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-rw-r--r--media/libstagefright/AudioPlayer.cpp400
1 files changed, 320 insertions, 80 deletions
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 4208019..05ee34e 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -17,6 +17,7 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioPlayer"
#include <utils/Log.h>
+#include <cutils/compiler.h>
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
@@ -27,6 +28,7 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
#include "include/AwesomePlayer.h"
@@ -34,10 +36,9 @@ namespace android {
AudioPlayer::AudioPlayer(
const sp<MediaPlayerBase::AudioSink> &audioSink,
- bool allowDeepBuffering,
+ uint32_t flags,
AwesomePlayer *observer)
- : mAudioTrack(NULL),
- mInputBuffer(NULL),
+ : mInputBuffer(NULL),
mSampleRate(0),
mLatencyUs(0),
mFrameSize(0),
@@ -48,14 +49,17 @@ AudioPlayer::AudioPlayer(
mSeeking(false),
mReachedEOS(false),
mFinalStatus(OK),
+ mSeekTimeUs(0),
mStarted(false),
mIsFirstBuffer(false),
mFirstBufferResult(OK),
mFirstBuffer(NULL),
mAudioSink(audioSink),
- mAllowDeepBuffering(allowDeepBuffering),
mObserver(observer),
- mPinnedTimeUs(-1ll) {
+ mPinnedTimeUs(-1ll),
+ mPlaying(false),
+ mStartPosUs(0),
+ mCreateFlags(flags) {
}
AudioPlayer::~AudioPlayer() {
@@ -110,7 +114,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
const char *mime;
bool success = format->findCString(kKeyMIMEType, &mime);
CHECK(success);
- CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
+ CHECK(useOffload() || !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
success = format->findInt32(kKeySampleRate, &mSampleRate);
CHECK(success);
@@ -126,16 +130,74 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
}
+ audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
+
+ if (useOffload()) {
+ if (mapMimeToAudioFormat(audioFormat, mime) != OK) {
+ ALOGE("Couldn't map mime type \"%s\" to a valid AudioSystem::audio_format", mime);
+ audioFormat = AUDIO_FORMAT_INVALID;
+ } else {
+ ALOGV("Mime type \"%s\" mapped to audio_format 0x%x", mime, audioFormat);
+ }
+ }
+
+ int avgBitRate = -1;
+ format->findInt32(kKeyBitRate, &avgBitRate);
+
if (mAudioSink.get() != NULL) {
+ uint32_t flags = AUDIO_OUTPUT_FLAG_NONE;
+ audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+
+ if (allowDeepBuffering()) {
+ flags |= AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ if (useOffload()) {
+ flags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
+
+ int64_t durationUs;
+ if (format->findInt64(kKeyDuration, &durationUs)) {
+ offloadInfo.duration_us = durationUs;
+ } else {
+ offloadInfo.duration_us = -1;
+ }
+
+ offloadInfo.sample_rate = mSampleRate;
+ offloadInfo.channel_mask = channelMask;
+ offloadInfo.format = audioFormat;
+ offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
+ offloadInfo.bit_rate = avgBitRate;
+ offloadInfo.has_video = ((mCreateFlags & HAS_VIDEO) != 0);
+ offloadInfo.is_streaming = ((mCreateFlags & IS_STREAMING) != 0);
+ }
+
status_t err = mAudioSink->open(
- mSampleRate, numChannels, channelMask, AUDIO_FORMAT_PCM_16_BIT,
+ mSampleRate, numChannels, channelMask, audioFormat,
DEFAULT_AUDIOSINK_BUFFERCOUNT,
&AudioPlayer::AudioSinkCallback,
this,
- (mAllowDeepBuffering ?
- AUDIO_OUTPUT_FLAG_DEEP_BUFFER :
- AUDIO_OUTPUT_FLAG_NONE));
+ (audio_output_flags_t)flags,
+ useOffload() ? &offloadInfo : NULL);
+
+ if (err == OK) {
+ mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
+ mFrameSize = mAudioSink->frameSize();
+
+ if (useOffload()) {
+ // 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 will be going
+ // through the AudioFlinger mixer before reaching the hardware
+ sendMetaDataToHal(mAudioSink, format);
+ }
+
+ err = mAudioSink->start();
+ // do not alter behavior for non offloaded tracks: ignore start status.
+ if (!useOffload()) {
+ err = OK;
+ }
+ }
+
if (err != OK) {
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
@@ -149,10 +211,6 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
return err;
}
- mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
- mFrameSize = mAudioSink->frameSize();
-
- mAudioSink->start();
} else {
// playing to an AudioTrack, set up mask if necessary
audio_channel_mask_t audioMask = channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER ?
