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/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index c972e62..6f7c6cf 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -37,12 +37,11 @@
{{pltype, {"L16", kSampleRateHz, 1}}};
acm_->SetReceiveCodecs(receive_codecs);
- rtp_info_.header.payloadType = pltype;
- rtp_info_.header.timestamp = 0;
- rtp_info_.header.ssrc = 0x12345678;
- rtp_info_.header.markerBit = false;
- rtp_info_.header.sequenceNumber = 0;
- rtp_info_.frameType = kAudioFrameSpeech;
+ rtp_header_.payloadType = pltype;
+ rtp_header_.timestamp = 0;
+ rtp_header_.ssrc = 0x12345678;
+ rtp_header_.markerBit = false;
+ rtp_header_.sequenceNumber = 0;
int16_t audio[kFrameSizeSamples];
const int kRange = 0x7FF; // 2047, easy for masking.
@@ -98,10 +97,10 @@
static const int kInterarrivalJitterPacket = 2;
void Push() {
- rtp_info_.header.timestamp += kFrameSizeSamples;
- rtp_info_.header.sequenceNumber++;
- ASSERT_EQ(0,
- acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_info_));
+ rtp_header_.timestamp += kFrameSizeSamples;
+ rtp_header_.sequenceNumber++;
+ ASSERT_EQ(
+ 0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_header_));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
@@ -150,7 +149,7 @@
}
std::unique_ptr<AudioCodingModule> acm_;
- WebRtcRTPHeader rtp_info_;
+ RTPHeader rtp_header_;
uint8_t payload_[kPayloadLenBytes];
};