Merge pull request #10502 from kencooke/audio-noisegate-new

Advanced noise gate
This commit is contained in:
Brad Hefta-Gaub 2017-05-25 08:21:49 -07:00 committed by GitHub
commit 756e00e9a9
12 changed files with 1432 additions and 711 deletions

View file

@ -55,7 +55,8 @@ static const int RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES = 10;
Agent::Agent(ReceivedMessage& message) :
ThreadedAssignment(message),
_receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES, RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES)
_receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES, RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES),
_audioGate(AudioConstants::SAMPLE_RATE, AudioConstants::MONO)
{
_entityEditSender.setPacketsPerSecond(DEFAULT_ENTITY_PPS_PER_SCRIPT);
DependencyManager::get<EntityScriptingInterface>()->setPacketSender(&_entityEditSender);
@ -397,16 +398,23 @@ void Agent::executeScript() {
QByteArray audio(frame->data);
if (_isNoiseGateEnabled) {
static int numSamples = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL;
_noiseGate.gateSamples(reinterpret_cast<int16_t*>(audio.data()), numSamples);
int16_t* samples = reinterpret_cast<int16_t*>(audio.data());
int numSamples = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL;
_audioGate.render(samples, samples, numSamples);
}
computeLoudness(&audio, scriptedAvatar);
// the codec needs a flush frame before sending silent packets, so
// do not send one if the gate closed in this block (eventually this can be crossfaded).
// state machine to detect gate opening and closing
bool audioGateOpen = (scriptedAvatar->getAudioLoudness() != 0.0f);
bool openedInLastBlock = !_audioGateOpen && audioGateOpen; // the gate just opened
bool closedInLastBlock = _audioGateOpen && !audioGateOpen; // the gate just closed
_audioGateOpen = audioGateOpen;
// the codec must be flushed to silence before sending silent packets,
// so delay the transition to silent packets by one packet after becoming silent.
auto packetType = PacketType::MicrophoneAudioNoEcho;
if (scriptedAvatar->getAudioLoudness() == 0.0f && !_noiseGate.closedInLastBlock()) {
if (!audioGateOpen && !closedInLastBlock) {
packetType = PacketType::SilentAudioFrame;
}
@ -620,19 +628,21 @@ void Agent::encodeFrameOfZeros(QByteArray& encodedZeros) {
}
void Agent::computeLoudness(const QByteArray* decodedBuffer, QSharedPointer<ScriptableAvatar> scriptableAvatar) {
float loudness = 0.0f;
float lastInputLoudness = 0.0f;
if (decodedBuffer) {
auto soundData = reinterpret_cast<const int16_t*>(decodedBuffer->constData());
int numFrames = decodedBuffer->size() / sizeof(int16_t);
// now iterate and come up with average
if (numFrames > 0) {
for(int i = 0; i < numFrames; i++) {
loudness += (float) std::abs(soundData[i]);
auto samples = reinterpret_cast<const int16_t*>(decodedBuffer->constData());
int numSamples = decodedBuffer->size() / AudioConstants::SAMPLE_SIZE;
assert(numSamples < 65536); // int32_t loudness cannot overflow
if (numSamples > 0) {
int32_t loudness = 0;
for (int i = 0; i < numSamples; ++i) {
loudness += std::abs((int32_t)samples[i]);
}
loudness /= numFrames;
lastInputLoudness = (float)loudness / numSamples;
}
}
scriptableAvatar->setAudioLoudness(loudness);
scriptableAvatar->setAudioLoudness(lastInputLoudness);
}
void Agent::processAgentAvatarAudio() {

View file

@ -29,7 +29,7 @@
#include <plugins/CodecPlugin.h>
#include "AudioNoiseGate.h"
#include "AudioGate.h"
#include "MixedAudioStream.h"
#include "avatars/ScriptableAvatar.h"
@ -111,7 +111,8 @@ private:
QTimer* _avatarIdentityTimer = nullptr;
QHash<QUuid, quint16> _outgoingScriptAudioSequenceNumbers;
AudioNoiseGate _noiseGate;
AudioGate _audioGate;
bool _audioGateOpen { false };
bool _isNoiseGateEnabled { false };
CodecPluginPointer _codec;

View file

@ -1007,30 +1007,27 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
_timeSinceLastClip = 0.0f;
} else {
int16_t* samples = reinterpret_cast<int16_t*>(audioBuffer.data());
int numSamples = audioBuffer.size() / sizeof(AudioConstants::SAMPLE_SIZE);
bool didClip = false;
int numSamples = audioBuffer.size() / AudioConstants::SAMPLE_SIZE;
int numFrames = numSamples / (_isStereoInput ? AudioConstants::STEREO : AudioConstants::MONO);
bool shouldRemoveDCOffset = !_isPlayingBackRecording && !_isStereoInput;
if (shouldRemoveDCOffset) {
_noiseGate.removeDCOffset(samples, numSamples);
}
bool shouldNoiseGate = (_isPlayingBackRecording || !_isStereoInput) && _isNoiseGateEnabled;
if (shouldNoiseGate) {
_noiseGate.gateSamples(samples, numSamples);
_lastInputLoudness = _noiseGate.getLastLoudness();
didClip = _noiseGate.clippedInLastBlock();
if (_isNoiseGateEnabled) {
// The audio gate includes DC removal
_audioGate->render(samples, samples, numFrames);
} else {
float loudness = 0.0f;
for (int i = 0; i < numSamples; ++i) {
int16_t sample = std::abs(samples[i]);
loudness += (float)sample;
didClip = didClip ||
(sample > (AudioConstants::MAX_SAMPLE_VALUE * AudioNoiseGate::CLIPPING_THRESHOLD));
}
_lastInputLoudness = fabs(loudness / numSamples);
_audioGate->removeDC(samples, samples, numFrames);
}
int32_t loudness = 0;
assert(numSamples < 65536); // int32_t loudness cannot overflow
bool didClip = false;
for (int i = 0; i < numSamples; ++i) {
const int32_t CLIPPING_THRESHOLD = (int32_t)(AudioConstants::MAX_SAMPLE_VALUE * 0.9f);
int32_t sample = std::abs((int32_t)samples[i]);
loudness += sample;
didClip |= (sample > CLIPPING_THRESHOLD);
}
_lastInputLoudness = (float)loudness / numSamples;
if (didClip) {
_timeSinceLastClip = 0.0f;
} else if (_timeSinceLastClip >= 0.0f) {
@ -1038,19 +1035,24 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
}
emit inputReceived(audioBuffer);
if (_noiseGate.openedInLastBlock()) {
emit noiseGateOpened();
} else if (_noiseGate.closedInLastBlock()) {
emit noiseGateClosed();
}
}
// the codec needs a flush frame before sending silent packets, so
// do not send one if the gate closed in this block (eventually this can be crossfaded).
auto packetType = _shouldEchoToServer ?
PacketType::MicrophoneAudioWithEcho : PacketType::MicrophoneAudioNoEcho;
if (_lastInputLoudness == 0.0f && !_noiseGate.closedInLastBlock()) {
// state machine to detect gate opening and closing
bool audioGateOpen = (_lastInputLoudness != 0.0f);
bool openedInLastBlock = !_audioGateOpen && audioGateOpen; // the gate just opened
bool closedInLastBlock = _audioGateOpen && !audioGateOpen; // the gate just closed
_audioGateOpen = audioGateOpen;
if (openedInLastBlock) {
emit noiseGateOpened();
} else if (closedInLastBlock) {
emit noiseGateClosed();
}
// the codec must be flushed to silence before sending silent packets,
// so delay the transition to silent packets by one packet after becoming silent.
auto packetType = _shouldEchoToServer ? PacketType::MicrophoneAudioWithEcho : PacketType::MicrophoneAudioNoEcho;
if (!audioGateOpen && !closedInLastBlock) {
packetType = PacketType::SilentAudioFrame;
_silentOutbound.increment();
} else {
@ -1415,6 +1417,10 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
delete _inputToNetworkResampler;
_inputToNetworkResampler = NULL;
}
if (_audioGate) {
delete _audioGate;
_audioGate = nullptr;
}
if (!inputDeviceInfo.isNull()) {
qCDebug(audioclient) << "The audio input device " << inputDeviceInfo.deviceName() << "is available.";
@ -1440,6 +1446,10 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
qCDebug(audioclient) << "No resampling required for audio input to match desired network format.";
}
// the audio gate runs after the resampler
_audioGate = new AudioGate(_desiredInputFormat.sampleRate(), _desiredInputFormat.channelCount());
qCDebug(audioclient) << "Noise gate created with" << _desiredInputFormat.channelCount() << "channels.";
// if the user wants stereo but this device can't provide then bail
if (!_isStereoInput || _inputFormat.channelCount() == 2) {
_audioInput = new QAudioInput(inputDeviceInfo, _inputFormat, this);

