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Move_clipping_to_audiomixer #221

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1 change: 0 additions & 1 deletion sources/Application/Audio/DummyAudioOut.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -53,7 +53,6 @@ void DummyAudioOut::Stop() {
SAFE_DELETE(thread_) ;
} ;

void DummyAudioOut::SetSoftclip(int clip, int gain) {}
void DummyAudioOut::SetMasterVolume(int volume) {}

void DummyAudioOut::SendPulse()
Expand Down
5 changes: 1 addition & 4 deletions sources/Application/Audio/DummyAudioOut.h
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Original file line number Diff line number Diff line change
Expand Up @@ -28,8 +28,6 @@ class DummyAudioOut: public AudioOut {

virtual void Trigger();

virtual bool Clipped() { return false; };

virtual int GetPlayedBufferPercentage() { return 0; };

void SendPulse();
Expand All @@ -39,8 +37,7 @@ class DummyAudioOut: public AudioOut {
virtual int GetAudioBufferSize() ;
virtual int GetAudioRequestedBufferSize() ;
virtual int GetAudioPreBufferCount() ;
virtual double GetStreamTime() ;
virtual void SetSoftclip(int, int);
virtual double GetStreamTime();
virtual void SetMasterVolume(int);

private:
Expand Down
98 changes: 92 additions & 6 deletions sources/Services/Audio/AudioMixer.cpp
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Original file line number Diff line number Diff line change
@@ -1,5 +1,9 @@
#include "AudioMixer.h"
#include "System/System/System.h"
#include <math.h>

#define MAX_POSITIVE_FIXED i2fp(32767)
#define MAX_NEGATIVE_FIXED i2fp(-32768)

AudioMixer::AudioMixer(const char *name):
T_SimpleList<AudioModule>(false),
Expand All @@ -8,6 +12,32 @@ AudioMixer::AudioMixer(const char *name):
name_(name)
{
volume_=(i2fp(1)) ;
softclip_ = -1;
softclipGain_ = 0 ;
masterVolume_ = 100 ;
clipped_ = false ;

// Precalculate constant values for softclipping algorithm
softClipData_[0].alpha = 1.45f; // -1.5db (approx.)
softClipData_[1].alpha = 1.07f; // -3db (approx.)
softClipData_[2].alpha = 0.75f; // -6db (approx.)
softClipData_[3].alpha = 0.53f; // -9db (approx.)

for (int i = 0; i < 4; i++) {
softClipData_[i].alpha23 = softClipData_[i].alpha * (2.0f / 3.0f);
softClipData_[i].alphaInv = 1.0f / softClipData_[i].alpha;

if (softClipData_[i].alpha > 1.0f) {
/* calculates gain compensation differently for
* modes with alpha > 1, so there's no drop in loudness
* and we can still drive the hard clipper when the input
* goes over 1.0
*/
softClipData_[i].gainCmp = 1.0f / (1.0f - (pow(softClipData_[i].alphaInv, 2.0f) / 3.0f));
} else {
softClipData_[i].gainCmp = 1.0f / softClipData_[i].alpha23;
}
}
} ;

AudioMixer::~AudioMixer() {
Expand Down Expand Up @@ -35,13 +65,14 @@ void AudioMixer::EnableRendering(bool enable) {
} ;

bool AudioMixer::Render(fixed *buffer,int samplecount) {
clipped_ = false;

fixed *mixBuffer=0 ;
bool gotData=false ;
IteratorPtr<AudioModule>it(GetIterator()) ;
for (it->Begin();!it->IsDone();it->Next()) {
AudioModule &current=it->CurrentItem() ;
if (!gotData) {
fixed *mixBuffer = 0;
bool gotData = false;
IteratorPtr<AudioModule> it(GetIterator());
for (it->Begin(); !it->IsDone(); it->Next()) {
AudioModule &current = it->CurrentItem();
if (!gotData) {
gotData=current.Render(buffer,samplecount) ;
} else {
if (!mixBuffer) {
Expand All @@ -64,12 +95,22 @@ bool AudioMixer::Render(fixed *buffer,int samplecount) {

if (gotData) {
fixed *c = buffer;
float damp = pow((float)masterVolume_ / 100, 4.0f);

if (volume_ != i2fp(1)) {
for (int i = 0; i < samplecount * 2; i++) {
fixed v = fp_mul(*c, volume_);
*c++ = v;
}
}

