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complete piping of stereo audio through mixer

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This commit is contained in:
Stephen Birarda 2014-06-06 10:55:04 -07:00
parent 681ce247d6
commit ae2f6a3cb6
8 changed files with 160 additions and 117 deletions
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242
assignment-client/src/audio/AudioMixer.cpp
View file

@ -173,134 +173,160 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
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) {
if (!bufferToAdd->isStereo()) {
// this is a mono buffer, which means it gets full attenuation and spatialization
// setup the int16_t variables for the two sample sets
correctBufferSample[0] = nextOutputStart[s / 2] * attenuationCoefficient;
correctBufferSample[1] = nextOutputStart[(s / 2) + 1] * attenuationCoefficient;
// 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;
delayedChannelIndex = s + (numSamplesDelay * 2) + delayedChannelOffset;
const int16_t* bufferStart = bufferToAdd->getBuffer();
int ringBufferSampleCapacity = bufferToAdd->getSampleCapacity();
delayBufferSample[0] = correctBufferSample[0] * weakChannelAmplitudeRatio;
delayBufferSample[1] = correctBufferSample[1] * weakChannelAmplitudeRatio;
int16_t correctBufferSample[2], delayBufferSample[2];
int delayedChannelIndex = 0;
__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]);
const int SINGLE_STEREO_OFFSET = 2;
// 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;
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];
}
int i = 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.
while (i + 3 < numSamplesDelay) {
// handle the first cases where we can MMX add four samples at once
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;
__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;
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];
}
}
} else {
// stereo buffer - do attenuation but no sample delay for spatialization
qDebug() << "Adding a stereo buffer";
int parentIndex = i * 2;
if (i + 2 < numSamplesDelay) {
// MMX add only three delayed samples
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 4) {
// use MMX to clamp four additions at a time
__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 bufferSamples = _mm_set_pi16(_clientSamples[s], _clientSamples[s + 1],
_clientSamples[s + 2], _clientSamples[s + 3]);
__m64 addSamples = _mm_set_pi16(nextOutputStart[s] * attenuationCoefficient,
nextOutputStart[s + 1] * attenuationCoefficient,
nextOutputStart[s + 2] * attenuationCoefficient,
nextOutputStart[s + 3] * attenuationCoefficient);
__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];
_clientSamples[s] = shortResults[3];
_clientSamples[s + 1] = shortResults[2];
_clientSamples[s + 2] = shortResults[1];
_clientSamples[s + 3] = shortResults[0];
}
}
}

14
assignment-client/src/audio/AudioMixerClientData.cpp
View file

@ -50,10 +50,22 @@ int AudioMixerClientData::parseData(const QByteArray& packet) {
// 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();
avatarRingBuffer = new AvatarAudioRingBuffer(isStereo);
_ringBuffers.push_back(avatarRingBuffer);
}

4
assignment-client/src/audio/AudioMixerClientData.h
View file

@ -24,14 +24,14 @@ public:
AudioMixerClientData();
~AudioMixerClientData();
const std::vector<PositionalAudioRingBuffer*> getRingBuffers() const { return _ringBuffers; }
const QList<PositionalAudioRingBuffer*> getRingBuffers() const { return _ringBuffers; }
AvatarAudioRingBuffer* getAvatarAudioRingBuffer() const;
int parseData(const QByteArray& packet);
void checkBuffersBeforeFrameSend(int jitterBufferLengthSamples);
void pushBuffersAfterFrameSend();
private:
std::vector<PositionalAudioRingBuffer*> _ringBuffers;
QList<PositionalAudioRingBuffer*> _ringBuffers;
};
#endif // hifi_AudioMixerClientData_h

4
assignment-client/src/audio/AvatarAudioRingBuffer.cpp
View file

@ -13,8 +13,8 @@
#include "AvatarAudioRingBuffer.h"
AvatarAudioRingBuffer::AvatarAudioRingBuffer() :
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Microphone) {
AvatarAudioRingBuffer::AvatarAudioRingBuffer(bool isStereo) :
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Microphone, isStereo) {
}

2
assignment-client/src/audio/AvatarAudioRingBuffer.h
View file

@ -18,7 +18,7 @@
class AvatarAudioRingBuffer : public PositionalAudioRingBuffer {
public:
AvatarAudioRingBuffer();
AvatarAudioRingBuffer(bool isStereo = false);
int parseData(const QByteArray& packet);
private:

6
interface/src/Audio.cpp
View file

@ -640,6 +640,9 @@ void Audio::handleAudioInput() {
}
char* currentPacketPtr = audioDataPacket + populatePacketHeader(audioDataPacket, packetType);
// set the mono/stereo byte
*currentPacketPtr++ = isStereo;
// memcpy the three float positions
memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
@ -649,9 +652,6 @@ void Audio::handleAudioInput() {
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
currentPacketPtr += sizeof(headOrientation);
// set the mono/stereo byte
*currentPacketPtr++ = isStereo;
nodeList->writeDatagram(audioDataPacket, numAudioBytes + leadingBytes, audioMixer);
Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO)

3
libraries/audio/src/PositionalAudioRingBuffer.cpp
View file

@ -41,6 +41,9 @@ int PositionalAudioRingBuffer::parseData(const QByteArray& packet) {
// skip the packet header (includes the source UUID)
int readBytes = numBytesForPacketHeader(packet);
// hop over the channel flag that has already been read in AudioMixerClientData
readBytes += sizeof(quint8);
// read the positional data
readBytes += parsePositionalData(packet.mid(readBytes));
if (packetTypeForPacket(packet) == PacketTypeSilentAudioFrame) {

2
libraries/audio/src/PositionalAudioRingBuffer.h
View file

@ -41,6 +41,8 @@ public:
bool shouldLoopbackForNode() const { return _shouldLoopbackForNode; }
bool isStereo() const { return _isStereo; }
PositionalAudioRingBuffer::Type getType() const { return _type; }
const glm::vec3& getPosition() const { return _position; }
const glm::quat& getOrientation() const { return _orientation; }