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}diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index fd76224..cbe066f 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -44,7 +44,7 @@
bool ReceiverWithPacketLoss::IncomingPacket() {
if (!_rtpStream->EndOfFile()) {
if (packet_counter_ == 0) {
- _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0) {
if (_rtpStream->EndOfFile()) {
@@ -57,11 +57,10 @@
}
if (!PacketLost()) {
- _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
- _rtpInfo.header);
+ _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
}
packet_counter_++;
- _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+ _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
packet_counter_ = 0;