Move OnDecoderInitDone() from decoder to pipeline thread.

media/filters/audio_renderer_impl.cc

 // 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/filters/audio_renderer_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/callback.h"
 #include "base/callback_helpers.h"
 #include "base/command_line.h"
 #include "base/logging.h"
+#include "base/message_loop_proxy.h"
 #include "media/audio/audio_util.h"
+#include "media/base/bind_to_loop.h"
 #include "media/base/demuxer_stream.h"
 #include "media/base/media_switches.h"
 
 namespace media {
 
 AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink)
-    : state_(kUninitialized),
+    : sink_(sink),
+      state_(kUninitialized),
       pending_read_(false),
       received_end_of_stream_(false),
       rendered_end_of_stream_(false),
       audio_time_buffered_(kNoTimestamp()),
       current_time_(kNoTimestamp()),
-      bytes_per_frame_(0),
-      stopped_(false),
-      sink_(sink),
       underflow_disabled_(false),
-      preroll_aborted_(false) {
+      preroll_aborted_(false),
+      actual_frames_per_buffer_(0) {
+  // We're created on the render thread, but this thread checker is for another.
+  pipeline_thread_checker_.DetachFromThread();
 }
 
 void AudioRendererImpl::Play(const base::Closure& callback) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+
+  float playback_rate = 0;
   {
     base::AutoLock auto_lock(lock_);
     DCHECK_EQ(kPaused, state_);
     state_ = kPlaying;
     callback.Run();
+    playback_rate = algorithm_->playback_rate();
   }
 
-  if (stopped_)
-    return;
-
-  if (GetPlaybackRate() != 0.0f) {
+  if (playback_rate != 0.0f) {
     DoPlay();
   } else {
     DoPause();
   }
 }
 
 void AudioRendererImpl::DoPlay() {
-  earliest_end_time_ = base::Time::Now();
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+  DCHECK(sink_);
+  {
+    base::AutoLock auto_lock(lock_);
+    earliest_end_time_ = base::Time::Now();
+  }
   sink_->Play();
 }
 
 void AudioRendererImpl::Pause(const base::Closure& callback) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+
   {
     base::AutoLock auto_lock(lock_);
     DCHECK(state_ == kPlaying || state_ == kUnderflow ||
            state_ == kRebuffering);
     pause_cb_ = callback;
     state_ = kPaused;
 
     // Pause only when we've completed our pending read.
     if (!pending_read_)
       base::ResetAndReturn(&pause_cb_).Run();
   }
 
-  if (stopped_)
-    return;
-
   DoPause();
 }
 
 void AudioRendererImpl::DoPause() {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+  DCHECK(sink_);
   sink_->Pause(false);
 }
 
 void AudioRendererImpl::Flush(const base::Closure& callback) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   decoder_->Reset(callback);
 }
 
 void AudioRendererImpl::Stop(const base::Closure& callback) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   DCHECK(!callback.is_null());
 
+  if (sink_) {
+    sink_->Stop();
+    sink_ = NULL;
+  }
+
   {
     base::AutoLock auto_lock(lock_);
-    if (!stopped_) {
-      sink_->Stop();
-      stopped_ = true;
-    }
-
     state_ = kStopped;
     algorithm_.reset(NULL);
     init_cb_.Reset();
     underflow_cb_.Reset();
     time_cb_.Reset();
   }
 
   callback.Run();
 }
 
 void AudioRendererImpl::Preroll(base::TimeDelta time,
                                 const PipelineStatusCB& cb) {
-  base::AutoLock auto_lock(lock_);
-  DCHECK_EQ(kPaused, state_);
-  DCHECK(!pending_read_) << "Pending read must complete before seeking";
-  DCHECK(pause_cb_.is_null());
-  DCHECK(preroll_cb_.is_null());
-  state_ = kPrerolling;
-  preroll_cb_ = cb;
-  preroll_timestamp_ = time;
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+  DCHECK(sink_);
 
