Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1505253004 .
Cr-Commit-Position: refs/heads/master@{#10995}
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index fe223bb..44b9d4f 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -31,6 +31,7 @@
#include <string>
#include <vector>
+#include "talk/media/base/audiorenderer.h"
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
@@ -51,9 +52,12 @@
class Timing;
}
+namespace webrtc {
+class AudioSinkInterface;
+}
+
namespace cricket {
-class AudioRenderer;
struct RtpHeader;
class ScreencastId;
struct VideoFormat;
@@ -1028,6 +1032,10 @@
virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
+
+ virtual void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
};
struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {