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audio_renderer_impl.cc
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audio_renderer_impl.cc
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// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/renderers/audio_renderer_impl.h"
#include <math.h>
#include <stddef.h>
#include <algorithm>
#include <memory>
#include <utility>
#include "base/bind.h"
#include "base/callback.h"
#include "base/callback_helpers.h"
#include "base/command_line.h"
#include "base/logging.h"
#include "base/power_monitor/power_monitor.h"
#include "base/single_thread_task_runner.h"
#include "base/time/default_tick_clock.h"
#include "base/trace_event/trace_event.h"
#include "build/build_config.h"
#include "media/base/audio_buffer.h"
#include "media/base/audio_buffer_converter.h"
#include "media/base/audio_latency.h"
#include "media/base/audio_parameters.h"
#include "media/base/bind_to_current_loop.h"
#include "media/base/channel_mixing_matrix.h"
#include "media/base/demuxer_stream.h"
#include "media/base/media_client.h"
#include "media/base/media_log.h"
#include "media/base/renderer_client.h"
#include "media/base/timestamp_constants.h"
#include "media/filters/audio_clock.h"
#include "media/filters/decrypting_demuxer_stream.h"
#if defined(OS_WIN)
#include "media/base/media_switches.h"
#endif // defined(OS_WIN)
namespace media {
AudioRendererImpl::AudioRendererImpl(
const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
media::AudioRendererSink* sink,
const CreateAudioDecodersCB& create_audio_decoders_cb,
MediaLog* media_log)
: task_runner_(task_runner),
expecting_config_changes_(false),
sink_(sink),
media_log_(media_log),
client_(nullptr),
tick_clock_(base::DefaultTickClock::GetInstance()),
last_audio_memory_usage_(0),
last_decoded_sample_rate_(0),
last_decoded_channel_layout_(CHANNEL_LAYOUT_NONE),
is_encrypted_(false),
last_decoded_channels_(0),
playback_rate_(0.0),
state_(kUninitialized),
create_audio_decoders_cb_(create_audio_decoders_cb),
buffering_state_(BUFFERING_HAVE_NOTHING),
rendering_(false),
sink_playing_(false),
pending_read_(false),
received_end_of_stream_(false),
rendered_end_of_stream_(false),
is_suspending_(false),
is_passthrough_(false),
weak_factory_(this) {
DCHECK(create_audio_decoders_cb_);
// Tests may not have a power monitor.
base::PowerMonitor* monitor = base::PowerMonitor::Get();
if (!monitor)
return;
// PowerObserver's must be added and removed from the same thread, but we
// won't remove the observer until we're destructed on |task_runner_| so we
// must post it here if we're on the wrong thread.
if (task_runner_->BelongsToCurrentThread()) {
monitor->AddObserver(this);
} else {
// Safe to post this without a WeakPtr because this class must be destructed
// on the same thread and construction has not completed yet.
task_runner_->PostTask(FROM_HERE,
base::Bind(&base::PowerMonitor::AddObserver,
base::Unretained(monitor), this));
}
// Do not add anything below this line since the above actions are only safe
// as the last lines of the constructor.
}
AudioRendererImpl::~AudioRendererImpl() {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
if (base::PowerMonitor::Get())
base::PowerMonitor::Get()->RemoveObserver(this);
// If Render() is in progress, this call will wait for Render() to finish.
// After this call, the |sink_| will not call back into |this| anymore.
sink_->Stop();
// Trying to track down AudioClock crash, http://crbug.com/674856. If the sink
// hasn't truly stopped above we will fail to acquire the lock. The sink must
// be stopped to avoid destroying the AudioClock while its still being used.
CHECK(lock_.Try());
lock_.Release();
if (!init_cb_.is_null())
base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_ABORT);
}
void AudioRendererImpl::StartTicking() {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK(!rendering_);
rendering_ = true;
// Wait for an eventual call to SetPlaybackRate() to start rendering.
if (playback_rate_ == 0) {
DCHECK(!sink_playing_);
return;
}
StartRendering_Locked();
}
void AudioRendererImpl::StartRendering_Locked() {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK_EQ(state_, kPlaying);
DCHECK(!sink_playing_);
DCHECK_NE(playback_rate_, 0.0);
lock_.AssertAcquired();
sink_playing_ = true;
base::AutoUnlock auto_unlock(lock_);
sink_->Play();
}
void AudioRendererImpl::StopTicking() {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK(rendering_);
rendering_ = false;
// Rendering should have already been stopped with a zero playback rate.
