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/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index d5392dc..e5a7684 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -50,13 +50,12 @@
acm_->InitializeReceiver();
acm_->RegisterTransportCallback(this);
- rtp_header_.header.sequenceNumber = 0;
- rtp_header_.header.timestamp = 0;
- rtp_header_.header.markerBit = false;
- rtp_header_.header.ssrc = 0x12345678; // Arbitrary.
- rtp_header_.header.numCSRCs = 0;
- rtp_header_.header.payloadType = 0;
- rtp_header_.frameType = kAudioFrameSpeech;
+ rtp_header_.sequenceNumber = 0;
+ rtp_header_.timestamp = 0;
+ rtp_header_.markerBit = false;
+ rtp_header_.ssrc = 0x12345678; // Arbitrary.
+ rtp_header_.numCSRCs = 0;
+ rtp_header_.payloadType = 0;
}
void TearDown() override {}
@@ -113,9 +112,8 @@
if (frame_type == kEmptyFrame)
return 0;
- rtp_header_.header.payloadType = payload_type;
- rtp_header_.frameType = frame_type;
- rtp_header_.header.timestamp = timestamp;
+ rtp_header_.payloadType = payload_type;
+ rtp_header_.timestamp = timestamp;
int ret_val = receiver_->InsertPacket(
rtp_header_,
@@ -124,7 +122,7 @@
assert(false);
return -1;
}
- rtp_header_.header.sequenceNumber++;
+ rtp_header_.sequenceNumber++;
packet_sent_ = true;
last_frame_type_ = frame_type;
return 0;
@@ -137,7 +135,7 @@
AudioCodingModule::Config config_;
std::unique_ptr<AcmReceiver> receiver_;
std::unique_ptr<AudioCodingModule> acm_;
- WebRtcRTPHeader rtp_header_;
+ RTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
uint32_t last_packet_send_timestamp_;