Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
| index 90e19c06459ad7c24a3230c53803b835e1755b02..3d51fe010ee28b5352a9cb93728700d8728082c9 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
| @@ -44,9 +44,8 @@ bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const { |
| } |
| int32_t RTPReceiverVideo::OnNewPayloadTypeCreated( |
| - const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| - int8_t payload_type, |
| - uint32_t frequency) { |
| + const CodecInst& audio_codec) { |
| + // Ignore, this function will never be called. |
|
danilchap
2016/11/23 17:12:50
use RTC_NOTREACHABLE() instead of the comment to b
magjed_webrtc
2016/11/24 09:39:40
Done.
|
| return 0; |
| } |