if (playback_rate_ == 0) {
DCHECK(!sink_playing_);
return;
}
StopRendering_Locked();
}
void AudioRendererImpl::StopRendering_Locked() {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK_EQ(state_, kPlaying);
DCHECK(sink_playing_);
lock_.AssertAcquired();
sink_playing_ = false;
base::AutoUnlock auto_unlock(lock_);
sink_->Pause();
stop_rendering_time_ = last_render_time_;
}
void AudioRendererImpl::SetMediaTime(base::TimeDelta time) {
DVLOG(1) << __func__ << "(" << time << ")";
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK(!rendering_);
DCHECK_EQ(state_, kFlushed);
start_timestamp_ = time;
ended_timestamp_ = kInfiniteDuration;
last_render_time_ = stop_rendering_time_ = base::TimeTicks();
first_packet_timestamp_ = kNoTimestamp;
audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
}
base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
base::AutoLock auto_lock(lock_);
// Return the current time based on the known extents of the rendered audio
// data plus an estimate based on the last time those values were calculated.
base::TimeDelta current_media_time = audio_clock_->front_timestamp();
if (!last_render_time_.is_null()) {
current_media_time +=
(tick_clock_->NowTicks() - last_render_time_) * playback_rate_;
if (current_media_time > audio_clock_->back_timestamp())
current_media_time = audio_clock_->back_timestamp();
}
return current_media_time;
}
bool AudioRendererImpl::GetWallClockTimes(
const std::vector<base::TimeDelta>& media_timestamps,
std::vector<base::TimeTicks>* wall_clock_times) {
base::AutoLock auto_lock(lock_);
DCHECK(wall_clock_times->empty());
// When playback is paused (rate is zero), assume a rate of 1.0.
const double playback_rate = playback_rate_ ? playback_rate_ : 1.0;
const bool is_time_moving = sink_playing_ && playback_rate_ &&
!last_render_time_.is_null() &&
stop_rendering_time_.is_null() && !is_suspending_;
// Pre-compute the time until playback of the audio buffer extents, since
// these values are frequently used below.
const base::TimeDelta time_until_front =
audio_clock_->TimeUntilPlayback(audio_clock_->front_timestamp());
const base::TimeDelta time_until_back =
audio_clock_->TimeUntilPlayback(audio_clock_->back_timestamp());
if (media_timestamps.empty()) {
// Return the current media time as a wall clock time while accounting for
// frames which may be in the process of play out.
wall_clock_times->push_back(std::min(
std::max(tick_clock_->NowTicks(), last_render_time_ + time_until_front),
last_render_time_ + time_until_back));
return is_time_moving;
}
wall_clock_times->reserve(media_timestamps.size());
for (const auto& media_timestamp : media_timestamps) {
// When time was or is moving and the requested media timestamp is within
// range of played out audio, we can provide an exact conversion.
if (!last_render_time_.is_null() &&
media_timestamp >= audio_clock_->front_timestamp() &&
media_timestamp <= audio_clock_->back_timestamp()) {
wall_clock_times->push_back(
last_render_time_ + audio_clock_->TimeUntilPlayback(media_timestamp));
continue;
}
base::TimeDelta base_timestamp, time_until_playback;
if (media_timestamp < audio_clock_->front_timestamp()) {
base_timestamp = audio_clock_->front_timestamp();
time_until_playback = time_until_front;
} else {
base_timestamp = audio_clock_->back_timestamp();
time_until_playback = time_until_back;
}
// In practice, most calls will be estimates given the relatively small
// window in which clients can get the actual time.
