrapper> &wrapper)
- : mWrapper(wrapper) {
+ SetSurfaceAction(const sp<Surface> &surface)
+ : mSurface(surface) {
}
virtual void execute(NuPlayer *player) {
- player->performSetSurface(mWrapper);
+ player->performSetSurface(mSurface);
}
private:
- sp<NativeWindowWrapper> mWrapper;
+ sp<Surface> mSurface;
DISALLOW_EVIL_CONSTRUCTORS(SetSurfaceAction);
};
@@ -180,7 +185,8 @@
mFlushingVideo(NONE),
mResumePending(false),
mVideoScalingMode(NATIVE_WINDOW_SCALING_MODE_SCALE_TO_WINDOW),
- mPlaybackRate(1.0),
+ mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT),
+ mVideoFpsHint(-1.f),
mStarted(false),
mPaused(false),
mPausedByClient(false) {
@@ -307,15 +313,12 @@
void NuPlayer::setVideoSurfaceTextureAsync(
const sp<IGraphicBufferProducer> &bufferProducer) {
- sp<AMessage> msg = new AMessage(kWhatSetVideoNativeWindow, this);
+ sp<AMessage> msg = new AMessage(kWhatSetVideoSurface, this);
if (bufferProducer == NULL) {
- msg->setObject("native-window", NULL);
+ msg->setObject("surface", NULL);
} else {
- msg->setObject(
- "native-window",
- new NativeWindowWrapper(
- new Surface(bufferProducer, true /* controlledByApp */)));
+ msg->setObject("surface", new Surface(bufferProducer, true /* controlledByApp */));
}
msg->post();
@@ -331,10 +334,61 @@
(new AMessage(kWhatStart, this))->post();
}
-void NuPlayer::setPlaybackRate(float rate) {
- sp<AMessage> msg = new AMessage(kWhatSetRate, this);
- msg->setFloat("rate", rate);
- msg->post();
+status_t NuPlayer::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ // do some cursory validation of the settings here. audio modes are
+ // only validated when set on the audiosink.
+ if ((rate.mSpeed != 0.f && rate.mSpeed < AUDIO_TIMESTRETCH_SPEED_MIN)
+ || rate.mSpeed > AUDIO_TIMESTRETCH_SPEED_MAX
+ || rate.mPitch < AUDIO_TIMESTRETCH_SPEED_MIN
+ || rate.mPitch > AUDIO_TIMESTRETCH_SPEED_MAX) {
+ return BAD_VALUE;
+ }
+ sp<AMessage> msg = new AMessage(kWhatConfigPlayback, this);
+ writeToAMessage(msg, rate);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ }
+ return err;
+}
+
+status_t NuPlayer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
+ sp<AMessage> msg = new AMessage(kWhatGetPlaybackSettings, this);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ if (err == OK) {
+ readFromAMessage(response, rate);
+ }
+ }
+ return err;
+}
+
+status_t NuPlayer::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
+ sp<AMessage> msg = new AMessage(kWhatConfigSync, this);
+ writeToAMessage(msg, sync, videoFpsHint);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ }
+ return err;
+}
+
+status_t NuPlayer::getSyncSettings(
+ AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */) {
+ sp<AMessage> msg = new AMessage(kWhatGetSyncSettings, this);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ if (err == OK) {
+ readFromAMessage(response, sync, videoFps);
+ }
+ }
+ return err;
}
void NuPlayer::pause() {
@@ -368,23 +422,23 @@
int32_t trackType;
CHECK(format->findInt32("type", &trackType));
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
AString lang;
CHECK(format->findString("language", &lang));
reply->writeInt32(2); // write something non-zero
reply->writeInt32(trackType);
+ reply->writeString16(String16(mime.c_str()));
reply->writeString16(String16(lang.c_str()));
if (trackType == MEDIA_TRACK_TYPE_SUBTITLE) {
- AString mime;
- CHECK(format->findString("mime", &mime));
-
int32_t isAuto, isDefault, isForced;
CHECK(format->findInt32("auto", &isAuto));
CHECK(format->findInt32("default", &isDefault));
CHECK(format->findInt32("forced", &isForced));
- reply->writeString16(String16(mime.c_str()));
reply->writeInt32(isAuto);
reply->writeInt32(isDefault);
reply->writeInt32(isForced);
@@ -555,15 +609,15 @@
break;
}
- case kWhatSetVideoNativeWindow:
+ case kWhatSetVideoSurface:
{
- ALOGV("kWhatSetVideoNativeWindow");
+ ALOGV("kWhatSetVideoSurface");
sp<RefBase> obj;
- CHECK(msg->findObject("native-window", &obj));
-
+ CHECK(msg->findObject("surface", &obj));
+ sp<Surface> surface = static_cast<Surface *>(obj.get());
if (mSource == NULL || mSource->getFormat(false /* audio */) == NULL) {
- performSetSurface(static_cast<NativeWindowWrapper *>(obj.get()));
+ performSetSurface(surface);
break;
}
@@ -571,9 +625,7 @@
new FlushDecoderAction(FLUSH_CMD_FLUSH /* audio */,
FLUSH_CMD_SHUTDOWN /* video */));
- mDeferredActions.push_back(
- new SetSurfaceAction(
- static_cast<NativeWindowWrapper *>(obj.get())));
+ mDeferredActions.push_back(new SetSurfaceAction(surface));
if (obj != NULL) {
if (mStarted) {
@@ -627,12 +679,28 @@
break;
}
- case kWhatSetRate:
+ case kWhatConfigPlayback:
{
- ALOGV("kWhatSetRate");
- CHECK(msg->findFloat("rate", &mPlaybackRate));
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AudioPlaybackRate rate /* sanitized */;
+ readFromAMessage(msg, &rate);
+ status_t err = OK;
if (mRenderer != NULL) {
- mRenderer->setPlaybackRate(mPlaybackRate);
+ err = mRenderer->setPlaybackSettings(rate);
+ }
+ if (err == OK) {
+ if (rate.mSpeed == 0.f) {
+ onPause();
+ // save all other settings (using non-paused speed)
+ // so we can restore them on start
+ AudioPlaybackRate newRate = rate;
+ newRate.mSpeed = mPlaybackSettings.mSpeed;
+ mPlaybackSettings = newRate;
+ } else { /* rate.mSpeed != 0.f */
+ onResume();
+ mPlaybackSettings = rate;
+ }
}
if (mVideoDecoder != NULL) {
@@ -640,11 +708,86 @@
int32_t rate;
if (meta != NULL && meta->findInt32(kKeyFrameRate, &rate) && rate > 0) {
sp<AMessage> params = new AMessage();
- params->setFloat("operating-rate", rate * mPlaybackRate);
+ params->setFloat("operating-rate", rate * mPlaybackSettings.mSpeed);
mVideoDecoder->setParameters(params);
}
}
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetPlaybackSettings:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AudioPlaybackRate rate = mPlaybackSettings;
+ status_t err = OK;
+ if (mRenderer != NULL) {
+ err = mRenderer->getPlaybackSettings(&rate);
+ }
+ if (err == OK) {
+ // get playback settings used by renderer, as it may be
+ // slightly off due to audiosink not taking small changes.
+ mPlaybackSettings = rate;
+ if (mPaused) {
+ rate.mSpeed = 0.f;
+ }
+ }
+ sp<AMessage> response = new AMessage;
+ if (err == OK) {
+ writeToAMessage(response, rate);
+ }
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatConfigSync:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ ALOGV("kWhatConfigSync");
+ AVSyncSettings sync;
+ float videoFpsHint;
+ readFromAMessage(msg, &sync, &videoFpsHint);
+ status_t err = OK;
+ if (mRenderer != NULL) {
+ err = mRenderer->setSyncSettings(sync, videoFpsHint);
+ }
+ if (err == OK) {
+ mSyncSettings = sync;
+ mVideoFpsHint = videoFpsHint;
+ }
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetSyncSettings:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AVSyncSettings sync = mSyncSettings;
+ float videoFps = mVideoFpsHint;
+ status_t err = OK;
+ if (mRenderer != NULL) {
+ err = mRenderer->getSyncSettings(&sync, &videoFps);
+ if (err == OK) {
+ mSyncSettings = sync;
+ mVideoFpsHint = videoFps;
+ }
+ }
+ sp<AMessage> response = new AMessage;
+ if (err == OK) {
+ writeToAMessage(response, sync, videoFps);
+ }
+ response->setInt32("err", err);
+ response->postReply(replyID);
break;
}
@@ -667,7 +810,7 @@
// initialize video before audio because successful initialization of
// video may change deep buffer mode of audio.
- if (mNativeWindow != NULL) {
+ if (mSurface != NULL) {
instantiateDecoder(false, &mVideoDecoder);
}
@@ -715,7 +858,7 @@
}
if ((mAudioDecoder == NULL && mAudioSink != NULL)
- || (mVideoDecoder == NULL && mNativeWindow != NULL)) {
+ || (mVideoDecoder == NULL && mSurface != NULL)) {
msg->post(100000ll);
mScanSourcesPending = true;
}
@@ -1061,7 +1204,7 @@
// TRICKY: We rely on mRenderer being null, so that decoder does not start requesting
// data on instantiation.
- if (mNativeWindow != NULL) {
+ if (mSurface != NULL) {
err = instantiateDecoder(false, &mVideoDecoder);
if (err != OK) {
return err;
@@ -1114,8 +1257,12 @@
mRendererLooper->setName("NuPlayerRenderer");
mRendererLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
mRendererLooper->registerHandler(mRenderer);
- if (mPlaybackRate != 1.0) {
- mRenderer->setPlaybackRate(mPlaybackRate);
+
+ status_t err = mRenderer->setPlaybackSettings(mPlaybackSettings);
+ if (err != OK) {
+ mSource->stop();
+ notifyListener(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
+ return;
}
sp<MetaData> meta = getFileMeta();
@@ -1282,7 +1429,7 @@
sp<MetaData> meta = getFileMeta();
int32_t rate;
if (meta != NULL && meta->findInt32(kKeyFrameRate, &rate) && rate > 0) {
- format->setFloat("operating-rate", rate * mPlaybackRate);
+ format->setFloat("operating-rate", rate * mPlaybackSettings.mSpeed);
}
}
@@ -1304,10 +1451,10 @@
notify->setInt32("generation", mVideoDecoderGeneration);
*decoder = new Decoder(
- notify, mSource, mRenderer, mNativeWindow, mCCDecoder);
+ notify, mSource, mRenderer, mSurface, mCCDecoder);
// enable FRC if high-quality AV sync is requested, even if not
- // queuing to native window, as this will even improve textureview
+ // directly queuing to display, as this will even improve textureview
// playback.
{
char value[PROPERTY_VALUE_MAX];
@@ -1479,9 +1626,8 @@
status_t NuPlayer::setVideoScalingMode(int32_t mode) {
mVideoScalingMode = mode;
- if (mNativeWindow != NULL) {
- status_t ret = native_window_set_scaling_mode(
- mNativeWindow->getNativeWindow().get(), mVideoScalingMode);
+ if (mSurface != NULL) {
+ status_t ret = native_window_set_scaling_mode(mSurface.get(), mVideoScalingMode);
if (ret != OK) {
ALOGE("Failed to set scaling mode (%d): %s",
-ret, strerror(-ret));
@@ -1675,10 +1821,10 @@
}
}
-void NuPlayer::performSetSurface(const sp<NativeWindowWrapper> &wrapper) {
+void NuPlayer::performSetSurface(const sp<Surface> &surface) {
ALOGV("performSetSurface");
- mNativeWindow = wrapper;
+ mSurface = surface;
// XXX - ignore error from setVideoScalingMode for now
setVideoScalingMode(mVideoScalingMode);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 14bdb01..fcf6841 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -18,14 +18,16 @@
#define NU_PLAYER_H_
+#include <media/AudioResamplerPublic.h>
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/foundation/AHandler.h>
-#include <media/stagefright/NativeWindowWrapper.h>
namespace android {
struct ABuffer;
struct AMessage;
+struct AudioPlaybackRate;
+struct AVSyncSettings;
class IDataSource;
class MetaData;
struct NuPlayerDriver;
@@ -54,7 +56,11 @@
const sp<IGraphicBufferProducer> &bufferProducer);
void setAudioSink(const sp<MediaPlayerBase::AudioSink> &sink);
- void setPlaybackRate(float rate);
+ status_t setPlaybackSettings(const AudioPlaybackRate &rate);
+ status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
+ status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint);
+ status_t getSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
+
void start();
void pause();
@@ -105,10 +111,13 @@
enum {
kWhatSetDataSource = '=DaS',
kWhatPrepare = 'prep',
- kWhatSetVideoNativeWindow = '=NaW',
+ kWhatSetVideoSurface = '=VSu',
kWhatSetAudioSink = '=AuS',
kWhatMoreDataQueued = 'more',
- kWhatSetRate = 'setR',
+ kWhatConfigPlayback = 'cfPB',
+ kWhatConfigSync = 'cfSy',
+ kWhatGetPlaybackSettings = 'gPbS',
+ kWhatGetSyncSettings = 'gSyS',
kWhatStart = 'strt',
kWhatScanSources = 'scan',
kWhatVideoNotify = 'vidN',
@@ -131,7 +140,7 @@
uid_t mUID;
sp<Source> mSource;
uint32_t mSourceFlags;
- sp<NativeWindowWrapper> mNativeWindow;
+ sp<Surface> mSurface;
sp<MediaPlayerBase::AudioSink> mAudioSink;
sp<DecoderBase> mVideoDecoder;
bool mOffloadAudio;
@@ -180,7 +189,9 @@
int32_t mVideoScalingMode;
- float mPlaybackRate;
+ AudioPlaybackRate mPlaybackSettings;
+ AVSyncSettings mSyncSettings;
+ float mVideoFpsHint;
bool mStarted;
// Actual pause state, either as requested by client or due to buffering.
@@ -239,7 +250,7 @@
void performDecoderFlush(FlushCommand audio, FlushCommand video);
void performReset();
void performScanSources();
- void performSetSurface(const sp<NativeWindowWrapper> &wrapper);
+ void performSetSurface(const sp<Surface> &wrapper);
void performResumeDecoders(bool needNotify);
void onSourceNotify(const sp<AMessage> &msg);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 3c4d695..376c93a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -33,6 +33,8 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
+#include <gui/Surface.h>
+
#include "avc_utils.h"
#include "ATSParser.h"
@@ -42,10 +44,10 @@
const sp<AMessage> ¬ify,
const sp<Source> &source,
const sp<Renderer> &renderer,
- const sp<NativeWindowWrapper> &nativeWindow,
+ const sp<Surface> &surface,
const sp<CCDecoder> &ccDecoder)
: DecoderBase(notify),
- mNativeWindow(nativeWindow),
+ mSurface(surface),
mSource(source),
mRenderer(renderer),
mCCDecoder(ccDecoder),
@@ -178,14 +180,9 @@
mIsAudio = !strncasecmp("audio/", mime.c_str(), 6);
mIsVideoAVC = !strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime.c_str());
- sp<Surface> surface = NULL;
- if (mNativeWindow != NULL) {
- surface = mNativeWindow->getSurfaceTextureClient();
- }
-
mComponentName = mime;
mComponentName.append(" decoder");
- ALOGV("[%s] onConfigure (surface=%p)", mComponentName.c_str(), surface.get());
+ ALOGV("[%s] onConfigure (surface=%p)", mComponentName.c_str(), mSurface.get());
mCodec = MediaCodec::CreateByType(mCodecLooper, mime.c_str(), false /* encoder */);
int32_t secure = 0;
@@ -210,17 +207,17 @@
mCodec->getName(&mComponentName);
status_t err;
- if (mNativeWindow != NULL) {
+ if (mSurface != NULL) {
// disconnect from surface as MediaCodec will reconnect
err = native_window_api_disconnect(
- surface.get(), NATIVE_WINDOW_API_MEDIA);
+ mSurface.get(), NATIVE_WINDOW_API_MEDIA);
// We treat this as a warning, as this is a preparatory step.
// Codec will try to connect to the surface, which is where
// any error signaling will occur.