-  // Throw away everything and schedule our reads.
-  audio_time_buffered_ = kNoTimestamp();
-  current_time_ = kNoTimestamp();
-  received_end_of_stream_ = false;
-  rendered_end_of_stream_ = false;
-  preroll_aborted_ = false;
+  {
+    base::AutoLock auto_lock(lock_);
+    DCHECK_EQ(kPaused, state_);
+    DCHECK(!pending_read_) << "Pending read must complete before seeking";
+    DCHECK(pause_cb_.is_null());
+    DCHECK(preroll_cb_.is_null());
+    state_ = kPrerolling;
+    preroll_cb_ = cb;
+    preroll_timestamp_ = time;
 
-  // |algorithm_| will request more reads.
-  algorithm_->FlushBuffers();
+    // Throw away everything and schedule our reads.
+    audio_time_buffered_ = kNoTimestamp();
+    current_time_ = kNoTimestamp();
+    received_end_of_stream_ = false;
+    rendered_end_of_stream_ = false;
+    preroll_aborted_ = false;
 
-  if (stopped_)
-    return;
+    // |algorithm_| will request more reads.
+    algorithm_->FlushBuffers();
+    earliest_end_time_ = base::Time::Now();
+  }
 
   // Pause and flush the stream when we preroll to a new location.
-  earliest_end_time_ = base::Time::Now();
   sink_->Pause(true);
 }
 
 void AudioRendererImpl::Initialize(const scoped_refptr<DemuxerStream>& stream,
                                    const AudioDecoderList& decoders,
                                    const PipelineStatusCB& init_cb,
                                    const StatisticsCB& statistics_cb,
                                    const base::Closure& underflow_cb,
                                    const TimeCB& time_cb,
                                    const base::Closure& ended_cb,
                                    const base::Closure& disabled_cb,
                                    const PipelineStatusCB& error_cb) {
-  base::AutoLock auto_lock(lock_);
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   DCHECK(stream);
   DCHECK(!decoders.empty());
   DCHECK_EQ(stream->type(), DemuxerStream::AUDIO);
   DCHECK(!init_cb.is_null());
   DCHECK(!statistics_cb.is_null());
   DCHECK(!underflow_cb.is_null());
   DCHECK(!time_cb.is_null());
   DCHECK(!ended_cb.is_null());
   DCHECK(!disabled_cb.is_null());
   DCHECK(!error_cb.is_null());
   DCHECK_EQ(kUninitialized, state_);
+  DCHECK(sink_);
 
   init_cb_ = init_cb;
   statistics_cb_ = statistics_cb;
   underflow_cb_ = underflow_cb;
   time_cb_ = time_cb;
   ended_cb_ = ended_cb;
   disabled_cb_ = disabled_cb;
   error_cb_ = error_cb;
 
   scoped_ptr<AudioDecoderList> decoder_list(new AudioDecoderList(decoders));
   InitializeNextDecoder(stream, decoder_list.Pass());
 }
 
 void AudioRendererImpl::InitializeNextDecoder(
     const scoped_refptr<DemuxerStream>& demuxer_stream,
     scoped_ptr<AudioDecoderList> decoders) {
-  lock_.AssertAcquired();
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   DCHECK(!decoders->empty());
 
   scoped_refptr<AudioDecoder> decoder = decoders->front();
   decoders->pop_front();
 
   DCHECK(decoder);
   decoder_ = decoder;
-
-  base::AutoUnlock auto_unlock(lock_);
   decoder->Initialize(
-      demuxer_stream,
-      base::Bind(&AudioRendererImpl::OnDecoderInitDone, this,
-                 demuxer_stream,
-                 base::Passed(&decoders)),
+      demuxer_stream, BindToLoop(base::MessageLoopProxy::current(), base::Bind(
+          &AudioRendererImpl::OnDecoderInitDone, this, demuxer_stream,
+          base::Passed(&decoders))),
       statistics_cb_);
 }
 
 void AudioRendererImpl::OnDecoderInitDone(
     const scoped_refptr<DemuxerStream>& demuxer_stream,
     scoped_ptr<AudioDecoderList> decoders,
     PipelineStatus status) {
-  base::AutoLock auto_lock(lock_);
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
 
   if (state_ == kStopped) {
-    DCHECK(stopped_);
+    DCHECK(!sink_);
     return;
   }
 
   if (!decoders->empty() && status == DECODER_ERROR_NOT_SUPPORTED) {
     InitializeNextDecoder(demuxer_stream, decoders.Pass());
     return;
   }
 
   if (status != PIPELINE_OK) {
     base::ResetAndReturn(&init_cb_).Run(status);
     return;
   }
 
   // We're all good! Continue initializing the rest of the audio renderer based
   // on the decoder format.
 
