/*
* Copyright (C) 2010 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 "NuPlayerRenderer"
#include <utils/Log.h>
#include "NuPlayerRenderer.h"
#include <algorithm>
#include <cutils/properties.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
#include <media/stagefright/foundation/AWakeLock.h>
#include <media/stagefright/MediaClock.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/VideoFrameScheduler.h>
#include <inttypes.h>
namespace android {
/*
* Example of common configuration settings in shell script form
#Turn offload audio off (use PCM for Play Music) -- AudioPolicyManager
adb shell setprop audio.offload.disable 1
#Allow offload audio with video (requires offloading to be enabled) -- AudioPolicyManager
adb shell setprop audio.offload.video 1
#Use audio callbacks for PCM data
adb shell setprop media.stagefright.audio.cbk 1
#Use deep buffer for PCM data with video (it is generally enabled for audio-only)
adb shell setprop media.stagefright.audio.deep 1
#Set size of buffers for pcm audio sink in msec (example: 1000 msec)
adb shell setprop media.stagefright.audio.sink 1000
* These configurations take effect for the next track played (not the current track).
*/
static inline bool getUseAudioCallbackSetting() {
return property_get_bool("media.stagefright.audio.cbk", false /* default_value */);
}
static inline int32_t getAudioSinkPcmMsSetting() {
return property_get_int32(
"media.stagefright.audio.sink", 500 /* default_value */);
}
// Maximum time in paused state when offloading audio decompression. When elapsed, the AudioSink
// is closed to allow the audio DSP to power down.
static const int64_t kOffloadPauseMaxUs = 10000000ll;
// Maximum allowed delay from AudioSink, 1.5 seconds.
static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000ll;
static const int64_t kMinimumAudioClockUpdatePeriodUs = 20 /* msec */ * 1000;
// static
const NuPlayer::Renderer::PcmInfo NuPlayer::Renderer::AUDIO_PCMINFO_INITIALIZER = {
AUDIO_CHANNEL_NONE,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID,
0, // mNumChannels
0 // mSampleRate
};
// static
const int64_t NuPlayer::Renderer::kMinPositionUpdateDelayUs = 100000ll;
NuPlayer::Renderer::Renderer(
const sp<MediaPlayerBase::AudioSink> &sink,
const sp<AMessage> ¬ify,
uint32_t flags)
: mAudioSink(sink),
mUseVirtualAudioSink(false),
mNotify(notify),
mFlags(flags),
mNumFramesWritten(0),
mDrainAudioQueuePending(false),
mDrainVideoQueuePending(false),
mAudioQueueGeneration(0),
mVideoQueueGeneration(0),
mAudioDrainGeneration(0),
mVideoDrainGeneration(0),
mAudioEOSGeneration(0),
mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT),
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
mAnchorNumFramesWritten(-1),
mVideoLateByUs(0ll),
mHasAudio(false),
mHasVideo(false),
mNotifyCompleteAudio(false),
mNotifyCompleteVideo(false),
mSyncQueues(false),
mPaused(false),
mPauseDrainAudioAllowedUs(0),
mVideoSampleReceived(false),
mVideoRenderingStarted(false),
mVideoRenderingStartGeneration(0),
mAudioRenderingStartGeneration(0),
mRenderingDataDelivered(false),
mNextAudioClockUpdateTimeUs(-1),
mLastAudioMediaTimeUs(-1),
mAudioOffloadPauseTimeoutGeneration(0),
mAudioTornDown(false),
mCurrentOffloadInfo(AUDIO_INFO_INITIALIZER),
mCurrentPcmInfo(AUDIO_PCMINFO_INITIALIZER),
mTotalBuffersQueued(0),
mLastAudioBufferDrained(0),
mUseAudioCallback(false),
mWakeLock(new AWakeLock()) {
mMediaClock = new MediaClock;
mPlaybackRate = mPlaybackSettings.mSpeed;
mMediaClock->setPlaybackRate(mPlaybackRate);
}
NuPlayer::Renderer::~Renderer() {
if (offloadingAudio()) {
mAudioSink->stop();
mAudioSink->flush();
mAudioSink->close();
}
}
void NuPlayer::Renderer::queueBuffer(
bool audio,
const sp<ABuffer> &buffer,
const sp<AMessage> ¬ifyConsumed) {
sp<AMessage> msg = new AMessage(kWhatQueueBuffer, this);
msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setBuffer("buffer", buffer);
msg->setMessage("notifyConsumed", notifyConsumed);
msg->post();
}
void NuPlayer::Renderer::queueEOS(bool audio, status_t finalResult) {
CHECK_NE(finalResult, (status_t)OK);
sp<AMessage> msg = new AMessage(kWhatQueueEOS, this);
msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setInt32("finalResult", finalResult);
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 && mAudioSink->ready()) {
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 && mAudioSink->ready()) {
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) {
{
Mutex::Autolock autoLock(mLock);
if (audio) {
mNotifyCompleteAudio |= notifyComplete;
clearAudioFirstAnchorTime_l();
++mAudioQueueGeneration;
++mAudioDrainGeneration;
} else {
mNotifyCompleteVideo |= notifyComplete;
++mVideoQueueGeneration;
++mVideoDrainGeneration;
}
clearAnchorTime_l();
mVideoLateByUs = 0;
mSyncQueues = false;
}
sp<AMessage> msg = new AMessage(kWhatFlush, this);
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->post();
}
void NuPlayer::Renderer::signalTimeDiscontinuity() {
}
void NuPlayer::Renderer::signalDisableOffloadAudio() {
(new AMessage(kWhatDisableOffloadAudio, this))->post();
}
void NuPlayer::Renderer::signalEnableOffloadAudio() {
(new AMessage(kWhatEnableOffloadAudio, this))->post();
}
void NuPlayer::Renderer::pause() {
(new AMessage(kWhatPause, this))->post();
}
void NuPlayer::Renderer::resume() {
(new AMessage(kWhatResume, this))->post();
}
void NuPlayer::Renderer::setVideoFrameRate(float fps) {
sp<AMessage> msg = new AMessage(kWhatSetVideoFrameRate, this);
msg->setFloat("frame-rate", fps);
msg->post();
}
// Called on any threads without mLock acquired.
status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
status_t result = mMediaClock->getMediaTime(ALooper::GetNowUs(), mediaUs);
if (result == OK) {
return result;
}
// MediaClock has not started yet. Try to start it if possible.
