Merge changes Id28b35fd,Ie4e64977,I2950f31e into klp-dev
* changes:
DO NOT MERGE: Camera: fix focusArea wrong indexing issue
DO NOT MERGE: camera2: Fix race with stream deletion during disconnect.
DO NOT MERGE: camera2/3: Add protection for still capture path
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f379ee5..f6646ab 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -661,7 +661,7 @@
sp<AudioTrackThread> mAudioTrackThread;
float mVolume[2];
float mSendLevel;
- uint32_t mSampleRate;
+ mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
size_t mFrameCount; // corresponds to current IAudioTrack
size_t mReqFrameCount; // frame count to request the next time a new
// IAudioTrack is needed
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index cc244f0..26d8729 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -100,6 +100,7 @@
virtual status_t getFramesWritten(uint32_t *frameswritten) const = 0;
virtual int getSessionId() const = 0;
virtual audio_stream_type_t getAudioStreamType() const = 0;
+ virtual uint32_t getSampleRate() const = 0;
// If no callback is specified, use the "write" API below to submit
// audio data.
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 912a43c..14afb85 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -129,7 +129,7 @@
void reset();
uint32_t getNumFramesPendingPlayout() const;
- int64_t getOutputPlayPositionUs_l() const;
+ int64_t getOutputPlayPositionUs_l();
bool allowDeepBuffering() const { return (mCreateFlags & ALLOW_DEEP_BUFFERING) != 0; }
bool useOffload() const { return (mCreateFlags & USE_OFFLOAD) != 0; }
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index de3fc36..3a87474 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -134,6 +134,7 @@
kKeyRequiresSecureBuffers = 'secu', // bool (int32_t)
kKeyIsADTS = 'adts', // bool (int32_t)
+ kKeyAACAOT = 'aaot', // int32_t
// If a MediaBuffer's data represents (at least partially) encrypted
// data, the following fields aid in decryption.
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 5aeba4f..8d656c4 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -585,4 +585,11 @@
return mSessionId;
}
+uint32_t VideoEditorPlayer::VeAudioOutput::getSampleRate() const {
+ if (mMsecsPerFrame == 0) {
+ return 0;
+ }
+ return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
} // namespace android
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index 5862c08..b8c1254 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -48,6 +48,7 @@
virtual status_t getPosition(uint32_t *position) const;
virtual status_t getFramesWritten(uint32_t*) const;
virtual int getSessionId() const;
+ virtual uint32_t getSampleRate() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 666fafa..ccbc5a3 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -545,13 +545,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7c4a990..11b0b89 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -603,6 +603,19 @@
}
AutoMutex lock(mLock);
+
+ // sample rate can be updated during playback by the offloaded decoder so we need to
+ // query the HAL and update if needed.
+// FIXME use Proxy return channel to update the rate from server and avoid polling here
+ if (isOffloaded()) {
+ if (mOutput != 0) {
+ uint32_t sampleRate = 0;
+ status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate);
+ if (status == NO_ERROR) {
+ mSampleRate = sampleRate;
+ }
+ }
+ }
return mSampleRate;
}
@@ -866,7 +879,8 @@
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
- // n = 1 fast track; nBuffering is ignored
+ // n = 1 fast track with single buffering; nBuffering is ignored
+ // n = 2 fast track with double buffering
// n = 2 normal track, no sample rate conversion
// n = 3 normal track, with sample rate conversion
// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
@@ -1006,9 +1020,11 @@
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
if (sharedBuffer == 0) {
- // double-buffering is not required for fast tracks, due to tighter scheduling
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) {
- mNotificationFramesAct = frameCount;
+ // Theoretically double-buffering is not required for fast tracks,
+ // due to tighter scheduling. But in practice, to accommodate kernels with
+ // scheduling jitter, and apps with computation jitter, we use double-buffering.