ALOGW_IF(err != OK, "failed to disconnect from surface: %d", err);
}
err = mCodec->configure(
- format, surface, NULL /* crypto */, 0 /* flags */);
+ format, mSurface, NULL /* crypto */, 0 /* flags */);
if (err != OK) {
ALOGE("Failed to configure %s decoder (err=%d)", mComponentName.c_str(), err);
mCodec->release();
@@ -337,12 +334,10 @@
mCodec = NULL;
++mBufferGeneration;
- if (mNativeWindow != NULL) {
+ if (mSurface != NULL) {
// reconnect to surface as MediaCodec disconnected from it
status_t error =
- native_window_api_connect(
- mNativeWindow->getNativeWindow().get(),
- NATIVE_WINDOW_API_MEDIA);
+ native_window_api_connect(mSurface.get(), NATIVE_WINDOW_API_MEDIA);
ALOGW_IF(error != NO_ERROR,
"[%s] failed to connect to native window, error=%d",
mComponentName.c_str(), error);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index dd84620..070d51a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -27,7 +27,7 @@
Decoder(const sp<AMessage> ¬ify,
const sp<Source> &source,
const sp<Renderer> &renderer = NULL,
- const sp<NativeWindowWrapper> &nativeWindow = NULL,
+ const sp<Surface> &surface = NULL,
const sp<CCDecoder> &ccDecoder = NULL);
virtual void getStats(
@@ -54,7 +54,7 @@
kWhatRenderBuffer = 'rndr',
};
- sp<NativeWindowWrapper> mNativeWindow;
+ sp<Surface> mSurface;
sp<Source> mSource;
sp<Renderer> mRenderer;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 04a324c..231f2e1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -360,9 +360,32 @@
return mState == STATE_RUNNING && !mAtEOS;
}
-status_t NuPlayerDriver::setPlaybackRate(float rate) {
- mPlayer->setPlaybackRate(rate);
- return OK;
+status_t NuPlayerDriver::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ Mutex::Autolock autoLock(mLock);
+ status_t err = mPlayer->setPlaybackSettings(rate);
+ if (err == OK) {
+ if (rate.mSpeed == 0.f && mState == STATE_RUNNING) {
+ mState = STATE_PAUSED;
+ // try to update position
+ (void)mPlayer->getCurrentPosition(&mPositionUs);
+ notifyListener_l(MEDIA_PAUSED);
+ } else if (rate.mSpeed != 0.f && mState == STATE_PAUSED) {
+ mState = STATE_RUNNING;
+ }
+ }
+ return err;
+}
+
+status_t NuPlayerDriver::getPlaybackSettings(AudioPlaybackRate *rate) {
+ return mPlayer->getPlaybackSettings(rate);
+}
+
+status_t NuPlayerDriver::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
+ return mPlayer->setSyncSettings(sync, videoFpsHint);
+}
+
+status_t NuPlayerDriver::getSyncSettings(AVSyncSettings *sync, float *videoFps) {
+ return mPlayer->getSyncSettings(sync, videoFps);
}
status_t NuPlayerDriver::seekTo(int msec) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index 65f170e..9da7fc1 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -49,7 +49,10 @@
virtual status_t stop();
virtual status_t pause();
virtual bool isPlaying();
- virtual status_t setPlaybackRate(float rate);
+ virtual status_t setPlaybackSettings(const AudioPlaybackRate &rate);
+ virtual status_t getPlaybackSettings(AudioPlaybackRate *rate);
+ virtual status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint);
+ virtual status_t getSyncSettings(AVSyncSettings *sync, float *videoFps);
virtual status_t seekTo(int msec);
virtual status_t getCurrentPosition(int *msec);
virtual status_t getDuration(int *msec);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index f229452..ae01cfc 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -66,7 +66,7 @@
mVideoQueueGeneration(0),
mAudioDrainGeneration(0),
mVideoDrainGeneration(0),
- mPlaybackRate(1.0),
+ mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT),
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
mAnchorNumFramesWritten(-1),
@@ -89,6 +89,8 @@
mLastAudioBufferDrained(0),
mWakeLock(new AWakeLock()) {
mMediaClock = new MediaClock;
+ mPlaybackRate = mPlaybackSettings.mSpeed;
+ mMediaClock->setPlaybackRate(mPlaybackRate);
}
NuPlayer::Renderer::~Renderer() {
@@ -121,10 +123,111 @@
msg->post();
}
-void NuPlayer::Renderer::setPlaybackRate(float rate) {
- sp<AMessage> msg = new AMessage(kWhatSetRate, this);
- msg->setFloat("rate", rate);
- msg->post();
+status_t NuPlayer::Renderer::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ sp<AMessage> msg = new AMessage(kWhatConfigPlayback, this);
+ writeToAMessage(msg, rate);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ }
+ return err;
+}
+
+status_t NuPlayer::Renderer::onConfigPlayback(const AudioPlaybackRate &rate /* sanitized */) {
+ if (rate.mSpeed == 0.f) {
+ onPause();
+ // don't call audiosink's setPlaybackRate if pausing, as pitch does not
+ // have to correspond to the any non-0 speed (e.g old speed). Keep
+ // settings nonetheless, using the old speed, in case audiosink changes.
+ AudioPlaybackRate newRate = rate;
+ newRate.mSpeed = mPlaybackSettings.mSpeed;
+ mPlaybackSettings = newRate;
+ return OK;
+ }
+
+ if (mAudioSink != NULL) {
+ status_t err = mAudioSink->setPlaybackRate(rate);
+ if (err != OK) {
+ return err;
+ }
+ }
+ mPlaybackSettings = rate;
+ mPlaybackRate = rate.mSpeed;
+ mMediaClock->setPlaybackRate(mPlaybackRate);
+ return OK;
+}
+
+status_t NuPlayer::Renderer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
+ sp<AMessage> msg = new AMessage(kWhatGetPlaybackSettings, this);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ if (err == OK) {
+ readFromAMessage(response, rate);
+ }
+ }
+ return err;
+}
+
+status_t NuPlayer::Renderer::onGetPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
+ if (mAudioSink != NULL) {
+ status_t err = mAudioSink->getPlaybackRate(rate);
+ if (err == OK) {
+ if (!isAudioPlaybackRateEqual(*rate, mPlaybackSettings)) {
+ ALOGW("correcting mismatch in internal/external playback rate");
+ }
+ // get playback settings used by audiosink, as it may be
+ // slightly off due to audiosink not taking small changes.
+ mPlaybackSettings = *rate;
+ if (mPaused) {
+ rate->mSpeed = 0.f;
+ }
+ }
+ return err;
+ }
+ *rate = mPlaybackSettings;
+ return OK;
+}
+
+status_t NuPlayer::Renderer::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
+ sp<AMessage> msg = new AMessage(kWhatConfigSync, this);
+ writeToAMessage(msg, sync, videoFpsHint);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ }
+ return err;
+}
+
+status_t NuPlayer::Renderer::onConfigSync(const AVSyncSettings &sync, float videoFpsHint __unused) {
+ if (sync.mSource != AVSYNC_SOURCE_DEFAULT) {
+ return BAD_VALUE;
+ }
+ // TODO: support sync sources
+ return INVALID_OPERATION;
+}
+
+status_t NuPlayer::Renderer::getSyncSettings(AVSyncSettings *sync, float *videoFps) {
+ sp<AMessage> msg = new AMessage(kWhatGetSyncSettings, this);
+ sp<AMessage> response;
+ status_t err = msg->postAndAwaitResponse(&response);
+ if (err == OK && response != NULL) {
+ CHECK(response->findInt32("err", &err));
+ if (err == OK) {
+ readFromAMessage(response, sync, videoFps);
+ }
+ }
+ return err;
+}
+
+status_t NuPlayer::Renderer::onGetSyncSettings(
+ AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */) {
+ *sync = mSyncSettings;
+ *videoFps = -1.f;
+ return OK;
}
void NuPlayer::Renderer::flush(bool audio, bool notifyComplete) {
@@ -365,13 +468,63 @@
break;
}
- case kWhatSetRate:
+ case kWhatConfigPlayback:
{
- CHECK(msg->findFloat("rate", &mPlaybackRate));
- int32_t ratePermille = (int32_t)(0.5f + 1000 * mPlaybackRate);
- mPlaybackRate = ratePermille / 1000.0f;
- mMediaClock->setPlaybackRate(mPlaybackRate);
- mAudioSink->setPlaybackRatePermille(ratePermille);
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AudioPlaybackRate rate;
+ readFromAMessage(msg, &rate);
+ status_t err = onConfigPlayback(rate);
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetPlaybackSettings:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AudioPlaybackRate rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ status_t err = onGetPlaybackSettings(&rate);
+ sp<AMessage> response = new AMessage;
+ if (err == OK) {
+ writeToAMessage(response, rate);
+ }
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatConfigSync:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+ AVSyncSettings sync;
+ float videoFpsHint;
+ readFromAMessage(msg, &sync, &videoFpsHint);
+ status_t err = onConfigSync(sync, videoFpsHint);
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetSyncSettings:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ ALOGV("kWhatGetSyncSettings");
+ AVSyncSettings sync;
+ float videoFps = -1.f;
+ status_t err = onGetSyncSettings(&sync, &videoFps);
+ sp<AMessage> response = new AMessage;
+ if (err == OK) {
+ writeToAMessage(response, sync, videoFps);
+ }
+ response->setInt32("err", err);
+ response->postReply(replyID);
break;
}
@@ -1176,7 +1329,6 @@
void NuPlayer::Renderer::onPause() {
if (mPaused) {
- ALOGW("Renderer::onPause() called while already paused!");
return;
}
@@ -1214,6 +1366,12 @@
{
Mutex::Autolock autoLock(mLock);
mPaused = false;
+
+ // configure audiosink as we did not do it when pausing
+ if (mAudioSink != NULL) {
+ mAudioSink->setPlaybackRate(mPlaybackSettings);
+ }
+
mMediaClock->setPlaybackRate(mPlaybackRate);
if (!mAudioQueue.empty()) {
@@ -1433,10 +1591,10 @@
&offloadInfo);
if (err == OK) {
- if (mPlaybackRate != 1.0) {
- mAudioSink->setPlaybackRatePermille(
- (int32_t)(mPlaybackRate * 1000 + 0.5f));
- }
+ err = mAudioSink->setPlaybackRate(mPlaybackSettings);
+ }
+
+ if (err == OK) {
// If the playback is offloaded to h/w, we pass
// the HAL some metadata information.
// We don't want to do this for PCM because it
@@ -1486,16 +1644,15 @@
NULL,
NULL,
(audio_output_flags_t)pcmFlags);
+ if (err == OK) {
+ err = mAudioSink->setPlaybackRate(mPlaybackSettings);
+ }
if (err != OK) {
ALOGW("openAudioSink: non offloaded open failed status: %d", err);
mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
return err;
}
mCurrentPcmInfo = info;
- if (mPlaybackRate != 1.0) {
- mAudioSink->setPlaybackRatePermille(
- (int32_t)(mPlaybackRate * 1000 + 0.5f));
- }
mAudioSink->start();
}
if (audioSinkChanged) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 38843d5..928b71b 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -18,6 +18,9 @@
#define NUPLAYER_RENDERER_H_
+#include <media/AudioResamplerPublic.h>
+#include <media/AVSyncSettings.h>
+
#include "NuPlayer.h"
namespace android {
@@ -48,7 +51,10 @@
void queueEOS(bool audio, status_t finalResult);
- void setPlaybackRate(float rate);
+ status_t setPlaybackSettings(const AudioPlaybackRate &rate /* sanitized */);
+ status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
+ status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint);
+ status_t getSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
void flush(bool audio, bool notifyComplete);
@@ -102,7 +108,10 @@
kWhatPostDrainVideoQueue = 'pDVQ',
kWhatQueueBuffer = 'queB',
kWhatQueueEOS = 'qEOS',
- kWhatSetRate = 'setR',
+ kWhatConfigPlayback = 'cfPB',
+ kWhatConfigSync = 'cfSy',
+ kWhatGetPlaybackSettings = 'gPbS',
+ kWhatGetSyncSettings = 'gSyS',
kWhatFlush = 'flus',
kWhatPause = 'paus',
kWhatResume = 'resm',
@@ -141,7 +150,12 @@
int32_t mVideoDrainGeneration;
sp<MediaClock> mMediaClock;
- float mPlaybackRate;
+ float mPlaybackRate; // audio track rate
+
+ AudioPlaybackRate mPlaybackSettings;
+ AVSyncSettings mSyncSettings;
+ float mVideoFpsHint;
+
int64_t mAudioFirstAnchorTimeMediaUs;
int64_t mAnchorTimeMediaUs;
int64_t mAnchorNumFramesWritten;
@@ -217,6 +231,11 @@
void onAudioSinkChanged();
void onDisableOffloadAudio();
void onEnableOffloadAudio();
+ status_t onConfigPlayback(const AudioPlaybackRate &rate /* sanitized */);
+ status_t onGetPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
+ status_t onConfigSync(const AVSyncSettings &sync, float videoFpsHint);
+ status_t onGetSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
+
void onPause();
void onResume();
void onSetVideoFrameRate(float fps);
@@ -252,6 +271,6 @@
DISALLOW_EVIL_CONSTRUCTORS(Renderer);
};
-} // namespace android
+} // namespace android
#endif // NUPLAYER_RENDERER_H_
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index c7df5a0..7b87676 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -24,6 +24,8 @@
#include <inttypes.h>
#include <utils/Trace.h>
+#include <gui/Surface.h>
+
#include <media/stagefright/ACodec.h>
#include <binder/MemoryDealer.h>
@@ -37,7 +39,6 @@
#include <media/stagefright/BufferProducerWrapper.h>
#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/OMXClient.h>
#include <media/stagefright/OMXCodec.h>
@@ -806,6 +807,11 @@
return err;
mNumUndequeuedBuffers = minUndequeuedBuffers;
+ if (!mStoreMetaDataInOutputBuffers) {
+ static_cast<Surface*>(mNativeWindow.get())
+ ->getIGraphicBufferProducer()->allowAllocation(true);
+ }
+
ALOGV("[%s] Allocating %u buffers from a native window of size %u on "
"output port",
mComponentName.c_str(), bufferCount, bufferSize);
@@ -864,6 +870,11 @@
}
}
+ if (!mStoreMetaDataInOutputBuffers) {
+ static_cast<Surface*>(mNativeWindow.get())
+ ->getIGraphicBufferProducer()->allowAllocation(false);
+ }
+
return err;
}
@@ -1306,9 +1317,8 @@
}
}
if (haveNativeWindow) {
- sp<NativeWindowWrapper> windowWrapper(
- static_cast<NativeWindowWrapper *>(obj.get()));
- sp<ANativeWindow> nativeWindow = windowWrapper->getNativeWindow();
+ sp<ANativeWindow> nativeWindow =
+ static_cast<ANativeWindow *>(static_cast<Surface *>(obj.get()));
// START of temporary support for automatic FRC - THIS WILL BE REMOVED
int32_t autoFrc;
@@ -1467,10 +1477,8 @@
}
if (haveNativeWindow) {
- sp<NativeWindowWrapper> nativeWindow(
- static_cast<NativeWindowWrapper *>(obj.get()));
- CHECK(nativeWindow != NULL);
- mNativeWindow = nativeWindow->getNativeWindow();
+ mNativeWindow = static_cast<Surface *>(obj.get());
+ CHECK(mNativeWindow != NULL);
native_window_set_scaling_mode(
mNativeWindow.get(), NATIVE_WINDOW_SCALING_MODE_SCALE_TO_WINDOW);
@@ -4909,7 +4917,10 @@
CHECK(mCodec->mNode == 0);
OMXClient client;
- CHECK_EQ(client.connect(), (status_t)OK);
+ if (client.connect() != OK) {
+ mCodec->signalError(OMX_ErrorUndefined, NO_INIT);
+ return false;
+ }
sp<IOMX> omx = client.interface();
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index e24824b..dd9d393 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -408,11 +408,22 @@