   ChannelLayout channel_layout = decoder_->channel_layout();
   int channels = ChannelLayoutToChannelCount(channel_layout);
   int bits_per_channel = decoder_->bits_per_channel();
   int sample_rate = decoder_->samples_per_second();
-  // TODO(vrk): Add method to AudioDecoder to compute bytes per frame.
-  bytes_per_frame_ = channels * bits_per_channel / 8;
 
   algorithm_.reset(new AudioRendererAlgorithm());
   if (!algorithm_->ValidateConfig(channels, sample_rate, bits_per_channel)) {
     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
     return;
   }
 
   algorithm_->Initialize(
       channels, sample_rate, bits_per_channel, 0.0f,
       base::Bind(&AudioRendererImpl::ScheduleRead_Locked, this));
@@ skipping 30 matching lines @@
     // purposes it's okay that this buffer size might lead to jitter since it's
     // not a multiple of the hardware buffer size.
     format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
     buffer_size = 2048;
   }
 #endif
 
   audio_parameters_ = AudioParameters(
       format, channel_layout, sample_rate, bits_per_channel, buffer_size);
 
+  state_ = kPaused;
+
   sink_->Initialize(audio_parameters_, this);
   sink_->Start();
 
-  state_ = kPaused;
   base::ResetAndReturn(&init_cb_).Run(PIPELINE_OK);
 }
 
 void AudioRendererImpl::ResumeAfterUnderflow(bool buffer_more_audio) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   if (state_ == kUnderflow) {
     // The "&& preroll_aborted_" is a hack. If preroll is aborted, then we
     // shouldn't even reach the kUnderflow state to begin with. But for now
     // we're just making sure that the audio buffer capacity (i.e. the
     // number of bytes that need to be buffered for preroll to complete)
     // does not increase due to an aborted preroll.
     // TODO(vrk): Fix this bug correctly! (crbug.com/151352)
     if (buffer_more_audio && !preroll_aborted_)
       algorithm_->IncreaseQueueCapacity();
 
     state_ = kRebuffering;
   }
 }
 
 void AudioRendererImpl::SetVolume(float volume) {
-  if (stopped_)
-    return;
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
+  DCHECK(sink_);
   sink_->SetVolume(volume);
 }
 
 AudioRendererImpl::~AudioRendererImpl() {
   // Stop() should have been called and |algorithm_| should have been destroyed.
   DCHECK(state_ == kUninitialized || state_ == kStopped);
   DCHECK(!algorithm_.get());
 }
 
 void AudioRendererImpl::DecodedAudioReady(AudioDecoder::Status status,
@@ skipping 63 matching lines @@
 
 void AudioRendererImpl::ScheduleRead_Locked() {
   lock_.AssertAcquired();
   if (pending_read_ || state_ == kPaused)
     return;
   pending_read_ = true;
   decoder_->Read(base::Bind(&AudioRendererImpl::DecodedAudioReady, this));
 }
 
 void AudioRendererImpl::SetPlaybackRate(float playback_rate) {
+  DCHECK(pipeline_thread_checker_.CalledOnValidThread());
   DCHECK_LE(0.0f, playback_rate);
+  DCHECK(sink_);
 
