Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
| diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
| index 56f03d54093e14cd1160d65c8f17ee93ddd53f15..7262ba62fa8aeee8709e393754006898e1b57bd4 100644 |
| --- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
| @@ -451,6 +451,12 @@ class RtpPacketSender { |
| int64_t capture_time_ms, |
| size_t bytes, |
| bool retransmission) = 0; |
| + |
| + // Currently audio traffic is not accounted by pacer and passed through. |
| + // With the introduction of audio BWE audio traffic will be accounted for |
| + // the pacer budget calculation. The audio traffic still will be injected |
| + // at high priority. |
| + virtual void AccountForAudioPackets(bool account_for_audio) = 0; |
|
stefan-webrtc
2017/08/14 07:46:03
SetAccountForAudioPackets
alexnarest
2017/08/15 06:02:36
Done.
|
| }; |
| class TransportSequenceNumberAllocator { |