}
}
-status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
+status_t AudioPlayer::setPlaybackRate(const AudioPlaybackRate &rate) {
if (mAudioSink.get() != NULL) {
- return mAudioSink->setPlaybackRatePermille(ratePermille);
+ return mAudioSink->setPlaybackRate(rate);
} else if (mAudioTrack != 0){
- return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
+ return mAudioTrack->setPlaybackRate(rate);
+ } else {
+ return NO_INIT;
+ }
+}
+
+status_t AudioPlayer::getPlaybackRate(AudioPlaybackRate *rate /* nonnull */) {
+ if (mAudioSink.get() != NULL) {
+ return mAudioSink->getPlaybackRate(rate);
+ } else if (mAudioTrack != 0) {
+ *rate = mAudioTrack->getPlaybackRate();
+ return OK;
} else {
return NO_INIT;
}
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 804f131..e5a6a9b 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -50,7 +50,8 @@
}
AudioSource::AudioSource(
- audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount)
+ audio_source_t inputSource, const String16 &opPackageName, uint32_t sampleRate,
+ uint32_t channelCount)
: mStarted(false),
mSampleRate(sampleRate),
mPrevSampleTimeUs(0),
@@ -78,6 +79,7 @@
mRecord = new AudioRecord(
inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
audio_channel_in_mask_from_count(channelCount),
+ opPackageName,
(size_t) (bufCount * frameCount),
AudioRecordCallbackFunction,
this,
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index c14625d..df01e7c 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -241,6 +241,8 @@
mClockEstimator = new WindowedLinearFitEstimator();
+ mPlaybackSettings = AUDIO_PLAYBACK_RATE_DEFAULT;
+
reset();
}
@@ -1009,6 +1011,10 @@
return err;
}
}
+
+ if (mAudioPlayer != NULL) {
+ mAudioPlayer->setPlaybackRate(mPlaybackSettings);
+ }
}
if (mTimeSource == NULL && mAudioPlayer == NULL) {
@@ -1131,6 +1137,10 @@
}
if (err == OK) {
+ err = mAudioPlayer->setPlaybackRate(mPlaybackSettings);
+ }
+
+ if (err == OK) {
modifyFlags(AUDIO_RUNNING, SET);
mWatchForAudioEOS = true;
@@ -2553,14 +2563,6 @@
{
return setCacheStatCollectFreq(request);
}
- case KEY_PARAMETER_PLAYBACK_RATE_PERMILLE:
- {
- if (mAudioPlayer != NULL) {
- return mAudioPlayer->setPlaybackRatePermille(request.readInt32());
- } else {
- return NO_INIT;
- }
- }
default:
{
return ERROR_UNSUPPORTED;
@@ -2597,6 +2599,58 @@
}
}
+status_t AwesomePlayer::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ Mutex::Autolock autoLock(mLock);
+ // cursory sanity check for non-audio and paused cases
+ if ((rate.mSpeed != 0.f && rate.mSpeed < AUDIO_TIMESTRETCH_SPEED_MIN)
+ || rate.mSpeed > AUDIO_TIMESTRETCH_SPEED_MAX
+ || rate.mPitch < AUDIO_TIMESTRETCH_SPEED_MIN
+ || rate.mPitch > AUDIO_TIMESTRETCH_SPEED_MAX) {
+ return BAD_VALUE;
+ }
+
+ status_t err = OK;
+ if (rate.mSpeed == 0.f) {
+ if (mFlags & PLAYING) {
+ modifyFlags(CACHE_UNDERRUN, CLEAR); // same as pause
+ err = pause_l();
+ }
+ if (err == OK) {
+ // save settings (using old speed) in case player is resumed
+ AudioPlaybackRate newRate = rate;
+ newRate.mSpeed = mPlaybackSettings.mSpeed;
+ mPlaybackSettings = newRate;
+ }
+ return err;
+ }
+ if (mAudioPlayer != NULL) {
+ err = mAudioPlayer->setPlaybackRate(rate);
+ }
+ if (err == OK) {
+ mPlaybackSettings = rate;
+ if (!(mFlags & PLAYING)) {
+ play_l();
+ }
+ }
+ return err;
+}
+
+status_t AwesomePlayer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
+ if (mAudioPlayer != NULL) {
+ status_t err = mAudioPlayer->getPlaybackRate(rate);
+ if (err == OK) {
+ mPlaybackSettings = *rate;
+ Mutex::Autolock autoLock(mLock);
+ if (!(mFlags & PLAYING)) {
+ rate->mSpeed = 0.f;
+ }
+ }
+ return err;
+ }
+ *rate = mPlaybackSettings;
+ return OK;
+}
+
status_t AwesomePlayer::getTrackInfo(Parcel *reply) const {
Mutex::Autolock autoLock(mLock);
size_t trackCount = mExtractor->countTracks();
diff --git a/media/libstagefright/MP3Extractor.cpp b/media/libstagefright/MP3Extractor.cpp
index 55e3c19..2e54e8c 100644
--- a/media/libstagefright/MP3Extractor.cpp
+++ b/media/libstagefright/MP3Extractor.cpp
@@ -282,6 +282,41 @@
mFirstFramePos = pos;
mFixedHeader = header;
+ mMeta = new MetaData;
+ sp<XINGSeeker> seeker = XINGSeeker::CreateFromSource(mDataSource, mFirstFramePos);
+
+ if (seeker == NULL) {
+ mSeeker = VBRISeeker::CreateFromSource(mDataSource, post_id3_pos);
+ } else {
+ mSeeker = seeker;
+ int encd = seeker->getEncoderDelay();
+ int encp = seeker->getEncoderPadding();
+ if (encd != 0 || encp != 0) {
+ mMeta->setInt32(kKeyEncoderDelay, encd);
+ mMeta->setInt32(kKeyEncoderPadding, encp);
+ }
+ }
+
+ if (mSeeker != NULL) {
+ // While it is safe to send the XING/VBRI frame to the decoder, this will
+ // result in an extra 1152 samples being output. In addition, the bitrate
+ // of the Xing header might not match the rest of the file, which could
+ // lead to problems when seeking. The real first frame to decode is after
+ // the XING/VBRI frame, so skip there.
+ size_t frame_size;
+ int sample_rate;
+ int num_channels;
+ int bitrate;
+ GetMPEGAudioFrameSize(
+ header, &frame_size, &sample_rate, &num_channels, &bitrate);
+ pos += frame_size;
+ if (!Resync(mDataSource, 0, &pos, &post_id3_pos, &header)) {
+ // mInitCheck will remain NO_INIT
+ return;
+ }
+ mFirstFramePos = pos;
+ mFixedHeader = header;
+ }
size_t frame_size;
int sample_rate;
@@ -292,8 +327,6 @@
unsigned layer = 4 - ((header >> 17) & 3);
- mMeta = new MetaData;
-
switch (layer) {
case 1:
mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_I);
@@ -312,27 +345,6 @@
mMeta->setInt32(kKeyBitRate, bitrate * 1000);
mMeta->setInt32(kKeyChannelCount, num_channels);
- sp<XINGSeeker> seeker = XINGSeeker::CreateFromSource(mDataSource, mFirstFramePos);
-
- if (seeker == NULL) {
- mSeeker = VBRISeeker::CreateFromSource(mDataSource, post_id3_pos);
- } else {
- mSeeker = seeker;
- int encd = seeker->getEncoderDelay();
- int encp = seeker->getEncoderPadding();
- if (encd != 0 || encp != 0) {
- mMeta->setInt32(kKeyEncoderDelay, encd);
- mMeta->setInt32(kKeyEncoderPadding, encp);
- }
- }
-
- if (mSeeker != NULL) {
- // While it is safe to send the XING/VBRI frame to the decoder, this will
- // result in an extra 1152 samples being output. The real first frame to
- // decode is after the XING/VBRI frame, so skip there.
- mFirstFramePos += frame_size;
- }
-
int64_t durationUs;
if (mSeeker == NULL || !mSeeker->getDuration(&durationUs)) {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index aa0d2e6..25887ef 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -42,7 +42,6 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaFilter.h>
#include <media/stagefright/MetaData.h>
-#include <media/stagefright/NativeWindowWrapper.h>
#include <private/android_filesystem_config.h>
#include <utils/Log.h>
#include <utils/Singleton.h>
@@ -392,6 +391,10 @@
tmp.erase(tmp.size() - 7, 7);
}
const sp<IMediaCodecList> mcl = MediaCodecList::getInstance();
+ if (mcl == NULL) {
+ mCodec = NULL; // remove the codec.
+ return NO_INIT; // if called from Java should raise IOException
+ }
ssize_t codecIdx = mcl->findCodecByName(tmp.c_str());
if (codecIdx >= 0) {
const sp<MediaCodecInfo> info = mcl->getCodecInfo(codecIdx);
@@ -477,9 +480,7 @@
msg->setInt32("flags", flags);
if (nativeWindow != NULL) {
- msg->setObject(
- "native-window",
- new NativeWindowWrapper(nativeWindow));
+ msg->setObject("native-window", nativeWindow);
}
if (crypto != NULL) {
@@ -1610,11 +1611,7 @@
if (obj != NULL) {
format->setObject("native-window", obj);
-
- status_t err = setNativeWindow(
- static_cast<NativeWindowWrapper *>(obj.get())
- ->getSurfaceTextureClient());
-
+ status_t err = setNativeWindow(static_cast<Surface *>(obj.get()));
if (err != OK) {
PostReplyWithError(replyID, err);
break;
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
index 26798ae..f12a913 100644
--- a/media/libstagefright/MediaCodecList.cpp
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -80,6 +80,10 @@
infos.push_back(gCodecList->getCodecInfo(i));
}
}
+ } else {
+ // failure to initialize may be temporary. retry on next call.
+ delete gCodecList;
+ gCodecList = NULL;
}
}
}
@@ -168,7 +172,7 @@
OMXClient client;
mInitCheck = client.connect();
if (mInitCheck != OK) {
- return;
+ return; // this may fail if IMediaPlayerService is not available.
}
mOMX = client.interface();
parseXMLFile(codecs_xml);
@@ -882,14 +886,16 @@
return -EINVAL;
}
- // size, blocks, bitrate, frame-rate, blocks-per-second, aspect-ratio: range
+ // size, blocks, bitrate, frame-rate, blocks-per-second, aspect-ratio,
+ // measured-frame-rate, measured-blocks-per-second: range
// quality: range + default + [scale]
// complexity: range + default
bool found;
if (name == "aspect-ratio" || name == "bitrate" || name == "block-count"
|| name == "blocks-per-second" || name == "complexity"
- || name == "frame-rate" || name == "quality" || name == "size") {
+ || name == "frame-rate" || name == "quality" || name == "size"
+ || name == "measured-blocks-per-second" || name == "measured-frame-rate") {
AString min, max;
if (msg->findString("min", &min) && msg->findString("max", &max)) {
min.append("-");
diff --git a/media/libstagefright/MediaSync.cpp b/media/libstagefright/MediaSync.cpp
index 8030a36..85027ce 100644
--- a/media/libstagefright/MediaSync.cpp
+++ b/media/libstagefright/MediaSync.cpp
@@ -56,6 +56,10 @@
mPlaybackRate(0.0) {
mMediaClock = new MediaClock;
+ // initialize settings
+ mPlaybackSettings = AUDIO_PLAYBACK_RATE_DEFAULT;
+ mPlaybackSettings.mSpeed = mPlaybackRate;
+
mLooper = new ALooper;
mLooper->setName("MediaSync");
mLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
@@ -84,6 +88,11 @@
return INVALID_OPERATION;
}
+ if (output == NULL && mSyncSettings.mSource == AVSYNC_SOURCE_VSYNC) {
+ ALOGE("configureSurface: output surface is used as sync source and cannot be removed.");
+ return INVALID_OPERATION;
+ }
+
if (output != NULL) {
IGraphicBufferProducer::QueueBufferOutput queueBufferOutput;
sp<OutputListener> listener(new OutputListener(this));
@@ -105,8 +114,7 @@
}
// |audioTrack| is used only for querying information.
-status_t MediaSync::configureAudioTrack(
- const sp<AudioTrack> &audioTrack, uint32_t nativeSampleRateInHz) {
+status_t MediaSync::configureAudioTrack(const sp<AudioTrack> &audioTrack) {
Mutex::Autolock lock(mMutex);
// TODO: support audio track change.
@@ -115,15 +123,35 @@
return INVALID_OPERATION;
}
- if (audioTrack != NULL && nativeSampleRateInHz <= 0) {
- ALOGE("configureAudioTrack: native sample rate should be positive.");
- return BAD_VALUE;
+ if (audioTrack == NULL && mSyncSettings.mSource == AVSYNC_SOURCE_AUDIO) {
+ ALOGE("configureAudioTrack: audioTrack is used as sync source and cannot be removed.");
+ return INVALID_OPERATION;
}
- mAudioTrack = audioTrack;
- mNativeSampleRateInHz = nativeSampleRateInHz;
+ if (audioTrack != NULL) {
+ // check if audio track supports the playback settings
+ if (mPlaybackSettings.mSpeed != 0.f
+ && audioTrack->setPlaybackRate(mPlaybackSettings) != OK) {
+ ALOGE("playback settings are not supported by the audio track");
+ return INVALID_OPERATION;
+ }
+ uint32_t nativeSampleRateInHz = audioTrack->getOriginalSampleRate();
+ if (nativeSampleRateInHz <= 0) {
+ ALOGE("configureAudioTrack: native sample rate should be positive.");
+ return BAD_VALUE;
+ }
+ mAudioTrack = audioTrack;
+ mNativeSampleRateInHz = nativeSampleRateInHz;
+ (void)setPlaybackSettings_l(mPlaybackSettings);
+ }
+ else {
+ mAudioTrack = NULL;
+ mNativeSampleRateInHz = 0;
+ }
- return NO_ERROR;
+ // potentially resync to new source
+ resync_l();
+ return OK;
}
status_t MediaSync::createInputSurface(
@@ -158,21 +186,27 @@
return status;
}
-status_t MediaSync::setPlaybackRate(float rate) {
- if (rate < 0.0) {
- return BAD_VALUE;
+void MediaSync::resync_l() {
+ AVSyncSource src = mSyncSettings.mSource;
+ if (src == AVSYNC_SOURCE_DEFAULT) {
+ if (mAudioTrack != NULL) {
+ src = AVSYNC_SOURCE_AUDIO;
+ } else {
+ src = AVSYNC_SOURCE_SYSTEM_CLOCK;
+ }
}
- Mutex::Autolock lock(mMutex);
+ // TODO: resync ourselves to the current clock (e.g. on sync source change)
+ updatePlaybackRate_l(mPlaybackRate);
+}
+void MediaSync::updatePlaybackRate_l(float rate) {
if (rate > mPlaybackRate) {
mNextBufferItemMediaUs = -1;
}
mPlaybackRate = rate;
mMediaClock->setPlaybackRate(rate);
onDrainVideo_l();
-
- return OK;
}
sp<const MediaClock> MediaSync::getMediaClock() {
@@ -264,6 +298,95 @@
mInput->setConsumerName(String8(name.c_str()));
}
+status_t MediaSync::setVideoFrameRateHint(float rate) {
+ // ignored until we add the FrameScheduler
+ return rate >= 0.f ? OK : BAD_VALUE;
+}
+
+float MediaSync::getVideoFrameRate() {
+ // we don't know the frame rate
+ return -1.f;
+}
+
+status_t MediaSync::setSyncSettings(const AVSyncSettings &syncSettings) {
+ // validate settings
+ if (syncSettings.mSource >= AVSYNC_SOURCE_MAX
+ || syncSettings.mAudioAdjustMode >= AVSYNC_AUDIO_ADJUST_MODE_MAX
+ || syncSettings.mTolerance < 0.f
+ || syncSettings.mTolerance >= AVSYNC_TOLERANCE_MAX) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMutex);
+
+ // verify that we have the sync source
+ switch (syncSettings.mSource) {
+ case AVSYNC_SOURCE_AUDIO:
+ if (mAudioTrack == NULL) {
+ ALOGE("setSyncSettings: audio sync source requires an audio track");
+ return BAD_VALUE;
+ }
+ break;
+ case AVSYNC_SOURCE_VSYNC:
+ if (mOutput == NULL) {
+ ALOGE("setSyncSettings: vsync sync source requires an output surface");
+ return BAD_VALUE;
+ }
+ break;
+ default:
+ break;
+ }
+
+ mSyncSettings = syncSettings;
+ resync_l();
+ return OK;
+}
+
+void MediaSync::getSyncSettings(AVSyncSettings *syncSettings) {
+ Mutex::Autolock lock(mMutex);
+ *syncSettings = mSyncSettings;
+}
+
+status_t MediaSync::setPlaybackSettings(const AudioPlaybackRate &rate) {
+ Mutex::Autolock lock(mMutex);
+
+ status_t err = setPlaybackSettings_l(rate);
+ if (err == OK) {
+ // TODO: adjust rate if using VSYNC as source
+ updatePlaybackRate_l(rate.mSpeed);
+ }
+ return err;
+}
+
+status_t MediaSync::setPlaybackSettings_l(const AudioPlaybackRate &rate) {
+ if (rate.mSpeed < 0.f || rate.mPitch < 0.f) {
+ // We don't validate other audio settings.