-  if (!stopped_) {
-    // We have two cases here:
-    // Play: GetPlaybackRate() == 0.0 && playback_rate != 0.0
-    // Pause: GetPlaybackRate() != 0.0 && playback_rate == 0.0
-    if (GetPlaybackRate() == 0.0f && playback_rate != 0.0f) {
-      DoPlay();
-    } else if (GetPlaybackRate() != 0.0f && playback_rate == 0.0f) {
-      // Pause is easy, we can always pause.
-      DoPause();
-    }
+  // We have two cases here:
+  // Play: current_playback_rate == 0.0 && playback_rate != 0.0
+  // Pause: current_playback_rate != 0.0 && playback_rate == 0.0
+  float current_playback_rate = algorithm_->playback_rate();
+  if (current_playback_rate == 0.0f && playback_rate != 0.0f) {
+    DoPlay();
+  } else if (current_playback_rate != 0.0f && playback_rate == 0.0f) {
+    // Pause is easy, we can always pause.
+    DoPause();
   }
 
   base::AutoLock auto_lock(lock_);
   algorithm_->SetPlaybackRate(playback_rate);
 }
 
-float AudioRendererImpl::GetPlaybackRate() {
-  base::AutoLock auto_lock(lock_);
-  return algorithm_->playback_rate();
-}
-
 bool AudioRendererImpl::IsBeforePrerollTime(
     const scoped_refptr<Buffer>& buffer) {
   return (state_ == kPrerolling) && buffer && !buffer->IsEndOfStream() &&
       (buffer->GetTimestamp() + buffer->GetDuration()) < preroll_timestamp_;
 }
 
 int AudioRendererImpl::Render(AudioBus* audio_bus,
                               int audio_delay_milliseconds) {
-  if (stopped_ || GetPlaybackRate() == 0.0f) {
-    // Output silence if stopped.
-    audio_bus->Zero();
-    return 0;
+  if (actual_frames_per_buffer_ != audio_bus->frames()) {
+    audio_buffer_.reset(
+        new uint8[audio_bus->frames() * audio_parameters_.GetBytesPerFrame()]);
+    actual_frames_per_buffer_ = audio_bus->frames();
   }
 
-  // Adjust the playback delay.
-  base::TimeDelta request_delay =
-      base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
-
-  // Finally we need to adjust the delay according to playback rate.
-  if (GetPlaybackRate() != 1.0f) {
-    request_delay = base::TimeDelta::FromMicroseconds(
-        static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                GetPlaybackRate())));
-  }
-
-  int bytes_per_frame = audio_parameters_.GetBytesPerFrame();
-
-  const int buf_size = audio_bus->frames() * bytes_per_frame;
-  scoped_array<uint8> buf(new uint8[buf_size]);
-
-  int frames_filled = FillBuffer(buf.get(), audio_bus->frames(), request_delay);
-  int bytes_filled = frames_filled * bytes_per_frame;
-  DCHECK_LE(bytes_filled, buf_size);
-  UpdateEarliestEndTime(bytes_filled, request_delay, base::Time::Now());
+  int frames_filled = FillBuffer(
+      audio_buffer_.get(), audio_bus->frames(), audio_delay_milliseconds);
+  DCHECK_LE(frames_filled, actual_frames_per_buffer_);
 
   // Deinterleave audio data into the output bus.
   audio_bus->FromInterleaved(
-      buf.get(), frames_filled, audio_parameters_.bits_per_sample() / 8);
+      audio_buffer_.get(), frames_filled,
+      audio_parameters_.bits_per_sample() / 8);
 
   return frames_filled;
 }
 
 uint32 AudioRendererImpl::FillBuffer(uint8* dest,
                                      uint32 requested_frames,
-                                     const base::TimeDelta& playback_delay) {
+                                     int audio_delay_milliseconds) {
   base::TimeDelta current_time = kNoTimestamp();
   base::TimeDelta max_time = kNoTimestamp();
 
   size_t frames_written = 0;
   base::Closure underflow_cb;
   {
     base::AutoLock auto_lock(lock_);
 
+    // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
+    if (!algorithm_)
+      return 0;
+
+    float playback_rate = algorithm_->playback_rate();
+    if (playback_rate == 0.0f)
+      return 0;
+
+    // Adjust the delay according to playback rate.
+    base::TimeDelta playback_delay =
+        base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
+    if (playback_rate != 1.0f) {
+      playback_delay = base::TimeDelta::FromMicroseconds(static_cast<int64>(
+          ceil(playback_delay.InMicroseconds() * playback_rate)));
+    }
+
     if (state_ == kRebuffering && algorithm_->IsQueueFull())
       state_ = kPlaying;
 