View file

@ -45,7 +45,7 @@
#include <AudioReverb.h>
#include <AudioLimiter.h>
#include <AudioConstants.h>
#include <AudioNoiseGate.h>
#include <AudioGate.h>
#include <shared/RateCounter.h>
@ -108,7 +108,7 @@ public:
void selectAudioFormat(const QString& selectedCodecName);
Q_INVOKABLE QString getSelectedAudioFormat() const { return _selectedCodecName; }
Q_INVOKABLE bool getNoiseGateOpen() const { return _noiseGate.isOpen(); }
Q_INVOKABLE bool getNoiseGateOpen() const { return _audioGateOpen; }
Q_INVOKABLE float getSilentInboundPPS() const { return _silentInbound.rate(); }
Q_INVOKABLE float getAudioInboundPPS() const { return _audioInbound.rate(); }
Q_INVOKABLE float getSilentOutboundPPS() const { return _silentOutbound.rate(); }
@ -117,7 +117,7 @@ public:
const MixedProcessedAudioStream& getReceivedAudioStream() const { return _receivedAudioStream; }
MixedProcessedAudioStream& getReceivedAudioStream() { return _receivedAudioStream; }
float getLastInputLoudness() const { return glm::max(_lastInputLoudness - _noiseGate.getMeasuredFloor(), 0.0f); }
float getLastInputLoudness() const { return _lastInputLoudness; } // TODO: relative to noise floor?
float getTimeSinceLastClip() const { return _timeSinceLastClip; }
float getAudioAverageInputLoudness() const { return _lastInputLoudness; }
@ -359,7 +359,8 @@ private:
AudioIOStats _stats;
AudioNoiseGate _noiseGate;
AudioGate* _audioGate { nullptr };
bool _audioGateOpen { false };
AudioPositionGetter _positionGetter;
AudioOrientationGetter _orientationGetter;

