start/stop the source of the capturer when 1st audiotrack/last audiotrack is added/removed

content/renderer/media/webrtc_audio_capturer.h

 // 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 <string>
 
@@ skipping 34 matching lines @@
 
   // Creates and configures the default audio capturing source using the
   // provided audio parameters.  |render_view_id| specifies the render view
   // consuming audio for capture.  |session_id| is passed to the browser to
   // decide which device to use.  Called on the main render thread.
   bool Initialize(int render_view_id,
                   media::ChannelLayout channel_layout,
                   int sample_rate,
                   int session_id);
 
+  // Called by the WebRtcAudioDeviceImpl to add the ADM as the default sink to
+  // the capturer. This function is needed since WebRTC supports only one ADM
+  // but multiple audio tracks, so the ADM can't be the sink of certain audio
+  // track now. And we also need to distinguish the WebRtcAudioDeviceImpl from
+  // the audio track in order to start/stop the source of the capturer
+  // correctly.
+  // TODO(xians): Remove this function after WebRtc supports multiple ADMs.
+  void SetDefaultSink(WebRtcAudioCapturerSink* sink);
+
   // 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.
-  // Called on the main render thread.
+  // Called on the main render thread or libjingle working thread.
   void AddSink(WebRtcAudioCapturerSink* track);
 
   // Remove a audio track from the sinks of the capturer.
-  // Called on the main render thread.
+  // Called on the main render thread or libjingle working thread.
   void RemoveSink(WebRtcAudioCapturerSink* track);
 
   // SetCapturerSource() 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 have the ball rolling.
   // Called on the main render thread.
   void SetCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
       float sample_rate);
 
-  // Starts recording audio.
-  // Called on the main render thread or a Libjingle working thread.
-  void Start();
-
-  // Stops recording audio.
-  // Called on the main render thread or a Libjingle working thread.
-  void Stop();
-
   // Sets the microphone volume.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(double volume);
 
   // Enables or disables the WebRtc AGC control.
   // Called from a Libjingle working thread.
   void SetAutomaticGainControl(bool enable);
 
   bool is_recording() const { return running_; }
 
@@ skipping 17 matching lines @@
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
                        double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // Reconfigures the capturer with a new buffer size and capture parameters.
   // Must be called without holding the lock. Returns true on success.
   bool Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
 
+  // 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();
+
+  // Stops recording audio.
+  // Triggered by RemoveSink() on the main render thread or a Libjingle working
+  // thread. It should NOT be called under |lock_|.
+  void Stop();
+
+
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   // Protects |source_|, |audio_tracks_|, |running_|, |loopback_fifo_|,
   // |params_|, |buffering_| and |agc_is_enabled_|.
   mutable base::Lock lock_;
 
   // A list of audio tracks that the audio data is fed to.
   TrackList tracks_;
 
+  // A pointer to WebRtcAudioDeviceImpl which is the default destination of
+  // all the audio tracks data flow.
+  WebRtcAudioCapturerSink* default_sink_;
+
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Buffers used for temporary storage during capture callbacks.
   // Allocated during initialization.
   class ConfiguredBuffer;
   scoped_refptr<ConfiguredBuffer> buffer_;
   bool running_;
 
   // True when automatic gain control is enabled, false otherwise.
   bool agc_is_enabled_;
 
   // The media session ID used to identify which input device to be started.
   int session_id_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_