+ // They will be validated when/if audiotrack is set.
+ return BAD_VALUE;
+ }
+
+ if (mAudioTrack != NULL) {
+ if (rate.mSpeed == 0.f) {
+ mAudioTrack->pause();
+ } else {
+ status_t err = mAudioTrack->setPlaybackRate(rate);
+ if (err != OK) {
+ return BAD_VALUE;
+ }
+
+ // ignore errors
+ (void)mAudioTrack->start();
+ }
+ }
+ mPlaybackSettings = rate;
+ return OK;
+}
+
+void MediaSync::getPlaybackSettings(AudioPlaybackRate *rate) {
+ Mutex::Autolock lock(mMutex);
+ *rate = mPlaybackSettings;
+}
+
int64_t MediaSync::getRealTime(int64_t mediaTimeUs, int64_t nowUs) {
int64_t realUs;
if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) {
@@ -430,7 +553,16 @@
return;
}
+ if (mBuffersFromInput.indexOfKey(bufferItem.mGraphicBuffer->getId()) >= 0) {
+ // Something is wrong since this buffer should be at our hands, bail.
+ mInput->consumerDisconnect();
+ onAbandoned_l(true /* isInput */);
+ return;
+ }
+ mBuffersFromInput.add(bufferItem.mGraphicBuffer->getId(), bufferItem.mGraphicBuffer);
+
mBufferItems.push_back(bufferItem);
+
if (mBufferItems.size() == 1) {
onDrainVideo_l();
}
@@ -497,9 +629,19 @@
void MediaSync::returnBufferToInput_l(
const sp<GraphicBuffer> &buffer, const sp<Fence> &fence) {
+ ssize_t ix = mBuffersFromInput.indexOfKey(buffer->getId());
+ if (ix < 0) {
+ // The buffer is unknown, something is wrong, bail.
+ mOutput->disconnect(NATIVE_WINDOW_API_MEDIA);
+ onAbandoned_l(false /* isInput */);
+ return;
+ }
+ sp<GraphicBuffer> oldBuffer = mBuffersFromInput.valueAt(ix);
+ mBuffersFromInput.removeItemsAt(ix);
+
// Attach and release the buffer back to the input.
int consumerSlot;
- status_t status = mInput->attachBuffer(&consumerSlot, buffer);
+ status_t status = mInput->attachBuffer(&consumerSlot, oldBuffer);
ALOGE_IF(status != NO_ERROR, "attaching buffer to input failed (%d)", status);
if (status == NO_ERROR) {
status = mInput->releaseBuffer(consumerSlot, 0 /* frameNumber */,
@@ -512,7 +654,7 @@
return;
}
- ALOGV("released buffer %#llx to input", (long long)buffer->getId());
+ ALOGV("released buffer %#llx to input", (long long)oldBuffer->getId());
// Notify any waiting onFrameAvailable calls.
--mNumOutstandingBuffers;
diff --git a/media/libstagefright/OMXClient.cpp b/media/libstagefright/OMXClient.cpp
index 230c1f7..06a598f 100644
--- a/media/libstagefright/OMXClient.cpp
+++ b/media/libstagefright/OMXClient.cpp
@@ -400,10 +400,16 @@
sp<IBinder> binder = sm->getService(String16("media.player"));
sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
- CHECK(service.get() != NULL);
+ if (service.get() == NULL) {
+ ALOGE("Cannot obtain IMediaPlayerService");
+ return NO_INIT;
+ }
mOMX = service->getOMX();
- CHECK(mOMX.get() != NULL);
+ if (mOMX.get() == NULL) {
+ ALOGE("Cannot obtain IOMX");
+ return NO_INIT;
+ }
if (!mOMX->livesLocally(0 /* node */, getpid())) {
ALOGI("Using client-side OMX mux.");
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 0d8e64a..413628d 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -181,6 +181,11 @@
msg->setInt32("rotation-degrees", rotationDegrees);
}
+ int32_t fps;
+ if (meta->findInt32(kKeyFrameRate, &fps)) {
+ msg->setInt32("frame-rate", fps);
+ }
+
uint32_t type;
const void *data;
size_t size;
@@ -588,6 +593,11 @@
meta->setInt32(kKeyMaxHeight, maxHeight);
}
+ int32_t fps;
+ if (msg->findInt32("frame-rate", &fps)) {
+ meta->setInt32(kKeyFrameRate, fps);
+ }
+
// reassemble the csd data into its original form
sp<ABuffer> csd0;
if (msg->findBuffer("csd-0", &csd0)) {
@@ -891,5 +901,39 @@
|| (t0.mSeq == t1.mSeq && t0.mTimeUs < t1.mTimeUs);
}
+void writeToAMessage(sp<AMessage> msg, const AudioPlaybackRate &rate) {
+ msg->setFloat("speed", rate.mSpeed);
+ msg->setFloat("pitch", rate.mPitch);
+ msg->setInt32("audio-fallback-mode", rate.mFallbackMode);
+ msg->setInt32("audio-stretch-mode", rate.mStretchMode);
+}
+
+void readFromAMessage(const sp<AMessage> &msg, AudioPlaybackRate *rate /* nonnull */) {
+ *rate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ CHECK(msg->findFloat("speed", &rate->mSpeed));
+ CHECK(msg->findFloat("pitch", &rate->mPitch));
+ CHECK(msg->findInt32("audio-fallback-mode", (int32_t *)&rate->mFallbackMode));
+ CHECK(msg->findInt32("audio-stretch-mode", (int32_t *)&rate->mStretchMode));
+}
+
+void writeToAMessage(sp<AMessage> msg, const AVSyncSettings &sync, float videoFpsHint) {
+ msg->setInt32("sync-source", sync.mSource);
+ msg->setInt32("audio-adjust-mode", sync.mAudioAdjustMode);
+ msg->setFloat("tolerance", sync.mTolerance);
+ msg->setFloat("video-fps", videoFpsHint);
+}
+
+void readFromAMessage(
+ const sp<AMessage> &msg,
+ AVSyncSettings *sync /* nonnull */,
+ float *videoFps /* nonnull */) {
+ AVSyncSettings settings;
+ CHECK(msg->findInt32("sync-source", (int32_t *)&settings.mSource));
+ CHECK(msg->findInt32("audio-adjust-mode", (int32_t *)&settings.mAudioAdjustMode));
+ CHECK(msg->findFloat("tolerance", &settings.mTolerance));
+ CHECK(msg->findFloat("video-fps", videoFps));
+ *sync = settings;
+}
+
} // namespace android
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 8bba804..758b2c9 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -21,6 +21,7 @@
#include "HTTPBase.h"
#include "TimedEventQueue.h"
+#include <media/AudioResamplerPublic.h>
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/OMXClient.h>
@@ -93,6 +94,8 @@
status_t setParameter(int key, const Parcel &request);
status_t getParameter(int key, Parcel *reply);
+ status_t setPlaybackSettings(const AudioPlaybackRate &rate);
+ status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
status_t invoke(const Parcel &request, Parcel *reply);
status_t setCacheStatCollectFreq(const Parcel &request);
@@ -180,6 +183,7 @@
sp<MediaSource> mOmxSource;
sp<MediaSource> mAudioSource;
AudioPlayer *mAudioPlayer;
+ AudioPlaybackRate mPlaybackSettings;
int64_t mDurationUs;
int32_t mDisplayWidth;
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index ddca437..70d2c69 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -925,6 +925,11 @@
ALOGV("codec id = %s", codecID);
ALOGV("codec name = %s", track->GetCodecNameAsUTF8());
+ if (codecID == NULL) {
+ ALOGW("unknown codecID is not supported.");
+ continue;
+ }
+
size_t codecPrivateSize;
const unsigned char *codecPrivate =
track->GetCodecPrivate(codecPrivateSize);
@@ -941,10 +946,7 @@
const mkvparser::VideoTrack *vtrack =
static_cast<const mkvparser::VideoTrack *>(track);
- if (codecID == NULL) {
- ALOGW("unknown codecID is not supported.");
- continue;
- } else if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
+ if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
meta->setData(kKeyAVCC, 0, codecPrivate, codecPrivateSize);
} else if (!strcmp("V_MPEG4/ISO/ASP", codecID)) {
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 5411821..0d071b2 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -302,9 +302,13 @@
// The two checks below shouldn't happen,
// we already checked above the stream count matches
ssize_t index = newType2PIDs.indexOfKey(temp[i]->type());
- CHECK(index >= 0);
+ if (index < 0) {
+ return false;
+ }
Vector<int32_t> &newPIDs = newType2PIDs.editValueAt(index);
- CHECK(newPIDs.size() > 0);
+ if (newPIDs.isEmpty()) {
+ return false;
+ }
// get the next PID for temp[i]->type() in the new PID map
Vector<int32_t>::iterator it = newPIDs.begin();
@@ -335,13 +339,11 @@
return ERROR_MALFORMED;
}
- CHECK_EQ(br->getBits(1), 0u);
+ br->skipBits(1); // '0'
MY_LOGV(" reserved = %u", br->getBits(2));
unsigned section_length = br->getBits(12);
ALOGV(" section_length = %u", section_length);
- CHECK_EQ(section_length & 0xc00, 0u);
- CHECK_LE(section_length, 1021u);
MY_LOGV(" program_number = %u", br->getBits(16));
MY_LOGV(" reserved = %u", br->getBits(2));
@@ -358,7 +360,6 @@
unsigned program_info_length = br->getBits(12);
ALOGV(" program_info_length = %u", program_info_length);
- CHECK_EQ(program_info_length & 0xc00, 0u);
br->skipBits(program_info_length * 8); // skip descriptors
@@ -369,8 +370,7 @@
// final CRC.
size_t infoBytesRemaining = section_length - 9 - program_info_length - 4;
- while (infoBytesRemaining > 0) {
- CHECK_GE(infoBytesRemaining, 5u);
+ while (infoBytesRemaining >= 5) {
unsigned streamType = br->getBits(8);
ALOGV(" stream_type = 0x%02x", streamType);
@@ -384,9 +384,6 @@
unsigned ES_info_length = br->getBits(12);
ALOGV(" ES_info_length = %u", ES_info_length);
- CHECK_EQ(ES_info_length & 0xc00, 0u);
-
- CHECK_GE(infoBytesRemaining - 5, ES_info_length);
#if 0
br->skipBits(ES_info_length * 8); // skip descriptors
@@ -398,13 +395,13 @@
unsigned descLength = br->getBits(8);
ALOGV(" len = %u", descLength);
- CHECK_GE(info_bytes_remaining, 2 + descLength);
-
+ if (info_bytes_remaining < descLength) {
+ return ERROR_MALFORMED;
+ }
br->skipBits(descLength * 8);
info_bytes_remaining -= descLength + 2;
}
- CHECK_EQ(info_bytes_remaining, 0u);
#endif
StreamInfo info;
@@ -415,7 +412,9 @@
infoBytesRemaining -= 5 + ES_info_length;
}
- CHECK_EQ(infoBytesRemaining, 0u);
+ if (infoBytesRemaining != 0) {
+ ALOGW("Section data remains unconsumed");
+ }
MY_LOGV(" CRC = 0x%08x", br->getBits(32));
bool PIDsChanged = false;
@@ -680,7 +679,10 @@
}
size_t payloadSizeBits = br->numBitsLeft();
- CHECK_EQ(payloadSizeBits % 8, 0u);
+ if (payloadSizeBits % 8 != 0u) {
+ ALOGE("Wrong value");
+ return BAD_VALUE;
+ }
size_t neededSize = mBuffer->size() + payloadSizeBits / 8;
if (mBuffer->capacity() < neededSize) {
@@ -797,8 +799,6 @@
return ERROR_MALFORMED;
}
- CHECK_EQ(packet_startcode_prefix, 0x000001u);
-
unsigned stream_id = br->getBits(8);
ALOGV("stream_id = 0x%02x", stream_id);
@@ -813,7 +813,9 @@
&& stream_id != 0xff // program_stream_directory
&& stream_id != 0xf2 // DSMCC
&& stream_id != 0xf8) { // H.222.1 type E
- CHECK_EQ(br->getBits(2), 2u);
+ if (br->getBits(2) != 2u) {
+ return ERROR_MALFORMED;
+ }
MY_LOGV("PES_scrambling_control = %u", br->getBits(2));
MY_LOGV("PES_priority = %u", br->getBits(1));
@@ -847,34 +849,51 @@
uint64_t PTS = 0, DTS = 0;
if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {
- CHECK_GE(optional_bytes_remaining, 5u);
+ if (optional_bytes_remaining < 5u) {
+ return ERROR_MALFORMED;
+ }
if (br->getBits(4) != PTS_DTS_flags) {
- ALOGE("PES data Error!");
return ERROR_MALFORMED;
}
PTS = ((uint64_t)br->getBits(3)) << 30;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
PTS |= ((uint64_t)br->getBits(15)) << 15;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
PTS |= br->getBits(15);
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("PTS = 0x%016" PRIx64 " (%.2f)", PTS, PTS / 90000.0);
optional_bytes_remaining -= 5;
if (PTS_DTS_flags == 3) {
- CHECK_GE(optional_bytes_remaining, 5u);
+ if (optional_bytes_remaining < 5u) {
+ return ERROR_MALFORMED;
+ }
- CHECK_EQ(br->getBits(4), 1u);
+ if (br->getBits(4) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS = ((uint64_t)br->getBits(3)) << 30;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS |= ((uint64_t)br->getBits(15)) << 15;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS |= br->getBits(15);
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("DTS = %" PRIu64, DTS);
@@ -883,31 +902,47 @@
}
if (ESCR_flag) {
- CHECK_GE(optional_bytes_remaining, 6u);
+ if (optional_bytes_remaining < 6u) {
+ return ERROR_MALFORMED;
+ }
br->getBits(2);
uint64_t ESCR = ((uint64_t)br->getBits(3)) << 30;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ESCR |= ((uint64_t)br->getBits(15)) << 15;
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ESCR |= br->getBits(15);
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("ESCR = %" PRIu64, ESCR);
MY_LOGV("ESCR_extension = %u", br->getBits(9));
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
optional_bytes_remaining -= 6;
}
if (ES_rate_flag) {
- CHECK_GE(optional_bytes_remaining, 3u);
+ if (optional_bytes_remaining < 3u) {
+ return ERROR_MALFORMED;
+ }
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
MY_LOGV("ES_rate = %u", br->getBits(22));
- CHECK_EQ(br->getBits(1), 1u);
+ if (br->getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
optional_bytes_remaining -= 3;
}
@@ -917,7 +952,9 @@
// ES data follows.
