589 lines
19 KiB
C++
589 lines
19 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 "prprf.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 "mozilla/UniquePtr.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 "cubeb/cubeb.h"
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#include "CubebUtils.h"
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#include "AudioPacketizer.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/modules/audio_processing/include/audio_processing.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 "CamerasChild.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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class MediaEngineWebRTCAudioCaptureSource : public MediaEngineAudioSource
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{
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public:
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NS_DECL_THREADSAFE_ISUPPORTS
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explicit MediaEngineWebRTCAudioCaptureSource(const char* aUuid)
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: MediaEngineAudioSource(kReleased)
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{
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}
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void GetName(nsAString& aName) override;
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void GetUUID(nsACString& aUUID) override;
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nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
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const MediaEnginePrefs& aPrefs,
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const nsString& aDeviceId,
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const nsACString& aOrigin) override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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return NS_OK;
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}
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nsresult Deallocate() override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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return NS_OK;
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}
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void Shutdown() override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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}
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nsresult Start(SourceMediaStream* aMediaStream, TrackID aId) override;
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nsresult Stop(SourceMediaStream* aMediaStream, TrackID aId) override;
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nsresult Restart(const dom::MediaTrackConstraints& aConstraints,
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const MediaEnginePrefs &aPrefs,
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const nsString& aDeviceId) override;
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void SetDirectListeners(bool aDirect) override
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{}
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void NotifyOutputData(MediaStreamGraph* aGraph,
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AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override
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{}
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void NotifyInputData(MediaStreamGraph* aGraph,
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const AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override
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{}
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void NotifyPull(MediaStreamGraph* aGraph, SourceMediaStream* aSource,
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TrackID aID, StreamTime aDesiredTime) override
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{}
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dom::MediaSourceEnum GetMediaSource() const override
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{
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return dom::MediaSourceEnum::AudioCapture;
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}
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bool IsFake() override
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{
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return false;
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}
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nsresult TakePhoto(PhotoCallback* aCallback) override
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{
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return NS_ERROR_NOT_IMPLEMENTED;
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}
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uint32_t GetBestFitnessDistance(
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const nsTArray<const dom::MediaTrackConstraintSet*>& aConstraintSets,
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const nsString& aDeviceId) override;
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protected:
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virtual ~MediaEngineWebRTCAudioCaptureSource() { Shutdown(); }
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nsCString mUUID;
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};
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// Small subset of VoEHardware
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class AudioInput
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{
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public:
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explicit AudioInput(webrtc::VoiceEngine* aVoiceEngine) : mVoiceEngine(aVoiceEngine) {};
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// Threadsafe because it's referenced from an MicrophoneSource, which can
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// had references to it on other threads.
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NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioInput)
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virtual int GetNumOfRecordingDevices(int& aDevices) = 0;
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virtual int GetRecordingDeviceName(int aIndex, char aStrNameUTF8[128],
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char aStrGuidUTF8[128]) = 0;
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virtual int GetRecordingDeviceStatus(bool& aIsAvailable) = 0;
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virtual void StartRecording(SourceMediaStream *aStream, AudioDataListener *aListener) = 0;
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virtual void StopRecording(SourceMediaStream *aStream) = 0;
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virtual int SetRecordingDevice(int aIndex) = 0;
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protected:
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// Protected destructor, to discourage deletion outside of Release():
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virtual ~AudioInput() {}
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webrtc::VoiceEngine* mVoiceEngine;
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};
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class AudioInputCubeb final : public AudioInput
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{
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public:
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explicit AudioInputCubeb(webrtc::VoiceEngine* aVoiceEngine, int aIndex = 0) :
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AudioInput(aVoiceEngine), mSelectedDevice(aIndex), mInUseCount(0)
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{
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if (!mDeviceIndexes) {
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mDeviceIndexes = new nsTArray<int>;
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mDeviceNames = new nsTArray<nsCString>;
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}
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}
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static void CleanupGlobalData()
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{
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if (mDevices) {
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// This doesn't require anything more than support for free()
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cubeb_device_collection_destroy(mDevices);
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mDevices = nullptr;
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}
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delete mDeviceIndexes;
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mDeviceIndexes = nullptr;
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delete mDeviceNames;
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mDeviceNames = nullptr;
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}
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int GetNumOfRecordingDevices(int& aDevices)
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{
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UpdateDeviceList();
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aDevices = mDeviceIndexes->Length();
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return 0;
