void JackAudioSystem::initializeInput() {
QMutexLocker lock(&qmWait);
if (!jasys->bJackIsGood) {
return;
}
AudioInputPtr ai = g.ai;
JackAudioInput * const jai = dynamic_cast<JackAudioInput *>(ai.get());
if (jai) {
jai->qmMutex.lock();
}
in_port = jack_port_register(client, "input", JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
if (in_port == NULL) {
qWarning("JackAudioSystem: unable to register 'input' port");
return;
}
bInputIsGood = true;
if (jai) {
jai->qmMutex.unlock();
}
}
void AudioInput::setMaxBandwidth(int bitspersec) {
if (bitspersec == g.iMaxBandwidth)
return;
int frames;
int bitrate;
adjustBandwidth(bitspersec, bitrate, frames);
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;
return;
}
ai.reset();
Audio::stopInput();
Audio::startInput();
}
void JackAudioSystem::destroyInput() {
AudioInputPtr ai = g.ai;
JackAudioInput * const jai = dynamic_cast<JackAudioInput *>(ai.get());
if (jai) {
jai->qmMutex.lock();
}
if (in_port != NULL) {
int err = jack_port_unregister(client, in_port);
if (err != 0) {
qWarning("JackAudioSystem: unable to unregister in port - jack_port_unregister() returned %i", err);
return;
}
}
bInputIsGood = false;
if (jai) {
jai->qmMutex.unlock();
}
}
void PulseAudioSystem::read_callback(pa_stream *s, size_t bytes, void *userdata) {
PulseAudioSystem *pas = reinterpret_cast<PulseAudioSystem *>(userdata);
size_t length = bytes;
const void *data = NULL;
pa_stream_peek(s, &data, &length);
if (data == NULL && length > 0) {
qWarning("PulseAudio: pa_stream_peek reports no data at current read index.");
} else if (data == NULL && length == 0) {
qWarning("PulseAudio: pa_stream_peek reports empty memblockq.");
} else if (data == NULL || length == 0) {
qWarning("PulseAudio: invalid pa_stream_peek state encountered.");
return;
}
AudioInputPtr ai = g.ai;
PulseAudioInput *pai = dynamic_cast<PulseAudioInput *>(ai.get());
if (! pai) {
if (length > 0) {
pa_stream_drop(s);
}
pas->wakeup();
return;
}
const pa_sample_spec *pss = pa_stream_get_sample_spec(s);
if (s == pas->pasInput) {
if (!pa_sample_spec_equal(pss, &pai->pssMic)) {
pai->pssMic = *pss;
pai->iMicFreq = pss->rate;
pai->iMicChannels = pss->channels;
if (pss->format == PA_SAMPLE_FLOAT32NE)
pai->eMicFormat = PulseAudioInput::SampleFloat;
else
pai->eMicFormat = PulseAudioInput::SampleShort;
pai->initializeMixer();
}
if (data != NULL) {
pai->addMic(data, static_cast<unsigned int>(length) / pai->iMicSampleSize);
}
} else if (s == pas->pasSpeaker) {
if (!pa_sample_spec_equal(pss, &pai->pssEcho)) {
pai->pssEcho = *pss;
pai->iEchoFreq = pss->rate;
pai->iEchoChannels = pss->channels;
if (pss->format == PA_SAMPLE_FLOAT32NE)
pai->eEchoFormat = PulseAudioInput::SampleFloat;
else
pai->eEchoFormat = PulseAudioInput::SampleShort;
pai->initializeMixer();
}
if (data != NULL) {
pai->addEcho(data, static_cast<unsigned int>(length) / pai->iEchoSampleSize);
}
}
if (length > 0) {
pa_stream_drop(s);
}
}
void PulseAudioSystem::eventCallback(pa_mainloop_api *api, pa_defer_event *) {
api->defer_enable(pade, false);
if (! bSourceDone || ! bSinkDone || ! bServerDone)
return;
AudioInputPtr ai = g.ai;
AudioOutputPtr ao = g.ao;
AudioInput *raw_ai = ai.get();
AudioOutput *raw_ao = ao.get();
PulseAudioInput *pai = dynamic_cast<PulseAudioInput *>(raw_ai);