// Apply soft/hard clipping before recording
c = buffer;
for (int i = 0; i < samplecount * 2; i++) {
fixed sample = *c;
sample = fl2fp(damp * fp2fl(hardClip(softClip(sample))));
*c++ = sample;
}
}
if (enableRendering_&&writer_) {
if (!gotData) {
Expand All @@ -82,3 +123,48 @@ bool AudioMixer::Render(fixed *buffer,int samplecount) {
} ;

void AudioMixer::SetVolume(fixed volume) { volume_ = volume; }

void AudioMixer::SetSoftclip(int clip, int gain) {
softclip_ = clip - 1;
softclipGain_ = gain;
}

void AudioMixer::SetMasterVolume(int volume) {
masterVolume_ = volume;
}

bool AudioMixer::Clipped() { return clipped_; }

fixed AudioMixer::hardClip(fixed sample) {
if (sample > MAX_POSITIVE_FIXED || sample < MAX_NEGATIVE_FIXED) {
clipped_ = true;
return sample > 0 ? MAX_POSITIVE_FIXED : MAX_NEGATIVE_FIXED;
}
return sample;
}

/* Implements standard cubic algorithm
* https://wiki.analog.com/resources/tools-software/sigmastudio/toolbox/nonlinearprocessors/standardcubic
*/
fixed AudioMixer::softClip(fixed sample) {
if (softclip_ == -1 || sample == 0)
return sample;

float x;
float sampleFloat = fp2fl(sample);
float maxFloat = fp2fl(sampleFloat > 0 ? MAX_POSITIVE_FIXED : MAX_NEGATIVE_FIXED);
SoftClipData* data = &softClipData_[softclip_];

x = data->alphaInv * (sampleFloat / maxFloat);
if (x > -1.0f && x < 1.0f) {
sampleFloat = maxFloat * (data->alpha * (x - (pow(x, 3.0f) / 3.0f)));
} else {
sampleFloat = maxFloat * data->alpha23;
}

if (softclipGain_) {
sampleFloat = sampleFloat * data->gainCmp;
}

return fl2fp(sampleFloat);
}
28 changes: 23 additions & 5 deletions sources/Services/Audio/AudioMixer.h
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Expand Up @@ -6,6 +6,13 @@
#include "Application/Instruments/WavFileWriter.h"
#include <string>

struct SoftClipData {
float alpha;
float alpha23;
float alphaInv;
float gainCmp;
};

class AudioMixer: public AudioModule,public T_SimpleList<AudioModule> {
public:
AudioMixer(const char *name) ;
Expand All @@ -14,11 +21,22 @@ class AudioMixer: public AudioModule,public T_SimpleList<AudioModule> {
void SetFileRenderer(const char *path) ;
void EnableRendering(bool enable) ;
void SetVolume(fixed volume) ;
virtual void SetSoftclip(int clip, int gain);
virtual void SetMasterVolume(int volume) ;
virtual bool Clipped() ;

private:
bool enableRendering_ ;
std::string renderPath_ ;
WavFileWriter *writer_ ;
fixed volume_ ;
std::string name_ ;
fixed hardClip(fixed sample);
fixed softClip(fixed sample);
bool enableRendering_;
std::string renderPath_;
WavFileWriter *writer_;
fixed volume_;
std::string name_;
SoftClipData softClipData_[4];
int softclip_;
int softclipGain_;
int masterVolume_;
bool clipped_;
} ;
#endif
3 changes: 0 additions & 3 deletions sources/Services/Audio/AudioOut.h
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Original file line number Diff line number Diff line change
Expand Up @@ -18,13 +18,10 @@ class AudioOut: public AudioMixer,public Observable {
virtual bool Start()=0 ;
virtual void Stop()=0 ;

virtual void SetSoftclip(int clip, int gain) = 0;
virtual void SetMasterVolume(int volume) = 0;

virtual void Trigger()=0 ;

virtual bool Clipped()=0 ;

virtual int GetPlayedBufferPercentage()=0 ;

virtual std::string GetAudioAPI()=0 ;
Expand Down
85 changes: 7 additions & 78 deletions sources/Services/Audio/AudioOutDriver.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -9,31 +9,9 @@
#include <math.h>

