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