PulseAudioOutput *pao = dynamic_cast<PulseAudioOutput *>(raw_ao);
if (raw_ao) {
QString odev = outputDevice();
pa_stream_state ost = pasOutput ? pa_stream_get_state(pasOutput) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if (! pao && (ost == PA_STREAM_READY)) {
do_stop = true;
} else if (pao) {
switch (ost) {
case PA_STREAM_TERMINATED: {
if (pasOutput)
pa_stream_unref(pasOutput);
pa_sample_spec pss = qhSpecMap.value(odev);
pa_channel_map pcm = qhChanMap.value(odev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
if ((pss.channels == 0) || (! g.s.doPositionalAudio()))
pss.channels = 1;
pasOutput = pa_stream_new(pacContext, mumble_sink_input, &pss, (pss.channels == 1) ? NULL : &pcm);
pa_stream_set_state_callback(pasOutput, stream_callback, this);
pa_stream_set_write_callback(pasOutput, write_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (g.s.iOutputDelay != iDelayCache) {
do_stop = true;
} else if (g.s.doPositionalAudio() != bPositionalCache) {
do_stop = true;
} else if (odev != qsOutputCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping output");
pa_stream_disconnect(pasOutput);
iSinkId = -1;
} else if (do_start) {
qWarning("PulseAudio: Starting output: %s", qPrintable(odev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasOutput);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pao->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen * (g.s.iOutputDelay+1);
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
iDelayCache = g.s.iOutputDelay;
bPositionalCache = g.s.doPositionalAudio();
qsOutputCache = odev;
pa_stream_connect_playback(pasOutput, qPrintable(odev), &buff, PA_STREAM_ADJUST_LATENCY, NULL, NULL);
pa_context_get_sink_info_by_name(pacContext, qPrintable(odev), sink_info_callback, this);
}
}
if (raw_ai) {
QString idev = inputDevice();
pa_stream_state ist = pasInput ? pa_stream_get_state(pasInput) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if (! pai && (ist == PA_STREAM_READY)) {
do_stop = true;
} else if (pai) {
switch (ist) {
case PA_STREAM_TERMINATED: {
if (pasInput)
pa_stream_unref(pasInput);
pa_sample_spec pss = qhSpecMap.value(idev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
pss.channels = 1;
pasInput = pa_stream_new(pacContext, "Microphone", &pss, NULL);
pa_stream_set_state_callback(pasInput, stream_callback, this);
pa_stream_set_read_callback(pasInput, read_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (idev != qsInputCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping input");
pa_stream_disconnect(pasInput);
} else if (do_start) {
qWarning("PulseAudio: Starting input %s",qPrintable(idev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasInput);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pai->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen;
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
qsInputCache = idev;
pa_stream_connect_record(pasInput, qPrintable(idev), &buff, PA_STREAM_ADJUST_LATENCY);
}
}
if (raw_ai) {
QString odev = outputDevice();
QString edev = qhEchoMap.value(odev);
pa_stream_state est = pasSpeaker ? pa_stream_get_state(pasSpeaker) : PA_STREAM_TERMINATED;
bool do_stop = false;
bool do_start = false;
if ((! pai || ! g.s.doEcho()) && (est == PA_STREAM_READY)) {