{
Mutex::Autolock autoLock(mLock);
if (mAudioFirstAnchorTimeMediaUs == -1) {
return result;
}
AudioTimestamp ts;
status_t res = mAudioSink->getTimestamp(ts);
if (res != OK) {
return result;
}
// AudioSink has rendered some frames.
int64_t nowUs = ALooper::GetNowUs();
int64_t nowMediaUs = mAudioSink->getPlayedOutDurationUs(nowUs)
+ mAudioFirstAnchorTimeMediaUs;
mMediaClock->updateAnchor(nowMediaUs, nowUs, -1);
}
return mMediaClock->getMediaTime(ALooper::GetNowUs(), mediaUs);
}
void NuPlayer::Renderer::clearAudioFirstAnchorTime_l() {
mAudioFirstAnchorTimeMediaUs = -1;
mMediaClock->setStartingTimeMedia(-1);
}
void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) {
if (mAudioFirstAnchorTimeMediaUs == -1) {
mAudioFirstAnchorTimeMediaUs = mediaUs;
mMediaClock->setStartingTimeMedia(mediaUs);
}
}
void NuPlayer::Renderer::clearAnchorTime_l() {
mMediaClock->clearAnchor();
mAnchorTimeMediaUs = -1;
mAnchorNumFramesWritten = -1;
}
void NuPlayer::Renderer::setVideoLateByUs(int64_t lateUs) {
Mutex::Autolock autoLock(mLock);
mVideoLateByUs = lateUs;
}
int64_t NuPlayer::Renderer::getVideoLateByUs() {
Mutex::Autolock autoLock(mLock);
return mVideoLateByUs;
}
status_t NuPlayer::Renderer::openAudioSink(
const sp<AMessage> &format,
bool offloadOnly,
bool hasVideo,
uint32_t flags,
bool *isOffloaded) {
sp<AMessage> msg = new AMessage(kWhatOpenAudioSink, this);
msg->setMessage("format", format);
msg->setInt32("offload-only", offloadOnly);
msg->setInt32("has-video", hasVideo);
msg->setInt32("flags", flags);
sp<AMessage> response;
msg->postAndAwaitResponse(&response);
int32_t err;
if (!response->findInt32("err", &err)) {
err = INVALID_OPERATION;
} else if (err == OK && isOffloaded != NULL) {
int32_t offload;
CHECK(response->findInt32("offload", &offload));
*isOffloaded = (offload != 0);
}
return err;
}
void NuPlayer::Renderer::closeAudioSink() {
sp<AMessage> msg = new AMessage(kWhatCloseAudioSink, this);
sp<AMessage> response;
msg->postAndAwaitResponse(&response);
}
void NuPlayer::Renderer::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
case kWhatOpenAudioSink:
{
sp<AMessage> format;
CHECK(msg->findMessage("format", &format));
int32_t offloadOnly;
CHECK(msg->findInt32("offload-only", &offloadOnly));
int32_t hasVideo;
CHECK(msg->findInt32("has-video", &hasVideo));
uint32_t flags;
CHECK(msg->findInt32("flags", (int32_t *)&flags));
status_t err = onOpenAudioSink(format, offloadOnly, hasVideo, flags);
sp<AMessage> response = new AMessage;
response->setInt32("err", err);
response->setInt32("offload", offloadingAudio());
sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
break;
}
case kWhatCloseAudioSink:
{
sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
onCloseAudioSink();
sp<AMessage> response = new AMessage;
response->postReply(replyID);
break;
}
case kWhatStopAudioSink:
{
mAudioSink->stop();
break;
}
case kWhatDrainAudioQueue:
{
mDrainAudioQueuePending = false;
int32_t generation;
CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != getDrainGeneration(true /* audio */)) {
break;
}
if (onDrainAudioQueue()) {
uint32_t numFramesPlayed;
CHECK_EQ(mAudioSink->getPosition(&numFramesPlayed),
(status_t)OK);
uint32_t numFramesPendingPlayout =
mNumFramesWritten - numFramesPlayed;
// This is how long the audio sink will have data to
// play back.
int64_t delayUs =
mAudioSink->msecsPerFrame()
* numFramesPendingPlayout * 1000ll;
if (mPlaybackRate > 1.0f) {
delayUs /= mPlaybackRate;
}
// Let's give it more data after about half that time
// has elapsed.
delayUs /= 2;
// check the buffer size to estimate maximum delay permitted.
const int64_t maxDrainDelayUs = std::max(
mAudioSink->getBufferDurationInUs(), (int64_t)500000 /* half second */);
ALOGD_IF(delayUs > maxDrainDelayUs, "postDrainAudioQueue long delay: %lld > %lld",
(long long)delayUs, (long long)maxDrainDelayUs);
Mutex::Autolock autoLock(mLock);
postDrainAudioQueue_l(delayUs);
}
break;
}
case kWhatDrainVideoQueue:
{
int32_t generation;
CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != getDrainGeneration(false /* audio */)) {
break;
}
mDrainVideoQueuePending = false;
onDrainVideoQueue();
postDrainVideoQueue();
break;
}
case kWhatPostDrainVideoQueue:
{
int32_t generation;
CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != getDrainGeneration(false /* audio */)) {
break;
}
mDrainVideoQueuePending = false;
postDrainVideoQueue();
break;
}
case kWhatQueueBuffer:
{
onQueueBuffer(msg);
break;
}
case kWhatQueueEOS:
{
onQueueEOS(msg);
break;
}
case kWhatEOS:
{
int32_t generation;
CHECK(msg->findInt32("audioEOSGeneration", &generation));
if (generation != mAudioEOSGeneration) {
break;
}
status_t finalResult;
CHECK(msg->findInt32("finalResult", &finalResult));
notifyEOS(true /* audio */, finalResult);
break;
}
case kWhatConfigPlayback:
{
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;
}
case kWhatFlush:
{
onFlush(msg);
break;
}
case kWhatDisableOffloadAudio:
{
onDisableOffloadAudio();
break;
}
case kWhatEnableOffloadAudio:
{
onEnableOffloadAudio();
break;
}
case kWhatPause:
{
onPause();
break;
}
case kWhatResume:
{
onResume();
break;
}
case kWhatSetVideoFrameRate:
{
float fps;
CHECK(msg->findFloat("frame-rate", &fps));
onSetVideoFrameRate(fps);
break;
}
case kWhatAudioTearDown:
{
int32_t reason;
CHECK(msg->findInt32("reason", &reason));
onAudioTearDown((AudioTearDownReason)reason);
break;
}
case kWhatAudioOffloadPauseTimeout:
{
int32_t generation;
CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != mAudioOffloadPauseTimeoutGeneration) {
break;
}
ALOGV("Audio Offload tear down due to pause timeout.");
onAudioTearDown(kDueToTimeout);
mWakeLock->release();
break;
}
default:
TRESPASS();
break;
}
}
void NuPlayer::Renderer::postDrainAudioQueue_l(int64_t delayUs) {
if (mDrainAudioQueuePending || mSyncQueues || mUseAudioCallback) {
return;
}
if (mAudioQueue.empty()) {
return;
}
// FIXME: if paused, wait until AudioTrack stop() is complete before delivering data.
