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Код Задачи Пакеты Проекты Релизы Вики Активность

Bug 1404977 - Part 13 - Remove useless mutex, and assert why they were useless, in WebRTCAudioDataListeners. r=pehrsons 

MozReview-Commit-ID: 2Mb5WZXbYgS

--HG--
extra : rebase_source : c4a7c8874919901eb11327cfb5f86d6b185be388
This commit is contained in:
Paul Adenot 2018-05-31 16:44:00 +02:00
Родитель ff9c7304e1
Коммит a1777772d7
5 изменённых файлов: 90 добавлений и 41 удалений
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6
dom/media/MediaStreamGraph.cpp
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@ -878,6 +878,10 @@ MediaStreamGraphImpl::CloseAudioInputImpl(Maybe<CubebUtils::AudioDeviceID>& aID,
MOZ_ASSERT(listeners);
DebugOnly<bool> wasPresent = listeners->RemoveElement(aListener);
MOZ_ASSERT(wasPresent);
// Breaks the cycle between the MSG and the listener.
aListener->Disconnect(this);
if (!listeners->IsEmpty()) {
// There is still a consumer for this audio input device
return;
@ -990,7 +994,7 @@ void MediaStreamGraphImpl::DeviceChangedImpl()
nsTArray<RefPtr<AudioDataListener>>* listeners =
mInputDeviceUsers.GetValue(mInputDeviceID);
for (auto& listener : *listeners) {
listener->DeviceChanged();
listener->DeviceChanged(this);
}
}

13
dom/media/MediaStreamGraph.h
Просмотреть файл

@ -106,26 +106,31 @@ public:
* cancellation. This is not guaranteed to be in any particular size
* chunks.
*/
virtual void NotifyOutputData(MediaStreamGraph* aGraph,
virtual void NotifyOutputData(MediaStreamGraphImpl* aGraph,
AudioDataValue* aBuffer, size_t aFrames,
TrackRate aRate, uint32_t aChannels) = 0;
/**
* Input data from a microphone (or other audio source. This is not
* guaranteed to be in any particular size chunks.
*/
virtual void NotifyInputData(MediaStreamGraph* aGraph,
virtual void NotifyInputData(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer, size_t aFrames,
TrackRate aRate, uint32_t aChannels) = 0;
/**
* Number of audio input channels.
*/
virtual uint32_t InputChannelCount() = 0;
virtual uint32_t RequestedInputChannelCount(MediaStreamGraphImpl* aGraph) = 0;
/**
* Called when the underlying audio device has changed.
*/
virtual void DeviceChanged() = 0;
virtual void DeviceChanged(MediaStreamGraphImpl* aGraph) = 0;
/**
* Called when the underlying audio device is being closed.
*/
virtual void Disconnect(MediaStreamGraphImpl* aGraph) = 0;
};
class AudioDataListener : public AudioDataListenerInterface {

2
dom/media/MediaStreamGraphImpl.h
Просмотреть файл

@ -484,7 +484,7 @@ public:
MOZ_ASSERT(listeners);
for (const auto& listener : *listeners) {
maxInputChannels =
std::max(maxInputChannels, listener->InputChannelCount());
std::max(maxInputChannels, listener->RequestedInputChannelCount(this));
}
return maxInputChannels;
}