wall_clock_times->push_back(last_render_time_ + time_until_playback +
(media_timestamp - base_timestamp) /
playback_rate);
}
return is_time_moving;
}
TimeSource* AudioRendererImpl::GetTimeSource() {
return this;
}
void AudioRendererImpl::Flush(const base::Closure& callback) {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK_EQ(state_, kPlaying);
DCHECK(flush_cb_.is_null());
flush_cb_ = callback;
ChangeState_Locked(kFlushing);
if (pending_read_)
return;
ChangeState_Locked(kFlushed);
DoFlush_Locked();
}
void AudioRendererImpl::DoFlush_Locked() {
DCHECK(task_runner_->BelongsToCurrentThread());
lock_.AssertAcquired();
DCHECK(!pending_read_);
DCHECK_EQ(state_, kFlushed);
ended_timestamp_ = kInfiniteDuration;
audio_buffer_stream_->Reset(base::Bind(&AudioRendererImpl::ResetDecoderDone,
weak_factory_.GetWeakPtr()));
}
void AudioRendererImpl::ResetDecoderDone() {
DCHECK(task_runner_->BelongsToCurrentThread());
{
base::AutoLock auto_lock(lock_);
DCHECK_EQ(state_, kFlushed);
DCHECK(!flush_cb_.is_null());
received_end_of_stream_ = false;
rendered_end_of_stream_ = false;
// Flush() may have been called while underflowed/not fully buffered.
if (buffering_state_ != BUFFERING_HAVE_NOTHING)
SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
if (buffer_converter_)
buffer_converter_->Reset();
algorithm_->FlushBuffers();
}
// Changes in buffering state are always posted. Flush callback must only be
// run after buffering state has been set back to nothing.
task_runner_->PostTask(FROM_HERE, base::ResetAndReturn(&flush_cb_));
}
void AudioRendererImpl::StartPlaying() {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK(!sink_playing_);
DCHECK_EQ(state_, kFlushed);
DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING);
DCHECK(!pending_read_) << "Pending read must complete before seeking";
ChangeState_Locked(kPlaying);
AttemptRead_Locked();
}
void AudioRendererImpl::Initialize(DemuxerStream* stream,
CdmContext* cdm_context,
RendererClient* client,
const PipelineStatusCB& init_cb) {
DVLOG(1) << __func__;
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(client);
DCHECK(stream);
DCHECK_EQ(stream->type(), DemuxerStream::AUDIO);
DCHECK(!init_cb.is_null());
DCHECK(state_ == kUninitialized || state_ == kFlushed);
DCHECK(sink_.get());
// Trying to track down AudioClock crash, http://crbug.com/674856.
// Initialize should never be called while Rendering is ongoing. This can lead
// to race conditions between media/audio-device threads.
CHECK(!sink_playing_);
// This lock is not required by Initialize, but failing to acquire the lock
// would indicate a race with the rendering thread, which should not be active
// at this time. This is just extra verification on the |sink_playing_| CHECK
// above. We hold |lock_| while setting |sink_playing_|, but release the lock
// when calling sink_->Pause() to avoid deadlocking with the AudioMixer.
CHECK(lock_.Try());
lock_.Release();
// If we are re-initializing playback (e.g. switching media tracks), stop the
// sink first.
if (state_ == kFlushed) {
sink_->Stop();
audio_clock_.reset();
}
state_ = kInitializing;
client_ = client;
current_decoder_config_ = stream->audio_decoder_config();
DCHECK(current_decoder_config_.IsValidConfig());
auto output_device_info = sink_->GetOutputDeviceInfo();
const AudioParameters& hw_params = output_device_info.output_params();
ChannelLayout hw_channel_layout =
hw_params.IsValid() ? hw_params.channel_layout() : CHANNEL_LAYOUT_NONE;
audio_buffer_stream_ = std::make_unique<AudioBufferStream>(
std::make_unique<AudioBufferStream::StreamTraits>(media_log_,
hw_channel_layout),
task_runner_, create_audio_decoders_cb_, media_log_);
audio_buffer_stream_->set_config_change_observer(base::Bind(
&AudioRendererImpl::OnConfigChange, weak_factory_.GetWeakPtr()));
// Always post |init_cb_| because |this| could be destroyed if initialization
// failed.
init_cb_ = BindToCurrentLoop(init_cb);
AudioCodec codec = stream->audio_decoder_config().codec();
if (auto* mc = GetMediaClient())
is_passthrough_ = mc->IsSupportedBitstreamAudioCodec(codec);
else
is_passthrough_ = false;
expecting_config_changes_ = stream->SupportsConfigChanges();
bool use_stream_params = !expecting_config_changes_ || !hw_params.IsValid() ||
hw_params.format() == AudioParameters::AUDIO_FAKE ||
!sink_->IsOptimizedForHardwareParameters();
if (stream->audio_decoder_config().channel_layout() ==
CHANNEL_LAYOUT_DISCRETE &&
sink_->IsOptimizedForHardwareParameters()) {
use_stream_params = false;
}
// Target ~20ms for our buffer size (which is optimal for power efficiency and
// responsiveness to play/pause events), but if the hardware needs something
// even larger (say for Bluetooth devices) prefer that.