if (mPaused) {
const int64_t diffUs = mPauseDrainAudioAllowedUs - ALooper::GetNowUs();
if (diffUs > delayUs) {
delayUs = diffUs;
}
}
mDrainAudioQueuePending = true;
sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, this);
msg->setInt32("drainGeneration", mAudioDrainGeneration);
msg->post(delayUs);
}
void NuPlayer::Renderer::prepareForMediaRenderingStart_l() {
mAudioRenderingStartGeneration = mAudioDrainGeneration;
mVideoRenderingStartGeneration = mVideoDrainGeneration;
mRenderingDataDelivered = false;
}
void NuPlayer::Renderer::notifyIfMediaRenderingStarted_l() {
if (mVideoRenderingStartGeneration == mVideoDrainGeneration &&
mAudioRenderingStartGeneration == mAudioDrainGeneration) {
mRenderingDataDelivered = true;
if (mPaused) {
return;
}
mVideoRenderingStartGeneration = -1;
mAudioRenderingStartGeneration = -1;
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatMediaRenderingStart);
notify->post();
}
}
// static
size_t NuPlayer::Renderer::AudioSinkCallback(
MediaPlayerBase::AudioSink * /* audioSink */,
void *buffer,
size_t size,
void *cookie,
MediaPlayerBase::AudioSink::cb_event_t event) {
NuPlayer::Renderer *me = (NuPlayer::Renderer *)cookie;
switch (event) {
case MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER:
{
return me->fillAudioBuffer(buffer, size);
break;
}
case MediaPlayerBase::AudioSink::CB_EVENT_STREAM_END:
{
ALOGV("AudioSink::CB_EVENT_STREAM_END");
me->notifyEOS(true /* audio */, ERROR_END_OF_STREAM);
break;
}
case MediaPlayerBase::AudioSink::CB_EVENT_TEAR_DOWN:
{
ALOGV("AudioSink::CB_EVENT_TEAR_DOWN");
me->notifyAudioTearDown(kDueToError);
break;
}
}
return 0;
}
size_t NuPlayer::Renderer::fillAudioBuffer(void *buffer, size_t size) {
Mutex::Autolock autoLock(mLock);
if (!mUseAudioCallback) {
return 0;
}
bool hasEOS = false;
size_t sizeCopied = 0;
bool firstEntry = true;
QueueEntry *entry; // will be valid after while loop if hasEOS is set.
while (sizeCopied < size && !mAudioQueue.empty()) {
entry = &*mAudioQueue.begin();
if (entry->mBuffer == NULL) { // EOS
hasEOS = true;
mAudioQueue.erase(mAudioQueue.begin());
break;
}
if (firstEntry && entry->mOffset == 0) {
firstEntry = false;
int64_t mediaTimeUs;
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
ALOGV("fillAudioBuffer: rendering audio at media time %.2f secs", mediaTimeUs / 1E6);
setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
}
size_t copy = entry->mBuffer->size() - entry->mOffset;
size_t sizeRemaining = size - sizeCopied;
if (copy > sizeRemaining) {
copy = sizeRemaining;
}
memcpy((char *)buffer + sizeCopied,
entry->mBuffer->data() + entry->mOffset,
copy);
entry->mOffset += copy;
if (entry->mOffset == entry->mBuffer->size()) {
entry->mNotifyConsumed->post();
mAudioQueue.erase(mAudioQueue.begin());
entry = NULL;
}
sizeCopied += copy;
notifyIfMediaRenderingStarted_l();
}
if (mAudioFirstAnchorTimeMediaUs >= 0) {
int64_t nowUs = ALooper::GetNowUs();
int64_t nowMediaUs =
mAudioFirstAnchorTimeMediaUs + mAudioSink->getPlayedOutDurationUs(nowUs);
// we don't know how much data we are queueing for offloaded tracks.
mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX);
}
// for non-offloaded audio, we need to compute the frames written because
// there is no EVENT_STREAM_END notification. The frames written gives
// an estimate on the pending played out duration.
if (!offloadingAudio()) {
mNumFramesWritten += sizeCopied / mAudioSink->frameSize();
}
if (hasEOS) {
(new AMessage(kWhatStopAudioSink, this))->post();
// As there is currently no EVENT_STREAM_END callback notification for
// non-offloaded audio tracks, we need to post the EOS ourselves.
if (!offloadingAudio()) {
int64_t postEOSDelayUs = 0;
if (mAudioSink->needsTrailingPadding()) {
postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs());
}
ALOGV("fillAudioBuffer: notifyEOS "
"mNumFramesWritten:%u finalResult:%d postEOSDelay:%lld",
mNumFramesWritten, entry->mFinalResult, (long long)postEOSDelayUs);
notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs);
}
}
return sizeCopied;
}
void NuPlayer::Renderer::drainAudioQueueUntilLastEOS() {
List<QueueEntry>::iterator it = mAudioQueue.begin(), itEOS = it;
bool foundEOS = false;
while (it != mAudioQueue.end()) {
int32_t eos;
QueueEntry *entry = &*it++;
if (entry->mBuffer == NULL
|| (entry->mNotifyConsumed->findInt32("eos", &eos) && eos != 0)) {
itEOS = it;
foundEOS = true;
}
}
if (foundEOS) {
// post all replies before EOS and drop the samples
for (it = mAudioQueue.begin(); it != itEOS; it++) {
if (it->mBuffer == NULL) {
// delay doesn't matter as we don't even have an AudioTrack
notifyEOS(true /* audio */, it->mFinalResult);
} else {
it->mNotifyConsumed->post();
}
}
mAudioQueue.erase(mAudioQueue.begin(), itEOS);
}
}
bool NuPlayer::Renderer::onDrainAudioQueue() {
// do not drain audio during teardown as queued buffers may be invalid.
if (mAudioTornDown) {
return false;
}
// TODO: This call to getPosition checks if AudioTrack has been created
// in AudioSink before draining audio. If AudioTrack doesn't exist, then
// CHECKs on getPosition will fail.
