int AudioMixerClientData::parseData(const QByteArray& packet) {
PacketType packetType = packetTypeForPacket(packet);
if (packetType == PacketTypeMicrophoneAudioWithEcho
|| packetType == PacketTypeMicrophoneAudioNoEcho
|| packetType == PacketTypeSilentAudioFrame) {
// grab the AvatarAudioRingBuffer from the vector (or create it if it doesn't exist)
AvatarAudioRingBuffer* avatarRingBuffer = getAvatarAudioRingBuffer();
// read the first byte after the header to see if this is a stereo or mono buffer
quint8 channelFlag = packet.at(numBytesForPacketHeader(packet));
bool isStereo = channelFlag == 1;
if (avatarRingBuffer && avatarRingBuffer->isStereo() != isStereo) {
// there's a mismatch in the buffer channels for the incoming and current buffer
// so delete our current buffer and create a new one
_ringBuffers.removeOne(avatarRingBuffer);
avatarRingBuffer->deleteLater();
avatarRingBuffer = NULL;
}
if (!avatarRingBuffer) {
// we don't have an AvatarAudioRingBuffer yet, so add it
avatarRingBuffer = new AvatarAudioRingBuffer(isStereo);
_ringBuffers.push_back(avatarRingBuffer);
}
// ask the AvatarAudioRingBuffer instance to parse the data
avatarRingBuffer->parseData(packet);
} else {
// this is injected audio
// grab the stream identifier for this injected audio
QUuid streamIdentifier = QUuid::fromRfc4122(packet.mid(numBytesForPacketHeader(packet), NUM_BYTES_RFC4122_UUID));
InjectedAudioRingBuffer* matchingInjectedRingBuffer = NULL;
for (unsigned int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->getType() == PositionalAudioRingBuffer::Injector
&& ((InjectedAudioRingBuffer*) _ringBuffers[i])->getStreamIdentifier() == streamIdentifier) {
matchingInjectedRingBuffer = (InjectedAudioRingBuffer*) _ringBuffers[i];
}
}
if (!matchingInjectedRingBuffer) {
// we don't have a matching injected audio ring buffer, so add it
matchingInjectedRingBuffer = new InjectedAudioRingBuffer(streamIdentifier);
_ringBuffers.push_back(matchingInjectedRingBuffer);
}
matchingInjectedRingBuffer->parseData(packet);
}
return 0;
}
int AudioMixerClientData::parseData(const QByteArray& packet) {
PacketType packetType = packetTypeForPacket(packet);
if (packetType == PacketTypeMicrophoneAudioWithEcho
|| packetType == PacketTypeMicrophoneAudioNoEcho
|| packetType == PacketTypeSilentAudioFrame) {
// grab the AvatarAudioRingBuffer from the vector (or create it if it doesn't exist)
AvatarAudioRingBuffer* avatarRingBuffer = getAvatarAudioRingBuffer();
if (!avatarRingBuffer) {
// we don't have an AvatarAudioRingBuffer yet, so add it
avatarRingBuffer = new AvatarAudioRingBuffer();
_ringBuffers.push_back(avatarRingBuffer);
}
// ask the AvatarAudioRingBuffer instance to parse the data
avatarRingBuffer->parseData(packet);
} else {
// this is injected audio
// grab the stream identifier for this injected audio
QUuid streamIdentifier = QUuid::fromRfc4122(packet.mid(numBytesForPacketHeader(packet), NUM_BYTES_RFC4122_UUID));
InjectedAudioRingBuffer* matchingInjectedRingBuffer = NULL;
for (unsigned int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->getType() == PositionalAudioRingBuffer::Injector
&& ((InjectedAudioRingBuffer*) _ringBuffers[i])->getStreamIdentifier() == streamIdentifier) {
matchingInjectedRingBuffer = (InjectedAudioRingBuffer*) _ringBuffers[i];
}
}
if (!matchingInjectedRingBuffer) {
// we don't have a matching injected audio ring buffer, so add it
matchingInjectedRingBuffer = new InjectedAudioRingBuffer(streamIdentifier);
_ringBuffers.push_back(matchingInjectedRingBuffer);
}
matchingInjectedRingBuffer->parseData(packet);
}
return 0;
}
void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
AvatarAudioRingBuffer* listeningNodeBuffer) {
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
const int PHASE_DELAY_AT_90 = 20;
if (bufferToAdd != listeningNodeBuffer) {
// if the two buffer pointers do not match then these are different buffers
glm::vec3 listenerPosition = listeningNodeBuffer->getPosition();