//
// Even if |use_stream_params| is true we should choose a value here based on
// hardware parameters since it affects the initial buffer size used by
// AudioRendererAlgorithm. Too small and we will underflow if the hardware
// asks for a buffer larger than the initial algorithm capacity.
const int preferred_buffer_size =
std::max(2 * stream->audio_decoder_config().samples_per_second() / 100,
hw_params.IsValid() ? hw_params.frames_per_buffer() : 0);
if (is_passthrough_) {
AudioParameters::Format format = AudioParameters::AUDIO_FAKE;
if (codec == kCodecAC3) {
format = AudioParameters::AUDIO_BITSTREAM_AC3;
} else if (codec == kCodecEAC3) {
format = AudioParameters::AUDIO_BITSTREAM_EAC3;
} else {
NOTREACHED();
}
// If we want the precise PCM frame count here, we have to somehow peek the
// audio bitstream and parse the header ahead of time. Instead, we ensure
// audio bus being large enough to accommodate
// kMaxFramesPerCompressedAudioBuffer frames. The real data size and frame
// count for bitstream formats will be carried in additional fields of
// AudioBus.
const int buffer_size =
AudioParameters::kMaxFramesPerCompressedAudioBuffer *
stream->audio_decoder_config().bytes_per_frame();
audio_parameters_.Reset(
format, stream->audio_decoder_config().channel_layout(),
stream->audio_decoder_config().samples_per_second(), buffer_size);
buffer_converter_.reset();
} else if (use_stream_params) {
audio_parameters_.Reset(AudioParameters::AUDIO_PCM_LOW_LATENCY,
stream->audio_decoder_config().channel_layout(),
stream->audio_decoder_config().samples_per_second(),
preferred_buffer_size);
audio_parameters_.set_channels_for_discrete(
stream->audio_decoder_config().channels());
buffer_converter_.reset();
} else {
// To allow for seamless sample rate adaptations (i.e. changes from say
// 16kHz to 48kHz), always resample to the hardware rate.
int sample_rate = hw_params.sample_rate();
// If supported by the OS and the initial sample rate is not too low, let
// the OS level resampler handle resampling for power efficiency.
if (AudioLatency::IsResamplingPassthroughSupported(
AudioLatency::LATENCY_PLAYBACK) &&
stream->audio_decoder_config().samples_per_second() >= 44100) {
sample_rate = stream->audio_decoder_config().samples_per_second();
}
int stream_channel_count = stream->audio_decoder_config().channels();
bool try_supported_channel_layouts = false;
#if defined(OS_WIN)
try_supported_channel_layouts =
base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kTrySupportedChannelLayouts);
#endif
// We don't know how to up-mix for DISCRETE layouts (fancy multichannel
// hardware with non-standard speaker arrangement). Instead, pretend the
// hardware layout is stereo and let the OS take care of further up-mixing
// to the discrete layout (http://crbug.com/266674). Additionally, pretend
// hardware is stereo whenever kTrySupportedChannelLayouts is set. This flag
// is for savvy users who want stereo content to output in all surround
// speakers. Using the actual layout (likely 5.1 or higher) will mean our
// mixer will attempt to up-mix stereo source streams to just the left/right
// speaker of the 5.1 setup, nulling out the other channels
// (http://crbug.com/177872).
ChannelLayout hw_channel_layout =
hw_params.channel_layout() == CHANNEL_LAYOUT_DISCRETE ||
try_supported_channel_layouts
? CHANNEL_LAYOUT_STEREO
: hw_params.channel_layout();
int hw_channel_count = ChannelLayoutToChannelCount(hw_channel_layout);
// The layout we pass to |audio_parameters_| will be used for the lifetime
// of this audio renderer, regardless of changes to hardware and/or stream
// properties. Below we choose the max of stream layout vs. hardware layout
// to leave room for changes to the hardware and/or stream (i.e. avoid
// premature down-mixing - http://crbug.com/379288).
