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|
// Copyright 2005-2021 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>.
#include "AudioOutput.h"
#include "AudioInput.h"
#include "AudioOutputSample.h"
#include "AudioOutputSpeech.h"
#include "Channel.h"
#include "ChannelListener.h"
#include "Message.h"
#include "PacketDataStream.h"
#include "Plugins.h"
#include "ServerHandler.h"
#include "SpeechFlags.h"
#include "Timer.h"
#include "User.h"
#include "Utils.h"
#include "VoiceRecorder.h"
#include "Global.h"
#include <cmath>
// Remember that we cannot use static member classes that are not pointers, as the constructor
// for AudioOutputRegistrar() might be called before they are initialized, as the constructor
// is called from global initialization.
// Hence, we allocate upon first call.
QMap< QString, AudioOutputRegistrar * > *AudioOutputRegistrar::qmNew;
QString AudioOutputRegistrar::current = QString();
AudioOutputRegistrar::AudioOutputRegistrar(const QString &n, int p) : name(n), priority(p) {
if (!qmNew)
qmNew = new QMap< QString, AudioOutputRegistrar * >();
qmNew->insert(name, this);
}
AudioOutputRegistrar::~AudioOutputRegistrar() {
qmNew->remove(name);
}
AudioOutputPtr AudioOutputRegistrar::newFromChoice(QString choice) {
if (!qmNew)
return AudioOutputPtr();
if (!choice.isEmpty() && qmNew->contains(choice)) {
Global::get().s.qsAudioOutput = choice;
current = choice;
return AudioOutputPtr(qmNew->value(choice)->create());
}
choice = Global::get().s.qsAudioOutput;
if (qmNew->contains(choice)) {
current = choice;
return AudioOutputPtr(qmNew->value(choice)->create());
}
AudioOutputRegistrar *r = nullptr;
foreach (AudioOutputRegistrar *aor, *qmNew)
if (!r || (aor->priority > r->priority))
r = aor;
if (r) {
current = r->name;
return AudioOutputPtr(r->create());
}
return AudioOutputPtr();
}
bool AudioOutputRegistrar::canMuteOthers() const {
return false;
}
bool AudioOutputRegistrar::usesOutputDelay() const {
return true;
}
bool AudioOutputRegistrar::canExclusive() const {
return false;
}
AudioOutput::~AudioOutput() {
bRunning = false;
wait();
wipe();
delete[] fSpeakers;
delete[] fSpeakerVolume;
delete[] bSpeakerPositional;
delete[] fStereoPanningFactor;
}
// Here's the theory.
// We support sound "bloom"ing. That is, if sound comes directly from the left, if it is sufficiently
// close, we'll hear it full intensity from the left side, and "bloom" intensity from the right side.
float AudioOutput::calcGain(float dotproduct, float distance) {
float dotfactor = (dotproduct + 1.0f) / 2.0f;
float att;
// No distance attenuation
if (Global::get().s.fAudioMaxDistVolume > 0.99f) {
att = qMin(1.0f, dotfactor + Global::get().s.fAudioBloom);
} else if (distance < Global::get().s.fAudioMinDistance) {
float bloomfac = Global::get().s.fAudioBloom * (1.0f - distance / Global::get().s.fAudioMinDistance);
att = qMin(1.0f, bloomfac + dotfactor);
} else {
float datt;
if (distance >= Global::get().s.fAudioMaxDistance) {
datt = Global::get().s.fAudioMaxDistVolume;
} else {
float mvol = Global::get().s.fAudioMaxDistVolume;
if (mvol < 0.01f)
mvol = 0.01f;
float drel = (distance - Global::get().s.fAudioMinDistance) / (Global::get().s.fAudioMaxDistance - Global::get().s.fAudioMinDistance);
datt = powf(10.0f, log10f(mvol) * drel);
}
att = datt * dotfactor;
}
return att;
}
void AudioOutput::wipe() {
foreach (AudioOutputUser *aop, qmOutputs)
removeBuffer(aop);
}
const float *AudioOutput::getSpeakerPos(unsigned int &speakers) {
if ((iChannels > 0) && fSpeakers) {
speakers = iChannels;
return fSpeakers;
}
return nullptr;
}
void AudioOutput::addFrameToBuffer(ClientUser *user, const QByteArray &qbaPacket, unsigned int iSeq,
MessageHandler::UDPMessageType type) {
if (iChannels == 0)
return;
qrwlOutputs.lockForRead();
// qmOutputs is a map of users and their AudioOutputUser objects, which will be create when audio from that user
// is received. This map will be iterated in mix(). After one's audio is finished, his AudioOutputUser will be
// removed from this map.