View file

@ -0,0 +1,585 @@
//
// AudioDynamics.h
// libraries/audio/src
//
// Created by Ken Cooke on 5/5/17.
// Copyright 2017 High Fidelity, Inc.
//
//
// Inline functions to implement audio dynamics processing
//
#include <math.h>
#include <stdint.h>
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif
#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
#ifdef _MSC_VER
#include <intrin.h>
#define MUL64(a,b) __emul((a), (b))
#else
#define MUL64(a,b) ((int64_t)(a) * (int64_t)(b))
#endif
#define MULHI(a,b) ((int32_t)(MUL64(a, b) >> 32))
#define MULQ31(a,b) ((int32_t)(MUL64(a, b) >> 31))
#define MULDIV64(a,b,c) (int32_t)(MUL64(a, b) / (c))
//
// on x86 architecture, assume that SSE2 is present
//
#if defined(_M_IX86) || defined(_M_X64) || defined(__i386__) || defined(__x86_64__)
#include <xmmintrin.h>
// convert float to int using round-to-nearest
static inline int32_t floatToInt(float x) {
return _mm_cvt_ss2si(_mm_load_ss(&x));
}
#else
// convert float to int using round-to-nearest
static inline int32_t floatToInt(float x) {
x += (x < 0.0f ? -0.5f : 0.5f); // round
return (int32_t)x;
}
#endif // _M_IX86
static const double FIXQ31 = 2147483648.0; // convert float to Q31
static const double DB_TO_LOG2 = 0.16609640474436813; // convert dB to log2
// convert dB to amplitude
static inline double dBToGain(double dB) {
return pow(10.0, dB / 20.0);
}
// convert milliseconds to first-order time constant
static inline int32_t msToTc(double ms, double sampleRate) {
double tc = exp(-1000.0 / (ms * sampleRate));
return (int32_t)(FIXQ31 * tc); // Q31
}
// log2 domain values are Q26
static const int LOG2_INTBITS = 5;
static const int LOG2_FRACBITS = 31 - LOG2_INTBITS;
// log2 domain headroom bits above 0dB
static const int LOG2_HEADROOM = 15;
// log2 domain offsets so error < 0
static const int32_t LOG2_BIAS = 347;
static const int32_t EXP2_BIAS = 64;
//
// P(x) = log2(1+x) for x=[0,1]
// scaled by 1, 0.5, 0.25
//
// |error| < 347 ulp, smooth
//
static const int LOG2_TABBITS = 4;
static const int32_t log2Table[1 << LOG2_TABBITS][3] = {
{ -0x56dfe26d, 0x5c46daff, 0x00000000 },
{ -0x4d397571, 0x5bae58e7, 0x00025a75 },
{ -0x4518f84b, 0x5aabcac4, 0x000a62db },
{ -0x3e3075ec, 0x596168c0, 0x0019d0e6 },
{ -0x384486e9, 0x57e769c7, 0x00316109 },
{ -0x332742ba, 0x564f1461, 0x00513776 },
{ -0x2eb4bad4, 0x54a4cdfe, 0x00791de2 },
{ -0x2ad07c6c, 0x52f18320, 0x00a8aa46 },
{ -0x2763c4d6, 0x513ba123, 0x00df574c },
{ -0x245c319b, 0x4f87c5c4, 0x011c9399 },
{ -0x21aac79f, 0x4dd93bef, 0x015fcb52 },
{ -0x1f433872, 0x4c325584, 0x01a86ddc },
{ -0x1d1b54b4, 0x4a94ac6e, 0x01f5f13e },
{ -0x1b2a9f81, 0x4901524f, 0x0247d3f2 },
{ -0x1969fa57, 0x4778f3a7, 0x029d9dbf },
{ -0x17d36370, 0x45fbf1e8, 0x02f6dfe8 },
};
//
// P(x) = exp2(x) for x=[0,1]
// scaled by 2, 1, 0.5
// Uses exp2(-x) = exp2(1-x)/2
//
// |error| < 1387 ulp, smooth
//
static const int EXP2_TABBITS = 4;
static const int32_t exp2Table[1 << EXP2_TABBITS][3] = {
{ 0x3ed838c8, 0x58b574b7, 0x40000000 },
{ 0x41a0821c, 0x5888db8f, 0x4000b2b7 },
{ 0x4488548d, 0x582bcbc6, 0x40039be1 },
{ 0x4791158a, 0x579a1128, 0x400a71ae },
{ 0x4abc3a53, 0x56cf3089, 0x4017212e },
{ 0x4e0b48af, 0x55c66396, 0x402bd31b },
{ 0x517fd7a7, 0x547a946d, 0x404af0ec },
{ 0x551b9049, 0x52e658f9, 0x40772a57 },
{ 0x58e02e75, 0x5103ee08, 0x40b37b31 },
{ 0x5ccf81b1, 0x4ecd321f, 0x410331b5 },
{ 0x60eb6e09, 0x4c3ba007, 0x4169f548 },
{ 0x6535ecf9, 0x49484909, 0x41ebcdaf },
{ 0x69b10e5b, 0x45ebcede, 0x428d2acd },
{ 0x6e5ef96c, 0x421e5d48, 0x4352ece7 },
{ 0x7341edcb, 0x3dd7a354, 0x44426d7b },
{ 0x785c4499, 0x390ecc3a, 0x456188bd },
};
static const int IEEE754_FABS_MASK = 0x7fffffff;
static const int IEEE754_MANT_BITS = 23;
static const int IEEE754_EXPN_BIAS = 127;
//
// Peak detection and -log2(x) for float input (mono)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input) {
// float as integer bits
int32_t u = *(int32_t*)input;
// absolute value
int32_t peak = u & IEEE754_FABS_MASK;
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Peak detection and -log2(x) for float input (stereo)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input0, float* input1) {
// float as integer bits
int32_t u0 = *(int32_t*)input0;
int32_t u1 = *(int32_t*)input1;
// max absolute value
u0 &= IEEE754_FABS_MASK;
u1 &= IEEE754_FABS_MASK;
int32_t peak = MAX(u0, u1);
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Peak detection and -log2(x) for float input (quad)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input0, float* input1, float* input2, float* input3) {
// float as integer bits
int32_t u0 = *(int32_t*)input0;
int32_t u1 = *(int32_t*)input1;
int32_t u2 = *(int32_t*)input2;
int32_t u3 = *(int32_t*)input3;
// max absolute value
u0 &= IEEE754_FABS_MASK;
u1 &= IEEE754_FABS_MASK;
u2 &= IEEE754_FABS_MASK;
u3 &= IEEE754_FABS_MASK;
int32_t peak = MAX(MAX(u0, u1), MAX(u2, u3));
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Count Leading Zeros
// Emulates the CLZ (ARM) and LZCNT (x86) instruction
//
static inline int CLZ(uint32_t x) {
if (x == 0) {
return 32;
}
int e = 0;
if (x < 0x00010000) {
x <<= 16;
e += 16;
}
if (x < 0x01000000) {
x <<= 8;
e += 8;
}
if (x < 0x10000000) {
x <<= 4;
e += 4;
}
if (x < 0x40000000) {
x <<= 2;
e += 2;
}
if (x < 0x80000000) {
e += 1;
}
return e;
}
//
// Compute -log2(x) for x=[0,1] in Q31, result in Q26
// x = 0 returns 0x7fffffff
// x < 0 undefined
//
static inline int32_t fixlog2(int32_t x) {
if (x == 0) {
return 0x7fffffff;
}
// split into e and x - 1.0
int e = CLZ((uint32_t)x);
x <<= e; // normalize to [0x80000000, 0xffffffff]
x &= 0x7fffffff; // x - 1.0
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Compute exp2(-x) for x=[0,32] in Q26, result in Q31
// x < 0 undefined
//
static inline int32_t fixexp2(int32_t x) {
// split into e and 1.0 - x
int e = x >> LOG2_FRACBITS;
x = ~(x << LOG2_INTBITS) & 0x7fffffff;
int k = x >> (31 - EXP2_TABBITS);
// polynomial for exp2(x)
int32_t c0 = exp2Table[k][0];
int32_t c1 = exp2Table[k][1];
int32_t c2 = exp2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q31
return c2 >> e;
}
// fast TPDF dither in [-1.0f, 1.0f]
static inline float dither() {
static uint32_t rz = 0;
rz = rz * 69069 + 1;
int32_t r0 = rz & 0xffff;
int32_t r1 = rz >> 16;
return (int32_t)(r0 - r1) * (1/65536.0f);
}
//
// Min-hold lowpass filter
//
// Bandlimits the gain control signal to greatly reduce the modulation distortion,
// while still reaching the peak attenuation after exactly N-1 samples of delay.
// N completely determines the attack time.
//
template<int N, int CIC1, int CIC2>
class MinFilterT {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
static_assert((CIC1 - 1) + (CIC2 - 1) == (N - 1), "Total CIC delay must be N-1");
int32_t _buffer[2*N] = {}; // shared FIFO
size_t _index = 0;
int32_t _acc1 = 0; // CIC1 integrator
int32_t _acc2 = 0; // CIC2 integrator
public:
MinFilterT() {
// fill history
for (size_t n = 0; n < N-1; n++) {
process(0x7fffffff);
}
}
int32_t process(int32_t x) {
const size_t MASK = 2*N - 1; // buffer wrap
size_t i = _index;
// Fast min-hold using a running-min filter. Finds the peak (min) value
// in the sliding window of N-1 samples, using only log2(N) comparisons.
// Hold time of N-1 samples exactly cancels the step response of FIR filter.
for (size_t n = 1; n < N; n <<= 1) {
_buffer[i] = x;
i = (i + n) & MASK;
x = MIN(x, _buffer[i]);
}
// Fast FIR attack/lowpass filter using a 2-stage CIC filter.
// The step response reaches final value after N-1 samples.
const int32_t CICGAIN = 0xffffffff / (CIC1 * CIC2); // Q32
x = MULHI(x, CICGAIN);
_buffer[i] = _acc1;
_acc1 += x; // integrator
i = (i + CIC1 - 1) & MASK;
x = _acc1 - _buffer[i]; // comb
_buffer[i] = _acc2;
_acc2 += x; // integrator
i = (i + CIC2 - 1) & MASK;
x = _acc2 - _buffer[i]; // comb
_index = (i + 1) & MASK; // skip unused tap
return x;
}
};
//
// Max-hold lowpass filter
//
// Bandlimits the gain control signal to greatly reduce the modulation distortion,
// while still reaching the peak attenuation after exactly N-1 samples of delay.
// N completely determines the attack time.
//
template<int N, int CIC1, int CIC2>
class MaxFilterT {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
static_assert((CIC1 - 1) + (CIC2 - 1) == (N - 1), "Total CIC delay must be N-1");
int32_t _buffer[2*N] = {}; // shared FIFO
size_t _index = 0;
int32_t _acc1 = 0; // CIC1 integrator
int32_t _acc2 = 0; // CIC2 integrator
public:
MaxFilterT() {
// fill history
for (size_t n = 0; n < N-1; n++) {
process(0);
}
}
int32_t process(int32_t x) {
const size_t MASK = 2*N - 1; // buffer wrap
size_t i = _index;
// Fast max-hold using a running-max filter. Finds the peak (max) value
// in the sliding window of N-1 samples, using only log2(N) comparisons.
// Hold time of N-1 samples exactly cancels the step response of FIR filter.
for (size_t n = 1; n < N; n <<= 1) {
_buffer[i] = x;
i = (i + n) & MASK;
x = MAX(x, _buffer[i]);
}
// Fast FIR attack/lowpass filter using a 2-stage CIC filter.
// The step response reaches final value after N-1 samples.
const int32_t CICGAIN = 0xffffffff / (CIC1 * CIC2); // Q32
x = MULHI(x, CICGAIN);
_buffer[i] = _acc1;
_acc1 += x; // integrator
i = (i + CIC1 - 1) & MASK;
x = _acc1 - _buffer[i]; // comb
_buffer[i] = _acc2;
_acc2 += x; // integrator
i = (i + CIC2 - 1) & MASK;
x = _acc2 - _buffer[i]; // comb
_index = (i + 1) & MASK; // skip unused tap
return x;
}
};
//
// Specializations that define the optimum lowpass filter for each length.
//
template<int N> class MinFilter;
template<> class MinFilter< 16> : public MinFilterT< 16, 7, 10> {};
template<> class MinFilter< 32> : public MinFilterT< 32, 14, 19> {};
template<> class MinFilter< 64> : public MinFilterT< 64, 27, 38> {};
template<> class MinFilter<128> : public MinFilterT<128, 53, 76> {};
template<> class MinFilter<256> : public MinFilterT<256, 106, 151> {};
template<int N> class MaxFilter;
template<> class MaxFilter< 16> : public MaxFilterT< 16, 7, 10> {};
template<> class MaxFilter< 32> : public MaxFilterT< 32, 14, 19> {};
template<> class MaxFilter< 64> : public MaxFilterT< 64, 27, 38> {};
template<> class MaxFilter<128> : public MaxFilterT<128, 53, 76> {};
template<> class MaxFilter<256> : public MaxFilterT<256, 106, 151> {};
//
// N-1 sample delay (mono)
//
template<int N, typename T = float>
class MonoDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
T _buffer[N] = {};
size_t _index = 0;
public:
void process(T& x) {
const size_t MASK = N - 1; // buffer wrap
size_t i = _index;
_buffer[i] = x;
i = (i + (N - 1)) & MASK;
x = _buffer[i];
_index = i;
}
};
//
// N-1 sample delay (stereo)
//
template<int N, typename T = float>
class StereoDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
T _buffer[2*N] = {};
size_t _index = 0;
public:
void process(T& x0, T& x1) {
const size_t MASK = 2*N - 1; // buffer wrap
size_t i = _index;
_buffer[i+0] = x0;
_buffer[i+1] = x1;
i = (i + 2*(N - 1)) & MASK;
x0 = _buffer[i+0];
x1 = _buffer[i+1];
_index = i;
}
};
//
// N-1 sample delay (quad)
//
template<int N, typename T = float>
class QuadDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
T _buffer[4*N] = {};
size_t _index = 0;
public:
void process(T& x0, T& x1, T& x2, T& x3) {
const size_t MASK = 4*N - 1; // buffer wrap
size_t i = _index;
_buffer[i+0] = x0;
_buffer[i+1] = x1;
_buffer[i+2] = x2;
_buffer[i+3] = x3;
i = (i + 4*(N - 1)) & MASK;
x0 = _buffer[i+0];
x1 = _buffer[i+1];
x2 = _buffer[i+2];
x3 = _buffer[i+3];
_index = i;
}
};

