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Implement changes suggested by @rsta2

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Description:
https://github.com/probonopd/MiniDexed/pull/86#issuecomment-1097912829

Thanks @rsta2
pull/86/head
probonopd 3 years ago
parent f597359dc4
commit d6f70cdddd
5 changed files with 47 additions and 118 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
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  1. 77
      src/effect_compressor.cpp
  2. 29
      src/effect_mixer.hpp
  3. 32
      src/minidexed.cpp
  4. 3
      src/performanceconfig.cpp
  5. 24
      src/uimenu.cpp

77
src/effect_compressor.cpp
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@ -37,12 +37,7 @@ void Compressor::setDefaultValues(const float32_t sample_rate_Hz) {
void Compressor::calcInstantaneousTargetGain(float32_t *audio_level_dB_block, float32_t *inst_targ_gain_dB_block, uint16_t len)
{
// how much are we above the compression threshold?
float32_t* above_thresh_dB_block=(float32_t*)malloc(sizeof(float32_t)*len);
if(!above_thresh_dB_block)
{
LOGERR("Cannot allocate memory for \"above_thresh_dB_block\" - stopping\n");
while(1);
}
float32_t above_thresh_dB_block[len];
//arm_copy_f32(zeroblock_f32,above_thresh_dB_block,len);
@ -68,10 +63,6 @@ void Compressor::calcInstantaneousTargetGain(float32_t *audio_level_dB_block, fl
if (inst_targ_gain_dB_block[i] > 0.0f) inst_targ_gain_dB_block[i] = 0.0f;
}
// release memory before returning
if(above_thresh_dB_block)
delete(above_thresh_dB_block);
return; //output is passed through inst_targ_gain_dB_block
}
@ -105,12 +96,7 @@ void Compressor::calcSmoothedGain_dB(float32_t *inst_targ_gain_dB_block, float32
void Compressor::calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_block, uint16_t len) {
// calculate the instantaneous signal power (square the signal)
float32_t* wav_pow_block=(float32_t*)malloc(sizeof(float32_t)*len);
if(!wav_pow_block)
{
LOGERR("Cannot allocate memory for \"wav_pow_block\" - stopping\n");
while(1);
}
float32_t wav_pow_block[len];
//arm_copy_f32(zeroblock_f32,wav_pow_block,len);
@ -135,10 +121,6 @@ void Compressor::calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_blo
//scale the wav_pow_block by 10.0 to complete the conversion to dB
arm_scale_f32(level_dB_block, 10.0f, level_dB_block, len); //use ARM DSP for speed!
//release memory and return
if(wav_pow_block)
delete(wav_pow_block);
return; //output is passed through level_dB_block
}
@ -147,23 +129,13 @@ void Compressor::calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_blo
void Compressor::calcGain(float32_t *audio_level_dB_block, float32_t *gain_block,uint16_t len)
{
//first, calculate the instantaneous target gain based on the compression ratio
float32_t* inst_targ_gain_dB_block=(float32_t*)malloc(sizeof(float32_t)*len);
if(!inst_targ_gain_dB_block)
{
LOGERR("Cannot allocate memory for \"inst_targ_gain_dB_block\" - stopping\n");
while(1);
}
float32_t inst_targ_gain_dB_block[len];
//arm_copy_f32(zeroblock_f32,inst_targ_gain_dB_block,len);
calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block,len);
//second, smooth in time (attack and release) by stepping through each sample
float32_t *gain_dB_block = (float32_t*)malloc(sizeof(float32_t)*len);
if(!gain_dB_block)
{
LOGERR("Cannot allocate memory for \"gain_dB_block\" - stopping\n");
while(1);
}
float32_t gain_dB_block[len];
//arm_copy_f32(zeroblock_f32,gain_dB_block,len);
calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block, len);
@ -172,13 +144,6 @@ void Compressor::calcGain(float32_t *audio_level_dB_block, float32_t *gain_block
arm_scale_f32(gain_dB_block, 1.0f/20.0f, gain_dB_block, len); //divide by 20
for (uint16_t i = 0; i < len; i++) gain_block[i] = pow10f(gain_dB_block[i]); //do the 10^(x)
//release memory and return
if(inst_targ_gain_dB_block)
delete(inst_targ_gain_dB_block);
if(gain_dB_block)
delete(gain_dB_block);
return; //output is passed through gain_block
}
@ -186,7 +151,7 @@ void Compressor::calcGain(float32_t *audio_level_dB_block, float32_t *gain_block
void Compressor::doCompression(float32_t *audio_block, uint16_t len) {
//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
if (!audio_block) {
LOGERR("No audio_block available for Compressor!\n");
LOGERR("No audio_block available for Compressor!");
return;
}
@ -199,41 +164,21 @@ void Compressor::doCompression(float32_t *audio_block, uint16_t len) {
arm_scale_f32(audio_block, pre_gain, audio_block, len); //use ARM DSP for speed!
//calculate the level of the audio (ie, calculate a smoothed version of the signal power)
float32_t* audio_level_dB_block = (float32_t*)malloc(sizeof(float32_t)*len);
if(!audio_level_dB_block)
{
LOGERR("Cannot allocate memory for \"audio_level_dB_block\" - stopping\n");
while(1);
}
float32_t audio_level_dB_block[len];
//arm_copy_f32(zeroblock_f32,audio_level_dB_block,len);
if(audio_level_dB_block)
calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
//compute the desired gain based on the observed audio level
float32_t* gain_block=(float32_t*)malloc(sizeof(float32_t)*len);
if(!gain_block)
{
LOGERR("Cannot allocate memory for \"gain_block\" - stopping\n");
while(1);
}
float32_t gain_block[len];
//arm_copy_f32(zeroblock_f32,gain_block,len);
if(gain_block)
{
calcGain(audio_level_dB_block, gain_block, len); //returns through gain_block
//apply the desired gain...store the processed audio back into audio_block
arm_mult_f32(audio_block, gain_block, audio_block, len);
}
calcGain(audio_level_dB_block, gain_block, len); //returns through gain_block
//release memory
if(audio_level_dB_block)
delete(audio_level_dB_block);
if(gain_block)
delete(gain_block);
//apply the desired gain...store the processed audio back into audio_block
arm_mult_f32(audio_block, gain_block, audio_block, len);
}
//methods to set parameters of this module