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}
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int32_t DeviceIndex(int aIndex)
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{
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if (aIndex == -1) {
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aIndex = 0; // -1 = system default
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}
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if (aIndex >= (int) mDeviceIndexes->Length()) {
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return -1;
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}
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// Note: if the device is gone, this will be -1
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return (*mDeviceIndexes)[aIndex]; // translate to mDevices index
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}
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int GetRecordingDeviceName(int aIndex, char aStrNameUTF8[128],
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char aStrGuidUTF8[128])
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{
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int32_t devindex = DeviceIndex(aIndex);
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if (!mDevices || devindex < 0) {
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return 1;
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}
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PR_snprintf(aStrNameUTF8, 128, "%s%s", aIndex == -1 ? "default: " : "",
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mDevices->device[devindex]->friendly_name);
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aStrGuidUTF8[0] = '\0';
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return 0;
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}
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int GetRecordingDeviceStatus(bool& aIsAvailable)
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{
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// With cubeb, we only expose devices of type CUBEB_DEVICE_TYPE_INPUT,
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// so unless it was removed, say it's available
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aIsAvailable = true;
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return 0;
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}
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void StartRecording(SourceMediaStream *aStream, AudioDataListener *aListener)
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{
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MOZ_ASSERT(mDevices);
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if (mInUseCount == 0) {
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ScopedCustomReleasePtr<webrtc::VoEExternalMedia> ptrVoERender;
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ptrVoERender = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
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if (ptrVoERender) {
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ptrVoERender->SetExternalRecordingStatus(true);
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}
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mAnyInUse = true;
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}
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mInUseCount++;
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// Always tell the stream we're using it for input
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aStream->OpenAudioInput(mDevices->device[mSelectedDevice]->devid, aListener);
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}
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void StopRecording(SourceMediaStream *aStream)
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{
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aStream->CloseAudioInput();
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if (--mInUseCount == 0) {
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mAnyInUse = false;
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}
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}
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int SetRecordingDevice(int aIndex)
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{
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int32_t devindex = DeviceIndex(aIndex);
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if (!mDevices || devindex < 0) {
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return 1;
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}
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mSelectedDevice = devindex;
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return 0;
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}
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protected:
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~AudioInputCubeb() {
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MOZ_RELEASE_ASSERT(mInUseCount == 0);
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}
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private:
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// It would be better to watch for device-change notifications
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void UpdateDeviceList()
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{
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cubeb_device_collection *devices = nullptr;
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if (CUBEB_OK != cubeb_enumerate_devices(CubebUtils::GetCubebContext(),
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CUBEB_DEVICE_TYPE_INPUT,
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&devices)) {
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return;
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}
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for (auto& device_index : (*mDeviceIndexes)) {
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device_index = -1; // unmapped
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}
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// We keep all the device names, but wipe the mappings and rebuild them
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// Calculate translation from existing mDevices to new devices. Note we
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// never end up with less devices than before, since people have
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// stashed indexes.
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// For some reason the "fake" device for automation is marked as DISABLED,
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// so white-list it.
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for (uint32_t i = 0; i < devices->count; i++) {
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if (devices->device[i]->type == CUBEB_DEVICE_TYPE_INPUT && // paranoia
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(devices->device[i]->state == CUBEB_DEVICE_STATE_ENABLED ||
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devices->device[i]->state == CUBEB_DEVICE_STATE_UNPLUGGED ||
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(devices->device[i]->state == CUBEB_DEVICE_STATE_DISABLED &&
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strcmp(devices->device[i]->friendly_name, "Sine source at 440 Hz") == 0)))
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{
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auto j = mDeviceNames->IndexOf(devices->device[i]->device_id);
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if (j != nsTArray<nsCString>::NoIndex) {
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// match! update the mapping
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(*mDeviceIndexes)[j] = i;
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} else {
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// new device, add to the array
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mDeviceIndexes->AppendElement(i);
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mDeviceNames->AppendElement(devices->device[i]->device_id);
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}
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}
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}
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// swap state
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if (mDevices) {
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cubeb_device_collection_destroy(mDevices);
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}
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mDevices = devices;
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}
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// We have an array, which consists of indexes to the current mDevices
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// list. This is updated on mDevices updates. Many devices in mDevices
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// won't be included in the array (wrong type, etc), or if a device is
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// removed it will map to -1 (and opens of this device will need to check
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// for this - and be careful of threading access. The mappings need to
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// updated on each re-enumeration.
