Merge "Fix HEVC profile idc parsing in HevcParameterSets" into pi-dev
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index b4fa3c5..ca119d5 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -60,6 +60,8 @@
volatile int32_t mRear; // written by producer (output: client, input: server)
volatile int32_t mFlush; // incremented by client to indicate a request to flush;
// server notices and discards all data between mFront and mRear
+ volatile int32_t mStop; // set by client to indicate a stop frame position; server
+ // will not read beyond this position until start is called.
volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame
volatile uint32_t mUnderrunCount; // server increments for each underrun occurrence
};
@@ -335,6 +337,8 @@
mTimestamp.clear();
}
+ virtual void stop() { }; // called by client in AudioTrack::stop()
+
private:
// This is a copy of mCblk->mBufferSizeInFrames
uint32_t mBufferSizeInFrames; // effective size of the buffer
@@ -383,8 +387,14 @@
mPlaybackRateMutator.push(playbackRate);
}
+ // Sends flush and stop position information from the client to the server,
+ // used by streaming AudioTrack flush() or stop().
+ void sendStreamingFlushStop(bool flush);
+
virtual void flush();
+ void stop() override;
+
virtual uint32_t getUnderrunFrames() const {
return mCblk->u.mStreaming.mUnderrunFrames;
}
@@ -410,6 +420,8 @@
virtual void flush();
+ void stop() override;
+
#define MIN_LOOP 16 // minimum length of each loop iteration in frames
// setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the
@@ -532,6 +544,10 @@
// client will be notified via Futex
virtual void flushBufferIfNeeded();
+ // Returns the rear position of the AudioTrack shared ring buffer, limited by
+ // the stop frame position level.
+ virtual int32_t getRear() const = 0;
+
// Total count of the number of flushed frames since creation (never reset).
virtual int64_t framesFlushed() const { return mFlushed; }
@@ -607,10 +623,18 @@
return mDrained.load();
}
+ int32_t getRear() const override;
+
+ // Called on server side track start().
+ virtual void start();
+
private:
AudioPlaybackRate mPlaybackRate; // last observed playback rate
PlaybackRateQueue::Observer mPlaybackRateObserver;
+ // Last client stop-at position when start() was called. Used for streaming AudioTracks.
+ std::atomic<int32_t> mStopLast{0};
+
// The server keeps a copy here where it is safe from the client.
uint32_t mUnderrunCount; // echoed to mCblk
bool mUnderrunning; // used to detect edge of underrun
@@ -634,6 +658,10 @@
virtual void tallyUnderrunFrames(uint32_t frameCount);
virtual uint32_t getUnderrunFrames() const { return 0; }
+ int32_t getRear() const override;
+
+ void start() override { } // ignore for static tracks
+
private:
status_t updateStateWithLoop(StaticAudioTrackState *localState,
const StaticAudioTrackState &update) const;
@@ -661,6 +689,10 @@
size_t frameSize, bool clientInServer)
: ServerProxy(cblk, buffers, frameCount, frameSize, false /*isOut*/, clientInServer) { }
+ int32_t getRear() const override {
+ return mCblk->u.mStreaming.mRear; // For completeness only; mRear written by server.
+ }
+
protected:
virtual ~AudioRecordServerProxy() { }
};
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index ac2e46e..86791c2 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -770,6 +770,7 @@
mReleased = 0;
}
+ mProxy->stop(); // notify server not to read beyond current client position until start().
mProxy->interrupt();
mAudioTrack->stop();
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index 7bf4f99..b4c179d 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -393,19 +393,50 @@
// ---------------------------------------------------------------------------
-__attribute__((no_sanitize("integer")))
void AudioTrackClientProxy::flush()
{
+ sendStreamingFlushStop(true /* flush */);
+}
+
+void AudioTrackClientProxy::stop()
+{
+ sendStreamingFlushStop(false /* flush */);
+}
+
+// Sets the client-written mFlush and mStop positions, which control server behavior.
+//
+// @param flush indicates whether the operation is a flush or stop.
+// A client stop sets mStop to the current write position;
+// the server will not read past this point until start() or subsequent flush().
+// A client flush sets both mStop and mFlush to the current write position.
+// This advances the server read limit (if previously set) and on the next
+// server read advances the server read position to this limit.
+//
+void AudioTrackClientProxy::sendStreamingFlushStop(bool flush)
+{
+ // TODO: Replace this by 64 bit counters - avoids wrap complication.
