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/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* 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 MediaEngineWebRTCAudio_h
#define MediaEngineWebRTCAudio_h
#include "AudioPacketizer.h"
#include "AudioSegment.h"
#include "AudioDeviceInfo.h"
#include "DeviceInputTrack.h"
#include "MediaEngineWebRTC.h"
#include "MediaEnginePrefs.h"
#include "MediaTrackListener.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace mozilla {
class AudioInputProcessing;
class AudioProcessingTrack;
// This class is created and used exclusively on the Media Manager thread, with
// exactly two exceptions:
// - Pull is always called on the MTG thread. It only ever uses
// mInputProcessing. mInputProcessing is set, then a message is sent first to
// the main thread and then the MTG thread so that it can be used as part of
// the graph processing. On destruction, similarly, a message is sent to the
// graph so that it stops using it, and then it is deleted.
// - mSettings is created on the MediaManager thread is always ever accessed on
// the Main Thread. It is const.
class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
public:
explicit MediaEngineWebRTCMicrophoneSource(const MediaDevice* aMediaDevice);
nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
const char** aOutBadConstraint) override;
nsresult Deallocate() override;
void SetTrack(const RefPtr<MediaTrack>& aTrack,
const PrincipalHandle& aPrincipal) override;
nsresult Start() override;
nsresult Stop() override;
nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs,
const char** aOutBadConstraint) override;
/**
* Assigns the current settings of the capture to aOutSettings.
* Main thread only.
*/
void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
return NS_ERROR_NOT_IMPLEMENTED;
}
protected:
~MediaEngineWebRTCMicrophoneSource() = default;
private:
/**
* From a set of constraints and about:config preferences, output the correct
* set of preferences that can be sent to AudioInputProcessing.
*
* This can fail if the number of channels requested is zero, negative, or
* more than the device supports.
*/
nsresult EvaluateSettings(const NormalizedConstraints& aConstraintsUpdate,
const MediaEnginePrefs& aInPrefs,
MediaEnginePrefs* aOutPrefs,
const char** aOutBadConstraint);
/**
* From settings output by EvaluateSettings, send those settings to the
* AudioInputProcessing instance and the main thread (for use in GetSettings).
*/
void ApplySettings(const MediaEnginePrefs& aPrefs);
PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
const RefPtr<AudioDeviceInfo> mDeviceInfo;
// The maximum number of channels that this device supports.
const uint32_t mDeviceMaxChannelCount;
// The current settings for the underlying device.
// Constructed on the MediaManager thread, and then only ever accessed on the
// main thread.
const nsMainThreadPtrHandle<media::Refcountable<dom::MediaTrackSettings>>
mSettings;
// Current state of the resource for this source.
MediaEngineSourceState mState;
// The current preferences that will be forwarded to mAudioProcessingConfig
// below.
MediaEnginePrefs mCurrentPrefs;
// The AudioProcessingTrack used to inteface with the MediaTrackGraph. Set in
// SetTrack as part of the initialization, and nulled in ::Deallocate.
RefPtr<AudioProcessingTrack> mTrack;
// See note at the top of this class.
RefPtr<AudioInputProcessing> mInputProcessing;
// Copy of the config currently applied to AudioProcessing through
// mInputProcessing.
webrtc::AudioProcessing::Config mAudioProcessingConfig;
};
// This class is created on the MediaManager thread, and then exclusively used
// on the MTG thread.
// All communication is done via message passing using MTG ControlMessages
class AudioInputProcessing : public AudioDataListener {
public:
explicit AudioInputProcessing(uint32_t aMaxChannelCount);
void Process(MediaTrackGraph* aGraph, GraphTime aFrom, GraphTime aTo,
AudioSegment* aInput, AudioSegment* aOutput);
void ProcessOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
bool IsVoiceInput(MediaTrackGraph* aGraph) const override {
// If we're passing data directly without AEC or any other process, this
// means that all voice-processing has been disabled intentionaly. In this
// case, consider that the device is not used for voice input.
return !PassThrough(aGraph);
}
void Start(MediaTrackGraph* aGraph);
void Stop(MediaTrackGraph* aGraph);
void DeviceChanged(MediaTrackGraph* aGraph) override;
uint32_t RequestedInputChannelCount(MediaTrackGraph*) override {
return GetRequestedInputChannelCount();
}
void Disconnect(MediaTrackGraph* aGraph) override;
void PacketizeAndProcess(MediaTrackGraph* aGraph,
const AudioSegment& aSegment);
void SetPassThrough(MediaTrackGraph* aGraph, bool aPassThrough);
uint32_t GetRequestedInputChannelCount();
void SetRequestedInputChannelCount(MediaTrackGraph* aGraph,
CubebUtils::AudioDeviceID aDeviceId,
uint32_t aRequestedInputChannelCount);
// This is true when all processing is disabled, we can skip
// packetization, resampling and other processing passes.
bool PassThrough(MediaTrackGraph* aGraph) const;
// This allow changing the APM options, enabling or disabling processing
// steps. The config gets applied the next time we're about to process input
// data.
void ApplyConfig(MediaTrackGraph* aGraph,
const webrtc::AudioProcessing::Config& aConfig);
void End();
TrackTime NumBufferedFrames(MediaTrackGraph* aGraph) const;
// The packet size contains samples in 10ms. The unit of aRate is hz.