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int mSelectedDevice;
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uint32_t mInUseCount;
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// pointers to avoid static constructors
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static nsTArray<int>* mDeviceIndexes;
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static nsTArray<nsCString>* mDeviceNames;
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static cubeb_device_collection *mDevices;
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static bool mAnyInUse;
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};
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class AudioInputWebRTC final : public AudioInput
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{
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public:
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explicit AudioInputWebRTC(webrtc::VoiceEngine* aVoiceEngine) : AudioInput(aVoiceEngine) {}
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int GetNumOfRecordingDevices(int& aDevices)
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{
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ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
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ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
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if (!ptrVoEHw) {
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return 1;
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}
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return ptrVoEHw->GetNumOfRecordingDevices(aDevices);
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}
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int GetRecordingDeviceName(int aIndex, char aStrNameUTF8[128],
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char aStrGuidUTF8[128])
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{
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ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
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ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
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if (!ptrVoEHw) {
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return 1;
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}
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return ptrVoEHw->GetRecordingDeviceName(aIndex, aStrNameUTF8,
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aStrGuidUTF8);
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}
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int GetRecordingDeviceStatus(bool& aIsAvailable)
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{
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ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
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ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
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if (!ptrVoEHw) {
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return 1;
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}
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ptrVoEHw->GetRecordingDeviceStatus(aIsAvailable);
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return 0;
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}
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void StartRecording(SourceMediaStream *aStream, AudioDataListener *aListener) {}
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void StopRecording(SourceMediaStream *aStream) {}
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int SetRecordingDevice(int aIndex)
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{
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ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
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ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
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if (!ptrVoEHw) {
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return 1;
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}
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return ptrVoEHw->SetRecordingDevice(aIndex);
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}
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protected:
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// Protected destructor, to discourage deletion outside of Release():
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~AudioInputWebRTC() {}
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};
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class WebRTCAudioDataListener : public AudioDataListener
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{
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protected:
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// Protected destructor, to discourage deletion outside of Release():
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virtual ~WebRTCAudioDataListener() {}
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public:
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explicit WebRTCAudioDataListener(MediaEngineAudioSource* aAudioSource) :
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mAudioSource(aAudioSource)
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{}
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// AudioDataListenerInterface methods
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virtual void NotifyOutputData(MediaStreamGraph* aGraph,
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AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override
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{
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mAudioSource->NotifyOutputData(aGraph, aBuffer, aFrames, aRate, aChannels);
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}
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virtual void NotifyInputData(MediaStreamGraph* aGraph,
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const AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override
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{
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mAudioSource->NotifyInputData(aGraph, aBuffer, aFrames, aRate, aChannels);
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}
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private:
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RefPtr<MediaEngineAudioSource> mAudioSource;
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};
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class MediaEngineWebRTCMicrophoneSource : public MediaEngineAudioSource,
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public webrtc::VoEMediaProcess,
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private MediaConstraintsHelper
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{
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public:
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MediaEngineWebRTCMicrophoneSource(nsIThread* aThread,
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webrtc::VoiceEngine* aVoiceEnginePtr,
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mozilla::AudioInput* aAudioInput,
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int aIndex,
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const char* name,
