Revert 237311 "Added an "enable-audio-processor" flag and WebRtc..."

trunk/src/content/renderer/media/media_stream_audio_processor.h

-// 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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
-#define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
-
-#include "base/atomicops.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "base/time/time.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 owns an object of webrtc::AudioProcessing which contains signal
-// processing components like AGC, AEC and NS. It enables the components based
-// on the getUserMedia constraints, processes the data and outputs it in a unit
-// of 10 ms data chunk.
-class CONTENT_EXPORT MediaStreamAudioProcessor {
- public:
-  explicit MediaStreamAudioProcessor(
-      const webrtc::MediaConstraintsInterface* constraints);
-  ~MediaStreamAudioProcessor();
-
-  // Pushes capture data in |audio_source| to the internal FIFO.
-  // Called on the capture audio thread.
-  void PushCaptureData(media::AudioBus* audio_source);
-
-  // Push the render audio to webrtc::AudioProcessing for analysis. This is
-  // needed iff echo processing is enabled.
-  // |render_audio| is the pointer to the render audio data, its format
-  // is specified by |sample_rate|, |number_of_channels| and |number_of_frames|.
-  // Called on the render audio thread.
-  void PushRenderData(const int16* render_audio,
-                      int sample_rate,
-                      int number_of_channels,
-                      int number_of_frames,
-                      base::TimeDelta render_delay);
-
-  // Processes a block of 10 ms data from the internal FIFO and outputs it via
-  // |out|. |out| is the address of the pointer that will be pointed to
-  // the post-processed data if the method is returning a true. The lifetime
-  // of the data represeted by |out| is guaranteed to outlive the method call.
-  // That also says *|out| won't change until this method is called again.
-  // Returns true if the internal FIFO has at least 10 ms data for processing,
-  // otherwise false.
-  // |capture_delay|, |volume| and |key_pressed| will be passed to
-  // webrtc::AudioProcessing to help processing the data.
-  // Called on the capture audio thread.
-  bool ProcessAndConsumeData(base::TimeDelta capture_delay,
-                             int volume,
-                             bool key_pressed,
-                             int16** out);
-
-  // Called when the format of the capture data has changed.
-  // This has to be called before PushCaptureData() and ProcessAndConsumeData().
-  // Called on the main render thread.
-  void SetCaptureFormat(const media::AudioParameters& source_params);
-
-  // The audio format of the output from the processor.
-  const media::AudioParameters& OutputFormat() const;
-
-  // Accessor to check if the audio processing is enabled or not.
-  bool has_audio_processing() const { return audio_processing_.get() != NULL; }
-
- private:
-  class MediaStreamAudioConverter;
-
-  // Helper to initialize the WebRtc AudioProcessing.
-  void InitializeAudioProcessingModule(
-      const webrtc::MediaConstraintsInterface* constraints);
-
-  // Helper to initialize the render converter.
-  void InitializeRenderConverterIfNeeded(int sample_rate,
-                                         int number_of_channels,
-                                         int frames_per_buffer);
-
-  // Called by ProcessAndConsumeData().
-  void ProcessData(webrtc::AudioFrame* audio_frame,
-                   base::TimeDelta capture_delay,
-                   int volume,
-                   bool key_pressed);
-
-  // Called when the processor is going away.
-  void StopAudioProcessing();
-
-  // Cached value for the render delay latency. This member is accessed by
-  // both the capture audio thread and the render audio thread.
-  base::subtle::Atomic32 render_delay_ms_;
-
-  // webrtc::AudioProcessing module which does AEC, AGC, NS, HighPass filter,
-  // ..etc.
-  scoped_ptr<webrtc::AudioProcessing> audio_processing_;
-
-  // Converter used for the down-mixing and resampling of the capture data.
-  scoped_ptr<MediaStreamAudioConverter> capture_converter_;
-
-  // AudioFrame used to hold the output of |capture_converter_|.
-  webrtc::AudioFrame capture_frame_;
-
-  // Converter used for the down-mixing and resampling of the render data when
-  // the AEC is enabled.
-  scoped_ptr<MediaStreamAudioConverter> render_converter_;
-
-  // AudioFrame used to hold the output of |render_converter_|.
-  webrtc::AudioFrame render_frame_;
-
-  // Data bus to help converting interleaved data to an AudioBus.
-  scoped_ptr<media::AudioBus> render_data_bus_;
-
-  // Used to DCHECK that some methods are called on the main render thread.
-  base::ThreadChecker main_thread_checker_;
-
-  // Used to DCHECK that some methods are called on the capture audio thread.
-  base::ThreadChecker capture_thread_checker_;
-
-  // Used to DCHECK that PushRenderData() is called on the render audio thread.
-  base::ThreadChecker render_thread_checker_;
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_