// If stream_channels < hw_channels:
// Taking max means we up-mix to hardware layout. If stream later changes
// to have more channels, we aren't locked into down-mixing to the
// initial stream layout.
// If stream_channels > hw_channels:
// We choose to output stream's layout, meaning mixing is a no-op for the
// renderer. Browser-side will down-mix to the hardware config. If the
// hardware later changes to equal stream channels, browser-side will stop
// down-mixing and use the data from all stream channels.
ChannelLayout renderer_channel_layout =
hw_channel_count > stream_channel_count
? hw_channel_layout
: stream->audio_decoder_config().channel_layout();
audio_parameters_.Reset(hw_params.format(), renderer_channel_layout,
sample_rate,
media::AudioLatency::GetHighLatencyBufferSize(
sample_rate, preferred_buffer_size));
}
audio_parameters_.set_latency_tag(AudioLatency::LATENCY_PLAYBACK);
last_decoded_channel_layout_ =
stream->audio_decoder_config().channel_layout();
is_encrypted_ = stream->audio_decoder_config().is_encrypted();
last_decoded_channels_ = stream->audio_decoder_config().channels();
audio_clock_.reset(
new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate()));
audio_buffer_stream_->Initialize(
stream, base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
weak_factory_.GetWeakPtr()),
cdm_context, base::Bind(&AudioRendererImpl::OnStatisticsUpdate,
weak_factory_.GetWeakPtr()),
base::Bind(&AudioRendererImpl::OnWaitingForDecryptionKey,
weak_factory_.GetWeakPtr()));
}
void AudioRendererImpl::OnAudioBufferStreamInitialized(bool success) {
DVLOG(1) << __func__ << ": " << success;
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
if (!success) {
state_ = kUninitialized;
base::ResetAndReturn(&init_cb_).Run(DECODER_ERROR_NOT_SUPPORTED);
return;
}
if (!audio_parameters_.IsValid()) {
DVLOG(1) << __func__ << ": Invalid audio parameters: "
<< audio_parameters_.AsHumanReadableString();
ChangeState_Locked(kUninitialized);
// TODO(flim): If the channel layout is discrete but channel count is 0, a
// possible cause is that the input stream has > 8 channels but there is no
// Web Audio renderer attached and no channel mixing matrices defined for
// hardware renderers. Adding one for previewing content could be useful.
base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
return;
}
if (expecting_config_changes_)
buffer_converter_.reset(new AudioBufferConverter(audio_parameters_));
// We're all good! Continue initializing the rest of the audio renderer
// based on the decoder format.
algorithm_.reset(new AudioRendererAlgorithm());
algorithm_->Initialize(audio_parameters_, is_encrypted_);
ConfigureChannelMask();
ChangeState_Locked(kFlushed);
{
base::AutoUnlock auto_unlock(lock_);
sink_->Initialize(audio_parameters_, this);
sink_->Start();
// Some sinks play on start...
sink_->Pause();
}
DCHECK(!sink_playing_);
base::ResetAndReturn(&init_cb_).Run(PIPELINE_OK);
}
void AudioRendererImpl::OnPlaybackError(PipelineStatus error) {
DCHECK(task_runner_->BelongsToCurrentThread());
client_->OnError(error);
}
void AudioRendererImpl::OnPlaybackEnded() {
DCHECK(task_runner_->BelongsToCurrentThread());
client_->OnEnded();
}
void AudioRendererImpl::OnStatisticsUpdate(const PipelineStatistics& stats) {
DCHECK(task_runner_->BelongsToCurrentThread());
client_->OnStatisticsUpdate(stats);
}
void AudioRendererImpl::OnBufferingStateChange(BufferingState state) {
DCHECK(task_runner_->BelongsToCurrentThread());
media_log_->AddEvent(media_log_->CreateBufferingStateChangedEvent(
"audio_buffering_state", state));
client_->OnBufferingStateChange(state);
}
void AudioRendererImpl::OnWaitingForDecryptionKey() {
DCHECK(task_runner_->BelongsToCurrentThread());
client_->OnWaitingForDecryptionKey();
}
