[tor-commits] [tor-browser] 92/311: Backed out changeset 974fb4e6468c (bug 1754027) for breaking Google Voice on beta (bug 1756222) a=pascalc

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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 at 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 {
       RefPtr<mozilla::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>&());

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