AudioOutDriver::AudioOutDriver(AudioDriver &driver) {
// Precalculate constant values for softclipping algorithm
softClipData_[0].alpha = 1.45f; // -1.5db (approx.)
softClipData_[1].alpha = 1.07f; // -3db (approx.)
softClipData_[2].alpha = 0.75f; // -6db (approx.)
softClipData_[3].alpha = 0.53f; // -9db (approx.)

for (int i = 0; i < 4; i++) {
softClipData_[i].alpha23 = softClipData_[i].alpha * (2.0f / 3.0f);
softClipData_[i].alphaInv = 1.0f / softClipData_[i].alpha;

if (softClipData_[i].alpha > 1.0f) {
/* calculates gain compensation differently for
* modes with alpha > 1, so there's no drop in loudness
* and we can still drive the hard clipper when the input
* goes over 1.0
*/
softClipData_[i].gainCmp = 1.0f / (1.0f - (pow(softClipData_[i].alphaInv, 2.0f) / 3.0f));
} else {
softClipData_[i].gainCmp = 1.0f / softClipData_[i].alpha23;
}
}

driver_=&driver ;
driver.AddObserver(*this) ;
primarySoundBuffer_=0 ;
driver_ = &driver;
driver.AddObserver(*this) ;
primarySoundBuffer_=0 ;
mixBuffer_ = 0;
SetOwnership(false);
}
Expand All @@ -56,7 +34,6 @@ void AudioOutDriver::Close() {
}

bool AudioOutDriver::Start() {
clipped_ = false;
sampleCount_=0 ;
return driver_->Start() ;
}
Expand All @@ -65,8 +42,6 @@ void AudioOutDriver::Stop() {
driver_->Stop() ;
}

bool AudioOutDriver::Clipped() { return clipped_; };

void AudioOutDriver::Trigger() {

TimeService *ts=TimeService::GetInstance() ;
Expand All @@ -84,58 +59,15 @@ void AudioOutDriver::Update(Observable &o,I_ObservableData *d)
}

void AudioOutDriver::prepareMixBuffers() {
sampleCount_=getPlaySampleCount() ;
clipped_=false ;
sampleCount_ = getPlaySampleCount();
} ;

void AudioOutDriver::SetSoftclip(int clip, int gain) {
softclip_ = clip - 1;
softclipGain_ = gain;
}

void AudioOutDriver::SetMasterVolume(int volume) {
masterVolume_ = volume;
}

fixed AudioOutDriver::hardClip(fixed sample) {

if (sample > MAX_POSITIVE_FIXED || sample < MAX_NEGATIVE_FIXED) {
clipped_ = true;
return sample > 0 ? MAX_POSITIVE_FIXED : MAX_NEGATIVE_FIXED;
}

return sample;
}

/* Implements standard cubic algorithm
* https://wiki.analog.com/resources/tools-software/sigmastudio/toolbox/nonlinearprocessors/standardcubic
*/
fixed AudioOutDriver::softClip(fixed sample) {
if (softclip_ == -1 || sample == 0)
return sample;

float x;
float sampleFloat = fp2fl(sample);
float maxFloat = fp2fl(sampleFloat > 0 ? MAX_POSITIVE_FIXED : MAX_NEGATIVE_FIXED);
SoftClipData* data = &softClipData_[softclip_];

x = data->alphaInv * (sampleFloat / maxFloat);
if (x > -1.0f && x < 1.0f) {
sampleFloat = maxFloat * (data->alpha * (x - (pow(x, 3.0f) / 3.0f)));
} else {
sampleFloat = maxFloat * data->alpha23;
}

if (softclipGain_) {
sampleFloat = sampleFloat * data->gainCmp;
}

return fl2fp(sampleFloat);
AudioMixer::SetMasterVolume(volume);
}

void AudioOutDriver::clipToMix() {

float damp = pow((float)masterVolume_ / 100, 4.0f);
bool interlaced = driver_->Interlaced();

if (!hasSound_) {
Expand All @@ -147,13 +79,10 @@ void AudioOutDriver::clipToMix() {

fixed *p = primarySoundBuffer_;

fixed leftSample;
fixed rightSample;

for (int i = 0; i < sampleCount_; i++) {

leftSample = damp * hardClip(softClip(*p++));
rightSample = damp * hardClip(softClip(*p++));
fixed leftSample = *p++;
fixed rightSample = *p++;

*s1 = short(fp2i(leftSample));
s1 += offset;
Expand Down
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