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PCM3060 volume control (#13) 

* Initial set of changes to offload the volume controls to the DAC.

* Minor cleanup. Added Independent Left/Right volume control.

* Code cleanup.
This commit is contained in:
George Norton 2023-05-17 20:13:24 +01:00 committed by GitHub
parent a8f9b522de
commit 47b94b8d73
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GPG Key ID: 4AEE18F83AFDEB23
6 changed files with 130 additions and 87 deletions
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132
firmware/code/run.c
View File

@ -46,15 +46,21 @@
i2s_obj_t i2s_write_obj;
static uint8_t *userbuf;
bqf_coeff_t bqf_filters_left[FILTER_STAGES];
bqf_coeff_t bqf_filters_right[FILTER_STAGES];
bqf_mem_t bqf_filters_mem_left[FILTER_STAGES];
bqf_mem_t bqf_filters_mem_right[FILTER_STAGES];
bqf_coeff_t bqf_filters_left[MAX_FILTER_STAGES];
bqf_coeff_t bqf_filters_right[MAX_FILTER_STAGES];
bqf_mem_t bqf_filters_mem_left[MAX_FILTER_STAGES];
bqf_mem_t bqf_filters_mem_right[MAX_FILTER_STAGES];
static struct {
uint32_t freq;
int16_t volume;
int16_t vol_mul;
union {
int16_t volume[2];
int32_t _volume;
};
union {
int16_t target_volume[2];
int32_t _target_volume;
};
bool mute;
} audio_state = {
.freq = 48000,
@ -92,7 +98,6 @@ static void _as_audio_packet(struct usb_endpoint *ep) {
struct usb_buffer *usb_buffer = usb_current_out_packet_buffer(ep);
int16_t *in = (int16_t *) usb_buffer->data;
int32_t *out = (int32_t *) userbuf;
uint16_t vol_mul = audio_state.vol_mul;
int samples = usb_buffer->data_len / 2;
for (int i = 0; i < samples; i++)
@ -101,7 +106,7 @@ static void _as_audio_packet(struct usb_endpoint *ep) {
multicore_fifo_push_blocking(CORE0_READY);
multicore_fifo_push_blocking(samples);
for (int j = 0; j < FILTER_STAGES; j++) {
for (int j = 0; j < filter_stages; j++) {
// Left channel filter
for (int i = 0; i < samples; i += 2) {
fix16_t x_f16 = fix16_from_int((int16_t) out[i]);
@ -116,10 +121,6 @@ static void _as_audio_packet(struct usb_endpoint *ep) {
// Block until core 1 has finished transforming the data
uint32_t ready = multicore_fifo_pop_blocking();
// Multiply the outgoing signal with the volume multiple
for (int i = 0; i < samples; i++)
out[i] = out[i] * (int32_t) vol_mul;
i2s_stream_write(&i2s_write_obj, userbuf, samples * 4);
// keep on truckin'
@ -127,6 +128,23 @@ static void _as_audio_packet(struct usb_endpoint *ep) {
usb_packet_done(ep);
}
static void update_volume()
{
if (audio_state._volume != audio_state._target_volume) {
// PCM3060 volume attenuation:
// 0: 0db (default)
// 55: -100db
// 56..: Mute
uint8_t buf[3];
buf[0] = 65; // register addr
buf[1] = 255 + (audio_state.target_volume[0] / 128); // data left
buf[2] = 255 + (audio_state.target_volume[1] / 128); // data right
i2c_write_blocking(i2c0, PCM_I2C_ADDR, buf, 3, false);
