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Merge pull request #13 from luar123/NETCONN

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Fix dsp and add software volume mixer
This commit is contained in:
CarlosDerSeher
2022-12-11 20:46:50 +01:00
committed by GitHub
Unverified
parent ec1a7f5ee8 b6b295f6d6
commit cf1530cfc1
4 changed files with 118 additions and 28 deletions
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50
components/dsp_processor/dsp_processor.c
View File

@@ -37,13 +37,15 @@ static uint32_t currentChunkDurationMs = 0;
static ptype_t bq[8];
static double dynamic_vol = 1.0;
int
dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
{
double dynamic_vol = 1.0;
int16_t len = chunk_size / 4;
int16_t valint;
uint16_t i;
volatile uint32_t *audio_tmp = (uint32_t *)audio; //volatile needed to ensure 32 bit access
if ((sbuffer0 == NULL) || (sbufout0 == NULL) || (sbuftmp0 == NULL))
{
@@ -57,6 +59,17 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
{
case dspfStereo:
{
//set volume
if (dynamic_vol != 1.0)
{
for (i = 0; i < len; i++)
{
audio_tmp[i] = ((uint32_t) (dynamic_vol
* ((float)((int16_t)((audio_tmp[i] & 0xFFFF0000) >> 16)))) << 16)
+ (uint32_t) (dynamic_vol
* ((float)((int16_t)(audio_tmp[i] & 0xFFFF))));
}
}
}
break;
@@ -66,8 +79,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 1] << 8)
+ audio[i * 4 + 0]))
* ((float)((int16_t)(audio_tmp[i] & 0xFFFF)))
/ 32768;
}
BIQUAD (sbuffer0, sbufout0, len, bq[6].coeffs, bq[6].w);
@@ -75,17 +87,14 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 0] = (valint & 0x00ff);
audio[i * 4 + 1] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF0000) + (uint32_t)valint;
}
// channel 1
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 3] << 8)
+ audio[i * 4 + 2]))
* ((float)((int16_t)((audio_tmp[i] & 0xFFFF0000) >> 16)))
/ 32768;
}
BIQUAD (sbuffer0, sbufout0, len, bq[7].coeffs, bq[7].w);
@@ -93,8 +102,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 2] = (valint & 0x00ff);
audio[i * 4 + 3] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF) + ((uint32_t)valint << 16);
}
}
break;
@@ -105,8 +113,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 1] << 8)
+ audio[i * 4 + 0]))
* ((float)((int16_t)(audio_tmp[i] & 0xFFFF)))
/ 32768;
}
BIQUAD (sbuffer0, sbuftmp0, len, bq[0].coeffs, bq[0].w);
@@ -115,16 +122,14 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 0] = (valint & 0x00ff);
audio[i * 4 + 1] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF0000) + (uint32_t)valint;
}
// Process audio ch1 HIGH PASS FILTER
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 3] << 8)
+ audio[i * 4 + 2]))
* ((float)((int16_t)((audio_tmp[i] & 0xFFFF0000) >> 16)))
/ 32768;
}
BIQUAD (sbuffer0, sbuftmp0, len, bq[2].coeffs, bq[2].w);
@@ -133,8 +138,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 2] = (valint & 0x00ff);
audio[i * 4 + 3] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF) + ((uint32_t)valint << 16);
}
}
break;
@@ -382,4 +386,14 @@ dsp_set_xoverfreq (uint8_t freqh, uint8_t freql, uint32_t samplerate)
}
}
}
void
dsp_set_vol (double volume)
{
if (volume >= 0 && volume <= 1.0)
{
ESP_LOGI (TAG, "Set volume to %f", volume);
dynamic_vol = volume;
}
}
#endif

1
components/dsp_processor/include/dsp_processor.h
View File

@@ -42,5 +42,6 @@ typedef struct pnode {
void dsp_setup_flow(double freq, uint32_t samplerate, uint32_t chunkDurationMs);
int dsp_processor(char *audio, size_t chunk_size, dspFlows_t dspFlow);
void dsp_set_xoverfreq(uint8_t, uint8_t, uint32_t);
void dsp_set_vol(double volume);
#endif /* _DSP_PROCESSOR_H_ */

32
main/Kconfig.projbuild
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@@ -19,18 +19,36 @@ menu "Snapcast Configuration"
depends on !SNAPSERVER_USE_MDNS
help
Port of the snapserver to connect to.
config SNAPCLIENT_BUFF_LEN
int "snapcast protocol buffer len"
default 4000
help
Size of the snapclient protocol messages buffer (in bytes).
config SNAPCLIENT_NAME
string "snapclient name"
default "esp-snapclient"
default "ESP32-Caster"
help
Name of the client to register the snapserver.
config SNAPCLIENT_USE_SOFT_VOL
bool "Use software volume"
default false
depends on USE_DSP_PROCESSOR
help
Use software volume mixer instead of hardware mixer.
choice SNAPCLIENT_DSP_FLOW
prompt "DSP flow"
default SNAPCLIENT_DSP_FLOW_STEREO
help
Select the DSP flow to use.
config SNAPCLIENT_DSP_FLOW_STEREO
bool "Stereo flow"
config SNAPCLIENT_DSP_FLOW_BASSBOOST
bool "Bassboost flow"
config SNAPCLIENT_DSP_FLOW_BIAMP
bool "Bi-Amp flow"
endchoice
endmenu
menu "SNTP Configuration"

