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fix low volume due to wrong init of audioDAC_data

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add support for mixing stereo to mono through dsp_processor (fixes #106)
add sdkconfigs to github action workflow
suppress warnings from flac git submodule
remove unused variable and address some compiler warnings
remove some dead code

Signed-off-by: Karl Osterseher <karl_osterseher@gmx.at>
This commit is contained in:
Karl Osterseher
2025-02-22 20:58:38 +01:00
committed by Karl Osterseher
Unverified
parent 3cc8572caf
commit 4eb01805fe
13 changed files with 123 additions and 123 deletions
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94
components/dsp_processor/dsp_processor.c
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@@ -7,12 +7,12 @@
#include "freertos/FreeRTOS.h"
#if CONFIG_USE_DSP_PROCESSOR
#include "dsp_processor.h"
#include "dsps_biquad.h"
#include "dsps_biquad_gen.h"
#include "esp_log.h"
#include "freertos/queue.h"
#include "dsp_processor.h"
#include "player.h"
#ifdef CONFIG_USE_BIQUAD_ASM
#define BIQUAD dsps_biquad_f32_ae32
@@ -112,7 +112,9 @@ void dsp_processor_init(void) {
break;
}
default: { break; }
default: {
break;
}
}
ESP_LOGI(TAG, "%s: init done", __func__);
@@ -203,12 +205,36 @@ static int32_t dsp_processor_gen_filter(ptype_t *filter, uint32_t cnt) {
/**
*
*/
int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
int16_t len = chunk_size / 4;
int dsp_processor_worker(void *p_pcmChnk, const void *p_scSet) {
const snapcastSetting_t *scSet = (const snapcastSetting_t *)p_scSet;
pcm_chunk_message_t *pcmChnk = (pcm_chunk_message_t *)p_pcmChnk;
uint32_t samplerate = scSet->sr;
if (!pcmChnk || !pcmChnk->fragment->payload) {
return -1;
}
int bits = scSet->bits;
int ch = scSet->ch;
if (bits == 0) {
bits = 16;
}
if (ch == 0) {
ch = 2;
}
if (samplerate == 0) {
samplerate = 44100;
}
int16_t len = pcmChnk->fragment->size / ((bits / 8) * ch);
int16_t valint;
uint16_t i;
// volatile needed to ensure 32 bit access
volatile uint32_t *audio_tmp = (volatile uint32_t *)audio;
volatile uint32_t *audio_tmp =
(volatile uint32_t *)(pcmChnk->fragment->payload);
dspFlows_t dspFlow;
// check if we need to update filters
@@ -332,7 +358,9 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
break;
}
default: { break; }
default: {
break;
}
}
dsp_processor_gen_filter(filter, cnt);
@@ -360,6 +388,28 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
return -1;
}
#if CONFIG_SNAPCLIENT_MIX_LR_TO_MONO
if (ch == 2) {
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;
}
for (i = 0; i < max; i++) {
int16_t channel0 = (int16_t)((tmp[i] & 0xFFFF0000) >> 16);
int16_t channel1 = (int16_t)(tmp[i] & 0x0000FFFF);
int16_t mixMono = (float)channel0 / 2.0 + (float)channel1 / 2.0;
tmp[i] = ((uint32_t)mixMono << 16) | ((uint32_t)mixMono & 0x0000FFFF);
}
}
}
#endif
switch (dspFlow) {
case dspfEQBassTreble: {
for (int k = 0; k < len; k += DSP_PROCESSOR_LEN) {
@@ -411,6 +461,7 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
}
case dspfStereo: {
#if SNAPCAST_USE_SOFT_VOL
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;
@@ -432,6 +483,7 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
}
}
}
#endif
break;
}
@@ -448,8 +500,11 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
// channel 0
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
sbuffer0[i] =
0.5 * ((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
#if SNAPCAST_USE_SOFT_VOL
sbuffer0[i] *= dynamic_vol;
#endif
}
BIQUAD(sbuffer0, sbufout0, max, filter[0].coeffs, filter[0].w);
@@ -460,9 +515,12 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
// channel 1
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
sbuffer0[i] = 0.5 *
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))) /
INT16_MAX;
#if SNAPCAST_USE_SOFT_VOL
sbuffer0[i] *= dynamic_vol;
#endif
}
BIQUAD(sbuffer0, sbufout0, max, filter[1].coeffs, filter[1].w);
@@ -487,8 +545,11 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
// 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;
sbuffer0[i] =
0.5 * ((float)((int16_t)(tmp[i] & 0xFFFF))) / INT16_MAX;
#if SNAPCAST_USE_SOFT_VOL
sbuffer0[i] *= dynamic_vol;
#endif
}
BIQUAD(sbuffer0, sbufout0, max, filter[0].coeffs, filter[0].w);
BIQUAD(sbufout0, sbuffer0, max, filter[1].coeffs, filter[1].w);
@@ -500,9 +561,12 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
// Process audio ch1 HIGH PASS FILTER
for (i = 0; i < max; i++) {
sbuffer0[i] = dynamic_vol * 0.5 *
sbuffer0[i] = 0.5 *
((float)((int16_t)((tmp[i] & 0xFFFF0000) >> 16))) /
INT16_MAX;
#if SNAPCAST_USE_SOFT_VOL
sbuffer0[i] *= dynamic_vol;
#endif
}
BIQUAD(sbuffer0, sbufout0, max, filter[2].coeffs, filter[2].w);
BIQUAD(sbufout0, sbuffer0, max, filter[3].coeffs, filter[3].w);
@@ -602,7 +666,9 @@ int dsp_processor_worker(char *audio, size_t chunk_size, uint32_t samplerate) {
"dspfFunkyHonda, not implemented yet, using stereo instead");
} break;
default: { } break; }
default: {
} break;
}
free(sbuffer0);
sbuffer0 = NULL;