/
AudioInput.cpp
1206 lines (1013 loc) · 32.3 KB
/
AudioInput.cpp
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// Copyright 2005-2020 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>.
#include "AudioInput.h"
#include "AudioOutput.h"
#include "CELTCodec.h"
#ifdef USE_OPUS
# include "OpusCodec.h"
#endif
#include "ServerHandler.h"
#include "MainWindow.h"
#include "User.h"
#include "PacketDataStream.h"
#include "Plugins.h"
#include "Message.h"
#include "Global.h"
#include "NetworkConfig.h"
#include "Utils.h"
#include "VoiceRecorder.h"
#ifdef USE_RNNOISE
extern "C" {
#include "rnnoise.h"
}
#endif
void Resynchronizer::addMic(short *mic)
{
bool drop = false;
{
std::unique_lock<std::mutex> l(m);
micQueue.push_back(mic);
switch(state)
{
case S0: state = S1a; break;
case S1a: state = S2; break;
case S1b: state = S2; break;
case S2: state = S3; break;
case S3: state = S4a; break;
case S4a: state = S5; break;
case S4b: drop = true; break;
case S5: drop = true; break;
}
if(drop)
{
delete[] micQueue.front();
micQueue.pop_front();
}
}
if(bDebugPrintQueue)
{
if(drop) printf("dropped microphone chunk due to overflow\n");
printQueue('+');
}
}
AudioChunk Resynchronizer::addSpeaker(short *speaker)
{
AudioChunk result;
bool drop = false;
{
std::unique_lock<std::mutex> l(m);
switch(state)
{
case S0: drop = true; break;
case S1a: drop = true; break;
case S1b: state = S0; break;
case S2: state = S1b; break;
case S3: state = S2; break;
case S4a: state = S3; break;
case S4b: state = S3; break;
case S5: state = S4b; break;
}
if(drop == false)
{
result = AudioChunk(micQueue.front(), speaker);
micQueue.pop_front();
}
}
if(drop) delete[] speaker;
if(bDebugPrintQueue)
{
if(drop) printf("dropped speaker chunk due to underflow\n");
printQueue('-');
}
return result;
}
void Resynchronizer::reset()
{
if(bDebugPrintQueue) printf("resetting echo queue\n");
std::unique_lock<std::mutex> l(m);
state = S0;
while(micQueue.empty() == false)
{
delete[] micQueue.front();
micQueue.pop_front();
}
}
Resynchronizer::~Resynchronizer()
{
reset();
}
void Resynchronizer::printQueue(char who)
{
unsigned int mic;
{
std::unique_lock<std::mutex> l(m);
mic = static_cast<unsigned int>(micQueue.size());
}
std::string line;
line.reserve(32);
line += who;
line += " Echo queue [";
for(unsigned int i = 0; i < 5; i++) line += i < mic ? '#' : ' ';
line += "]\r";
printf("%s",line.c_str());
fflush(stdout);
}
// Remember that we cannot use static member classes that are not pointers, as the constructor
// for AudioInputRegistrar() might be called before they are initialized, as the constructor
// is called from global initialization.
// Hence, we allocate upon first call.