AudioOutputSpeech *aop = qobject_cast< AudioOutputSpeech * >(qmOutputs.value(user));
if (!UDPMessageTypeIsValidVoicePacket(type)) {
qWarning("AudioOutput: ignored frame with invalid message type 0x%x in addFrameToBuffer().",
static_cast< unsigned char >(type));
return;
}
if (!aop || (aop->umtType != type)) {
qrwlOutputs.unlock();
if (aop)
removeBuffer(aop);
while ((iMixerFreq == 0) && isAlive()) {
QThread::yieldCurrentThread();
}
if (!iMixerFreq)
return;
qrwlOutputs.lockForWrite();
aop = new AudioOutputSpeech(user, iMixerFreq, type, iBufferSize);
qmOutputs.replace(user, aop);
}
aop->addFrameToBuffer(qbaPacket, iSeq);
qrwlOutputs.unlock();
}
void AudioOutput::removeBuffer(const ClientUser *user) {
removeBuffer(qmOutputs.value(user));
}
void AudioOutput::removeBuffer(AudioOutputUser *aop) {
QWriteLocker locker(&qrwlOutputs);
QMultiHash< const ClientUser *, AudioOutputUser * >::iterator i;
for (i = qmOutputs.begin(); i != qmOutputs.end(); ++i) {
if (i.value() == aop) {
qmOutputs.erase(i);
delete aop;
break;
}
}
}
AudioOutputSample *AudioOutput::playSample(const QString &filename, bool loop) {
SoundFile *handle = AudioOutputSample::loadSndfile(filename);
if (!handle)
return nullptr;
Timer t;
const quint64 oneSecond = 1000000;
while (!t.isElapsed(oneSecond) && (iMixerFreq == 0) && isAlive()) {
QThread::yieldCurrentThread();
}
// If we've waited for more than one second, we declare timeout.
if (t.isElapsed(oneSecond)) {
qWarning("AudioOutput: playSample() timed out after 1 second: device not ready");
return nullptr;
}
if (!iMixerFreq)
return nullptr;
QWriteLocker locker(&qrwlOutputs);
AudioOutputSample *aos = new AudioOutputSample(filename, handle, loop, iMixerFreq, iBufferSize);
qmOutputs.insert(nullptr, aos);
return aos;
}
void AudioOutput::initializeMixer(const unsigned int *chanmasks, bool forceheadphone) {
delete[] fSpeakers;
delete[] bSpeakerPositional;
delete[] fSpeakerVolume;
delete[] fStereoPanningFactor;
fSpeakers = new float[iChannels * 3];
bSpeakerPositional = new bool[iChannels];
fSpeakerVolume = new float[iChannels];
fStereoPanningFactor = new float[iChannels * 2];
memset(fSpeakers, 0, sizeof(float) * iChannels * 3);
memset(bSpeakerPositional, 0, sizeof(bool) * iChannels);
for (unsigned int i = 0; i < iChannels; ++i)
fSpeakerVolume[i] = 1.0f;
if (iChannels > 1) {
for (unsigned int i = 0; i < iChannels; i++) {
float *s = &fSpeakers[3 * i];
bSpeakerPositional[i] = true;
switch (chanmasks[i]) {
case SPEAKER_FRONT_LEFT:
s[0] = -0.5f;
s[2] = 1.0f;
break;
case SPEAKER_FRONT_RIGHT:
s[0] = 0.5f;
s[2] = 1.0f;
break;
case SPEAKER_FRONT_CENTER:
s[2] = 1.0f;
break;
case SPEAKER_LOW_FREQUENCY:
break;
case SPEAKER_BACK_LEFT:
s[0] = -0.5f;
s[2] = -1.0f;
break;
case SPEAKER_BACK_RIGHT:
s[0] = 0.5f;
s[2] = -1.0f;
break;
case SPEAKER_FRONT_LEFT_OF_CENTER:
s[0] = -0.25;
s[2] = 1.0f;
break;
case SPEAKER_FRONT_RIGHT_OF_CENTER:
s[0] = 0.25;
s[2] = 1.0f;
break;
case SPEAKER_BACK_CENTER:
s[2] = -1.0f;
break;
case SPEAKER_SIDE_LEFT:
s[0] = -1.0f;
break;
case SPEAKER_SIDE_RIGHT:
s[0] = 1.0f;
break;
case SPEAKER_TOP_CENTER:
