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more MMX optimizations to AudioMixer for delayed samples

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This commit is contained in:
Stephen Birarda 2014-03-17 10:02:00 -07:00
parent 72449fdb6a
commit 05d6522279
2 changed files with 103 additions and 35 deletions
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133
assignment-client/src/audio/AudioMixer.cpp
View file

@ -74,9 +74,7 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
float attenuationCoefficient = 1.0f; float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0; int numSamplesDelay = 0;
float weakChannelAmplitudeRatio = 1.0f; float weakChannelAmplitudeRatio = 1.0f;
const int PHASE_DELAY_AT_90 = 20;
if (bufferToAdd != listeningNodeBuffer) { if (bufferToAdd != listeningNodeBuffer) {
// if the two buffer pointers do not match then these are different buffers // if the two buffer pointers do not match then these are different buffers
@ -150,7 +148,7 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
// figure out the number of samples of delay and the ratio of the amplitude // figure out the number of samples of delay and the ratio of the amplitude
// in the weak channel for audio spatialization // in the weak channel for audio spatialization
float sinRatio = fabsf(sinf(bearingRelativeAngleToSource)); float sinRatio = fabsf(sinf(bearingRelativeAngleToSource));
numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio; numSamplesDelay = SAMPLE_PHASE_DELAY_AT_90 * sinRatio;
weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio); weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
} }
} }
@ -160,45 +158,28 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
int goodChannelOffset = delayedChannelOffset == 0 ? 1 : 0; int goodChannelOffset = delayedChannelOffset == 0 ? 1 : 0;
const int16_t* nextOutputStart = bufferToAdd->getNextOutput(); const int16_t* nextOutputStart = bufferToAdd->getNextOutput();
const int16_t* delayNextOutputStart = nextOutputStart - numSamplesDelay;
const int16_t* bufferStart = bufferToAdd->getBuffer(); const int16_t* bufferStart = bufferToAdd->getBuffer();
int ringBufferSampleCapacity = bufferToAdd->getSampleCapacity(); int ringBufferSampleCapacity = bufferToAdd->getSampleCapacity();
if (delayNextOutputStart < bufferStart) {
delayNextOutputStart = bufferStart + ringBufferSampleCapacity - numSamplesDelay;
}
int16_t correctMixSample[2], correctBufferSample[2], delayMixSample[2], delayBufferSample[2]; int16_t correctBufferSample[2], delayBufferSample[2];
int delayedChannelIndex[2]; int delayedChannelIndex = 0;
float attenuationAndWeakChannelRatio = attenuationCoefficient * weakChannelAmplitudeRatio;
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 4) { for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 4) {
// setup the int16_t variables for the two sample sets // setup the int16_t variables for the two sample sets
correctBufferSample[0] = nextOutputStart[s / 2] * attenuationCoefficient; correctBufferSample[0] = nextOutputStart[s / 2] * attenuationCoefficient;
correctBufferSample[1] = nextOutputStart[(s / 2) + 1] * attenuationCoefficient; correctBufferSample[1] = nextOutputStart[(s / 2) + 1] * attenuationCoefficient;
correctMixSample[0] = _clientSamples[s + goodChannelOffset];
correctMixSample[1] = _clientSamples[s + goodChannelOffset + 2];
for (int i = 0; i < 2; ++i) { delayedChannelIndex = s + (numSamplesDelay * 2) + delayedChannelOffset;
if ((s / 2) + numSamplesDelay + i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {
// for the delayed channel we fill the range (n + numSamplesDelay) to NETWORK_BUFFER_LENGTH_SAMPLES_STEREO first
delayedChannelIndex[i] = s + (numSamplesDelay * 2) + (i * 2) + delayedChannelOffset;
delayBufferSample[i] = correctBufferSample[i] * weakChannelAmplitudeRatio;
} else {
// now that the right most range has been filled, we go back to fill in numSamples delay at the beginning
int samplesBack = (s / 2) - NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL + i;
delayBufferSample[i] = delayNextOutputStart[numSamplesDelay + samplesBack] * attenuationAndWeakChannelRatio;
delayedChannelIndex[i] = (numSamplesDelay + samplesBack) * 2 + delayedChannelOffset;
