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Remove call to RtpRtcp::RegisterSendPayload for the default RTP module.

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The send payload type is only checked in RTPSender::CheckPayloadType,
which in turn is only called from SendOutgoingData and never from the
default module anylonger.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39949004

Cr-Commit-Position: refs/heads/master@{#8357}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8357 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2015-02-13 09:52:01 +00:00
parent 40367f984b
commit 2bd299a172
2 changed files with 2 additions and 8 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -283,6 +283,7 @@ int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
const CodecInst& voice_codec) {
assert(!IsDefaultModule());
return rtp_sender_.RegisterPayload(
voice_codec.plname,
voice_codec.pltype,

9
webrtc/video_engine/vie_encoder.cc
View File

@ -219,11 +219,7 @@ bool ViEEncoder::Init() {
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
PayloadRouter::DefaultMaxPayloadLength()) !=
0) {
return false;
}
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
PayloadRouter::DefaultMaxPayloadLength()) != 0) {
return false;
}
}
@ -377,9 +373,6 @@ int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
return -1;
}
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
return -1;
}
// Convert from kbps to bps.
std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
video_codec.startBitrate * 1000,