29
src/effect_mixer.hpp
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@ -24,28 +24,24 @@ public:
for (uint8_t i=0; i<NN; i++)
multiplier[i] = UNITY_GAIN;
sumbufL=(float32_t*)malloc(sizeof(float32_t) * buffer_length);
sumbufL=new float32_t[buffer_length];
arm_fill_f32(0.0f, sumbufL, len);
}
~AudioMixer()
{
if(sumbufL)
free(sumbufL);
delete [] sumbufL;
}
void doAddMix(uint8_t channel, float32_t* in)
{
float32_t* tmp=(float32_t*)malloc(sizeof(float32_t)*buffer_length);
float32_t tmp[buffer_length];
assert(tmp!=NULL);
assert(in);
if(multiplier[channel]!=UNITY_GAIN)
arm_scale_f32(in,multiplier[channel],tmp,buffer_length);
arm_add_f32(sumbufL, tmp, sumbufL, buffer_length);
free(tmp);
}
void gain(uint8_t channel, float32_t gain)
@ -95,16 +91,13 @@ public:
for (uint8_t i=0; i<NN; i++)
panorama[i] = UNITY_PANORAMA;
sumbufR=(float32_t*)malloc(sizeof(float32_t) * buffer_length);
sumbufR=new float32_t[buffer_length];
arm_fill_f32(0.0f, sumbufR, buffer_length);
}
~AudioStereoMixer()
{
if(sumbufL)
free(sumbufL);
if(sumbufR)
free(sumbufR);
delete [] sumbufR;
}
void pan(uint8_t channel, float32_t pan)
@ -120,9 +113,8 @@ public:
void doAddMix(uint8_t channel, float32_t* in)
{
float32_t* tmp=(float32_t*)malloc(sizeof(float32_t)*buffer_length);
float32_t tmp[buffer_length];
assert(tmp!=NULL);
assert(in);
// left
@ -135,16 +127,12 @@ public:
if(multiplier[channel]!=UNITY_GAIN)
arm_scale_f32(tmp,multiplier[channel],tmp,buffer_length);
arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
if(tmp)
free(tmp);
}
void doAddMix(uint8_t channel, float32_t* inL, float32_t* inR)
{
float32_t* tmp=malloc(sizeof(float32_t)*buffer_length);
float32_t tmp[buffer_length];
assert(tmp!=NULL);
assert(inL);
assert(inR);
@ -156,9 +144,6 @@ public:
if(multiplier[channel]!=UNITY_GAIN)
arm_scale_f32(inR,multiplier[channel],tmp,buffer_length);
arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
if(tmp)
free(tmp);
}
void getMix(float32_t* bufferL, float32_t* bufferR)