QMap<QString, AudioInputRegistrar *> *AudioInputRegistrar::qmNew;
QString AudioInputRegistrar::current = QString();
AudioInputRegistrar::AudioInputRegistrar(const QString &n, int p) : name(n), priority(p) {
if (! qmNew)
qmNew = new QMap<QString, AudioInputRegistrar *>();
qmNew->insert(name,this);
}
AudioInputRegistrar::~AudioInputRegistrar() {
qmNew->remove(name);
}
AudioInputPtr AudioInputRegistrar::newFromChoice(QString choice) {
if (! qmNew)
return AudioInputPtr();
if (!choice.isEmpty() && qmNew->contains(choice)) {
g.s.qsAudioInput = choice;
current = choice;
return AudioInputPtr(qmNew->value(current)->create());
}
choice = g.s.qsAudioInput;
if (qmNew->contains(choice)) {
current = choice;
return AudioInputPtr(qmNew->value(choice)->create());
}
AudioInputRegistrar *r = NULL;
foreach(AudioInputRegistrar *air, *qmNew)
if (!r || (air->priority > r->priority))
r = air;
if (r) {
current = r->name;
return AudioInputPtr(r->create());
}
return AudioInputPtr();
}
bool AudioInputRegistrar::canExclusive() const {
return false;
}
AudioInput::AudioInput() : opusBuffer(g.s.iFramesPerPacket * (SAMPLE_RATE / 100)) {
bDebugDumpInput = g.bDebugDumpInput;
resync.bDebugPrintQueue = g.bDebugPrintQueue;
if(bDebugDumpInput)
{
outMic.open("raw_microphone_dump", std::ios::binary);
outSpeaker.open("speaker_dump", std::ios::binary);
outProcessed.open("processed_microphone_dump", std::ios::binary);
}
adjustBandwidth(g.iMaxBandwidth, iAudioQuality, iAudioFrames, bAllowLowDelay);
g.iAudioBandwidth = getNetworkBandwidth(iAudioQuality, iAudioFrames);
umtType = MessageHandler::UDPVoiceCELTAlpha;
activityState = ActivityStateActive;
oCodec = NULL;
opusState = NULL;
cCodec = NULL;
ceEncoder = NULL;
#ifdef USE_OPUS
oCodec = g.oCodec;
if (oCodec) {
if (bAllowLowDelay && iAudioQuality >= 64000) { // > 64 kbit/s bitrate and low delay allowed
opusState = oCodec->opus_encoder_create(SAMPLE_RATE, 1, OPUS_APPLICATION_RESTRICTED_LOWDELAY, NULL);
qWarning("AudioInput: Opus encoder set for low delay");
} else if (iAudioQuality >= 32000) { // > 32 kbit/s bitrate
opusState = oCodec->opus_encoder_create(SAMPLE_RATE, 1, OPUS_APPLICATION_AUDIO, NULL);
qWarning("AudioInput: Opus encoder set for high quality speech");
} else {
opusState = oCodec->opus_encoder_create(SAMPLE_RATE, 1, OPUS_APPLICATION_VOIP, NULL);
qWarning("AudioInput: Opus encoder set for low quality speech");
}
oCodec->opus_encoder_ctl(opusState, OPUS_SET_VBR(0)); // CBR
}
#endif
#ifdef USE_RNNOISE
denoiseState = rnnoise_create();
#endif
qWarning("AudioInput: %d bits/s, %d hz, %d sample", iAudioQuality, iSampleRate, iFrameSize);
iEchoFreq = iMicFreq = iSampleRate;
iFrameCounter = 0;
iSilentFrames = 0;
iHoldFrames = 0;
iBufferedFrames = 0;
bResetProcessor = true;
bEchoMulti = false;
sppPreprocess = NULL;
sesEcho = NULL;
srsMic = srsEcho = NULL;
iEchoChannels = iMicChannels = 0;
iEchoFilled = iMicFilled = 0;
eMicFormat = eEchoFormat = SampleFloat;
iMicSampleSize = iEchoSampleSize = 0;
bPreviousVoice = false;
bResetEncoder = true;
pfMicInput = pfEchoInput = NULL;