// This works for mFrameCountP2 <= 2^30
- size_t increment = mFrameCountP2 << 1;
- size_t mask = increment - 1;
- audio_track_cblk_t* cblk = mCblk;
// mFlush is 32 bits concatenated as [ flush_counter ] [ newfront_offset ]
// Should newFlush = cblk->u.mStreaming.mRear? Only problem is
// if you want to flush twice to the same rear location after a 32 bit wrap.
- int32_t newFlush = (cblk->u.mStreaming.mRear & mask) |
- ((cblk->u.mStreaming.mFlush & ~mask) + increment);
- android_atomic_release_store(newFlush, &cblk->u.mStreaming.mFlush);
+
+ const size_t increment = mFrameCountP2 << 1;
+ const size_t mask = increment - 1;
+ // No need for client atomic synchronization on mRear, mStop, mFlush
+ // as AudioTrack client only read/writes to them under client lock. Server only reads.
+ const int32_t rearMasked = mCblk->u.mStreaming.mRear & mask;
+
+ // update stop before flush so that the server front
+ // never advances beyond a (potential) previous stop's rear limit.
+ int32_t stopBits; // the following add can overflow
+ __builtin_add_overflow(mCblk->u.mStreaming.mStop & ~mask, increment, &stopBits);
+ android_atomic_release_store(rearMasked | stopBits, &mCblk->u.mStreaming.mStop);
+
+ if (flush) {
+ int32_t flushBits; // the following add can overflow
+ __builtin_add_overflow(mCblk->u.mStreaming.mFlush & ~mask, increment, &flushBits);
+ android_atomic_release_store(rearMasked | flushBits, &mCblk->u.mStreaming.mFlush);
+ }
}
bool AudioTrackClientProxy::clearStreamEndDone() {
@@ -540,6 +571,11 @@
LOG_ALWAYS_FATAL("static flush");
}
+void StaticAudioTrackClientProxy::stop()
+{
+ ; // no special handling required for static tracks.
+}
+
void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
{
// This can only happen on a 64-bit client
@@ -638,6 +674,7 @@
if (flush != mFlush) {
ALOGV("ServerProxy::flushBufferIfNeeded() mStreaming.mFlush = 0x%x, mFlush = 0x%0x",
flush, mFlush);
+ // shouldn't matter, but for range safety use mRear instead of getRear().
int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
int32_t front = cblk->u.mStreaming.mFront;
@@ -677,6 +714,45 @@
}
__attribute__((no_sanitize("integer")))
+int32_t AudioTrackServerProxy::getRear() const
+{
+ const int32_t stop = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop);
+ const int32_t rear = android_atomic_acquire_load(&mCblk->u.mStreaming.mRear);
+ const int32_t stopLast = mStopLast.load(std::memory_order_acquire);
+ if (stop != stopLast) {
+ const int32_t front = mCblk->u.mStreaming.mFront;
+ const size_t overflowBit = mFrameCountP2 << 1;
+ const size_t mask = overflowBit - 1;
+ int32_t newRear = (rear & ~mask) | (stop & mask);
+ ssize_t filled = newRear - front;
+ if (filled < 0) {
+ // front and rear offsets span the overflow bit of the p2 mask
+ // so rebasing newrear.
+ ALOGV("stop wrap: filled %zx >= overflowBit %zx", filled, overflowBit);
+ newRear += overflowBit;
+ filled += overflowBit;
+ }
+ if (0 <= filled && (size_t) filled <= mFrameCount) {
+ // we're stopped, return the stop level as newRear
+ return newRear;
+ }
+
+ // A corrupt stop. Log error and ignore.
+ ALOGE("mStopLast %#x -> stop %#x, front %#x, rear %#x, mask %#x, newRear %#x, "
+ "filled %zd=%#x",
+ stopLast, stop, front, rear,
+ (unsigned)mask, newRear, filled, (unsigned)filled);
+ // Don't reset mStopLast as this is const.
+ }
+ return rear;
+}
+
+void AudioTrackServerProxy::start()
+{
+ mStopLast = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop);
+}
+
+__attribute__((no_sanitize("integer")))
status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
{
LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0,
@@ -693,7 +769,7 @@
// See notes on barriers at ClientProxy::obtainBuffer()
if (mIsOut) {
flushBufferIfNeeded(); // might modify mFront
- rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ rear = getRear();
front = cblk->u.mStreaming.mFront;
} else {
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
@@ -825,8 +901,7 @@
// FIXME should return an accurate value, but over-estimate is better than under-estimate
return mFrameCount;
}
- // the acquire might not be necessary since not doing a subsequent read
- int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ const int32_t rear = getRear();
ssize_t filled = rear - cblk->u.mStreaming.mFront;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
@@ -852,7 +927,7 @@
if (flush != mFlush) {
return mFrameCount;
}
- const int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+ const int32_t rear = getRear();
const ssize_t filled = rear - cblk->u.mStreaming.mFront;
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
return 0; // error condition, silently return 0.
