Reland 196894 - First step to add audio track support to local stream.

First step to add audio track support to local stream.
This patch builds up a correct structure for the local audio track, and added a correct chromium audio track implementation to local media stream.

This patch does not change the current functionalities and behaviour of local media stream, it is more like a refactoring on the structure.

TBR=tommi@chromium.org

BUG=164813
TEST=content_unittests --gtest_filter="*WebRtcLocalAudio*"
apprtc.appspot.com and webrtc.googlecode.com/svn/trunk/samples/js/demos/html/local-audio-rendering.html

Review URL: https://codereview.chromium.org/14640006

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@197349 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 98 insertions, 0 deletions

diff --git a/content/renderer/media/webrtc_local_audio_track.h b/content/renderer/media/webrtc_local_audio_track.h
new file mode 100644
index 0000000..660d696
--- /dev/null
+++ b/content/renderer/media/webrtc_local_audio_track.h
@@ -0,0 +1,98 @@
+// 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/synchronization/lock.h"
+#include "base/threading/thread_checker.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
+
+
+namespace content {
+
+class WebRtcAudioCapturer;
+class WebRtcAudioCapturerSinkOwner;
+
+// A WebRtcLocalAudioTrack instance contains the implementations of
+// MediaStreamTrack and WebRtcAudioCapturerSink.
+// 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.
+class CONTENT_EXPORT WebRtcLocalAudioTrack
+    : NON_EXPORTED_BASE(public WebRtcAudioCapturerSink),
+      NON_EXPORTED_BASE(
+          public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
+ public:
+  static scoped_refptr<WebRtcLocalAudioTrack> Create(
+      const std::string& id,
+      const scoped_refptr<WebRtcAudioCapturer>& capturer,
+      webrtc::AudioSourceInterface* stream_source);
+
+  // Add a sink to the track. This function will trigger a SetCaptureFormat()
+  // call on the |sink|.
+  // Called on the main render thread.
+  void AddSink(WebRtcAudioCapturerSink* sink);
+
+  // Remove a sink from the track.
+  // Called on the main render thread.
+  void RemoveSink(WebRtcAudioCapturerSink* sink);
+
+ protected:
+  WebRtcLocalAudioTrack(const std::string& label,
+                        const scoped_refptr<WebRtcAudioCapturer>& capturer,
+                        webrtc::AudioSourceInterface* stream_source);
+  virtual ~WebRtcLocalAudioTrack();
+
+ private:
+  typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList;
+
+  // content::WebRtcAudioCapturerSink implementation.
+  // Called on the AudioInputDevice worker thread.
+  virtual void CaptureData(const int16* audio_data,
+                           int number_of_channels,
+                           int number_of_frames,
+                           int audio_delay_milliseconds,
+                           double volume) OVERRIDE;
+
+  // Can be called on different user threads.
+  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
+
+  // webrtc::AudioTrackInterface implementation.
+  virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
+
+  // webrtc::MediaStreamTrack implementation.
+  virtual std::string kind() const OVERRIDE;
+
+  // The provider of captured data to render.
+  // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
+  scoped_refptr<WebRtcAudioCapturer> capturer_;
+
+  // The source of the audio track which handles the audio constraints.
+  // TODO(xians): merge |track_source_| to |capturer_|.
+  talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
+
+  // A list of sinks that the audio data is fed to.
+  SinkList sinks_;
+
+  // Used to DCHECK that we are called on the correct thread.
+  base::ThreadChecker thread_checker_;
+
+  // Cached values of the audio parameters used by the |source_| and |sinks_|.
+  media::AudioParameters params_;
+
+  // Protects |params_| and |sinks_|.
+  mutable base::Lock lock_;
+
+  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_