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/Channel.cc b/modules/audio_coding/test/Channel.cc
index 4cb6c35..e54a29a 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -116,7 +116,8 @@
return 0;
}
- status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
+ status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize,
+ rtpInfo.header);
return status;
}
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index fbbc9d3..cd57ecd 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -168,7 +168,7 @@
}
EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
- _rtpInfo));
+ _rtpInfo.header));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 000041b..fd76224 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -57,7 +57,8 @@
}
if (!PacketLost()) {
- _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
+ _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
+ _rtpInfo.header);
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 1d0ddf8..141075b 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -15,6 +15,7 @@
#include <queue>
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
#include "rtc_base/synchronization/rw_lock_wrapper.h"
namespace webrtc {
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_;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 70065fb..2c71f46 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -46,14 +46,14 @@
const uint8_t* payload_data,
const size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
- WebRtcRTPHeader rtp_info;
+ RTPHeader rtp_header;
int32_t status = 0;
- rtp_info.header.markerBit = false;
- rtp_info.header.ssrc = 0;
- rtp_info.header.sequenceNumber = seq_no_++;
- rtp_info.header.payloadType = payload_type;
- rtp_info.header.timestamp = timestamp;
+ rtp_header.markerBit = false;
+ rtp_header.ssrc = 0;
+ rtp_header.sequenceNumber = seq_no_++;
+ rtp_header.payloadType = payload_type;
+ rtp_header.timestamp = timestamp;
if (frame_type == kEmptyFrame) {
// Skip this frame
return 0;
@@ -61,7 +61,7 @@
if (lost_packet_ == false) {
status =
- receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_info);
+ receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
if (frame_type != kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
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];
};