@@ -1149,6 +1224,12 @@
}
}
+int32_t StaticAudioTrackServerProxy::getRear() const
+{
+ LOG_ALWAYS_FATAL("getRear() not permitted for static tracks");
+ return 0;
+}
+
// ---------------------------------------------------------------------------
} // namespace android
diff --git a/media/libmedia/include/media/mediametadataretriever.h b/media/libmedia/include/media/mediametadataretriever.h
index 3511253..b41da80 100644
--- a/media/libmedia/include/media/mediametadataretriever.h
+++ b/media/libmedia/include/media/mediametadataretriever.h
@@ -66,6 +66,8 @@
METADATA_KEY_IMAGE_HEIGHT = 30,
METADATA_KEY_IMAGE_ROTATION = 31,
METADATA_KEY_VIDEO_FRAME_COUNT = 32,
+ METADATA_KEY_EXIF_OFFSET = 33,
+ METADATA_KEY_EXIF_LENGTH = 34,
// Add more here...
};
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index cc7f688..a762e76 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1617,14 +1617,7 @@
// internal buffer before resuming playback.
// FIXME: this is ignored after flush().
mAudioSink->stop();
- if (mPaused) {
- // Race condition: if renderer is paused and audio sink is stopped,
- // we need to make sure that the audio track buffer fully drains
- // before delivering data.
- // FIXME: remove this if we can detect if stop() is complete.
- const int delayUs = 2 * 50 * 1000; // (2 full mixer thread cycles at 50ms)
- mPauseDrainAudioAllowedUs = ALooper::GetNowUs() + delayUs;
- } else {
+ if (!mPaused) {
mAudioSink->start();
}
mNumFramesWritten = 0;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index a8c6d15..3bbba49 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -3291,6 +3291,22 @@
return err;
}
+ if (compressionFormat == OMX_VIDEO_CodingHEVC) {
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ // verify if Main10 profile is supported at all, and fail
+ // immediately if it's not supported.
+ if (profile == OMX_VIDEO_HEVCProfileMain10 ||
+ profile == OMX_VIDEO_HEVCProfileMain10HDR10) {
+ err = verifySupportForProfileAndLevel(
+ kPortIndexInput, profile, 0);
+ if (err != OK) {
+ return err;
+ }
+ }
+ }
+ }
+
if (compressionFormat == OMX_VIDEO_CodingVP9) {
OMX_VIDEO_PARAM_PROFILELEVELTYPE params;
InitOMXParams(¶ms);
@@ -4059,7 +4075,7 @@
return INVALID_OPERATION;
}
- err = verifySupportForProfileAndLevel(profile, level);
+ err = verifySupportForProfileAndLevel(kPortIndexOutput, profile, level);
if (err != OK) {
return err;
@@ -4131,7 +4147,7 @@
return INVALID_OPERATION;
}
- err = verifySupportForProfileAndLevel(profile, level);
+ err = verifySupportForProfileAndLevel(kPortIndexOutput, profile, level);
if (err != OK) {
return err;
@@ -4266,7 +4282,7 @@
return INVALID_OPERATION;
}
- err = verifySupportForProfileAndLevel(profile, level);
+ err = verifySupportForProfileAndLevel(kPortIndexOutput, profile, level);
if (err != OK) {
return err;
@@ -4280,7 +4296,7 @@
// Use largest supported profile for AVC recording if profile is not specified.
for (OMX_VIDEO_AVCPROFILETYPE profile : {
OMX_VIDEO_AVCProfileHigh, OMX_VIDEO_AVCProfileMain }) {
- if (verifySupportForProfileAndLevel(profile, 0) == OK) {
+ if (verifySupportForProfileAndLevel(kPortIndexOutput, profile, 0) == OK) {
h264type.eProfile = profile;
break;
}
@@ -4457,7 +4473,7 @@
return INVALID_OPERATION;
}
- err = verifySupportForProfileAndLevel(profile, level);
+ err = verifySupportForProfileAndLevel(kPortIndexOutput, profile, level);
if (err != OK) {
return err;
}
@@ -4602,10 +4618,10 @@
}
status_t ACodec::verifySupportForProfileAndLevel(
- int32_t profile, int32_t level) {
+ OMX_U32 portIndex, int32_t profile, int32_t level) {
OMX_VIDEO_PARAM_PROFILELEVELTYPE params;
InitOMXParams(¶ms);
- params.nPortIndex = kPortIndexOutput;
+ params.nPortIndex = portIndex;
for (OMX_U32 index = 0; index <= kMaxIndicesToCheck; ++index) {
params.nProfileIndex = index;
@@ -4906,8 +4922,8 @@
rect.nHeight = videoDef->nFrameHeight;
}
- if (rect.nLeft < 0 ||
- rect.nTop < 0 ||
+ if (rect.nLeft < 0 || rect.nTop < 0 ||
+ rect.nWidth == 0 || rect.nHeight == 0 ||
rect.nLeft + rect.nWidth > videoDef->nFrameWidth ||
rect.nTop + rect.nHeight > videoDef->nFrameHeight) {
ALOGE("Wrong cropped rect (%d, %d, %u, %u) vs. frame (%u, %u)",
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index b874df4..f25d1f1 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -82,7 +82,6 @@
// NB: These are not yet exposed as public Java API constants.
