MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
deleted file mode 100644
index df992e1b333ed69fa3ba8aa4854193027b617c69..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ /dev/null
@@ -1,207 +0,0 @@
-// Copyright (c) 2012 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_CAPTURER_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
-
-#include <list>
-#include <memory>
-#include <string>
-
-#include "base/callback.h"
-#include "base/files/file.h"
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "base/time/time.h"
-#include "content/common/media/media_stream_options.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/tagged_list.h"
-#include "media/audio/audio_input_device.h"
-#include "media/base/audio_capturer_source.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-
-namespace media {
-class AudioBus;
-}
-
-namespace content {
-
-class MediaStreamAudioProcessor;
-class MediaStreamAudioSource;
-class WebRtcAudioDeviceImpl;
-class WebRtcLocalAudioRenderer;
-class WebRtcLocalAudioTrack;
-
-// This class manages the capture data flow by getting data from its
-// |source_|, and passing it to its |tracks_|.
-// The threading model for this class is rather complex since it will be
-// created on the main render thread, captured data is provided on a dedicated
-// AudioInputDevice thread, and methods can be called either on the Libjingle
-// thread or on the main render thread but also other client threads
-// if an alternative AudioCapturerSource has been set.
-class CONTENT_EXPORT WebRtcAudioCapturer
-    : NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
- public:
-  // Used to construct the audio capturer. |render_frame_id| specifies the
-  // RenderFrame consuming audio for capture; -1 is used for tests.
-  // |device_info| contains all the device information that the capturer is
-  // created for. |constraints| contains the settings for audio processing.
-  // TODO(xians): Implement the interface for the audio source and move the
-  // |constraints| to ApplyConstraints(). Called on the main render thread.
-  static std::unique_ptr<WebRtcAudioCapturer> CreateCapturer(
-      int render_frame_id,
-      const StreamDeviceInfo& device_info,
-      const blink::WebMediaConstraints& constraints,
-      WebRtcAudioDeviceImpl* audio_device,
-      MediaStreamAudioSource* audio_source);
-
-  ~WebRtcAudioCapturer() override;
-
-  // Add a audio track to the sinks of the capturer.
-  // WebRtcAudioDeviceImpl calls this method on the main render thread but
-  // other clients may call it from other threads. The current implementation
-  // does not support multi-thread calling.
-  // The first AddTrack will implicitly trigger the Start() of this object.
-  void AddTrack(WebRtcLocalAudioTrack* track);
-
-  // Remove a audio track from the sinks of the capturer.
-  // If the track has been added to the capturer, it  must call RemoveTrack()
-  // before it goes away.
-  // Called on the main render thread or libjingle working thread.
-  void RemoveTrack(WebRtcLocalAudioTrack* track);
-
-  // Called when a stream is connecting to a peer connection. This will set
-  // up the native buffer size for the stream in order to optimize the
-  // performance for peer connection.
-  void EnablePeerConnectionMode();
-
-  // Volume APIs used by WebRtcAudioDeviceImpl.
-  // Called on the AudioInputDevice audio thread.
-  void SetVolume(int volume);
-  int Volume() const;
-  int MaxVolume() const;
-
-  // Audio parameters utilized by the source of the audio capturer.
-  // TODO(phoglund): Think over the implications of this accessor and if we can
-  // remove it.
-  media::AudioParameters GetInputFormat() const;
-
-  const StreamDeviceInfo& device_info() const { return device_info_; }
-
-  // Stops recording audio. This method will empty its track lists since
-  // stopping the capturer will implicitly invalidate all its tracks.
-  // This method is exposed to the public because the MediaStreamAudioSource can
-  // call Stop()
-  void Stop();
-
-  // Returns the output format.
-  // Called on the main render thread.
-  media::AudioParameters GetOutputFormat() const;
-
-  // Used by clients to inject their own source to the capturer.
-  void SetCapturerSource(
-      const scoped_refptr<media::AudioCapturerSource>& source,
-      media::AudioParameters params);
-
- private:
-  class TrackOwner;
-  typedef TaggedList<TrackOwner> TrackList;
-
-  WebRtcAudioCapturer(int render_frame_id,
-                      const StreamDeviceInfo& device_info,
-                      const blink::WebMediaConstraints& constraints,
-                      WebRtcAudioDeviceImpl* audio_device,
-                      MediaStreamAudioSource* audio_source);
-
-  // AudioCapturerSource::CaptureCallback implementation.
-  // Called on the AudioInputDevice audio thread.
-  void Capture(const media::AudioBus* audio_source,
-               int audio_delay_milliseconds,
-               double volume,
-               bool key_pressed) override;
-  void OnCaptureError(const std::string& message) override;
-
-  // Initializes the default audio capturing source using the provided render
-  // frame id and device information. Return true if success, otherwise false.
-  bool Initialize();
-
-  // SetCapturerSourceInternal() is called if the client on the source side
-  // desires to provide their own captured audio data. Client is responsible
-  // for calling Start() on its own source to get the ball rolling.
-  // Called on the main render thread.
-  // buffer_size is optional. Set to 0 to let it be chosen automatically.
-  void SetCapturerSourceInternal(
-      const scoped_refptr<media::AudioCapturerSource>& source,
-      media::ChannelLayout channel_layout,
-      int sample_rate);
-
-  // Starts recording audio.
-  // Triggered by AddSink() on the main render thread or a Libjingle working
-  // thread. It should NOT be called under |lock_|.
-  void Start();
-
-  // Helper function to get the buffer size based on |peer_connection_mode_|
-  // and sample rate;
-  int GetBufferSize(int sample_rate) const;
-
-  // Used to DCHECK that we are called on the correct thread.
-  base::ThreadChecker thread_checker_;
-
-  // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|,
-  // |params_| and |buffering_|.
-  mutable base::Lock lock_;
-
-  // A tagged list of audio tracks that the audio data is fed
-  // to. Tagged items need to be notified that the audio format has
-  // changed.
-  TrackList tracks_;
-
-  // The audio data source from the browser process.
-  scoped_refptr<media::AudioCapturerSource> source_;
-
-  // Cached audio constraints for the capturer.
-  blink::WebMediaConstraints constraints_;
-
-  // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
-  // data is in a unit of 10 ms data chunk.
-  const scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
-
-  bool running_;
-
-  int render_frame_id_;
-
-  // Cached information of the device used by the capturer.
-  const StreamDeviceInfo device_info_;
-
-  // Stores latest microphone volume received in a CaptureData() callback.
-  // Range is [0, 255].
-  int volume_;
-
-  // Flag which affects the buffer size used by the capturer.
-  bool peer_connection_mode_;
-
-  // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
-  // of RenderThread.
-  WebRtcAudioDeviceImpl* audio_device_;
-
-  // Raw pointer to the MediaStreamAudioSource object that holds a reference
-  // to this WebRtcAudioCapturer.
-  // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and
-  // blink guarantees that the blink::WebMediaStreamSource outlives any
-  // blink::WebMediaStreamTrack connected to the source, |audio_source_| is
-  // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
-  // WebRtcAudioCapturer.
-  MediaStreamAudioSource* const audio_source_;
-
-  // Used to calculate the signal level that shows in the UI.
-  MediaStreamAudioLevelCalculator level_calculator_;
-
-  DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_