iBitrate = 0;
dPeakSignal = dPeakSpeaker = dPeakMic = dPeakCleanMic = 0.0;
if (g.uiSession) {
setMaxBandwidth(g.iMaxBandwidth);
}
bRunning = true;
connect(this, SIGNAL(doDeaf()), g.mw->qaAudioDeaf, SLOT(trigger()), Qt::QueuedConnection);
connect(this, SIGNAL(doMute()), g.mw->qaAudioMute, SLOT(trigger()), Qt::QueuedConnection);
}
AudioInput::~AudioInput() {
bRunning = false;
wait();
#ifdef USE_OPUS
if (opusState) {
oCodec->opus_encoder_destroy(opusState);
}
#endif
#ifdef USE_RNNOISE
if (denoiseState) {
rnnoise_destroy(denoiseState);
}
#endif
if (ceEncoder) {
cCodec->celt_encoder_destroy(ceEncoder);
}
if (sppPreprocess)
speex_preprocess_state_destroy(sppPreprocess);
if (sesEcho)
speex_echo_state_destroy(sesEcho);
if (srsMic)
speex_resampler_destroy(srsMic);
if (srsEcho)
speex_resampler_destroy(srsEcho);
delete [] pfMicInput;
delete [] pfEchoInput;
}
bool AudioInput::isTransmitting() const {
return bPreviousVoice;
};
#define IN_MIXER_FLOAT(channels) \
static void inMixerFloat##channels ( float * RESTRICT buffer, const void * RESTRICT ipt, unsigned int nsamp, unsigned int N, quint64 mask) { \
const float * RESTRICT input = reinterpret_cast<const float *>(ipt); \
const float m = 1.0f / static_cast<float>(channels); \
Q_UNUSED(N); \
Q_UNUSED(mask); \
for(unsigned int i=0;i<nsamp;++i) {\
float v= 0.0f; \
for(unsigned int j=0;j<channels;++j) \
v += input[i*channels+j]; \
buffer[i] = v * m; \
} \
}
#define IN_MIXER_SHORT(channels) \
static void inMixerShort##channels ( float * RESTRICT buffer, const void * RESTRICT ipt, unsigned int nsamp, unsigned int N, quint64 mask) { \
const short * RESTRICT input = reinterpret_cast<const short *>(ipt); \
const float m = 1.0f / (32768.f * static_cast<float>(channels)); \
Q_UNUSED(N); \
Q_UNUSED(mask); \
for(unsigned int i=0;i<nsamp;++i) {\
float v= 0.0f; \
for(unsigned int j=0;j<channels;++j) \
v += static_cast<float>(input[i*channels+j]); \
buffer[i] = v * m; \
} \
}
static void inMixerFloatMask(float * RESTRICT buffer, const void * RESTRICT ipt, unsigned int nsamp, unsigned int N, quint64 mask) { \
const float * RESTRICT input = reinterpret_cast<const float *>(ipt);
unsigned int chancount = 0;
STACKVAR(unsigned int, chanindex, N);
for (unsigned int j = 0; j < N; ++j) {
if ((mask & (1ULL << j)) == 0) {
continue;
}
chanindex[chancount] = j; // Use chancount as index into chanindex.
++chancount;
}
const float m = 1.0f / static_cast<float>(chancount);
for(unsigned int i = 0; i < nsamp; ++i) {
float v = 0.0f;
for(unsigned int j = 0; j < chancount; ++j) {
v += input[i * N + chanindex[j]];
}
buffer[i] = v * m;
}
}
static void inMixerShortMask(float * RESTRICT buffer, const void * RESTRICT ipt, unsigned int nsamp, unsigned int N, quint64 mask) {
const short * RESTRICT input = reinterpret_cast<const short *>(ipt);
unsigned int chancount = 0;
STACKVAR(unsigned int, chanindex, N);
for (unsigned int j = 0; j < N; ++j) {
if ((mask & (1ULL << j)) == 0) {
continue;
}
chanindex[chancount] = j; // Use chancount as index into chanindex.