audio_state._volume = audio_state._target_volume;
}
}
void core1_entry() {
uint8_t *userbuf = (uint8_t *) multicore_fifo_pop_blocking();
int32_t *out = (int32_t *) userbuf;
@ -141,7 +159,7 @@ void core1_entry() {
uint32_t limit = multicore_fifo_pop_blocking();
for (int j = 0; j < FILTER_STAGES; j++) {
for (int j = 0; j < filter_stages; j++) {
for (int i = 1; i < limit; i += 2) {
fix16_t x_f16 = fix16_from_int((int16_t) out[i]);
@ -154,6 +172,11 @@ void core1_entry() {
// Signal to core 0 that the data has all been transformed
multicore_fifo_push_blocking(CORE1_READY);
// Update the volume if required. We do this from core1 as
// core0 is more heavily loaded, doing this from core0 can
// lead to audio crackling.
update_volume();
}
}
@ -185,7 +208,10 @@ void setup() {
gpio_set_dir(PCM3060_RST_PIN, GPIO_OUT);
gpio_put(PCM3060_RST_PIN, true);
i2c_init(i2c0, 50000);
// The PCM3060 supports standard mode (100kbps) or fast mode (400kbps)
// we run in fast mode so we dont block the core for too long while
// updating the volume.
i2c_init(i2c0, 400000);
gpio_set_function(PCM3060_SDA_PIN, GPIO_FUNC_I2C);
gpio_set_function(PCM3060_SCL_PIN, GPIO_FUNC_I2C);
gpio_pull_up(PCM3060_SDA_PIN);
@ -203,6 +229,11 @@ void setup() {
// Don't remove this. Don't do it.
sleep_ms(200);
// Set data format to 16 bit right justified, MSB first
buf[0] = 67; // register addr
buf[1] = 0x03; // data
i2c_write_blocking(i2c0, PCM_I2C_ADDR, buf, 2, false);
// Enable DAC
buf[0] = 64; // register addr
buf[1] = 0xE0; // data
@ -312,9 +343,9 @@ static const audio_device_config ad_conf = {
.bSourceID = 1,
.bControlSize = 1,
.bmaControls = {
AUDIO_FEATURE_MUTE | AUDIO_FEATURE_VOLUME,
0,
0
AUDIO_FEATURE_MUTE, // Master channel
AUDIO_FEATURE_VOLUME, // Left channel
AUDIO_FEATURE_VOLUME, // Right channel
},
.iFeature = 0,
},
@ -498,7 +529,16 @@ static bool do_get_current(struct usb_setup_packet *setup) {
}
case 2: { // volume
/* Current volume. See UAC Spec 1.0 p.77 */
usb_start_tiny_control_in_transfer(audio_state.volume, 2);
const uint8_t cn = (uint8_t) setup->wValue;
if (cn == AUDIO_CHANNEL_LEFT_FRONT) {
usb_start_tiny_control_in_transfer(audio_state.target_volume[0], 2);
}
else if (cn == AUDIO_CHANNEL_RIGHT_FRONT) {
usb_start_tiny_control_in_transfer(audio_state.target_volume[1], 2);
}
else {
return false;
}
return true;
}
}
@ -512,22 +552,6 @@ static bool do_get_current(struct usb_setup_packet *setup) {
return false;
}
// todo this seemed like aood guess, but is not correct
uint16_t db_to_vol[91] = {
0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0002, 0x0002,
0x0002, 0x0002, 0x0003, 0x0003, 0x0004, 0x0004, 0x0005, 0x0005,
0x0006, 0x0007, 0x0008, 0x0009, 0x000a, 0x000b, 0x000d, 0x000e,
0x0010, 0x0012, 0x0014, 0x0017, 0x001a, 0x001d, 0x0020, 0x0024,