if (PES_packet_length != 0) {
- CHECK_GE(PES_packet_length, PES_header_data_length + 3);
+ if (PES_packet_length < PES_header_data_length + 3) {
+ return ERROR_MALFORMED;
+ }
unsigned dataLength =
PES_packet_length - 3 - PES_header_data_length;
@@ -930,7 +967,9 @@
return ERROR_MALFORMED;
}
- CHECK_GE(br->numBitsLeft(), dataLength * 8);
+ if (br->numBitsLeft() < dataLength * 8) {
+ return ERROR_MALFORMED;
+ }
onPayloadData(
PTS_DTS_flags, PTS, DTS, br->data(), dataLength);
@@ -942,15 +981,21 @@
br->data(), br->numBitsLeft() / 8);
size_t payloadSizeBits = br->numBitsLeft();
- CHECK_EQ(payloadSizeBits % 8, 0u);
+ if (payloadSizeBits % 8 != 0u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("There's %zu bytes of payload.", payloadSizeBits / 8);
}
} else if (stream_id == 0xbe) { // padding_stream
- CHECK_NE(PES_packet_length, 0u);
+ if (PES_packet_length == 0u) {
+ return ERROR_MALFORMED;
+ }
br->skipBits(PES_packet_length * 8);
} else {
- CHECK_NE(PES_packet_length, 0u);
+ if (PES_packet_length == 0u) {
+ return ERROR_MALFORMED;
+ }
br->skipBits(PES_packet_length * 8);
}
@@ -1082,7 +1127,10 @@
}
status_t ATSParser::feedTSPacket(const void *data, size_t size) {
- CHECK_EQ(size, kTSPacketSize);
+ if (size != kTSPacketSize) {
+ ALOGE("Wrong TS packet size");
+ return BAD_VALUE;
+ }
ABitReader br((const uint8_t *)data, kTSPacketSize);
return parseTS(&br);
@@ -1108,14 +1156,23 @@
}
} else if (type == DISCONTINUITY_ABSOLUTE_TIME) {
int64_t timeUs;
- CHECK(extra->findInt64("timeUs", &timeUs));
+ if (!extra->findInt64("timeUs", &timeUs)) {
+ ALOGE("timeUs not found");
+ return;
+ }
- CHECK(mPrograms.empty());
+ if (!mPrograms.empty()) {
+ ALOGE("mPrograms is not empty");
+ return;
+ }
mAbsoluteTimeAnchorUs = timeUs;
return;
} else if (type == DISCONTINUITY_TIME_OFFSET) {
int64_t offset;
- CHECK(extra->findInt64("offset", &offset));
+ if (!extra->findInt64("offset", &offset)) {
+ ALOGE("offset not found");
+ return;
+ }
mTimeOffsetValid = true;
mTimeOffsetUs = offset;
@@ -1128,7 +1185,10 @@
}
void ATSParser::signalEOS(status_t finalResult) {
- CHECK_NE(finalResult, (status_t)OK);
+ if (finalResult == (status_t) OK) {
+ ALOGE("finalResult not OK");
+ return;
+ }
for (size_t i = 0; i < mPrograms.size(); ++i) {
mPrograms.editItemAt(i)->signalEOS(finalResult);
@@ -1144,14 +1204,12 @@
}
unsigned section_syntax_indictor = br->getBits(1);
ALOGV(" section_syntax_indictor = %u", section_syntax_indictor);
- CHECK_EQ(section_syntax_indictor, 1u);
- CHECK_EQ(br->getBits(1), 0u);
+ br->skipBits(1); // '0'
MY_LOGV(" reserved = %u", br->getBits(2));
unsigned section_length = br->getBits(12);
ALOGV(" section_length = %u", section_length);
- CHECK_EQ(section_length & 0xc00, 0u);
MY_LOGV(" transport_stream_id = %u", br->getBits(16));
MY_LOGV(" reserved = %u", br->getBits(2));
@@ -1161,7 +1219,6 @@
MY_LOGV(" last_section_number = %u", br->getBits(8));
size_t numProgramBytes = (section_length - 5 /* header */ - 4 /* crc */);
- CHECK_EQ((numProgramBytes % 4), 0u);
for (size_t i = 0; i < numProgramBytes / 4; ++i) {
unsigned program_number = br->getBits(16);
@@ -1221,7 +1278,9 @@
br->skipBits(skip * 8);
}
- CHECK((br->numBitsLeft() % 8) == 0);
+ if (br->numBitsLeft() % 8 != 0) {
+ return ERROR_MALFORMED;
+ }
status_t err = section->append(br->data(), br->numBitsLeft() / 8);
if (err != OK) {
@@ -1291,7 +1350,7 @@
return OK;
}
-void ATSParser::parseAdaptationField(ABitReader *br, unsigned PID) {
+status_t ATSParser::parseAdaptationField(ABitReader *br, unsigned PID) {
unsigned adaptation_field_length = br->getBits(8);
if (adaptation_field_length > 0) {
@@ -1307,6 +1366,9 @@
size_t numBitsRead = 4;
if (PCR_flag) {
+ if (adaptation_field_length * 8 < 52) {
+ return ERROR_MALFORMED;
+ }
br->skipBits(4);
uint64_t PCR_base = br->getBits(32);
PCR_base = (PCR_base << 1) | br->getBits(1);
@@ -1337,10 +1399,9 @@
numBitsRead += 52;
}
- CHECK_GE(adaptation_field_length * 8, numBitsRead);
-
br->skipBits(adaptation_field_length * 8 - numBitsRead);
}
+ return OK;
}
status_t ATSParser::parseTS(ABitReader *br) {
@@ -1375,15 +1436,16 @@
// ALOGI("PID = 0x%04x, continuity_counter = %u", PID, continuity_counter);
- if (adaptation_field_control == 2 || adaptation_field_control == 3) {
- parseAdaptationField(br, PID);
- }
-
status_t err = OK;
- if (adaptation_field_control == 1 || adaptation_field_control == 3) {
- err = parsePID(
- br, PID, continuity_counter, payload_unit_start_indicator);
+ if (adaptation_field_control == 2 || adaptation_field_control == 3) {
+ err = parseAdaptationField(br, PID);
+ }
+ if (err == OK) {
+ if (adaptation_field_control == 1 || adaptation_field_control == 3) {
+ err = parsePID(
+ br, PID, continuity_counter, payload_unit_start_indicator);
+ }
}
++mNumTSPacketsParsed;
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index 87ab1a0..4def333 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -133,7 +133,7 @@
unsigned continuity_counter,
unsigned payload_unit_start_indicator);
- void parseAdaptationField(ABitReader *br, unsigned PID);
+ status_t parseAdaptationField(ABitReader *br, unsigned PID);
status_t parseTS(ABitReader *br);
void updatePCR(unsigned PID, uint64_t PCR, size_t byteOffsetFromStart);
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index f28a1fd..7b5b46a 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -421,8 +421,8 @@
}
default:
- TRESPASS();
- break;
+ ALOGE("Unknown mode: %d", mMode);
+ return ERROR_MALFORMED;
}
}
@@ -503,7 +503,10 @@
case METADATA:
return dequeueAccessUnitMetadata();
default:
- CHECK_EQ((unsigned)mMode, (unsigned)MPEG_AUDIO);
+ if (mMode != MPEG_AUDIO) {
+ ALOGE("Unknown mode");
+ return NULL;
+ }
return dequeueAccessUnitMPEGAudio();
}
}
@@ -540,7 +543,10 @@
memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("negative timeUs");
+ return NULL;
+ }
accessUnit->meta()->setInt64("timeUs", timeUs);
accessUnit->meta()->setInt32("isSync", 1);
@@ -560,15 +566,24 @@
}
ABitReader bits(mBuffer->data(), 4);
- CHECK_EQ(bits.getBits(8), 0xa0);
+ if (bits.getBits(8) != 0xa0) {
+ ALOGE("Unexpected bit values");
+ return NULL;
+ }
unsigned numAUs = bits.getBits(8);
bits.skipBits(8);
unsigned quantization_word_length __unused = bits.getBits(2);
unsigned audio_sampling_frequency = bits.getBits(3);
unsigned num_channels = bits.getBits(3);
- CHECK_EQ(audio_sampling_frequency, 2); // 48kHz
- CHECK_EQ(num_channels, 1u); // stereo!
+ if (audio_sampling_frequency != 2) {
+ ALOGE("Wrong sampling freq");
+ return NULL;
+ }
+ if (num_channels != 1u) {
+ ALOGE("Wrong channel #");
+ return NULL;
+ }
if (mFormat == NULL) {
mFormat = new MetaData;
@@ -590,7 +605,10 @@
memcpy(accessUnit->data(), mBuffer->data() + 4, payloadSize);
int64_t timeUs = fetchTimestamp(payloadSize + 4);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("Negative timeUs");
+ return NULL;
+ }
accessUnit->meta()->setInt64("timeUs", timeUs);
accessUnit->meta()->setInt32("isSync", 1);
@@ -614,14 +632,19 @@
return NULL;
}
- CHECK(!mRangeInfos.empty());
+ if (mRangeInfos.empty()) {
+ return NULL;
+ }
const RangeInfo &info = *mRangeInfos.begin();
if (mBuffer->size() < info.mLength) {
return NULL;
}
- CHECK_GE(info.mTimestampUs, 0ll);
+ if (info.mTimestampUs < 0ll) {
+ ALOGE("Negative info.mTimestampUs");
+ return NULL;
+ }
// The idea here is consume all AAC frames starting at offsets before
// info.mLength so we can assign a meaningful timestamp without
@@ -638,17 +661,26 @@
// adts_fixed_header
- CHECK_EQ(bits.getBits(12), 0xfffu);
+ if (bits.getBits(12) != 0xfffu) {
+ ALOGE("Wrong atds_fixed_header");
+ return NULL;
+ }
bits.skipBits(3); // ID, layer
bool protection_absent __unused = bits.getBits(1) != 0;
if (mFormat == NULL) {
unsigned profile = bits.getBits(2);
- CHECK_NE(profile, 3u);
+ if (profile == 3u) {
+ ALOGE("profile should not be 3");
+ return NULL;
+ }
unsigned sampling_freq_index = bits.getBits(4);
bits.getBits(1); // private_bit
unsigned channel_configuration = bits.getBits(3);
- CHECK_NE(channel_configuration, 0u);
+ if (channel_configuration == 0u) {
+ ALOGE("channel_config should not be 0");
+ return NULL;
+ }
bits.skipBits(2); // original_copy, home
mFormat = MakeAACCodecSpecificData(
@@ -658,8 +690,14 @@
int32_t sampleRate;
int32_t numChannels;
- CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate));
- CHECK(mFormat->findInt32(kKeyChannelCount, &numChannels));
+ if (!mFormat->findInt32(kKeySampleRate, &sampleRate)) {
+ ALOGE("SampleRate not found");
+ return NULL;
+ }
+ if (!mFormat->findInt32(kKeyChannelCount, &numChannels)) {
+ ALOGE("ChannelCount not found");
+ return NULL;
+ }
ALOGI("found AAC codec config (%d Hz, %d channels)",
sampleRate, numChannels);
@@ -682,7 +720,8 @@
if (number_of_raw_data_blocks_in_frame != 0) {
// To be implemented.
- TRESPASS();
+ ALOGE("Should not reach here.");
+ return NULL;
}
if (offset + aac_frame_length > mBuffer->size()) {
@@ -714,7 +753,9 @@
bool first = true;
while (size > 0) {
- CHECK(!mRangeInfos.empty());
+ if (mRangeInfos.empty()) {
+ return timeUs;
+ }
RangeInfo *info = &*mRangeInfos.begin();
@@ -813,7 +854,10 @@
unsigned nalType = mBuffer->data()[pos.nalOffset] & 0x1f;
if (nalType == 6 && pos.nalSize > 0) {
- CHECK_LT(seiIndex, sei->size() / sizeof(NALPosition));
+ if (seiIndex >= sei->size() / sizeof(NALPosition)) {
+ ALOGE("Wrong seiIndex");
+ return NULL;
+ }
NALPosition &seiPos = ((NALPosition *)sei->data())[seiIndex++];
seiPos.nalOffset = dstOffset + 4;
seiPos.nalSize = pos.nalSize;
@@ -851,7 +895,10 @@
mBuffer->setRange(0, mBuffer->size() - nextScan);
int64_t timeUs = fetchTimestamp(nextScan);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("Negative timeUs");
+ return NULL;
+ }
accessUnit->meta()->setInt64("timeUs", timeUs);
if (foundIDR) {
@@ -873,7 +920,10 @@
totalSize += nalSize;
}
- CHECK_EQ(err, (status_t)-EAGAIN);
+ if (err != (status_t)-EAGAIN) {
+ ALOGE("Unexpeted err");
+ return NULL;
+ }
return NULL;
}
@@ -890,9 +940,12 @@
size_t frameSize;
int samplingRate, numChannels, bitrate, numSamples;
- CHECK(GetMPEGAudioFrameSize(
+ if (!GetMPEGAudioFrameSize(
header, &frameSize, &samplingRate, &numChannels,
- &bitrate, &numSamples));
+ &bitrate, &numSamples)) {
+ ALOGE("Failed to get audio frame size");
+ return NULL;
+ }
if (size < frameSize) {
return NULL;
@@ -910,7 +963,10 @@
mBuffer->setRange(0, mBuffer->size() - frameSize);
int64_t timeUs = fetchTimestamp(frameSize);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("Negative timeUs");
+ return NULL;
+ }
accessUnit->meta()->setInt64("timeUs", timeUs);
accessUnit->meta()->setInt32("isSync", 1);
@@ -932,7 +988,7 @@
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
break;
default:
- TRESPASS();
+ return NULL;
}
mFormat->setInt32(kKeySampleRate, samplingRate);
@@ -943,7 +999,10 @@
}
static void EncodeSize14(uint8_t **_ptr, size_t size) {
- CHECK_LE(size, 0x3fff);
+ if (size > 0x3fff) {
+ ALOGE("Wrong size");
+ return;
+ }
uint8_t *ptr = *_ptr;
@@ -1018,7 +1077,10 @@
// seqHeader without/with extension
if (mFormat == NULL) {
- CHECK_GE(size, 7u);
+ if (size < 7u) {
+ ALOGE("Size too small");
+ return NULL;
+ }
unsigned width =
(data[4] << 4) | data[5] >> 4;
@@ -1078,7 +1140,10 @@
mBuffer->setRange(0, mBuffer->size() - offset);
int64_t timeUs = fetchTimestamp(offset);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("Negative timeUs");
+ return NULL;
+ }
offset = 0;
@@ -1111,7 +1176,7 @@
}
if (memcmp(kStartCode, data, 3)) {
- TRESPASS();
+ return -EAGAIN;
}
size_t offset = 3;
@@ -1171,25 +1236,37 @@
case EXPECT_VISUAL_OBJECT_START:
{
- CHECK_EQ(chunkType, 0xb5);
+ if (chunkType != 0xb5) {
+ ALOGE("Unexpected chunkType");
+ return NULL;
+ }
state = EXPECT_VO_START;
break;
}
case EXPECT_VO_START:
{
- CHECK_LE(chunkType, 0x1f);
+ if (chunkType > 0x1f) {
+ ALOGE("Unexpected chunkType");
+ return NULL;
+ }
state = EXPECT_VOL_START;
break;
}
case EXPECT_VOL_START:
{
- CHECK((chunkType & 0xf0) == 0x20);
+ if ((chunkType & 0xf0) != 0x20) {
+ ALOGE("Wrong chunkType");
+ return NULL;
+ }
- CHECK(ExtractDimensionsFromVOLHeader(
+ if (!ExtractDimensionsFromVOLHeader(
&data[offset], chunkSize,
- &width, &height));
+ &width, &height)) {
+ ALOGE("Failed to get dimension");
+ return NULL;
+ }
state = WAIT_FOR_VOP_START;
break;
@@ -1242,7 +1319,10 @@
mBuffer->setRange(0, size);
int64_t timeUs = fetchTimestamp(offset);
- CHECK_GE(timeUs, 0ll);
+ if (timeUs < 0ll) {
+ ALOGE("Negative timeus");
+ return NULL;
+ }
offset = 0;
@@ -1266,7 +1346,8 @@
}
default:
- TRESPASS();
+ ALOGE("Unknown state: %d", state);
+ return NULL;
}
if (discard) {
diff --git a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
index 85859f7..6d9fe9d 100644
--- a/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2PSExtractor.cpp
@@ -265,7 +265,10 @@
}
unsigned PES_packet_length = U16_AT(mBuffer->data() + 4);
- CHECK_NE(PES_packet_length, 0u);
+ if (PES_packet_length == 0u) {
+ ALOGE("PES_packet_length is 0");
+ return -EAGAIN;
+ }
size_t n = PES_packet_length + 6;
@@ -286,7 +289,10 @@
return ERROR_MALFORMED;
}
- CHECK_EQ(packet_startcode_prefix, 0x000001u);
+ if (packet_startcode_prefix != 0x000001u) {
+ ALOGE("Wrong PES prefix");
+ return ERROR_MALFORMED;
+ }
unsigned stream_id = br.getBits(8);
ALOGV("stream_id = 0x%02x", stream_id);
@@ -366,8 +372,7 @@
&& stream_id != 0xff // program_stream_directory
&& stream_id != 0xf2 // DSMCC
&& stream_id != 0xf8) { // H.222.1 type E
- CHECK_EQ(br.getBits(2), 2u);
-
+ /* unsigned PES_marker_bits = */br.getBits(2); // should be 0x2(hex)
/* unsigned PES_scrambling_control = */br.getBits(2);
/* unsigned PES_priority = */br.getBits(1);
/* unsigned data_alignment_indicator = */br.getBits(1);
@@ -400,16 +405,26 @@
uint64_t PTS = 0, DTS = 0;
if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {
- CHECK_GE(optional_bytes_remaining, 5u);
+ if (optional_bytes_remaining < 5u) {
+ return ERROR_MALFORMED;
+ }
- CHECK_EQ(br.getBits(4), PTS_DTS_flags);
+ if (br.getBits(4) != PTS_DTS_flags) {
+ return ERROR_MALFORMED;
+ }
PTS = ((uint64_t)br.getBits(3)) << 30;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
PTS |= ((uint64_t)br.getBits(15)) << 15;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
PTS |= br.getBits(15);
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("PTS = %" PRIu64, PTS);
// ALOGI("PTS = %.2f secs", PTS / 90000.0f);
@@ -417,16 +432,26 @@
optional_bytes_remaining -= 5;
if (PTS_DTS_flags == 3) {
- CHECK_GE(optional_bytes_remaining, 5u);
+ if (optional_bytes_remaining < 5u) {
+ return ERROR_MALFORMED;
+ }
- CHECK_EQ(br.getBits(4), 1u);
+ if (br.getBits(4) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS = ((uint64_t)br.getBits(3)) << 30;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS |= ((uint64_t)br.getBits(15)) << 15;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
DTS |= br.getBits(15);
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("DTS = %" PRIu64, DTS);
@@ -435,40 +460,62 @@
}
if (ESCR_flag) {
- CHECK_GE(optional_bytes_remaining, 6u);
+ if (optional_bytes_remaining < 6u) {
+ return ERROR_MALFORMED;
+ }
br.getBits(2);
uint64_t ESCR = ((uint64_t)br.getBits(3)) << 30;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ESCR |= ((uint64_t)br.getBits(15)) << 15;
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ESCR |= br.getBits(15);
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
ALOGV("ESCR = %" PRIu64, ESCR);
/* unsigned ESCR_extension = */br.getBits(9);
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
optional_bytes_remaining -= 6;
}
if (ES_rate_flag) {
- CHECK_GE(optional_bytes_remaining, 3u);
+ if (optional_bytes_remaining < 3u) {
+ return ERROR_MALFORMED;
+ }
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
/* unsigned ES_rate = */br.getBits(22);
- CHECK_EQ(br.getBits(1), 1u);
+ if (br.getBits(1) != 1u) {
+ return ERROR_MALFORMED;
+ }
optional_bytes_remaining -= 3;
}
+ if (br.numBitsLeft() < optional_bytes_remaining * 8) {
+ return ERROR_MALFORMED;
+ }
+
br.skipBits(optional_bytes_remaining * 8);
// ES data follows.