32
src/minidexed.cpp
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@ -371,8 +371,8 @@ void CMiniDexed::SetPan (unsigned nPan, unsigned nTG)
assert (nTG < CConfig::ToneGenerators);
m_nPan[nTG] = nPan;
tg_mixer->pan(nTG,mapfloat(nPan,-99,99,0.0f,1.0f));
reverb_send_mixer->pan(nTG,mapfloat(nPan,-99,99,0.0f,1.0f));
tg_mixer->pan(nTG,mapfloat(nPan,0,127,0.0f,1.0f));
reverb_send_mixer->pan(nTG,mapfloat(nPan,0,127,0.0f,1.0f));
m_UI.ParameterChanged ();
}
@ -391,7 +391,7 @@ void CMiniDexed::SetReverbSend (unsigned nReverbSend, unsigned nTG)
void CMiniDexed::SetMasterTune (int nMasterTune, unsigned nTG)
{
constrain((int)nMasterTune,-99,99);
nMasterTune=constrain((int)nMasterTune,-99,99);
assert (nTG < CConfig::ToneGenerators);
m_nMasterTune[nTG] = nMasterTune;
@ -529,49 +529,49 @@ void CMiniDexed::SetParameter (TParameter Parameter, int nValue)
break;
case ParameterReverbEnable:
constrain((int)nValue,0,1);
nValue=constrain((int)nValue,0,1);
m_ReverbSpinLock.Acquire ();
reverb->set_bypass (!nValue);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbSize:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->size (nValue / 99.0f);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbHighDamp:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->hidamp (nValue / 99.0f);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbLowDamp:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->lodamp (nValue / 99.0f);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbLowPass:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->lowpass (nValue / 99.0f);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbDiffusion:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->diffusion (nValue / 99.0f);
m_ReverbSpinLock.Release ();
break;
case ParameterReverbLevel:
constrain((int)nValue,0,99);
nValue=constrain((int)nValue,0,99);
m_ReverbSpinLock.Acquire ();
reverb->level (nValue / 99.0f);
m_ReverbSpinLock.Release ();
@ -699,15 +699,9 @@ void CMiniDexed::ProcessSound (void)
float32_t SampleBuffer[nFrames];
m_pTG[0]->getSamples (SampleBuffer, nFrames);
// Convert dual float array (left, right) to single int16 array (left/right)
float32_t tmp_float[nFrames*2];
int16_t tmp_int[nFrames*2];
for(uint16_t i=0; i<nFrames;i++)
{
tmp_float[i*2]=SampleBuffer[i];
tmp_float[(i*2)+1]=SampleBuffer[i];
}
arm_float_to_q15(tmp_float,tmp_int,nFrames*2);
// Convert single float array (mono) to int16 array
int16_t tmp_int[nFrames];
arm_float_to_q15(SampleBuffer,tmp_int,nFrames);
if (m_pSoundDevice->Write (tmp_int, sizeof(tmp_int)) != (int) sizeof(tmp_int))
{

3
src/performanceconfig.cpp
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@ -154,6 +154,9 @@ bool CPerformanceConfig::Save (void)
PropertyName.Format ("NoteShift%u", nTG+1);
m_Properties.SetSignedNumber (PropertyName, m_nNoteShift[nTG]);
PropertyName.Format ("ReverbSend%u", nTG+1);
m_Properties.SetNumber (PropertyName, m_nReverbSend[nTG]);
}
m_Properties.SetNumber ("CompressorEnable", m_bCompressorEnable ? 1 : 0);

24
src/uimenu.cpp
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@ -158,6 +158,19 @@ const CUIMenu::TMenuItem CUIMenu::s_SaveMenu[] =
{0}
};
// must match CMiniDexed::TParameter
const CUIMenu::TParameter CUIMenu::s_GlobalParameter[CMiniDexed::ParameterUnknown] =
{
{0, 1, 1, ToOnOff}, // ParameterCompessorEnable
{0, 1, 1, ToOnOff}, // ParameterReverbEnable
{0, 99, 1}, // ParameterReverbSize
{0, 99, 1}, // ParameterReverbHighDamp
{0, 99, 1}, // ParameterReverbLowDamp
{0, 99, 1}, // ParameterReverbLowPass
{0, 99, 1}, // ParameterReverbDiffusion
{0, 99, 1} // ParameterReverbLevel
};
// must match CMiniDexed::TTGParameter
const CUIMenu::TParameter CUIMenu::s_TGParameter[CMiniDexed::TGParameterUnknown] =
{
@ -170,17 +183,6 @@ const CUIMenu::TParameter CUIMenu::s_TGParameter[CMiniDexed::TGParameterUnknown]
{0, 99, 1} // TGParameterReverbSend
};
// must match CMiniDexed::TTGParameter
const CUIMenu::TParameter CUIMenu::s_TGParameter[CMiniDexed::TGParameterUnknown] =
{
{0, CSysExFileLoader::MaxVoiceBankID, 1}, // TGParameterVoiceBank
{0, CSysExFileLoader::VoicesPerBank-1, 1}, // TGParameterProgram
{0, 127, 8, ToVolume}, // TGParameterVolume
{0, 127, 8, ToPan}, // TGParameterPan
{-99, 99, 1}, // TGParameterMasterTune
{0, CMIDIDevice::ChannelUnknown-1, 1, ToMIDIChannel} // TGParameterMIDIChannel
};
// must match DexedVoiceParameters in Synth_Dexed
const CUIMenu::TParameter CUIMenu::s_VoiceParameter[] =
{

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