}
}
delayMixSample[0] = _clientSamples[delayedChannelIndex[0]]; delayBufferSample[0] = correctBufferSample[0] * weakChannelAmplitudeRatio;
delayMixSample[1] = _clientSamples[delayedChannelIndex[1]]; delayBufferSample[1] = correctBufferSample[1] * weakChannelAmplitudeRatio;
__m64 bufferSamples = _mm_set_pi16(correctMixSample[0], correctMixSample[1], delayMixSample[0], delayMixSample[1]); __m64 bufferSamples = _mm_set_pi16(_clientSamples[s + goodChannelOffset],
_clientSamples[s + goodChannelOffset + 2],
_clientSamples[delayedChannelIndex],
_clientSamples[delayedChannelIndex + 2]);
__m64 addedSamples = _mm_set_pi16(correctBufferSample[0], correctBufferSample[1], __m64 addedSamples = _mm_set_pi16(correctBufferSample[0], correctBufferSample[1],
delayBufferSample[0], delayBufferSample[1]); delayBufferSample[0], delayBufferSample[1]);
@ -209,8 +190,92 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
// assign the results from the result of the mmx arithmetic // assign the results from the result of the mmx arithmetic
_clientSamples[s + goodChannelOffset] = shortResults[3]; _clientSamples[s + goodChannelOffset] = shortResults[3];
_clientSamples[s + goodChannelOffset + 2] = shortResults[2]; _clientSamples[s + goodChannelOffset + 2] = shortResults[2];
_clientSamples[delayedChannelIndex[0]] = shortResults[1]; _clientSamples[delayedChannelIndex] = shortResults[1];
_clientSamples[delayedChannelIndex[1]] = shortResults[0]; _clientSamples[delayedChannelIndex + 2] = 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.
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_setr_pi16(_clientSamples[parentIndex + delayedChannelOffset],
_clientSamples[parentIndex + 2 + delayedChannelOffset],
_clientSamples[parentIndex + 4 + delayedChannelOffset],
_clientSamples[parentIndex + 6 + 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 + 2 + delayedChannelOffset] = shortResults[2];
_clientSamples[parentIndex + 4 + delayedChannelOffset] = shortResults[1];
_clientSamples[parentIndex + 6 + 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 + 2 + delayedChannelOffset],
_clientSamples[parentIndex + 4 + 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 + 2 + delayedChannelOffset] = shortResults[2];
_clientSamples[parentIndex + 4 + delayedChannelOffset] = shortResults[1];
} else if (i + 1 < numSamplesDelay) {
// MMX add two delayed samples
__m64 bufferSamples = _mm_set_pi16(_clientSamples[parentIndex + delayedChannelOffset],
_clientSamples[parentIndex + 2 + 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 + 2 + 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];
}
} }
} }
@ -303,7 +368,7 @@ void AudioMixer::run() {
&& ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer()) { && ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioRingBuffer()) {
prepareMixForListeningNode(node.data()); prepareMixForListeningNode(node.data());
memcpy(_clientMixBuffer.data() + numBytesPacketHeader, _clientSamples, sizeof(_clientSamples)); memcpy(_clientMixBuffer.data() + numBytesPacketHeader, _clientSamples, NETWORK_BUFFER_LENGTH_BYTES_STEREO);
nodeList->writeDatagram(_clientMixBuffer, node); nodeList->writeDatagram(_clientMixBuffer, node);
} }
} }

5
assignment-client/src/audio/AudioMixer.h
View file

@ -16,6 +16,8 @@
class PositionalAudioRingBuffer; class PositionalAudioRingBuffer;
class AvatarAudioRingBuffer; class AvatarAudioRingBuffer;
const int SAMPLE_PHASE_DELAY_AT_90 = 20;
/// Handles assignments of type AudioMixer - mixing streams of audio and re-distributing to various clients. /// Handles assignments of type AudioMixer - mixing streams of audio and re-distributing to various clients.
class AudioMixer : public ThreadedAssignment { class AudioMixer : public ThreadedAssignment {
Q_OBJECT Q_OBJECT
@ -38,7 +40,8 @@ private:
QByteArray _clientMixBuffer; QByteArray _clientMixBuffer;
int16_t _clientSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO]; // client samples capacity is larger than what will be sent to optimize mixing
int16_t _clientSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + SAMPLE_PHASE_DELAY_AT_90];
}; };
#endif /* defined(__hifi__AudioMixer__) */ #endif /* defined(__hifi__AudioMixer__) */