s[1] = 1.0f;
s[2] = 1.0f;
break;
case SPEAKER_TOP_FRONT_LEFT:
s[0] = -0.5f;
s[1] = 1.0f;
s[2] = 1.0f;
break;
case SPEAKER_TOP_FRONT_CENTER:
s[1] = 1.0f;
s[2] = 1.0f;
break;
case SPEAKER_TOP_FRONT_RIGHT:
s[0] = 0.5f;
s[1] = 1.0f;
s[2] = 1.0f;
break;
case SPEAKER_TOP_BACK_LEFT:
s[0] = -0.5f;
s[1] = 1.0f;
s[2] = -1.0f;
break;
case SPEAKER_TOP_BACK_CENTER:
s[1] = 1.0f;
s[2] = -1.0f;
break;
case SPEAKER_TOP_BACK_RIGHT:
s[0] = 0.5f;
s[1] = 1.0f;
s[2] = -1.0f;
break;
default:
bSpeakerPositional[i] = false;
fSpeakerVolume[i] = 0.0f;
qWarning("AudioOutput: Unknown speaker %d: %08x", i, chanmasks[i]);
break;
}
if (Global::get().s.bPositionalHeadphone || forceheadphone) {
s[1] = 0.0f;
s[2] = 0.0f;
if (s[0] == 0.0f)
fSpeakerVolume[i] = 0.0f;
}
}
for (unsigned int i = 0; i < iChannels; i++) {
float d = sqrtf(fSpeakers[3 * i + 0] * fSpeakers[3 * i + 0] + fSpeakers[3 * i + 1] * fSpeakers[3 * i + 1]
+ fSpeakers[3 * i + 2] * fSpeakers[3 * i + 2]);
if (d > 0.0f) {
fSpeakers[3 * i + 0] /= d;
fSpeakers[3 * i + 1] /= d;
fSpeakers[3 * i + 2] /= d;
}
float *spf = &fStereoPanningFactor[2 * i];
spf[0] = (1.0 - fSpeakers[i * 3 + 0]) / 2.0;
spf[1] = (1.0 + fSpeakers[i * 3 + 0]) / 2.0;
}
} else if (iChannels == 1) {
fStereoPanningFactor[0] = 0.5;
fStereoPanningFactor[1] = 0.5;
}
iSampleSize = static_cast< int >(iChannels * ((eSampleFormat == SampleFloat) ? sizeof(float) : sizeof(short)));
qWarning("AudioOutput: Initialized %d channel %d hz mixer", iChannels, iMixerFreq);
}
bool AudioOutput::mix(void *outbuff, unsigned int frameCount) {
#ifdef USE_MANUAL_PLUGIN
positions.clear();
#endif
// A list of users that have audio to contribute
QList< AudioOutputUser * > qlMix;
// A list of users that no longer have any audio to play and can thus be deleted
QList< AudioOutputUser * > qlDel;
if (Global::get().s.fVolume < 0.01f) {
return false;
}
const float adjustFactor = std::pow(10.f, -18.f / 20);
const float mul = Global::get().s.fVolume;
const unsigned int nchan = iChannels;
ServerHandlerPtr sh = Global::get().sh;
VoiceRecorderPtr recorder;
if (sh) {
recorder = Global::get().sh->recorder;
}
qrwlOutputs.lockForRead();
bool prioritySpeakerActive = false;
// Get the users that are currently talking (and are thus serving as an audio source)
QMultiHash< const ClientUser *, AudioOutputUser * >::const_iterator it = qmOutputs.constBegin();
while (it != qmOutputs.constEnd()) {
AudioOutputUser *aop = it.value();
if (!aop->prepareSampleBuffer(frameCount)) {
qlDel.append(aop);
} else {
qlMix.append(aop);
const ClientUser *user = it.key();
if (user && user->bPrioritySpeaker) {
prioritySpeakerActive = true;
}
}
++it;
}
if (Global::get().prioritySpeakerActiveOverride) {
prioritySpeakerActive = true;
}
if (!qlMix.isEmpty()) {
// There are audio sources available -> mix those sources together and feed them into the audio backend
STACKVAR(float, speaker, iChannels * 3);
STACKVAR(float, svol, iChannels);
STACKVAR(float, fOutput, iChannels *frameCount);
// If the audio backend uses a float-array we can sample and mix the audio sources directly into the output.
