| Index: content/renderer/media/webrtc_audio_renderer.h
|
| diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
|
| index 577c993cc7b887c1a5bb61c277cd1f07c57313f3..1112c61bafa140337db6c33bf652dabe7cc0a207 100644
|
| --- a/content/renderer/media/webrtc_audio_renderer.h
|
| +++ b/content/renderer/media/webrtc_audio_renderer.h
|
| @@ -13,6 +13,7 @@
|
| #include "media/base/audio_decoder.h"
|
| #include "media/base/audio_pull_fifo.h"
|
| #include "media/base/audio_renderer_sink.h"
|
| +#include "media/base/channel_layout.h"
|
|
|
| namespace media {
|
| class AudioOutputDevice;
|
| @@ -28,7 +29,10 @@ class CONTENT_EXPORT WebRtcAudioRenderer
|
| : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
|
| NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
|
| public:
|
| - explicit WebRtcAudioRenderer(int source_render_view_id);
|
| + WebRtcAudioRenderer(int source_render_view_id,
|
| + int session_id,
|
| + int sample_rate,
|
| + int frames_per_buffer);
|
|
|
| // Initialize function called by clients like WebRtcAudioDeviceImpl.
|
| // Stop() has to be called before |source| is deleted.
|
| @@ -72,6 +76,7 @@ class CONTENT_EXPORT WebRtcAudioRenderer
|
|
|
| // 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_;
|
| @@ -100,6 +105,10 @@ class CONTENT_EXPORT WebRtcAudioRenderer
|
| // Delay due to the FIFO in milliseconds.
|
| int fifo_delay_milliseconds_;
|
|
|
| + // The preferred sample rate and buffer sizes provided via the ctor.
|
| + const int sample_rate_;
|
| + const int frames_per_buffer_;
|
| +
|
| DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
|
| };
|
|
|
|
|