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

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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
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8
modules/audio_coding/test/Channel.cc
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@ -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;
}
}
}