Index: content/renderer/media/media_stream_audio_processor.cc |
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc |
index 01cee0c6e6a601f2a9541aec8c7dffba381fe7c8..c82a1b654e0d4d05922957d4a32adf5c6c2b0bf1 100644 |
--- a/content/renderer/media/media_stream_audio_processor.cc |
+++ b/content/renderer/media/media_stream_audio_processor.cc |
@@ -33,6 +33,8 @@ const int kAudioProcessingSampleRate = 16000; |
const int kAudioProcessingSampleRate = 32000; |
#endif |
const int kAudioProcessingNumberOfChannels = 1; |
+const AudioProcessing::ChannelLayout kAudioProcessingChannelLayout = |
+ AudioProcessing::kMono; |
const int kMaxNumberOfBuffersInFifo = 2; |
@@ -351,11 +353,13 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule( |
} |
// Create and configure the webrtc::AudioProcessing. |
- audio_processing_.reset(webrtc::AudioProcessing::Create(0)); |
- // TODO(ajm): Replace with AudioProcessing::Initialize() when this rolls to |
- // Chromium: http://review.webrtc.org/9919004/ |
- CHECK_EQ(0, |
- audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate)); |
+ audio_processing_.reset(webrtc::AudioProcessing::Create()); |
+ CHECK_EQ(0, audio_processing_->Initialize(kAudioProcessingSampleRate, |
+ kAudioProcessingSampleRate, |
+ kAudioProcessingSampleRate, |
+ kAudioProcessingChannelLayout, |
+ kAudioProcessingChannelLayout, |
+ kAudioProcessingChannelLayout)); |
// Enable the audio processing components. |
if (enable_aec) { |
@@ -459,7 +463,7 @@ int MediaStreamAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame, |
return 0; |
TRACE_EVENT0("audio", "MediaStreamAudioProcessor::ProcessData"); |
- DCHECK_EQ(audio_processing_->sample_rate_hz(), |
+ DCHECK_EQ(audio_processing_->input_sample_rate_hz(), |
capture_converter_->sink_parameters().sample_rate()); |
DCHECK_EQ(audio_processing_->num_input_channels(), |
capture_converter_->sink_parameters().channels()); |