Adding upstream version 127.0.upstream/127.0

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

3 files changed, 10 insertions, 12 deletions

diff --git a/media/webrtc/signaling/gtest/MockCall.cpp b/media/webrtc/signaling/gtest/MockCall.cpp
index 7190974ab7..fb00c1a4b1 100644
--- a/media/webrtc/signaling/gtest/MockCall.cpp
+++ b/media/webrtc/signaling/gtest/MockCall.cpp
@@ -17,7 +17,7 @@ void MockAudioSendStream::Reconfigure(const Config& config,
 
 void MockAudioReceiveStream::SetDecoderMap(
     std::map<int, webrtc::SdpAudioFormat> decoder_map) {
-  MOZ_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome());
+  MOZ_RELEASE_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome());
   mCallWrapper->GetMockCall()->mAudioReceiveConfig->decoder_map =
       std::move(decoder_map);
 }
diff --git a/media/webrtc/signaling/gtest/MockCall.h b/media/webrtc/signaling/gtest/MockCall.h
index e0e9e92dcb..b1e12673f7 100644
--- a/media/webrtc/signaling/gtest/MockCall.h
+++ b/media/webrtc/signaling/gtest/MockCall.h
@@ -79,16 +79,16 @@ class MockAudioReceiveStream : public webrtc::AudioReceiveStreamInterface {
   virtual void SetDepacketizerToDecoderFrameTransformer(
       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
       override {
-    // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure
-    // method.
-    MOZ_ASSERT(false);
+    MOZ_CRASH(
+        "Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure "
+        "method.");
   }
   virtual void SetDecoderMap(
       std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
   virtual void SetNackHistory(int history_ms) override {
-    // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure
-    // method.
-    MOZ_ASSERT(false);
+    MOZ_CRASH(
+        "Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure "
+        "method.");
   }
   virtual void SetNonSenderRttMeasurement(bool enabled) override {}
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
@@ -126,8 +126,6 @@ class MockVideoSendStream : public webrtc::VideoSendStream {
 
   Stats GetStats() override { return mStats; }
 
-  void StartPerRtpStream(const std::vector<bool> active_layers) override {}
-
   void AddAdaptationResource(
       rtc::scoped_refptr<webrtc::Resource> resource) override {}
 
diff --git a/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp b/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
index 032023d95f..4aa7f18b3c 100644
--- a/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
+++ b/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
@@ -69,7 +69,7 @@ class FakeAudioTrack : public ProcessedMediaTrack {
 
   void Destroy() override {
     MutexAutoLock lock(mMutex);
-    MOZ_ASSERT(!mMainThreadDestroyed);
+    MOZ_RELEASE_ASSERT(!mMainThreadDestroyed);
     mMainThreadDestroyed = true;
     mTimer->Cancel();
     mTimer = nullptr;
@@ -79,14 +79,14 @@ class FakeAudioTrack : public ProcessedMediaTrack {
 
   void AddListener(MediaTrackListener* aListener) override {
     MutexAutoLock lock(mMutex);
-    MOZ_ASSERT(!mListener);
+    MOZ_RELEASE_ASSERT(!mListener);
     mListener = aListener;
   }
 
   RefPtr<GenericPromise> RemoveListener(
       MediaTrackListener* aListener) override {
     MutexAutoLock lock(mMutex);
-    MOZ_ASSERT(mListener == aListener);
+    MOZ_RELEASE_ASSERT(mListener == aListener);
     mListener = nullptr;
     return GenericPromise::CreateAndResolve(true, __func__);
   }