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Don't check MIDs when demuxing RTP packets in Call

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The MID header extension is handled by the RtpTransport
which lives above Call and takes care of demuxing to SSRC.

Bug: webrtc:4050
Change-Id: I27135e296ae9c7b15e926ba17547c26c75684ad6
Reviewed-on: https://webrtc-review.googlesource.com/65025
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22682}
This commit is contained in:
Steve Anton
2018-03-29 12:59:17 -07:00
committed by Commit Bot
parent 003930a3ce
commit ed09dc6f56
3 changed files with 13 additions and 2 deletions
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7
call/rtp_stream_receiver_controller.cc
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@ -35,7 +35,12 @@ RtpStreamReceiverController::Receiver::~Receiver() {
controller_->RemoveSink(sink_);
}
RtpStreamReceiverController::RtpStreamReceiverController() = default;
RtpStreamReceiverController::RtpStreamReceiverController() {
// At this level the demuxer is only configured to demux by SSRC, so don't
// worry about MIDs (MIDs are handled by upper layers).
demuxer_.set_use_mid(false);
}
RtpStreamReceiverController::~RtpStreamReceiverController() = default;
std::unique_ptr<RtpStreamReceiverInterface>