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const char* uuid)
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: MediaEngineAudioSource(kReleased)
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, mVoiceEngine(aVoiceEnginePtr)
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, mAudioInput(aAudioInput)
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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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, mNrAllocations(0)
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, mInitDone(false)
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, mStarted(false)
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, mSampleFrequency(MediaEngine::DEFAULT_SAMPLE_RATE)
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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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, mInputBufferLen(0) {
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MOZ_ASSERT(aVoiceEnginePtr);
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MOZ_ASSERT(aAudioInput);
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mDeviceName.Assign(NS_ConvertUTF8toUTF16(name));
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mDeviceUUID.Assign(uuid);
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mListener = new mozilla::WebRTCAudioDataListener(this);
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Init();
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}
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void GetName(nsAString& aName) override;
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void GetUUID(nsACString& aUUID) override;
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nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
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const MediaEnginePrefs& aPrefs,
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const nsString& aDeviceId,
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const nsACString& aOrigin) override;
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nsresult Deallocate() override;
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nsresult Start(SourceMediaStream* aStream, TrackID aID) override;
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nsresult Stop(SourceMediaStream* aSource, TrackID aID) override;
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nsresult Restart(const dom::MediaTrackConstraints& aConstraints,
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const MediaEnginePrefs &aPrefs,
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const nsString& aDeviceId) override;
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void SetDirectListeners(bool aHasDirectListeners) override {};
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void NotifyPull(MediaStreamGraph* aGraph,
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SourceMediaStream* aSource,
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TrackID aId,
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StreamTime aDesiredTime) override;
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// AudioDataListenerInterface methods
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void NotifyOutputData(MediaStreamGraph* aGraph,
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AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override;
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void NotifyInputData(MediaStreamGraph* aGraph,
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const AudioDataValue* aBuffer, size_t aFrames,
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TrackRate aRate, uint32_t aChannels) override;
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bool IsFake() override {
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return false;
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}
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dom::MediaSourceEnum GetMediaSource() const override {
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return dom::MediaSourceEnum::Microphone;
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}
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nsresult TakePhoto(PhotoCallback* aCallback) override
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{
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return NS_ERROR_NOT_IMPLEMENTED;
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}
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uint32_t GetBestFitnessDistance(
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const nsTArray<const dom::MediaTrackConstraintSet*>& aConstraintSets,
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const nsString& aDeviceId) override;
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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) override;
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void Shutdown() override;
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NS_DECL_THREADSAFE_ISUPPORTS
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protected:
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~MediaEngineWebRTCMicrophoneSource() { Shutdown(); }
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private:
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void Init();
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|
webrtc::VoiceEngine* mVoiceEngine;
|
|
RefPtr<mozilla::AudioInput> mAudioInput;
|
|
RefPtr<WebRTCAudioDataListener> mListener;
|
|
|
|
ScopedCustomReleasePtr<webrtc::VoEBase> mVoEBase;
|
|
ScopedCustomReleasePtr<webrtc::VoEExternalMedia> mVoERender;
|
|
ScopedCustomReleasePtr<webrtc::VoENetwork> mVoENetwork;
|
|
ScopedCustomReleasePtr<webrtc::VoEAudioProcessing> mVoEProcessing;
|
|
|
|
nsAutoPtr<AudioPacketizer<AudioDataValue, int16_t>> mPacketizer;
|
|
|
|
// 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<RefPtr<SourceMediaStream>> mSources;
|
|
nsCOMPtr<nsIThread> mThread;
|
|
int mCapIndex;
|
|
int mChannel;
|
|
int mNrAllocations; // When this becomes 0, we shut down HW
|
|
TrackID mTrackID;
|
|
bool mInitDone;
|
|
bool mStarted;
|
|
|
|
nsString mDeviceName;
|
|
nsCString mDeviceUUID;
|
|
|
|
uint32_t mSampleFrequency;
|
|
bool mEchoOn, mAgcOn, mNoiseOn;
|
|
webrtc::EcModes mEchoCancel;
|
|
webrtc::AgcModes mAGC;
|
|
webrtc::NsModes mNoiseSuppress;
|
|
int32_t mPlayoutDelay;
|
|
|
|
NullTransport *mNullTransport;
|
|
|
|
// For full_duplex packetizer output
|
|
size_t mInputBufferLen;
|
|
UniquePtr<int16_t[]> mInputBuffer;
|
|
};
|
|
|
|
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;
|
|
|
|
void EnumerateVideoDevices(dom::MediaSourceEnum,
|
|
nsTArray<RefPtr<MediaEngineVideoSource>>*) override;
|
|
void EnumerateAudioDevices(dom::MediaSourceEnum,
|
|
nsTArray<RefPtr<MediaEngineAudioSource>>*) override;
|
|
private:
|
|
~MediaEngineWebRTC() {
|
|
Shutdown();
|
|
#if defined(MOZ_B2G_CAMERA) && defined(MOZ_WIDGET_GONK)
|
|
AsyncLatencyLogger::Get()->Release();
|
|
#endif
|
|
gFarendObserver = nullptr;
|
|
}
|
|
|
|
nsCOMPtr<nsIThread> mThread;
|
|
|
|
// gUM runnables can e.g. Enumerate from multiple threads
|
|
Mutex mMutex;
|
|
webrtc::VoiceEngine* mVoiceEngine;
|
|
webrtc::Config mConfig;
|
|
RefPtr<mozilla::AudioInput> mAudioInput;
|
|
bool mAudioEngineInit;
|
|
bool mFullDuplex;
|
|
bool mExtendedFilter;
|
|
bool mDelayAgnostic;
|
|
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, MediaEngineAudioSource> mAudioSources;
|
|
};
|
|
|
|
}
|
|
|
|
#endif /* NSMEDIAENGINEWEBRTC_H_ */
|