mirror of
https://github.com/mozilla/gecko-dev.git
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419 lines
13 KiB
C++
419 lines
13 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/Mutex.h"
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#include "mozilla/Monitor.h"
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#include "nsCOMPtr.h"
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#include "nsDOMFile.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 "MediaEngine.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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// WebRTC library includes follow
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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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#ifdef MOZ_B2G_CAMERA
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#include "CameraControlListener.h"
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#include "ICameraControl.h"
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#include "ImageContainer.h"
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#include "nsGlobalWindow.h"
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#include "prprf.h"
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#include "mozilla/Hal.h"
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#endif
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#include "NullTransport.h"
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#include "AudioOutputObserver.h"
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namespace mozilla {
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#ifdef MOZ_B2G_CAMERA
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class CameraAllocateRunnable;
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class GetCameraNameRunnable;
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#endif
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/**
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* The WebRTC implementation of the MediaEngine interface.
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*
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* On B2G platform, member data may accessed from different thread after construction:
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*
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* MediaThread:
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* mState, mImage, mWidth, mHeight, mCapability, mPrefs, mDeviceName, mUniqueId, mInitDone,
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* mImageContainer, mSources, mState, mImage
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*
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* MainThread:
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* mCaptureIndex, mLastCapture, mState, mWidth, mHeight,
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*
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* Where mWidth, mHeight, mImage are protected by mMonitor
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* mState is protected by mCallbackMonitor
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* Other variable is accessed only from single thread
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*/
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class MediaEngineWebRTCVideoSource : public MediaEngineVideoSource
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, public nsRunnable
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#ifdef MOZ_B2G_CAMERA
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, public CameraControlListener
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, public mozilla::hal::ScreenConfigurationObserver
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#else
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, public webrtc::ExternalRenderer
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#endif
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{
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public:
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#ifdef MOZ_B2G_CAMERA
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MediaEngineWebRTCVideoSource(int aIndex)
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: mCameraControl(nullptr)
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, mCallbackMonitor("WebRTCCamera.CallbackMonitor")
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, mRotation(0)
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, mBackCamera(false)
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, mCaptureIndex(aIndex)
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, mMonitor("WebRTCCamera.Monitor")
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, mWidth(0)
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, mHeight(0)
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, mHasDirectListeners(false)
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, mInitDone(false)
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, mInSnapshotMode(false)
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, mSnapshotPath(nullptr)
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{
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mState = kReleased;
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Init();
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}
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#else
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// ViEExternalRenderer.
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virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int);
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virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t,
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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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: mVideoEngine(aVideoEnginePtr)
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, mCaptureIndex(aIndex)
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, mFps(-1)
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, mMinFps(-1)
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, mMonitor("WebRTCCamera.Monitor")
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, mWidth(0)
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, mHeight(0)
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, mHasDirectListeners(false)
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, mInitDone(false)
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, mInSnapshotMode(false)
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, mSnapshotPath(nullptr) {
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MOZ_ASSERT(aVideoEnginePtr);
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mState = kReleased;
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Init();
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}
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#endif
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virtual void GetName(nsAString&);
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virtual void GetUUID(nsAString&);
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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 SetDirectListeners(bool aHasListeners);
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virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
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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) { return NS_OK; };
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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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TrackTicks &aLastEndTime);
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virtual bool IsFake() {
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return false;
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}
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#ifndef MOZ_B2G_CAMERA
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NS_DECL_THREADSAFE_ISUPPORTS
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#else
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// We are subclassed from CameraControlListener, which implements a
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// threadsafe reference-count for us.