30
dom/media/webrtc/MediaEngineWebRTC.h
Просмотреть файл

@ -160,6 +160,8 @@ private:
bool mManualInvalidation;
};
// This class is instantiated on the MediaManager thread, and is then sent and
// only ever access again on the MediaStreamGraph.
class WebRTCAudioDataListener : public AudioDataListener
{
protected:
@ -168,31 +170,29 @@ protected:
public:
explicit WebRTCAudioDataListener(MediaEngineWebRTCMicrophoneSource* aAudioSource)
: mMutex("WebRTCAudioDataListener::mMutex")
, mAudioSource(aAudioSource)
: mAudioSource(aAudioSource)
{}
// AudioDataListenerInterface methods
void NotifyOutputData(MediaStreamGraph* aGraph,
void NotifyOutputData(MediaStreamGraphImpl* aGraph,
AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
uint32_t aChannels) override;
void NotifyInputData(MediaStreamGraph* aGraph,
void NotifyInputData(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
uint32_t aChannels) override;
uint32_t InputChannelCount() override;
uint32_t RequestedInputChannelCount(MediaStreamGraphImpl* aGraph) override;
void DeviceChanged() override;
void DeviceChanged(MediaStreamGraphImpl* aGraph) override;
void Shutdown();
void Disconnect(MediaStreamGraphImpl* aGraph) override;
private:
Mutex mMutex;
RefPtr<MediaEngineWebRTCMicrophoneSource> mAudioSource;
};
@ -247,20 +247,22 @@ public:
const PrincipalHandle& aPrincipalHandle) override;
// AudioDataListenerInterface methods
void NotifyOutputData(MediaStreamGraph* aGraph,
void NotifyOutputData(MediaStreamGraphImpl* aGraph,
AudioDataValue* aBuffer, size_t aFrames,
TrackRate aRate, uint32_t aChannels) override;
void NotifyInputData(MediaStreamGraph* aGraph,
void NotifyInputData(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer, size_t aFrames,
TrackRate aRate, uint32_t aChannels) override;
void DeviceChanged() override;
void DeviceChanged(MediaStreamGraphImpl* aGraph) override;
uint32_t RequestedInputChannelCount(MediaStreamGraphImpl* aGraph) override
{
return GetRequestedInputChannelCount(aGraph);
}
void Disconnect(MediaStreamGraphImpl* aGraph) override;
dom::MediaSourceEnum GetMediaSource() const override
{
return dom::MediaSourceEnum::Microphone;
@ -364,7 +366,7 @@ private:
size_t aFrames,
uint32_t aChannels);
void PacketizeAndProcess(MediaStreamGraph* aGraph,
void PacketizeAndProcess(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
@ -381,7 +383,9 @@ private:
uint32_t GetRequestedInputChannelCount(MediaStreamGraphImpl* aGraphImpl);
void SetRequestedInputChannelCount(uint32_t aRequestedInputChannelCount);
// Owning thread only.
// mListener is created on the MediaManager thread, and then sent to the MSG
// thread. On shutdown, we send this pointer to the MSG thread again, telling
// it to clean up.
RefPtr<WebRTCAudioDataListener> mListener;
// Can be shared on any thread.