++chancount;
}
const float m = 1.0f / static_cast<float>(chancount);
for(unsigned int i = 0; i < nsamp; ++i) {
float v = 0.0f;
for(unsigned int j = 0; j < chancount; ++j) {
v += static_cast<float>(input[i * N + chanindex[j]]);
}
buffer[i] = v * m;
}
}
IN_MIXER_FLOAT(1)
IN_MIXER_FLOAT(2)
IN_MIXER_FLOAT(3)
IN_MIXER_FLOAT(4)
IN_MIXER_FLOAT(5)
IN_MIXER_FLOAT(6)
IN_MIXER_FLOAT(7)
IN_MIXER_FLOAT(8)
IN_MIXER_FLOAT(N)
IN_MIXER_SHORT(1)
IN_MIXER_SHORT(2)
IN_MIXER_SHORT(3)
IN_MIXER_SHORT(4)
IN_MIXER_SHORT(5)
IN_MIXER_SHORT(6)
IN_MIXER_SHORT(7)
IN_MIXER_SHORT(8)
IN_MIXER_SHORT(N)
AudioInput::inMixerFunc AudioInput::chooseMixer(const unsigned int nchan, SampleFormat sf, quint64 chanmask) {
inMixerFunc r = NULL;
if (chanmask != 0xffffffffffffffffULL) {
if (sf == SampleFloat) {
r = inMixerFloatMask;
} else if (sf == SampleShort) {
r = inMixerShortMask;
}
return r;
}
if (sf == SampleFloat) {
switch (nchan) {
case 1:
r = inMixerFloat1;
break;
case 2:
r = inMixerFloat2;
break;
case 3:
r = inMixerFloat3;
break;
case 4:
r = inMixerFloat4;
break;
case 5:
r = inMixerFloat5;
break;
case 6:
r = inMixerFloat6;
break;
case 7:
r = inMixerFloat7;
break;
case 8:
r = inMixerFloat8;
break;
default:
r = inMixerFloatN;
break;
}
} else {
switch (nchan) {
case 1:
r = inMixerShort1;
break;
case 2:
r = inMixerShort2;
break;
case 3:
r = inMixerShort3;
break;
case 4:
r = inMixerShort4;
break;
case 5:
r = inMixerShort5;
break;
case 6:
r = inMixerShort6;
break;
case 7:
r = inMixerShort7;
break;
case 8:
r = inMixerShort8;
break;
default:
r = inMixerShortN;
break;
}
}
return r;
}
void AudioInput::initializeMixer() {
int err;
if (srsMic)
speex_resampler_destroy(srsMic);
if (srsEcho)
speex_resampler_destroy(srsEcho);
delete [] pfMicInput;
delete [] pfEchoInput;
if (iMicFreq != iSampleRate)
srsMic = speex_resampler_init(1, iMicFreq, iSampleRate, 3, &err);
iMicLength = (iFrameSize * iMicFreq) / iSampleRate;
pfMicInput = new float[iMicLength];
if (iEchoChannels > 0) {
bEchoMulti = g.s.bEchoMulti;
if (iEchoFreq != iSampleRate)
srsEcho = speex_resampler_init(bEchoMulti ? iEchoChannels : 1, iEchoFreq, iSampleRate, 3, &err);
iEchoLength = (iFrameSize * iEchoFreq) / iSampleRate;
iEchoMCLength = bEchoMulti ? iEchoLength * iEchoChannels : iEchoLength;
iEchoFrameSize = bEchoMulti ? iFrameSize * iEchoChannels : iFrameSize;
pfEchoInput = new float[iEchoMCLength];
} else {
srsEcho = NULL;
pfEchoInput = NULL;
}
uiMicChannelMask = g.s.uiAudioInputChannelMask;
// There is no channel mask setting for the echo canceller, so allow all channels.
uiEchoChannelMask = 0xffffffffffffffffULL;
imfMic = chooseMixer(iMicChannels, eMicFormat, uiMicChannelMask);
imfEcho = chooseMixer(iEchoChannels, eEchoFormat, uiEchoChannelMask);
iMicSampleSize = static_cast<int>(iMicChannels * ((eMicFormat == SampleFloat) ? sizeof(float) : sizeof(short)));
iEchoSampleSize = static_cast<int>(iEchoChannels * ((eEchoFormat == SampleFloat) ? sizeof(float) : sizeof(short)));
bResetProcessor = true;
qWarning("AudioInput: Initialized mixer for %d channel %d hz mic and %d channel %d hz echo", iMicChannels, iMicFreq, iEchoChannels, iEchoFreq);
if (uiMicChannelMask != 0xffffffffffffffffULL) {
qWarning("AudioInput: using mic channel mask 0x%llx", static_cast<unsigned long long>(uiMicChannelMask));
}
}
void AudioInput::addMic(const void *data, unsigned int nsamp) {
while (nsamp > 0) {
// Make sure we don't overrun the frame buffer
const unsigned int left = qMin(nsamp, iMicLength - iMicFilled);
// Append mix into pfMicInput frame buffer (converts 16bit pcm->float if necessary)
imfMic(pfMicInput + iMicFilled, data, left, iMicChannels, uiMicChannelMask);
iMicFilled += left;
nsamp -= left;
// If new samples are left offset data pointer to point at the first one for next iteration
if (nsamp > 0) {
if (eMicFormat == SampleFloat)
data = reinterpret_cast<const float *>(data) + left * iMicChannels;
else
data = reinterpret_cast<const short *>(data) + left * iMicChannels;
}
if (iMicFilled == iMicLength) {
// Frame complete
iMicFilled = 0;
// If needed resample frame
float *pfOutput = srsMic ? (float*)alloca(iFrameSize*sizeof(float)) : nullptr;
float *ptr = srsMic ? pfOutput : pfMicInput;
if (srsMic) {
spx_uint32_t inlen = iMicLength;
spx_uint32_t outlen = iFrameSize;
speex_resampler_process_float(srsMic, 0, pfMicInput, &inlen, pfOutput, &outlen);
}
// If echo cancellation is enabled the pointer ends up in the resynchronizer queue
// and may need to outlive this function's frame
short *psMic = iEchoChannels > 0 ? new short[iFrameSize] : (short*)alloca(iFrameSize*sizeof(short));
// Convert float to 16bit PCM
const float mul = 32768.f;
for (int j = 0; j < iFrameSize; ++j)
psMic[j] = static_cast<short>(qBound(-32768.f, (ptr[j] * mul), 32767.f));
// If we have echo cancellation enabled...