63
main/main.c
View File

@@ -101,8 +101,8 @@ static audio_board_handle_t board_handle = NULL;
#define SNAPCAST_SERVER_HOST CONFIG_SNAPSERVER_HOST
#define SNAPCAST_SERVER_PORT CONFIG_SNAPSERVER_PORT
#endif
#define SNAPCAST_BUFF_LEN CONFIG_SNAPCLIENT_BUFF_LEN
#define SNAPCAST_CLIENT_NAME CONFIG_SNAPCLIENT_NAME
#define SNAPCAST_USE_SOFT_VOL CONFIG_SNAPCLIENT_USE_SOFT_VOL
/* Logging tag */
static const char *TAG = "SC";
@@ -113,7 +113,15 @@ extern char mac_address[18];
SemaphoreHandle_t timeSyncSemaphoreHandle = NULL;
#if CONFIG_USE_DSP_PROCESSOR
uint8_t dspFlow = dspfStereo; // dspfBiamp; // dspfStereo; // dspfBassBoost;
#if CONFIG_SNAPCLIENT_DSP_FLOW_STEREO
dspFlows_t dspFlow = dspfStereo; // dspfBiamp; // dspfStereo; // dspfBassBoost;
#endif
#if CONFIG_SNAPCLIENT_DSP_FLOW_BASSBOOST
dspFlows_t dspFlow = dspfBassBoost;
#endif
#if CONFIG_SNAPCLIENT_DSP_FLOW_BIAMP
dspFlows_t dspFlow = dspfBiamp;
#endif
#endif
typedef struct flacData_s {
@@ -722,7 +730,7 @@ static void http_get_task(void *pvParameters) {
// init hello message
hello_message.mac = mac_address;
hello_message.hostname = "ESP32-Caster";
hello_message.hostname = SNAPCAST_CLIENT_NAME;
hello_message.version = (char *)VERSION_STRING;
hello_message.client_name = "libsnapcast";
hello_message.os = "esp32";
@@ -785,6 +793,7 @@ static void http_get_task(void *pvParameters) {
uint32_t typedMsgLen = 0;
uint32_t offset = 0;
uint32_t tmpData = 0;
int flow_drain_counter = 0;
#define BASE_MESSAGE_STATE 0
#define TYPED_MESSAGE_STATE 1
@@ -1405,6 +1414,20 @@ static void http_get_task(void *pvParameters) {
endTime = esp_timer_get_time();
#if CONFIG_USE_DSP_PROCESSOR
if (flow_drain_counter > 0) {
flow_drain_counter--;
double dynamic_vol = ((double)scSet.volume/100 / (20 - flow_drain_counter));
if (flow_drain_counter == 0) {
#if SNAPCAST_USE_SOFT_VOL
dynamic_vol = 0;
#else
dynamic_vol = 1;
#endif
audio_hal_set_mute(board_handle->audio_hal,
server_settings_message.muted);
}
dsp_set_vol(dynamic_vol);
}
dsp_setup_flow(500, scSet.sr, scSet.chkDur_ms);
dsp_processor(pcmData->fragment->payload,
pcmData->fragment->size, dspFlow);
@@ -1443,6 +1466,20 @@ static void http_get_task(void *pvParameters) {
}
#if CONFIG_USE_DSP_PROCESSOR
if (flow_drain_counter > 0) {
flow_drain_counter--;
double dynamic_vol = ((double)scSet.volume/100 / (20 - flow_drain_counter));
if (flow_drain_counter == 0) {
#if SNAPCAST_USE_SOFT_VOL
dynamic_vol = 0;
#else
dynamic_vol = 1;
#endif
audio_hal_set_mute(board_handle->audio_hal,
server_settings_message.muted);
}
dsp_set_vol(dynamic_vol);
}
dsp_setup_flow(500, scSet.sr, scSet.chkDur_ms);
dsp_processor(pcmData->fragment->payload,
pcmData->fragment->size, dspFlow);
@@ -1982,12 +2019,32 @@ static void http_get_task(void *pvParameters) {
// Volume setting using ADF HAL
// abstraction
if (scSet.muted != server_settings_message.muted) {
#if CONFIG_USE_DSP_PROCESSOR
if (server_settings_message.muted) {
flow_drain_counter = 20;
}
else {
flow_drain_counter = 0;
audio_hal_set_mute(board_handle->audio_hal,
server_settings_message.muted);
#if SNAPCAST_USE_SOFT_VOL
dsp_set_vol((double)server_settings_message.volume/100);
#else
dsp_set_vol(1.0);
#endif
}
#else
audio_hal_set_mute(board_handle->audio_hal,
server_settings_message.muted);
#endif
}
if (scSet.volume != server_settings_message.volume) {
#if SNAPCAST_USE_SOFT_VOL
dsp_set_vol((double)server_settings_message.volume/100);
#else
audio_hal_set_volume(board_handle->audio_hal,
server_settings_message.volume);
#endif
}
scSet.cDacLat_ms = server_settings_message.latency;