// We still need to figure out why AudioTrack is not created when
// this function is called. One possible reason could be leftover
// audio. Another possible place is to check whether decoder
// has received INFO_FORMAT_CHANGED as the first buffer since
// AudioSink is opened there, and possible interactions with flush
// immediately after start. Investigate error message
// "vorbis_dsp_synthesis returned -135", along with RTSP.
uint32_t numFramesPlayed;
if (mAudioSink->getPosition(&numFramesPlayed) != OK) {
// When getPosition fails, renderer will not reschedule the draining
// unless new samples are queued.
// If we have pending EOS (or "eos" marker for discontinuities), we need
// to post these now as NuPlayerDecoder might be waiting for it.
drainAudioQueueUntilLastEOS();
ALOGW("onDrainAudioQueue(): audio sink is not ready");
return false;
}
#if 0
ssize_t numFramesAvailableToWrite =
mAudioSink->frameCount() - (mNumFramesWritten - numFramesPlayed);
if (numFramesAvailableToWrite == mAudioSink->frameCount()) {
ALOGI("audio sink underrun");
} else {
ALOGV("audio queue has %d frames left to play",
mAudioSink->frameCount() - numFramesAvailableToWrite);
}
#endif
uint32_t prevFramesWritten = mNumFramesWritten;
while (!mAudioQueue.empty()) {
QueueEntry *entry = &*mAudioQueue.begin();
mLastAudioBufferDrained = entry->mBufferOrdinal;
if (entry->mBuffer == NULL) {
// EOS
int64_t postEOSDelayUs = 0;
if (mAudioSink->needsTrailingPadding()) {
postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs());
}
notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs);
mLastAudioMediaTimeUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten);
mAudioQueue.erase(mAudioQueue.begin());
entry = NULL;
if (mAudioSink->needsTrailingPadding()) {
// If we're not in gapless playback (i.e. through setNextPlayer), we
// need to stop the track here, because that will play out the last
// little bit at the end of the file. Otherwise short files won't play.
mAudioSink->stop();
mNumFramesWritten = 0;
}
return false;
}
// ignore 0-sized buffer which could be EOS marker with no data
if (entry->mOffset == 0 && entry->mBuffer->size() > 0) {
int64_t mediaTimeUs;
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
ALOGV("onDrainAudioQueue: rendering audio at media time %.2f secs",
mediaTimeUs / 1E6);
onNewAudioMediaTime(mediaTimeUs);
}
size_t copy = entry->mBuffer->size() - entry->mOffset;
ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset,
copy, false /* blocking */);
if (written < 0) {
// An error in AudioSink write. Perhaps the AudioSink was not properly opened.
if (written == WOULD_BLOCK) {
ALOGV("AudioSink write would block when writing %zu bytes", copy);
} else {
ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy);
// This can only happen when AudioSink was opened with doNotReconnect flag set to
// true, in which case the NuPlayer will handle the reconnect.
notifyAudioTearDown(kDueToError);
}
break;
}
entry->mOffset += written;
size_t remainder = entry->mBuffer->size() - entry->mOffset;
if ((ssize_t)remainder < mAudioSink->frameSize()) {
if (remainder > 0) {
ALOGW("Corrupted audio buffer has fractional frames, discarding %zu bytes.",
remainder);
entry->mOffset += remainder;
copy -= remainder;
}
entry->mNotifyConsumed->post();
mAudioQueue.erase(mAudioQueue.begin());
entry = NULL;
}
size_t copiedFrames = written / mAudioSink->frameSize();
mNumFramesWritten += copiedFrames;
{
Mutex::Autolock autoLock(mLock);
int64_t maxTimeMedia;
maxTimeMedia =
mAnchorTimeMediaUs +
(int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
* 1000LL * mAudioSink->msecsPerFrame());
mMediaClock->updateMaxTimeMedia(maxTimeMedia);
notifyIfMediaRenderingStarted_l();
}
if (written != (ssize_t)copy) {
// A short count was received from AudioSink::write()
//
// AudioSink write is called in non-blocking mode.
// It may return with a short count when:
//
// 1) Size to be copied is not a multiple of the frame size. Fractional frames are
// discarded.
// 2) The data to be copied exceeds the available buffer in AudioSink.
// 3) An error occurs and data has been partially copied to the buffer in AudioSink.
// 4) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded.
// (Case 1)
// Must be a multiple of the frame size. If it is not a multiple of a frame size, it
// needs to fail, as we should not carry over fractional frames between calls.
CHECK_EQ(copy % mAudioSink->frameSize(), 0);
// (Case 2, 3, 4)
// Return early to the caller.
// Beware of calling immediately again as this may busy-loop if you are not careful.
ALOGV("AudioSink write short frame count %zd < %zu", written, copy);
break;
}
}
// calculate whether we need to reschedule another write.
bool reschedule = !mAudioQueue.empty()
&& (!mPaused
|| prevFramesWritten != mNumFramesWritten); // permit pause to fill buffers
//ALOGD("reschedule:%d empty:%d mPaused:%d prevFramesWritten:%u mNumFramesWritten:%u",
// reschedule, mAudioQueue.empty(), mPaused, prevFramesWritten, mNumFramesWritten);
return reschedule;
}
int64_t NuPlayer::Renderer::getDurationUsIfPlayedAtSampleRate(uint32_t numFrames) {
int32_t sampleRate = offloadingAudio() ?
mCurrentOffloadInfo.sample_rate : mCurrentPcmInfo.mSampleRate;
if (sampleRate == 0) {
ALOGE("sampleRate is 0 in %s mode", offloadingAudio() ? "offload" : "non-offload");
return 0;
}
// TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
return (int64_t)((int32_t)numFrames * 1000000LL / sampleRate);
}
// Calculate duration of pending samples if played at normal rate (i.e., 1.0).
int64_t NuPlayer::Renderer::getPendingAudioPlayoutDurationUs(int64_t nowUs) {
int64_t writtenAudioDurationUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten);
if (mUseVirtualAudioSink) {
int64_t nowUs = ALooper::GetNowUs();
int64_t mediaUs;
if (mMediaClock->getMediaTime(nowUs, &mediaUs) != OK) {
return 0ll;
} else {
return writtenAudioDurationUs - (mediaUs - mAudioFirstAnchorTimeMediaUs);
}
}
return writtenAudioDurationUs - mAudioSink->getPlayedOutDurationUs(nowUs);
}
int64_t NuPlayer::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) {
int64_t realUs;
if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) {
// If failed to get current position, e.g. due to audio clock is
// not ready, then just play out video immediately without delay.
