MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_local_audio_track.h

Index: content/renderer/media/webrtc_local_audio_track.h
diff --git a/content/renderer/media/webrtc_local_audio_track.h b/content/renderer/media/webrtc_local_audio_track.h
deleted file mode 100644
index 2eafbd160ee49b95cd42709946e504e082d094da..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track.h
+++ /dev/null
@@ -1,136 +0,0 @@
-// 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_LOCAL_AUDIO_TRACK_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
-
-#include <list>
-#include <string>
-
-#include "base/macros.h"
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_ptr.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
-#include "content/renderer/media/media_stream_audio_track.h"
-#include "content/renderer/media/tagged_list.h"
-#include "media/audio/audio_parameters.h"
-
-namespace media {
-class AudioBus;
-}
-
-namespace content {
-
-class MediaStreamAudioLevelCalculator;
-class MediaStreamAudioProcessor;
-class MediaStreamAudioSink;
-class MediaStreamAudioSinkOwner;
-class MediaStreamAudioTrackSink;
-class WebAudioCapturerSource;
-class WebRtcAudioCapturer;
-class WebRtcLocalAudioTrackAdapter;
-
-// A WebRtcLocalAudioTrack instance contains the implementations of
-// MediaStreamTrackExtraData.
-// When an instance is created, it will register itself as a track to the
-// WebRtcAudioCapturer to get the captured data, and forward the data to
-// its |sinks_|. The data flow can be stopped by disabling the audio track.
-// TODO(tommi): Rename to MediaStreamLocalAudioTrack.
-class CONTENT_EXPORT WebRtcLocalAudioTrack
-    : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
- public:
-  WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
-                        const scoped_refptr<WebRtcAudioCapturer>& capturer,
-                        WebAudioCapturerSource* webaudio_source);
-
-  ~WebRtcLocalAudioTrack() override;
-
-  // Add a sink to the track. This function will trigger a OnSetFormat()
-  // call on the |sink|.
-  // Called on the main render thread.
-  void AddSink(MediaStreamAudioSink* sink) override;
-
-  // Remove a sink from the track.
-  // Called on the main render thread.
-  void RemoveSink(MediaStreamAudioSink* sink) override;
-
-  // Starts the local audio track. Called on the main render thread and
-  // should be called only once when audio track is created.
-  void Start();
-
-  // Overrides for MediaStreamTrack.
-
-  void SetEnabled(bool enabled) override;
-
-  // Stops the local audio track. Called on the main render thread and
-  // should be called only once when audio track going away.
-  void Stop() override;
-
-  webrtc::AudioTrackInterface* GetAudioAdapter() override;
-
-  // Returns the output format of the capture source. May return an invalid
-  // AudioParameters if the format is not yet available.
-  // Called on the main render thread.
-  media::AudioParameters GetOutputFormat() const override;
-
-  // Method called by the capturer to deliver the capture data.
-  // Called on the capture audio thread.
-  void Capture(const media::AudioBus& audio_bus,
-               base::TimeTicks estimated_capture_time,
-               bool force_report_nonzero_energy);
-
-  // Method called by the capturer to set the audio parameters used by source
-  // of the capture data..
-  // Called on the capture audio thread.
-  void OnSetFormat(const media::AudioParameters& params);
-
-  // Method called by the capturer to set the processor that applies signal
-  // processing on the data of the track.
-  // Called on the capture audio thread.
-  void SetAudioProcessor(
-      const scoped_refptr<MediaStreamAudioProcessor>& processor);
-
- private:
-  typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
-
-  // All usage of libjingle is through this adapter. The adapter holds
-  // a pointer to this object, but no reference.
-  const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
-
-  // The provider of captured data to render.
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
-
-  // The source of the audio track which is used by WebAudio, which provides
-  // data to the audio track when hooking up with WebAudio.
-  scoped_refptr<WebAudioCapturerSource> webaudio_source_;
-
-  // A tagged list of sinks that the audio data is fed to. Tags
-  // indicate tracks that need to be notified that the audio format
-  // has changed.
-  SinkList sinks_;
-
-  // Tests that methods are called on libjingle's signaling thread.
-  base::ThreadChecker signal_thread_checker_;
-
-  // Used to DCHECK that some methods are called on the capture audio thread.
-  base::ThreadChecker capture_thread_checker_;
-
-  // Protects |params_| and |sinks_|.
-  mutable base::Lock lock_;
-
-  // Audio parameters of the audio capture stream.
-  // Accessed on only the audio capture thread.
-  media::AudioParameters audio_parameters_;
-
-  // Used to calculate the signal level that shows in the UI.
-  // Accessed on only the audio thread.
-  scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
-
-  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
-};
-
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_