Miscellaneous AudioRendererImpl cleanup salvaged from http://codereview.chromium.org/10918022/.

Review URL: https://chromiumcodereview.appspot.com/11183048

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@162766 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 12 insertions, 38 deletions

diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc
index 20dffd6..51c19df 100644
--- a/media/filters/audio_renderer_impl.cc
+++ b/media/filters/audio_renderer_impl.cc
@@ -25,10 +25,8 @@ AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink)
       audio_time_buffered_(kNoTimestamp()),
       current_time_(kNoTimestamp()),
       bytes_per_frame_(0),
-      bytes_per_second_(0),
       stopped_(false),
       sink_(sink),
-      is_initialized_(false),
       underflow_disabled_(false),
       preroll_aborted_(false) {
 }
@@ -53,7 +51,6 @@ void AudioRendererImpl::Play(const base::Closure& callback) {
 
 void AudioRendererImpl::DoPlay() {
   earliest_end_time_ = base::Time::Now();
-  DCHECK(sink_.get());
   sink_->Play();
 }
 
@@ -77,7 +74,6 @@ void AudioRendererImpl::Pause(const base::Closure& callback) {
 }
 
 void AudioRendererImpl::DoPause() {
-  DCHECK(sink_.get());
   sink_->Pause(false);
 }
 
@@ -89,9 +85,7 @@ void AudioRendererImpl::Stop(const base::Closure& callback) {
   DCHECK(!callback.is_null());
 
   if (!stopped_) {
-    DCHECK(sink_.get());
     sink_->Stop();
-
     stopped_ = true;
   }
   {
@@ -149,6 +143,7 @@ void AudioRendererImpl::Initialize(const scoped_refptr<DemuxerStream>& 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());
@@ -210,14 +205,9 @@ void AudioRendererImpl::OnDecoderInitDone(
     return;
   }
 
-  // Create a callback so our algorithm can request more reads.
-  base::Closure cb = base::Bind(&AudioRendererImpl::ScheduleRead_Locked, this);
-
-  // Construct the algorithm.
-  algorithm_.reset(new AudioRendererAlgorithm());
+  // We're all good! Continue initializing the rest of the audio renderer based
+  // on the decoder format.
 
-  // Initialize our algorithm with media properties, initial playback rate,
-  // and a callback to request more reads from the data source.
   ChannelLayout channel_layout = decoder_->channel_layout();
   int channels = ChannelLayoutToChannelCount(channel_layout);
   int bits_per_channel = decoder_->bits_per_channel();
@@ -225,15 +215,15 @@ void AudioRendererImpl::OnDecoderInitDone(
   // TODO(vrk): Add method to AudioDecoder to compute bytes per frame.
   bytes_per_frame_ = channels * bits_per_channel / 8;
 
-  bool config_ok = algorithm_->ValidateConfig(channels, sample_rate,
-                                              bits_per_channel);
-  if (!config_ok || is_initialized_) {
+  algorithm_.reset(new AudioRendererAlgorithm());
+  if (!algorithm_->ValidateConfig(channels, sample_rate, bits_per_channel)) {
     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
     return;
   }
 
-  if (config_ok)
-    algorithm_->Initialize(channels, sample_rate, bits_per_channel, 0.0f, cb);
+  algorithm_->Initialize(
+      channels, sample_rate, bits_per_channel, 0.0f,
+      base::Bind(&AudioRendererImpl::ScheduleRead_Locked, this));
 
   // We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
   // does not currently support all the sample-rates that we require.
@@ -243,17 +233,9 @@ void AudioRendererImpl::OnDecoderInitDone(
       AudioParameters::AUDIO_PCM_LINEAR, channel_layout, sample_rate,
       bits_per_channel, GetHighLatencyOutputBufferSize(sample_rate));
 
-  bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
-
-  DCHECK(sink_.get());
-  DCHECK(!is_initialized_);
-
   sink_->Initialize(audio_parameters_, this);
-
   sink_->Start();
-  is_initialized_ = true;
 
-  // Finally, execute the start callback.
   state_ = kPaused;
   base::ResetAndReturn(&init_cb_).Run(PIPELINE_OK);
 }
@@ -547,11 +529,10 @@ void AudioRendererImpl::UpdateEarliestEndTime(int bytes_filled,
 }
 
 base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
-  if (bytes_per_second_) {
-    return base::TimeDelta::FromMicroseconds(
-        base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second_);
-  }
-  return base::TimeDelta();
+  int bytes_per_second = audio_parameters_.GetBytesPerSecond();
+  CHECK(bytes_per_second);
+  return base::TimeDelta::FromMicroseconds(
+      base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second);
 }
 
 void AudioRendererImpl::OnRenderError() {
@@ -559,7 +540,6 @@ void AudioRendererImpl::OnRenderError() {
 }
 
 void AudioRendererImpl::DisableUnderflowForTesting() {
-  DCHECK(!is_initialized_);
   underflow_disabled_ = true;
 }
 
diff --git a/media/filters/audio_renderer_impl.h b/media/filters/audio_renderer_impl.h
index b154acf..47a498b 100644
--- a/media/filters/audio_renderer_impl.h
+++ b/media/filters/audio_renderer_impl.h
@@ -197,18 +197,12 @@ class MEDIA_EXPORT AudioRendererImpl
 
   uint32 bytes_per_frame_;
 
-  // Used to calculate audio delay given bytes.
-  uint32 bytes_per_second_;
-
   // A flag that indicates this filter is called to stop.
   bool stopped_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioRendererSink> sink_;
 
-  // Set to true when OnInitialize() is called.
-  bool is_initialized_;
-
   // We're supposed to know amount of audio data OS or hardware buffered, but
   // that is not always so -- on my Linux box
   // AudioBuffersState::hardware_delay_bytes never reaches 0.