SubSeekableAudioStream::SubSeekableAudioStream(SeekableAudioStream *parent, const Timestamp start, const Timestamp end, DisposeAfterUse::Flag disposeAfterUse)
: _parent(parent), _disposeAfterUse(disposeAfterUse),
_start(convertTimeToStreamPos(start, getRate(), isStereo())),
_pos(0, getRate() * (isStereo() ? 2 : 1)),
_length(convertTimeToStreamPos(end - start, getRate(), isStereo())) {
assert(_length.totalNumberOfFrames() % (isStereo() ? 2 : 1) == 0);
_parent->seek(_start);
}
SubLoopingAudioStream::SubLoopingAudioStream(SeekableAudioStream *stream,
uint loops,
const Timestamp loopStart,
const Timestamp loopEnd,
DisposeAfterUse::Flag disposeAfterUse)
: _parent(stream), _disposeAfterUse(disposeAfterUse), _loops(loops),
_pos(0, getRate() * (isStereo() ? 2 : 1)),
_loopStart(convertTimeToStreamPos(loopStart, getRate(), isStereo())),
_loopEnd(convertTimeToStreamPos(loopEnd, getRate(), isStereo())),
_done(false) {
assert(loopStart < loopEnd);
if (!_parent->rewind())
_done = true;
}
EMISubLoopingAudioStream(Audio::SeekableAudioStream *stream, uint loops,
const Audio::Timestamp start,
const Audio::Timestamp loopStart,
const Audio::Timestamp loopEnd,
DisposeAfterUse::Flag disposeAfterUse = DisposeAfterUse::YES)
: _parent(stream, disposeAfterUse),
_pos(convertTimeToStreamPos(start, getRate(), isStereo())),
_loopStart(convertTimeToStreamPos(loopStart, getRate(), isStereo())),
_loopEnd(convertTimeToStreamPos(loopEnd, getRate(), isStereo())),
_done(false), _hasLooped(false) {
assert(loopStart < loopEnd);
if (!_parent->seek(_pos))
_done = true;
}
int SCXStream::readBuffer(int16 *buffer, const int numSamples) {
if (isStereo()) {
// Needs to be divisible by the channel count
assert((numSamples % 2) == 0);
// TODO: As per above, this probably should do more actual streaming
// Decode enough data from each channel
int samplesPerChannel = numSamples / 2;
int16 *leftSamples = new int16[samplesPerChannel];
int16 *rightSamples = new int16[samplesPerChannel];
int samplesDecodedLeft = _xaStreams[0]->readBuffer(leftSamples, samplesPerChannel);
int samplesDecodedRight = _xaStreams[1]->readBuffer(rightSamples, samplesPerChannel);
assert(samplesDecodedLeft == samplesDecodedRight);
// Now re-interleave the data
int samplesDecoded = 0;
int16 *leftSrc = leftSamples, *rightSrc = rightSamples;
for (; samplesDecoded < numSamples; samplesDecoded += 2) {
*buffer++ = *leftSrc++;
*buffer++ = *rightSrc++;
}
delete[] leftSamples;
delete[] rightSamples;
return samplesDecoded;
}
// Just read from the stream directly for mono
return _xaStreams[0]->readBuffer(buffer, numSamples);
}
void AudioMixerClientData::setupCodec(CodecPluginPointer codec, const QString& codecName) {
cleanupCodec(); // cleanup any previously allocated coders first
_codec = codec;
_selectedCodecName = codecName;
if (codec) {
_encoder = codec->createEncoder(AudioConstants::SAMPLE_RATE, AudioConstants::STEREO);
_decoder = codec->createDecoder(AudioConstants::SAMPLE_RATE, AudioConstants::MONO);
}
auto avatarAudioStream = getAvatarAudioStream();
if (avatarAudioStream) {
avatarAudioStream->setupCodec(codec, codecName, avatarAudioStream->isStereo() ? AudioConstants::STEREO : AudioConstants::MONO);
qCDebug(audio) << "setting AvatarAudioStream... codec:" << _selectedCodecName << "isStereo:" << avatarAudioStream->isStereo();
}
#if INJECTORS_SUPPORT_CODECS
// fixup codecs for any active injectors...
