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move filter to single instance in mixer, and do filtering on pre-mix buffer not the mixed buffer

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This commit is contained in:
ZappoMan 2014-09-17 12:48:47 -07:00
parent c30ef9c331
commit ab447e20ee
4 changed files with 33 additions and 27 deletions
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35
assignment-client/src/audio/AudioMixer.cpp
View file

@ -57,6 +57,8 @@
#include "AvatarAudioStream.h" #include "AvatarAudioStream.h"
#include "InjectedAudioStream.h" #include "InjectedAudioStream.h"
#include "AudioMixer.h" #include "AudioMixer.h"
const float LOUDNESS_TO_DISTANCE_RATIO = 0.00001f; const float LOUDNESS_TO_DISTANCE_RATIO = 0.00001f;
@ -92,7 +94,9 @@ AudioMixer::AudioMixer(const QByteArray& packet) :
_timeSpentPerHashMatchCallStats(0, READ_DATAGRAMS_STATS_WINDOW_SECONDS), _timeSpentPerHashMatchCallStats(0, READ_DATAGRAMS_STATS_WINDOW_SECONDS),
_readPendingCallsPerSecondStats(1, READ_DATAGRAMS_STATS_WINDOW_SECONDS) _readPendingCallsPerSecondStats(1, READ_DATAGRAMS_STATS_WINDOW_SECONDS)
{ {
// constant defined in AudioMixer.h. However, we don't want to include this here
// we will soon find a better common home for these audio-related constants
_penumbraFilter.initialize(SAMPLE_RATE, (NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)) / 2);
} }
AudioMixer::~AudioMixer() { AudioMixer::~AudioMixer() {
@ -257,6 +261,7 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
// we need to do several things in this process: // we need to do several things in this process:
// 1) convert from mono to stereo by copying each input sample into the left and right output samples // 1) convert from mono to stereo by copying each input sample into the left and right output samples
// 2)
// 2) apply an attenuation AND fade to all samples (left and right) // 2) apply an attenuation AND fade to all samples (left and right)
// 3) based on the bearing relative angle to the source we will weaken and delay either the left or // 3) based on the bearing relative angle to the source we will weaken and delay either the left or
// right channel of the input into the output // right channel of the input into the output
@ -314,7 +319,7 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
for (int i = 0; i < numSamplesDelay; i++) { for (int i = 0; i < numSamplesDelay; i++) {
int16_t originalHistoricalSample = *delayStreamSourceSamples; int16_t originalHistoricalSample = *delayStreamSourceSamples;
_clientSamples[delayedChannelHistoricalAudioOutputIndex] += originalHistoricalSample _preMixSamples[delayedChannelHistoricalAudioOutputIndex] += originalHistoricalSample
* attenuationAndWeakChannelRatioAndFade; * attenuationAndWeakChannelRatioAndFade;
++delayStreamSourceSamples; // move our input pointer ++delayStreamSourceSamples; // move our input pointer
delayedChannelHistoricalAudioOutputIndex += OUTPUT_SAMPLES_PER_INPUT_SAMPLE; // move our output sample delayedChannelHistoricalAudioOutputIndex += OUTPUT_SAMPLES_PER_INPUT_SAMPLE; // move our output sample
@ -329,10 +334,10 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
// since we might be delayed, don't write beyond our maxOutputIndex // since we might be delayed, don't write beyond our maxOutputIndex
if (leftDestinationIndex <= maxOutputIndex) { if (leftDestinationIndex <= maxOutputIndex) {
