Index: content/renderer/media/webrtc_local_audio_renderer.h |
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h |
index f58aee3c29390461e634cef8553e24d78f9a10d0..f77d523f4a6dc8413b51624f546e73537fb1ed14 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.h |
+++ b/content/renderer/media/webrtc_local_audio_renderer.h |
@@ -48,7 +48,10 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
// The |source| is owned by the WebRtcAudioDeviceImpl. |
// Called on the main thread. |
WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track, |
- int source_render_view_id); |
+ int source_render_view_id, |
+ int session_id, |
+ int sample_rate, |
+ int frames_per_buffer); |
// MediaStreamAudioRenderer implementation. |
// Called on the main thread. |
@@ -101,6 +104,7 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
// The render view in which the audio is rendered into |sink_|. |
const int source_render_view_id_; |
+ const int session_id_; |
// The sink (destination) for rendered audio. |
scoped_refptr<media::AudioOutputDevice> sink_; |
@@ -128,6 +132,14 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
// Protects |loopback_fifo_|, |playing_| and |sink_|. |
mutable base::Lock thread_lock_; |
+ // The preferred sample rate and buffer sizes provided via the ctor. |
+ const int sample_rate_; |
+ const int frames_per_buffer_; |
+ |
+ // The preferred device id of the output device or empty for the default |
+ // output device. |
+ const std::string output_device_id_; |
+ |
DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); |
}; |