0x0029, 0x002e, 0x0033, 0x003a, 0x0041, 0x0049, 0x0052, 0x005c,
0x0067, 0x0074, 0x0082, 0x0092, 0x00a4, 0x00b8, 0x00ce, 0x00e7,
0x0104, 0x0124, 0x0147, 0x016f, 0x019c, 0x01ce, 0x0207, 0x0246,
0x028d, 0x02dd, 0x0337, 0x039b, 0x040c, 0x048a, 0x0518, 0x05b7,
0x066a, 0x0732, 0x0813, 0x090f, 0x0a2a, 0x0b68, 0x0ccc, 0x0e5c,
0x101d, 0x1214, 0x1449, 0x16c3, 0x198a, 0x1ca7, 0x2026, 0x2413,
0x287a, 0x2d6a, 0x32f5, 0x392c, 0x4026, 0x47fa, 0x50c3, 0x5a9d,
0x65ac, 0x7214, 0x7fff
};
static bool do_get_minimum(struct usb_setup_packet *setup) {
if ((setup->bmRequestType & USB_REQ_TYPE_RECIPIENT_MASK) == USB_REQ_TYPE_RECIPIENT_INTERFACE) {
switch (setup->wValue >> 8u) {
@ -583,30 +607,48 @@ static void _audio_reconfigure() {
}
}
static void audio_set_volume(int16_t volume) {
audio_state.volume = volume;
// todo interpolate
volume += CENTER_VOLUME_INDEX * 256;
if (volume < 0)
volume = 0;
if (volume >= count_of(db_to_vol) * 256)
volume = count_of(db_to_vol) * 256 - 1;
audio_state.vol_mul = db_to_vol[((uint16_t)volume) >> 8u];
static void audio_set_volume(int8_t channel, int16_t volume) {
// volume is in the range 127.9961dB (0x7FFF) .. -127.9961dB (0x8001). 0x8000 = mute
// the old code reported a min..max volume of -90.9961dB (0xA500) .. 0dB (0x0)
if (volume == 0x8000) {
// Mute case
}
else if (volume > (int16_t) MAX_VOLUME) {
volume = MAX_VOLUME;
}
else if (volume < (int16_t) MIN_VOLUME) {
volume = MIN_VOLUME;
}
if (channel == AUDIO_CHANNEL_LEFT_FRONT || channel == 0) {
audio_state.target_volume[0] = volume;
}
if (channel == AUDIO_CHANNEL_RIGHT_FRONT || channel == 0) {
audio_state.target_volume[1] = volume;
}
}
static void audio_cmd_packet(struct usb_endpoint *ep) {
assert(audio_control_cmd_t.cmd == AUDIO_REQ_SetCurrent);
struct usb_buffer *buffer = usb_current_out_packet_buffer(ep);
// printf("%s: CMD: %u, Type: %u, CS: %u, CN: %u, Unit: %u, Len: %u\n", __PRETTY_FUNCTION__, audio_control_cmd_t.cmd, audio_control_cmd_t.type,
// audio_control_cmd_t.cs, audio_control_cmd_t.cn, audio_control_cmd_t.unit, audio_control_cmd_t.len);
audio_control_cmd_t.cmd = 0;
if (buffer->data_len >= audio_control_cmd_t.len) {
if (audio_control_cmd_t.type == USB_REQ_TYPE_RECIPIENT_INTERFACE) {
switch (audio_control_cmd_t.cs) {
case 1: { // mute
audio_state.mute = buffer->data[0];
uint8_t buf[2];
buf[0] = 68; // register addr
buf[1] = buffer->data[0] ? 0x3 : 0x0; // data
i2c_write_blocking(i2c0, PCM_I2C_ADDR, buf, 2, false);
break;
}
case 2: { // volume
audio_set_volume(*(int16_t *) buffer->data);
audio_set_volume(audio_control_cmd_t.cn, *(int16_t *) buffer->data);
break;
}
}
@ -747,7 +789,7 @@ void usb_sound_card_init() {
assert(device);
device->setup_request_handler = ad_setup_request_handler;
audio_set_volume(DEFAULT_VOLUME);
audio_set_volume(0, DEFAULT_VOLUME);
_audio_reconfigure();
usb_device_start();