- CHECK_GE(PES_packet_length, PES_header_data_length + 3);
+ if (PES_packet_length < PES_header_data_length + 3) {
+ return ERROR_MALFORMED;
+ }
unsigned dataLength =
PES_packet_length - 3 - PES_header_data_length;
@@ -481,7 +528,9 @@
return ERROR_MALFORMED;
}
- CHECK_GE(br.numBitsLeft(), dataLength * 8);
+ if (br.numBitsLeft() < dataLength * 8) {
+ return ERROR_MALFORMED;
+ }
ssize_t index = mTracks.indexOfKey(stream_id);
if (index < 0 && mScanning) {
@@ -521,10 +570,14 @@
return err;
}
} else if (stream_id == 0xbe) { // padding_stream
- CHECK_NE(PES_packet_length, 0u);
+ if (PES_packet_length == 0u) {
+ return ERROR_MALFORMED;
+ }
br.skipBits(PES_packet_length * 8);
} else {
- CHECK_NE(PES_packet_length, 0u);
+ if (PES_packet_length == 0u) {
+ return ERROR_MALFORMED;
+ }
br.skipBits(PES_packet_length * 8);
}
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 74cb5d8..f5c33cf 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -131,7 +131,10 @@
bool seekable = true;
if (mSourceImpls.size() > 1) {
- CHECK_EQ(mSourceImpls.size(), 2u);
+ if (mSourceImpls.size() != 2u) {
+ ALOGE("Wrong size");
+ return NULL;
+ }
sp<MetaData> meta = mSourceImpls.editItemAt(index)->getFormat();
const char *mime;
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index fd889f9..3860e9b 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -19,6 +19,7 @@
#include <gtest/gtest.h>
#include <utils/String8.h>
+#include <utils/String16.h>
#include <utils/Errors.h>
#include <fcntl.h>
#include <unistd.h>
@@ -466,7 +467,7 @@
// Set up the MediaRecorder which runs in the same process as mediaserver
sp<MediaRecorder> SurfaceMediaSourceGLTest::setUpMediaRecorder(int fd, int videoSource,
int outputFormat, int videoEncoder, int width, int height, int fps) {
- sp<MediaRecorder> mr = new MediaRecorder();
+ sp<MediaRecorder> mr = new MediaRecorder(String16());
mr->setVideoSource(videoSource);
mr->setOutputFormat(outputFormat);
mr->setVideoEncoder(videoEncoder);
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 5e2f0bf..ed5a404 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -345,12 +345,14 @@
////////////////////////////////////////////////////////////////////////////////
WifiDisplaySource::PlaybackSession::PlaybackSession(
+ const String16 &opPackageName,
const sp<ANetworkSession> &netSession,
const sp<AMessage> ¬ify,
const in_addr &interfaceAddr,
const sp<IHDCP> &hdcp,
const char *path)
- : mNetSession(netSession),
+ : mOpPackageName(opPackageName),
+ mNetSession(netSession),
mNotify(notify),
mInterfaceAddr(interfaceAddr),
mHDCP(hdcp),
@@ -1069,6 +1071,7 @@
status_t WifiDisplaySource::PlaybackSession::addAudioSource(bool usePCMAudio) {
sp<AudioSource> audioSource = new AudioSource(
AUDIO_SOURCE_REMOTE_SUBMIX,
+ mOpPackageName,
48000 /* sampleRate */,
2 /* channelCount */);
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.h b/media/libstagefright/wifi-display/source/PlaybackSession.h
index 4cd1a75..f6673df 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.h
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.h
@@ -22,6 +22,8 @@
#include "VideoFormats.h"
#include "WifiDisplaySource.h"
+#include <utils/String16.h>
+
namespace android {
struct ABuffer;
@@ -36,6 +38,7 @@
// display.
struct WifiDisplaySource::PlaybackSession : public AHandler {
PlaybackSession(
+ const String16 &opPackageName,
const sp<ANetworkSession> &netSession,
const sp<AMessage> ¬ify,
const struct in_addr &interfaceAddr,
@@ -96,6 +99,8 @@
kWhatPullExtractorSample,
};
+ String16 mOpPackageName;
+
sp<ANetworkSession> mNetSession;
sp<AMessage> mNotify;
in_addr mInterfaceAddr;
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
index 332fe16..e26165e 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
@@ -50,10 +50,12 @@
const AString WifiDisplaySource::sUserAgent = MakeUserAgent();
WifiDisplaySource::WifiDisplaySource(
+ const String16 &opPackageName,
const sp<ANetworkSession> &netSession,
const sp<IRemoteDisplayClient> &client,
const char *path)
- : mState(INITIALIZED),
+ : mOpPackageName(opPackageName),
+ mState(INITIALIZED),
mNetSession(netSession),
mClient(client),
mSessionID(0),
@@ -1245,7 +1247,7 @@
sp<PlaybackSession> playbackSession =
new PlaybackSession(
- mNetSession, notify, mInterfaceAddr, mHDCP, mMediaPath.c_str());
+ mOpPackageName, mNetSession, notify, mInterfaceAddr, mHDCP, mMediaPath.c_str());
looper()->registerHandler(playbackSession);
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.h b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
index c417cf5..c25a675 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.h
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
@@ -25,6 +25,8 @@
#include <netinet/in.h>
+#include <utils/String16.h>
+
namespace android {
struct AReplyToken;
@@ -38,6 +40,7 @@
static const unsigned kWifiDisplayDefaultPort = 7236;
WifiDisplaySource(
+ const String16 &opPackageName,
const sp<ANetworkSession> &netSession,
const sp<IRemoteDisplayClient> &client,
const char *path = NULL);
@@ -114,6 +117,8 @@
static const AString sUserAgent;
+ String16 mOpPackageName;
+
State mState;
VideoFormats mSupportedSourceVideoFormats;
sp<ANetworkSession> mNetSession;
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index 80c1c2f..cd0c462 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -154,6 +154,10 @@
} else {
mData->mCodec = android::MediaCodec::CreateByComponentName(mData->mLooper, name);
}
+ if (mData->mCodec == NULL) { // failed to create codec
+ AMediaCodec_delete(mData);
+ return NULL;
+ }
mData->mHandler = new CodecHandler(mData);
mData->mLooper->registerHandler(mData->mHandler);
mData->mGeneration = 1;
@@ -180,17 +184,21 @@
EXPORT
media_status_t AMediaCodec_delete(AMediaCodec *mData) {
- if (mData->mCodec != NULL) {
- mData->mCodec->release();
- mData->mCodec.clear();
- }
+ if (mData != NULL) {
+ if (mData->mCodec != NULL) {
+ mData->mCodec->release();
+ mData->mCodec.clear();
+ }
- if (mData->mLooper != NULL) {
- mData->mLooper->unregisterHandler(mData->mHandler->id());
- mData->mLooper->stop();
- mData->mLooper.clear();
+ if (mData->mLooper != NULL) {
+ if (mData->mHandler != NULL) {
+ mData->mLooper->unregisterHandler(mData->mHandler->id());
+ }
+ mData->mLooper->stop();
+ mData->mLooper.clear();
+ }
+ delete mData;
}
- delete mData;
return AMEDIA_OK;
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3e4b1fc..bd6889d 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1421,6 +1421,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& opPackageName,
size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
@@ -1440,7 +1441,7 @@
buffers.clear();
// check calling permissions
- if (!recordingAllowed()) {
+ if (!recordingAllowed(opPackageName)) {
ALOGE("openRecord() permission denied: recording not allowed");
lStatus = PERMISSION_DENIED;
goto Exit;
@@ -2456,6 +2457,7 @@
int32_t priority,
audio_io_handle_t io,
int sessionId,
+ const String16& opPackageName,
status_t *status,
int *id,
int *enabled)
@@ -2552,7 +2554,7 @@
// check recording permission for visualizer
if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
- !recordingAllowed()) {
+ !recordingAllowed(opPackageName)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8e06138..8085ec2 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -120,6 +120,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
+ const String16& opPackageName,
size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
@@ -216,6 +217,7 @@
int32_t priority,
audio_io_handle_t io,
int sessionId,
+ const String16& opPackageName,
status_t *status /*non-NULL*/,
int *id,
int *enabled);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index c51021b..7bc6f0c 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -156,11 +156,6 @@
bool mResumeToStopping; // track was paused in stopping state.
bool mFlushHwPending; // track requests for thread flush
- // for last call to getTimestamp
- bool mPreviousTimestampValid;
- // This is either the first timestamp or one that has passed
- // the check to prevent retrograde motion.
- AudioTimestamp mPreviousTimestamp;
}; // end of Track
class TimedTrack : public Track {
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
index 8246fef..0a718fb 100644
--- a/services/audioflinger/ServiceUtilities.cpp
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -14,38 +14,97 @@
* limitations under the License.
*/
+#include <binder/AppOpsManager.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <binder/PermissionCache.h>
#include "ServiceUtilities.h"
+/* When performing permission checks we do not use permission cache for
+ * runtime permissions (protection level dangerous) as they may change at
+ * runtime. All other permissions (protection level normal and dangerous)
+ * can be cached as they never change. Of course all permission checked
+ * here are platform defined.
+ */
+
namespace android {
// Not valid until initialized by AudioFlinger constructor. It would have to be
// re-initialized if the process containing AudioFlinger service forks (which it doesn't).
pid_t getpid_cached;
-bool recordingAllowed() {
+bool recordingAllowed(const String16& opPackageName) {
+ // Note: We are getting the UID from the calling IPC thread state because all
+ // clients that perform recording create AudioRecord in their own processes
+ // and the system does not create AudioRecord objects on behalf of apps. This
+ // differs from playback where in some situations the system recreates AudioTrack
+ // instances associated with a client's MediaPlayer on behalf of this client.
+ // In the latter case we have to store the client UID and pass in along for
+ // security checks.
+
if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
static const String16 sRecordAudio("android.permission.RECORD_AUDIO");
- // don't use PermissionCache; this is not a system permission
- bool ok = checkCallingPermission(sRecordAudio);
- if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
- return ok;
+
+ // IMPORTANT: Don't use PermissionCache - a runtime permission and may change.
+ const bool ok = checkCallingPermission(sRecordAudio);
+ if (!ok) {
+ ALOGE("Request requires android.permission.RECORD_AUDIO");
+ return false;
+ }
+
+ const uid_t uid = IPCThreadState::self()->getCallingUid();
+ String16 checkedOpPackageName = opPackageName;
+
+ // In some cases the calling code has no access to the package it runs under.
+ // For example, code using the wilhelm framework's OpenSL-ES APIs. In this
+ // case we will get the packages for the calling UID and pick the first one
+ // for attributing the app op. This will work correctly for runtime permissions
+ // as for legacy apps we will toggle the app op for all packages in the UID.
+ // The caveat is that the operation may be attributed to the wrong package and
+ // stats based on app ops may be slightly off.
+ if (checkedOpPackageName.size() <= 0) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder = sm->getService(String16("permission"));
+ if (binder == 0) {
+ ALOGE("Cannot get permission service");
+ return false;
+ }
+
+ sp<IPermissionController> permCtrl = interface_cast<IPermissionController>(binder);
+ Vector<String16> packages;
+
+ permCtrl->getPackagesForUid(uid, packages);
+
+ if (packages.isEmpty()) {
+ ALOGE("No packages for calling UID");
+ return false;
+ }
+ checkedOpPackageName = packages[0];
+ }
+
+ AppOpsManager appOps;
+ if (appOps.noteOp(AppOpsManager::OP_RECORD_AUDIO, uid, opPackageName)
+ != AppOpsManager::MODE_ALLOWED) {
+ ALOGE("Request denied by app op OP_RECORD_AUDIO");
+ return false;
+ }
+
+ return true;
}
bool captureAudioOutputAllowed() {
if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
static const String16 sCaptureAudioOutput("android.permission.CAPTURE_AUDIO_OUTPUT");
- // don't use PermissionCache; this is not a system permission
- bool ok = checkCallingPermission(sCaptureAudioOutput);
+ // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
+ bool ok = PermissionCache::checkCallingPermission(sCaptureAudioOutput);
if (!ok) ALOGE("Request requires android.permission.CAPTURE_AUDIO_OUTPUT");
return ok;
}
bool captureHotwordAllowed() {
static const String16 sCaptureHotwordAllowed("android.permission.CAPTURE_AUDIO_HOTWORD");
- bool ok = checkCallingPermission(sCaptureHotwordAllowed);
+ // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
+ bool ok = PermissionCache::checkCallingPermission(sCaptureHotwordAllowed);
if (!ok) ALOGE("android.permission.CAPTURE_AUDIO_HOTWORD");
return ok;
}
@@ -53,15 +112,16 @@
bool settingsAllowed() {
if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS");
- // don't use PermissionCache; this is not a system permission
- bool ok = checkCallingPermission(sAudioSettings);
+ // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
+ bool ok = PermissionCache::checkCallingPermission(sAudioSettings);
if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
return ok;
}
bool modifyAudioRoutingAllowed() {
static const String16 sModifyAudioRoutingAllowed("android.permission.MODIFY_AUDIO_ROUTING");
- bool ok = checkCallingPermission(sModifyAudioRoutingAllowed);
+ // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
+ bool ok = PermissionCache::checkCallingPermission(sModifyAudioRoutingAllowed);
if (!ok) ALOGE("android.permission.MODIFY_AUDIO_ROUTING");
return ok;
}
@@ -69,7 +129,7 @@
bool dumpAllowed() {
// don't optimize for same pid, since mediaserver never dumps itself
static const String16 sDump("android.permission.DUMP");
- // OK to use PermissionCache; this is a system permission
+ // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
bool ok = PermissionCache::checkCallingPermission(sDump);
// convention is for caller to dump an error message to fd instead of logging here
//if (!ok) ALOGE("Request requires android.permission.DUMP");
diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h
index df6f6f4..fba6dce 100644
--- a/services/audioflinger/ServiceUtilities.h
+++ b/services/audioflinger/ServiceUtilities.h
@@ -20,11 +20,10 @@
extern pid_t getpid_cached;
-bool recordingAllowed();
+bool recordingAllowed(const String16& opPackageName);
bool captureAudioOutputAllowed();
bool captureHotwordAllowed();
bool settingsAllowed();
bool modifyAudioRoutingAllowed();
bool dumpAllowed();
-
}
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index c6e9745..1b03060 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -404,8 +404,7 @@
mIsInvalid(false),
mAudioTrackServerProxy(NULL),
mResumeToStopping(false),
- mFlushHwPending(false),
- mPreviousTimestampValid(false)
+ mFlushHwPending(false)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -863,7 +862,6 @@
if (mState == FLUSHED) {
mState = IDLE;
}
- mPreviousTimestampValid = false;
}
}
@@ -885,12 +883,10 @@
{
// Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
if (isFastTrack()) {
- // FIXME no lock held to set mPreviousTimestampValid = false
return INVALID_OPERATION;
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
- // FIXME no lock held to set mPreviousTimestampValid = false
return INVALID_OPERATION;
}
@@ -900,7 +896,6 @@
status_t result = INVALID_OPERATION;
if (!isOffloaded() && !isDirect()) {
if (!playbackThread->mLatchQValid) {
- mPreviousTimestampValid = false;
return INVALID_OPERATION;
}
// FIXME Not accurate under dynamic changes of sample rate and speed.