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+ mNotificationFramesAct = frameCount/nBuffering;
}
}
} else {
@@ -1091,13 +1107,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index cd052e6..9ac9105 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1813,6 +1813,12 @@
return mSessionId;
}
+uint32_t MediaPlayerService::AudioOutput::getSampleRate() const
+{
+ if (mTrack == 0) return 0;
+ return mTrack->getSampleRate();
+}
+
#undef LOG_TAG
#define LOG_TAG "AudioCache"
MediaPlayerService::AudioCache::AudioCache(const sp<IMemoryHeap>& heap) :
@@ -2015,6 +2021,14 @@
return 0;
}
+uint32_t MediaPlayerService::AudioCache::getSampleRate() const
+{
+ if (mMsecsPerFrame == 0) {
+ return 0;
+ }
+ return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
void MediaPlayerService::addBatteryData(uint32_t params)
{
Mutex::Autolock lock(mLock);
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index a486cb5..9c084e1 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -86,6 +86,7 @@
virtual status_t getPosition(uint32_t *position) const;
virtual status_t getFramesWritten(uint32_t *frameswritten) const;
virtual int getSessionId() const;
+ virtual uint32_t getSampleRate() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -195,6 +196,7 @@
virtual status_t getPosition(uint32_t *position) const;
virtual status_t getFramesWritten(uint32_t *frameswritten) const;
virtual int getSessionId() const;
+ virtual uint32_t getSampleRate() const;
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index a8a8786..05ee34e 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -721,16 +721,27 @@
return result + diffUs;
}
-int64_t AudioPlayer::getOutputPlayPositionUs_l() const
+int64_t AudioPlayer::getOutputPlayPositionUs_l()
{
uint32_t playedSamples = 0;
+ uint32_t sampleRate;
if (mAudioSink != NULL) {
mAudioSink->getPosition(&playedSamples);
+ sampleRate = mAudioSink->getSampleRate();
} else {
mAudioTrack->getPosition(&playedSamples);
+ sampleRate = mAudioTrack->getSampleRate();
+ }
+ if (sampleRate != 0) {
+ mSampleRate = sampleRate;
}
- const int64_t playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+ int64_t playedUs;
+ if (mSampleRate != 0) {
+ playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+ } else {
+ playedUs = 0;
+ }
// HAL position is relative to the first buffer we sent at mStartPosUs
const int64_t renderedDuration = mStartPosUs + playedUs;
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 1ba1c6e..491b4d1 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2285,6 +2285,11 @@
return ERROR_MALFORMED;
}
+ static uint32_t kSamplingRate[] = {
+ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350
+ };
+
ABitReader br(csd, csd_size);
uint32_t objectType = br.getBits(5);
@@ -2292,6 +2297,9 @@
objectType = 32 + br.getBits(6);
}
+ //keep AOT type
+ mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
+
uint32_t freqIndex = br.getBits(4);
int32_t sampleRate = 0;
@@ -2304,29 +2312,30 @@
numChannels = br.getBits(4);
} else {
numChannels = br.getBits(4);
- if (objectType == 5) {
- // SBR specific config per 14496-3 table 1.13
- freqIndex = br.getBits(4);
- if (freqIndex == 15) {
- if (csd_size < 8) {
- return ERROR_MALFORMED;
- }
- sampleRate = br.getBits(24);
- }
+
+ if (freqIndex == 13 || freqIndex == 14) {
+ return ERROR_MALFORMED;
}
- if (sampleRate == 0) {
- static uint32_t kSamplingRate[] = {
- 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
- 16000, 12000, 11025, 8000, 7350
- };
+ sampleRate = kSamplingRate[freqIndex];
+ }
- if (freqIndex == 13 || freqIndex == 14) {
+ if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+ uint32_t extFreqIndex = br.getBits(4);
+ int32_t extSampleRate;
+ if (extFreqIndex == 15) {
+ if (csd_size < 8) {
return ERROR_MALFORMED;
}
-
- sampleRate = kSamplingRate[freqIndex];
+ extSampleRate = br.getBits(24);
+ } else {
+ if (extFreqIndex == 13 || extFreqIndex == 14) {
+ return ERROR_MALFORMED;
+ }
+ extSampleRate = kSamplingRate[extFreqIndex];
}
+ //TODO: save the extension sampling rate value in meta data =>
+ // mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
}
if (numChannels == 0) {
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 1a9a26b..dedd186 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -217,6 +217,7 @@
for (;;) {
int64_t now_us = 0;
sp<Event> event;
+ bool wakeLocked = false;
{
Mutex::Autolock autoLock(mLock);
@@ -283,26 +284,28 @@
// removeEventFromQueue_l will return NULL.
// Otherwise, the QueueItem will be removed
// from the queue and the referenced event returned.