static uint32_t GetPacketSize(TrackRate aRate) {
return webrtc::AudioProcessing::GetFrameSize(aRate);
}
bool IsEnded() const { return mEnded; }
private:
~AudioInputProcessing() = default;
void EnsureAudioProcessing(MediaTrackGraph* aGraph, uint32_t aChannels);
void ResetAudioProcessing(MediaTrackGraph* aGraph);
PrincipalHandle GetCheckedPrincipal(const AudioSegment& aSegment);
// This implements the processing algoritm to apply to the input (e.g. a
// microphone). If all algorithms are disabled, this class in not used. This
// class only accepts audio chunks of 10ms. It has two inputs and one output:
// it is fed the speaker data and the microphone data. It outputs processed
// input data.
const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
// Packetizer to be able to feed 10ms packets to the input side of
// mAudioProcessing. Not used if the processing is bypassed.
Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
// The number of channels asked for by content, after clamping to the range of
// legal channel count for this particular device.
uint32_t mRequestedInputChannelCount;
// mSkipProcessing is true if none of the processing passes are enabled,
// because of prefs or constraints. This allows simply copying the audio into
// the MTG, skipping resampling and the whole webrtc.org code.
bool mSkipProcessing;
// Buffer for up to one 10ms packet of planar mixed audio output for the
// reverse-stream (speaker data) of mAudioProcessing AEC.
// Length is packet size * channel count, regardless of how many frames are
// buffered. Not used if the processing is bypassed.
AlignedFloatBuffer mOutputBuffer;
// Number of channels into which mOutputBuffer is divided.
uint32_t mOutputBufferChannelCount = 0;
// Number of frames buffered in mOutputBuffer for the reverse stream.
uint32_t mOutputBufferFrameCount = 0;
// Stores the input audio, to be processed by the APM.
AlignedFloatBuffer mInputBuffer;
// Stores the deinterleaved microphone audio
AlignedFloatBuffer mDeinterleavedBuffer;
// Stores the mixed down input audio
AlignedFloatBuffer mInputDownmixBuffer;
// Stores data waiting to be pulled.
AudioSegment mSegment;
// Whether or not this MediaEngine is enabled. If it's not enabled, it
// operates in "pull" mode, and we append silence only, releasing the audio
// input track.
bool mEnabled;
// Whether or not we've ended and removed the AudioProcessingTrack.
bool mEnded;
// When processing is enabled, the number of packets received by this
// instance, to implement periodic logging.
uint64_t mPacketCount;
// Temporary descriptor for a slice of an AudioChunk parameter passed to
// ProcessOutputData(). This is a member rather than on the stack so that
// any memory allocated for its mChannelData pointer array is not
// reallocated on each iteration.
AudioChunk mSubChunk;
// A storage holding the interleaved audio data converted the AudioSegment.
// This will be used as an input parameter for PacketizeAndProcess. This
// should be removed once bug 1729041 is done.
AutoTArray<AudioDataValue,
SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
mInterleavedBuffer;
// Tracks the pending frames with paired principals piled up in packetizer.
std::deque<std::pair<TrackTime, PrincipalHandle>> mChunksInPacketizer;
};
// MediaTrack subclass tailored for MediaEngineWebRTCMicrophoneSource.
class AudioProcessingTrack : public DeviceInputConsumerTrack {
// Only accessed on the graph thread.
RefPtr<AudioInputProcessing> mInputProcessing;
explicit AudioProcessingTrack(TrackRate aSampleRate)
: DeviceInputConsumerTrack(aSampleRate) {}
~AudioProcessingTrack() = default;
public:
// Main Thread API
void Destroy() override;
void SetInputProcessing(RefPtr<AudioInputProcessing> aInputProcessing);
static AudioProcessingTrack* Create(MediaTrackGraph* aGraph);
// Graph Thread API
void DestroyImpl() override;
void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
uint32_t NumberOfChannels() const override {
MOZ_DIAGNOSTIC_ASSERT(
mInputProcessing,
"Must set mInputProcessing before exposing to content");
return mInputProcessing->GetRequestedInputChannelCount();
}
// Pass the graph's mixed audio output to mInputProcessing for processing as
// the reverse stream.
void NotifyOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
// Any thread
AudioProcessingTrack* AsAudioProcessingTrack() override { return this; }
private:
// Graph thread API
void SetInputProcessingImpl(RefPtr<AudioInputProcessing> aInputProcessing);
};
class MediaEngineWebRTCAudioCaptureSource : public MediaEngineSource {
public:
explicit MediaEngineWebRTCAudioCaptureSource(const MediaDevice* aMediaDevice);
static nsString GetUUID();
static nsString GetGroupId();
nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
const char** aOutBadConstraint) override {
// Nothing to do here, everything is managed in MediaManager.cpp
return NS_OK;
}
nsresult Deallocate() override {
// Nothing to do here, everything is managed in MediaManager.cpp
return NS_OK;
}
void SetTrack(const RefPtr<MediaTrack>& aTrack,
const PrincipalHandle& aPrincipal) override;
nsresult Start() override;
nsresult Stop() override;
nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
const MediaEnginePrefs& aPrefs,
const char** aOutBadConstraint) override;
nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
return NS_ERROR_NOT_IMPLEMENTED;
}
void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
protected:
virtual ~MediaEngineWebRTCAudioCaptureSource() = default;
};
} // end namespace mozilla
#endif // MediaEngineWebRTCAudio_h