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-rw-r--r--media/webrtc/signaling/gtest/MockCall.cpp2
-rw-r--r--media/webrtc/signaling/gtest/MockCall.h14
-rw-r--r--media/webrtc/signaling/gtest/mediapipeline_unittest.cpp6
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__);
}