return nowUs;
}
return realUs;
}
void NuPlayer::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) {
Mutex::Autolock autoLock(mLock);
// TRICKY: vorbis decoder generates multiple frames with the same
// timestamp, so only update on the first frame with a given timestamp
if (mediaTimeUs == mAnchorTimeMediaUs) {
return;
}
setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
// mNextAudioClockUpdateTimeUs is -1 if we're waiting for audio sink to start
if (mNextAudioClockUpdateTimeUs == -1) {
AudioTimestamp ts;
if (mAudioSink->getTimestamp(ts) == OK && ts.mPosition > 0) {
mNextAudioClockUpdateTimeUs = 0; // start our clock updates
}
}
int64_t nowUs = ALooper::GetNowUs();
if (mNextAudioClockUpdateTimeUs >= 0) {
if (nowUs >= mNextAudioClockUpdateTimeUs) {
int64_t nowMediaUs = mediaTimeUs - getPendingAudioPlayoutDurationUs(nowUs);
mMediaClock->updateAnchor(nowMediaUs, nowUs, mediaTimeUs);
mUseVirtualAudioSink = false;
mNextAudioClockUpdateTimeUs = nowUs + kMinimumAudioClockUpdatePeriodUs;
}
} else {
int64_t unused;
if ((mMediaClock->getMediaTime(nowUs, &unused) != OK)
&& (getDurationUsIfPlayedAtSampleRate(mNumFramesWritten)
> kMaxAllowedAudioSinkDelayUs)) {
// Enough data has been sent to AudioSink, but AudioSink has not rendered
// any data yet. Something is wrong with AudioSink, e.g., the device is not
// connected to audio out.
// Switch to system clock. This essentially creates a virtual AudioSink with
// initial latenty of getDurationUsIfPlayedAtSampleRate(mNumFramesWritten).
// This virtual AudioSink renders audio data starting from the very first sample
// and it's paced by system clock.
ALOGW("AudioSink stuck. ARE YOU CONNECTED TO AUDIO OUT? Switching to system clock.");
mMediaClock->updateAnchor(mAudioFirstAnchorTimeMediaUs, nowUs, mediaTimeUs);
mUseVirtualAudioSink = true;
}
}
mAnchorNumFramesWritten = mNumFramesWritten;
mAnchorTimeMediaUs = mediaTimeUs;
}
// Called without mLock acquired.
void NuPlayer::Renderer::postDrainVideoQueue() {
if (mDrainVideoQueuePending
|| getSyncQueues()
|| (mPaused && mVideoSampleReceived)) {
return;
}
if (mVideoQueue.empty()) {
return;
}
QueueEntry &entry = *mVideoQueue.begin();
sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, this);
msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));
if (entry.mBuffer == NULL) {
// EOS doesn't carry a timestamp.
msg->post();
mDrainVideoQueuePending = true;
return;
}
bool needRepostDrainVideoQueue = false;
int64_t delayUs;
int64_t nowUs = ALooper::GetNowUs();
int64_t realTimeUs;
if (mFlags & FLAG_REAL_TIME) {
int64_t mediaTimeUs;
CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
realTimeUs = mediaTimeUs;
} else {
int64_t mediaTimeUs;
CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
{
Mutex::Autolock autoLock(mLock);
if (mAnchorTimeMediaUs < 0) {
mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
mAnchorTimeMediaUs = mediaTimeUs;
realTimeUs = nowUs;
} else if (!mVideoSampleReceived) {
// Always render the first video frame.
realTimeUs = nowUs;
} else if (mAudioFirstAnchorTimeMediaUs < 0
|| mMediaClock->getRealTimeFor(mediaTimeUs, &realTimeUs) == OK) {
realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
} else if (mediaTimeUs - mAudioFirstAnchorTimeMediaUs >= 0) {
needRepostDrainVideoQueue = true;
realTimeUs = nowUs;
} else {
realTimeUs = nowUs;
}
}
if (!mHasAudio) {
// smooth out videos >= 10fps
mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
}
// Heuristics to handle situation when media time changed without a
// discontinuity. If we have not drained an audio buffer that was
// received after this buffer, repost in 10 msec. Otherwise repost
// in 500 msec.
delayUs = realTimeUs - nowUs;
int64_t postDelayUs = -1;
if (delayUs > 500000) {
postDelayUs = 500000;
if (mHasAudio && (mLastAudioBufferDrained - entry.mBufferOrdinal) <= 0) {
postDelayUs = 10000;
}
} else if (needRepostDrainVideoQueue) {
// CHECK(mPlaybackRate > 0);
// CHECK(mAudioFirstAnchorTimeMediaUs >= 0);
// CHECK(mediaTimeUs - mAudioFirstAnchorTimeMediaUs >= 0);
postDelayUs = mediaTimeUs - mAudioFirstAnchorTimeMediaUs;
postDelayUs /= mPlaybackRate;
}
if (postDelayUs >= 0) {
msg->setWhat(kWhatPostDrainVideoQueue);
msg->post(postDelayUs);
mVideoScheduler->restart();
ALOGI("possible video time jump of %dms or uninitialized media clock, retrying in %dms",
(int)(delayUs / 1000), (int)(postDelayUs / 1000));
mDrainVideoQueuePending = true;
return;
}
}
realTimeUs = mVideoScheduler->schedule(realTimeUs * 1000) / 1000;
int64_t twoVsyncsUs = 2 * (mVideoScheduler->getVsyncPeriod() / 1000);
delayUs = realTimeUs - nowUs;
ALOGW_IF(delayUs > 500000, "unusually high delayUs: %" PRId64, delayUs);
// post 2 display refreshes before rendering is due
msg->post(delayUs > twoVsyncsUs ? delayUs - twoVsyncsUs : 0);
mDrainVideoQueuePending = true;
}
void NuPlayer::Renderer::onDrainVideoQueue() {
if (mVideoQueue.empty()) {
return;
}
QueueEntry *entry = &*mVideoQueue.begin();
if (entry->mBuffer == NULL) {
// EOS
notifyEOS(false /* audio */, entry->mFinalResult);
mVideoQueue.erase(mVideoQueue.begin());
entry = NULL;
setVideoLateByUs(0);
return;
}
int64_t nowUs = ALooper::GetNowUs();
int64_t realTimeUs;
int64_t mediaTimeUs = -1;
if (mFlags & FLAG_REAL_TIME) {
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &realTimeUs));
} else {
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
}
bool tooLate = false;
if (!mPaused) {
setVideoLateByUs(nowUs - realTimeUs);
tooLate = (mVideoLateByUs > 40000);
if (tooLate) {
ALOGV("video late by %lld us (%.2f secs)",
(long long)mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
int64_t mediaUs = 0;
mMediaClock->getMediaTime(realTimeUs, &mediaUs);
ALOGV("rendering video at media time %.2f secs",
(mFlags & FLAG_REAL_TIME ? realTimeUs :
mediaUs) / 1E6);
if (!(mFlags & FLAG_REAL_TIME)
&& mLastAudioMediaTimeUs != -1
&& mediaTimeUs > mLastAudioMediaTimeUs) {
// If audio ends before video, video continues to drive media clock.
