zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.

Browse Source 
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}
This commit is contained in:
philipel
2018-06-19 16:18:31 +02:00
committed by Commit Bot
parent 7e9a619599
commit 0a5fe77d23
17 changed files with 3 additions and 109 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
modules/audio_coding/test/Channel.cc
View File

@ -40,11 +40,6 @@ int32_t Channel::SendData(FrameType frameType,
? 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 @@ int32_t Channel::SendData(FrameType frameType,
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 @@ int32_t Channel::SendData(FrameType frameType,
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;
}
}
}

2
modules/audio_coding/test/RTPFile.cc
View File

@ -222,8 +222,6 @@ size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
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) {

7
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -69,18 +69,13 @@ int32_t TestPack::SendData(FrameType frame_type,
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);

7
modules/audio_coding/test/TestStereo.cc
View File

@ -63,13 +63,6 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
}
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);

2
modules/audio_coding/test/target_delay_unittest.cc
View File

@ -43,8 +43,6 @@ class TargetDelayTest : public ::testing::Test {
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];