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

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
new file mode 100644
index 0000000000000000000000000000000000000000..72b80194b08ed09a01673c98c4ed9816aa4e6d74
--- /dev/null
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
@@ -0,0 +1,107 @@
+// Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
+
+#include <vector>
+
+#include "base/memory/ref_counted.h"
+#include "base/memory/scoped_vector.h"
+#include "base/single_thread_task_runner.h"
+#include "base/synchronization/lock.h"
+#include "content/common/content_export.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
+#include "content/renderer/media/media_stream_audio_processor.h"
+#include "third_party/webrtc/api/mediastreamtrack.h"
+#include "third_party/webrtc/media/base/audiorenderer.h"
+
+namespace cricket {
+class AudioRenderer;
+}
+
+namespace webrtc {
+class AudioSourceInterface;
+class AudioProcessorInterface;
+}
+
+namespace content {
+
+class MediaStreamAudioProcessor;
+class WebRtcAudioSinkAdapter;
+class WebRtcLocalAudioTrack;
+
+// Provides an implementation of the webrtc::AudioTrackInterface that can be
+// bound/unbound to/from a MediaStreamAudioTrack.  In other words, this is an
+// adapter that sits between the media stream object graph and WebRtc's object
+// graph and proxies between the two.
+class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
+    : NON_EXPORTED_BASE(
+          public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
+ public:
+  static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
+      const std::string& label,
+      webrtc::AudioSourceInterface* track_source);
+
+  WebRtcLocalAudioTrackAdapter(
+      const std::string& label,
+      webrtc::AudioSourceInterface* track_source,
+      scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
+
+  ~WebRtcLocalAudioTrackAdapter() override;
+
+  void Initialize(WebRtcLocalAudioTrack* owner);
+
+  // Set the object that provides shared access to the current audio signal
+  // level.  This method may only be called once, before the audio data flow
+  // starts, and before any calls to GetSignalLevel() might be made.
+  void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
+
+  // Method called by the WebRtcLocalAudioTrack to set the processor that
+  // applies signal processing on the data of the track.
+  // This class will keep a reference of the |processor|.
+  // Called on the main render thread.
+  // This method may only be called once, before the audio data flow starts, and
+  // before any calls to GetAudioProcessor() might be made.
+  void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
+
+  // webrtc::MediaStreamTrack implementation.
+  std::string kind() const override;
+  bool set_enabled(bool enable) override;
+
+ private:
+  // webrtc::AudioTrackInterface implementation.
+  void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
+  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
+  bool GetSignalLevel(int* level) override;
+  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
+      override;
+  webrtc::AudioSourceInterface* GetSource() const override;
+
+  // Weak reference.
+  WebRtcLocalAudioTrack* owner_;
+
+  // The source of the audio track which handles the audio constraints.
+  // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
+  rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
+
+  // Libjingle's signaling thread.
+  const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
+
+  // The audio processsor that applies audio processing on the data of audio
+  // track.  This must be set before calls to GetAudioProcessor() are made.
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
+
+  // A vector of the peer connection sink adapters which receive the audio data
+  // from the audio track.
+  ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
+
+  // Thread-safe accessor to current audio signal level.  This must be set
+  // before calls to GetSignalLevel() are made.
+  scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_