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@ -0,0 +1,735 @@
//
// AudioGate.cpp
// libraries/audio/src
//
// Created by Ken Cooke on 5/5/17.
// Copyright 2017 High Fidelity, Inc.
//
#include <string.h>
#include <assert.h>
#include "AudioDynamics.h"
#include "AudioGate.h"
// log2 domain headroom bits above 0dB (int32_t)
static const int LOG2_HEADROOM_Q30 = 1;
// convert Q30 to Q15 with saturation
static inline int32_t saturateQ30(int32_t x) {
x = (x + (1 << 14)) >> 15;
x = MIN(MAX(x, -32768), 32767);
return x;
}
//
// First-order DC-blocking filter, with zero at 1.0 and pole at 0.99994
//
// -3dB @ 0.5 Hz (48KHz)
// -3dB @ 0.2 Hz (24KHz)
//
// input in Q15, output in Q30
//
class MonoDCBlock {
int32_t _dcOffset = {}; // Q30, cannot overflow
public:
void process(int32_t& x) {
x <<= 15; // scale to Q30
x -= _dcOffset; // remove DC
_dcOffset += x >> 14; // pole = (1.0 - 2^-14) = 0.99994
}
};
class StereoDCBlock {
int32_t _dcOffset[2] = {};
public:
void process(int32_t& x0, int32_t& x1) {
x0 <<= 15;
x1 <<= 15;
x0 -= _dcOffset[0];
x1 -= _dcOffset[1];
_dcOffset[0] += x0 >> 14;
_dcOffset[1] += x1 >> 14;
}
};
class QuadDCBlock {
int32_t _dcOffset[4] = {};
public:
void process(int32_t& x0, int32_t& x1, int32_t& x2, int32_t& x3) {
x0 <<= 15;
x1 <<= 15;
x2 <<= 15;
x3 <<= 15;
x0 -= _dcOffset[0];
x1 -= _dcOffset[1];
x2 -= _dcOffset[2];
x3 -= _dcOffset[3];
_dcOffset[0] += x0 >> 14;
_dcOffset[1] += x1 >> 14;
_dcOffset[2] += x2 >> 14;
_dcOffset[3] += x3 >> 14;
}
};
//
// Gate (common)
//
class GateImpl {
protected:
// histogram
static const int NHIST = 256;
int _update[NHIST] = {};
int _histogram[NHIST] = {};
// peakhold
int32_t _holdMin = 0x7fffffff;
int32_t _holdInc = 0x7fffffff;
uint32_t _holdMax = 0x7fffffff;
int32_t _holdRel = 0x7fffffff;
int32_t _holdPeak = 0x7fffffff;
// hysteresis
int32_t _hysteresis = 0;
int32_t _hystOffset = 0;
int32_t _hystPeak = 0x7fffffff;
int32_t _release = 0x7fffffff;
int32_t _threshFixed = 0;
int32_t _threshAdapt = 0;
int32_t _attn = 0x7fffffff;
int _sampleRate;
public:
GateImpl(int sampleRate);
virtual ~GateImpl() {}
void setThreshold(float threshold);
void setHold(float hold);
void setHysteresis(float hysteresis);
void setRelease(float release);
void clearHistogram() { memset(_update, 0, sizeof(_update)); }
void updateHistogram(int32_t value, int count);
int partitionHistogram();
void processHistogram(int numFrames);
int32_t peakhold(int32_t peak);
int32_t hysteresis(int32_t peak);
int32_t envelope(int32_t attn);
virtual void process(int16_t* input, int16_t* output, int numFrames) = 0;
virtual void removeDC(int16_t* input, int16_t* output, int numFrames) = 0;
};
GateImpl::GateImpl(int sampleRate) {
sampleRate = MAX(sampleRate, 8000);
sampleRate = MIN(sampleRate, 96000);
_sampleRate = sampleRate;
// defaults
setThreshold(-36.0);
setHold(20.0);
setHysteresis(6.0);
setRelease(1000.0);
}
//
// Set the gate threshold (dB)
// This is a base value that is modulated by the adaptive threshold algorithm.
//
void GateImpl::setThreshold(float threshold) {
// gate threshold = -96dB to 0dB
threshold = MAX(threshold, -96.0f);
threshold = MIN(threshold, 0.0f);
// gate threshold in log2 domain
_threshFixed = (int32_t)(-(double)threshold * DB_TO_LOG2 * (1 << LOG2_FRACBITS));
_threshFixed += LOG2_HEADROOM_Q30 << LOG2_FRACBITS;
_threshAdapt = _threshFixed;
}
//
// Set the detector hold time (milliseconds)
//
void GateImpl::setHold(float hold) {
const double RELEASE = 100.0; // release = 100ms
const double PROGHOLD = 0.100; // progressive hold = 100ms
// pure hold = 1 to 1000ms
hold = MAX(hold, 1.0f);
hold = MIN(hold, 1000.0f);
_holdMin = msToTc(RELEASE, _sampleRate);
_holdInc = (int32_t)((_holdMin - 0x7fffffff) / (PROGHOLD * _sampleRate));
_holdInc = MIN(_holdInc, -1); // prevent 0 on long releases
_holdMax = 0x7fffffff - (uint32_t)(_holdInc * (double)hold/1000.0 * _sampleRate);
}
//
// Set the detector hysteresis (dB)
//
void GateImpl::setHysteresis(float hysteresis) {
// gate hysteresis in log2 domain
_hysteresis = (int32_t)((double)hysteresis * DB_TO_LOG2 * (1 << LOG2_FRACBITS));
}
//
// Set the gate release time (milliseconds)
//
void GateImpl::setRelease(float release) {
// gate release = 50 to 5000ms
release = MAX(release, 50.0f);
release = MIN(release, 5000.0f);
_release = msToTc((double)release, _sampleRate);
}
//
// Update the histogram count of the bin which contains value
//
void GateImpl::updateHistogram(int32_t value, int count = 1) {
// quantize to LOG2 + 3 fraction bits (0.75dB steps)
int index = (NHIST-1) - (value >> (LOG2_FRACBITS - 3));
assert(index >= 0);
assert(index < NHIST);
_update[index] += count << 16; // Q16 for filtering
assert(_update[index] >= 0);
}
//
// Partition the histogram
//
// The idea behind the adaptive threshold:
//
// When processing a gaussian mixture of signal and noise, separated by minimal SNR,
// a bimodal distribution emerges in the histogram of preprocessed peak levels.
// In this case, the threshold adapts toward the level that optimally partitions the distributions.
// Partitioning is computed using Otsu's method.
//
// When only a single distribution is present, the threshold becomes level-dependent:
// At levels below the fixed threshold, the threshold adapts toward the upper edge
// of the distribution, presumed to be noise.
// At levels above the fixed threshold, the threshold adapts toward the lower edge
// of the distribution, presumed to be signal.
// This is implemented by adding a hidden (bias) distribution at the fixed threshold.
//
int GateImpl::partitionHistogram() {
// initialize
int total = 0;
float sum = 0.0f;
for (int i = 0; i < NHIST; i++) {
total += _histogram[i];
sum += (float)i * _histogram[i];
}
int w0 = 0;
float sum0 = 0.0f;
float max = 0.0f;
int index = 0;
// find the index that maximizes the between-class variance
for (int i = 0 ; i < NHIST; i++) {
// update weights
w0 += _histogram[i];
int w1 = total - w0;
if (w0 == 0) {
continue; // skip leading zeros
}
if (w1 == 0) {
break; // skip trailing zeros
}
// update means
sum0 += (float)i * _histogram[i];
float sum1 = sum - sum0;
float m0 = sum0 / (float)w0;
float m1 = sum1 / (float)w1;
// between-class variance
float variance = (float)w0 * (float)w1 * (m0 - m1) * (m0 - m1);
// update threshold
if (variance > max) {
max = variance;
index = i;
}
}
return index;
}
//
// Process the histogram to update the adaptive threshold
//
void GateImpl::processHistogram(int numFrames) {
const int32_t LOG2E_Q26 = (int32_t)(log2(exp(1.0)) * (1 << LOG2_FRACBITS) + 0.5);
// compute time constants, for sampleRate downsampled by numFrames
int32_t tcHistogram = fixexp2(MULDIV64(numFrames, LOG2E_Q26, _sampleRate * 10)); // 10 seconds
int32_t tcThreshold = fixexp2(MULDIV64(numFrames, LOG2E_Q26, _sampleRate * 1)); // 1 second
// add bias at the fixed threshold
updateHistogram(_threshFixed, (numFrames+7)/8);
// leaky integrate into long-term histogram
for (int i = 0; i < NHIST; i++) {
_histogram[i] = _update[i] + MULQ31((_histogram[i] - _update[i]), tcHistogram);
}
// compute new threshold
int index = partitionHistogram();
int32_t threshold = ((NHIST-1) - index) << (LOG2_FRACBITS - 3);
// smooth threshold update
_threshAdapt = threshold + MULQ31((_threshAdapt - threshold), tcThreshold);
//printf("threshold = %0.1f\n", (_threshAdapt - (LOG2_HEADROOM_Q15 << LOG2_FRACBITS)) * -6.02f / (1 << LOG2_FRACBITS));
}
//
// Gate detector peakhold
//
int32_t GateImpl::peakhold(int32_t peak) {
if (peak > _holdPeak) {
// RELEASE
// 3-stage progressive hold
//
// (_holdRel > 0x7fffffff) pure hold
// (_holdRel > _holdMin) progressive hold
// (_holdRel = _holdMin) release
_holdRel += _holdInc; // update progressive hold
_holdRel = MAX((uint32_t)_holdRel, (uint32_t)_holdMin); // saturate at final value
int32_t tc = MIN((uint32_t)_holdRel, 0x7fffffff);
