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snapclient/components/dsp_processor/dsp_processor.c
Karl Osterseher 016a131f33 add new component: user interface http server 
work in progress which eventually will enable the user to configure dsp processor on the fly using an on device http server.

first try and possible fix for #22

Signed-off-by: Karl Osterseher <karli_o@gmx.at>
2023-01-15 20:21:49 +01:00

654 lines
19 KiB
C
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#include <stdint.h>
#include <string.h>
#include <sys/time.h>
#include "freertos/FreeRTOS.h"
#if CONFIG_USE_DSP_PROCESSOR
#include "dsps_biquad.h"
#include "dsps_biquad_gen.h"
#include "esp_log.h"
#include "freertos/queue.h"
#include "dsp_processor.h"
#ifdef CONFIG_USE_BIQUAD_ASM
#define BIQUAD dsps_biquad_f32_ae32
#else
#define BIQUAD dsps_biquad_f32
#endif
static const char *TAG = "dspProc";
#define DSP_PROCESSOR_LEN 16
static QueueHandle_t filterUpdateQHdl = NULL;
static filterParams_t filterParams;
static ptype_t *filter = NULL;
static double dynamic_vol = 1.0;
static bool init = false;
static float *sbuffer0 = NULL;
static float *sbufout0 = NULL;
#if CONFIG_USE_DSP_PROCESSOR
#if CONFIG_SNAPCLIENT_DSP_FLOW_STEREO
dspFlows_t dspFlowInit = dspfStereo;
#endif
#if CONFIG_SNAPCLIENT_DSP_FLOW_BASSBOOST
dspFlows_t dspFlowInit = dspfBassBoost;
#endif
#if CONFIG_SNAPCLIENT_DSP_FLOW_BIAMP
dspFlows_t dspFlowInit = dspfBiamp;
#endif
#if CONFIG_SNAPCLIENT_DSP_FLOW_BASS_TREBLE_EQ
dspFlows_t dspFlowInit = dspfEQBassTreble;
#endif
#endif
/**
*
*/
void dsp_processor_init(void) {
init = false;
if (filterUpdateQHdl) {
vQueueDelete(filterUpdateQHdl);
filterUpdateQHdl = NULL;
}
// have a max queue length of 1 here because we use xQueueOverwrite
// to write to the queue
filterUpdateQHdl = xQueueCreate(1, sizeof(filterParams_t));
if (filterUpdateQHdl == NULL) {
ESP_LOGE(TAG, "%s: Failed to create filter update queue", __func__);
return;
}
// TODO: load this data from NVM if available
filterParams.dspFlow = dspFlowInit;
switch (filterParams.dspFlow) {
case dspfEQBassTreble: {
filterParams.fc_1 = 300.0;
filterParams.gain_1 = 0.0;
filterParams.fc_3 = 4000.0;
filterParams.gain_3 = 0.0;
break;
}
case dspfStereo: {
break;
}
case dspfBassBoost: {
filterParams.fc_1 = 300.0;
filterParams.gain_1 = 6.0;
break;
}
case dspfBiamp: {
filterParams.fc_1 = 300.0;
filterParams.gain_1 = 0;
filterParams.fc_3 = 100.0;
filterParams.gain_3 = 0.0;
break;
}
case dspf2DOT1: { // Process audio L + R LOW PASS FILTER
ESP_LOGW(TAG, "dspf2DOT1, not implemented yet, using stereo instead");
} break;
case dspfFunkyHonda: { // Process audio L + R LOW PASS FILTER
ESP_LOGW(TAG,
"dspfFunkyHonda, not implemented yet, using stereo instead");
break;
}
default: { break; }
}
ESP_LOGI(TAG, "%s: init done", __func__);
}
/**
* free previously allocated memories
*/
void dsp_processor_uninit(void) {
if (sbuffer0) {
free(sbuffer0);
sbuffer0 = NULL;
}
if (sbufout0) {
free(sbufout0);
sbufout0 = NULL;
}
