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// Copyright 2007-2021 The Mumble Developers. All rights reserved.
// Use of this source code is governed by a BSD-style license
// that can be found in the LICENSE file at the root of the
// Mumble source tree or at <https://www.mumble.info/LICENSE>.
#ifndef MUMBLE_MUMBLE_AUDIOINPUT_H_
#define MUMBLE_MUMBLE_AUDIOINPUT_H_
#include <QtCore/QObject>
#include <QtCore/QThread>
#include <boost/array.hpp>
#include <boost/shared_ptr.hpp>
#include <fstream>
#include <list>
#include <mutex>
#include <speex/speex.h>
#include <speex/speex_echo.h>
#include <speex/speex_preprocess.h>
#include <speex/speex_resampler.h>
#include <vector>
#include "Audio.h"
#include "EchoCancelOption.h"
#include "Message.h"
#include "Settings.h"
#include "Timer.h"
class AudioInput;
class CELTCodec;
class OpusCodec;
struct CELTEncoder;
struct OpusEncoder;
struct DenoiseState;
typedef boost::shared_ptr< AudioInput > AudioInputPtr;
/**
* A chunk of audio data to process
* This struct wraps pointers to two dynamically allocated arrays, containing
* PCM samples of microphone and speaker readback data (for echo cancellation).
* Does not handle pointer ownership, so you'll have to deallocate them yourself.
*/
struct AudioChunk {
AudioChunk() : mic(nullptr), speaker(nullptr) {}
explicit AudioChunk(short *mic) : mic(mic), speaker(nullptr) {}
AudioChunk(short *mic, short *speaker) : mic(mic), speaker(speaker) {}
bool empty() const { return mic == nullptr; }
short *mic; ///< Pointer to microphone samples
short *speaker; ///< Pointer to speaker samples, nullptr if echo cancellation is disabled
};
/*
* According to https://www.speex.org/docs/manual/speex-manual/node7.html
* "It is important that, at any time, any echo that is present in the input
* has already been sent to the echo canceller as echo_frame."
* Thus, we artificially introduce a small lag in the microphone by means of
* a queue, so as to be sure the speaker data always precedes the microphone.
*
* There are conflicting requirements for the queue:
* - it has to be small enough not to cause a noticeable lag in the voice
* - it has to be large enough not to force us to drop packets frequently
* when the addMic() and addEcho() callbacks are called in a jittery way
* - its fill level must be controlled so it does not operate towards zero
* elements size, as this would not provide the lag required for the
* echo canceller to work properly.
*
* The current implementation uses a 5 elements queue, with a control
* statemachine that introduces packet drops to control the fill level
* to at least 2 (plus or minus one) and less than 4 elements.
* With a 10ms chunk, this queue should introduce a ~20ms lag to the voice.
*/
class Resynchronizer {
public:
/**
* Add a microphone sample to the resynchronizer queue
* The resynchronizer may decide to drop the sample, and in that case
* the pointer will be deallocated not lo leak memory
*
* \param mic pointer to a dynamically allocated array with PCM data
*/
void addMic(short *mic);
/**
* Add a speaker sample to the resynchronizer
* The resynchronizer may decide to drop the sample, and in that case
* the pointer will be deallocated not lo leak memory
*
* \param mic pointer to a dynamically allocated array with PCM data
* \return If microphone data is available, the resynchronizer will return a
* valid audio chunk to encode, otherwise an empty chunk will be returned
*/
AudioChunk addSpeaker(short *speaker);
/**
* Reinitialize the resynchronizer, emptying the queue in the process.
*/
void reset();
/**
* \return the nominal lag that the resynchronizer tries to enforce on the
* microphone data, in order to make sure the speaker data is always passed
* first to the echo canceller
*/
int getNominalLag() const { return 2; }
~Resynchronizer();
bool bDebugPrintQueue = false; ///< Enables printing queue fill level stats
private:
/**
* Print queue level stats for debugging purposes
* \param mic used to distinguish between addMic() and addSpeaker()
*/
void printQueue(char who);
// TODO: there was a mutex (qmEcho), but can the callbacks be called concurrently?
mutable std::mutex m;
std::list< short * > micQueue; ///< Queue of microphone samples
enum { S0, S1a, S1b, S2, S3, S4a, S4b, S5 } state = S0; ///< Queue fill control statemachine
};
class AudioInputRegistrar {
private:
Q_DISABLE_COPY(AudioInputRegistrar)
public:
static QMap< QString, AudioInputRegistrar * > *qmNew;
static QString current;
static AudioInputPtr newFromChoice(QString choice = QString());
const QString name;
int priority;
/// A list of echo cancellation options available for this backend.
std::vector< EchoCancelOptionID > echoOptions;
AudioInputRegistrar(const QString &n, int priority = 0);
virtual ~AudioInputRegistrar();
virtual AudioInput *create() = 0;
virtual const QList< audioDevice > getDeviceChoices() = 0;
virtual void setDeviceChoice(const QVariant &, Settings &) = 0;
/// Check that given combination of echoOption and outputSystem combination is suitable for echo cancellation
virtual bool canEcho(EchoCancelOptionID echoOptionId, const QString &outputSystem) const = 0;
virtual bool canExclusive() const;
/**
* Check if Mumble's microphone access has been denied by the OS.
