| 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],
|
|
|