80
dom/media/webrtc/MediaEngineWebRTCAudio.cpp
Просмотреть файл

@ -58,49 +58,52 @@ WebRTCAudioDataListener::NotifyOutputData(MediaStreamGraph* aGraph,
TrackRate aRate,
uint32_t aChannels)
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (mAudioSource) {
mAudioSource->NotifyOutputData(aGraph, aBuffer, aFrames, aRate, aChannels);
}
}
void
WebRTCAudioDataListener::NotifyInputData(MediaStreamGraph* aGraph,
WebRTCAudioDataListener::NotifyInputData(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
uint32_t aChannels)
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (mAudioSource) {
mAudioSource->NotifyInputData(aGraph, aBuffer, aFrames, aRate, aChannels);
}
}
void
WebRTCAudioDataListener::DeviceChanged()
WebRTCAudioDataListener::DeviceChanged(MediaStreamGraphImpl* aGraph)
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (mAudioSource) {
mAudioSource->DeviceChanged();
mAudioSource->DeviceChanged(aGraph);
}
}
uint32_t
WebRTCAudioDataListener::InputChannelCount()
WebRTCAudioDataListener::RequestedInputChannelCount(MediaStreamGraphImpl* aGraph)
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (mAudioSource) {
return mAudioSource->InputChannelCount();
return mAudioSource->RequestedInputChannelCount(aGraph);
}
return 0;
}
void
WebRTCAudioDataListener::Shutdown()
WebRTCAudioDataListener::Disconnect(MediaStreamGraphImpl* aGraph)
{
MutexAutoLock lock(mMutex);
mAudioSource = nullptr;
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (mAudioSource) {
mAudioSource->Disconnect(aGraph);
mAudioSource = nullptr;
}
}
/**
@ -529,6 +532,9 @@ MediaEngineWebRTCMicrophoneSource::SetPassThrough(bool aPassThrough)
uint32_t
MediaEngineWebRTCMicrophoneSource::GetRequestedInputChannelCount(MediaStreamGraphImpl* aGraphImpl)
{
MOZ_ASSERT(aGraphImpl->CurrentDriver()->OnThread(),
"Wrong calling pattern, don't call this before ::SetTrack.");
if (mState == kReleased) {
// This source has been released, and is waiting for collection. Simply
// return 0, this source won't contribute to the channel count decision.
@ -544,6 +550,9 @@ MediaEngineWebRTCMicrophoneSource::SetRequestedInputChannelCount(
uint32_t aRequestedInputChannelCount)
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(mAllocations.Length() <= 1);
if (mAllocations.IsEmpty()) {
return;
}
@ -561,6 +570,12 @@ MediaEngineWebRTCMicrophoneSource::ApplySettings(const MediaEnginePrefs& aPrefs,
{
AssertIsOnOwningThread();
MOZ_DIAGNOSTIC_ASSERT(aGraph);
#ifdef DEBUG
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(mAllocations.Length() <= 1);
}
#endif
RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
NS_DispatchToMainThread(media::NewRunnableFrom([that, graph = std::move(aGraph), aPrefs]() mutable {
@ -616,6 +631,12 @@ MediaEngineWebRTCMicrophoneSource::Allocate(const dom::MediaTrackConstraints &aC
auto handle = MakeRefPtr<AllocationHandle>(aConstraints, aPrincipalInfo,
aDeviceId);
#ifdef DEBUG
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(mAllocations.Length() <= 1);
}
#endif
LOG(("Mic source %p allocation %p Allocate()", this, handle.get()));
nsresult rv = ReevaluateAllocation(handle, nullptr, aPrefs, aDeviceId,
@ -626,6 +647,7 @@ MediaEngineWebRTCMicrophoneSource::Allocate(const dom::MediaTrackConstraints &aC
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(mAllocations.IsEmpty(), "Only allocate once.");
mAllocations.AppendElement(Allocation(handle));
}
@ -638,6 +660,8 @@ MediaEngineWebRTCMicrophoneSource::Deallocate(const RefPtr<const AllocationHandl
{
AssertIsOnOwningThread();
MOZ_ASSERT(mState == kStopped);
size_t i = mAllocations.IndexOf(aHandle, 0, AllocationHandleComparator());
MOZ_DIAGNOSTIC_ASSERT(i != mAllocations.NoIndex);
MOZ_DIAGNOSTIC_ASSERT(!mAllocations[i].mEnabled,
@ -649,6 +673,7 @@ MediaEngineWebRTCMicrophoneSource::Deallocate(const RefPtr<const AllocationHandl
