| commit | afb5dbbf4ef5a8462380f3ba4a61937f6b802e35 | [log] [tgz] |
|---|---|---|
| author | Niels Möller <nisse@webrtc.org> | Fri Feb 15 15:21:47 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 18 08:01:31 2019 +0000 |
| tree | bde8e7a328495ee52832e6a286108d98b48f9098 | |
| parent | 389b1672a32f2dd49af6c6ed40e8ddf394b986de [diff] [blame] |
Update ACM to use RTPHeader instead of WebRtcRTPHeader Bug: webrtc:5876 Change-Id: Id3311dcf508cca34495349197eeac2edf8783772 Reviewed-on: https://webrtc-review.googlesource.com/c/123188 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26729}
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc index 000041b..fd76224 100644 --- a/modules/audio_coding/test/PacketLossTest.cc +++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -57,7 +57,8 @@ } if (!PacketLost()) { - _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo); + _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, + _rtpInfo.header); } packet_counter_++; _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,