auto it = _audioStreams.begin();
while (it != _audioStreams.end()) {
SharedStreamPointer stream = it->second;
if (stream->getType() == PositionalAudioStream::Injector) {
stream->setupCodec(codec, codecName, stream->isStereo() ? AudioConstants::STEREO : AudioConstants::MONO);
}
++it;
}
#endif
}
void QueuingAudioStreamImpl::queueAudioStream(AudioStream *stream, DisposeAfterUse::Flag disposeAfterUse) {
assert(!_finished);
if ((stream->getRate() != getRate()) || (stream->isStereo() != isStereo()))
error("QueuingAudioStreamImpl::queueAudioStream: stream has mismatched parameters");
Common::StackLock lock(_mutex);
_queue.push(StreamHolder(stream, disposeAfterUse));
}
void AvatarManager::handleCollisionEvents(const CollisionEvents& collisionEvents) {
bool playedCollisionSound { false };
for (Collision collision : collisionEvents) {
// TODO: The plan is to handle MOTIONSTATE_TYPE_AVATAR, and then MOTIONSTATE_TYPE_MYAVATAR. As it is, other
// people's avatars will have an id that doesn't match any entities, and one's own avatar will have
// an id of null. Thus this code handles any collision in which one of the participating objects is
// my avatar. (Other user machines will make a similar analysis and inject sound for their collisions.)
if (collision.idA.isNull() || collision.idB.isNull()) {
auto myAvatar = getMyAvatar();
myAvatar->collisionWithEntity(collision);
if (!playedCollisionSound) {
playedCollisionSound = true;
auto collisionSound = myAvatar->getCollisionSound();
if (collisionSound) {
const auto characterController = myAvatar->getCharacterController();
const float avatarVelocityChange =
(characterController ? glm::length(characterController->getVelocityChange()) : 0.0f);
const float velocityChange = glm::length(collision.velocityChange) + avatarVelocityChange;
const float MIN_AVATAR_COLLISION_ACCELERATION = 2.4f; // walking speed
const bool isSound =
(collision.type == CONTACT_EVENT_TYPE_START) && (velocityChange > MIN_AVATAR_COLLISION_ACCELERATION);
if (!isSound) {
return; // No sense iterating for others. We only have one avatar.
}
// Your avatar sound is personal to you, so let's say the "mass" part of the kinetic energy is already accounted for.
const float energy = velocityChange * velocityChange;
const float COLLISION_ENERGY_AT_FULL_VOLUME = 10.0f;
const float energyFactorOfFull = fmin(1.0f, energy / COLLISION_ENERGY_AT_FULL_VOLUME);
// For general entity collisionSoundURL, playSound supports changing the pitch for the sound based on the size of the object,
// but most avatars are roughly the same size, so let's not be so fancy yet.
const float AVATAR_STRETCH_FACTOR = 1.0f;
_collisionInjectors.remove_if([](const AudioInjectorPointer& injector) { return !injector; });
static const int MAX_INJECTOR_COUNT = 3;
if (_collisionInjectors.size() < MAX_INJECTOR_COUNT) {
AudioInjectorOptions options;
options.stereo = collisionSound->isStereo();
options.position = myAvatar->getWorldPosition();
options.volume = energyFactorOfFull;
options.pitch = 1.0f / AVATAR_STRETCH_FACTOR;
auto injector = DependencyManager::get<AudioInjectorManager>()->playSound(collisionSound, options, true);
_collisionInjectors.emplace_back(injector);
}
}
}
}
}
}
bool SubSeekableAudioStream::seek(const Timestamp &where) {
_pos = convertTimeToStreamPos(where, getRate(), isStereo());
if (_pos > _length) {
_pos = _length;
return false;
}
if (_parent->seek(_pos + _start)) {
return true;
} else {
_pos = _length;
return false;
}
}
uint64_t SoundEmitter::getDuration() {
if (m_soundClip) {
//convert to milliseconds
double samplerate = static_cast<double>(getSampleRate()) / 1000.0;
double bitres = static_cast<double>(getBitResolution());
// convert to bits
double size = static_cast<double>(getDecodedLength()) * 8.0;
double stereo = (isStereo() ? 2.0 : 1.0);
double time = ( size / (samplerate * bitres) ) / stereo;
return static_cast<uint64_t>(time);
}
return 0;
}
void GlWindow::setViewBuffer(vrj::Viewport::View view)
{
if(!isStereo())
{
glDrawBuffer(GL_BACK);
}
else if(Viewport::LEFT_EYE == view)
{
glDrawBuffer(GL_BACK_LEFT);
}
else if(Viewport::RIGHT_EYE == view)
{
glDrawBuffer(GL_BACK_RIGHT);
}
}
bool RawStream<is16Bit, isUnsigned, isLE>::seek(const Timestamp &where) {
_endOfData = true;
if (where > _playtime)
return false;
const uint32 seekSample = convertTimeToStreamPos(where, getRate(), isStereo()).totalNumberOfFrames();
_stream->seek(seekSample * (is16Bit ? 2 : 1), SEEK_SET);
// In case of an error we will not continue stream playback.