if (filter) {
free(filter);
filter = NULL;
}
if (filterUpdateQHdl) {
vQueueDelete(filterUpdateQHdl);
filterUpdateQHdl = NULL;
}
init = false;
ESP_LOGI(TAG, "%s: uninit done", __func__);
}
/**
*
*/
esp_err_t dsp_processor_update_filter_params(filterParams_t *params) {
if (filterUpdateQHdl) {
if (xQueueOverwrite(filterUpdateQHdl, params) == pdTRUE) {
return ESP_OK;
}
}
return ESP_FAIL;
}
/**
*
*/
static int32_t dsp_processor_gen_filter(ptype_t *filter, uint32_t cnt) {
if ((filter == NULL) && (cnt > 0)) {
return ESP_FAIL;
}
for (int n = 0; n < cnt; n++) {
switch (filter[n].filtertype) {
case HIGHSHELF:
dsps_biquad_gen_highShelf_f32(filter[n].coeffs, filter[n].freq,
filter[n].gain, filter[n].q);
break;
case LOWSHELF:
dsps_biquad_gen_lowShelf_f32(filter[n].coeffs, filter[n].freq,
filter[n].gain, filter[n].q);
break;
case LPF:
dsps_biquad_gen_lpf_f32(filter[n].coeffs, filter[n].freq, filter[n].q);
break;
case HPF:
dsps_biquad_gen_hpf_f32(filter[n].coeffs, filter[n].freq, filter[n].q);
break;
default:
break;
}
// for (uint8_t i = 0; i <= 4; i++) {
// printf("%.6f ", filter[n].coeffs[i]);
// }
// printf("\n");
}
return ESP_OK;
}
/**
*
*/
int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
int16_t len = chunk_size / 4;
int16_t valint;
uint16_t i;
// volatile needed to ensure 32 bit access
volatile uint32_t *audio_tmp = (volatile uint32_t *)audio;
dspFlows_t dspFlow;
// check if we need to update filters
if (xQueueReceive(filterUpdateQHdl, &filterParams, pdMS_TO_TICKS(0)) ==
pdTRUE) {
init = false;
// TODO: store filterParams in NVM
}
dspFlow = filterParams.dspFlow;
if (init == false) {
uint32_t cnt = 0;
if (filter) {
free(filter);
filter = NULL;
}
switch (dspFlow) {
case dspfEQBassTreble: {
cnt = 4;
filter =
(ptype_t *)heap_caps_malloc(sizeof(ptype_t) * cnt, MALLOC_CAP_8BIT);
if (filter) {
// simple EQ control of low and high frequencies (bass, treble)
float bass_fc = filterParams.fc_1 / samplerate;
float bass_gain = filterParams.gain_1;
float treble_fc = filterParams.fc_3 / samplerate;
float treble_gain = filterParams.gain_3;
// filters for CH 0
filter[0] = (ptype_t){LOWSHELF, bass_fc, bass_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[1] = (ptype_t){HIGHSHELF, treble_fc, treble_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
// filters for CH 1
filter[2] = (ptype_t){LOWSHELF, bass_fc, bass_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[3] = (ptype_t){HIGHSHELF, treble_fc, treble_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
ESP_LOGI(TAG, "got new setting for dspfEQBassTreble");
} else {
ESP_LOGE(TAG, "failed to get memory for filter");
}
break;
}
case dspfStereo: {
cnt = 0;
break;
}
case dspfBassBoost: {
cnt = 2;
filter =
(ptype_t *)heap_caps_malloc(sizeof(ptype_t) * cnt, MALLOC_CAP_8BIT);
if (filter) {
float bass_fc = filterParams.fc_1 / samplerate;
float bass_gain = 6.0;
filter[0] = (ptype_t){LOWSHELF, bass_fc, bass_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[1] = (ptype_t){LOWSHELF, bass_fc, bass_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