glm::vec3 relativePosition = bufferToAdd->getPosition() - listeningNodeBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(listeningNodeBuffer->getOrientation());
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
if (bufferToAdd->getType() == PositionalAudioRingBuffer::Injector) {
InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) bufferToAdd;
radius = injectedBuffer->getRadius();
attenuationCoefficient *= injectedBuffer->getAttenuationRatio();
}
if (radius == 0 || (distanceSquareToSource > radius * radius)) {
// this is either not a spherical source, or the listener is outside the sphere
if (radius > 0) {
// this is a spherical source - the distance used for the coefficient
// needs to be the closest point on the boundary to the source
// ovveride the distance to the node with the distance to the point on the
// boundary of the sphere
distanceSquareToSource -= (radius * radius);
} else {
// calculate the angle delivery for off-axis attenuation
glm::vec3 rotatedListenerPosition = glm::inverse(bufferToAdd->getOrientation()) * relativePosition;
float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedListenerPosition));
const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / 90.0f));
// multiply the current attenuation coefficient by the calculated off axis coefficient
attenuationCoefficient *= offAxisCoefficient;
}
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
const float DISTANCE_SCALE = 2.5f;
const float GEOMETRIC_AMPLITUDE_SCALAR = 0.3f;
const float DISTANCE_LOG_BASE = 2.5f;
const float DISTANCE_SCALE_LOG = logf(DISTANCE_SCALE) / logf(DISTANCE_LOG_BASE);
// calculate the distance coefficient using the distance to this node
float distanceCoefficient = powf(GEOMETRIC_AMPLITUDE_SCALAR,
DISTANCE_SCALE_LOG +
(0.5f * logf(distanceSquareToSource) / logf(DISTANCE_LOG_BASE)) - 1);
distanceCoefficient = std::min(1.0f, distanceCoefficient);
// multiply the current attenuation coefficient by the distance coefficient
attenuationCoefficient *= distanceCoefficient;
// project the rotated source position vector onto the XZ plane
rotatedSourcePosition.y = 0.0f;
// produce an oriented angle about the y-axis
bearingRelativeAngleToSource = glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedSourcePosition),
glm::vec3(0.0f, 1.0f, 0.0f));
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
// figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization
float sinRatio = fabsf(sinf(glm::radians(bearingRelativeAngleToSource)));
numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
}
}
// if the bearing relative angle to source is > 0 then the delayed channel is the right one
int delayedChannelOffset = (bearingRelativeAngleToSource > 0.0f) ? 1 : 0;
int goodChannelOffset = delayedChannelOffset == 0 ? 1 : 0;
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 2) {
if ((s / 2) < numSamplesDelay) {
// pull the earlier sample for the delayed channel
int earlierSample = (*bufferToAdd)[(s / 2) - numSamplesDelay] * attenuationCoefficient * weakChannelAmplitudeRatio;
_clientSamples[s + delayedChannelOffset] = glm::clamp(_clientSamples[s + delayedChannelOffset] + earlierSample,
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
}
// pull the current sample for the good channel
int16_t currentSample = (*bufferToAdd)[s / 2] * attenuationCoefficient;
_clientSamples[s + goodChannelOffset] = glm::clamp(_clientSamples[s + goodChannelOffset] + currentSample,
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
if ((s / 2) + numSamplesDelay < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {
// place the current sample at the right spot in the delayed channel
int16_t clampedSample = glm::clamp((int) (_clientSamples[s + (numSamplesDelay * 2) + delayedChannelOffset]
+ (currentSample * weakChannelAmplitudeRatio)),
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
_clientSamples[s + (numSamplesDelay * 2) + delayedChannelOffset] = clampedSample;
}
}
}
int AudioMixerClientData::parseData(const QByteArray& packet) {
// parse sequence number for this packet
int numBytesPacketHeader = numBytesForPacketHeader(packet);
