wirtz
/
hexefx_audiolib_F32
mirror of https://github.com/hexeguitar/hexefx_audiolib_F32
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity

stereo 3 band eq

Browse Source
pull/2/head
pio 9 months ago
parent ad10d9a088
commit f1f8aa711d
1 changed files with 190 additions and 11 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 201
      src/filter_3bandeq.h

201
src/filter_3bandeq.h
Unescape View File

@ -25,7 +25,6 @@
#include "mathDSP_F32.h" #include "mathDSP_F32.h"
#include "basic_components.h" #include "basic_components.h"
class AudioFilterEqualizer3band_F32 : public AudioStream_F32 class AudioFilterEqualizer3band_F32 : public AudioStream_F32
{ {
public: public:
@ -33,14 +32,14 @@ public:
{ {
setBands(500.0f, 3000.0f); setBands(500.0f, 3000.0f);
} }
void setBands(float32_t bassF, float32_t trebleF) void setBands(float32_t bassF, float32_t trebleF)
{ {
trebleF = 2.0f * sinf(M_PI * (trebleF / AUDIO_SAMPLE_RATE_EXACT)); trebleF = 2.0f * sinf(M_PI * (trebleF / AUDIO_SAMPLE_RATE_EXACT));
bassF = 2.0f * sinf(M_PI * (bassF / AUDIO_SAMPLE_RATE_EXACT)); bassF = 2.0f * sinf(M_PI * (bassF / AUDIO_SAMPLE_RATE_EXACT));
__disable_irq(); __disable_irq();
lowpass_f = bassF; lowpass_f = bassF;
hipass_f = trebleF; hipass_f = trebleF;
__enable_irq(); __enable_irq();
} }
void treble(float32_t t) void treble(float32_t t)
@ -66,7 +65,7 @@ public:
{ {
__disable_irq(); __disable_irq();
treble_g = t; treble_g = t;
mid_g = m; mid_g = m;
bass_g = b; bass_g = b;
__enable_irq(); __enable_irq();
} }
@ -85,7 +84,7 @@ public:
} }
for (i = 0; i < block->length; i++) for (i = 0; i < block->length; i++)
{ {
float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values
float32_t sample = block->data[i]; // give some headroom float32_t sample = block->data[i]; // give some headroom
// Filter #1 (lowpass) // Filter #1 (lowpass)
f1p0 += (lowpass_f * (sample - f1p0)) + vsa; f1p0 += (lowpass_f * (sample - f1p0)) + vsa;
@ -96,9 +95,9 @@ public:
// // Filter #2 (highpass) // // Filter #2 (highpass)
f2p0 += (hipass_f * (sample - f2p0)) + vsa; f2p0 += (hipass_f * (sample - f2p0)) + vsa;
f2p1 += (hipass_f * (f2p0 - f2p1)); f2p1 += (hipass_f * (f2p0 - f2p1));
f2p2 += (hipass_f * (f2p1 - f2p2)); f2p2 += (hipass_f * (f2p1 - f2p2));
f2p3 += (hipass_f * (f2p2 - f2p3)); f2p3 += (hipass_f * (f2p2 - f2p3));
hpOut = sdm3 - f2p3; hpOut = sdm3 - f2p3;
midOut = sdm3 - (lpOut + hpOut); midOut = sdm3 - (lpOut + hpOut);
@ -123,6 +122,7 @@ public:
bp ^= 1; bp ^= 1;
return bp; return bp;
} }
private: private:
audio_block_f32_t *inputQueueArray[1]; audio_block_f32_t *inputQueueArray[1];
bool bp = false; bool bp = false;
@ -136,9 +136,9 @@ private:
float32_t f2p2 = 0.0f; float32_t f2p2 = 0.0f;
float32_t f2p3 = 0.0f; float32_t f2p3 = 0.0f;
float32_t sdm1 = 0.0f; float32_t sdm1 = 0.0f;
float32_t sdm2 = 0.0f; // 2 float32_t sdm2 = 0.0f; // 2
float32_t sdm3 = 0.0f; // 3 float32_t sdm3 = 0.0f; // 3
static constexpr float32_t vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix) static constexpr float32_t vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)
float32_t lpreg; float32_t lpreg;
@ -148,6 +148,185 @@ private:
float32_t hipass_f; float32_t hipass_f;
float32_t treble_g = 1.0f; float32_t treble_g = 1.0f;
float32_t mid_g = 1.0f; float32_t mid_g = 1.0f;
};
class AudioFilterEqualizer3bandStereo_F32 : public AudioStream_F32
{
public:
AudioFilterEqualizer3bandStereo_F32(void) : AudioStream_F32(2, inputQueueArray)
{
setBands(500.0f, 3000.0f);
}
void setBands(float32_t bassF, float32_t trebleF)
{
trebleF = 2.0f * sinf(M_PI * (trebleF / AUDIO_SAMPLE_RATE_EXACT));
bassF = 2.0f * sinf(M_PI * (bassF / AUDIO_SAMPLE_RATE_EXACT));
__disable_irq();
lowpass_f = bassF;
hipass_f = trebleF;