peak += MULQ31((_holdPeak - peak), tc); // apply release
} else {
// ATTACK
_holdRel = _holdMax; // reset release
}
_holdPeak = peak;
return peak;
}
//
// Gate hysteresis
// Implemented as detector hysteresis instead of high/low thresholds, to simplify adaptive threshold.
//
int32_t GateImpl::hysteresis(int32_t peak) {
// by construction, cannot overflow/underflow
assert((double)_hystOffset + (peak - _hystPeak) <= +(double)0x7fffffff);
assert((double)_hystOffset + (peak - _hystPeak) >= -(double)0x80000000);
// accumulate the offset, with saturation
_hystOffset += peak - _hystPeak;
_hystOffset = MIN(MAX(_hystOffset, 0), _hysteresis);
_hystPeak = peak;
peak -= _hystOffset; // apply hysteresis
assert(peak >= 0);
return peak;
}
//
// Gate envelope processing
// zero attack, fixed release
//
int32_t GateImpl::envelope(int32_t attn) {
if (attn > _attn) {
attn += MULQ31((_attn - attn), _release); // apply release
}
_attn = attn;
return attn;
}
//
// Gate (mono)
//
template<int N>
class GateMono : public GateImpl {
MonoDCBlock _dc;
MaxFilter<N> _filter;
MonoDelay<N, int32_t> _delay;
public:
GateMono(int sampleRate) : GateImpl(sampleRate) {}
// mono input/output (in-place is allowed)
void process(int16_t* input, int16_t* output, int numFrames) override;
void removeDC(int16_t* input, int16_t* output, int numFrames) override;
};
template<int N>
void GateMono<N>::process(int16_t* input, int16_t* output, int numFrames) {
clearHistogram();
for (int n = 0; n < numFrames; n++) {
int32_t x = input[n];
// remove DC
_dc.process(x);
// peak detect
int32_t peak = abs(x);
// convert to log2 domain
peak = fixlog2(peak);
// apply peak hold
peak = peakhold(peak);
// count peak level
updateHistogram(peak);
// apply hysteresis
peak = hysteresis(peak);
// compute gate attenuation
int32_t attn = (peak > _threshAdapt) ? 0x7fffffff : 0; // hard-knee, 1:inf ratio
// apply envelope
attn = envelope(attn);
// convert from log2 domain
attn = fixexp2(attn);
// lowpass filter
attn = _filter.process(attn);
// delay audio
_delay.process(x);
// apply gain
x = MULQ31(x, attn);
// store 16-bit output
output[n] = (int16_t)saturateQ30(x);
}
// update adaptive threshold
processHistogram(numFrames);
}
template<int N>
void GateMono<N>::removeDC(int16_t* input, int16_t* output, int numFrames) {
for (int n = 0; n < numFrames; n++) {
int32_t x = input[n];
// remove DC
_dc.process(x);
// store 16-bit output
output[n] = (int16_t)saturateQ30(x);
}
}
//
// Gate (stereo)
//
template<int N>
class GateStereo : public GateImpl {
StereoDCBlock _dc;
MaxFilter<N> _filter;
StereoDelay<N, int32_t> _delay;
public:
GateStereo(int sampleRate) : GateImpl(sampleRate) {}
// interleaved stereo input/output (in-place is allowed)
void process(int16_t* input, int16_t* output, int numFrames) override;
void removeDC(int16_t* input, int16_t* output, int numFrames) override;
};
template<int N>
void GateStereo<N>::process(int16_t* input, int16_t* output, int numFrames) {
clearHistogram();
for (int n = 0; n < numFrames; n++) {
int32_t x0 = input[2*n+0];
int32_t x1 = input[2*n+1];
// remove DC
_dc.process(x0, x1);
// peak detect
int32_t peak = MAX(abs(x0), abs(x1));
// convert to log2 domain
peak = fixlog2(peak);
// apply peak hold
peak = peakhold(peak);
// count peak level
updateHistogram(peak);
// apply hysteresis
peak = hysteresis(peak);
// compute gate attenuation
int32_t attn = (peak > _threshAdapt) ? 0x7fffffff : 0; // hard-knee, 1:inf ratio
// apply envelope
attn = envelope(attn);
// convert from log2 domain
attn = fixexp2(attn);
// lowpass filter
attn = _filter.process(attn);
// delay audio
_delay.process(x0, x1);
// apply gain
x0 = MULQ31(x0, attn);
x1 = MULQ31(x1, attn);
// store 16-bit output
output[2*n+0] = (int16_t)saturateQ30(x0);
output[2*n+1] = (int16_t)saturateQ30(x1);
}
// update adaptive threshold
processHistogram(numFrames);
}
template<int N>
void GateStereo<N>::removeDC(int16_t* input, int16_t* output, int numFrames) {
for (int n = 0; n < numFrames; n++) {
int32_t x0 = input[2*n+0];
int32_t x1 = input[2*n+1];
// remove DC
_dc.process(x0, x1);
// store 16-bit output
output[2*n+0] = (int16_t)saturateQ30(x0);
output[2*n+1] = (int16_t)saturateQ30(x1);
}
}
//
// Gate (quad)
//
template<int N>
class GateQuad : public GateImpl {
QuadDCBlock _dc;
MaxFilter<N> _filter;
QuadDelay<N, int32_t> _delay;
public:
GateQuad(int sampleRate) : GateImpl(sampleRate) {}
// interleaved quad input/output (in-place is allowed)
void process(int16_t* input, int16_t* output, int numFrames) override;
void removeDC(int16_t* input, int16_t* output, int numFrames) override;
};
template<int N>
void GateQuad<N>::process(int16_t* input, int16_t* output, int numFrames) {
clearHistogram();
for (int n = 0; n < numFrames; n++) {
int32_t x0 = input[4*n+0];
int32_t x1 = input[4*n+1];
int32_t x2 = input[4*n+2];
int32_t x3 = input[4*n+3];
// remove DC
_dc.process(x0, x1, x2, x3);
// peak detect
int32_t peak = MAX(MAX(abs(x0), abs(x1)), MAX(abs(x2), abs(x3)));
// convert to log2 domain
peak = fixlog2(peak);
// apply peak hold
peak = peakhold(peak);
// count peak level
updateHistogram(peak);
// apply hysteresis
peak = hysteresis(peak);
// compute gate attenuation
int32_t attn = (peak > _threshAdapt) ? 0x7fffffff : 0; // hard-knee, 1:inf ratio
// apply envelope
attn = envelope(attn);
// convert from log2 domain
attn = fixexp2(attn);
// lowpass filter
attn = _filter.process(attn);
// delay audio
_delay.process(x0, x1, x2, x3);
// apply gain
x0 = MULQ31(x0, attn);
x1 = MULQ31(x1, attn);
x2 = MULQ31(x2, attn);
x3 = MULQ31(x3, attn);
// store 16-bit output
output[4*n+0] = (int16_t)saturateQ30(x0);
output[4*n+1] = (int16_t)saturateQ30(x1);
output[4*n+2] = (int16_t)saturateQ30(x2);
output[4*n+3] = (int16_t)saturateQ30(x3);
}
// update adaptive threshold
processHistogram(numFrames);
}
template<int N>
void GateQuad<N>::removeDC(int16_t* input, int16_t* output, int numFrames) {
for (int n = 0; n < numFrames; n++) {
int32_t x0 = input[4*n+0];
int32_t x1 = input[4*n+1];
int32_t x2 = input[4*n+2];
int32_t x3 = input[4*n+3];
// remove DC
_dc.process(x0, x1, x2, x3);
// store 16-bit output
output[4*n+0] = (int16_t)saturateQ30(x0);
output[4*n+1] = (int16_t)saturateQ30(x1);
output[4*n+2] = (int16_t)saturateQ30(x2);
output[4*n+3] = (int16_t)saturateQ30(x3);
}
}
//
// Public API
//
AudioGate::AudioGate(int sampleRate, int numChannels) {
if (numChannels == 1) {
// ~3ms lookahead for all rates
if (sampleRate < 16000) {
_impl = new GateMono<32>(sampleRate);
} else if (sampleRate < 32000) {
_impl = new GateMono<64>(sampleRate);
} else if (sampleRate < 64000) {
_impl = new GateMono<128>(sampleRate);
} else {
_impl = new GateMono<256>(sampleRate);
}
} else if (numChannels == 2) {
// ~3ms lookahead for all rates
if (sampleRate < 16000) {
_impl = new GateStereo<32>(sampleRate);
} else if (sampleRate < 32000) {
_impl = new GateStereo<64>(sampleRate);
} else if (sampleRate < 64000) {
_impl = new GateStereo<128>(sampleRate);
} else {
_impl = new GateStereo<256>(sampleRate);
}
} else if (numChannels == 4) {
// ~3ms lookahead for all rates
if (sampleRate < 16000) {
_impl = new GateQuad<32>(sampleRate);
} else if (sampleRate < 32000) {
_impl = new GateQuad<64>(sampleRate);
} else if (sampleRate < 64000) {
_impl = new GateQuad<128>(sampleRate);
} else {
_impl = new GateQuad<256>(sampleRate);
}
} else {
assert(0); // unsupported
}
}
AudioGate::~AudioGate() {
delete _impl;
}
void AudioGate::render(int16_t* input, int16_t* output, int numFrames) {
_impl->process(input, output, numFrames);
}
void AudioGate::removeDC(int16_t* input, int16_t* output, int numFrames) {
_impl->removeDC(input, output, numFrames);
}
void AudioGate::setThreshold(float threshold) {
_impl->setThreshold(threshold);
}
void AudioGate::setRelease(float release) {
_impl->setRelease(release);
}

