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/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 6afc161..ba8937e 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -199,7 +199,6 @@
WebRtcRTPHeader header;
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
- memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_EQ(0,
acm_->IncomingPacket(
packet->payload(),
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 350183b..d1cff23 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -82,7 +82,6 @@
rtp_header_.header.numCSRCs = 0;
rtp_header_.header.payloadType = 0;
rtp_header_.frameType = kAudioFrameSpeech;
- rtp_header_.type.Audio.isCNG = false;
}
void TearDown() override {}
@@ -135,10 +134,6 @@
rtp_header_.header.payloadType = payload_type;
rtp_header_.frameType = frame_type;
- if (frame_type == kAudioFrameSpeech)
- rtp_header_.type.Audio.isCNG = false;
- else
- rtp_header_.type.Audio.isCNG = true;
rtp_header_.header.timestamp = timestamp;
int ret_val = receiver_->InsertPacket(
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 7592300..ce2832a 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -78,8 +78,6 @@
rtp_header->frameType = kAudioFrameSpeech;
rtp_header->header.payload_type_frequency = kSampleRateHz;
- rtp_header->type.Audio.channel = 1;
- rtp_header->type.Audio.isCNG = false;
}
void Forward(WebRtcRTPHeader* rtp_header) {
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 8fdb677..bb970c1 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -40,11 +40,6 @@
? timeStamp
: static_cast<uint32_t>(external_send_timestamp_);
- if (frameType == kAudioFrameCN) {
- rtpInfo.type.Audio.isCNG = true;
- } else {
- rtpInfo.type.Audio.isCNG = false;
- }
if (frameType == kEmptyFrame) {
// When frame is empty, we should not transmit it. The frame size of the
// next non-empty frame will be based on the previous frame size.
@@ -52,7 +47,6 @@
return 0;
}
- rtpInfo.type.Audio.channel = 1;
// Treat fragmentation separately
if (fragmentation != NULL) {
// If silence for too long, send only new data.
@@ -89,11 +83,9 @@
if (_leftChannel) {
memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
_leftChannel = false;
- rtpInfo.type.Audio.channel = 1;
} else {
memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
_leftChannel = true;
- rtpInfo.type.Audio.channel = 2;
}
}
}
diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index a1329e7..d058384 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -222,8 +222,6 @@
EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
ParseRTPHeader(rtpInfo, rtpHeader);
- rtpInfo->type.Audio.isCNG = false;
- rtpInfo->type.Audio.channel = 1;
EXPECT_EQ(lengthBytes, plen + 8);
if (plen == 0) {
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);
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 2704d3d..31b1d07 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -63,13 +63,6 @@
}
if (lost_packet_ == false) {
- if (frame_type != kAudioFrameCN) {
- rtp_info.type.Audio.isCNG = false;
- rtp_info.type.Audio.channel = static_cast<int>(codec_mode_);
- } else {
- rtp_info.type.Audio.isCNG = true;
- rtp_info.type.Audio.channel = static_cast<int>(kMono);
- }
status =
receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_info);
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 7579d62..3b129ea 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -43,8 +43,6 @@
rtp_info_.header.ssrc = 0x12345678;
rtp_info_.header.markerBit = false;
rtp_info_.header.sequenceNumber = 0;
- rtp_info_.type.Audio.channel = 1;
- rtp_info_.type.Audio.isCNG = false;
rtp_info_.frameType = kAudioFrameSpeech;
int16_t audio[kFrameSizeSamples];