__enable_irq();
}
void treble(float32_t t)
{
__disable_irq();
treble_g = t;
__enable_irq();
}
void mid(float32_t m)
{
__disable_irq();
mid_g = m;
__enable_irq();
}
void bass(float32_t b)
{
__disable_irq();
bass_g = b;
__enable_irq();
}
void set(float32_t t, float32_t m, float32_t b)
{
__disable_irq();
treble_g = t;
mid_g = m;
bass_g = b;
__enable_irq();
}
void update()
{
audio_block_f32_t *blockL, *blockR;
int i;
if (bp) // bypass mode
{
blockL = AudioStream_F32::receiveReadOnly_f32(0);
blockR = AudioStream_F32::receiveReadOnly_f32(1);
if (!blockL || !blockR)
{
if (blockL)
AudioStream_F32::release(blockL);
if (blockR)
AudioStream_F32::release(blockR);
return;
}
AudioStream_F32::transmit(blockL, 0);
AudioStream_F32::transmit(blockR, 1);
AudioStream_F32::release(blockL);
AudioStream_F32::release(blockR);
return;
}
blockL = AudioStream_F32::receiveWritable_f32(0);
blockR = AudioStream_F32::receiveWritable_f32(1);
if (!blockL || !blockR)
{
if (blockL)
AudioStream_F32::release(blockL);
if (blockR)
AudioStream_F32::release(blockR);
return;
}
for (i = 0; i < blockL->length; i++)
{
float32_t lpOut, midOut, hpOut; // Low / Mid / High - Sample Values
// channel L
float32_t sample = blockL->data[i]; // give some headroom
// Filter #1 (lowpass)
f1p0[0] += (lowpass_f * (sample - f1p0[0])) + vsa;
f1p1[0] += (lowpass_f * (f1p0[0] - f1p1[0]));
f1p2[0] += (lowpass_f * (f1p1[0] - f1p2[0]));
f1p3[0] += (lowpass_f * (f1p2[0] - f1p3[0]));
lpOut = f1p3[0];
// // Filter #2 (highpass)
f2p0[0] += (hipass_f * (sample - f2p0[0])) + vsa;
f2p1[0] += (hipass_f * (f2p0[0] - f2p1[0]));
f2p2[0] += (hipass_f * (f2p1[0] - f2p2[0]));
f2p3[0] += (hipass_f * (f2p2[0] - f2p3[0]));
hpOut = sdm3[0] - f2p3[0];
midOut = sdm3[0] - (lpOut + hpOut);
// // Scale, Combine and store
lpOut *= bass_g;
midOut *= mid_g;
hpOut *= treble_g;
// // Shuffle history buffer
sdm3[0] = sdm2[0];
sdm2[0] = sdm1[0];
sdm1[0] = sample;
blockL->data[i] = (lpOut + midOut + hpOut);
// channel R
sample = blockR->data[i]; // give some headroom
// Filter #1 (lowpass)
f1p0[1] += (lowpass_f * (sample - f1p0[1])) + vsa;
f1p1[1] += (lowpass_f * (f1p0[1] - f1p1[1]));
f1p2[1] += (lowpass_f * (f1p1[1] - f1p2[1]));
f1p3[1] += (lowpass_f * (f1p2[1] - f1p3[1]));
lpOut = f1p3[1];
// // Filter #2 (highpass)
f2p0[1] += (hipass_f * (sample - f2p0[1])) + vsa;
f2p1[1] += (hipass_f * (f2p0[1] - f2p1[1]));
f2p2[1] += (hipass_f * (f2p1[1] - f2p2[1]));
f2p3[1] += (hipass_f * (f2p2[1] - f2p3[1]));
hpOut = sdm3[1] - f2p3[1];
midOut = sdm3[1] - (lpOut + hpOut);
// // Scale, Combine and store
lpOut *= bass_g;
midOut *= mid_g;
hpOut *= treble_g;
// // Shuffle history buffer
sdm3[1] = sdm2[1];
sdm2[1] = sdm1[1];
sdm1[1] = sample;
blockR->data[i] = (lpOut + midOut + hpOut);
}
AudioStream_F32::transmit(blockL, 0);
AudioStream_F32::transmit(blockR, 1);
AudioStream_F32::release(blockL);
AudioStream_F32::release(blockR);
}
void bypass_set(bool s) { bp = s; }
bool bypass_tgl()
{
bp ^= 1;
return bp;
}
private:
audio_block_f32_t *inputQueueArray[2];
bool bp = false;
float32_t f1p0[2] = {0.0f, 0.0f};
float32_t f1p1[2] = {0.0f, 0.0f};
float32_t f1p2[2] = {0.0f, 0.0f};
float32_t f1p3[2] = {0.0f, 0.0f};
float32_t f2p0[2] = {0.0f, 0.0f};
float32_t f2p1[2] = {0.0f, 0.0f};
float32_t f2p2[2] = {0.0f, 0.0f};
float32_t f2p3[2] = {0.0f, 0.0f};
float32_t sdm1[2] = {0.0f, 0.0f};
float32_t sdm2[2] = {0.0f, 0.0f};
float32_t sdm3[2] = {0.0f, 0.0f};
static constexpr float32_t vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)
float32_t lpreg[2];
float32_t hpreg[2];
float32_t lowpass_f;
float32_t bass_g = 1.0f;
float32_t hipass_f;
float32_t treble_g = 1.0f;
float32_t mid_g = 1.0f;
}; };
#endif #endif
Write Preview
Loading…
Cancel
Save