8
firmware/code/run.h
View File

@ -40,10 +40,10 @@
#define ENCODE_DB(x) ((uint16_t)(int16_t)((x)*256))
#define MIN_VOLUME ENCODE_DB(-CENTER_VOLUME_INDEX)
#define MIN_VOLUME ENCODE_DB(-100)
#define DEFAULT_VOLUME ENCODE_DB(0)
#define MAX_VOLUME ENCODE_DB(count_of(db_to_vol)-CENTER_VOLUME_INDEX)
#define VOLUME_RESOLUTION ENCODE_DB(1)
#define MAX_VOLUME ENCODE_DB(0)
#define VOLUME_RESOLUTION ENCODE_DB(0.5f)
typedef struct _audio_device_config {
struct usb_configuration_descriptor descriptor;
@ -125,7 +125,7 @@ static bool do_get_minimum(struct usb_setup_packet *);
static bool do_get_maximum(struct usb_setup_packet *);
static bool do_get_resolution(struct usb_setup_packet *);
static void _audio_reconfigure(void);
static void audio_set_volume(int16_t);
static void audio_set_volume(int8_t, int16_t);
static void audio_cmd_packet(struct usb_endpoint *);
static bool as_set_alternate(struct usb_interface *, uint);
static bool do_set_current(struct usb_setup_packet *);

51
firmware/code/user.c
View File

@ -19,47 +19,46 @@
#include "bqf.h"
#include "run.h"
int filter_stages = 0;
/*****************************************************************************
* Here is where your digital signal processing journey begins. Follow this
* guide, and don't forget any steps!
*
* 1. Go to user.h and change FILTER_STAGES to the number of filter stages you
* want.
* 2. Define the filters that you want to use. Check out "bqf.c" for a
* 1. Define the filters that you want to use. Check out "bqf.c" for a
* complete list of what they are and how they work. Using those filters, you
* can create ANY digital signal shape you want. Anything you can dream of.
* 3. You're done! Enjoy the sounds of anything you want.
* 2. You're done! Enjoy the sounds of anything you want.
****************************************************************************/
void define_filters() {
// First filter.
bqf_memreset(&bqf_filters_mem_left[0]);
bqf_memreset(&bqf_filters_mem_right[0]);
bqf_peaking_config(SAMPLING_FREQ, 38.0, -19.0, 0.9, &bqf_filters_left[0]);
bqf_peaking_config(SAMPLING_FREQ, 38.0, -19.0, 0.9, &bqf_filters_right[0]);
bqf_memreset(&bqf_filters_mem_left[filter_stages]);
bqf_memreset(&bqf_filters_mem_right[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 38.0, -19.0, 0.9, &bqf_filters_left[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 38.0, -19.0, 0.9, &bqf_filters_right[filter_stages++]);
// Second filter.
bqf_memreset(&bqf_filters_mem_left[1]);
bqf_memreset(&bqf_filters_mem_right[1]);
bqf_lowshelf_config(SAMPLING_FREQ, 2900.0, 3.0, 4.0, &bqf_filters_left[1]);
bqf_lowshelf_config(SAMPLING_FREQ, 2900.0, 3.0, 4.0, &bqf_filters_right[1]);
bqf_memreset(&bqf_filters_mem_left[filter_stages]);
bqf_memreset(&bqf_filters_mem_right[filter_stages]);
bqf_lowshelf_config(SAMPLING_FREQ, 2900.0, 3.0, 4.0, &bqf_filters_left[filter_stages]);
bqf_lowshelf_config(SAMPLING_FREQ, 2900.0, 3.0, 4.0, &bqf_filters_right[filter_stages++]);
// Third filter.
bqf_memreset(&bqf_filters_mem_left[2]);
bqf_memreset(&bqf_filters_mem_right[2]);
bqf_peaking_config(SAMPLING_FREQ, 430.0, 6.0, 3.5, &bqf_filters_left[2]);
bqf_peaking_config(SAMPLING_FREQ, 430.0, 6.0, 3.5, &bqf_filters_right[2]);
bqf_memreset(&bqf_filters_mem_left[filter_stages]);
bqf_memreset(&bqf_filters_mem_right[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 430.0, 6.0, 3.5, &bqf_filters_left[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 430.0, 6.0, 3.5, &bqf_filters_right[filter_stages++]);
// Fourth filter.
bqf_memreset(&bqf_filters_mem_left[3]);
bqf_memreset(&bqf_filters_mem_right[3]);
bqf_highshelf_config(SAMPLING_FREQ, 8400.0, 3.0, 4.0, &bqf_filters_left[3]);
bqf_highshelf_config(SAMPLING_FREQ, 8400.0, 3.0, 4.0, &bqf_filters_right[3]);
bqf_memreset(&bqf_filters_mem_left[filter_stages]);
bqf_memreset(&bqf_filters_mem_right[filter_stages]);
bqf_highshelf_config(SAMPLING_FREQ, 8400.0, 3.0, 4.0, &bqf_filters_left[filter_stages]);
bqf_highshelf_config(SAMPLING_FREQ, 8400.0, 3.0, 4.0, &bqf_filters_right[filter_stages++]);
// Fifth filter.
bqf_memreset(&bqf_filters_mem_left[4]);
bqf_memreset(&bqf_filters_mem_right[4]);
bqf_peaking_config(SAMPLING_FREQ, 4800.0, 6.0, 5.0, &bqf_filters_left[4]);
bqf_peaking_config(SAMPLING_FREQ, 4800.0, 6.0, 5.0, &bqf_filters_right[4]);
bqf_memreset(&bqf_filters_mem_left[filter_stages]);
bqf_memreset(&bqf_filters_mem_right[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 4800.0, 6.0, 5.0, &bqf_filters_left[filter_stages]);
bqf_peaking_config(SAMPLING_FREQ, 4800.0, 6.0, 5.0, &bqf_filters_right[filter_stages++]);
}