if(iEchoChannels > 0)
{
resync.addMic(psMic);
} else {
encodeAudioFrame(AudioChunk(psMic));
}
}
}
}
void AudioInput::addEcho(const void *data, unsigned int nsamp) {
while (nsamp > 0) {
// Make sure we don't overrun the echo frame buffer
const unsigned int left = qMin(nsamp, iEchoLength - iEchoFilled);
if (bEchoMulti) {
const unsigned int samples = left * iEchoChannels;
if (eEchoFormat == SampleFloat) {
for (unsigned int i=0;i<samples;++i)
pfEchoInput[i + iEchoFilled * iEchoChannels] = reinterpret_cast<const float *>(data)[i];
}
else {
// 16bit PCM -> float
for (unsigned int i=0;i<samples;++i)
pfEchoInput[i + iEchoFilled * iEchoChannels] = static_cast<float>(reinterpret_cast<const short *>(data)[i]) * (1.0f / 32768.f);
}
} else {
// Mix echo channels (converts 16bit PCM -> float if needed)
imfEcho(pfEchoInput + iEchoFilled, data, left, iEchoChannels, uiEchoChannelMask);
}
iEchoFilled += left;
nsamp -= left;
// If new samples are left offset data pointer to point at the first one for next iteration
if (nsamp > 0) {
if (eEchoFormat == SampleFloat)
data = reinterpret_cast<const float *>(data) + left * iEchoChannels;
else
data = reinterpret_cast<const short *>(data) + left * iEchoChannels;
}
if (iEchoFilled == iEchoLength) {
//Frame complete
iEchoFilled = 0;
// Resample if necessary
float *pfOutput = srsEcho ? (float*)alloca(iEchoFrameSize*sizeof(float)) : nullptr;
float *ptr = srsEcho ? pfOutput : pfEchoInput;
if (srsEcho) {
spx_uint32_t inlen = iEchoLength;
spx_uint32_t outlen = iFrameSize;
speex_resampler_process_interleaved_float(srsEcho, pfEchoInput, &inlen, pfOutput, &outlen);
}
short *outbuff = new short[iEchoFrameSize];
// float -> 16bit PCM
const float mul = 32768.f;
for (int j=0;j<iEchoFrameSize;++j)
outbuff[j] = static_cast<short>(qBound(-32768.f, (ptr[j] * mul), 32767.f));
auto chunk = resync.addSpeaker(outbuff);
if(chunk.empty() == false)
{
encodeAudioFrame(chunk);
delete[] chunk.mic;
delete[] chunk.speaker;
}
}
}
}
void AudioInput::adjustBandwidth(int bitspersec, int &bitrate, int &frames, bool &allowLowDelay) {
frames = g.s.iFramesPerPacket;
bitrate = g.s.iQuality;
allowLowDelay = g.s.bAllowLowDelay;
if (bitspersec == -1) {
// No limit
} else {
if (getNetworkBandwidth(bitrate, frames) > bitspersec) {
if ((frames <= 4) && (bitspersec <= 32000))
frames = 4;
else if ((frames == 1) && (bitspersec <= 64000))
frames = 2;
else if ((frames == 2) && (bitspersec <= 48000))
frames = 4;
if (getNetworkBandwidth(bitrate, frames) > bitspersec) {
do {
bitrate -= 1000;
} while ((bitrate > 8000) && (getNetworkBandwidth(bitrate, frames) > bitspersec));
}
}
}
if (bitrate <= 8000)
bitrate = 8000;
}
void AudioInput::setMaxBandwidth(int bitspersec) {
if (bitspersec == g.iMaxBandwidth)
return;
int frames;
int bitrate;
bool allowLowDelay;
adjustBandwidth(bitspersec, bitrate, frames, allowLowDelay);
g.iMaxBandwidth = bitspersec;
