зеркало из https://github.com/mozilla/gecko-dev.git
308 строки
9.8 KiB
C++
308 строки
9.8 KiB
C++
/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef MEDIAENGINEWEBRTC_H_
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#define MEDIAENGINEWEBRTC_H_
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#include "prcvar.h"
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#include "prthread.h"
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#include "nsIThread.h"
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#include "nsIRunnable.h"
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#include "mozilla/dom/File.h"
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#include "mozilla/Mutex.h"
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#include "mozilla/Monitor.h"
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#include "nsCOMPtr.h"
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#include "nsThreadUtils.h"
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#include "DOMMediaStream.h"
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#include "nsDirectoryServiceDefs.h"
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#include "nsComponentManagerUtils.h"
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#include "nsRefPtrHashtable.h"
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#include "VideoUtils.h"
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#include "MediaEngineCameraVideoSource.h"
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#include "VideoSegment.h"
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#include "AudioSegment.h"
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#include "StreamBuffer.h"
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#include "MediaStreamGraph.h"
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#include "MediaEngineWrapper.h"
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#include "mozilla/dom/MediaStreamTrackBinding.h"
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// WebRTC library includes follow
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#include "webrtc/common.h"
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// Audio Engine
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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#include "webrtc/voice_engine/include/voe_hardware.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_volume_control.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/include/voe_call_report.h"
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// Video Engine
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// conflicts with #include of scoped_ptr.h
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#undef FF
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#include "webrtc/video_engine/include/vie_base.h"
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#include "webrtc/video_engine/include/vie_codec.h"
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#include "webrtc/video_engine/include/vie_render.h"
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#include "webrtc/video_engine/include/vie_capture.h"
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#include "NullTransport.h"
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#include "AudioOutputObserver.h"
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namespace mozilla {
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/**
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* The WebRTC implementation of the MediaEngine interface.
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*/
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class MediaEngineWebRTCVideoSource : public MediaEngineCameraVideoSource
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, public webrtc::ExternalRenderer
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{
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public:
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NS_DECL_THREADSAFE_ISUPPORTS
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// ViEExternalRenderer.
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virtual int FrameSizeChange(unsigned int w, unsigned int h, unsigned int streams);
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virtual int DeliverFrame(unsigned char* buffer,
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int size,
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uint32_t time_stamp,
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int64_t render_time,
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void *handle);
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/**
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* Does DeliverFrame() support a null buffer and non-null handle
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* (video texture)?
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* XXX Investigate! Especially for Android/B2G
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*/
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virtual bool IsTextureSupported() { return false; }
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MediaEngineWebRTCVideoSource(webrtc::VideoEngine* aVideoEnginePtr, int aIndex,
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MediaSourceType aMediaSource = MediaSourceType::Camera)
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: MediaEngineCameraVideoSource(aIndex, "WebRTCCamera.Monitor")
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, mVideoEngine(aVideoEnginePtr)
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, mMinFps(-1)
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, mMediaSource(aMediaSource)
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{
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MOZ_ASSERT(aVideoEnginePtr);
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Init();
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}
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virtual nsresult Allocate(const VideoTrackConstraintsN& aConstraints,
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const MediaEnginePrefs& aPrefs);
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virtual nsresult Deallocate();
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virtual nsresult Start(SourceMediaStream*, TrackID);
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virtual nsresult Stop(SourceMediaStream*, TrackID);
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virtual void NotifyPull(MediaStreamGraph* aGraph,
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SourceMediaStream* aSource,
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TrackID aId,
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StreamTime aDesiredTime,
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StreamTime &aLastEndTime);
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virtual const MediaSourceType GetMediaSource() {
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return mMediaSource;
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}
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virtual nsresult TakePhoto(PhotoCallback* aCallback)
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{
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return NS_ERROR_NOT_IMPLEMENTED;
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}
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void Refresh(int aIndex);
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bool SatisfiesConstraintSets(
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const nsTArray<const dom::MediaTrackConstraintSet*>& aConstraintSets);
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protected:
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~MediaEngineWebRTCVideoSource() { Shutdown(); }
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private:
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// Initialize the needed Video engine interfaces.
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void Init();
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void Shutdown();
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// Engine variables.
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webrtc::VideoEngine* mVideoEngine; // Weak reference, don't free.