{
MutexAutoLock lock(mMutex);
MOZ_ASSERT(mAllocations.Length() == 1, "Only allocate once.");
mAllocations.RemoveElementAt(i);
}
@ -682,6 +707,8 @@ MediaEngineWebRTCMicrophoneSource::SetTrack(const RefPtr<const AllocationHandle>
return NS_ERROR_NOT_AVAILABLE;
}
MOZ_ASSERT(mAllocations.Length() == 1, "Only allocate once.");
size_t i = mAllocations.IndexOf(aHandle, 0, AllocationHandleComparator());
MOZ_DIAGNOSTIC_ASSERT(i != mAllocations.NoIndex);
MOZ_ASSERT(!mAllocations[i].mStream);
@ -766,6 +793,7 @@ MediaEngineWebRTCMicrophoneSource::Start(const RefPtr<const AllocationHandle>& a
nsresult
MediaEngineWebRTCMicrophoneSource::Stop(const RefPtr<const AllocationHandle>& aHandle)
{
MOZ_ASSERT(mAllocations.Length() <= 1);
AssertIsOnOwningThread();
LOG(("Mic source %p allocation %p Stop()", this, aHandle.get()));
@ -774,6 +802,7 @@ MediaEngineWebRTCMicrophoneSource::Stop(const RefPtr<const AllocationHandle>& aH
MOZ_DIAGNOSTIC_ASSERT(i != mAllocations.NoIndex,
"Cannot stop track that we don't know about");
Allocation& allocation = mAllocations[i];
MOZ_ASSERT(allocation.mStream, "SetTrack must have been called before ::Stop");
if (!allocation.mEnabled) {
// Already stopped - this is allowed
@ -788,6 +817,7 @@ MediaEngineWebRTCMicrophoneSource::Stop(const RefPtr<const AllocationHandle>& aH
CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
Maybe<CubebUtils::AudioDeviceID> id = Some(deviceID);
allocation.mStream->CloseAudioInput(id, mListener);
mListener = nullptr;
if (HasEnabledTrack()) {
// Another track is keeping us from stopping
@ -798,12 +828,6 @@ MediaEngineWebRTCMicrophoneSource::Stop(const RefPtr<const AllocationHandle>& aH
mState = kStopped;
}
if (mListener) {
// breaks a cycle, since the WebRTCAudioDataListener has a RefPtr to us
mListener->Shutdown();
mListener = nullptr;
}
return NS_OK;
}
@ -884,12 +908,14 @@ MediaEngineWebRTCMicrophoneSource::Pull(const RefPtr<const AllocationHandle>& aH
}
void
MediaEngineWebRTCMicrophoneSource::NotifyOutputData(MediaStreamGraph* aGraph,
MediaEngineWebRTCMicrophoneSource::NotifyOutputData(MediaStreamGraphImpl* aGraph,
AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
uint32_t aChannels)
{
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
if (!mPacketizerOutput ||
mPacketizerOutput->PacketSize() != aRate/100u ||
mPacketizerOutput->Channels() != aChannels) {
@ -977,7 +1003,7 @@ MediaEngineWebRTCMicrophoneSource::NotifyOutputData(MediaStreamGraph* aGraph,
// Only called if we're not in passthrough mode
void
MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess(MediaStreamGraph* aGraph,
MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
@ -1202,12 +1228,13 @@ MediaEngineWebRTCMicrophoneSource::InsertInGraph(const T* aBuffer,
// Called back on GraphDriver thread!
// Note this can be called back after ::Shutdown()
void
MediaEngineWebRTCMicrophoneSource::NotifyInputData(MediaStreamGraph* aGraph,
MediaEngineWebRTCMicrophoneSource::NotifyInputData(MediaStreamGraphImpl* aGraph,
const AudioDataValue* aBuffer,
size_t aFrames,
TrackRate aRate,
uint32_t aChannels)
{
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
TRACE_AUDIO_CALLBACK();
{
@ -1253,14 +1280,23 @@ do { \
} while(0)
void
MediaEngineWebRTCMicrophoneSource::DeviceChanged()
MediaEngineWebRTCMicrophoneSource::DeviceChanged(MediaStreamGraphImpl* aGraph)
{
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
// Reset some processing
ResetProcessingIfNeeded(gain_control);
ResetProcessingIfNeeded(echo_cancellation);
ResetProcessingIfNeeded(noise_suppression);
}
void
MediaEngineWebRTCMicrophoneSource::Disconnect(MediaStreamGraphImpl* aGraph)
{
// This method is just for asserts.
MOZ_ASSERT(aGraph->CurrentDriver()->OnThread());
MOZ_ASSERT(!mListener);
}
void
MediaEngineWebRTCMicrophoneSource::Shutdown()
{