const char* sequenceAt = packet.constData() + numBytesPacketHeader;
quint16 sequence = *(reinterpret_cast<const quint16*>(sequenceAt));
PacketType packetType = packetTypeForPacket(packet);
if (packetType == PacketTypeMicrophoneAudioWithEcho
|| packetType == PacketTypeMicrophoneAudioNoEcho
|| packetType == PacketTypeSilentAudioFrame) {
_incomingAvatarAudioSequenceNumberStats.sequenceNumberReceived(sequence);
// grab the AvatarAudioRingBuffer from the vector (or create it if it doesn't exist)
AvatarAudioRingBuffer* avatarRingBuffer = getAvatarAudioRingBuffer();
// read the first byte after the header to see if this is a stereo or mono buffer
quint8 channelFlag = packet.at(numBytesForPacketHeader(packet) + sizeof(quint16));
bool isStereo = channelFlag == 1;
if (avatarRingBuffer && avatarRingBuffer->isStereo() != isStereo) {
// there's a mismatch in the buffer channels for the incoming and current buffer
// so delete our current buffer and create a new one
_ringBuffers.removeOne(avatarRingBuffer);
avatarRingBuffer->deleteLater();
avatarRingBuffer = NULL;
}
if (!avatarRingBuffer) {
// we don't have an AvatarAudioRingBuffer yet, so add it
avatarRingBuffer = new AvatarAudioRingBuffer(isStereo, AudioMixer::getUseDynamicJitterBuffers());
_ringBuffers.push_back(avatarRingBuffer);
}
// ask the AvatarAudioRingBuffer instance to parse the data
avatarRingBuffer->parseData(packet);
} else if (packetType == PacketTypeInjectAudio) {
// this is injected audio
// grab the stream identifier for this injected audio
QUuid streamIdentifier = QUuid::fromRfc4122(packet.mid(numBytesForPacketHeader(packet) + sizeof(quint16), NUM_BYTES_RFC4122_UUID));
if (!_incomingInjectedAudioSequenceNumberStatsMap.contains(streamIdentifier)) {
_incomingInjectedAudioSequenceNumberStatsMap.insert(streamIdentifier, SequenceNumberStats(INCOMING_SEQ_STATS_HISTORY_LENGTH));
}
_incomingInjectedAudioSequenceNumberStatsMap[streamIdentifier].sequenceNumberReceived(sequence);
InjectedAudioRingBuffer* matchingInjectedRingBuffer = NULL;
for (int i = 0; i < _ringBuffers.size(); i++) {
if (_ringBuffers[i]->getType() == PositionalAudioRingBuffer::Injector
&& ((InjectedAudioRingBuffer*) _ringBuffers[i])->getStreamIdentifier() == streamIdentifier) {
matchingInjectedRingBuffer = (InjectedAudioRingBuffer*) _ringBuffers[i];
}
}
if (!matchingInjectedRingBuffer) {
// we don't have a matching injected audio ring buffer, so add it
matchingInjectedRingBuffer = new InjectedAudioRingBuffer(streamIdentifier, AudioMixer::getUseDynamicJitterBuffers());
_ringBuffers.push_back(matchingInjectedRingBuffer);
}
matchingInjectedRingBuffer->parseData(packet);
} else if (packetType == PacketTypeAudioStreamStats) {
const char* dataAt = packet.data();
// skip over header, appendFlag, and num stats packed
dataAt += (numBytesPacketHeader + sizeof(quint8) + sizeof(quint16));
// read the downstream audio stream stats
memcpy(&_downstreamAudioStreamStats, dataAt, sizeof(AudioStreamStats));
}
return 0;
}
void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuffer* bufferToAdd,
AvatarAudioRingBuffer* listeningNodeBuffer) {
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f;
if (bufferToAdd != listeningNodeBuffer) {
// if the two buffer pointers do not match then these are different buffers
glm::vec3 relativePosition = bufferToAdd->getPosition() - listeningNodeBuffer->getPosition();
glm::quat inverseOrientation = glm::inverse(listeningNodeBuffer->getOrientation());
float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
float radius = 0.0f;
if (bufferToAdd->getType() == PositionalAudioRingBuffer::Injector) {
InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) bufferToAdd;
radius = injectedBuffer->getRadius();
attenuationCoefficient *= injectedBuffer->getAttenuationRatio();
}
if (radius == 0 || (distanceSquareToSource > radius * radius)) {
// this is either not a spherical source, or the listener is outside the sphere
if (radius > 0) {
// this is a spherical source - the distance used for the coefficient