ESP_LOGI(TAG, "got new setting for dspfBassBoost");
} else {
ESP_LOGE(TAG, "failed to get memory for filter");
}
break;
}
case dspfBiamp: {
cnt = 4;
filter =
(ptype_t *)heap_caps_malloc(sizeof(ptype_t) * cnt, MALLOC_CAP_8BIT);
if (filter) {
float lp_fc = filterParams.fc_1 / samplerate;
float lp_gain = filterParams.gain_1;
float hp_fc = filterParams.fc_3 / samplerate;
float hp_gain = filterParams.gain_3;
filter[0] = (ptype_t){LPF, lp_fc, lp_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[1] = (ptype_t){LPF, lp_fc, lp_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[2] = (ptype_t){HPF, hp_fc, hp_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
filter[3] = (ptype_t){HPF, hp_fc, hp_gain, 0.707,
NULL, NULL, {0, 0, 0, 0, 0}, {0, 0}};
ESP_LOGI(TAG, "got new setting for dspfBiamp");
} else {
ESP_LOGE(TAG, "failed to get memory for filter");
}
break;
}
case dspf2DOT1: { // Process audio L + R LOW PASS FILTER
cnt = 0;
dspFlow = dspfStereo;
ESP_LOGW(TAG, "dspf2DOT1, not implemented yet, using stereo instead");
} break;
case dspfFunkyHonda: { // Process audio L + R LOW PASS FILTER
cnt = 0;
dspFlow = dspfStereo;
ESP_LOGW(TAG,
"dspfFunkyHonda, not implemented yet, using stereo instead");
break;
}
default: { break; }
}
dsp_processor_gen_filter(filter, cnt);
init = true;
}
// only process data if it is valid
if (audio_tmp) {
sbuffer0 = (float *)heap_caps_malloc(sizeof(float) * DSP_PROCESSOR_LEN,
MALLOC_CAP_8BIT);
if (sbuffer0 == NULL) {
ESP_LOGE(TAG, "No Memory allocated for dsp_processor sbuffer0");
return -1;
}
sbufout0 = (float *)heap_caps_malloc(sizeof(float) * DSP_PROCESSOR_LEN,
MALLOC_CAP_8BIT);
if (sbufout0 == NULL) {
ESP_LOGE(TAG, "No Memory allocated for dsp_processor sbufout0");
free(sbuffer0);
return -1;
}
switch (dspFlow) {
case dspfEQBassTreble: {
for (int k = 0; k < len; k += DSP_PROCESSOR_LEN) {
volatile uint32_t *tmp = (uint32_t *)(&audio_tmp[k]);
uint32_t max = DSP_PROCESSOR_LEN;
uint32_t test = len - k;
if (test < DSP_PROCESSOR_LEN) {
max = test;
}
// channel 0
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * /*0.5 **/
((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
}
// BASS
BIQUAD(sbuffer0, sbufout0, max, filter[0].coeffs, filter[0].w);
// TREBLE
BIQUAD(sbufout0, sbuffer0, max, filter[1].coeffs, filter[1].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbuffer0[i] * INT16_MAX);
tmp[i] =
(volatile uint32_t)((tmp[i] & 0xFFFF0000) + (uint32_t)valint);
}
// channel 1
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * /*0.5 **/
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))) /
INT16_MAX;
}
// BASS
BIQUAD(sbuffer0, sbufout0, max, filter[2].coeffs, filter[2].w);
// TREBLE
BIQUAD(sbufout0, sbuffer0, max, filter[3].coeffs, filter[3].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbuffer0[i] * INT16_MAX);
tmp[i] = (volatile uint32_t)((tmp[i] & 0xFFFF) +
((uint32_t)valint << 16));
}
}
break;