void AudioRendererImpl::SetVolume(float volume) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(sink_.get());
sink_->SetVolume(volume);
}
void AudioRendererImpl::OnSuspend() {
base::AutoLock auto_lock(lock_);
is_suspending_ = true;
}
void AudioRendererImpl::OnResume() {
base::AutoLock auto_lock(lock_);
is_suspending_ = false;
}
void AudioRendererImpl::SetPlayDelayCBForTesting(PlayDelayCBForTesting cb) {
DCHECK_EQ(state_, kUninitialized);
play_delay_cb_for_testing_ = std::move(cb);
}
void AudioRendererImpl::DecodedAudioReady(
AudioBufferStream::Status status,
const scoped_refptr<AudioBuffer>& buffer) {
DVLOG(2) << __func__ << "(" << status << ")";
DCHECK(task_runner_->BelongsToCurrentThread());
base::AutoLock auto_lock(lock_);
DCHECK(state_ != kUninitialized);
CHECK(pending_read_);
pending_read_ = false;
if (status == AudioBufferStream::ABORTED ||
status == AudioBufferStream::DEMUXER_READ_ABORTED) {
HandleAbortedReadOrDecodeError(PIPELINE_OK);
return;
}
if (status == AudioBufferStream::DECODE_ERROR) {
HandleAbortedReadOrDecodeError(PIPELINE_ERROR_DECODE);
return;
}
DCHECK_EQ(status, AudioBufferStream::OK);
DCHECK(buffer.get());
if (state_ == kFlushing) {
ChangeState_Locked(kFlushed);
DoFlush_Locked();
return;
}
bool need_another_buffer = true;
if (expecting_config_changes_) {
if (!buffer->end_of_stream()) {
if (last_decoded_sample_rate_ &&
buffer->sample_rate() != last_decoded_sample_rate_) {
DVLOG(1) << __func__ << " Updating audio sample_rate."
<< " ts:" << buffer->timestamp().InMicroseconds()
<< " old:" << last_decoded_sample_rate_
<< " new:" << buffer->sample_rate();
// Send a bogus config to reset timestamp state.
OnConfigChange(AudioDecoderConfig());
}
last_decoded_sample_rate_ = buffer->sample_rate();
if (last_decoded_channel_layout_ != buffer->channel_layout()) {
if (buffer->channel_layout() == CHANNEL_LAYOUT_DISCRETE) {
MEDIA_LOG(ERROR, media_log_)
<< "Unsupported midstream configuration change! Discrete channel"
<< " layout not allowed by sink.";
HandleAbortedReadOrDecodeError(PIPELINE_ERROR_DECODE);
return;
} else {
last_decoded_channel_layout_ = buffer->channel_layout();
last_decoded_channels_ = buffer->channel_count();
ConfigureChannelMask();
}
}
}
DCHECK(buffer_converter_);
buffer_converter_->AddInput(buffer);
while (buffer_converter_->HasNextBuffer()) {
need_another_buffer =
HandleDecodedBuffer_Locked(buffer_converter_->GetNextBuffer());
}
} else {
// TODO(chcunningham, tguilbert): Figure out if we want to support implicit
// config changes during src=. Doing so requires resampling each individual
// stream which is inefficient when there are many tags in a page.
//
// Check if the buffer we received matches the expected configuration.
// Note: We explicitly do not check channel layout here to avoid breaking
// weird behavior with multichannel wav files: http://crbug.com/600538.
if (!buffer->end_of_stream() &&
(buffer->sample_rate() != audio_parameters_.sample_rate() ||
buffer->channel_count() != audio_parameters_.channels())) {
MEDIA_LOG(ERROR, media_log_)
<< "Unsupported midstream configuration change!"
<< " Sample Rate: " << buffer->sample_rate() << " vs "
<< audio_parameters_.sample_rate()
<< ", Channels: " << buffer->channel_count() << " vs "
<< audio_parameters_.channels();
HandleAbortedReadOrDecodeError(PIPELINE_ERROR_DECODE);
return;
}
need_another_buffer = HandleDecodedBuffer_Locked(buffer);
}
if (!need_another_buffer && !CanRead_Locked())
return;
AttemptRead_Locked();
}
bool AudioRendererImpl::HandleDecodedBuffer_Locked(
const scoped_refptr<AudioBuffer>& buffer) {
lock_.AssertAcquired();
if (buffer->end_of_stream()) {
received_end_of_stream_ = true;
} else {
if (buffer->IsBitstreamFormat() && state_ == kPlaying) {
if (IsBeforeStartTime(buffer))
return true;
// Adjust the start time since we are unable to trim a compressed audio
// buffer.
