commit bf47495979b8e938fd4f05ca13e6a3a6ab639387
author Niels Möller <nisse@webrtc.org> Mon Feb 18 12:00:06 2019 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Feb 18 13:29:35 2019 +0000
tree 9dc8ad343e4253a688988f94ff4bb50301c62b8c
parent a0b1fb9ac7698948736aa53c6d48972cb40779b5

Update remaining audio test code to not use WebRtcRTPHeader.

Bug: webrtc:5876
Change-Id: I5b1abcec4a0ef52b6dd36d1fe94dbfd3f88f28a7
Reviewed-on: https://webrtc-review.googlesource.com/c/123235
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26736}

modules/audio_coding/test/EncodeDecodeTest.cc

diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index cd57ecd..28ee8aa 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -155,7 +155,7 @@
   if (!_rtpStream->EndOfFile()) {
     if (_firstTime) {
       _firstTime = false;
-      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                                _payloadSizeBytes, &_nextTime);
       if (_realPayloadSizeBytes == 0) {
         if (_rtpStream->EndOfFile()) {
@@ -168,8 +168,8 @@
     }
 
     EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
-                                      _rtpInfo.header));
-    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                      _rtpHeader));
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
                                              _payloadSizeBytes, &_nextTime);
     if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
       _firstTime = true;