}
case dspfStereo: {
for (int k = 0; k < len; k += DSP_PROCESSOR_LEN) {
volatile uint32_t *tmp = (uint32_t *)(&audio_tmp[k]);
uint32_t max = DSP_PROCESSOR_LEN;
uint32_t test = len - k;
if (test < DSP_PROCESSOR_LEN) {
max = test;
}
// set volume
if (dynamic_vol != 1.0) {
for (i = 0; i < max; i++) {
tmp[i] =
((uint32_t)(dynamic_vol *
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))))
<< 16) +
(uint32_t)(dynamic_vol *
((float)((int16_t)(tmp[i] & 0xFFFF))));
}
}
}
break;
}
case dspfBassBoost: { // CH0 low shelf 6dB @ 400Hz
for (int k = 0; k < len; k += DSP_PROCESSOR_LEN) {
volatile uint32_t *tmp = (uint32_t *)(&audio_tmp[k]);
uint32_t max = DSP_PROCESSOR_LEN;
uint32_t test = len - k;
if (test < DSP_PROCESSOR_LEN) {
max = test;
}
// channel 0
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
}
BIQUAD(sbuffer0, sbufout0, max, filter[0].coeffs, filter[0].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbufout0[i] * INT16_MAX);
tmp[i] = (tmp[i] & 0xFFFF0000) + (uint32_t)valint;
}
// channel 1
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))) /
INT16_MAX;
}
BIQUAD(sbuffer0, sbufout0, max, filter[1].coeffs, filter[1].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbufout0[i] * INT16_MAX);
tmp[i] = (tmp[i] & 0xFFFF) + ((uint32_t)valint << 16);
}
}
break;
}
case dspfBiamp: {
for (int k = 0; k < len; k += DSP_PROCESSOR_LEN) {
volatile uint32_t *tmp = (uint32_t *)(&audio_tmp[k]);
uint32_t max = DSP_PROCESSOR_LEN;
uint32_t test = len - k;
if (test < DSP_PROCESSOR_LEN) {
max = test;
}
// Process audio ch0 LOW PASS FILTER
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
}
BIQUAD(sbuffer0, sbufout0, max, filter[0].coeffs, filter[0].w);
BIQUAD(sbufout0, sbuffer0, max, filter[1].coeffs, filter[1].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbuffer0[i] * INT16_MAX);
tmp[i] = (tmp[i] & 0xFFFF0000) + (uint32_t)valint;
}
// Process audio ch1 HIGH PASS FILTER
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))) /
INT16_MAX;
}
BIQUAD(sbuffer0, sbufout0, max, filter[2].coeffs, filter[2].w);
BIQUAD(sbufout0, sbuffer0, max, filter[3].coeffs, filter[3].w);
for (i = 0; i < max; i++) {
valint = (int16_t)(sbuffer0[i] * INT16_MAX);
tmp[i] = (tmp[i] & 0xFFFF) + ((uint32_t)valint << 16);
}
}
break;
}
case dspf2DOT1: { // Process audio L + R LOW PASS FILTER
/*
BIQUAD(sbuffer2, sbuftmp0, len, bq[0].coeffs, bq[0].w);
BIQUAD(sbuftmp0, sbufout2, len, bq[1].coeffs, bq[1].w);
// Process audio L HIGH PASS FILTER
BIQUAD(sbuffer0, sbuftmp0, len, bq[2].coeffs, bq[2].w);
BIQUAD(sbuftmp0, sbufout0, len, bq[3].coeffs, bq[3].w);
// Process audio R HIGH PASS FILTER
BIQUAD(sbuffer1, sbuftmp0, len, bq[4].coeffs, bq[4].w);
BIQUAD(sbuftmp0, sbufout1, len, bq[5].coeffs, bq[5].w);
int16_t valint[5];
for (uint16_t i = 0; i < len; i++) {
valint[0] =
(muteCH[0] == 1) ? (int16_t)0 : (int16_t)(sbufout0[i] *
INT16_MAX); valint[1] = (muteCH[1] == 1) ? (int16_t)0 :
(int16_t)(sbufout1[i] * INT16_MAX); valint[2] = (muteCH[2] == 1) ?