if (buffer->timestamp() < start_timestamp_ &&
(buffer->timestamp() + buffer->duration()) > start_timestamp_) {
start_timestamp_ = buffer->timestamp();
audio_clock_.reset(new AudioClock(buffer->timestamp(),
audio_parameters_.sample_rate()));
}
} else if (state_ == kPlaying) {
if (IsBeforeStartTime(buffer))
return true;
// Trim off any additional time before the start timestamp.
const base::TimeDelta trim_time = start_timestamp_ - buffer->timestamp();
if (trim_time > base::TimeDelta()) {
buffer->TrimStart(buffer->frame_count() *
(static_cast<double>(trim_time.InMicroseconds()) /
buffer->duration().InMicroseconds()));
buffer->set_timestamp(start_timestamp_);
}
// If the entire buffer was trimmed, request a new one.
if (!buffer->frame_count())
return true;
}
if (state_ != kUninitialized)
algorithm_->EnqueueBuffer(buffer);
}
// Store the timestamp of the first packet so we know when to start actual
// audio playback.
if (first_packet_timestamp_ == kNoTimestamp)
first_packet_timestamp_ = buffer->timestamp();
const size_t memory_usage = algorithm_->GetMemoryUsage();
PipelineStatistics stats;
stats.audio_memory_usage = memory_usage - last_audio_memory_usage_;
last_audio_memory_usage_ = memory_usage;
task_runner_->PostTask(FROM_HERE,
base::Bind(&AudioRendererImpl::OnStatisticsUpdate,
weak_factory_.GetWeakPtr(), stats));
switch (state_) {
case kUninitialized:
case kInitializing:
case kFlushing:
NOTREACHED();
return false;
case kFlushed:
DCHECK(!pending_read_);
return false;
case kPlaying:
if (buffer->end_of_stream() || algorithm_->IsQueueFull()) {
if (buffering_state_ == BUFFERING_HAVE_NOTHING)
SetBufferingState_Locked(BUFFERING_HAVE_ENOUGH);
return false;
}
return true;
}
return false;
}
void AudioRendererImpl::AttemptRead() {
base::AutoLock auto_lock(lock_);
AttemptRead_Locked();
}
void AudioRendererImpl::AttemptRead_Locked() {
DCHECK(task_runner_->BelongsToCurrentThread());
lock_.AssertAcquired();
if (!CanRead_Locked())
return;
pending_read_ = true;
audio_buffer_stream_->Read(base::Bind(&AudioRendererImpl::DecodedAudioReady,
weak_factory_.GetWeakPtr()));
}
bool AudioRendererImpl::CanRead_Locked() {
lock_.AssertAcquired();
switch (state_) {
case kUninitialized:
case kInitializing:
case kFlushing:
case kFlushed:
return false;
case kPlaying:
break;
}
return !pending_read_ && !received_end_of_stream_ &&
!algorithm_->IsQueueFull();
}
void AudioRendererImpl::SetPlaybackRate(double playback_rate) {
DVLOG(1) << __func__ << "(" << playback_rate << ")";
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK_GE(playback_rate, 0);
DCHECK(sink_.get());
base::AutoLock auto_lock(lock_);
if (is_passthrough_ && playback_rate != 0 && playback_rate != 1) {
MEDIA_LOG(INFO, media_log_) << "Playback rate changes are not supported "
"when output compressed bitstream."