@@ -166,8 +224,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
- delete mAudioTrack;
- mAudioTrack = NULL;
+ mAudioTrack.clear();
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
@@ -188,6 +245,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
}
mStarted = true;
+ mPlaying = true;
mPinnedTimeUs = -1ll;
return OK;
@@ -214,29 +272,57 @@ void AudioPlayer::pause(bool playPendingSamples) {
mPinnedTimeUs = ALooper::GetNowUs();
}
+
+ mPlaying = false;
}
-void AudioPlayer::resume() {
+status_t AudioPlayer::resume() {
CHECK(mStarted);
+ status_t err;
if (mAudioSink.get() != NULL) {
- mAudioSink->start();
+ err = mAudioSink->start();
} else {
- mAudioTrack->start();
+ err = mAudioTrack->start();
}
+
+ if (err == OK) {
+ mPlaying = true;
+ }
+
+ return err;
}
void AudioPlayer::reset() {
CHECK(mStarted);
+ ALOGV("reset: mPlaying=%d mReachedEOS=%d useOffload=%d",
+ mPlaying, mReachedEOS, useOffload() );
+
if (mAudioSink.get() != NULL) {
mAudioSink->stop();
+ // If we're closing and have reached EOS, we don't want to flush
+ // the track because if it is offloaded there could be a small
+ // amount of residual data in the hardware buffer which we must
+ // play to give gapless playback.
+ // But if we're resetting when paused or before we've reached EOS
+ // we can't be doing a gapless playback and there could be a large
+ // amount of data queued in the hardware if the track is offloaded,
+ // so we must flush to prevent a track switch being delayed playing
+ // the buffered data that we don't want now
+ if (!mPlaying || !mReachedEOS) {
+ mAudioSink->flush();
+ }
+
mAudioSink->close();
} else {
mAudioTrack->stop();
- delete mAudioTrack;
- mAudioTrack = NULL;
+ if (!mPlaying || !mReachedEOS) {
+ mAudioTrack->flush();
+ }
+
+ mAudioTrack.clear();
}
// Make sure to release any buffer we hold onto so that the
@@ -259,10 +345,16 @@ void AudioPlayer::reset() {
// The following hack is necessary to ensure that the OMX
// component is completely released by the time we may try
// to instantiate it again.
- wp<MediaSource> tmp = mSource;
- mSource.clear();
- while (tmp.promote() != NULL) {
- usleep(1000);
+ // When offloading, the OMX component is not used so this hack
+ // is not needed
+ if (!useOffload()) {
+ wp<MediaSource> tmp = mSource;
+ mSource.clear();
+ while (tmp.promote() != NULL) {
+ usleep(1000);
+ }
+ } else {
+ mSource.clear();
}
IPCThreadState::self()->flushCommands();
@@ -271,9 +363,12 @@ void AudioPlayer::reset() {
mPositionTimeMediaUs = -1;
mPositionTimeRealUs = -1;
mSeeking = false;
+ mSeekTimeUs = 0;
mReachedEOS = false;
mFinalStatus = OK;
mStarted = false;
+ mPlaying = false;
+ mStartPosUs = 0;
}
// static
@@ -294,10 +389,19 @@ bool AudioPlayer::reachedEOS(status_t *finalStatus) {
return mReachedEOS;
}
+void AudioPlayer::notifyAudioEOS() {
+ ALOGV("AudioPlayer@0x%p notifyAudioEOS", this);
+
+ if (mObserver != NULL) {
+ mObserver->postAudioEOS(0);
+ ALOGV("Notified observer of EOS!");
+ }
+}
+
status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
if (mAudioSink.get() != NULL) {
return mAudioSink->setPlaybackRatePermille(ratePermille);
- } else if (mAudioTrack != NULL){
+ } else if (mAudioTrack != 0){
return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
} else {
return NO_INIT;
@@ -307,21 +411,44 @@ status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
// static
size_t AudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *buffer, size_t size, void *cookie) {
+ void *buffer, size_t size, void *cookie,
+ MediaPlayerBase::AudioSink::cb_event_t event) {
AudioPlayer *me = (AudioPlayer *)cookie;
- return me->fillBuffer(buffer, size);
-}
+ switch(event) {
+ case MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER:
+ return me->fillBuffer(buffer, size);
-void AudioPlayer::AudioCallback(int event, void *info) {
- if (event != AudioTrack::EVENT_MORE_DATA) {
- return;
+ case MediaPlayerBase::AudioSink::CB_EVENT_STREAM_END:
+ ALOGV("AudioSinkCallback: stream end");
+ me->mReachedEOS = true;
+ me->notifyAudioEOS();
+ break;
+
+ case MediaPlayerBase::AudioSink::CB_EVENT_TEAR_DOWN:
+ ALOGV("AudioSinkCallback: Tear down event");
+ me->mObserver->postAudioTearDown();
+ break;
}
- AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
- size_t numBytesWritten = fillBuffer(buffer->raw, buffer->size);
+ return 0;
+}
- buffer->size = numBytesWritten;
+void AudioPlayer::AudioCallback(int event, void *info) {
+ switch (event) {
+ case AudioTrack::EVENT_MORE_DATA:
+ {
+ AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
+ size_t numBytesWritten = fillBuffer(buffer->raw, buffer->size);
+ buffer->size = numBytesWritten;
+ }
+ break;
+
+ case AudioTrack::EVENT_STREAM_END:
+ mReachedEOS = true;
+ notifyAudioEOS();
+ break;
+ }
}
uint32_t AudioPlayer::getNumFramesPendingPlayout() const {
@@ -361,6 +488,7 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
size_t size_remaining = size;
while (size_remaining > 0) {
MediaSource::ReadOptions options;
+ bool refreshSeekTime = false;
{
Mutex::Autolock autoLock(mLock);
@@ -375,6 +503,7 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
}
options.setSeekTo(mSeekTimeUs);
+ refreshSeekTime = true;
if (mInputBuffer != NULL) {
mInputBuffer->release();
@@ -407,43 +536,56 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
Mutex::Autolock autoLock(mLock);
if (err != OK) {
- if (mObserver && !mReachedEOS) {
- // We don't want to post EOS right away but only
- // after all frames have actually been played out.
