Add support for midstream audio configuration changes.

media/filters/audio_renderer_impl.cc

Index: media/filters/audio_renderer_impl.cc
diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc
index 2dab6018b8982cc9ab31afa29a568ecdeddf47c4..7998a8354be26949ee8a9eb32ee984d3d8642cec 100644
--- a/media/filters/audio_renderer_impl.cc
+++ b/media/filters/audio_renderer_impl.cc
@@ -15,6 +15,8 @@
 #include "base/metrics/histogram.h"
 #include "base/single_thread_task_runner.h"
 #include "media/base/audio_buffer.h"
+#include "media/base/audio_buffer_converter.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/audio_splicer.h"
 #include "media/base/bind_to_current_loop.h"
 #include "media/base/demuxer_stream.h"
@@ -41,12 +43,14 @@ AudioRendererImpl::AudioRendererImpl(
     const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
     media::AudioRendererSink* sink,
     ScopedVector<AudioDecoder> decoders,
-    const SetDecryptorReadyCB& set_decryptor_ready_cb)
+    const SetDecryptorReadyCB& set_decryptor_ready_cb,
+    const AudioHardwareConfig& hardware_config)
     : task_runner_(task_runner),
       sink_(sink),
       audio_buffer_stream_(task_runner,
                            decoders.Pass(),
                            set_decryptor_ready_cb),
+      hardware_config_(hardware_config),
       now_cb_(base::Bind(&base::TimeTicks::Now)),
       state_(kUninitialized),
       sink_playing_(false),
@@ -168,6 +172,8 @@ void AudioRendererImpl::ResetDecoderDone() {
 
     earliest_end_time_ = now_cb_.Run();
     splicer_->Reset();
+    // TODO: add a Reset() or Flush()

[rileya (GONE FROM CHROMIUM), 2014/03/20 00:55:11]

This is an immediate TODO for this CL that I forgot to address, I'll do so in
the next patchset.

+    buffer_converter_.reset(new AudioBufferConverter(audio_parameters_));
     algorithm_->FlushBuffers();
   }
   base::ResetAndReturn(&flush_cb_).Run();
@@ -273,27 +279,17 @@ void AudioRendererImpl::OnAudioBufferStreamInitialized(bool success) {
     return;
   }
 
-  int sample_rate = audio_buffer_stream_.decoder()->samples_per_second();
+  // TODO(rileya): Support hardware config changes
+  audio_parameters_ = hardware_config_.GetOutputConfig();
 
-  // The actual buffer size is controlled via the size of the AudioBus
-  // provided to Render(), so just choose something reasonable here for looks.
-  int buffer_size = audio_buffer_stream_.decoder()->samples_per_second() / 100;
-
-  // TODO(rileya): Remove the channel_layout/bits_per_channel/samples_per_second
-  // getters from AudioDecoder, and adjust this accordingly.
-  audio_parameters_ =
-      AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                      audio_buffer_stream_.decoder()->channel_layout(),
-                      sample_rate,
-                      audio_buffer_stream_.decoder()->bits_per_channel(),
-                      buffer_size);
   if (!audio_parameters_.IsValid()) {
     ChangeState_Locked(kUninitialized);
     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
     return;
   }
 
-  splicer_.reset(new AudioSplicer(sample_rate));
+  buffer_converter_.reset(new AudioBufferConverter(audio_parameters_));
+  splicer_.reset(new AudioSplicer(audio_parameters_.sample_rate()));
 
   // We're all good! Continue initializing the rest of the audio renderer
   // based on the decoder format.
@@ -373,19 +369,26 @@ void AudioRendererImpl::DecodedAudioReady(
     return;
   }
 
-  if (!splicer_->AddInput(buffer)) {
-    HandleAbortedReadOrDecodeError(true);
-    return;
-  }
+  buffer_converter_->AddInput(buffer);
+  bool need_another_buffer = false;

[DaleCurtis, 2014/03/20 19:29:01]

Move with while below?

[rileya (GONE FROM CHROMIUM), 2014/03/21 20:58:49]

Done.

 
-  if (!splicer_->HasNextBuffer()) {
+  if (!buffer_converter_->HasNextBuffer()) {
     AttemptRead_Locked();
     return;
   }
 
-  bool need_another_buffer = false;
-  while (splicer_->HasNextBuffer())
-    need_another_buffer = HandleSplicerBuffer(splicer_->GetNextBuffer());
+  while (buffer_converter_->HasNextBuffer()) {
+    scoped_refptr<AudioBuffer> converted_buffer =
+        buffer_converter_->GetNextBuffer();
+
+    if (!splicer_->AddInput(converted_buffer)) {
+      HandleAbortedReadOrDecodeError(true);
+      return;
+    }
+
+    while (splicer_->HasNextBuffer())
+      need_another_buffer = HandleSplicerBuffer(splicer_->GetNextBuffer());
+  }
 
   if (!need_another_buffer && !CanRead_Locked())
     return;
@@ -395,6 +398,7 @@ void AudioRendererImpl::DecodedAudioReady(
 
 bool AudioRendererImpl::HandleSplicerBuffer(
     const scoped_refptr<AudioBuffer>& buffer) {
+

[DaleCurtis, 2014/03/20 19:29:01]

No extra line.

[rileya (GONE FROM CHROMIUM), 2014/03/21 20:58:49]

Done.

   if (buffer->end_of_stream()) {
     received_end_of_stream_ = true;