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pierov pushed a commit to branch geckoview-99.0.1-11.0-1 in repository tor-browser.
commit 7a92d1a19bd2c580a12d1db297ce7710cc56420b Author: Pascal Chevrel pchevrel@mozilla.com AuthorDate: Mon Feb 28 17:28:54 2022 +0100
Backed out changeset 974fb4e6468c (bug 1754027) for breaking Google Voice on beta (bug 1756222) a=pascalc --- dom/media/webrtc/jsapi/RTCRtpReceiver.cpp | 12 ------------ dom/media/webrtc/libwebrtcglue/AudioConduit.cpp | 2 +- dom/media/webrtc/libwebrtcglue/AudioConduit.h | 9 --------- dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h | 4 ++-- media/webrtc/signaling/gtest/MockConduit.h | 1 - 5 files changed, 3 insertions(+), 25 deletions(-)
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp index 20843da496b75..8c2c214f6d936 100644 --- a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp +++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp @@ -719,18 +719,6 @@ nsresult RTCRtpReceiver::UpdateAudioConduit() { 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)); - } - if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() && mJsepTransceiver->mRecvTrack.GetActive()) { const auto& details(*mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()); diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp index 7fff2abfdd25f..da61ffa79095b 100644 --- a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp +++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp @@ -488,7 +488,7 @@ void WebrtcAudioConduit::OnRtpReceived(MediaPacket&& aPacket, webrtc::RTPHeader&& aHeader) { MOZ_ASSERT(mCallThread->IsOnCurrentThread());
- if (mAllowSsrcChange && mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc) { + if (mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc) { CSFLogDebug(LOGTAG, "%s: switching from SSRC %u to %u", __FUNCTION__, mRecvStreamConfig.rtp.remote_ssrc, aHeader.ssrc); OverrideRemoteSSRC(aHeader.ssrc); diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.h b/dom/media/webrtc/libwebrtcglue/AudioConduit.h index c503cff854df9..26d968938a01f 100644 --- a/dom/media/webrtc/libwebrtcglue/AudioConduit.h +++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.h @@ -153,11 +153,6 @@ class WebrtcAudioConduit : public AudioSessionConduit, Ssrcs GetLocalSSRCs() const override; Maybe<Ssrc> GetRemoteSSRC() const override;
- void DisableSsrcChanges() override { - MOZ_ASSERT(mCallThread->IsOnCurrentThread()); - mAllowSsrcChange = false; - } - private: /** * Override the remote ssrc configured on mRecvStreamConfig. @@ -209,10 +204,6 @@ class WebrtcAudioConduit : public AudioSessionConduit, void CreateRecvStream(); void DeleteRecvStream();
- // Are SSRC changes without signaling allowed or not. - // Call thread only. - bool mAllowSsrcChange = true; - // Const so can be accessed on any thread. Most methods are called on the Call // thread. const RefPtr<WebrtcCallWrapper> mCall; diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h index ad4fbef42fed1..7797267d1679e 100644 --- a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h +++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h @@ -149,8 +149,6 @@ class MediaSessionConduit { virtual Maybe<Ssrc> GetRemoteSSRC() const = 0; virtual void UnsetRemoteSSRC(Ssrc aSsrc) = 0;
- virtual void DisableSsrcChanges() = 0; - virtual bool HasCodecPluginID(uint64_t aPluginID) const = 0;
virtual MediaEventSource<void>& RtcpByeEvent() = 0; @@ -363,6 +361,8 @@ class VideoSessionConduit : public MediaSessionConduit { RefPtrmozilla::VideoRenderer aRenderer) = 0; virtual void DetachRenderer() = 0;
+ virtual void DisableSsrcChanges() = 0; + /** * Function to deliver a capture video frame for encoding and transport. * If the frame's timestamp is 0, it will be automatcally generated. diff --git a/media/webrtc/signaling/gtest/MockConduit.h b/media/webrtc/signaling/gtest/MockConduit.h index a4be2e1a0c88e..d76fce8dd2b01 100644 --- a/media/webrtc/signaling/gtest/MockConduit.h +++ b/media/webrtc/signaling/gtest/MockConduit.h @@ -42,7 +42,6 @@ class MockConduit : public MediaSessionConduit { MOCK_CONST_METHOD0(GetLocalSSRCs, Ssrcs()); MOCK_CONST_METHOD0(GetRemoteSSRC, Maybe<Ssrc>()); MOCK_METHOD1(UnsetRemoteSSRC, void(Ssrc)); - MOCK_METHOD0(DisableSsrcChanges, void()); MOCK_CONST_METHOD1(HasCodecPluginID, bool(uint64_t)); MOCK_METHOD0(RtcpByeEvent, MediaEventSource<void>&()); MOCK_METHOD0(RtcpTimeoutEvent, MediaEventSource<void>&());