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NS_DECL_ISUPPORTS_INHERITED
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void OnHardwareStateChange(HardwareState aState);
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void GetRotation();
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bool OnNewPreviewFrame(layers::Image* aImage, uint32_t aWidth, uint32_t aHeight);
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void OnUserError(UserContext aContext, nsresult aError);
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void OnTakePictureComplete(uint8_t* aData, uint32_t aLength, const nsAString& aMimeType);
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void AllocImpl();
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void DeallocImpl();
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void StartImpl(webrtc::CaptureCapability aCapability);
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void StopImpl();
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void SnapshotImpl();
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void RotateImage(layers::Image* aImage, uint32_t aWidth, uint32_t aHeight);
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void Notify(const mozilla::hal::ScreenConfiguration& aConfiguration);
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#endif
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// This runnable is for creating a temporary file on the main thread.
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NS_IMETHODIMP
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Run()
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{
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nsCOMPtr<nsIFile> tmp;
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nsresult rv = NS_GetSpecialDirectory(NS_OS_TEMP_DIR, getter_AddRefs(tmp));
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NS_ENSURE_SUCCESS(rv, rv);
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tmp->Append(NS_LITERAL_STRING("webrtc_snapshot.jpeg"));
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rv = tmp->CreateUnique(nsIFile::NORMAL_FILE_TYPE, 0600);
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NS_ENSURE_SUCCESS(rv, rv);
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mSnapshotPath = new nsString();
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rv = tmp->GetPath(*mSnapshotPath);
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NS_ENSURE_SUCCESS(rv, rv);
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return NS_OK;
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}
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protected:
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~MediaEngineWebRTCVideoSource() { Shutdown(); }
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private:
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static const unsigned int KMaxDeviceNameLength = 128;
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static const unsigned int KMaxUniqueIdLength = 256;
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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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#ifdef MOZ_B2G_CAMERA
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mozilla::ReentrantMonitor mCallbackMonitor; // Monitor for camera callback handling
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// This is only modified on MainThread (AllocImpl and DeallocImpl)
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nsRefPtr<ICameraControl> mCameraControl;
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nsCOMPtr<nsIDOMFile> mLastCapture;
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// These are protected by mMonitor below
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int mRotation;
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int mCameraAngle; // See dom/base/ScreenOrientation.h
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bool mBackCamera;
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#else
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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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#endif
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webrtc::CaptureCapability mCapability; // Doesn't work on OS X.
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int mCaptureIndex;
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int mFps; // Track rate (30 fps by default)
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int mMinFps; // Min rate we want to accept
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// mMonitor protects mImage access/changes, and transitions of mState
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// from kStarted to kStopped (which are combined with EndTrack() and
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// image changes). Note that mSources is not accessed from other threads
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// for video and is not protected.
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Monitor mMonitor; // Monitor for processing WebRTC frames.
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int mWidth, mHeight;
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nsRefPtr<layers::Image> mImage;
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nsRefPtr<layers::ImageContainer> mImageContainer;
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bool mHasDirectListeners;
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nsTArray<SourceMediaStream *> mSources; // When this goes empty, we shut down HW
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bool mInitDone;
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bool mInSnapshotMode;
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nsString* mSnapshotPath;
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nsString mDeviceName;
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nsString mUniqueId;
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void ChooseCapability(const VideoTrackConstraintsN &aConstraints,
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const MediaEnginePrefs &aPrefs);
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void GuessCapability(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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: 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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mState = kReleased;
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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&);
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virtual void GetUUID(nsAString&);
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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*, TrackID);
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virtual nsresult Stop(SourceMediaStream*, TrackID);
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virtual void SetDirectListeners(bool aHasDirectListeners) {};
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virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
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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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TrackTicks &aLastEndTime);
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virtual bool IsFake() {
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return false;
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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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static const unsigned int KMaxDeviceNameLength = 128;
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static const unsigned int KMaxUniqueIdLength = 256;
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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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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(nsTArray<nsRefPtr<MediaEngineVideoSource> >*);
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virtual void EnumerateAudioDevices(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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// XXX
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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* mVideoEngine;
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webrtc::VoiceEngine* mVoiceEngine;
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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 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, MediaEngineWebRTCVideoSource > 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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