if (!_stream->err() && !_stream->eos() && _stream->pos() != _stream->size())
_endOfData = false;
return true;
}
void JitterSample::run(const render::RenderContextPointer& renderContext) {
auto& current = _sampleSequence.currentIndex;
if (!_freeze) {
if (current >= 0) {
current = (current + 1) % SEQUENCE_LENGTH;
} else {
current = -1;
}
}
auto args = renderContext->args;
auto viewFrustum = args->getViewFrustum();
auto jit = _sampleSequence.offsets[(current < 0 ? SEQUENCE_LENGTH : current)];
auto width = (float)args->_viewport.z;
auto height = (float)args->_viewport.w;
auto jx = 2.0f * jit.x / width;
auto jy = 2.0f * jit.y / height;
if (!args->isStereo()) {
auto projMat = viewFrustum.getProjection();
projMat[2][0] += jx;
projMat[2][1] += jy;
viewFrustum.setProjection(projMat);
viewFrustum.calculate();
args->pushViewFrustum(viewFrustum);
} else {
mat4 projMats[2];
args->_context->getStereoProjections(projMats);
jx *= 2.0f;
for (int i = 0; i < 2; i++) {
auto& projMat = projMats[i];
projMat[2][0] += jx;
projMat[2][1] += jy;
}
args->_context->setStereoProjections(projMats);
}
}
Sample* Wav::makeSample(int chan)
{
if(!isok)
{
DERROR("not loaded");
return NULL;
}
Sample* samp;
if(chan==EXTRACT_ALL)
{
samp = new Sample(head.nbytes/2);
memcpy(samp->data, bytes, head.nbytes);
}
else
{
DASSERTP(isStereo(),"(W): trying to extract single channel from mono!");
samp = new Sample(head.nbytes/4);
short* dp = (short*)bytes;
if(chan==EXTRACT_LEFT)
{
for(int i=0;i<head.nbytes/4;i++)
{
samp->data[i] = dp[i<<1];
}
}
if(chan==EXTRACT_RIGHT)
{
for(int i=0;i<head.nbytes/4;i++)
{
samp->data[i] = dp[ 1+ (i<<1)];
}
}
}
return samp;
}
bool SCXStream::rewind() {
if (!_xaStreams[0]->rewind())
return false;
return !isStereo() || _xaStreams[1]->rewind();
}
SCXStream::SCXStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse) {
static const uint32 stereoChannelNames[MAX_CHANNELS] = { MKTAG('L', 'E', 'F', 'T'), MKTAG('R', 'G', 'H', 'T') };
stream->readUint32BE(); // 'SCRX'
stream->readUint32LE();
_blockSize = stream->readUint16LE();
/* totalBlockSize = */ stream->readUint16LE();
if (_blockSize & 0xf)
error("Bad SCX block size %04x", _blockSize);
// Base our channel count based off the block size
_channels = (_blockSize == 0) ? 1 : 2;
stream->skip(12);
uint32 channelSize[MAX_CHANNELS];
for (int i = 0; i < _channels; i++) {
uint32 tag = stream->readUint32BE();
if (isStereo()) {
if (tag != stereoChannelNames[i])
error("Bad stereo channel tag found '%s'", tag2str(tag));
} else if (tag != MKTAG('M', 'O', 'N', 'O'))
error("Bad mono channel tag found '%s'", tag2str(tag));
channelSize[i] = stream->readUint32LE();
}
stream->seek(0x80);
uint32 leftRate = 0, rightRate = 0;
for (int i = 0; i < _channels; i++) {
if (stream->readUint32BE() != MKTAG('V', 'A', 'G', 'p'))
error("Bad VAG header");
/* uint32 version = */ stream->readUint32BE();
stream->readUint32BE();
stream->readUint32BE();
if (i == 0)
leftRate = stream->readUint32BE();
else
rightRate = stream->readUint32BE();
stream->skip(12); // skip useless info
stream->skip(16); // skip name
stream->skip(16); // skip zeroes
}
if (isStereo() && leftRate != rightRate)
error("Mismatching SCX rates");
_rate = leftRate;
if (isStereo()) {
// TODO: Make XAStream allow for appending data (similar to how ScummVM
// handles AAC/QDM2. For now, we de-interleave the XA ADPCM data and then
// re-interleave in readBuffer().