-
- // These are the number of frames submitted to the
- // AudioTrack that you haven't heard yet.
- uint32_t numFramesPendingPlayout =
- getNumFramesPendingPlayout();
-
- // These are the number of frames we're going to
- // submit to the AudioTrack by returning from this
- // callback.
- uint32_t numAdditionalFrames = size_done / mFrameSize;
-
- numFramesPendingPlayout += numAdditionalFrames;
-
- int64_t timeToCompletionUs =
- (1000000ll * numFramesPendingPlayout) / mSampleRate;
-
- ALOGV("total number of frames played: %lld (%lld us)",
- (mNumFramesPlayed + numAdditionalFrames),
- 1000000ll * (mNumFramesPlayed + numAdditionalFrames)
- / mSampleRate);
-
- ALOGV("%d frames left to play, %lld us (%.2f secs)",
- numFramesPendingPlayout,
- timeToCompletionUs, timeToCompletionUs / 1E6);
-
- postEOS = true;
- if (mAudioSink->needsTrailingPadding()) {
- postEOSDelayUs = timeToCompletionUs + mLatencyUs;
+ if (!mReachedEOS) {
+ if (useOffload()) {
+ // no more buffers to push - stop() and wait for STREAM_END
+ // don't set mReachedEOS until stream end received
+ if (mAudioSink != NULL) {
+ mAudioSink->stop();
+ } else {
+ mAudioTrack->stop();
+ }
} else {
- postEOSDelayUs = 0;
+ if (mObserver) {
+ // We don't want to post EOS right away but only
+ // after all frames have actually been played out.
+
+ // These are the number of frames submitted to the
+ // AudioTrack that you haven't heard yet.
+ uint32_t numFramesPendingPlayout =
+ getNumFramesPendingPlayout();
+
+ // These are the number of frames we're going to
+ // submit to the AudioTrack by returning from this
+ // callback.
+ uint32_t numAdditionalFrames = size_done / mFrameSize;
+
+ numFramesPendingPlayout += numAdditionalFrames;
+
+ int64_t timeToCompletionUs =
+ (1000000ll * numFramesPendingPlayout) / mSampleRate;
+
+ ALOGV("total number of frames played: %lld (%lld us)",
+ (mNumFramesPlayed + numAdditionalFrames),
+ 1000000ll * (mNumFramesPlayed + numAdditionalFrames)
+ / mSampleRate);
+
+ ALOGV("%d frames left to play, %lld us (%.2f secs)",
+ numFramesPendingPlayout,
+ timeToCompletionUs, timeToCompletionUs / 1E6);
+
+ postEOS = true;
+ if (mAudioSink->needsTrailingPadding()) {
+ postEOSDelayUs = timeToCompletionUs + mLatencyUs;
+ } else {
+ postEOSDelayUs = 0;
+ }
+ }
+
+ mReachedEOS = true;
}
}
- mReachedEOS = true;
mFinalStatus = err;
break;
}
@@ -454,17 +596,43 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
}
- CHECK(mInputBuffer->meta_data()->findInt64(
+ if(mInputBuffer->range_length() != 0) {
+ CHECK(mInputBuffer->meta_data()->findInt64(
kKeyTime, &mPositionTimeMediaUs));
+ }
- mPositionTimeRealUs =
- ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
- / mSampleRate;
+ // need to adjust the mStartPosUs for offload decoding since parser
+ // might not be able to get the exact seek time requested.