<< " Playback Rate: " << playback_rate;
return;
}
// We have two cases here:
// Play: current_playback_rate == 0 && playback_rate != 0
// Pause: current_playback_rate != 0 && playback_rate == 0
double current_playback_rate = playback_rate_;
playback_rate_ = playback_rate;
if (!rendering_)
return;
if (current_playback_rate == 0 && playback_rate != 0) {
StartRendering_Locked();
return;
}
if (current_playback_rate != 0 && playback_rate == 0) {
StopRendering_Locked();
return;
}
}
bool AudioRendererImpl::IsBeforeStartTime(
const scoped_refptr<AudioBuffer>& buffer) {
DCHECK_EQ(state_, kPlaying);
return buffer.get() && !buffer->end_of_stream() &&
(buffer->timestamp() + buffer->duration()) < start_timestamp_;
}
int AudioRendererImpl::Render(base::TimeDelta delay,
base::TimeTicks delay_timestamp,
int prior_frames_skipped,
AudioBus* audio_bus) {
TRACE_EVENT1("media", "AudioRendererImpl::Render", "id", media_log_->id());
int frames_requested = audio_bus->frames();
DVLOG(4) << __func__ << " delay:" << delay
<< " prior_frames_skipped:" << prior_frames_skipped
<< " frames_requested:" << frames_requested;
int frames_written = 0;
{
base::AutoLock auto_lock(lock_);
last_render_time_ = tick_clock_->NowTicks();
int64_t frames_delayed = AudioTimestampHelper::TimeToFrames(
delay, audio_parameters_.sample_rate());
if (!stop_rendering_time_.is_null()) {
audio_clock_->CompensateForSuspendedWrites(
last_render_time_ - stop_rendering_time_, frames_delayed);
stop_rendering_time_ = base::TimeTicks();
}
// Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
if (!algorithm_) {
audio_clock_->WroteAudio(0, frames_requested, frames_delayed,
playback_rate_);
return 0;
}
if (playback_rate_ == 0 || is_suspending_) {
audio_clock_->WroteAudio(0, frames_requested, frames_delayed,
playback_rate_);
return 0;
}
// Mute audio by returning 0 when not playing.
if (state_ != kPlaying) {
audio_clock_->WroteAudio(0, frames_requested, frames_delayed,
playback_rate_);
return 0;
}
if (is_passthrough_ && algorithm_->frames_buffered() > 0) {
// TODO(tsunghung): For compressed bitstream formats, play zeroed buffer
// won't generate delay. It could be discarded immediately. Need another
// way to generate audio delay.
const base::TimeDelta play_delay =
first_packet_timestamp_ - audio_clock_->back_timestamp();
if (play_delay > base::TimeDelta()) {
MEDIA_LOG(ERROR, media_log_)
<< "Cannot add delay for compressed audio bitstream foramt."
<< " Requested delay: " << play_delay;
}
frames_written += algorithm_->FillBuffer(audio_bus, 0, frames_requested,
playback_rate_);
// See Initialize(), the |audio_bus| should be bigger than we need in
// bitstream cases. Fix |frames_requested| to avoid incorrent time
// calculation of |audio_clock_| below.
frames_requested = frames_written;
} else if (algorithm_->frames_buffered() > 0) {
// Delay playback by writing silence if we haven't reached the first
// timestamp yet; this can occur if the video starts before the audio.
CHECK_NE(first_packet_timestamp_, kNoTimestamp);
CHECK_GE(first_packet_timestamp_, base::TimeDelta());
const base::TimeDelta play_delay =
first_packet_timestamp_ - audio_clock_->back_timestamp();
if (play_delay > base::TimeDelta()) {
DCHECK_EQ(frames_written, 0);
if (!play_delay_cb_for_testing_.is_null())
play_delay_cb_for_testing_.Run(play_delay);
// Don't multiply |play_delay| out since it can be a huge value on
// poorly encoded media and multiplying by the sample rate could cause
// the value to overflow.
if (play_delay.InSecondsF() > static_cast<double>(frames_requested) /
audio_parameters_.sample_rate()) {
frames_written = frames_requested;
} else {
frames_written =
play_delay.InSecondsF() * audio_parameters_.sample_rate();
}
audio_bus->ZeroFramesPartial(0, frames_written);
}
// If there's any space left, actually render the audio; this is where the
// aural magic happens.
if (frames_written < frames_requested) {
frames_written += algorithm_->FillBuffer(
audio_bus, frames_written, frames_requested - frames_written,
playback_rate_);
}
}
// We use the following conditions to determine end of playback:
// 1) Algorithm can not fill the audio callback buffer
// 2) We received an end of stream buffer
// 3) We haven't already signalled that we've ended
// 4) We've played all known audio data sent to hardware
//
// We use the following conditions to determine underflow:
// 1) Algorithm can not fill the audio callback buffer
// 2) We have NOT received an end of stream buffer
// 3) We are in the kPlaying state
//
// Otherwise the buffer has data we can send to the device.
//
// Per the TimeSource API the media time should always increase even after
// we've rendered all known audio data. Doing so simplifies scenarios where