     // Mute audio by returning 0 when not playing.
     if (state_ != kPlaying) {
       // TODO(scherkus): To keep the audio hardware busy we write at most 8k of
       // zeros.  This gets around the tricky situation of pausing and resuming
       // the audio IPC layer in Chrome.  Ideally, we should return zero and then
       // the subclass can restart the conversation.
       //
       // This should get handled by the subclass http://crbug.com/106600
       const uint32 kZeroLength = 8192;
-      size_t zeros_to_write =
-          std::min(kZeroLength, requested_frames * bytes_per_frame_);
+      size_t zeros_to_write = std::min(
+          kZeroLength, requested_frames * audio_parameters_.GetBytesPerFrame());
       memset(dest, 0, zeros_to_write);
-      return zeros_to_write / bytes_per_frame_;
+      return zeros_to_write / audio_parameters_.GetBytesPerFrame();
     }
 
     // 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) Our estimated earliest end time has expired
     //
     // TODO(enal): we should replace (4) with a check that the browser has no
     // more audio data or at least use a delayed callback.
@@ skipping 45 matching lines @@
       // delay after we've revamped our audio IPC subsystem.
       if (current_time_ > previous_time && !rendered_end_of_stream_) {
         current_time = current_time_;
       }
     }
 
     // The call to FillBuffer() on |algorithm_| has increased the amount of
     // buffered audio data. Update the new amount of time buffered.
     max_time = algorithm_->GetTime();
     audio_time_buffered_ = max_time;
+
+    UpdateEarliestEndTime_Locked(
+        frames_written, playback_rate, playback_delay, base::Time::Now());
   }
 
   if (current_time != kNoTimestamp() && max_time != kNoTimestamp()) {
     time_cb_.Run(current_time, max_time);
   }
 
   if (!underflow_cb.is_null())
     underflow_cb.Run();
 
   return frames_written;
 }
 
-void AudioRendererImpl::UpdateEarliestEndTime(int bytes_filled,
-                                              base::TimeDelta request_delay,
-                                              base::Time time_now) {
-  if (bytes_filled != 0) {
-    base::TimeDelta predicted_play_time = ConvertToDuration(bytes_filled);
-    float playback_rate = GetPlaybackRate();
-    if (playback_rate != 1.0f) {
-      predicted_play_time = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
-                                  playback_rate)));
-    }
-    earliest_end_time_ =
-        std::max(earliest_end_time_,
-                 time_now + request_delay + predicted_play_time);
+void AudioRendererImpl::UpdateEarliestEndTime_Locked(
+    int frames_filled, float playback_rate, base::TimeDelta playback_delay,
+    base::Time time_now) {
+  if (frames_filled <= 0)
+    return;
+
+  base::TimeDelta predicted_play_time = base::TimeDelta::FromMicroseconds(
+      static_cast<float>(frames_filled) * base::Time::kMicrosecondsPerSecond /
+      audio_parameters_.sample_rate());
+
+  if (playback_rate != 1.0f) {
+    predicted_play_time = base::TimeDelta::FromMicroseconds(
+        static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
+                                playback_rate)));
   }
-}
 
-base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
-  int bytes_per_second = audio_parameters_.GetBytesPerSecond();
-  CHECK(bytes_per_second);
-  return base::TimeDelta::FromMicroseconds(
-      base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second);
+  lock_.AssertAcquired();
+  earliest_end_time_ = std::max(
+      earliest_end_time_, time_now + playback_delay + predicted_play_time);
 }
 
 void AudioRendererImpl::OnRenderError() {
   disabled_cb_.Run();
 }
 
 void AudioRendererImpl::DisableUnderflowForTesting() {
   underflow_disabled_ = true;
 }
 
@@ skipping 18 matching lines @@
     case kUnderflow:
     case kRebuffering:
     case kStopped:
       if (status != PIPELINE_OK)
         error_cb_.Run(status);
       return;
   }
 }
 
 }  // namespace media