14
firmware/code/user.h
View File

@ -20,13 +20,15 @@
#include "bqf.h"
// todo fix this. people will forget this.
#define FILTER_STAGES 5 // Don't forget to set this to the right size!
// In reality we do not have enough CPU resource to run 8 filtering
// stages without some optimisation.
#define MAX_FILTER_STAGES 8
extern int filter_stages;
extern bqf_coeff_t bqf_filters_left[FILTER_STAGES];
extern bqf_coeff_t bqf_filters_right[FILTER_STAGES];
extern bqf_mem_t bqf_filters_mem_left[FILTER_STAGES];
extern bqf_mem_t bqf_filters_mem_right[FILTER_STAGES];
extern bqf_coeff_t bqf_filters_left[MAX_FILTER_STAGES];
extern bqf_coeff_t bqf_filters_right[MAX_FILTER_STAGES];
extern bqf_mem_t bqf_filters_mem_left[MAX_FILTER_STAGES];
extern bqf_mem_t bqf_filters_mem_right[MAX_FILTER_STAGES];
void define_filters(void);

2
firmware/tools/README.md
View File

@ -5,7 +5,7 @@ This is a basic utility for testing the Ploopy headphones filtering routines on
Find a source file and use ffmpeg to convert it to 16bit stereo PCM samples:
```
ffmpeg -i <input file> -map 0:6 -vn -f s16le -acodec pcm_s16le input.pcm
ffmpeg -i <input file> -vn -f s16le -acodec pcm_s16le input.pcm
```
Run `filter_test` to process the PCM samples. The `filter_test` program takes two arguments an input file and an output file:

10
firmware/tools/filter_test.c
View File

@ -4,10 +4,10 @@
#include "fix16.h"
#include "user.h"
bqf_coeff_t bqf_filters_left[FILTER_STAGES];
bqf_coeff_t bqf_filters_right[FILTER_STAGES];
bqf_mem_t bqf_filters_mem_left[FILTER_STAGES];
bqf_mem_t bqf_filters_mem_right[FILTER_STAGES];
bqf_coeff_t bqf_filters_left[MAX_FILTER_STAGES];
bqf_coeff_t bqf_filters_right[MAX_FILTER_STAGES];
bqf_mem_t bqf_filters_mem_left[MAX_FILTER_STAGES];
bqf_mem_t bqf_filters_mem_right[MAX_FILTER_STAGES];
const char* usage = "Usage: %s INFILE OUTFILE\n\n"
"Reads 16bit stereo PCM data from INFILE, runs it through the Ploopy headphones\n"
@ -59,7 +59,7 @@ int main(int argc, char* argv[])
out[i] = in[i];
}
for (int j = 0; j < FILTER_STAGES; j++)
for (int j = 0; j < filter_stages; j++)
{
for (int i = 0; i < samples; i ++)
{