static const char *kCodecCrypto = "android.media.mediacodec.crypto"; /* 0,1 */
-static const char *kCodecBytesIn = "android.media.mediacodec.bytesin"; /* 0..n */
static const char *kCodecProfile = "android.media.mediacodec.profile"; /* 0..n */
static const char *kCodecLevel = "android.media.mediacodec.level"; /* 0..n */
static const char *kCodecMaxWidth = "android.media.mediacodec.maxwidth"; /* 0..n */
@@ -3202,10 +3201,6 @@
info->mData.clear();
statsBufferSent(timeUs);
-
- if (mAnalyticsItem != NULL) {
- mAnalyticsItem->addInt64(kCodecBytesIn, size);
- }
}
return err;
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 179e0e6..5ae5644 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -438,6 +438,15 @@
mMetaData.add(METADATA_KEY_CAPTURE_FRAMERATE, String8(tmp));
}
+ int64_t exifOffset, exifSize;
+ if (meta->findInt64(kKeyExifOffset, &exifOffset)
+ && meta->findInt64(kKeyExifSize, &exifSize)) {
+ sprintf(tmp, "%lld", (long long)exifOffset);
+ mMetaData.add(METADATA_KEY_EXIF_OFFSET, String8(tmp));
+ sprintf(tmp, "%lld", (long long)exifSize);
+ mMetaData.add(METADATA_KEY_EXIF_LENGTH, String8(tmp));
+ }
+
bool hasAudio = false;
bool hasVideo = false;
int32_t videoWidth = -1;
diff --git a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
index 13b6d05..2c0f224 100644
--- a/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
+++ b/media/libstagefright/codecs/flac/dec/SoftFlacDecoder.cpp
@@ -302,7 +302,7 @@
List<BufferInfo *> &outQueue = getPortQueue(1);
ALOGV("onQueueFilled %d/%d:", inQueue.empty(), outQueue.empty());
- while ((!inQueue.empty() || mSawInputEOS) && !outQueue.empty()) {
+ while ((!inQueue.empty() || mSawInputEOS) && !outQueue.empty() && !mFinishedDecoder) {
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
short *outBuffer = reinterpret_cast<short *>(outHeader->pBuffer + outHeader->nOffset);
@@ -318,6 +318,21 @@
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
ALOGV("saw EOS");
mSawInputEOS = true;
+ if (mInputBufferCount == 0 && inHeader->nFilledLen == 0) {
+ // first buffer was empty and EOS: signal EOS on output and return
+ ALOGV("empty first EOS");
+ outHeader->nFilledLen = 0;
+ outHeader->nTimeStamp = inHeader->nTimeStamp;
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ outInfo->mOwnedByUs = false;
+ outQueue.erase(outQueue.begin());
+ notifyFillBufferDone(outHeader);
+ mFinishedDecoder = true;
+ inInfo->mOwnedByUs = false;
+ inQueue.erase(inQueue.begin());
+ notifyEmptyBufferDone(inHeader);
+ return;
+ }
}
if (mInputBufferCount == 0 && !(inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
@@ -377,7 +392,7 @@
ALOGV("no output, trying again");
continue;
}
- } else if (mSawInputEOS && !mFinishedDecoder) {
+ } else if (mSawInputEOS) {
status_t decoderErr = mFLACDecoder->decodeOneFrame(NULL, 0, outBuffer, &outBufferSize);
mFinishedDecoder = true;
if (decoderErr != OK) {
@@ -388,10 +403,8 @@
}
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
} else {
- ALOGE("no input buffer but did not get EOS");
- mSignalledError = true;
- notify(OMX_EventError, OMX_ErrorStreamCorrupt, 0, NULL);
- return;
+ // no more input buffers at this time, loop and see if there is more output
+ continue;
}
outHeader->nFilledLen = outBufferSize;
@@ -412,9 +425,12 @@
void SoftFlacDecoder::drainDecoder() {
mFLACDecoder->flush();
+ mSawInputEOS = false;
+ mFinishedDecoder = false;
}
void SoftFlacDecoder::onReset() {
+ ALOGV("onReset");
drainDecoder();
memset(&mStreamInfo, 0, sizeof(mStreamInfo));
diff --git a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
index a0e46c3..fdc8975 100644
--- a/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
+++ b/media/libstagefright/codecs/flac/enc/SoftFlacEncoder.cpp
@@ -357,7 +357,7 @@
FLAC__bool ok = true;
- while ((!inQueue.empty() || mSawInputEOS) && !outQueue.empty()) {
+ while ((!inQueue.empty() || mSawInputEOS) && !outQueue.empty() && !mSentOutputEOS) {