// needs to be the closest point on the boundary to the source
// ovveride the distance to the node with the distance to the point on the
// boundary of the sphere
distanceSquareToSource -= (radius * radius);
} else {
// calculate the angle delivery for off-axis attenuation
glm::vec3 rotatedListenerPosition = glm::inverse(bufferToAdd->getOrientation()) * relativePosition;
float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedListenerPosition));
const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / PI_OVER_TWO));
// multiply the current attenuation coefficient by the calculated off axis coefficient
attenuationCoefficient *= offAxisCoefficient;
}
glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
const float DISTANCE_SCALE = 2.5f;
const float GEOMETRIC_AMPLITUDE_SCALAR = 0.3f;
const float DISTANCE_LOG_BASE = 2.5f;
const float DISTANCE_SCALE_LOG = logf(DISTANCE_SCALE) / logf(DISTANCE_LOG_BASE);
// calculate the distance coefficient using the distance to this node
float distanceCoefficient = powf(GEOMETRIC_AMPLITUDE_SCALAR,
DISTANCE_SCALE_LOG +
(0.5f * logf(distanceSquareToSource) / logf(DISTANCE_LOG_BASE)) - 1);
distanceCoefficient = std::min(1.0f, distanceCoefficient);
// multiply the current attenuation coefficient by the distance coefficient
attenuationCoefficient *= distanceCoefficient;
// project the rotated source position vector onto the XZ plane
rotatedSourcePosition.y = 0.0f;
// produce an oriented angle about the y-axis
bearingRelativeAngleToSource = glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f),
glm::normalize(rotatedSourcePosition),
glm::vec3(0.0f, 1.0f, 0.0f));
const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
// figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization
float sinRatio = fabsf(sinf(bearingRelativeAngleToSource));
numSamplesDelay = SAMPLE_PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
}
}
// if the bearing relative angle to source is > 0 then the delayed channel is the right one
int delayedChannelOffset = (bearingRelativeAngleToSource > 0.0f) ? 1 : 0;
int goodChannelOffset = delayedChannelOffset == 0 ? 1 : 0;
const int16_t* nextOutputStart = bufferToAdd->getNextOutput();
const int16_t* bufferStart = bufferToAdd->getBuffer();
int ringBufferSampleCapacity = bufferToAdd->getSampleCapacity();
int16_t correctBufferSample[2], delayBufferSample[2];
int delayedChannelIndex = 0;
const int SINGLE_STEREO_OFFSET = 2;
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 4) {
// setup the int16_t variables for the two sample sets
correctBufferSample[0] = nextOutputStart[s / 2] * attenuationCoefficient;
correctBufferSample[1] = nextOutputStart[(s / 2) + 1] * attenuationCoefficient;
delayedChannelIndex = s + (numSamplesDelay * 2) + delayedChannelOffset;
delayBufferSample[0] = correctBufferSample[0] * weakChannelAmplitudeRatio;
delayBufferSample[1] = correctBufferSample[1] * weakChannelAmplitudeRatio;
__m64 bufferSamples = _mm_set_pi16(_clientSamples[s + goodChannelOffset],
_clientSamples[s + goodChannelOffset + SINGLE_STEREO_OFFSET],
_clientSamples[delayedChannelIndex],
_clientSamples[delayedChannelIndex + SINGLE_STEREO_OFFSET]);
__m64 addedSamples = _mm_set_pi16(correctBufferSample[0], correctBufferSample[1],
delayBufferSample[0], delayBufferSample[1]);
// perform the MMX add (with saturation) of two correct and delayed samples
__m64 mmxResult = _mm_adds_pi16(bufferSamples, addedSamples);
int16_t* shortResults = reinterpret_cast<int16_t*>(&mmxResult);
// assign the results from the result of the mmx arithmetic
_clientSamples[s + goodChannelOffset] = shortResults[3];
_clientSamples[s + goodChannelOffset + SINGLE_STEREO_OFFSET] = shortResults[2];
_clientSamples[delayedChannelIndex] = shortResults[1];
_clientSamples[delayedChannelIndex + SINGLE_STEREO_OFFSET] = shortResults[0];
}
// The following code is pretty gross and redundant, but AFAIK it's the best way to avoid
// too many conditionals in handling the delay samples at the beginning of _clientSamples.