do_stop = true;
} else if (pai && g.s.doEcho()) {
switch (est) {
case PA_STREAM_TERMINATED: {
if (pasSpeaker)
pa_stream_unref(pasSpeaker);
pa_sample_spec pss = qhSpecMap.value(edev);
pa_channel_map pcm = qhChanMap.value(edev);
if ((pss.format != PA_SAMPLE_FLOAT32NE) && (pss.format != PA_SAMPLE_S16NE))
pss.format = PA_SAMPLE_FLOAT32NE;
if (pss.rate == 0)
pss.rate = SAMPLE_RATE;
if ((pss.channels == 0) || (! g.s.bEchoMulti))
pss.channels = 1;
pasSpeaker = pa_stream_new(pacContext, mumble_echo, &pss, (pss.channels == 1) ? NULL : &pcm);
pa_stream_set_state_callback(pasSpeaker, stream_callback, this);
pa_stream_set_read_callback(pasSpeaker, read_callback, this);
}
case PA_STREAM_UNCONNECTED:
do_start = true;
break;
case PA_STREAM_READY: {
if (g.s.bEchoMulti != bEchoMultiCache) {
do_stop = true;
} else if (edev != qsEchoCache) {
do_stop = true;
}
break;
}
default:
break;
}
}
if (do_stop) {
qWarning("PulseAudio: Stopping echo");
pa_stream_disconnect(pasSpeaker);
} else if (do_start) {
qWarning("PulseAudio: Starting echo: %s", qPrintable(edev));
pa_buffer_attr buff;
const pa_sample_spec *pss = pa_stream_get_sample_spec(pasSpeaker);
const size_t sampleSize = (pss->format == PA_SAMPLE_FLOAT32NE) ? sizeof(float) : sizeof(short);
const unsigned int iBlockLen = ((pai->iFrameSize * pss->rate) / SAMPLE_RATE) * pss->channels * static_cast<unsigned int>(sampleSize);
buff.tlength = iBlockLen;
buff.minreq = iBlockLen;
buff.maxlength = -1;
buff.prebuf = -1;
buff.fragsize = iBlockLen;
bEchoMultiCache = g.s.bEchoMulti;
qsEchoCache = edev;
pa_stream_connect_record(pasSpeaker, qPrintable(edev), &buff, PA_STREAM_ADJUST_LATENCY);
}
}
}
void AudioStats::on_Tick_timeout() {
AudioInputPtr ai = g.ai;
if (ai.get() == NULL || ! ai->sppPreprocess)
return;
bool nTalking = ai->isTransmitting();
QString txt;
txt.sprintf("%06.2f dB",ai->dPeakMic);
qlMicLevel->setText(txt);
txt.sprintf("%06.2f dB",ai->dPeakSpeaker);
qlSpeakerLevel->setText(txt);
txt.sprintf("%06.2f dB",ai->dPeakSignal);
qlSignalLevel->setText(txt);
spx_int32_t ps_size = 0;
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD_SIZE, &ps_size);
STACKVAR(spx_int32_t, noise, ps_size);
STACKVAR(spx_int32_t, ps, ps_size);
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD, ps);
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_NOISE_PSD, noise);
float s = 0.0f;
float n = 0.0001f;
int start = (ps_size * 300) / SAMPLE_RATE;
int stop = (ps_size * 2000) / SAMPLE_RATE;
for (int i=start;i<stop;i++) {
s += sqrtf(static_cast<float>(ps[i]));
n += sqrtf(static_cast<float>(noise[i]));
}
txt.sprintf("%06.3f",s / n);
qlMicSNR->setText(txt);
spx_int32_t v;
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_AGC_GAIN, &v);
float fv = powf(10.0f, (static_cast<float>(v) / 20.0f));
txt.sprintf("%03.0f%%",100.0f / fv);
qlMicVolume->setText(txt);
txt.sprintf("%03.0f%%",ai->fSpeechProb * 100.0f);
qlSpeechProb->setText(txt);
txt.sprintf("%04.1f kbit/s",static_cast<float>(ai->iBitrate) / 1000.0f);
qlBitrate->setText(txt);
if (nTalking != bTalking) {
bTalking = nTalking;
QFont f = qlSpeechProb->font();
f.setBold(bTalking);
qlSpeechProb->setFont(f);
}
if (g.uiDoublePush > 1000000)
txt = tr(">1000 ms");
else
txt.sprintf("%04llu ms",g.uiDoublePush / 1000);
