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/*
* Copyright (C) 2009 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioPlayer"
#include <utils/Log.h>
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
#include "include/AwesomePlayer.h"
namespace android {
AudioPlayer::AudioPlayer(
const sp<MediaPlayerBase::AudioSink> &audioSink,
AwesomePlayer *observer)
: mAudioTrack(NULL),
mInputBuffer(NULL),
mSampleRate(0),
mLatencyUs(0),
mFrameSize(0),
mNumFramesPlayed(0),
mPositionTimeMediaUs(-1),
mPositionTimeRealUs(-1),
mSeeking(false),
mReachedEOS(false),
mFinalStatus(OK),
mStarted(false),
mIsFirstBuffer(false),
mFirstBufferResult(OK),
mFirstBuffer(NULL),
mAudioSink(audioSink),
mObserver(observer) {
}
AudioPlayer::~AudioPlayer() {
if (mStarted) {
reset();
}
}
void AudioPlayer::setSource(const sp<MediaSource> &source) {
CHECK(mSource == NULL);
mSource = source;
}
status_t AudioPlayer::start(bool sourceAlreadyStarted) {
CHECK(!mStarted);
CHECK(mSource != NULL);
status_t err;
if (!sourceAlreadyStarted) {
err = mSource->start();
if (err != OK) {
return err;
}
}
// We allow an optional INFO_FORMAT_CHANGED at the very beginning
// of playback, if there is one, getFormat below will retrieve the
// updated format, if there isn't, we'll stash away the valid buffer
// of data to be used on the first audio callback.
CHECK(mFirstBuffer == NULL);
MediaSource::ReadOptions options;
if (mSeeking) {
options.setSeekTo(mSeekTimeUs);
mSeeking = false;
}
mFirstBufferResult = mSource->read(&mFirstBuffer, &options);
if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
ALOGV("INFO_FORMAT_CHANGED!!!");
CHECK(mFirstBuffer == NULL);
mFirstBufferResult = OK;
mIsFirstBuffer = false;
} else {
mIsFirstBuffer = true;
}
sp<MetaData> format = mSource->getFormat();
const char *mime;
bool success = format->findCString(kKeyMIMEType, &mime);
CHECK(success);
CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));
success = format->findInt32(kKeySampleRate, &mSampleRate);
CHECK(success);
int32_t numChannels, channelMask;
success = format->findInt32(kKeyChannelCount, &numChannels);
CHECK(success);
if(!format->findInt32(kKeyChannelMask, &channelMask)) {
ALOGW("source format didn't specify channel mask, using channel order");
channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
}
if (mAudioSink.get() != NULL) {
status_t err = mAudioSink->open(
mSampleRate, numChannels, channelMask, AUDIO_FORMAT_PCM_16_BIT,
DEFAULT_AUDIOSINK_BUFFERCOUNT,
&AudioPlayer::AudioSinkCallback, this);
if (err != OK) {
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
mFirstBuffer = NULL;
}
if (!sourceAlreadyStarted) {
mSource->stop();
}
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 ?
audio_channel_mask_from_count(numChannels) : channelMask;
if (0 == audioMask) {
return BAD_VALUE;
}
mAudioTrack = new AudioTrack(
AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
0, 0, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
delete mAudioTrack;
mAudioTrack = NULL;
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
mFirstBuffer = NULL;
}
if (!sourceAlreadyStarted) {
mSource->stop();
}
return err;
}
mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
mFrameSize = mAudioTrack->frameSize();
mAudioTrack->start();
}
mStarted = true;
return OK;
}
void AudioPlayer::pause(bool playPendingSamples) {
CHECK(mStarted);
if (playPendingSamples) {
if (mAudioSink.get() != NULL) {
mAudioSink->stop();
} else {
mAudioTrack->stop();
}
mNumFramesPlayed = 0;
} else {
if (mAudioSink.get() != NULL) {
mAudioSink->pause();
} else {
mAudioTrack->pause();
}
}
}
void AudioPlayer::resume() {
CHECK(mStarted);
if (mAudioSink.get() != NULL) {
mAudioSink->start();
} else {
mAudioTrack->start();
}
}
void AudioPlayer::reset() {
CHECK(mStarted);
if (mAudioSink.get() != NULL) {
mAudioSink->stop();
mAudioSink->close();
} else {
mAudioTrack->stop();
delete mAudioTrack;
mAudioTrack = NULL;
}
// Make sure to release any buffer we hold onto so that the
// source is able to stop().
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
mFirstBuffer = NULL;
}
if (mInputBuffer != NULL) {
ALOGV("AudioPlayer releasing input buffer.");
mInputBuffer->release();
mInputBuffer = NULL;
}
mSource->stop();
// 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);
}
IPCThreadState::self()->flushCommands();
mNumFramesPlayed = 0;
mPositionTimeMediaUs = -1;
mPositionTimeRealUs = -1;
mSeeking = false;
mReachedEOS = false;
mFinalStatus = OK;
mStarted = false;
}
// static
void AudioPlayer::AudioCallback(int event, void *user, void *info) {
static_cast<AudioPlayer *>(user)->AudioCallback(event, info);
}
bool AudioPlayer::isSeeking() {
Mutex::Autolock autoLock(mLock);
return mSeeking;
}
bool AudioPlayer::reachedEOS(status_t *finalStatus) {
*finalStatus = OK;
Mutex::Autolock autoLock(mLock);
*finalStatus = mFinalStatus;
return mReachedEOS;
}
status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
if (mAudioSink.get() != NULL) {
return mAudioSink->setPlaybackRatePermille(ratePermille);
} else if (mAudioTrack != NULL){
return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
} else {
return NO_INIT;
}
}
// static
size_t AudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
void *buffer, size_t size, void *cookie) {
AudioPlayer *me = (AudioPlayer *)cookie;
return me->fillBuffer(buffer, size);
}
void AudioPlayer::AudioCallback(int event, void *info) {
if (event != AudioTrack::EVENT_MORE_DATA) {
return;
}
AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
size_t numBytesWritten = fillBuffer(buffer->raw, buffer->size);
buffer->size = numBytesWritten;
}
uint32_t AudioPlayer::getNumFramesPendingPlayout() const {
uint32_t numFramesPlayedOut;
status_t err;
if (mAudioSink != NULL) {
err = mAudioSink->getPosition(&numFramesPlayedOut);
} else {
err = mAudioTrack->getPosition(&numFramesPlayedOut);
}
if (err != OK || mNumFramesPlayed < numFramesPlayedOut) {
return 0;
}
// mNumFramesPlayed is the number of frames submitted
// to the audio sink for playback, but not all of them
// may have played out by now.