@@ -919,10 +914,7 @@
uint32_t framesWritten = i >= 0 ?
playbackThread->mLatchQ.mFramesReleased[i] :
mAudioTrackServerProxy->framesReleased();
- if (framesWritten < unpresentedFrames) {
- mPreviousTimestampValid = false;
- // return invalid result
- } else {
+ if (framesWritten >= unpresentedFrames) {
timestamp.mPosition = framesWritten - unpresentedFrames;
timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
result = NO_ERROR;
@@ -931,41 +923,6 @@
result = playbackThread->getTimestamp_l(timestamp);
}
- // Prevent retrograde motion in timestamp.
- if (result == NO_ERROR) {
- if (mPreviousTimestampValid) {
- if (timestamp.mTime.tv_sec < mPreviousTimestamp.mTime.tv_sec ||
- (timestamp.mTime.tv_sec == mPreviousTimestamp.mTime.tv_sec &&
- timestamp.mTime.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) {
- ALOGW("WARNING - retrograde timestamp time");
- // FIXME Consider blocking this from propagating upwards.
- }
-
- // Looking at signed delta will work even when the timestamps
- // are wrapping around.
- int32_t deltaPosition = static_cast<int32_t>(timestamp.mPosition
- - mPreviousTimestamp.mPosition);
- // position can bobble slightly as an artifact; this hides the bobble
- static const int32_t MINIMUM_POSITION_DELTA = 8;
- if (deltaPosition < 0) {
-#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * 1000000000 + time.tv_nsec)
- ALOGW("WARNING - retrograde timestamp position corrected,"
- " %d = %u - %u, (at %llu, %llu nanos)",
- deltaPosition,
- timestamp.mPosition,
- mPreviousTimestamp.mPosition,
- TIME_TO_NANOS(timestamp.mTime),
- TIME_TO_NANOS(mPreviousTimestamp.mTime));
-#undef TIME_TO_NANOS
- }
- if (deltaPosition < MINIMUM_POSITION_DELTA) {
- // Current timestamp is bad. Use last valid timestamp.
- timestamp = mPreviousTimestamp;
- }
- }
- mPreviousTimestamp = timestamp;
- mPreviousTimestampValid = true;
- }
return result;
}
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 9230750..8523fc5 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -106,6 +106,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -129,6 +130,7 @@
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -209,7 +211,7 @@
struct audio_patch *patches,
unsigned int *generation) = 0;
virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;
- virtual void clearAudioPatches(uid_t uid) = 0;
+ virtual void releaseResourcesForUid(uid_t uid) = 0;
virtual status_t acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 144d8ad..a278375 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -315,13 +315,15 @@
mGlobalRefCount += delta;
}
if ((oldGlobalRefCount == 0) && (mGlobalRefCount > 0)) {
- if ((mPolicyMix != NULL) && ((mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) {
+ if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
+ {
mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mRegistrationId,
MIX_STATE_MIXING);
}
} else if ((oldGlobalRefCount > 0) && (mGlobalRefCount == 0)) {
- if ((mPolicyMix != NULL) && ((mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) {
+ if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
+ {
mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mRegistrationId,
MIX_STATE_IDLE);
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 77fc0b9..6f1998c 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -176,14 +176,14 @@
ssize_t index = indexOfKey(address);
if (index < 0) {
- ALOGW("getInputForAttr() no policy for address %s", address.string());
+ ALOGW("getInputMixForAttr() no policy for address %s", address.string());
return BAD_VALUE;
}
sp<AudioPolicyMix> audioPolicyMix = valueAt(index);
AudioMix *mix = audioPolicyMix->getMix();
if (mix->mMixType != MIX_TYPE_PLAYERS) {
- ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
+ ALOGW("getInputMixForAttr() bad policy mix type for address %s", address.string());
return BAD_VALUE;
}
*policyMix = mix;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 7de72de..b7eed62 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -620,6 +620,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -659,8 +660,22 @@
return BAD_VALUE;
}
- ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
- attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
+ ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x"
+ " session %d selectedDeviceId %d",
+ attributes.usage, attributes.content_type, attributes.tags, attributes.flags,
+ session, selectedDeviceId);
+
+ *stream = streamTypefromAttributesInt(&attributes);
+
+ // Explicit routing?
+ sp<DeviceDescriptor> deviceDesc;
+ for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
+ if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) {
+ deviceDesc = mAvailableOutputDevices[i];
+ break;
+ }
+ }
+ mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc, uid);
routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
@@ -672,24 +687,14 @@
ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
device, samplingRate, format, channelMask, flags);
- *stream = streamTypefromAttributesInt(&attributes);
*output = getOutputForDevice(device, session, *stream,
samplingRate, format, channelMask,
flags, offloadInfo);
if (*output == AUDIO_IO_HANDLE_NONE) {
+ mOutputRoutes.removeRoute(session);
return INVALID_OPERATION;
}
- // Explicit routing?
- sp<DeviceDescriptor> deviceDesc;
-
- for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
- if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) {
- deviceDesc = mAvailableOutputDevices[i];
- break;
- }
- }
- mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc);
return NO_ERROR;
}
@@ -966,24 +971,26 @@
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ // Routing?
+ mOutputRoutes.incRouteActivity(session);
+
audio_devices_t newDevice;
if (outputDesc->mPolicyMix != NULL) {
newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
} else if (mOutputRoutes.hasRouteChanged(session)) {
newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ checkStrategyRoute(getStrategy(stream), output);
} else {
newDevice = AUDIO_DEVICE_NONE;
}
uint32_t delayMs = 0;
- // Routing?
- mOutputRoutes.incRouteActivity(session);
-
status_t status = startSource(outputDesc, stream, newDevice, &delayMs);
if (status != NO_ERROR) {
mOutputRoutes.decRouteActivity(session);
+ return status;
}
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
@@ -1112,15 +1119,22 @@
}
// Routing?
+ bool forceDeviceUpdate = false;
if (outputDesc->mRefCount[stream] > 0) {
- mOutputRoutes.decRouteActivity(session);
+ int activityCount = mOutputRoutes.decRouteActivity(session);
+ forceDeviceUpdate = (mOutputRoutes.hasRoute(session) && (activityCount == 0));
+
+ if (forceDeviceUpdate) {
+ checkStrategyRoute(getStrategy(stream), AUDIO_IO_HANDLE_NONE);
+ }
}
- return stopSource(outputDesc, stream);
+ return stopSource(outputDesc, stream, forceDeviceUpdate);
}
status_t AudioPolicyManager::stopSource(sp<AudioOutputDescriptor> outputDesc,
- audio_stream_type_t stream)
+ audio_stream_type_t stream,
+ bool forceDeviceUpdate)
{
// always handle stream stop, check which stream type is stopping
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
@@ -1135,7 +1149,7 @@
outputDesc->changeRefCount(stream, -1);
// store time at which the stream was stopped - see isStreamActive()
- if (outputDesc->mRefCount[stream] == 0) {
+ if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
outputDesc->mStopTime[stream] = systemTime();
audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
// delay the device switch by twice the latency because stopOutput() is executed when
@@ -1222,6 +1236,7 @@
status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -1256,7 +1271,7 @@
break;
}
}
- mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc);
+ mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc, uid);
if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
@@ -1431,17 +1446,17 @@
}
}
+ // Routing?
+ mInputRoutes.incRouteActivity(session);
+
if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
- && ((inputDesc->mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) {
+ && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
MIX_STATE_MIXING);
}
- // Routing?
- mInputRoutes.incRouteActivity(session);
-
if (mInputs.activeInputsCount() == 0) {
SoundTrigger::setCaptureState(true);
}
@@ -1501,7 +1516,7 @@
if (inputDesc->mRefCount == 0) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
- && ((inputDesc->mPolicyMix->mFlags & MIX_FLAG_NOTIFY_ACTIVITY) != 0)) {
+ && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
MIX_STATE_IDLE);
}
@@ -2479,6 +2494,12 @@
return status;
}
+void AudioPolicyManager::releaseResourcesForUid(uid_t uid)
+{
+ clearAudioPatches(uid);
+ clearSessionRoutes(uid);
+}
+
void AudioPolicyManager::clearAudioPatches(uid_t uid)
{
for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
@@ -2489,6 +2510,82 @@
}
}
+
+void AudioPolicyManager::checkStrategyRoute(routing_strategy strategy,
+ audio_io_handle_t ouptutToSkip)
+{
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ if (mOutputs.keyAt(j) == ouptutToSkip) {
+ continue;
+ }
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(j);
+ if (!isStrategyActive(outputDesc, (routing_strategy)strategy)) {
+ continue;
+ }
+ // If the default device for this strategy is on another output mix,
+ // invalidate all tracks in this strategy to force re connection.
+ // Otherwise select new device on the output mix.
+ if (outputs.indexOf(mOutputs.keyAt(j)) < 0) {
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)stream) == strategy) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)stream);
+ }
+ }
+ } else {
+ audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ setOutputDevice(outputDesc, newDevice, false);
+ }
+ }
+}
+
+void AudioPolicyManager::clearSessionRoutes(uid_t uid)
+{
+ // remove output routes associated with this uid
+ SortedVector<routing_strategy> affectedStrategies;
+ for (ssize_t i = (ssize_t)mOutputRoutes.size() - 1; i >= 0; i--) {
+ sp<SessionRoute> route = mOutputRoutes.valueAt(i);
+ if (route->mUid == uid) {
+ mOutputRoutes.removeItemsAt(i);
+ if (route->mDeviceDescriptor != 0) {
+ affectedStrategies.add(getStrategy(route->mStreamType));
+ }
+ }
+ }
+ // reroute outputs if necessary
+ for (size_t i = 0; i < affectedStrategies.size(); i++) {
+ checkStrategyRoute(affectedStrategies[i], AUDIO_IO_HANDLE_NONE);
+ }
+
+ // remove input routes associated with this uid
+ SortedVector<audio_source_t> affectedSources;
+ for (ssize_t i = (ssize_t)mInputRoutes.size() - 1; i >= 0; i--) {
+ sp<SessionRoute> route = mInputRoutes.valueAt(i);
+ if (route->mUid == uid) {
+ mInputRoutes.removeItemsAt(i);
+ if (route->mDeviceDescriptor != 0) {
+ affectedSources.add(route->mSource);
+ }
+ }
+ }
+ // reroute inputs if necessary
+ SortedVector<audio_io_handle_t> inputsToClose;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(i);
+ if (affectedSources.indexOf(inputDesc->mInputSource) >= 0) {
+ inputsToClose.add(inputDesc->mIoHandle);
+ }
+ }
+ for (size_t i = 0; i < inputsToClose.size(); i++) {
+ closeInput(inputsToClose[i]);
+ }
+}
+
+
status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
audio_devices_t *device)
@@ -3563,7 +3660,8 @@
ALOGVV("getOutputsForDevice() device %04x", device);
for (size_t i = 0; i < openOutputs.size(); i++) {
ALOGVV("output %d isDuplicated=%d device=%04x",
- i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+ i, openOutputs.valueAt(i)->isDuplicated(),
+ openOutputs.valueAt(i)->supportedDevices());
if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
@@ -3925,7 +4023,7 @@
for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex);
routing_strategy strat = getStrategy(route->mStreamType);
- if (strat == strategy && route->mDeviceDescriptor != 0 /*&& route->mActivityCount != 0*/) {
+ if (strat == strategy && route->isActive()) {
return route->mDeviceDescriptor->type();
}
}
@@ -4315,8 +4413,7 @@
{
for (size_t routeIndex = 0; routeIndex < mInputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mInputRoutes.valueAt(routeIndex);
- if (inputSource == route->mSource && route->mDeviceDescriptor != 0
- /*&& route->mActivityCount != 0*/) {
+ if (inputSource == route->mSource && route->isActive()) {
return route->mDeviceDescriptor->type();
}
}
@@ -4605,7 +4702,8 @@
void AudioPolicyManager::SessionRouteMap::addRoute(audio_session_t session,
audio_stream_type_t streamType,
audio_source_t source,
- sp<DeviceDescriptor> descriptor)
+ sp<DeviceDescriptor> descriptor,
+ uid_t uid)
{
if (mMapType == MAPTYPE_INPUT && streamType != SessionRoute::STREAM_TYPE_NA) {
ALOGE("Adding Output Route to InputRouteMap");
@@ -4618,14 +4716,15 @@
sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0;
if (route != 0) {
- if ((route->mDeviceDescriptor == 0 && descriptor != 0) ||
- (!route->mDeviceDescriptor->equals(descriptor))) {
+ if (((route->mDeviceDescriptor == 0) && (descriptor != 0)) ||
+ ((route->mDeviceDescriptor != 0) &&
+ ((descriptor == 0) || (!route->mDeviceDescriptor->equals(descriptor))))) {
route->mChanged = true;
}
route->mRefCount++;
route->mDeviceDescriptor = descriptor;
} else {
- route = new AudioPolicyManager::SessionRoute(session, streamType, source, descriptor);
+ route = new AudioPolicyManager::SessionRoute(session, streamType, source, descriptor, uid);
route->mRefCount++;
add(session, route);
if (descriptor != 0) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index b965411..ea16864 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -109,6 +109,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -127,6 +128,7 @@
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -207,7 +209,6 @@
struct audio_patch *patches,
unsigned int *generation);
virtual status_t setAudioPortConfig(const struct audio_port_config *config);
- virtual void clearAudioPatches(uid_t uid);
virtual status_t acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
@@ -226,6 +227,8 @@
audio_io_handle_t *handle);
virtual status_t stopAudioSource(audio_io_handle_t handle);
+ virtual void releaseResourcesForUid(uid_t uid);
+
// Audio policy configuration file parsing (audio_policy.conf)
// TODO candidates to be moved to ConfigParsingUtils
void defaultAudioPolicyConfig(void);
@@ -248,31 +251,36 @@
SessionRoute(audio_session_t session,
audio_stream_type_t streamType,
audio_source_t source,
- sp<DeviceDescriptor> deviceDescriptor)
- : mSession(session),
+ sp<DeviceDescriptor> deviceDescriptor,
+ uid_t uid)
+ : mUid(uid),
+ mSession(session),
mDeviceDescriptor(deviceDescriptor),
mRefCount(0),
mActivityCount(0),
mChanged(false),
mStreamType(streamType),
- mSource(source) {}
-
- audio_session_t mSession;
-
- sp<DeviceDescriptor> mDeviceDescriptor;
+ mSource(source)
+ {}
void log(const char* prefix);
+ bool isActive() {
+ return (mDeviceDescriptor != 0) && (mChanged || (mActivityCount > 0));
+ }
+
+ uid_t mUid;
+ audio_session_t mSession;
+ sp<DeviceDescriptor> mDeviceDescriptor;
+
// "reference" counting
- int mRefCount; // +/- on references