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@ -0,0 +1,32 @@
//
// AudioGate.h
// libraries/audio/src
//
// Created by Ken Cooke on 5/5/17.
// Copyright 2016 High Fidelity, Inc.
//
#ifndef hifi_AudioGate_h
#define hifi_AudioGate_h
#include <stdint.h>
class GateImpl;
class AudioGate {
public:
AudioGate(int sampleRate, int numChannels);
~AudioGate();
// interleaved int16_t input/output (in-place is allowed)
void render(int16_t* input, int16_t* output, int numFrames);
void removeDC(int16_t* input, int16_t* output, int numFrames);
void setThreshold(float threshold);
void setRelease(float release);
private:
GateImpl* _impl;
};
#endif // hifi_AudioGate_h

View file

@ -139,7 +139,7 @@ bool AudioInjector::inject(bool(AudioInjectorManager::*injection)(AudioInjector*
if (_options.secondOffset > 0.0f) {
int numChannels = _options.ambisonic ? 4 : (_options.stereo ? 2 : 1);
byteOffset = (int)(AudioConstants::SAMPLE_RATE * _options.secondOffset * numChannels);
byteOffset *= sizeof(AudioConstants::SAMPLE_SIZE);
byteOffset *= AudioConstants::SAMPLE_SIZE;
}
_currentSendOffset = byteOffset;