// Also smooth out videos >= 10fps.
mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
}
}
} else {
setVideoLateByUs(0);
if (!mVideoSampleReceived && !mHasAudio) {
// This will ensure that the first frame after a flush won't be used as anchor
// when renderer is in paused state, because resume can happen any time after seek.
Mutex::Autolock autoLock(mLock);
clearAnchorTime_l();
}
}
// Always render the first video frame while keeping stats on A/V sync.
if (!mVideoSampleReceived) {
realTimeUs = nowUs;
tooLate = false;
}
entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000ll);
entry->mNotifyConsumed->setInt32("render", !tooLate);
entry->mNotifyConsumed->post();
mVideoQueue.erase(mVideoQueue.begin());
entry = NULL;
mVideoSampleReceived = true;
if (!mPaused) {
if (!mVideoRenderingStarted) {
mVideoRenderingStarted = true;
notifyVideoRenderingStart();
}
Mutex::Autolock autoLock(mLock);
notifyIfMediaRenderingStarted_l();
}
}
void NuPlayer::Renderer::notifyVideoRenderingStart() {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatVideoRenderingStart);
notify->post();
}
void NuPlayer::Renderer::notifyEOS(bool audio, status_t finalResult, int64_t delayUs) {
if (audio && delayUs > 0) {
sp<AMessage> msg = new AMessage(kWhatEOS, this);
msg->setInt32("audioEOSGeneration", mAudioEOSGeneration);
msg->setInt32("finalResult", finalResult);
msg->post(delayUs);
return;
}
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatEOS);
notify->setInt32("audio", static_cast<int32_t>(audio));
notify->setInt32("finalResult", finalResult);
notify->post(delayUs);
}
void NuPlayer::Renderer::notifyAudioTearDown(AudioTearDownReason reason) {
sp<AMessage> msg = new AMessage(kWhatAudioTearDown, this);
msg->setInt32("reason", reason);
msg->post();
}
void NuPlayer::Renderer::onQueueBuffer(const sp<AMessage> &msg) {
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
if (dropBufferIfStale(audio, msg)) {
return;
}
if (audio) {
mHasAudio = true;
} else {
mHasVideo = true;
}
if (mHasVideo) {
if (mVideoScheduler == NULL) {
mVideoScheduler = new VideoFrameScheduler();
mVideoScheduler->init();
}
}
sp<ABuffer> buffer;
CHECK(msg->findBuffer("buffer", &buffer));
sp<AMessage> notifyConsumed;
CHECK(msg->findMessage("notifyConsumed", ¬ifyConsumed));
QueueEntry entry;
entry.mBuffer = buffer;
entry.mNotifyConsumed = notifyConsumed;
entry.mOffset = 0;
entry.mFinalResult = OK;
entry.mBufferOrdinal = ++mTotalBuffersQueued;
if (audio) {
Mutex::Autolock autoLock(mLock);
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
mVideoQueue.push_back(entry);
postDrainVideoQueue();
}
Mutex::Autolock autoLock(mLock);
if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
return;
}
sp<ABuffer> firstAudioBuffer = (*mAudioQueue.begin()).mBuffer;
sp<ABuffer> firstVideoBuffer = (*mVideoQueue.begin()).mBuffer;
if (firstAudioBuffer == NULL || firstVideoBuffer == NULL) {
// EOS signalled on either queue.
syncQueuesDone_l();
return;
}
int64_t firstAudioTimeUs;
int64_t firstVideoTimeUs;
CHECK(firstAudioBuffer->meta()
->findInt64("timeUs", &firstAudioTimeUs));
CHECK(firstVideoBuffer->meta()
->findInt64("timeUs", &firstVideoTimeUs));
int64_t diff = firstVideoTimeUs - firstAudioTimeUs;
ALOGV("queueDiff = %.2f secs", diff / 1E6);
if (diff > 100000ll) {
// Audio data starts More than 0.1 secs before video.
// Drop some audio.
(*mAudioQueue.begin()).mNotifyConsumed->post();
mAudioQueue.erase(mAudioQueue.begin());
return;
}
syncQueuesDone_l();
}
void NuPlayer::Renderer::syncQueuesDone_l() {
if (!mSyncQueues) {
return;
}
mSyncQueues = false;
if (!mAudioQueue.empty()) {
postDrainAudioQueue_l();
}
if (!mVideoQueue.empty()) {
mLock.unlock();
postDrainVideoQueue();
mLock.lock();
}
}
void NuPlayer::Renderer::onQueueEOS(const sp<AMessage> &msg) {
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
if (dropBufferIfStale(audio, msg)) {
return;
}
int32_t finalResult;
CHECK(msg->findInt32("finalResult", &finalResult));
QueueEntry entry;
entry.mOffset = 0;
entry.mFinalResult = finalResult;
if (audio) {
Mutex::Autolock autoLock(mLock);
if (mAudioQueue.empty() && mSyncQueues) {
syncQueuesDone_l();
}
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
if (mVideoQueue.empty() && getSyncQueues()) {
Mutex::Autolock autoLock(mLock);
syncQueuesDone_l();
}
mVideoQueue.push_back(entry);
postDrainVideoQueue();
}
}
void NuPlayer::Renderer::onFlush(const sp<AMessage> &msg) {
int32_t audio, notifyComplete;
CHECK(msg->findInt32("audio", &audio));
{
Mutex::Autolock autoLock(mLock);
if (audio) {
notifyComplete = mNotifyCompleteAudio;
mNotifyCompleteAudio = false;
mLastAudioMediaTimeUs = -1;
} else {
notifyComplete = mNotifyCompleteVideo;
mNotifyCompleteVideo = false;
}
// If we're currently syncing the queues, i.e. dropping audio while
// aligning the first audio/video buffer times and only one of the
// two queues has data, we may starve that queue by not requesting
// more buffers from the decoder. If the other source then encounters
// a discontinuity that leads to flushing, we'll never find the
// corresponding discontinuity on the other queue.