// Basically we try to take the samples in batches of four, and then handle the remainder
// conditionally to get rid of the rest.
const int DOUBLE_STEREO_OFFSET = 4;
const int TRIPLE_STEREO_OFFSET = 6;
if (numSamplesDelay > 0) {
// if there was a sample delay for this buffer, we need to pull samples prior to the nextOutput
// to stick at the beginning
float attenuationAndWeakChannelRatio = attenuationCoefficient * weakChannelAmplitudeRatio;
const int16_t* delayNextOutputStart = nextOutputStart - numSamplesDelay;
if (delayNextOutputStart < bufferStart) {
delayNextOutputStart = bufferStart + ringBufferSampleCapacity - numSamplesDelay;
}
int i = 0;
while (i + 3 < numSamplesDelay) {
// handle the first cases where we can MMX add four samples at once
int parentIndex = i * 2;
__m64 bufferSamples = _mm_set_pi16(_clientSamples[parentIndex + delayedChannelOffset],
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset],
_clientSamples[parentIndex + DOUBLE_STEREO_OFFSET + delayedChannelOffset],
_clientSamples[parentIndex + TRIPLE_STEREO_OFFSET + delayedChannelOffset]);
__m64 addSamples = _mm_set_pi16(delayNextOutputStart[i] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 1] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 2] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 3] * attenuationAndWeakChannelRatio);
__m64 mmxResult = _mm_adds_pi16(bufferSamples, addSamples);
int16_t* shortResults = reinterpret_cast<int16_t*>(&mmxResult);
_clientSamples[parentIndex + delayedChannelOffset] = shortResults[3];
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[2];
_clientSamples[parentIndex + DOUBLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[1];
_clientSamples[parentIndex + TRIPLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[0];
// push the index
i += 4;
}
int parentIndex = i * 2;
if (i + 2 < numSamplesDelay) {
// MMX add only three delayed samples
__m64 bufferSamples = _mm_set_pi16(_clientSamples[parentIndex + delayedChannelOffset],
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset],
_clientSamples[parentIndex + DOUBLE_STEREO_OFFSET + delayedChannelOffset],
0);
__m64 addSamples = _mm_set_pi16(delayNextOutputStart[i] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 1] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 2] * attenuationAndWeakChannelRatio,
0);
__m64 mmxResult = _mm_adds_pi16(bufferSamples, addSamples);
int16_t* shortResults = reinterpret_cast<int16_t*>(&mmxResult);
_clientSamples[parentIndex + delayedChannelOffset] = shortResults[3];
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[2];
_clientSamples[parentIndex + DOUBLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[1];
} else if (i + 1 < numSamplesDelay) {
// MMX add two delayed samples
__m64 bufferSamples = _mm_set_pi16(_clientSamples[parentIndex + delayedChannelOffset],
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset], 0, 0);
__m64 addSamples = _mm_set_pi16(delayNextOutputStart[i] * attenuationAndWeakChannelRatio,
delayNextOutputStart[i + 1] * attenuationAndWeakChannelRatio, 0, 0);
__m64 mmxResult = _mm_adds_pi16(bufferSamples, addSamples);
int16_t* shortResults = reinterpret_cast<int16_t*>(&mmxResult);
_clientSamples[parentIndex + delayedChannelOffset] = shortResults[3];
_clientSamples[parentIndex + SINGLE_STEREO_OFFSET + delayedChannelOffset] = shortResults[2];
} else if (i < numSamplesDelay) {
// MMX add a single delayed sample
__m64 bufferSamples = _mm_set_pi16(_clientSamples[parentIndex + delayedChannelOffset], 0, 0, 0);
__m64 addSamples = _mm_set_pi16(delayNextOutputStart[i] * attenuationAndWeakChannelRatio, 0, 0, 0);
__m64 mmxResult = _mm_adds_pi16(bufferSamples, addSamples);
int16_t* shortResults = reinterpret_cast<int16_t*>(&mmxResult);
_clientSamples[parentIndex + delayedChannelOffset] = shortResults[3];
}
}
}
