From 41d4023961776fc75583e8ffa3830d1d6d3864fc Mon Sep 17 00:00:00 2001 From: George Norton <30636555+george-norton@users.noreply.github.com> Date: Thu, 14 Sep 2023 09:49:56 +0100 Subject: [PATCH] Run the DAC in 24bit mode. (#25) * Run the DAC in 24bit mode. * Update comment. * Remove accidental paste * Fix distortion. * Shift up the samples into -1..1, not much different, but we get an extra bit of resolution at the low end. --- firmware/code/fix16.h | 2 +- firmware/code/fix16.inl | 26 ++++++++++++------------- firmware/code/run.c | 25 +++++++++++------------- firmware/tools/CMakeLists.txt | 1 - firmware/tools/README.md | 2 +- firmware/tools/filter_test.c | 36 +++++++++++++++++++++-------------- 6 files changed, 47 insertions(+), 45 deletions(-) diff --git a/firmware/code/fix16.h b/firmware/code/fix16.h index 66799ad..c96ef31 100644 --- a/firmware/code/fix16.h +++ b/firmware/code/fix16.h @@ -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); diff --git a/firmware/code/fix16.inl b/firmware/code/fix16.inl index ca45a2e..ec511b3 100644 --- a/firmware/code/fix16.inl +++ b/firmware/code/fix16.inl @@ -32,18 +32,18 @@ static inline fix3_28_t norm_fix3_28_from_s16sample(int16_t a) { /* So, we're using a Q3.28 fixed point system here, and we want the incoming audio signal to be represented as a number between -1 and 1. To do this, we need the 16-bit value to map to the 28-bit right-of-decimal field in - our fixed point number. 28-16 = 12, so we shift the incoming value by - that much to covert it to the desired Q3.28 format and do the normalization - all in one go. + our fixed point number. 28-16 = 12 + the sign bit = 13, so we shift the + incoming value by that much to covert it to the desired Q3.28 format and + do the normalization all in one go. */ - return (fix3_28_t)a << 12; + return (fix3_28_t)a << 13; } /// @brief Convert fixed point samples into signed integer. Used to convert /// 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. @@ -56,22 +56,20 @@ static inline int16_t norm_fix3_28_to_s16sample(fix3_28_t a) { */ // Saturate the value if an overflow has occurred - uint32_t upper = (a >> 30); + uint32_t upper = (a >> 29); if (a < 0) { - if (~upper) - { - return SHRT_MIN; + if (~upper) { + return 0xff800000; } } else { - if (upper) - { - return SHRT_MAX; + if (upper) { + return 0x00efffff; } } /* 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); + by shifting it but we output 24bts, so the shift is reduced. */ + return (a >> 6); } static inline fix3_28_t fix3_28_from_flt(float a) { diff --git a/firmware/code/run.c b/firmware/code/run.c index 3a33f89..0acc1a8 100644 --- a/firmware/code/run.c +++ b/firmware/code/run.c @@ -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; diff --git a/firmware/tools/CMakeLists.txt b/firmware/tools/CMakeLists.txt index 2cf646a..56a34c6 100644 --- a/firmware/tools/CMakeLists.txt +++ b/firmware/tools/CMakeLists.txt @@ -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 ) diff --git a/firmware/tools/README.md b/firmware/tools/README.md index 1b21591..b7eb844 100644 --- a/firmware/tools/README.md +++ b/firmware/tools/README.md @@ -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. diff --git a/firmware/tools/filter_test.c b/firmware/tools/filter_test.c index 43b55df..a2dd190 100644 --- a/firmware/tools/filter_test.c +++ b/firmware/tools/filter_test.c @@ -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 +54,39 @@ 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 ++) - { - // Left channel filter - fix16_t x_f16 = fix16_from_s16sample((int16_t) out[i]); + const fix3_28_t preamp = fix3_28_from_flt(0.92f); + for (int i = 0; i < samples; i ++) + { + // Left channel filter + fix3_28_t x_f16 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), 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) 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 = fix16_mul(norm_fix3_28_from_s16sample((int16_t) out[i]), 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) 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