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Run the DAC in 24bit mode.

This commit is contained in:
George Norton 2023-08-31 18:08:16 +01:00
parent 1e6896f918
commit 6cd07952ec
6 changed files with 36 additions and 34 deletions
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2
firmware/code/fix16.h
View File

@ -41,7 +41,7 @@ static const fix3_28_t fix16_zero = 0x00000000;
static inline fix3_28_t norm_fix3_28_from_s16sample(int16_t);
static inline int16_t norm_fix3_28_to_s16sample(fix3_28_t);
static inline int32_t norm_fix3_28_to_s16sample(fix3_28_t);
static inline fix3_28_t fix3_28_from_flt(float);

8
firmware/code/fix16.inl
View File

@ -43,7 +43,7 @@ static inline fix3_28_t norm_fix3_28_from_s16sample(int16_t a) {
/// calculated sample to one that the DAC can understand.
/// @param a
/// @return Signed 16-bit integer.
static inline int16_t norm_fix3_28_to_s16sample(fix3_28_t a) {
static inline int32_t norm_fix3_28_to_s16sample(fix3_28_t a) {
// Handle rounding up front, adding one can cause an overflow/underflow
// It's not clear exactly how this works, so we'll disable it for now.
@ -60,18 +60,18 @@ static inline int16_t norm_fix3_28_to_s16sample(fix3_28_t a) {
if (a < 0) {
if (~upper)
{
return SHRT_MIN;
return 0xff800000;
}
} else {
if (upper)
{
return SHRT_MAX;
return 0x007fffff;
}
}
/* When we converted the USB audio sample to a fixed point number, we applied
a normalization, or a gain of 1/65536. To convert it back, we can undo that
by shifting it back by the same amount we shifted it in the first place. */
return (a >> 12);
return (a >> 4);
}
static inline fix3_28_t fix3_28_from_flt(float a) {

25
firmware/code/run.c
View File

@ -140,16 +140,15 @@ static void __no_inline_not_in_flash_func(_as_audio_packet)(struct usb_endpoint
multicore_fifo_push_blocking(CORE0_READY);
multicore_fifo_push_blocking(samples);
for (int j = 0; j < filter_stages; j++) {
// Left channel filter
for (int i = 0; i < samples; i += 2) {
fix3_28_t x_f16 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), preprocessing.preamp);
// Left channel filter
for (int i = 0; i < samples; i += 2) {
fix3_28_t x_f16 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), preprocessing.preamp);
for (int j = 0; j < filter_stages; j++) {
x_f16 = bqf_transform(x_f16, &bqf_filters_left[j],
&bqf_filters_mem_left[j]);
out[i] = (int32_t) norm_fix3_28_to_s16sample(x_f16);
}
out[i] = (int32_t) norm_fix3_28_to_s16sample(x_f16);
}
// Block until core 1 has finished transforming the data
@ -187,15 +186,13 @@ void __no_inline_not_in_flash_func(core1_entry)() {
const uint32_t samples = multicore_fifo_pop_blocking();
for (int j = 0; j < filter_stages; j++) {
for (int i = 1; i < samples; i += 2) {
fix3_28_t x_f16 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), preprocessing.preamp);
for (int i = 1; i < samples; i += 2) {
fix3_28_t x_f16 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), preprocessing.preamp);
for (int j = 0; j < filter_stages; j++) {
x_f16 = bqf_transform(x_f16, &bqf_filters_right[j],
&bqf_filters_mem_right[j]);
out[i] = (int16_t) norm_fix3_28_to_s16sample(x_f16);
}
out[i] = (int32_t) norm_fix3_28_to_s16sample(x_f16);
}
// Signal to core 0 that the data has all been transformed
@ -273,9 +270,9 @@ void setup() {
// Same here, pal. Hands off.
sleep_ms(100);
// Set data format to 16 bit right justified, MSB first
// Set data format to 24 bit right justified, MSB first
buf[0] = 67; // register addr
buf[1] = 0x03; // data
buf[1] = 0x02; // data
i2c_write_blocking(i2c0, PCM_I2C_ADDR, buf, 2, false);
i2s_write_obj.sck_pin = PCM3060_DAC_SCK_PIN;

1
firmware/tools/CMakeLists.txt
View File

@ -6,7 +6,6 @@ set(CMAKE_CXX_STANDARD 17)
add_executable(filter_test
filter_test.c
../code/fix16.c
../code/bqf.c
../code/configuration_manager.c
)

2
firmware/tools/README.md
View File

@ -17,7 +17,7 @@ Run `filter_test` to process the PCM samples. The `filter_test` program takes tw
You can listen to the PCM files using ffplay (which is usually included with ffmpeg):
```
ffplay -f s16le -ar 48000 -ac 2 output.pcm
ffplay -f s24le -ar 48000 -ac 2 output.pcm
```
If there are no obvious problems, go ahead and flash your firmware.

32
firmware/tools/filter_test.c
View File

@ -32,7 +32,7 @@ int main(int argc, char* argv[])
// we dont need to store the whole input and output files in memory.
int samples = input_size / 2;
int16_t *in = (int16_t *) calloc(samples, sizeof(int16_t));
int16_t *out = (int16_t *) calloc(samples, sizeof(int16_t));
int32_t *out = (int32_t *) calloc(samples, sizeof(int32_t));
fread(in, samples, sizeof(int16_t), input);
fclose(input);
@ -54,31 +54,37 @@ int main(int argc, char* argv[])
out[i] = in[i];
}
for (int j = 0; j < filter_stages; j++)
for (int i = 0; i < samples; i ++)
{
for (int i = 0; i < samples; i ++)
{
// Left channel filter
fix16_t x_f16 = fix16_from_s16sample((int16_t) out[i]);
// Left channel filter
fix3_28_t x_f16 = norm_fix3_28_from_s16sample((int16_t) out[i]);
for (int j = 0; j < filter_stages; j++)
{
x_f16 = bqf_transform(x_f16, &bqf_filters_left[j],
&bqf_filters_mem_left[j]);
}
out[i] = (int32_t) fix16_to_s16sample(x_f16);
out[i] = (int32_t) norm_fix3_28_to_s16sample(x_f16);
// Right channel filter
i++;
x_f16 = fix16_from_s16sample((int16_t) out[i]);
// Right channel filter
i++;
x_f16 = norm_fix3_28_from_s16sample((int16_t) out[i]);
for (int j = 0; j < filter_stages; j++)
{
x_f16 = bqf_transform(x_f16, &bqf_filters_right[j],
&bqf_filters_mem_right[j]);
out[i] = (int16_t) fix16_to_s16sample(x_f16);
}
out[i] = (int32_t) norm_fix3_28_to_s16sample(x_f16);
//printf("%08x\n", out[i]);
}
// Write out the processed audio.
fwrite(out, samples, sizeof(int16_t), output);
for (int i=0; i<samples; i++) {
fwrite(&out[i], 3, sizeof(int8_t), output);
}
fclose(output);
free(in);