+ if (refreshSeekTime) {
+ if (useOffload()) {
+ if (postSeekComplete) {
+ ALOGV("fillBuffer is going to post SEEK_COMPLETE");
+ mObserver->postAudioSeekComplete();
+ postSeekComplete = false;
+ }
+
+ mStartPosUs = mPositionTimeMediaUs;
+ ALOGV("adjust seek time to: %.2f", mStartPosUs/ 1E6);
+ }
+ // clear seek time with mLock locked and once we have valid mPositionTimeMediaUs
+ // and mPositionTimeRealUs
+ // before clearing mSeekTimeUs check if a new seek request has been received while
+ // we were reading from the source with mLock released.
+ if (!mSeeking) {
+ mSeekTimeUs = 0;
+ }
+ }
+
+ if (!useOffload()) {
+ mPositionTimeRealUs =
+ ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
+ / mSampleRate;
+ ALOGV("buffer->size() = %d, "
+ "mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
+ mInputBuffer->range_length(),
+ mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
+ }
- ALOGV("buffer->size() = %d, "
- "mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
- mInputBuffer->range_length(),
- mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
}
if (mInputBuffer->range_length() == 0) {
@@ -490,6 +658,13 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
size_remaining -= copy;
}
+ if (useOffload()) {
+ // We must ask the hardware what it has played
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ ALOGV("mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
+ mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
+ }
+
{
Mutex::Autolock autoLock(mLock);
mNumFramesPlayed += size_done / mFrameSize;
@@ -515,6 +690,14 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
int64_t AudioPlayer::getRealTimeUs() {
Mutex::Autolock autoLock(mLock);
+ if (useOffload()) {
+ if (mSeeking) {
+ return mSeekTimeUs;
+ }
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ return mPositionTimeRealUs;
+ }
+
return getRealTimeUsLocked();
}
@@ -538,15 +721,51 @@ int64_t AudioPlayer::getRealTimeUsLocked() const {
return result + diffUs;
}
+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;
+ }
+
+ 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;
+ ALOGV("getOutputPlayPositionUs_l %lld", renderedDuration);
+ return renderedDuration;
+}
+
int64_t AudioPlayer::getMediaTimeUs() {
Mutex::Autolock autoLock(mLock);
- if (mPositionTimeMediaUs < 0 || mPositionTimeRealUs < 0) {
+ if (useOffload()) {
if (mSeeking) {
return mSeekTimeUs;
}
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ ALOGV("getMediaTimeUs getOutputPlayPositionUs_l() mPositionTimeRealUs %lld",
+ mPositionTimeRealUs);
+ return mPositionTimeRealUs;
+ }
- return 0;
+
+ if (mPositionTimeMediaUs < 0 || mPositionTimeRealUs < 0) {
+ // mSeekTimeUs is either seek time while seeking or 0 if playback did not start.
+ return mSeekTimeUs;
}
int64_t realTimeOffset = getRealTimeUsLocked() - mPositionTimeRealUs;
@@ -561,8 +780,14 @@ bool AudioPlayer::getMediaTimeMapping(
int64_t *realtime_us, int64_t *mediatime_us) {
Mutex::Autolock autoLock(mLock);
- *realtime_us = mPositionTimeRealUs;
- *mediatime_us = mPositionTimeMediaUs;
+ if (useOffload()) {
+ mPositionTimeRealUs = getOutputPlayPositionUs_l();
+ *realtime_us = mPositionTimeRealUs;
+ *mediatime_us = mPositionTimeRealUs;
+ } else {
+ *realtime_us = mPositionTimeRealUs;
+ *mediatime_us = mPositionTimeMediaUs;
+ }
return mPositionTimeRealUs != -1 && mPositionTimeMediaUs != -1;
}
@@ -570,19 +795,34 @@ bool AudioPlayer::getMediaTimeMapping(
status_t AudioPlayer::seekTo(int64_t time_us) {
Mutex::Autolock autoLock(mLock);
+ ALOGV("seekTo( %lld )", time_us);
+
mSeeking = true;
mPositionTimeRealUs = mPositionTimeMediaUs = -1;
mReachedEOS = false;
mSeekTimeUs = time_us;
+ mStartPosUs = time_us;
// Flush resets the number of played frames
mNumFramesPlayed = 0;
mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();
if (mAudioSink != NULL) {
+ if (mPlaying) {
+ mAudioSink->pause();
+ }
mAudioSink->flush();
+ if (mPlaying) {
+ mAudioSink->start();
+ }
} else {
+ if (mPlaying) {
+ mAudioTrack->pause();
+ }
mAudioTrack->flush();
+ if (mPlaying) {
+ mAudioTrack->start();
+ }
}
return OK;
generated by cgit v1.1 at 2024-05-10 21:46:11 +0000