_clientSamples[leftDestinationIndex] += leftSideSample; _preMixSamples[leftDestinationIndex] += leftSideSample;
} }
if (rightDestinationIndex <= maxOutputIndex) { if (rightDestinationIndex <= maxOutputIndex) {
_clientSamples[rightDestinationIndex] += rightSideSample; _preMixSamples[rightDestinationIndex] += rightSideSample;
} }
leftDestinationIndex += OUTPUT_SAMPLES_PER_INPUT_SAMPLE; leftDestinationIndex += OUTPUT_SAMPLES_PER_INPUT_SAMPLE;
@ -345,7 +350,7 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
float attenuationAndFade = attenuationCoefficient * repeatedFrameFadeFactor; float attenuationAndFade = attenuationCoefficient * repeatedFrameFadeFactor;
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s++) { for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s++) {
_clientSamples[s] = glm::clamp(_clientSamples[s] + (int)(streamPopOutput[s / stereoDivider] * attenuationAndFade), _preMixSamples[s] = glm::clamp(_preMixSamples[s] + (int)(streamPopOutput[s / stereoDivider] * attenuationAndFade),
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE); MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
} }
} }
@ -409,12 +414,15 @@ int AudioMixer::addStreamToMixForListeningNodeWithStream(PositionalAudioStream*
#endif #endif
// set the gain on both filter channels // set the gain on both filter channels
AudioFilterHSF1s& penumbraFilter = streamToAdd->getFilter(); _penumbraFilter.reset();
_penumbraFilter.setParameters(0, 0, SAMPLE_RATE, penumbraFilterFrequency, penumbraFilterGainL, penumbraFilterSlope);
penumbraFilter.setParameters(0, 0, SAMPLE_RATE, penumbraFilterFrequency, penumbraFilterGainL, penumbraFilterSlope); _penumbraFilter.setParameters(0, 1, SAMPLE_RATE, penumbraFilterFrequency, penumbraFilterGainR, penumbraFilterSlope);
penumbraFilter.setParameters(0, 1, SAMPLE_RATE, penumbraFilterFrequency, penumbraFilterGainR, penumbraFilterSlope); _penumbraFilter.render(_preMixSamples, _preMixSamples, NETWORK_BUFFER_LENGTH_SAMPLES_STEREO / 2);
}
penumbraFilter.render(_clientSamples, _clientSamples, NETWORK_BUFFER_LENGTH_SAMPLES_STEREO / 2);
// Actually mix the _preMixSamples into the _mixSamples here.
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s++) {
_mixSamples[s] = glm::clamp(_mixSamples[s] + _preMixSamples[s], MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
} }
return 1; return 1;
@ -424,7 +432,8 @@ int AudioMixer::prepareMixForListeningNode(Node* node) {
AvatarAudioStream* nodeAudioStream = ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioStream(); AvatarAudioStream* nodeAudioStream = ((AudioMixerClientData*) node->getLinkedData())->getAvatarAudioStream();
// zero out the client mix for this node // zero out the client mix for this node
memset(_clientSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_STEREO); memset(_preMixSamples, 0, sizeof(_preMixSamples));
memset(_mixSamples, 0, sizeof(_mixSamples));
// loop through all other nodes that have sufficient audio to mix // loop through all other nodes that have sufficient audio to mix
int streamsMixed = 0; int streamsMixed = 0;
@ -817,7 +826,7 @@ void AudioMixer::run() {
dataAt += sizeof(quint16); dataAt += sizeof(quint16);
// pack mixed audio samples // pack mixed audio samples
memcpy(dataAt, _clientSamples, NETWORK_BUFFER_LENGTH_BYTES_STEREO); memcpy(dataAt, _mixSamples, NETWORK_BUFFER_LENGTH_BYTES_STEREO);
dataAt += NETWORK_BUFFER_LENGTH_BYTES_STEREO; dataAt += NETWORK_BUFFER_LENGTH_BYTES_STEREO;
} else { } else {
// pack header // pack header