if (bitspersec != -1) {
if ((bitrate != g.s.iQuality) || (frames != g.s.iFramesPerPacket))
g.mw->msgBox(tr("Server maximum network bandwidth is only %1 kbit/s. Audio quality auto-adjusted to %2 kbit/s (%3 ms)").arg(bitspersec / 1000).arg(bitrate / 1000).arg(frames*10));
}
AudioInputPtr ai = g.ai;
if (ai) {
g.iAudioBandwidth = getNetworkBandwidth(bitrate, frames);
ai->iAudioQuality = bitrate;
ai->iAudioFrames = frames;
ai->bAllowLowDelay = allowLowDelay;
return;
}
ai.reset();
Audio::stopInput();
Audio::startInput();
}
int AudioInput::getNetworkBandwidth(int bitrate, int frames) {
int overhead = 20 + 8 + 4 + 1 + 2 + (g.s.bTransmitPosition ? 12 : 0) + (NetworkConfig::TcpModeEnabled() ? 12 : 0) + frames;
overhead *= (800 / frames);
int bw = overhead + bitrate;
return bw;
}
void AudioInput::resetAudioProcessor() {
if (!bResetProcessor)
return;
int iArg;
if (sppPreprocess)
speex_preprocess_state_destroy(sppPreprocess);
if (sesEcho)
speex_echo_state_destroy(sesEcho);
resync.reset();
sppPreprocess = speex_preprocess_state_init(iFrameSize, iSampleRate);
iArg = 1;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_VAD, &iArg);
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_AGC, &iArg);
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_DENOISE, &iArg);
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_DEREVERB, &iArg);
iArg = 30000;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_AGC_TARGET, &iArg);
float v = 30000.0f / static_cast<float>(g.s.iMinLoudness);
iArg = iroundf(floorf(20.0f * log10f(v)));
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_AGC_MAX_GAIN, &iArg);
iArg = -60;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_AGC_DECREMENT, &iArg);
iArg = g.s.iNoiseSuppress;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &iArg);
if (iEchoChannels > 0) {
int filterSize = iFrameSize * (10 + resync.getNominalLag());
sesEcho = speex_echo_state_init_mc(iFrameSize, filterSize, 1, bEchoMulti ? iEchoChannels : 1);
iArg = iSampleRate;
speex_echo_ctl(sesEcho, SPEEX_ECHO_SET_SAMPLING_RATE, &iArg);
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_ECHO_STATE, sesEcho);
qWarning("AudioInput: ECHO CANCELLER ACTIVE");
} else {
sesEcho = NULL;
}
bResetEncoder = true;
bResetProcessor = false;
}
bool AudioInput::selectCodec() {
bool useOpus = false;
// Currently talking, use previous Opus status.
if (bPreviousVoice) {
useOpus = (umtType == MessageHandler::UDPVoiceOpus);
} else {
#ifdef USE_OPUS
if (g.bOpus || (g.s.lmLoopMode == Settings::Local)) {
useOpus = true;
}
#endif
}
if (!useOpus) {
CELTCodec *switchto = NULL;
if ((!g.uiSession || (g.s.lmLoopMode == Settings::Local)) && (!g.qmCodecs.isEmpty())) {
// Use latest for local loopback
QMap<int, CELTCodec *>::const_iterator i = g.qmCodecs.constEnd();
--i;
switchto = i.value();
} else {
// Currently talking, don't switch unless you must.
