Use AudioProcessing::Initialize().

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 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 "content/renderer/media/media_stream_audio_processor.h"
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram.h"
@@ skipping 15 matching lines @@
 
 using webrtc::AudioProcessing;
 using webrtc::MediaConstraintsInterface;
 
 #if defined(OS_ANDROID)
 const int kAudioProcessingSampleRate = 16000;
 #else
 const int kAudioProcessingSampleRate = 32000;
 #endif
 const int kAudioProcessingNumberOfChannels = 1;
+const AudioProcessing::ChannelLayout kAudioProcessingChannelLayout =
+    AudioProcessing::kMono;
 
 const int kMaxNumberOfBuffersInFifo = 2;
 
 // Used by UMA histograms and entries shouldn't be re-ordered or removed.
 enum AudioTrackProcessingStates {
   AUDIO_PROCESSING_ENABLED = 0,
   AUDIO_PROCESSING_DISABLED,
   AUDIO_PROCESSING_IN_WEBRTC,
   AUDIO_PROCESSING_MAX
 };
@@ skipping 298 matching lines @@
 
   // Return immediately if no audio processing component is enabled.
   if (!enable_aec && !enable_experimental_aec && !enable_ns &&
       !enable_high_pass_filter && !enable_typing_detection && !enable_agc &&
       !enable_experimental_ns) {
     RecordProcessingState(AUDIO_PROCESSING_DISABLED);
     return;
   }
 
   // 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) {
     EnableEchoCancellation(audio_processing_.get());
     if (enable_experimental_aec)
       EnableExperimentalEchoCancellation(audio_processing_.get());
 
     if (playout_data_source_)
       playout_data_source_->AddPlayoutSink(this);
   }
@@ skipping 83 matching lines @@
 
 int MediaStreamAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame,
                                            base::TimeDelta capture_delay,
                                            int volume,
                                            bool key_pressed) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   if (!audio_processing_)
     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());
   DCHECK_EQ(audio_processing_->num_output_channels(),
             capture_converter_->sink_parameters().channels());
 
   base::subtle::Atomic32 render_delay_ms =
       base::subtle::Acquire_Load(&render_delay_ms_);
   int64 capture_delay_ms = capture_delay.InMilliseconds();
   DCHECK_LT(capture_delay_ms,
@@ skipping 36 matching lines @@
 
   StopAecDump();
 
   if (playout_data_source_)
     playout_data_source_->RemovePlayoutSink(this);
 
   audio_processing_.reset();
 }
 
 }  // namespace content