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Make sure padding is sent on the first sending RTP module.

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R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
2014-07-24 16:41:25 +00:00
parent 45304ff0a7
commit f9460688a6
3 changed files with 6 additions and 10 deletions
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8
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -613,14 +613,10 @@ int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) {
}
} else {
CriticalSectionScoped lock(critical_section_module_ptrs_.get());
// Decide what media stream to pad on based on a round-robin scheme.
for (size_t i = 0; i < child_modules_.size(); ++i) {
padding_index_ = (padding_index_ + 1) % child_modules_.size();
// Send padding on one of the modules sending media.
if (child_modules_[padding_index_]->SendingMedia() &&
child_modules_[padding_index_]->rtp_sender_.GetTargetBitrate() > 0) {
return child_modules_[padding_index_]->rtp_sender_.TimeToSendPadding(
bytes);
if (child_modules_[i]->SendingMedia()) {
return child_modules_[i]->rtp_sender_.TimeToSendPadding(bytes);
}
}
}