Index: talk/media/webrtc/webrtcvoiceengine.h |
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h |
index 0f2f59e4928f0ea619adb5e7088fc3b524388bf5..96d8946c1b5d87fd915bb6a46da80022b535db81 100644 |
--- a/talk/media/webrtc/webrtcvoiceengine.h |
+++ b/talk/media/webrtc/webrtcvoiceengine.h |
@@ -198,7 +198,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
void SetRawAudioSink( |
uint32_t ssrc, |
- rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; |
+ rtc::scoped_refptr<webrtc::AudioSinkInterface> sink) override; |
// implements Transport interface |
bool SendRtp(const uint8_t* data, |
@@ -269,6 +269,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
int64_t default_recv_ssrc_ = -1; |
// Volume for unsignalled stream, which may be set before the stream exists. |
double default_recv_volume_ = 1.0; |
+ // Sink for default stream, which may be set before the stream exists. |
the sun
2016/01/08 13:14:26
nit: default->unsignalled since that is the termin
Taylor Brandstetter
2016/01/08 19:46:58
Done.
|
+ rtc::scoped_refptr<webrtc::AudioSinkInterface> default_sink_; |
// Default SSRC to use for RTCP receiver reports in case of no signaled |
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
// and https://code.google.com/p/chromium/issues/detail?id=547661 |