Cast: Basic cast_receiver API for chrome.

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 91 matching lines @@
   // 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 the unittests to inject their own source to the capturer.
-  void SetCapturerSourceForTesting(
+  // Used by clients to inject their own source to the capturer.
+  void SetCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   ~WebRtcAudioCapturer() override;
 
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
 
   WebRtcAudioCapturer(int render_view_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() override;
 
   // Initializes the default audio capturing source using the provided render
   // view id and device information. Return true if success, otherwise false.
   bool Initialize();
 
-  // 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.
+  // 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.
-  void SetCapturerSource(
+  void SetCapturerSourceInternal(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
       float 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_|
@@ skipping 48 matching lines @@
   // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_