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); |
}; |