Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 464d6133d0d60c558af76305019a3322f4d08962..d873d8b767ff1e7324f271877df4899265a42d0e 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1262,12 +1262,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { |
bool use_transport_cc, |
const std::string& sync_group, |
const std::vector<webrtc::RtpExtension>& extensions, |
- webrtc::Call* call) |
+ webrtc::Call* call, |
+ webrtc::Transport* rtcp_send_transport) |
: call_(call), config_() { |
RTC_DCHECK_GE(ch, 0); |
RTC_DCHECK(call); |
config_.rtp.remote_ssrc = remote_ssrc; |
config_.rtp.local_ssrc = local_ssrc; |
+ config_.rtcp_send_transport = rtcp_send_transport; |
config_.voe_channel_id = ch; |
config_.sync_group = sync_group; |
RecreateAudioReceiveStream(use_transport_cc, extensions); |
@@ -2099,7 +2101,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { |
ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_, |
recv_transport_cc_enabled_, |
sp.sync_label, recv_rtp_extensions_, |
- call_))); |
+ call_, this))); |
SetNack(channel, send_codec_spec_.nack_enabled); |
SetPlayout(channel, playout_); |