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@ -6,452 +6,11 @@
// Copyright 2016 High Fidelity, Inc.
//
#include <math.h>
#include <assert.h>
#include "AudioDynamics.h"
#include "AudioLimiter.h"
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif
#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
#ifdef _MSC_VER
#include <intrin.h>
#define MUL64(a,b) __emul((a), (b))
#define MULHI(a,b) ((int)(MUL64(a, b) >> 32))
#define MULQ31(a,b) ((int)(MUL64(a, b) >> 31))
#else
#define MUL64(a,b) ((long long)(a) * (b))
#define MULHI(a,b) ((int)(MUL64(a, b) >> 32))
#define MULQ31(a,b) ((int)(MUL64(a, b) >> 31))
#endif // _MSC_VER
//
// on x86 architecture, assume that SSE2 is present
//
#if defined(_M_IX86) || defined(_M_X64) || defined(__i386__) || defined(__x86_64__)
#include <xmmintrin.h>
// convert float to int using round-to-nearest
static inline int32_t floatToInt(float x) {
return _mm_cvt_ss2si(_mm_load_ss(&x));
}
#else
// convert float to int using round-to-nearest
static inline int32_t floatToInt(float x) {
x += (x < 0.0f ? -0.5f : 0.5f); // round
return (int32_t)x;
}
#endif // _M_IX86
static const double FIXQ31 = 2147483648.0; // convert float to Q31
static const double DB_TO_LOG2 = 0.16609640474436813; // convert dB to log2
// convert dB to amplitude
static double dBToGain(double dB) {
return pow(10.0, dB / 20.0);
}
// convert milliseconds to first-order time constant
static int32_t msToTc(double ms, double sampleRate) {
double tc = exp(-1000.0 / (ms * sampleRate));
return (int32_t)(FIXQ31 * tc); // Q31
}
// log2 domain values are Q26
static const int LOG2_INTBITS = 5;
static const int LOG2_FRACBITS = 31 - LOG2_INTBITS;
// log2 domain headroom bits above 0dB
static const int LOG2_HEADROOM = 15;
// log2 domain offsets so error < 0
static const int32_t LOG2_BIAS = 347;
static const int32_t EXP2_BIAS = 64;
//
// P(x) = log2(1+x) for x=[0,1]
// scaled by 1, 0.5, 0.25
//
// |error| < 347 ulp, smooth
//
static const int LOG2_TABBITS = 4;
static const int32_t log2Table[1 << LOG2_TABBITS][3] = {
{ -0x56dfe26d, 0x5c46daff, 0x00000000 },
{ -0x4d397571, 0x5bae58e7, 0x00025a75 },
{ -0x4518f84b, 0x5aabcac4, 0x000a62db },
{ -0x3e3075ec, 0x596168c0, 0x0019d0e6 },
{ -0x384486e9, 0x57e769c7, 0x00316109 },
{ -0x332742ba, 0x564f1461, 0x00513776 },
{ -0x2eb4bad4, 0x54a4cdfe, 0x00791de2 },
{ -0x2ad07c6c, 0x52f18320, 0x00a8aa46 },
{ -0x2763c4d6, 0x513ba123, 0x00df574c },
{ -0x245c319b, 0x4f87c5c4, 0x011c9399 },
{ -0x21aac79f, 0x4dd93bef, 0x015fcb52 },
{ -0x1f433872, 0x4c325584, 0x01a86ddc },
{ -0x1d1b54b4, 0x4a94ac6e, 0x01f5f13e },
{ -0x1b2a9f81, 0x4901524f, 0x0247d3f2 },
{ -0x1969fa57, 0x4778f3a7, 0x029d9dbf },
{ -0x17d36370, 0x45fbf1e8, 0x02f6dfe8 },
};
//
// P(x) = exp2(x) for x=[0,1]
// scaled by 2, 1, 0.5
// Uses exp2(-x) = exp2(1-x)/2
//
// |error| < 1387 ulp, smooth
//
static const int EXP2_TABBITS = 4;
static const int32_t exp2Table[1 << EXP2_TABBITS][3] = {
{ 0x3ed838c8, 0x58b574b7, 0x40000000 },
{ 0x41a0821c, 0x5888db8f, 0x4000b2b7 },
{ 0x4488548d, 0x582bcbc6, 0x40039be1 },
{ 0x4791158a, 0x579a1128, 0x400a71ae },
{ 0x4abc3a53, 0x56cf3089, 0x4017212e },
{ 0x4e0b48af, 0x55c66396, 0x402bd31b },
{ 0x517fd7a7, 0x547a946d, 0x404af0ec },
{ 0x551b9049, 0x52e658f9, 0x40772a57 },
{ 0x58e02e75, 0x5103ee08, 0x40b37b31 },
{ 0x5ccf81b1, 0x4ecd321f, 0x410331b5 },
{ 0x60eb6e09, 0x4c3ba007, 0x4169f548 },
{ 0x6535ecf9, 0x49484909, 0x41ebcdaf },
{ 0x69b10e5b, 0x45ebcede, 0x428d2acd },
{ 0x6e5ef96c, 0x421e5d48, 0x4352ece7 },
{ 0x7341edcb, 0x3dd7a354, 0x44426d7b },
{ 0x785c4499, 0x390ecc3a, 0x456188bd },
};
static const int IEEE754_FABS_MASK = 0x7fffffff;
static const int IEEE754_MANT_BITS = 23;
static const int IEEE754_EXPN_BIAS = 127;
//
// Peak detection and -log2(x) for float input (mono)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input) {
// float as integer bits
int32_t u = *(int32_t*)input;
// absolute value
int32_t peak = u & IEEE754_FABS_MASK;
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Peak detection and -log2(x) for float input (stereo)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input0, float* input1) {
// float as integer bits
int32_t u0 = *(int32_t*)input0;
int32_t u1 = *(int32_t*)input1;
// max absolute value
u0 &= IEEE754_FABS_MASK;
u1 &= IEEE754_FABS_MASK;
int32_t peak = MAX(u0, u1);
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Peak detection and -log2(x) for float input (quad)
// x < 2^(31-LOG2_HEADROOM) returns 0x7fffffff
// x > 2^LOG2_HEADROOM undefined
//
static inline int32_t peaklog2(float* input0, float* input1, float* input2, float* input3) {
// float as integer bits
int32_t u0 = *(int32_t*)input0;
int32_t u1 = *(int32_t*)input1;
int32_t u2 = *(int32_t*)input2;
int32_t u3 = *(int32_t*)input3;
// max absolute value
u0 &= IEEE754_FABS_MASK;
u1 &= IEEE754_FABS_MASK;
u2 &= IEEE754_FABS_MASK;
u3 &= IEEE754_FABS_MASK;
int32_t peak = MAX(MAX(u0, u1), MAX(u2, u3));
// split into e and x - 1.0
int32_t e = IEEE754_EXPN_BIAS - (peak >> IEEE754_MANT_BITS) + LOG2_HEADROOM;
int32_t x = (peak << (31 - IEEE754_MANT_BITS)) & 0x7fffffff;
// saturate
if (e > 31) {
return 0x7fffffff;
}
int k = x >> (31 - LOG2_TABBITS);
// polynomial for log2(1+x) over x=[0,1]
int32_t c0 = log2Table[k][0];
int32_t c1 = log2Table[k][1];
int32_t c2 = log2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q26
return (e << LOG2_FRACBITS) - (c2 >> 3);
}
//
// Compute exp2(-x) for x=[0,32] in Q26, result in Q31
// x < 0 undefined
//
static inline int32_t fixexp2(int32_t x) {
// split into e and 1.0 - x
int32_t e = x >> LOG2_FRACBITS;
x = ~(x << LOG2_INTBITS) & 0x7fffffff;
int k = x >> (31 - EXP2_TABBITS);
// polynomial for exp2(x)
int32_t c0 = exp2Table[k][0];
int32_t c1 = exp2Table[k][1];
int32_t c2 = exp2Table[k][2];
c1 += MULHI(c0, x);
c2 += MULHI(c1, x);
// reconstruct result in Q31
return c2 >> e;
}
// fast TPDF dither in [-1.0f, 1.0f]
static inline float dither() {
static uint32_t rz = 0;
rz = rz * 69069 + 1;
int32_t r0 = rz & 0xffff;
int32_t r1 = rz >> 16;
return (int32_t)(r0 - r1) * (1/65536.0f);
}
//
// Peak-hold lowpass filter
//
// Bandlimits the gain control signal to greatly reduce the modulation distortion,
// while still reaching the peak attenuation after exactly N-1 samples of delay.
// N completely determines the limiter attack time.
//
template<int N, int CIC1, int CIC2>
class PeakFilterT {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
static_assert((CIC1 - 1) + (CIC2 - 1) == (N - 1), "Total CIC delay must be N-1");
int32_t _buffer[2*N] = {}; // shared FIFO
size_t _index = 0;
int32_t _acc1 = 0; // CIC1 integrator
int32_t _acc2 = 0; // CIC2 integrator
public:
PeakFilterT() {
// fill history
for (size_t n = 0; n < N-1; n++) {
process(0x7fffffff);
}
}
int32_t process(int32_t x) {
const size_t MASK = 2*N - 1; // buffer wrap
size_t i = _index;
// Fast peak-hold using a running-min filter. Finds the peak (min) value
// in the sliding window of N-1 samples, using only log2(N) comparisons.
// Hold time of N-1 samples exactly cancels the step response of FIR filter.
for (size_t n = 1; n < N; n <<= 1) {
_buffer[i] = x;
i = (i + n) & MASK;
x = MIN(x, _buffer[i]);
}
// Fast FIR attack/lowpass filter using a 2-stage CIC filter.
// The step response reaches final value after N-1 samples.
const int32_t CICGAIN = 0xffffffff / (CIC1 * CIC2); // Q32
x = MULHI(x, CICGAIN);
_buffer[i] = _acc1;
_acc1 += x; // integrator
i = (i + CIC1 - 1) & MASK;
x = _acc1 - _buffer[i]; // comb
_buffer[i] = _acc2;
_acc2 += x; // integrator
i = (i + CIC2 - 1) & MASK;
x = _acc2 - _buffer[i]; // comb
_index = (i + 1) & MASK; // skip unused tap
return x;
}
};
//
// Specializations that define the optimum lowpass filter for each length.
//
template<int N> class PeakFilter;
template<> class PeakFilter< 16> : public PeakFilterT< 16, 7, 10> {};
template<> class PeakFilter< 32> : public PeakFilterT< 32, 14, 19> {};
template<> class PeakFilter< 64> : public PeakFilterT< 64, 27, 38> {};
template<> class PeakFilter<128> : public PeakFilterT<128, 53, 76> {};
template<> class PeakFilter<256> : public PeakFilterT<256, 106, 151> {};
//
// N-1 sample delay (mono)
//
template<int N>
class MonoDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
float _buffer[N] = {};
size_t _index = 0;
public:
void process(float& x) {
const size_t MASK = N - 1; // buffer wrap
size_t i = _index;
_buffer[i] = x;
i = (i + (N - 1)) & MASK;
x = _buffer[i];
_index = i;
}
};
//
// N-1 sample delay (stereo)
//
template<int N>
class StereoDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
float _buffer[2*N] = {};
size_t _index = 0;
public:
void process(float& x0, float& x1) {
const size_t MASK = 2*N - 1; // buffer wrap
size_t i = _index;
_buffer[i+0] = x0;
_buffer[i+1] = x1;
i = (i + 2*(N - 1)) & MASK;
x0 = _buffer[i+0];
x1 = _buffer[i+1];
_index = i;
}
};
//
// N-1 sample delay (quad)
//
template<int N>
class QuadDelay {
static_assert((N & (N - 1)) == 0, "N must be a power of 2");
float _buffer[4*N] = {};
size_t _index = 0;
public:
void process(float& x0, float& x1, float& x2, float& x3) {
const size_t MASK = 4*N - 1; // buffer wrap
size_t i = _index;
_buffer[i+0] = x0;
_buffer[i+1] = x1;
_buffer[i+2] = x2;
_buffer[i+3] = x3;
i = (i + 4*(N - 1)) & MASK;
x0 = _buffer[i+0];
x1 = _buffer[i+1];
x2 = _buffer[i+2];
x3 = _buffer[i+3];
_index = i;
}
};
//
// Limiter (common)
//
@ -637,7 +196,7 @@ int32_t LimiterImpl::envelope(int32_t attn) {
template<int N>
class LimiterMono : public LimiterImpl {
PeakFilter<N> _filter;
MinFilter<N> _filter;
MonoDelay<N> _delay;
public:
@ -688,7 +247,7 @@ void LimiterMono<N>::process(float* input, int16_t* output, int numFrames) {
template<int N>
class LimiterStereo : public LimiterImpl {
PeakFilter<N> _filter;
MinFilter<N> _filter;
StereoDelay<N> _delay;
public:
@ -745,7 +304,7 @@ void LimiterStereo<N>::process(float* input, int16_t* output, int numFrames) {
template<int N>
class LimiterQuad : public LimiterImpl {
PeakFilter<N> _filter;
MinFilter<N> _filter;
QuadDelay<N> _delay;
public:

