move the APM to chrome.

content/renderer/media/webrtc_audio_processor.h

Index: content/renderer/media/webrtc_audio_processor.h
diff --git a/content/renderer/media/webrtc_audio_processor.h b/content/renderer/media/webrtc_audio_processor.h
new file mode 100644
index 0000000000000000000000000000000000000000..94b5d305ea77fc21e576b37f72f859aef0d1cec7
--- /dev/null
+++ b/content/renderer/media/webrtc_audio_processor.h
@@ -0,0 +1,88 @@
+// 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.
+
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_
+
+#include "base/synchronization/lock.h"
+#include "content/common/content_export.h"
+#include "media/base/audio_converter.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
+#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
+#include "third_party/webrtc/modules/interface/module_common_types.h"
+
+namespace media {
+class AudioBus;
+class AudioFifo;
+class AudioParameters;
+}  // namespace media
+
+namespace webrtc {
+class AudioFrame;
+}
+
+namespace content {
+
+// This class is a wrapper class of webrtc::AudioProcessing.
+class CONTENT_EXPORT WebRtcAudioProcessor {
+ public:
+  explicit WebRtcAudioProcessor(
+      const webrtc::MediaConstraintsInterface* constraints);
+  ~WebRtcAudioProcessor();
+
+  // TODO(xians): Add comment.
+  void SetFormat(const media::AudioParameters& source_params);
+
+  void Push(media::AudioBus* audio_source);

[Henrik Grunell, 2013/10/31 11:56:12]

Add comment.

+
+  // Returns true if it has 10ms data for processing, otherwise false.
+  bool ProcessAndConsume10MsData(int capture_audio_delay_ms,
+                                 int volume,
+                                 bool key_pressed);
+
+  const int16* OutputBuffer() const;
+  const media::AudioParameters& OutputFormat() const;
+
+  // Feed render audio to AudioProcessing for analysis. This is needed
+  // if and only if echo processing is enabled.
+  void FeedRenderDataToAudioProcessing(const int16* render_audio,
+                                       int sample_rate,
+                                       int number_of_channels,
+                                       int number_of_frames,
+                                       int render_delay_ms);
+
+  bool has_audio_processing() const { return audio_processing_.get() != NULL; }
+
+ private:
+  class WebRtcAudioConverter;
+
+  void InitializeAudioProcessingModule(
+      const webrtc::MediaConstraintsInterface* constraints);
+  void InitializeRenderConverterIfNeeded(int sample_rate,
+                                         int number_of_channels,
+                                         int frames_per_buffer);
+  // Processes 10ms data.
+  void Process10MsData(int audio_delay_milliseconds,
+                       int volume,
+                       bool key_pressed);
+
+  void StopAudioProcessing();
+
+  // Cached value for the render delay latency.
+  int render_delay_ms_;
+
+  // Protects |render_delay_ms_|.
+  // TODO(xians): Can we get rid of the lock?
+  mutable base::Lock lock_;
+
+  // Hanles processing the audio data.
+  scoped_ptr<webrtc::AudioProcessing> audio_processing_;
+
+  scoped_ptr<WebRtcAudioConverter> capture_converter_;
+  scoped_ptr<WebRtcAudioConverter> render_converter_;
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_PROCESSOR_H_