// Otherwise we'll have to use an intermediate buffer which we will convert to an array of shorts later
float *output = (eSampleFormat == SampleFloat) ? reinterpret_cast< float * >(outbuff) : fOutput;
bool validListener = false;
memset(output, 0, sizeof(float) * frameCount * iChannels);
// Initialize recorder if recording is enabled
boost::shared_array< float > recbuff;
if (recorder) {
recbuff = boost::shared_array< float >(new float[frameCount]);
memset(recbuff.get(), 0, sizeof(float) * frameCount);
recorder->prepareBufferAdds();
}
for (unsigned int i = 0; i < iChannels; ++i)
svol[i] = mul * fSpeakerVolume[i];
if (Global::get().s.bPositionalAudio && (iChannels > 1) && Global::get().p->fetch()
&& (Global::get().bPosTest || Global::get().p->fCameraPosition[0] != 0 || Global::get().p->fCameraPosition[1] != 0
|| Global::get().p->fCameraPosition[2] != 0)) {
// Calculate the positional audio effects if it is enabled
float front[3] = { Global::get().p->fCameraFront[0], Global::get().p->fCameraFront[1], Global::get().p->fCameraFront[2] };
float top[3] = { Global::get().p->fCameraTop[0], Global::get().p->fCameraTop[1], Global::get().p->fCameraTop[2] };
// Front vector is dominant; if it's zero we presume all is zero.
float flen = sqrtf(front[0] * front[0] + front[1] * front[1] + front[2] * front[2]);
if (flen > 0.0f) {
front[0] *= (1.0f / flen);
front[1] *= (1.0f / flen);
front[2] *= (1.0f / flen);
float tlen = sqrtf(top[0] * top[0] + top[1] * top[1] + top[2] * top[2]);
if (tlen > 0.0f) {
top[0] *= (1.0f / tlen);
top[1] *= (1.0f / tlen);
top[2] *= (1.0f / tlen);
} else {
top[0] = 0.0f;
top[1] = 1.0f;
top[2] = 0.0f;
}
const float dotproduct = front[0] * top[0] + front[1] * top[1] + front[2] * top[2];
const float error = std::abs(dotproduct);
if (error > 0.5f) {
// Not perpendicular by a large margin. Assume Y up and rotate 90 degrees.
float azimuth = 0.0f;
if ((front[0] != 0.0f) || (front[2] != 0.0f))
azimuth = atan2f(front[2], front[0]);
float inclination = acosf(front[1]) - static_cast< float >(M_PI) / 2.0f;
top[0] = sinf(inclination) * cosf(azimuth);
top[1] = cosf(inclination);
top[2] = sinf(inclination) * sinf(azimuth);
} else if (error > 0.01f) {
// Not perpendicular by a small margin. Find the nearest perpendicular vector.