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@ -9,7 +9,7 @@
#ifndef hifi_AudioLimiter_h
#define hifi_AudioLimiter_h
#include "stdint.h"
#include <stdint.h>
class LimiterImpl;

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@ -1,164 +0,0 @@
//
// AudioNoiseGate.cpp
// libraries/audio
//
// Created by Stephen Birarda on 2014-12-16.
// Copyright 2014 High Fidelity, Inc.
//
// Distributed under the Apache License, Version 2.0.
// See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//
#include "AudioNoiseGate.h"
#include <cstdlib>
#include <string.h>
#include "AudioConstants.h"
const float AudioNoiseGate::CLIPPING_THRESHOLD = 0.90f;
AudioNoiseGate::AudioNoiseGate() :
_lastLoudness(0.0f),
_didClipInLastBlock(false),
_dcOffset(0.0f),
_measuredFloor(0.0f),
_sampleCounter(0),
_isOpen(false),
_blocksToClose(0) {}
void AudioNoiseGate::removeDCOffset(int16_t* samples, int numSamples) {
//
// DC Offset correction
//
// Measure the DC offset over a trailing number of blocks, and remove it from the input signal.
// This causes the noise background measurements and server muting to be more accurate. Many off-board
// ADC's have a noticeable DC offset.
//
const float DC_OFFSET_AVERAGING = 0.99f;
float measuredDcOffset = 0.0f;
// Remove trailing DC offset from samples
for (int i = 0; i < numSamples; i++) {
measuredDcOffset += samples[i];
samples[i] -= (int16_t) _dcOffset;
}
// Update measured DC offset
measuredDcOffset /= numSamples;
if (_dcOffset == 0.0f) {
// On first block, copy over measured offset
_dcOffset = measuredDcOffset;
} else {
_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.0f - DC_OFFSET_AVERAGING) * measuredDcOffset;
}
}
void AudioNoiseGate::gateSamples(int16_t* samples, int numSamples) {
//
// Impose Noise Gate
//
// The Noise Gate is used to reject constant background noise by measuring the noise
// floor observed at the microphone and then opening the 'gate' to allow microphone
// signals to be transmitted when the microphone samples average level exceeds a multiple
// of the noise floor.
//
// NOISE_GATE_HEIGHT: How loud you have to speak relative to noise background to open the gate.
// Make this value lower for more sensitivity and less rejection of noise.
// NOISE_GATE_WIDTH: The number of samples in an audio block for which the height must be exceeded
// to open the gate.
// NOISE_GATE_CLOSE_BLOCK_DELAY: Once the noise is below the gate height for the block, how many blocks
// will we wait before closing the gate.
// NOISE_GATE_BLOCKS_TO_AVERAGE: How many audio blocks should we average together to compute noise floor.
// More means better rejection but also can reject continuous things like singing.
// NUMBER_OF_NOISE_SAMPLE_BLOCKS: How often should we re-evaluate the noise floor?
float loudness = 0;
int thisSample = 0;
int samplesOverNoiseGate = 0;
const float NOISE_GATE_HEIGHT = 7.0f;
const int NOISE_GATE_WIDTH = 5;
const int NOISE_GATE_CLOSE_BLOCK_DELAY = 5;
const int NOISE_GATE_BLOCKS_TO_AVERAGE = 5;
// Check clipping, and check if should open noise gate
_didClipInLastBlock = false;
for (int i = 0; i < numSamples; i++) {
thisSample = std::abs(samples[i]);
if (thisSample >= ((float) AudioConstants::MAX_SAMPLE_VALUE * CLIPPING_THRESHOLD)) {
_didClipInLastBlock = true;
}
loudness += thisSample;
// Noise Reduction: Count peaks above the average loudness
if (thisSample > (_measuredFloor * NOISE_GATE_HEIGHT)) {
samplesOverNoiseGate++;
}
}
_lastLoudness = fabs(loudness / numSamples);
// If Noise Gate is enabled, check and turn the gate on and off
float averageOfAllSampleBlocks = 0.0f;
_sampleBlocks[_sampleCounter++] = _lastLoudness;
if (_sampleCounter == NUMBER_OF_NOISE_SAMPLE_BLOCKS) {
float smallestSample = std::numeric_limits<float>::max();
for (int i = 0; i <= NUMBER_OF_NOISE_SAMPLE_BLOCKS - NOISE_GATE_BLOCKS_TO_AVERAGE; i += NOISE_GATE_BLOCKS_TO_AVERAGE) {
float thisAverage = 0.0f;
for (int j = i; j < i + NOISE_GATE_BLOCKS_TO_AVERAGE; j++) {
thisAverage += _sampleBlocks[j];
averageOfAllSampleBlocks += _sampleBlocks[j];
}
thisAverage /= NOISE_GATE_BLOCKS_TO_AVERAGE;
if (thisAverage < smallestSample) {
smallestSample = thisAverage;
}
}
averageOfAllSampleBlocks /= NUMBER_OF_NOISE_SAMPLE_BLOCKS;
_measuredFloor = smallestSample;
_sampleCounter = 0;
}
_closedInLastBlock = false;
_openedInLastBlock = false;
if (samplesOverNoiseGate > NOISE_GATE_WIDTH) {
_openedInLastBlock = !_isOpen;
_isOpen = true;
_blocksToClose = NOISE_GATE_CLOSE_BLOCK_DELAY;
} else {
if (--_blocksToClose == 0) {
_closedInLastBlock = _isOpen;
_isOpen = false;
}
}
if (!_isOpen) {
// First block after being closed gets faded to silence, we fade across
// the entire block on fading out. All subsequent blocks are muted by being slammed
// to zeros
if (_closedInLastBlock) {
float fadeSlope = (1.0f / numSamples);
for (int i = 0; i < numSamples; i++) {
float fadedSample = (1.0f - ((float)i * fadeSlope)) * (float)samples[i];
samples[i] = (int16_t)fadedSample;
}
} else {
memset(samples, 0, numSamples * sizeof(int16_t));
}
_lastLoudness = 0;
}
if (_openedInLastBlock) {
// would be nice to do a little crossfade from silence, but we only want to fade
// across the first 1/10th of the block, because we don't want to miss early
// transients.
int fadeSamples = numSamples / 10; // fade over 1/10th of the samples
float fadeSlope = (1.0f / fadeSamples);
for (int i = 0; i < fadeSamples; i++) {
float fadedSample = (float)i * fadeSlope * (float)samples[i];
samples[i] = (int16_t)fadedSample;
}
}
}

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@ -1,48 +0,0 @@
//
// AudioNoiseGate.h
// libraries/audio
//
// Created by Stephen Birarda on 2014-12-16.
// Copyright 2014 High Fidelity, Inc.
//
// Distributed under the Apache License, Version 2.0.
// See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//
#ifndef hifi_AudioNoiseGate_h
#define hifi_AudioNoiseGate_h
#include <stdint.h>
const int NUMBER_OF_NOISE_SAMPLE_BLOCKS = 300;
class AudioNoiseGate {
public:
AudioNoiseGate();
void gateSamples(int16_t* samples, int numSamples);
void removeDCOffset(int16_t* samples, int numSamples);
bool clippedInLastBlock() const { return _didClipInLastBlock; }
bool closedInLastBlock() const { return _closedInLastBlock; }
bool openedInLastBlock() const { return _openedInLastBlock; }
bool isOpen() const { return _isOpen; }
float getMeasuredFloor() const { return _measuredFloor; }
float getLastLoudness() const { return _lastLoudness; }
static const float CLIPPING_THRESHOLD;
private:
float _lastLoudness;
bool _didClipInLastBlock;
float _dcOffset;
float _measuredFloor;
float _sampleBlocks[NUMBER_OF_NOISE_SAMPLE_BLOCKS];
int _sampleCounter;
bool _isOpen;
bool _closedInLastBlock { false };
bool _openedInLastBlock { false };
int _blocksToClose;
};
#endif // hifi_AudioNoiseGate_h