// Therefore we'll stop syncing the queues if at least one of them
// is flushed.
syncQueuesDone_l();
clearAnchorTime_l();
}
ALOGV("flushing %s", audio ? "audio" : "video");
if (audio) {
{
Mutex::Autolock autoLock(mLock);
flushQueue(&mAudioQueue);
++mAudioDrainGeneration;
++mAudioEOSGeneration;
prepareForMediaRenderingStart_l();
// the frame count will be reset after flush.
clearAudioFirstAnchorTime_l();
}
mDrainAudioQueuePending = false;
if (offloadingAudio()) {
mAudioSink->pause();
mAudioSink->flush();
if (!mPaused) {
mAudioSink->start();
}
} else {
mAudioSink->pause();
mAudioSink->flush();
// Call stop() to signal to the AudioSink to completely fill the
// 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 {
mAudioSink->start();
}
mNumFramesWritten = 0;
}
mNextAudioClockUpdateTimeUs = -1;
} else {
flushQueue(&mVideoQueue);
mDrainVideoQueuePending = false;
if (mVideoScheduler != NULL) {
mVideoScheduler->restart();
}
Mutex::Autolock autoLock(mLock);
++mVideoDrainGeneration;
prepareForMediaRenderingStart_l();
}
mVideoSampleReceived = false;
if (notifyComplete) {
notifyFlushComplete(audio);
}
}
void NuPlayer::Renderer::flushQueue(List<QueueEntry> *queue) {
while (!queue->empty()) {
QueueEntry *entry = &*queue->begin();
if (entry->mBuffer != NULL) {
entry->mNotifyConsumed->post();
}
queue->erase(queue->begin());
entry = NULL;
}
}
void NuPlayer::Renderer::notifyFlushComplete(bool audio) {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatFlushComplete);
notify->setInt32("audio", static_cast<int32_t>(audio));
notify->post();
}
bool NuPlayer::Renderer::dropBufferIfStale(
bool audio, const sp<AMessage> &msg) {
int32_t queueGeneration;
CHECK(msg->findInt32("queueGeneration", &queueGeneration));
if (queueGeneration == getQueueGeneration(audio)) {
return false;
}
sp<AMessage> notifyConsumed;
if (msg->findMessage("notifyConsumed", ¬ifyConsumed)) {
notifyConsumed->post();
}
return true;
}
void NuPlayer::Renderer::onAudioSinkChanged() {
if (offloadingAudio()) {
return;
}
CHECK(!mDrainAudioQueuePending);
mNumFramesWritten = 0;
{
Mutex::Autolock autoLock(mLock);
mAnchorNumFramesWritten = -1;
}
uint32_t written;
if (mAudioSink->getFramesWritten(&written) == OK) {
mNumFramesWritten = written;
}
}
void NuPlayer::Renderer::onDisableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags &= ~FLAG_OFFLOAD_AUDIO;
++mAudioDrainGeneration;
if (mAudioRenderingStartGeneration != -1) {
prepareForMediaRenderingStart_l();
}
}
void NuPlayer::Renderer::onEnableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags |= FLAG_OFFLOAD_AUDIO;
++mAudioDrainGeneration;
if (mAudioRenderingStartGeneration != -1) {
prepareForMediaRenderingStart_l();
}
}
void NuPlayer::Renderer::onPause() {
if (mPaused) {
return;
}
{
Mutex::Autolock autoLock(mLock);
// we do not increment audio drain generation so that we fill audio buffer during pause.
++mVideoDrainGeneration;
prepareForMediaRenderingStart_l();
mPaused = true;
mMediaClock->setPlaybackRate(0.0);
}
mDrainAudioQueuePending = false;
mDrainVideoQueuePending = false;
// Note: audio data may not have been decoded, and the AudioSink may not be opened.
mAudioSink->pause();
startAudioOffloadPauseTimeout();
ALOGV("now paused audio queue has %zu entries, video has %zu entries",
mAudioQueue.size(), mVideoQueue.size());
}
void NuPlayer::Renderer::onResume() {
if (!mPaused) {
return;
}
// Note: audio data may not have been decoded, and the AudioSink may not be opened.
cancelAudioOffloadPauseTimeout();
if (mAudioSink->ready()) {
status_t err = mAudioSink->start();
if (err != OK) {
ALOGE("cannot start AudioSink err %d", err);
notifyAudioTearDown(kDueToError);
}
}
{
Mutex::Autolock autoLock(mLock);
mPaused = false;
// rendering started message may have been delayed if we were paused.
if (mRenderingDataDelivered) {
notifyIfMediaRenderingStarted_l();
}
// configure audiosink as we did not do it when pausing
if (mAudioSink != NULL && mAudioSink->ready()) {
mAudioSink->setPlaybackRate(mPlaybackSettings);
}
mMediaClock->setPlaybackRate(mPlaybackRate);
if (!mAudioQueue.empty()) {
postDrainAudioQueue_l();
}
}
if (!mVideoQueue.empty()) {
postDrainVideoQueue();
}
}
void NuPlayer::Renderer::onSetVideoFrameRate(float fps) {
if (mVideoScheduler == NULL) {
mVideoScheduler = new VideoFrameScheduler();
}
mVideoScheduler->init(fps);
}
int32_t NuPlayer::Renderer::getQueueGeneration(bool audio) {
Mutex::Autolock autoLock(mLock);
return (audio ? mAudioQueueGeneration : mVideoQueueGeneration);
}
int32_t NuPlayer::Renderer::getDrainGeneration(bool audio) {
Mutex::Autolock autoLock(mLock);
return (audio ? mAudioDrainGeneration : mVideoDrainGeneration);
}
bool NuPlayer::Renderer::getSyncQueues() {
Mutex::Autolock autoLock(mLock);
return mSyncQueues;
}
void NuPlayer::Renderer::onAudioTearDown(AudioTearDownReason reason) {
if (mAudioTornDown) {
return;
}
mAudioTornDown = true;
int64_t currentPositionUs;
sp<AMessage> notify = mNotify->dup();
if (getCurrentPosition(¤tPositionUs) == OK) {
notify->setInt64("positionUs", currentPositionUs);
}
mAudioSink->stop();
mAudioSink->flush();
notify->setInt32("what", kWhatAudioTearDown);
notify->setInt32("reason", reason);
notify->post();
}
void NuPlayer::Renderer::startAudioOffloadPauseTimeout() {
if (offloadingAudio()) {
mWakeLock->acquire();
sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, this);
msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
msg->post(kOffloadPauseMaxUs);
}
}
void NuPlayer::Renderer::cancelAudioOffloadPauseTimeout() {
// We may have called startAudioOffloadPauseTimeout() without
// the AudioSink open and with offloadingAudio enabled.
