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[tor-commits] [tor-browser] 147/311: Bug 1756222: Do not disable SSRC switching if there were no a=ssrc attributes in the remote SDP. r=mjf, a=dmeehan
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pierov pushed a commit to branch geckoview-99.0.1-11.0-1
in repository tor-browser.
commit 6ddff345736733dd1fb35dc20e98d5d5df90857d
Author: Byron Campen <docfaraday@xxxxxxxxx>
AuthorDate: Wed Mar 9 16:23:31 2022 +0000
Bug 1756222: Do not disable SSRC switching if there were no a=ssrc attributes in the remote SDP. r=mjf, a=dmeehan
Differential Revision: https://phabricator.services.mozilla.com/D140236
---
dom/media/webrtc/jsapi/RTCRtpReceiver.cpp | 46 ++++++++++++++++---------------
1 file changed, 24 insertions(+), 22 deletions(-)
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
index 20843da496b75..50cc40fc940bf 100644
--- a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -660,18 +660,19 @@ nsresult RTCRtpReceiver::UpdateVideoConduit() {
: mJsepTransceiver->mRecvTrack.GetRtxSsrcs().front();
mSsrc = mJsepTransceiver->mRecvTrack.GetSsrcs().front();
mVideoRtxSsrc = rtxSsrc;
- }
- // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
- // packets (see bug 1405495) we could make this depending on the presence of
- // MID in the RTP packets instead of relying on the signaling.
- if (mJsepTransceiver->HasBundleLevel() &&
- (!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
- !mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
- webrtc::RtpExtension::kMidUri))) {
- mCallThread->Dispatch(
- NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
- &VideoSessionConduit::DisableSsrcChanges));
+ // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
+ // packets (see bug 1405495) we could make this depending on the presence of
+ // MID in the RTP packets instead of relying on the signaling.
+ // In any case, do not disable SSRC changes if no SSRCs were negotiated
+ if (mJsepTransceiver->HasBundleLevel() &&
+ (!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
+ !mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
+ webrtc::RtpExtension::kMidUri))) {
+ mCallThread->Dispatch(
+ NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
+ &VideoSessionConduit::DisableSsrcChanges));
+ }
}
if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
@@ -717,18 +718,19 @@ nsresult RTCRtpReceiver::UpdateAudioConduit() {
GetMid().c_str(), __FUNCTION__,
mJsepTransceiver->mRecvTrack.GetSsrcs().front()));
mSsrc = mJsepTransceiver->mRecvTrack.GetSsrcs().front();
- }
- // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
- // packets (see bug 1405495) we could make this depending on the presence of
- // MID in the RTP packets instead of relying on the signaling.
- if (mJsepTransceiver->HasBundleLevel() &&
- (!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
- !mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
- webrtc::RtpExtension::kMidUri))) {
- mCallThread->Dispatch(
- NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit,
- &AudioSessionConduit::DisableSsrcChanges));
+ // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
+ // packets (see bug 1405495) we could make this depending on the presence of
+ // MID in the RTP packets instead of relying on the signaling.
+ // In any case, do not disable SSRC changes if no SSRCs were negotiated
+ if (mJsepTransceiver->HasBundleLevel() &&
+ (!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
+ !mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
+ webrtc::RtpExtension::kMidUri))) {
+ mCallThread->Dispatch(
+ NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit,
+ &AudioSessionConduit::DisableSsrcChanges));
+ }
}
if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
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