13
assignment-client/src/audio/AudioMixer.h
View file

@ -13,6 +13,10 @@
#define hifi_AudioMixer_h #define hifi_AudioMixer_h
#include <AABox.h> #include <AABox.h>
#include <AudioFormat.h> // For AudioFilterHSF1s and _penumbraFilter
#include <AudioBuffer.h> // For AudioFilterHSF1s and _penumbraFilter
#include <AudioFilter.h> // For AudioFilterHSF1s and _penumbraFilter
#include <AudioFilterBank.h> // For AudioFilterHSF1s and _penumbraFilter
#include <AudioRingBuffer.h> #include <AudioRingBuffer.h>
#include <ThreadedAssignment.h> #include <ThreadedAssignment.h>
@ -23,7 +27,6 @@ const int SAMPLE_PHASE_DELAY_AT_90 = 20;
const int READ_DATAGRAMS_STATS_WINDOW_SECONDS = 30; const int READ_DATAGRAMS_STATS_WINDOW_SECONDS = 30;
/// 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
@ -48,11 +51,17 @@ private:
/// prepares and sends a mix to one Node /// prepares and sends a mix to one Node
int prepareMixForListeningNode(Node* node); int prepareMixForListeningNode(Node* node);
// used on a per stream basis to run the filter on before mixing, large enough to handle the historical
// data from a phase delay as well as an entire network buffer
int16_t _preMixSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)];
// client samples capacity is larger than what will be sent to optimize mixing // client samples capacity is larger than what will be sent to optimize mixing
// we are MMX adding 4 samples at a time so we need client samples to have an extra 4 // we are MMX adding 4 samples at a time so we need client samples to have an extra 4
int16_t _clientSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)]; int16_t _mixSamples[NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)];
AudioFilterHSF1s _penumbraFilter;
void perSecondActions(); void perSecondActions();
QString getReadPendingDatagramsCallsPerSecondsStatsString() const; QString getReadPendingDatagramsCallsPerSecondsStatsString() const;

4
libraries/audio/src/PositionalAudioStream.cpp
View file

@ -33,10 +33,6 @@ PositionalAudioStream::PositionalAudioStream(PositionalAudioStream::Type type, b
_lastPopOutputLoudness(0.0f), _lastPopOutputLoudness(0.0f),
_listenerUnattenuatedZone(NULL) _listenerUnattenuatedZone(NULL)
{ {
// constant defined in AudioMixer.h. However, we don't want to include this here
// we will soon find a better common home for these audio-related constants
const int SAMPLE_PHASE_DELAY_AT_90 = 20;
_filter.initialize(SAMPLE_RATE, (NETWORK_BUFFER_LENGTH_SAMPLES_STEREO + (SAMPLE_PHASE_DELAY_AT_90 * 2)) / 2);
} }
void PositionalAudioStream::resetStats() { void PositionalAudioStream::resetStats() {

8
libraries/audio/src/PositionalAudioStream.h
View file

@ -16,10 +16,6 @@
#include <AABox.h> #include <AABox.h>
#include "InboundAudioStream.h" #include "InboundAudioStream.h"
#include "AudioFormat.h"
#include "AudioBuffer.h"
#include "AudioFilter.h"
#include "AudioFilterBank.h"
const int AUDIOMIXER_INBOUND_RING_BUFFER_FRAME_CAPACITY = 100; const int AUDIOMIXER_INBOUND_RING_BUFFER_FRAME_CAPACITY = 100;
@ -50,8 +46,6 @@ public:
void setListenerUnattenuatedZone(AABox* listenerUnattenuatedZone) { _listenerUnattenuatedZone = listenerUnattenuatedZone; } void setListenerUnattenuatedZone(AABox* listenerUnattenuatedZone) { _listenerUnattenuatedZone = listenerUnattenuatedZone; }
AudioFilterHSF1s& getFilter() { return _filter; }
protected: protected:
// disallow copying of PositionalAudioStream objects // disallow copying of PositionalAudioStream objects
PositionalAudioStream(const PositionalAudioStream&); PositionalAudioStream(const PositionalAudioStream&);
@ -70,8 +64,6 @@ protected:
float _lastPopOutputTrailingLoudness; float _lastPopOutputTrailingLoudness;
float _lastPopOutputLoudness; float _lastPopOutputLoudness;
AABox* _listenerUnattenuatedZone; AABox* _listenerUnattenuatedZone;
AudioFilterHSF1s _filter;
}; };
#endif // hifi_PositionalAudioStream_h #endif // hifi_PositionalAudioStream_h