Minor cleanup. Added Independent Left/Right volume control.

This commit is contained in:
george-norton 2023-04-28 17:30:49 +01:00
parent d7d40af360
commit 19c9cdd0f1
6 changed files with 81 additions and 59 deletions

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 target_volume;
union {
int16_t volume[2];
int32_t _volume;
};
union {
int16_t target_volume[2];
int32_t _target_volume;
};
bool mute;
} audio_state = {
.freq = 48000,
@ -153,20 +159,18 @@ void core1_entry() {
// 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.
if (audio_state.volume != audio_state.target_volume) {
// Volume attenuation:
if (audio_state._volume != audio_state._target_volume) {
// PCM3060 volume attenuation:
// 0: 0db (default)
// 55: -100db
// 56..: Mute
uint8_t value = 255 + (audio_state.target_volume / 128);
uint8_t buf[3];
buf[0] = 65; // register addr
buf[1] = value; // data left
buf[2] = value; // data right
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;
audio_state._volume = audio_state._target_volume;
}
}
}
@ -335,9 +339,9 @@ 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,
},
@ -521,7 +525,17 @@ 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;
}
}
@ -590,7 +604,7 @@ static void _audio_reconfigure() {
}
}
static void audio_set_volume(int16_t volume) {
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)
@ -603,12 +617,21 @@ static void audio_set_volume(int16_t volume) {
else if (volume < (int16_t) MIN_VOLUME) {
volume = MIN_VOLUME;
}
audio_state.target_volume = 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) {
@ -622,7 +645,7 @@ static void audio_cmd_packet(struct usb_endpoint *ep) {
break;
}
case 2: { // volume
audio_set_volume(*(int16_t *) buffer->data);
audio_set_volume(audio_control_cmd_t.cn, *(int16_t *) buffer->data);
break;
}
}
@ -763,7 +786,7 @@ 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();

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@ -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 *);

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++]);
}

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@ -20,13 +20,13 @@
#include "bqf.h"
// todo fix this. people will forget this.
#define FILTER_STAGES 5 // Don't forget to set this to the right size!
#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);

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@ -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:

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@ -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"