(int16_t)0 : (int16_t)(sbufout2[i] * INT16_MAX); dsp_audio[i * 4 + 0]
= (valint[2] & 0xff); dsp_audio[i * 4 + 1] = ((valint[2] & 0xff00) >>
8); dsp_audio[i * 4 + 2] = 0; dsp_audio[i * 4 + 3] = 0;
dsp_audio1[i * 4 + 0] = (valint[0] & 0xff);
dsp_audio1[i * 4 + 1] = ((valint[0] & 0xff00) >> 8);
dsp_audio1[i * 4 + 2] = (valint[1] & 0xff);
dsp_audio1[i * 4 + 3] = ((valint[1] & 0xff00) >> 8);
}
// TODO: this copy could be avoided if dsp_audio buffers are
// allocated dynamically and pointers are exchanged after
// audio was freed
memcpy(audio, dsp_audio, chunk_size);
ESP_LOGW(TAG, "Don't know what to do with dsp_audio1");
*/
ESP_LOGW(TAG, "dspf2DOT1, not implemented yet, using stereo instead");
} break;
case dspfFunkyHonda: { // Process audio L + R LOW PASS FILTER
/*
BIQUAD(sbuffer2, sbuftmp0, len, bq[0].coeffs, bq[0].w);
BIQUAD(sbuftmp0, sbufout2, len, bq[1].coeffs, bq[1].w);
// Process audio L HIGH PASS FILTER
BIQUAD(sbuffer0, sbuftmp0, len, bq[2].coeffs, bq[2].w);
BIQUAD(sbuftmp0, sbufout0, len, bq[3].coeffs, bq[3].w);
// Process audio R HIGH PASS FILTER
BIQUAD(sbuffer1, sbuftmp0, len, bq[4].coeffs, bq[4].w);
BIQUAD(sbuftmp0, sbufout1, len, bq[5].coeffs, bq[5].w);
uint16_t scale = 16384; // INT16_MAX
int16_t valint[5];
for (uint16_t i = 0; i < len; i++) {
valint[0] =
(muteCH[0] == 1) ? (int16_t)0 : (int16_t)(sbufout0[i] * scale);
valint[1] =
(muteCH[1] == 1) ? (int16_t)0 : (int16_t)(sbufout1[i] * scale);
valint[2] =
(muteCH[2] == 1) ? (int16_t)0 : (int16_t)(sbufout2[i] * scale);
valint[3] = valint[0] + valint[2];
valint[4] = -valint[2];
valint[5] = -valint[1] - valint[2];
dsp_audio[i * 4 + 0] = (valint[3] & 0xff);
dsp_audio[i * 4 + 1] = ((valint[3] & 0xff00) >> 8);
dsp_audio[i * 4 + 2] = (valint[2] & 0xff);
dsp_audio[i * 4 + 3] = ((valint[2] & 0xff00) >> 8);
dsp_audio1[i * 4 + 0] = (valint[4] & 0xff);
dsp_audio1[i * 4 + 1] = ((valint[4] & 0xff00) >> 8);
dsp_audio1[i * 4 + 2] = (valint[5] & 0xff);
dsp_audio1[i * 4 + 3] = ((valint[5] & 0xff00) >> 8);
}
// TODO: this copy could be avoided if dsp_audio buffers are
// allocated dynamically and pointers are exchanged after
// audio was freed
memcpy(audio, dsp_audio, chunk_size);
ESP_LOGW(TAG, "Don't know what to do with dsp_audio1");
*/
ESP_LOGW(TAG,
"dspfFunkyHonda, not implemented yet, using stereo instead");
} break;
default: { } break; }
free(sbuffer0);
sbuffer0 = NULL;
free(sbufout0);
sbufout0 = NULL;
}
return 0;
}
// void dsp_set_xoverfreq(uint8_t freqh, uint8_t freql, uint32_t samplerate) {
// float freq = freqh * 256 + freql;
// // printf("%f\n", freq);
// float f = freq / samplerate / 2.;
// for (int8_t n = 0; n <= 5; n++) {
// bq[n].freq = f;
// switch (bq[n].filtertype) {
// case LPF:
// // for (uint8_t i = 0; i <= 4; i++) {
// // printf("%.6f ", bq[n].coeffs[i]);
// // }
// // printf("\n");
// dsps_biquad_gen_lpf_f32(bq[n].coeffs, bq[n].freq, bq[n].q);
// // for (uint8_t i = 0; i <= 4; i++) {
// // printf("%.6f ", bq[n].coeffs[i]);
// // }
// // printf("%f \n", bq[n].freq);
// break;
// case HPF:
// dsps_biquad_gen_hpf_f32(bq[n].coeffs, bq[n].freq, bq[n].q);
// break;
// default:
// break;
// }
// }
//}
/**
*
*/
void dsp_processor_set_volome(double volume) {
if (volume >= 0 && volume <= 1.0) {
ESP_LOGI(TAG, "Set volume to %f", volume);
dynamic_vol = volume;
}
}
#endif
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