top[0] -= front[0] * dotproduct;
top[1] -= front[1] * dotproduct;
top[2] -= front[2] * dotproduct;
// normalize top again
tlen = sqrtf(top[0] * top[0] + top[1] * top[1] + top[2] * top[2]);
// tlen is guaranteed to be non-zero, otherwise error would have been larger than 0.5
top[0] *= (1.0f / tlen);
top[1] *= (1.0f / tlen);
top[2] *= (1.0f / tlen);
}
} else {
front[0] = 0.0f;
front[1] = 0.0f;
front[2] = 1.0f;
top[0] = 0.0f;
top[1] = 1.0f;
top[2] = 0.0f;
}
// Calculate right vector as front X top
float right[3] = { top[1] * front[2] - top[2] * front[1], top[2] * front[0] - top[0] * front[2],
top[0] * front[1] - top[1] * front[0] };
/*
qWarning("Front: %f %f %f", front[0], front[1], front[2]);
qWarning("Top: %f %f %f", top[0], top[1], top[2]);
qWarning("Right: %f %f %f", right[0], right[1], right[2]);
*/
// Rotate speakers to match orientation
for (unsigned int i = 0; i < iChannels; ++i) {
speaker[3 * i + 0] =
fSpeakers[3 * i + 0] * right[0] + fSpeakers[3 * i + 1] * top[0] + fSpeakers[3 * i + 2] * front[0];
speaker[3 * i + 1] =
fSpeakers[3 * i + 0] * right[1] + fSpeakers[3 * i + 1] * top[1] + fSpeakers[3 * i + 2] * front[1];
speaker[3 * i + 2] =
fSpeakers[3 * i + 0] * right[2] + fSpeakers[3 * i + 1] * top[2] + fSpeakers[3 * i + 2] * front[2];
}
validListener = true;
}
foreach (AudioOutputUser *aop, qlMix) {
// Iterate through all audio sources and mix them together into the output (or the intermediate array)
const float *RESTRICT pfBuffer = aop->pfBuffer;
float volumeAdjustment = 1;
// Check if the audio source is a user speaking (instead of a sample playback) and apply potential volume
// adjustments
AudioOutputSpeech *speech = qobject_cast< AudioOutputSpeech * >(aop);
if (speech) {
const ClientUser *user = speech->p;
volumeAdjustment *= user->getLocalVolumeAdjustments();
if (user->cChannel && ChannelListener::isListening(Global::get().uiSession, user->cChannel->iId)
&& (speech->ucFlags & SpeechFlags::Listen)) {
// We are receiving this audio packet only because we are listening to the channel
// the speaking user is in. Thus we receive the audio via our "listener proxy".
// Thus we'll apply the volume adjustment for our listener proxy as well
volumeAdjustment *= ChannelListener::getListenerLocalVolumeAdjustment(user->cChannel);
}
if (prioritySpeakerActive) {
if (user->tsState != Settings::Whispering && !user->bPrioritySpeaker) {
volumeAdjustment *= adjustFactor;
}
}
}
// If recording is enabled add the current audio source to the recording buffer
if (recorder) {
if (speech) {
if (speech->bStereo) {
// Mix down stereo to mono. TODO: stereo record support
// frame: for a stereo stream, the [LR] pair inside ...[LR]LRLRLR.... is a frame
for (unsigned int i = 0; i < frameCount; ++i) {
recbuff[i] += (pfBuffer[2 * i] / 2.0 + pfBuffer[2 * i + 1] / 2.0) * volumeAdjustment;
}
} else {
for (unsigned int i = 0; i < frameCount; ++i) {
recbuff[i] += pfBuffer[i] * volumeAdjustment;
}
}
if (!recorder->isInMixDownMode()) {
recorder->addBuffer(speech->p, recbuff, frameCount);
recbuff = boost::shared_array< float >(new float[frameCount]);
memset(recbuff.get(), 0, sizeof(float) * frameCount);
}
// Don't add the local audio to the real output
if (qobject_cast< RecordUser * >(speech->p)) {
continue;
}
}
}
if (validListener && ((aop->fPos[0] != 0.0f) || (aop->fPos[1] != 0.0f) || (aop->fPos[2] != 0.0f))) {
// Add position to position map
AudioOutputSpeech *speech = qobject_cast< AudioOutputSpeech * >(aop);
#ifdef USE_MANUAL_PLUGIN
if (speech) {
const ClientUser *user = speech->p;
// The coordinates in the plane are actually given by x and z instead of x and y (y is up)