return mNumFramesPlayed - numFramesPlayedOut;
}
size_t AudioPlayer::fillBuffer(void *data, size_t size) {
if (mNumFramesPlayed == 0) {
ALOGV("AudioCallback");
}
if (mReachedEOS) {
return 0;
}
bool postSeekComplete = false;
bool postEOS = false;
int64_t postEOSDelayUs = 0;
size_t size_done = 0;
size_t size_remaining = size;
while (size_remaining > 0) {
MediaSource::ReadOptions options;
{
Mutex::Autolock autoLock(mLock);
if (mSeeking) {
if (mIsFirstBuffer) {
if (mFirstBuffer != NULL) {
mFirstBuffer->release();
mFirstBuffer = NULL;
}
mIsFirstBuffer = false;
}
options.setSeekTo(mSeekTimeUs);
if (mInputBuffer != NULL) {
mInputBuffer->release();
mInputBuffer = NULL;
}
mSeeking = false;
if (mObserver) {
postSeekComplete = true;
}
}
}
if (mInputBuffer == NULL) {
status_t err;
if (mIsFirstBuffer) {
mInputBuffer = mFirstBuffer;
mFirstBuffer = NULL;
err = mFirstBufferResult;
mIsFirstBuffer = false;
} else {
err = mSource->read(&mInputBuffer, &options);
}
CHECK((err == OK && mInputBuffer != NULL)
|| (err != OK && mInputBuffer == NULL));
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;
postEOSDelayUs = timeToCompletionUs + mLatencyUs;
}
mReachedEOS = true;
mFinalStatus = err;
break;
}
if (mAudioSink != NULL) {
mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
} else {
mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
}
CHECK(mInputBuffer->meta_data()->findInt64(
kKeyTime, &mPositionTimeMediaUs));
mPositionTimeRealUs =
-mLatencyUs + ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
/ mSampleRate;
if (mPositionTimeRealUs < 0) {
mPositionTimeRealUs = 0;
}
ALOGV("buffer->size() = %d, "
"mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
mInputBuffer->range_length(),
mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
}
if (mInputBuffer->range_length() == 0) {
mInputBuffer->release();
mInputBuffer = NULL;
continue;
}
size_t copy = size_remaining;
if (copy > mInputBuffer->range_length()) {
copy = mInputBuffer->range_length();
}
memcpy((char *)data + size_done,
(const char *)mInputBuffer->data() + mInputBuffer->range_offset(),
copy);
mInputBuffer->set_range(mInputBuffer->range_offset() + copy,
mInputBuffer->range_length() - copy);
size_done += copy;
size_remaining -= copy;
}
{
Mutex::Autolock autoLock(mLock);
mNumFramesPlayed += size_done / mFrameSize;
}
if (postEOS) {
mObserver->postAudioEOS(postEOSDelayUs);
}
if (postSeekComplete) {
mObserver->postAudioSeekComplete();
}
return size_done;
}
int64_t AudioPlayer::getRealTimeUs() {
Mutex::Autolock autoLock(mLock);
return getRealTimeUsLocked();
}
int64_t AudioPlayer::getRealTimeUsLocked() const {
CHECK(mStarted);
CHECK_NE(mSampleRate, 0);
int64_t t = -mLatencyUs + (mNumFramesPlayed * 1000000) / mSampleRate;
if (t < 0) return 0;
return t;
}
int64_t AudioPlayer::getMediaTimeUs() {
Mutex::Autolock autoLock(mLock);
if (mPositionTimeMediaUs < 0 || mPositionTimeRealUs < 0) {
if (mSeeking) {
return mSeekTimeUs;
}
return 0;
}
int64_t realTimeOffset = getRealTimeUsLocked() - mPositionTimeRealUs;
if (realTimeOffset < 0) {
realTimeOffset = 0;
}
return mPositionTimeMediaUs + realTimeOffset;
}
bool AudioPlayer::getMediaTimeMapping(
int64_t *realtime_us, int64_t *mediatime_us) {
Mutex::Autolock autoLock(mLock);
*realtime_us = mPositionTimeRealUs;
*mediatime_us = mPositionTimeMediaUs;
return mPositionTimeRealUs != -1 && mPositionTimeMediaUs != -1;
}
status_t AudioPlayer::seekTo(int64_t time_us) {
Mutex::Autolock autoLock(mLock);
mSeeking = true;
mPositionTimeRealUs = mPositionTimeMediaUs = -1;
mReachedEOS = false;
mSeekTimeUs = time_us;
// Flush resets the number of played frames
mNumFramesPlayed = 0;
if (mAudioSink != NULL) {
mAudioSink->flush();
} else {
mAudioTrack->flush();
}
return OK;
}
}
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