Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| index 7c72e5917c8ab2feb8acdac1c119cd3e25b8b5c6..94ab763a26b1f8e00c999b61987f5e5bc35c0e70 100644 |
| --- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| @@ -286,7 +286,7 @@ class RtpRtcp : public Module { |
| virtual void BitrateSent(uint32_t* totalRate, |
| uint32_t* videoRate, |
| uint32_t* fecRate, |
| - uint32_t* nackRate) const = 0; |
|
danilchap
2016/06/20 16:53:46
this would break downstream code.
sprang_webrtc
2016/06/23 07:54:29
Acknowledged.
|
| + uint32_t* nackRate) = 0; |
| /* |
| * Used by the codec module to deliver a video or audio frame for |