qlDoublePush->setText(txt);
abSpeech->iBelow = iroundf(g.s.fVADmin * 32767.0f);
abSpeech->iAbove = iroundf(g.s.fVADmax * 32767.0f);
if (g.s.vsVAD == Settings::Amplitude) {
abSpeech->iValue = iroundf((32767.f/96.0f) * (96.0f + ai->dPeakMic));
} else {
abSpeech->iValue = iroundf(ai->fSpeechProb * 32767.0f);
}
abSpeech->update();
anwNoise->update();
if (aewEcho)
aewEcho->updateGL();
}
void AudioNoiseWidget::paintEvent(QPaintEvent *) {
QPainter paint(this);
QPalette pal;
paint.fillRect(rect(), pal.color(QPalette::Background));
AudioInputPtr ai = g.ai;
if (ai.get() == NULL || ! ai->sppPreprocess)
return;
QPolygonF poly;
ai->qmSpeex.lock();
spx_int32_t ps_size = 0;
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD_SIZE, &ps_size);
STACKVAR(spx_int32_t, noise, ps_size);
STACKVAR(spx_int32_t, ps, ps_size);
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD, ps);
speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_NOISE_PSD, noise);
ai->qmSpeex.unlock();
qreal sx, sy;
sx = (static_cast<float>(width()) - 1.0f) / static_cast<float>(ps_size);
sy = static_cast<float>(height()) - 1.0f;
poly << QPointF(0.0f, height() - 1);
float fftmul = 1.0 / (32768.0);
for (int i=0; i < ps_size; i++) {
qreal xp, yp;
xp = i * sx;
yp = sqrtf(sqrtf(static_cast<float>(noise[i]))) - 1.0f;
yp = yp * fftmul;
yp = qMin<qreal>(yp * 3000.0f, 1.0f);
yp = (1 - yp) * sy;
poly << QPointF(xp, yp);
}
poly << QPointF(width() - 1, height() - 1);
poly << QPointF(0.0f, height() - 1);
paint.setPen(Qt::blue);
paint.setBrush(Qt::blue);
paint.drawPolygon(poly);
poly.clear();
for (int i=0;i < ps_size; i++) {
qreal xp, yp;
xp = i * sx;
yp = sqrtf(sqrtf(static_cast<float>(ps[i]))) - 1.0f;
yp = yp * fftmul;
yp = qMin(yp * 3000.0, 1.0);
yp = (1 - yp) * sy;
poly << QPointF(xp, yp);
}
paint.setPen(Qt::red);
paint.drawPolyline(poly);
}
int JackAudioSystem::process_callback(jack_nframes_t nframes, void *arg) {
JackAudioSystem * const jas = static_cast<JackAudioSystem*>(arg);
if (jas && jas->bJackIsGood) {
AudioInputPtr ai = g.ai;
AudioOutputPtr ao = g.ao;
JackAudioInput * const jai = dynamic_cast<JackAudioInput *>(ai.get());
JackAudioOutput * const jao = dynamic_cast<JackAudioOutput *>(ao.get());
if (jai && jai->isRunning() && jai->iMicChannels > 0 && !jai->isFinished()) {
QMutexLocker(&jai->qmMutex);
void *input = jack_port_get_buffer(jas->in_port, nframes);
if (input != NULL) {
jai->addMic(input, nframes);
}
}
if (jao && jao->isRunning() && jao->iChannels > 0 && !jao->isFinished()) {
QMutexLocker(&jao->qmMutex);
jack_default_audio_sample_t *port_buffers[JACK_MAX_OUTPUT_PORTS];
for (unsigned int i = 0; i < jao->iChannels; ++i) {
port_buffers[i] = (jack_default_audio_sample_t*)jack_port_get_buffer(jas->out_ports[i], nframes);
if (port_buffers[i] == NULL) {
return 1;
}
}
jack_default_audio_sample_t * const buffer = jas->output_buffer;
memset(buffer, 0, sizeof(jack_default_audio_sample_t) * nframes * jao->iChannels);
jao->mix(buffer, nframes);
if (jao->iChannels == 1) {
memcpy(port_buffers[0], buffer, sizeof(jack_default_audio_sample_t) * nframes);
} else {
// de-interleave channels
for (unsigned int i = 0; i < nframes * jao->iChannels; ++i) {
port_buffers[i % jao->iChannels][i / jao->iChannels] = buffer[i];
}
}
}
}
return 0;
}