Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
By removing it we can in turn (next CL) get rid of RTPTypeHeader, which is a
union that cause some problems.
Bug: none
Change-Id: I9246ecbfe2c8b7eda27497cccbc5f438958b64bf
Reviewed-on: https://webrtc-review.googlesource.com/83985
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23666}diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index df9c731..74de1d9 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -69,18 +69,13 @@
rtp_info.header.sequenceNumber = sequence_number_++;
rtp_info.header.payloadType = payload_type;
rtp_info.header.timestamp = timestamp;
- if (frame_type == kAudioFrameCN) {
- rtp_info.type.Audio.isCNG = true;
- } else {
- rtp_info.type.Audio.isCNG = false;
- }
+
if (frame_type == kEmptyFrame) {
// Skip this frame.
return 0;
}
// Only run mono for all test cases.
- rtp_info.type.Audio.channel = 1;
memcpy(payload_data_, payload_data, payload_size);
status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);