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Update remaining audio test code to not use WebRtcRTPHeader.

Browse Source 
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}
This commit is contained in:
Niels Möller
2019-02-18 12:00:06 +01:00
committed by Commit Bot
parent a0b1fb9ac7
commit bf47495979
7 changed files with 56 additions and 58 deletions
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7
modules/audio_coding/test/PacketLossTest.cc
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@ -44,7 +44,7 @@ void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
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 @@ bool ReceiverWithPacketLoss::IncomingPacket() {
}
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;