if (!inQueue.empty()) {
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
@@ -415,7 +415,7 @@
mEncoderReturnedEncodedData = false;
} else {
ALOGV(" encoder process_interleaved returned without data to write");
- if (mSawInputEOS && !mSentOutputEOS) {
+ if (mSawInputEOS) {
ALOGV("finishing encoder");
mSentOutputEOS = true;
FLAC__stream_encoder_finish(mFlacStreamEncoder);
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index eae73fc..1b38852 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -818,7 +818,8 @@
uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ dst.mCropTop * dst.mWidth + dst.mCropLeft;
- const uint8_t *src_y = (const uint8_t *)src.mBits;
+ const uint8_t *src_y =
+ (const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
const uint8_t *src_u =
(const uint8_t *)src_y + src.mWidth * (src.mHeight - src.mCropTop / 2);
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 1a5304b..64caeed 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -493,7 +493,8 @@
status_t setupHEVCEncoderParameters(const sp<AMessage> &msg, sp<AMessage> &outputFormat);
status_t setupVPXEncoderParameters(const sp<AMessage> &msg, sp<AMessage> &outputFormat);
- status_t verifySupportForProfileAndLevel(int32_t profile, int32_t level);
+ status_t verifySupportForProfileAndLevel(
+ OMX_U32 portIndex, int32_t profile, int32_t level);
status_t configureImageGrid(const sp<AMessage> &msg, sp<AMessage> &outputFormat);
status_t configureBitrate(
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b5b50f8..160aa45 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2631,6 +2631,10 @@
StreamOutHalInterface::SourceMetadata metadata;
for (const sp<Track> &track : mActiveTracks) {
// No track is invalid as this is called after prepareTrack_l in the same critical section
+ if (track->isOutputTrack()) {
+ // TODO: OutputTrack (used for duplication) are currently not supported
+ continue;
+ }
metadata.tracks.push_back({
.usage = track->attributes().usage,
.content_type = track->attributes().content_type,
@@ -6151,13 +6155,12 @@
void AudioFlinger::DuplicatingThread::updateMetadata_l()
{
- // TODO: The duplicated track metadata are stored in other threads
- // (accessible through mActiveTracks::OutputTrack::thread()::mActiveTracks::Track::attributes())
- // but this information can be mutated at any time by the owning threads.
- // Taking the lock of any other owning threads is no possible due to timing constrains.
- // Similarly, the other threads can not push the metadatas in this thread as cross deadlock
- // would be possible.
- // A lock-free structure needs to be used to shared the metadata (maybe an atomic shared_ptr ?).
+ // TODO: The duplicated track metadata needs to be pushed to downstream
+ // but this information can be read at any time by the downstream threads.
+ // Taking the lock of any downstream threads is no possible due to cross deadlock risks
+ // (eg: during effect move).
+ // A lock-free structure needs to be used to shared the metadata, probably an atomic
+ // pointer to a metadata vector in each output tracks.
}
uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 44ce3aa..236412b 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -764,6 +764,12 @@
mState = state;
}
}
+
+ if (status == NO_ERROR || status == ALREADY_EXISTS) {
+ // for streaming tracks, remove the buffer read stop limit.
+ mAudioTrackServerProxy->start();
+ }
+
// track was already in the active list, not a problem
if (status == ALREADY_EXISTS) {
status = NO_ERROR;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index 094ff65..d85562e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -391,6 +391,7 @@
mSamplingRate = 0;
mChannelMask = AUDIO_CHANNEL_NONE;
mFormat = AUDIO_FORMAT_INVALID;
+ memset(&mGain, 0, sizeof(struct audio_gain_config));
mGain.index = -1;
}