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webrtc::ViEBase* mViEBase;
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webrtc::ViECapture* mViECapture;
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webrtc::ViERender* mViERender;
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int mMinFps; // Min rate we want to accept
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MediaSourceType mMediaSource; // source of media (camera | application | screen)
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static bool SatisfiesConstraintSet(const dom::MediaTrackConstraintSet& aConstraints,
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const webrtc::CaptureCapability& aCandidate);
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void ChooseCapability(const VideoTrackConstraintsN& aConstraints,
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const MediaEnginePrefs& aPrefs);
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};
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class MediaEngineWebRTCAudioSource : public MediaEngineAudioSource,
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public webrtc::VoEMediaProcess
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{
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public:
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MediaEngineWebRTCAudioSource(nsIThread* aThread, webrtc::VoiceEngine* aVoiceEnginePtr,
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int aIndex, const char* name, const char* uuid)
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: MediaEngineAudioSource(kReleased)
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, mSamples(0)
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, mVoiceEngine(aVoiceEnginePtr)
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, mMonitor("WebRTCMic.Monitor")
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, mThread(aThread)
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, mCapIndex(aIndex)
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, mChannel(-1)
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, mInitDone(false)
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, mStarted(false)
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, mEchoOn(false), mAgcOn(false), mNoiseOn(false)
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, mEchoCancel(webrtc::kEcDefault)
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, mAGC(webrtc::kAgcDefault)
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, mNoiseSuppress(webrtc::kNsDefault)
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, mPlayoutDelay(0)
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, mNullTransport(nullptr) {
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MOZ_ASSERT(aVoiceEnginePtr);
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mDeviceName.Assign(NS_ConvertUTF8toUTF16(name));
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mDeviceUUID.Assign(NS_ConvertUTF8toUTF16(uuid));
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Init();
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}
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virtual void GetName(nsAString& aName);
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virtual void GetUUID(nsAString& aUUID);
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virtual nsresult Allocate(const AudioTrackConstraintsN& aConstraints,
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const MediaEnginePrefs& aPrefs);
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virtual nsresult Deallocate();
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virtual nsresult Start(SourceMediaStream* aStream, TrackID aID);
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virtual nsresult Stop(SourceMediaStream* aSource, TrackID aID);
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virtual void SetDirectListeners(bool aHasDirectListeners) {};
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virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
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bool aAgcOn, uint32_t aAGC,
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bool aNoiseOn, uint32_t aNoise,
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int32_t aPlayoutDelay);
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virtual void NotifyPull(MediaStreamGraph* aGraph,
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SourceMediaStream* aSource,
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TrackID aId,
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StreamTime aDesiredTime,
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StreamTime &aLastEndTime);
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virtual bool IsFake() {
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return false;
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}
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virtual const MediaSourceType GetMediaSource() {
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return MediaSourceType::Microphone;
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}
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virtual nsresult TakePhoto(PhotoCallback* aCallback)
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{
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return NS_ERROR_NOT_IMPLEMENTED;
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}
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// VoEMediaProcess.
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void Process(int channel, webrtc::ProcessingTypes type,
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int16_t audio10ms[], int length,
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int samplingFreq, bool isStereo);
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NS_DECL_THREADSAFE_ISUPPORTS
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protected:
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~MediaEngineWebRTCAudioSource() { Shutdown(); }
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// mSamples is an int to avoid conversions when comparing/etc to
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// samplingFreq & length. Making mSamples protected instead of private is a
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// silly way to avoid -Wunused-private-field warnings when PR_LOGGING is not
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// #defined. mSamples is not actually expected to be used by a derived class.
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int mSamples;
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private:
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void Init();
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void Shutdown();
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webrtc::VoiceEngine* mVoiceEngine;
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ScopedCustomReleasePtr<webrtc::VoEBase> mVoEBase;
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ScopedCustomReleasePtr<webrtc::VoEExternalMedia> mVoERender;
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ScopedCustomReleasePtr<webrtc::VoENetwork> mVoENetwork;
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ScopedCustomReleasePtr<webrtc::VoEAudioProcessing> mVoEProcessing;
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ScopedCustomReleasePtr<webrtc::VoECallReport> mVoECallReport;
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// mMonitor protects mSources[] access/changes, and transitions of mState
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// from kStarted to kStopped (which are combined with EndTrack()).
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// mSources[] is accessed from webrtc threads.
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Monitor mMonitor;
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nsTArray<SourceMediaStream*> mSources; // When this goes empty, we shut down HW
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nsCOMPtr<nsIThread> mThread;
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int mCapIndex;
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int mChannel;
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TrackID mTrackID;
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bool mInitDone;
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bool mStarted;
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nsString mDeviceName;
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nsString mDeviceUUID;
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bool mEchoOn, mAgcOn, mNoiseOn;
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webrtc::EcModes mEchoCancel;
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webrtc::AgcModes mAGC;
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webrtc::NsModes mNoiseSuppress;
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int32_t mPlayoutDelay;
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NullTransport *mNullTransport;
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};
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class MediaEngineWebRTC : public MediaEngine
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{
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public:
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explicit MediaEngineWebRTC(MediaEnginePrefs& aPrefs);
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// Clients should ensure to clean-up sources video/audio sources
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// before invoking Shutdown on this class.
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void Shutdown();
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virtual void EnumerateVideoDevices(MediaSourceType,
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nsTArray<nsRefPtr<MediaEngineVideoSource> >*);
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virtual void EnumerateAudioDevices(MediaSourceType,
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nsTArray<nsRefPtr<MediaEngineAudioSource> >*);
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private:
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~MediaEngineWebRTC() {
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Shutdown();
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#ifdef MOZ_B2G_CAMERA
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AsyncLatencyLogger::Get()->Release();
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#endif
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gFarendObserver = nullptr;
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}
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nsCOMPtr<nsIThread> mThread;
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Mutex mMutex;
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// protected with mMutex:
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webrtc::VideoEngine* mScreenEngine;
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webrtc::VideoEngine* mBrowserEngine;
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webrtc::VideoEngine* mWinEngine;
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webrtc::VideoEngine* mAppEngine;
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webrtc::VideoEngine* mVideoEngine;
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webrtc::VoiceEngine* mVoiceEngine;
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// specialized configurations
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webrtc::Config mAppEngineConfig;
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webrtc::Config mWinEngineConfig;
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webrtc::Config mScreenEngineConfig;
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webrtc::Config mBrowserEngineConfig;
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// Need this to avoid unneccesary WebRTC calls while enumerating.
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bool mVideoEngineInit;
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bool mAudioEngineInit;
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bool mScreenEngineInit;
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bool mBrowserEngineInit;
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bool mWinEngineInit;
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bool mAppEngineInit;
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bool mHasTabVideoSource;
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// Store devices we've already seen in a hashtable for quick return.
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// Maps UUID to MediaEngineSource (one set for audio, one for video).
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nsRefPtrHashtable<nsStringHashKey, MediaEngineVideoSource> mVideoSources;
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nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCAudioSource> mAudioSources;
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};
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}
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#endif /* NSMEDIAENGINEWEBRTC_H_ */
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