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/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 75ba60a..81b83c0 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -66,14 +66,14 @@
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
- WebRtcRTPHeader rtp_info;
+ RTPHeader rtp_header;
int32_t status;
- rtp_info.header.markerBit = false;
- rtp_info.header.ssrc = 0;
- rtp_info.header.sequenceNumber = sequence_number_++;
- rtp_info.header.payloadType = payload_type;
- rtp_info.header.timestamp = timestamp;
+ rtp_header.markerBit = false;
+ rtp_header.ssrc = 0;
+ rtp_header.sequenceNumber = sequence_number_++;
+ rtp_header.payloadType = payload_type;
+ rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
// Skip this frame.
@@ -83,7 +83,8 @@
// Only run mono for all test cases.
memcpy(payload_data_, payload_data, payload_size);
- status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
+ status =
+ receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header);
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;