// Of course, in doing something that does better streaming, it would
// screw up the XA loop points. So, I'm not really sure what is best atm.
byte *leftOut = new byte[channelSize[0]];
byte *rightOut = new byte[channelSize[1]];
Common::MemoryWriteStream *leftStream = new Common::MemoryWriteStream(leftOut, channelSize[0]);
Common::MemoryWriteStream *rightStream = new Common::MemoryWriteStream(rightOut, channelSize[1]);
byte *buf = new byte[_blockSize];
while (stream->pos() < stream->size()) {
stream->read(buf, _blockSize);
leftStream->write(buf, _blockSize);
stream->read(buf, _blockSize);
rightStream->write(buf, _blockSize);
}
_xaStreams[0] = Audio::makeXAStream(new Common::MemoryReadStream(leftOut, channelSize[0], DisposeAfterUse::YES), _rate);
_xaStreams[1] = Audio::makeXAStream(new Common::MemoryReadStream(rightOut, channelSize[1], DisposeAfterUse::YES), _rate);
delete[] buf;
delete leftStream;
delete rightStream;
} else {
_xaStreams[0] = Audio::makeXAStream(stream->readStream(channelSize[0]), _rate);
_xaStreams[1] = 0;
}
if (disposeAfterUse == DisposeAfterUse::YES)
delete stream;
}
void Tool::play(char **argv)
{
AudioDevice *dev;
PlayStream playfile;
const char *path = *argv;
Linear buffer;
Info info;
unsigned bufcount, pages;
dev = getDevice();
if(!hasDevice() && !dev)
{
cerr << "no device supported" << endl;
exit(-1);
}
else if(!dev)
{
cerr << "device unavailable" << endl;
exit(-1);
}
playfile.open(argv);
if(!playfile.isOpen())
{
cerr << "audiotool: " << path << ": unable to access" << endl;
exit(-1);
}
if(!playfile.isStreamable())
{
cerr << "audiotool: " << path << ": missing needed codec" << endl;
exit(-1);
}
playfile.getInfo(&info);
if(!dev->setAudio((Rate)info.rate, isStereo(info.encoding), 10))
{
cerr << "audiotool: sound device does not support rate" << endl;
exit(-1);
}
bufcount = playfile.getCount();
if(isStereo(info.encoding))
buffer = new Sample[bufcount * 2];
else
buffer = new Sample[bufcount];
for(;;)
{
if(isStereo(info.encoding))
pages = playfile.getStereo(buffer, 1);
else
pages = playfile.getMono(buffer, 1);
if(!pages)
break;
dev->putSamples(buffer, bufcount);
}
dev->sync();
delete dev;
playfile.close();
exit(0);
}
void Tremolo::process( AudioBuffer* sampleBuffer, bool isMonoSource )
{
int bufferSize = sampleBuffer->bufferSize;
bool doStereo = ( sampleBuffer->amountOfChannels > 1 ) && isStereo();
SAMPLE_TYPE* envelopeTable;
SAMPLE_TYPE volume;
for ( int c = 0, ca = sampleBuffer->amountOfChannels; c < ca; ++c )
{
envelopeTable = getTableForChannel( c );
SAMPLE_TYPE* channelBuffer = sampleBuffer->getBufferForChannel( c );
bool useLeft = !doStereo || c % 2 == 0;
for ( int i = 0; i < bufferSize; ++i )
{
if ( useLeft )
{
if ( _leftState == 0 )
{
if (( _leftTableIndex += _leftAttackIncr ) >= ENVELOPE_PRECISION )
{
_leftTableIndex = ENVELOPE_PRECISION - 1;
_leftState = 1;
}
}
else if ( _leftState == 1 )
{
if (( _leftTableIndex -= _leftDecayIncr ) <= 0 )
{
_leftTableIndex = 0;
// optional: hold state
// if resulting volume is now smaller than sustain amp > increment _leftState
_leftState = 0;
}
}
volume = envelopeTable[( int ) _leftTableIndex ];
}
else {
// right channel
if ( _rightState == 0 )
{
if (( _rightTableIndex += _rightAttackIncr ) >= ENVELOPE_PRECISION )
{
_rightTableIndex = ENVELOPE_PRECISION - 1;
_rightState = 1;
}
}
else if ( _rightState == 1 )
{
if (( _rightTableIndex -= _rightDecayIncr ) <= 0 )
{
_rightTableIndex = 0;
// optional: hold state
// if resulting volume is now smaller than sustain amp > increment _rightState
_rightState = 0;
}
}
volume = envelopeTable[( int ) _rightTableIndex ];
}
channelBuffer[ i ] *= volume;
}
// save CPU cycles when source and output are mono
if ( isMonoSource && !doStereo )
{
sampleBuffer->applyMonoSource();
break;
}
}
}