* Both Windows and macOS have builtin privacy safeguards that display a message asking for users'
* consent when apps are trying to use the microphone, and/or provide ways to deny the microphone
* access of some apps.
* This function should check if Mumble has the permission to use the microphone.
* Note: It is possible that this result could only be known after trying to initialize the audio backend.
* Generally, call this function after attempts to initialize the AudioInput have been made.
* @return true if microphone access is denied.
*/
virtual bool isMicrophoneAccessDeniedByOS() = 0;
};
class AudioInput : public QThread {
friend class AudioNoiseWidget;
friend class AudioEchoWidget;
friend class AudioStats;
friend class AudioInputDialog;
private:
Q_OBJECT
Q_DISABLE_COPY(AudioInput)
protected:
typedef enum { CodecCELT, CodecSpeex } CodecFormat;
typedef enum { SampleShort, SampleFloat } SampleFormat;
typedef void (*inMixerFunc)(float *RESTRICT, const void *RESTRICT, unsigned int, unsigned int, quint64);
private:
bool bDebugDumpInput; ///< When true, dump pcm data to debug the echo canceller
std::ofstream outMic, outSpeaker, outProcessed; ///< Files to dump raw pcm data
SpeexResamplerState *srsMic, *srsEcho;
unsigned int iMicFilled, iEchoFilled;
inMixerFunc imfMic, imfEcho;
inMixerFunc chooseMixer(const unsigned int nchan, SampleFormat sf, quint64 mask);
void resetAudioProcessor();
OpusCodec *oCodec;
OpusEncoder *opusState;
DenoiseState *denoiseState;
bool selectCodec();
void selectNoiseCancel();
typedef boost::array< unsigned char, 960 > EncodingOutputBuffer;
int encodeOpusFrame(short *source, int size, EncodingOutputBuffer &buffer);
int encodeCELTFrame(short *pSource, EncodingOutputBuffer &buffer);
protected:
MessageHandler::UDPMessageType umtType;
SampleFormat eMicFormat, eEchoFormat;
unsigned int iMicChannels, iEchoChannels;
unsigned int iMicFreq, iEchoFreq;
unsigned int iMicLength, iEchoLength;
unsigned int iMicSampleSize, iEchoSampleSize;
int iEchoMCLength, iEchoFrameSize;
quint64 uiMicChannelMask, uiEchoChannelMask;
bool bEchoMulti;
Settings::NoiseCancel noiseCancel;
static const unsigned int iSampleRate = SAMPLE_RATE;
static const int iFrameSize = SAMPLE_RATE / 100;
QMutex qmSpeex;
SpeexPreprocessState *sppPreprocess;
SpeexEchoState *sesEcho;
CELTCodec *cCodec;
CELTEncoder *ceEncoder;
/// bResetEncoder is a flag that notifies
/// our encoder functions that the encoder
/// needs to be reset.
bool bResetEncoder;
/// Encoded audio rate in bit/s
int iAudioQuality;
bool bAllowLowDelay;
/// Number of 10ms audio "frames" per packet (!= frames in packet)
int iAudioFrames;
float *pfMicInput;
float *pfEchoInput;
Resynchronizer resync;
std::vector< short > opusBuffer;
void encodeAudioFrame(AudioChunk chunk);
void addMic(const void *data, unsigned int nsamp);
void addEcho(const void *data, unsigned int nsamp);
volatile bool bRunning;
volatile bool bPreviousVoice;
int iFrameCounter;
int iSilentFrames;
int iHoldFrames;
int iBufferedFrames;
QList< QByteArray > qlFrames;
void flushCheck(const QByteArray &, bool terminator, int voiceTargetID);
void initializeMixer();
static void adjustBandwidth(int bitspersec, int &bitrate, int &frames, bool &allowLowDelay);
signals:
void doDeaf();
void doMute();
public:
typedef enum { ActivityStateIdle, ActivityStateReturnedFromIdle, ActivityStateActive } ActivityState;
ActivityState activityState;
bool bResetProcessor;
Timer tIdle;
int iBitrate;
float dPeakSpeaker, dPeakSignal, dMaxMic, dPeakMic, dPeakCleanMic;
float fSpeechProb;
static int getNetworkBandwidth(int bitrate, int frames);
static void setMaxBandwidth(int bitspersec);
/// Construct an AudioInput.
///
/// This constructor is only ever called by Audio::startInput(), and is guaranteed
/// to be called on the application's main thread.
AudioInput();
/// Destroy an AudioInput.
///
/// This destructor is only ever called by Audio::stopInput() and Audio::stop(),
/// and is guaranteed to be called on the application's main thread.
~AudioInput() Q_DECL_OVERRIDE;
void run() Q_DECL_OVERRIDE = 0;
virtual bool isAlive() const;
bool isTransmitting() const;
};
#endif
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