- event = removeEventFromQueue_l(eventID);
+ event = removeEventFromQueue_l(eventID, &wakeLocked);
}
if (event != NULL) {
// Fire event with the lock NOT held.
event->fire(this, now_us);
+ if (wakeLocked) {
+ Mutex::Autolock autoLock(mLock);
+ releaseWakeLock_l();
+ }
}
}
}
sp<TimedEventQueue::Event> TimedEventQueue::removeEventFromQueue_l(
- event_id id) {
+ event_id id, bool *wakeLocked) {
for (List<QueueItem>::iterator it = mQueue.begin();
it != mQueue.end(); ++it) {
if ((*it).event->eventID() == id) {
sp<Event> event = (*it).event;
event->setEventID(0);
- if ((*it).has_wakelock) {
- releaseWakeLock_l();
- }
+ *wakeLocked = (*it).has_wakelock;
mQueue.erase(it);
return event;
}
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 9041c21..216a329 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -562,6 +562,17 @@
return false;
}
+ // check whether it is ELD/LD content -> no offloading
+ // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
+ // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+ int32_t aacaot = -1;
+ if (meta->findInt32(kKeyAACAOT, &aacaot)) {
+ if (aacaot == 23 || aacaot == 39 ) {
+ ALOGV("track of type '%s' is ELD/LD content", mime);
+ return false;
+ }
+ }
+
int32_t srate = -1;
if (!meta->findInt32(kKeySampleRate, &srate)) {
ALOGV("track of type '%s' does not publish sample rate", mime);
diff --git a/media/libstagefright/include/TimedEventQueue.h b/media/libstagefright/include/TimedEventQueue.h
index 38a08b1..3e84256 100644
--- a/media/libstagefright/include/TimedEventQueue.h
+++ b/media/libstagefright/include/TimedEventQueue.h
@@ -145,7 +145,7 @@
static void *ThreadWrapper(void *me);
void threadEntry();
- sp<Event> removeEventFromQueue_l(event_id id);
+ sp<Event> removeEventFromQueue_l(event_id id, bool *wakeLocked);
void acquireWakeLock_l();
void releaseWakeLock_l(bool force = false);
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..eeb3700 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@
uint8_t *ptr = dup->data();
*ptr++ = 0xff;
- *ptr++ = 0xf1; // b11110001, ID=0, layer=0, protection_absent=1
+ *ptr++ = 0xf9; // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
*ptr++ =
profile << 6
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index df4e029..07dc6dd 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1122,10 +1122,6 @@
t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
- // t.in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t.in == NULL)
- enabledTracks &= ~(1<<i);
}
e0 = enabledTracks;
@@ -1161,6 +1157,13 @@
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
+ // t.in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t.in == NULL) {
+ enabledTracks &= ~(1<<i);
+ e1 &= ~(1<<i);
+ break;
+ }
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames) {
t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 110e45c..3d657b3 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -135,12 +135,12 @@
// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
// for the track. The client then sub-divides this into smaller buffers for its use.
-// Currently the client uses double-buffering by default, but doesn't tell us about that.
-// So for now we just assume that client is double-buffered.
-// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
-// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// Currently the client uses N-buffering by default, but doesn't tell us about the value of N.
+// So for now we just assume that client is double-buffered for fast tracks.
+// FIXME It would be better for client to tell AudioFlinger the value of N,
+// so AudioFlinger could allocate the right amount of memory.
// See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 1;
+static const int kFastTrackMultiplier = 2;
// ----------------------------------------------------------------------------
@@ -1210,7 +1210,7 @@
(
(tid != -1) &&
((frameCount == 0) ||
- (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+ (frameCount >= mFrameCount))
)
) &&
// PCM data
@@ -1915,7 +1915,7 @@
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
- size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t);
+ size_t offset = (mCurrentWriteLength - mBytesRemaining);
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
@@ -1926,7 +1926,7 @@
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->stream->write(mOutput->stream,
- mMixBuffer + offset, mBytesRemaining);
+ (char *)mMixBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -3038,15 +3038,8 @@
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
minFrames = desiredFrames;
}
- // It's not safe to call framesReady() for a static buffer track, so assume it's ready
- size_t framesReady;
- if (track->sharedBuffer() == 0) {
- framesReady = track->framesReady();
- } else if (track->isStopped()) {
- framesReady = 0;
- } else {
- framesReady = 1;
- }
+
+ size_t framesReady = track->framesReady();
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
@@ -4694,7 +4687,7 @@
(
(tid != -1) &&
((frameCount == 0) ||
- (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+ (frameCount >= mFrameCount))
) &&
// FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format)
// mono or stereo