positions.insert(user->uiSession, { aop->fPos[0], aop->fPos[2] });
}
#endif
// If positional audio is enabled, calculate the respective audio effect here
float dir[3] = { aop->fPos[0] - Global::get().p->fCameraPosition[0], aop->fPos[1] - Global::get().p->fCameraPosition[1],
aop->fPos[2] - Global::get().p->fCameraPosition[2] };
float len = sqrtf(dir[0] * dir[0] + dir[1] * dir[1] + dir[2] * dir[2]);
if (len > 0.0f) {
dir[0] /= len;
dir[1] /= len;
dir[2] /= len;
}
/*
qWarning("Voice pos: %f %f %f", aop->fPos[0], aop->fPos[1], aop->fPos[2]);
qWarning("Voice dir: %f %f %f", dir[0], dir[1], dir[2]);
*/
if (!aop->pfVolume) {
aop->pfVolume = new float[nchan];
for (unsigned int s = 0; s < nchan; ++s)
aop->pfVolume[s] = -1.0;
}
for (unsigned int s = 0; s < nchan; ++s) {
const float dot = bSpeakerPositional[s] ? dir[0] * speaker[s * 3 + 0] + dir[1] * speaker[s * 3 + 1]
+ dir[2] * speaker[s * 3 + 2]
: 1.0f;
const float str = svol[s] * calcGain(dot, len) * volumeAdjustment;
float *RESTRICT o = output + s;
const float old = (aop->pfVolume[s] >= 0.0f) ? aop->pfVolume[s] : str;
const float inc = (str - old) / static_cast< float >(frameCount);
aop->pfVolume[s] = str;
/*
qWarning("%d: Pos %f %f %f : Dot %f Len %f Str %f", s, speaker[s*3+0],
speaker[s*3+1], speaker[s*3+2], dot, len, str);
*/
if ((old >= 0.00000001f) || (str >= 0.00000001f)) {
for (unsigned int i = 0; i < frameCount; ++i) {
if (speech && speech->bStereo) {
// Mix stereo user's stream into mono
// frame: for a stereo stream, the [LR] pair inside ...[LR]LRLRLR.... is a frame
o[i * nchan] += (pfBuffer[2 * i] / 2.0 + pfBuffer[2 * i + 1] / 2.0)
* (old + inc * static_cast< float >(i));
} else {
o[i * nchan] += pfBuffer[i] * (old + inc * static_cast< float >(i));
}
}
}
}
} else {
// Mix the current audio source into the output by adding it to the elements of the output buffer after
// having applied a volume adjustment
for (unsigned int s = 0; s < nchan; ++s) {
const float str = svol[s] * volumeAdjustment;
float *RESTRICT o = output + s;
if (aop->bStereo) {
// Linear-panning stereo stream according to the projection of fSpeaker vector on left-right
// direction.
// frame: for a stereo stream, the [LR] pair inside ...[LR]LRLRLR.... is a frame
for (unsigned int i = 0; i < frameCount; ++i)
o[i * nchan] += (pfBuffer[2 * i] * fStereoPanningFactor[2 * s + 0]
+ pfBuffer[2 * i + 1] * fStereoPanningFactor[2 * s + 1])
* str;
} else {
for (unsigned int i = 0; i < frameCount; ++i)
o[i * nchan] += pfBuffer[i] * str;
}
}
}
}
if (recorder && recorder->isInMixDownMode()) {
recorder->addBuffer(nullptr, recbuff, frameCount);
}
// Clip the output audio
if (eSampleFormat == SampleFloat)
for (unsigned int i = 0; i < frameCount * iChannels; i++)
output[i] = qBound(-1.0f, output[i], 1.0f);
else
// Also convert the intermediate float array into an array of shorts before writing it to the outbuff
for (unsigned int i = 0; i < frameCount * iChannels; i++)
reinterpret_cast< short * >(outbuff)[i] =
static_cast< short >(qBound(-32768.f, (output[i] * 32768.f), 32767.f));
}
qrwlOutputs.unlock();
// Delete all AudioOutputUsers that no longer provide any new audio
foreach (AudioOutputUser *aop, qlDel)
removeBuffer(aop);
#ifdef USE_MANUAL_PLUGIN
Manual::setSpeakerPositions(positions);
#endif
// Return whether data has been written to the outbuff
return (!qlMix.isEmpty());
}
bool AudioOutput::isAlive() const {
return isRunning();
}
unsigned int AudioOutput::getMixerFreq() const {
return iMixerFreq;
}
void AudioOutput::setBufferSize(unsigned int bufferSize) {
iBufferSize = bufferSize;
}
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