Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
index 1e0bf732ad7d0ca37ede10d196b54c99b79ad997..386336544b9ea49719681a21f54c40fbd1da4635 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc |
@@ -100,6 +100,11 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(const CodecInst& audio_codec) { |
return result; |
} |
+int32_t RtpReceiverImpl::RegisterReceivePayload(const VideoCodec& video_codec) { |
+ rtc::CritScope lock(&critical_section_rtp_receiver_); |
+ return rtp_payload_registry_->RegisterReceivePayload(video_codec); |
+} |
+ |
// TODO(magjed): Remove once external code is updated. |
int32_t RtpReceiverImpl::RegisterReceivePayload( |
const char payload_name[RTP_PAYLOAD_NAME_SIZE], |