- int mActivityCount; // +/- on start/stop
- bool mChanged;
-
+ int mRefCount; // +/- on references
+ int mActivityCount; // +/- on start/stop
+ bool mChanged;
// for outputs
- const audio_stream_type_t mStreamType;
-
+ const audio_stream_type_t mStreamType;
// for inputs
- const audio_source_t mSource;
+ const audio_source_t mSource;
};
class SessionRouteMap: public KeyedVector<audio_session_t, sp<SessionRoute>> {
@@ -292,6 +300,7 @@
}
bool hasRoute(audio_session_t session);
+
void removeRoute(audio_session_t session);
int incRouteActivity(audio_session_t session);
@@ -306,7 +315,8 @@
void addRoute(audio_session_t session,
audio_stream_type_t streamType,
audio_source_t source,
- sp<DeviceDescriptor> deviceDescriptor);
+ sp<DeviceDescriptor> deviceDescriptor,
+ uid_t uid);
private:
// Used to mark a SessionRoute as for either inputs (mMapType == kSessionRouteMap_Input)
@@ -559,7 +569,12 @@
audio_devices_t device,
uint32_t *delayMs);
status_t stopSource(sp<AudioOutputDescriptor> outputDesc,
- audio_stream_type_t stream);
+ audio_stream_type_t stream,
+ bool forceDeviceUpdate);
+
+ void clearAudioPatches(uid_t uid);
+ void clearSessionRoutes(uid_t uid);
+ void checkStrategyRoute(routing_strategy strategy, audio_io_handle_t ouptutToSkip);
uid_t mUidCached;
AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index e6ace20..282ddeb 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -109,8 +109,8 @@
Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
for (size_t i = 0; i < effects.size(); i++) {
EffectDesc *effect = effects[i];
- sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0,
- audioSession, input);
+ sp<AudioEffect> fx = new AudioEffect(NULL, String16("android"), &effect->mUuid, -1, 0,
+ 0, audioSession, input);
status_t status = fx->initCheck();
if (status != NO_ERROR && status != ALREADY_EXISTS) {
ALOGW("addInputEffects(): failed to create Fx %s on source %d",
@@ -254,7 +254,7 @@
Vector <EffectDesc *> effects = mOutputStreams.valueAt(index)->mEffects;
for (size_t i = 0; i < effects.size(); i++) {
EffectDesc *effect = effects[i];
- sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, 0, 0, 0,
+ sp<AudioEffect> fx = new AudioEffect(NULL, String16("android"), &effect->mUuid, 0, 0, 0,
audioSession, output);
status_t status = fx->initCheck();
if (status != NO_ERROR && status != ALREADY_EXISTS) {
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 5ceb1cf..65639c3 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -146,6 +146,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -158,7 +159,16 @@
}
ALOGV("getOutput()");
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, samplingRate,
+
+ // if the caller is us, trust the specified uid
+ if (IPCThreadState::self()->getCallingPid() != getpid_cached || uid == (uid_t)-1) {
+ uid_t newclientUid = IPCThreadState::self()->getCallingUid();
+ if (uid != (uid_t)-1 && uid != newclientUid) {
+ ALOGW("%s uid %d tried to pass itself off as %d", __FUNCTION__, newclientUid, uid);
+ }
+ uid = newclientUid;
+ }
+ return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, uid, samplingRate,
format, channelMask, flags, selectedDeviceId, offloadInfo);
}
@@ -248,6 +258,7 @@
status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -269,12 +280,22 @@
sp<AudioPolicyEffects>audioPolicyEffects;
status_t status;
AudioPolicyInterface::input_type_t inputType;
+ // if the caller is us, trust the specified uid
+ if (IPCThreadState::self()->getCallingPid() != getpid_cached || uid == (uid_t)-1) {
+ uid_t newclientUid = IPCThreadState::self()->getCallingUid();
+ if (uid != (uid_t)-1 && uid != newclientUid) {
+ ALOGW("%s uid %d tried to pass itself off as %d", __FUNCTION__, newclientUid, uid);
+ }
+ uid = newclientUid;
+ }
+
{
Mutex::Autolock _l(mLock);
// the audio_in_acoustics_t parameter is ignored by get_input()
- status = mAudioPolicyManager->getInputForAttr(attr, input, session,
+ status = mAudioPolicyManager->getInputForAttr(attr, input, session, uid,
samplingRate, format, channelMask,
- flags, selectedDeviceId, &inputType);
+ flags, selectedDeviceId,
+ &inputType);
audioPolicyEffects = mAudioPolicyEffects;
if (status == NO_ERROR) {
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
index 433e712..13af3ef 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
@@ -234,6 +234,7 @@
status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid __unused,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -565,6 +566,7 @@
audio_io_handle_t *output,
audio_session_t session __unused,
audio_stream_type_t *stream,
+ uid_t uid __unused,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index ccf9f9b..c5f4fb7 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -177,7 +177,7 @@
{
Mutex::Autolock _l(mLock);
if (mAudioPolicyManager) {
- mAudioPolicyManager->clearAudioPatches(uid);
+ mAudioPolicyManager->releaseResourcesForUid(uid);
}
}
#endif
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 07ea96b..eb50cdd 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -80,6 +80,7 @@
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
+ uid_t uid,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = 0,
@@ -98,6 +99,7 @@
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
+ uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 8c5c43a..e28464d 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -890,9 +890,12 @@
if (current != nullptr) {
auto clientSp = current->getValue();
if (clientSp.get() != nullptr) { // should never be needed
- if (clientSp->getRemote() == remoteCallback) {
+ if (!clientSp->canCastToApiClient(effectiveApiLevel)) {
+ ALOGW("CameraService connect called from same client, but with a different"
+ " API level, evicting prior client...");
+ } else if (clientSp->getRemote() == remoteCallback) {
ALOGI("CameraService::connect X (PID %d) (second call from same"
- "app binder, returning the same client)", clientPid);
+ " app binder, returning the same client)", clientPid);
*client = clientSp;
return NO_ERROR;
}
@@ -1754,6 +1757,11 @@
return mClientPid;
}
+bool CameraService::BasicClient::canCastToApiClient(apiLevel level) const {
+ // Defaults to API2.
+ return level == API_2;
+}
+
status_t CameraService::BasicClient::startCameraOps() {
int32_t res;
// Notify app ops that the camera is not available
@@ -1866,6 +1874,10 @@
BasicClient::disconnect();
}
+bool CameraService::Client::canCastToApiClient(apiLevel level) const {
+ return level == API_1;
+}
+
CameraService::Client::OpsCallback::OpsCallback(wp<BasicClient> client):
mClient(client) {
}
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 84e61c5..a8b4c4a 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -65,6 +65,7 @@
class Client;
class BasicClient;
+ // The effective API level. The Camera2 API running in LEGACY mode counts as API_1.
enum apiLevel {
API_1 = 1,
API_2 = 2
@@ -215,6 +216,10 @@
// Get the PID of the application client using this
virtual int getClientPid() const;
+
+ // Check what API level is used for this client. This is used to determine which
+ // superclass this can be cast to.
+ virtual bool canCastToApiClient(apiLevel level) const;
protected:
BasicClient(const sp<CameraService>& cameraService,
const sp<IBinder>& remoteCallback,
@@ -323,6 +328,10 @@
virtual void notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
const CaptureResultExtras& resultExtras);
+
+ // Check what API level is used for this client. This is used to determine which
+ // superclass this can be cast to.
+ virtual bool canCastToApiClient(apiLevel level) const;
protected:
// Convert client from cookie.
static sp<CameraService::Client> getClientFromCookie(void* user);
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 6b0f8b5..c3a6842 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -2100,12 +2100,7 @@
delete[] reqMeteringAreas;
- /* don't include jpeg thumbnail size - it's valid for
- it to be set to (0,0), meaning 'no thumbnail' */
- CropRegion crop = calculateCropRegion( (CropRegion::Outputs)(
- CropRegion::OUTPUT_PREVIEW |
- CropRegion::OUTPUT_VIDEO |
- CropRegion::OUTPUT_PICTURE ));
+ CropRegion crop = calculateCropRegion(/*previewOnly*/ false);
int32_t reqCropRegion[4] = {
static_cast<int32_t>(crop.left),
static_cast<int32_t>(crop.top),
@@ -2603,7 +2598,7 @@
ALOG_ASSERT(x >= 0, "Crop-relative X coordinate = '%d' is out of bounds"
"(lower = 0)", x);
- CropRegion previewCrop = calculateCropRegion(CropRegion::OUTPUT_PREVIEW);
+ CropRegion previewCrop = calculateCropRegion(/*previewOnly*/ true);
ALOG_ASSERT(x < previewCrop.width, "Crop-relative X coordinate = '%d' "
"is out of bounds (upper = %f)", x, previewCrop.width);
@@ -2619,7 +2614,7 @@
ALOG_ASSERT(y >= 0, "Crop-relative Y coordinate = '%d' is out of bounds "
"(lower = 0)", y);
- CropRegion previewCrop = calculateCropRegion(CropRegion::OUTPUT_PREVIEW);
+ CropRegion previewCrop = calculateCropRegion(/*previewOnly*/ true);
ALOG_ASSERT(y < previewCrop.height, "Crop-relative Y coordinate = '%d' is "
"out of bounds (upper = %f)", y, previewCrop.height);
@@ -2634,12 +2629,12 @@
}
int Parameters::normalizedXToCrop(int x) const {
- CropRegion previewCrop = calculateCropRegion(CropRegion::OUTPUT_PREVIEW);
+ CropRegion previewCrop = calculateCropRegion(/*previewOnly*/ true);
return (x + 1000) * (previewCrop.width - 1) / 2000;
}
int Parameters::normalizedYToCrop(int y) const {
- CropRegion previewCrop = calculateCropRegion(CropRegion::OUTPUT_PREVIEW);
+ CropRegion previewCrop = calculateCropRegion(/*previewOnly*/ true);
return (y + 1000) * (previewCrop.height - 1) / 2000;
}
@@ -2855,8 +2850,7 @@
return jpegSizes;
}
-Parameters::CropRegion Parameters::calculateCropRegion(
- Parameters::CropRegion::Outputs outputs) const {
+Parameters::CropRegion Parameters::calculateCropRegion(bool previewOnly) const {
float zoomLeft, zoomTop, zoomWidth, zoomHeight;
@@ -2880,90 +2874,45 @@
maxDigitalZoom.data.f[0], zoomIncrement, zoomRatio, previewWidth,
previewHeight, fastInfo.arrayWidth, fastInfo.arrayHeight);
- /*
- * Assumption: On the HAL side each stream buffer calculates its crop
- * rectangle as follows:
- * cropRect = (zoomLeft, zoomRight,
- * zoomWidth, zoomHeight * zoomWidth / outputWidth);
- *
- * Note that if zoomWidth > bufferWidth, the new cropHeight > zoomHeight
- * (we can then get into trouble if the cropHeight > arrayHeight).
- * By selecting the zoomRatio based on the smallest outputRatio, we
- * guarantee this will never happen.
- */
+ if (previewOnly) {
+ // Calculate a tight crop region for the preview stream only
+ float previewRatio = static_cast<float>(previewWidth) / previewHeight;
- // Enumerate all possible output sizes, select the one with the smallest
- // aspect ratio
- float minOutputWidth, minOutputHeight, minOutputRatio;
- {
- float outputSizes[][2] = {
- { static_cast<float>(previewWidth),
- static_cast<float>(previewHeight) },
- { static_cast<float>(videoWidth),
- static_cast<float>(videoHeight) },
- { static_cast<float>(jpegThumbSize[0]),
- static_cast<float>(jpegThumbSize[1]) },
- { static_cast<float>(pictureWidth),
- static_cast<float>(pictureHeight) },
- };
+ /* Ensure that the width/height never go out of bounds
+ * by scaling across a diffent dimension if an out-of-bounds
+ * possibility exists.
+ *
+ * e.g. if the previewratio < arrayratio and e.g. zoomratio = 1.0, then by
+ * calculating the zoomWidth from zoomHeight we'll actually get a
+ * zoomheight > arrayheight
+ */
+ float arrayRatio = 1.f * fastInfo.arrayWidth / fastInfo.arrayHeight;
+ if (previewRatio >= arrayRatio) {
+ // Adjust the height based on the width
+ zoomWidth = fastInfo.arrayWidth / zoomRatio;
+ zoomHeight = zoomWidth *
+ previewHeight / previewWidth;
- minOutputWidth = outputSizes[0][0];
- minOutputHeight = outputSizes[0][1];
- minOutputRatio = minOutputWidth / minOutputHeight;
- for (unsigned int i = 0;
- i < sizeof(outputSizes) / sizeof(outputSizes[0]);
- ++i) {
-
- // skip over outputs we don't want to consider for the crop region
- if ( !((1 << i) & outputs) ) {
- continue;
- }
-
- float outputWidth = outputSizes[i][0];
- float outputHeight = outputSizes[i][1];
- float outputRatio = outputWidth / outputHeight;
-
- if (minOutputRatio > outputRatio) {
- minOutputRatio = outputRatio;
- minOutputWidth = outputWidth;
- minOutputHeight = outputHeight;
- }
-
- // and then use this output ratio instead of preview output ratio
- ALOGV("Enumerating output ratio %f = %f / %f, min is %f",
- outputRatio, outputWidth, outputHeight, minOutputRatio);
+ } else {
+ // Adjust the width based on the height
+ zoomHeight = fastInfo.arrayHeight / zoomRatio;
+ zoomWidth = zoomHeight *
+ previewWidth / previewHeight;
}
- }
-
- /* Ensure that the width/height never go out of bounds
- * by scaling across a diffent dimension if an out-of-bounds
- * possibility exists.
- *
- * e.g. if the previewratio < arrayratio and e.g. zoomratio = 1.0, then by
- * calculating the zoomWidth from zoomHeight we'll actually get a
- * zoomheight > arrayheight
- */
- float arrayRatio = 1.f * fastInfo.arrayWidth / fastInfo.arrayHeight;
- if (minOutputRatio >= arrayRatio) {
- // Adjust the height based on the width
- zoomWidth = fastInfo.arrayWidth / zoomRatio;
- zoomHeight = zoomWidth *
- minOutputHeight / minOutputWidth;
-
} else {
- // Adjust the width based on the height
+ // Calculate the global crop region with a shape matching the active
+ // array.
+ zoomWidth = fastInfo.arrayWidth / zoomRatio;
zoomHeight = fastInfo.arrayHeight / zoomRatio;
- zoomWidth = zoomHeight *
- minOutputWidth / minOutputHeight;
}
- // centering the zoom area within the active area
+
+ // center the zoom area within the active area
zoomLeft = (fastInfo.arrayWidth - zoomWidth) / 2;
zoomTop = (fastInfo.arrayHeight - zoomHeight) / 2;
ALOGV("Crop region calculated (x=%d,y=%d,w=%f,h=%f) for zoom=%d",
(int32_t)zoomLeft, (int32_t)zoomTop, zoomWidth, zoomHeight, this->zoom);
-
CropRegion crop = { zoomLeft, zoomTop, zoomWidth, zoomHeight };
return crop;
}
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index e628a7e..46d48bc 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -271,21 +271,16 @@
// if video snapshot size is currently overridden
bool isJpegSizeOverridden();
- // Calculate the crop region rectangle based on current stream sizes
+ // Calculate the crop region rectangle, either tightly about the preview
+ // resolution, or a region just based on the active array; both take
+ // into account the current zoom level.
struct CropRegion {
float left;
float top;
float width;
float height;
-
- enum Outputs {
- OUTPUT_PREVIEW = 0x01,
- OUTPUT_VIDEO = 0x02,
- OUTPUT_JPEG_THUMBNAIL = 0x04,
- OUTPUT_PICTURE = 0x08,
- };
};
- CropRegion calculateCropRegion(CropRegion::Outputs outputs) const;
+ CropRegion calculateCropRegion(bool previewOnly) const;
// Calculate the field of view of the high-resolution JPEG capture
status_t calculatePictureFovs(float *horizFov, float *vertFov) const;