Use AudioProcessing::Initialize().

Use this in place of the deprecated set_sample_rate_hz(). Similarly,
switch sample_rate_hz() to input_sample_rate_hz().

TESTED=apprtc sounds as expected with constraints enabled/disabled.

Review URL: https://codereview.chromium.org/259663005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@266320 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 10 insertions, 6 deletions

diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc
index 01cee0c..c82a1b6 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());