if (cCodec && bPreviousVoice) {
int v = cCodec->bitstreamVersion();
if ((v == g.iCodecAlpha) || (v == g.iCodecBeta))
switchto = cCodec;
}
}
if (!switchto) {
switchto = g.qmCodecs.value(g.bPreferAlpha ? g.iCodecAlpha : g.iCodecBeta);
if (!switchto)
switchto = g.qmCodecs.value(g.bPreferAlpha ? g.iCodecBeta : g.iCodecAlpha);
}
if (switchto != cCodec) {
if (cCodec && ceEncoder) {
cCodec->celt_encoder_destroy(ceEncoder);
ceEncoder = NULL;
}
cCodec = switchto;
if (cCodec)
ceEncoder = cCodec->encoderCreate();
}
if (!cCodec)
return false;
}
MessageHandler::UDPMessageType previousType = umtType;
if (useOpus) {
umtType = MessageHandler::UDPVoiceOpus;
} else {
if (!g.uiSession) {
umtType = MessageHandler::UDPVoiceCELTAlpha;
} else {
int v = cCodec->bitstreamVersion();
if (v == g.iCodecAlpha)
umtType = MessageHandler::UDPVoiceCELTAlpha;
else if (v == g.iCodecBeta)
umtType = MessageHandler::UDPVoiceCELTBeta;
else {
qWarning() << "Couldn't find message type for codec version" << v;
}
}
}
if (umtType != previousType) {
iBufferedFrames = 0;
qlFrames.clear();
opusBuffer.clear();
}
return true;
}
int AudioInput::encodeOpusFrame(short *source, int size, EncodingOutputBuffer& buffer) {
#ifdef USE_OPUS
int len;
if (!oCodec) {
return 0;
}
if (bResetEncoder) {
oCodec->opus_encoder_ctl(opusState, OPUS_RESET_STATE, NULL);
bResetEncoder = false;
}
oCodec->opus_encoder_ctl(opusState, OPUS_SET_BITRATE(iAudioQuality));
len = oCodec->opus_encode(opusState, source, size, &buffer[0], static_cast<opus_int32>(buffer.size()));
const int tenMsFrameCount = (size / iFrameSize);
iBitrate = (len * 100 * 8) / tenMsFrameCount;
return len;
#else
return 0;
#endif
}
int AudioInput::encodeCELTFrame(short *psSource, EncodingOutputBuffer& buffer) {
int len;
if (!cCodec)
return 0;
if (bResetEncoder) {
cCodec->celt_encoder_ctl(ceEncoder, CELT_RESET_STATE);
bResetEncoder = false;
}
cCodec->celt_encoder_ctl(ceEncoder, CELT_SET_PREDICTION(0));
cCodec->celt_encoder_ctl(ceEncoder, CELT_SET_VBR_RATE(iAudioQuality));
len = cCodec->encode(ceEncoder, psSource, &buffer[0], qMin<int>(iAudioQuality / (8 * 100), static_cast<int>(buffer.size())));
iBitrate = len * 100 * 8;
return len;
}
void AudioInput::encodeAudioFrame(AudioChunk chunk) {
int iArg;
int i;
float sum;
short max;
short *psSource;
iFrameCounter++;
// As g.iTarget is not protected by any locks, we avoid race-conditions by
// copying it once at this point and stick to whatever value it is here. Thus
// if the value of g.iTarget changes during the execution of this function,
// it won't cause any inconsistencies and the change is reflected once this
// function is called again.
int voiceTargetID = g.iTarget;
if (! bRunning)
return;
sum=1.0f;
max = 1;
for (i=0;i<iFrameSize;i++) {
sum += static_cast<float>(chunk.mic[i] * chunk.mic[i]);
max = std::max(static_cast<short>(abs(chunk.mic[i])), max);
}
dPeakMic = qMax(20.0f*log10f(sqrtf(sum / static_cast<float>(iFrameSize)) / 32768.0f), -96.0f);
dMaxMic = max;
if (chunk.speaker && (iEchoChannels > 0)) {
sum=1.0f;
for (i=0;i<iEchoFrameSize;i++)
sum += static_cast<float>(chunk.speaker[i] * chunk.speaker[i]);
dPeakSpeaker = qMax(20.0f*log10f(sqrtf(sum / static_cast<float>(iFrameSize)) / 32768.0f), -96.0f);
} else {
dPeakSpeaker = 0.0;
}
QMutexLocker l(&qmSpeex);
resetAudioProcessor();
#ifdef USE_RNNOISE
// At the time of writing this code, RNNoise only supports a sample rate of 48000 Hz.