//
// When we cancel, it may be that offloadingAudio is subsequently disabled, so regardless
// we always release the wakelock and increment the pause timeout generation.
//
// Note: The acquired wakelock prevents the device from suspending
// immediately after offload pause (in case a resume happens shortly thereafter).
mWakeLock->release(true);
++mAudioOffloadPauseTimeoutGeneration;
}
status_t NuPlayer::Renderer::onOpenAudioSink(
const sp<AMessage> &format,
bool offloadOnly,
bool hasVideo,
uint32_t flags) {
ALOGV("openAudioSink: offloadOnly(%d) offloadingAudio(%d)",
offloadOnly, offloadingAudio());
bool audioSinkChanged = false;
int32_t numChannels;
CHECK(format->findInt32("channel-count", &numChannels));
int32_t channelMask;
if (!format->findInt32("channel-mask", &channelMask)) {
// signal to the AudioSink to derive the mask from count.
channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
}
int32_t sampleRate;
CHECK(format->findInt32("sample-rate", &sampleRate));
if (offloadingAudio()) {
audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
AString mime;
CHECK(format->findString("mime", &mime));
status_t err = mapMimeToAudioFormat(audioFormat, mime.c_str());
if (err != OK) {
ALOGE("Couldn't map mime \"%s\" to a valid "
"audio_format", mime.c_str());
onDisableOffloadAudio();
} else {
ALOGV("Mime \"%s\" mapped to audio_format 0x%x",
mime.c_str(), audioFormat);
int avgBitRate = -1;
format->findInt32("bitrate", &avgBitRate);
int32_t aacProfile = -1;
if (audioFormat == AUDIO_FORMAT_AAC
&& format->findInt32("aac-profile", &aacProfile)) {
// Redefine AAC format as per aac profile
mapAACProfileToAudioFormat(
audioFormat,
aacProfile);
}
audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
offloadInfo.duration_us = -1;
format->findInt64(
"durationUs", &offloadInfo.duration_us);
offloadInfo.sample_rate = sampleRate;
offloadInfo.channel_mask = channelMask;
offloadInfo.format = audioFormat;
offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
offloadInfo.bit_rate = avgBitRate;
offloadInfo.has_video = hasVideo;
offloadInfo.is_streaming = true;
if (memcmp(&mCurrentOffloadInfo, &offloadInfo, sizeof(offloadInfo)) == 0) {
ALOGV("openAudioSink: no change in offload mode");
// no change from previous configuration, everything ok.
return OK;
}
mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
ALOGV("openAudioSink: try to open AudioSink in offload mode");
uint32_t offloadFlags = flags;
offloadFlags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
offloadFlags &= ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
audioSinkChanged = true;
mAudioSink->close();
err = mAudioSink->open(
sampleRate,
numChannels,
(audio_channel_mask_t)channelMask,
audioFormat,
0 /* bufferCount - unused */,
&NuPlayer::Renderer::AudioSinkCallback,
this,
(audio_output_flags_t)offloadFlags,
&offloadInfo);
if (err == OK) {
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
// will be going through the AudioFlinger mixer
// before reaching the hardware.
// TODO
mCurrentOffloadInfo = offloadInfo;
if (!mPaused) { // for preview mode, don't start if paused
err = mAudioSink->start();
}
ALOGV_IF(err == OK, "openAudioSink: offload succeeded");
}
if (err != OK) {
// Clean up, fall back to non offload mode.
mAudioSink->close();
onDisableOffloadAudio();
mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
ALOGV("openAudioSink: offload failed");
if (offloadOnly) {
notifyAudioTearDown(kForceNonOffload);
}
} else {
mUseAudioCallback = true; // offload mode transfers data through callback
++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message.
}
}
}
if (!offloadOnly && !offloadingAudio()) {
ALOGV("openAudioSink: open AudioSink in NON-offload mode");
uint32_t pcmFlags = flags;
pcmFlags &= ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
const PcmInfo info = {
(audio_channel_mask_t)channelMask,
(audio_output_flags_t)pcmFlags,
AUDIO_FORMAT_PCM_16_BIT, // TODO: change to audioFormat
numChannels,
sampleRate
};
if (memcmp(&mCurrentPcmInfo, &info, sizeof(info)) == 0) {
ALOGV("openAudioSink: no change in pcm mode");
// no change from previous configuration, everything ok.
return OK;
}
audioSinkChanged = true;
mAudioSink->close();
mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
// Note: It is possible to set up the callback, but not use it to send audio data.
// This requires a fix in AudioSink to explicitly specify the transfer mode.
mUseAudioCallback = getUseAudioCallbackSetting();
if (mUseAudioCallback) {
++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message.
}
// Compute the desired buffer size.
// For callback mode, the amount of time before wakeup is about half the buffer size.
const uint32_t frameCount =
(unsigned long long)sampleRate * getAudioSinkPcmMsSetting() / 1000;
// The doNotReconnect means AudioSink will signal back and let NuPlayer to re-construct
// AudioSink. We don't want this when there's video because it will cause a video seek to
// the previous I frame. But we do want this when there's only audio because it will give
// NuPlayer a chance to switch from non-offload mode to offload mode.
// So we only set doNotReconnect when there's no video.
const bool doNotReconnect = !hasVideo;
// We should always be able to set our playback settings if the sink is closed.
LOG_ALWAYS_FATAL_IF(mAudioSink->setPlaybackRate(mPlaybackSettings) != OK,
"onOpenAudioSink: can't set playback rate on closed sink");
status_t err = mAudioSink->open(
sampleRate,
numChannels,
(audio_channel_mask_t)channelMask,
AUDIO_FORMAT_PCM_16_BIT,
0 /* bufferCount - unused */,
mUseAudioCallback ? &NuPlayer::Renderer::AudioSinkCallback : NULL,
mUseAudioCallback ? this : NULL,
(audio_output_flags_t)pcmFlags,
NULL,
doNotReconnect,
frameCount);
if (err != OK) {
ALOGW("openAudioSink: non offloaded open failed status: %d", err);
mAudioSink->close();
mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
return err;
}
mCurrentPcmInfo = info;
if (!mPaused) { // for preview mode, don't start if paused
mAudioSink->start();
}
}
if (audioSinkChanged) {
onAudioSinkChanged();
}
mAudioTornDown = false;
return OK;
}
void NuPlayer::Renderer::onCloseAudioSink() {
mAudioSink->close();
mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
}
} // namespace android