if (g.s.bDenoise && denoiseState && (iFrameSize == 480)) {
float denoiseFrames[480];
for (int i = 0; i < 480; i++) {
denoiseFrames[i] = chunk.mic[i];
}
rnnoise_process_frame(denoiseState, denoiseFrames, denoiseFrames);
for (int i = 0; i < 480; i++) {
chunk.mic[i] = denoiseFrames[i];
}
}
#endif
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_AGC_GAIN, &iArg);
float gainValue = static_cast<float>(iArg);
iArg = g.s.iNoiseSuppress - iArg;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &iArg);
short psClean[iFrameSize];
if (sesEcho && chunk.speaker) {
speex_echo_cancellation(sesEcho, chunk.mic, chunk.speaker, psClean);
speex_preprocess_run(sppPreprocess, psClean);
psSource = psClean;
} else {
speex_preprocess_run(sppPreprocess, chunk.mic);
psSource = chunk.mic;
}
sum=1.0f;
for (i=0;i<iFrameSize;i++)
sum += static_cast<float>(psSource[i] * psSource[i]);
float micLevel = sqrtf(sum / static_cast<float>(iFrameSize));
dPeakSignal = qMax(20.0f*log10f(micLevel / 32768.0f), -96.0f);
if(bDebugDumpInput)
{
outMic.write(reinterpret_cast<const char*>(chunk.mic),iFrameSize*sizeof(short));
if(chunk.speaker)
outSpeaker.write(reinterpret_cast<const char*>(chunk.speaker),iEchoFrameSize*sizeof(short));
outProcessed.write(reinterpret_cast<const char*>(psSource),iFrameSize*sizeof(short));
}
spx_int32_t prob = 0;
speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB, &prob);
fSpeechProb = static_cast<float>(prob) / 100.0f;
// clean microphone level: peak of filtered signal attenuated by AGC gain
dPeakCleanMic = qMax(dPeakSignal - gainValue, -96.0f);
float level = (g.s.vsVAD == Settings::SignalToNoise) ? fSpeechProb : (1.0f + dPeakCleanMic / 96.0f);
bool bIsSpeech = false;
if (level > g.s.fVADmax) {
// Voice-activation threshold has been reached
bIsSpeech = true;
} else if (level > g.s.fVADmin && bPreviousVoice) {
// Voice-deactivation threshold has not yet been reached
bIsSpeech = true;
}
if (! bIsSpeech) {
iHoldFrames++;
if (iHoldFrames < g.s.iVoiceHold)
bIsSpeech = true;
} else {
iHoldFrames = 0;
}
if (g.s.atTransmit == Settings::Continuous) {
// Continous transmission is enabled
bIsSpeech = true;
} else if (g.s.atTransmit == Settings::PushToTalk) {
// PTT is enabled, so check if it is currently active
bIsSpeech = g.s.uiDoublePush && ((g.uiDoublePush < g.s.uiDoublePush) || (g.tDoublePush.elapsed() < g.s.uiDoublePush));
}
// If g.iPushToTalk > 0 that means that we are currently in some sort of PTT action. For
// instance this could mean we're currently whispering
bIsSpeech = bIsSpeech || (g.iPushToTalk > 0);
ClientUser *p = ClientUser::get(g.uiSession);
if (g.s.bMute || ((g.s.lmLoopMode != Settings::Local) && p && (p->bMute || p->bSuppress)) || g.bPushToMute || (voiceTargetID < 0)) {
bIsSpeech = false;
}
if (bIsSpeech) {
iSilentFrames = 0;
} else {
iSilentFrames++;
if (iSilentFrames > 500)