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stm32-f4discovery-cc256x: adapt PDM ringbuffer to sample rate to keep constant audio recording callback interval and latency

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This commit is contained in:
Matthias Ringwald 2019-03-03 21:38:03 +01:00
parent 04cbc450ea
commit d4bf5cc155
1 changed files with 18 additions and 9 deletions
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27
port/stm32-f4discovery-cc256x/src/hal_audio_f4discovery.c
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@ -58,7 +58,7 @@ static uint32_t stream_start_ms;
static uint32_t stream_samples; static uint32_t stream_samples;
#endif #endif
// input // input - irq every 16 ms currently
#define INPUT_BUFFER_NUM_SAMPLES 256 #define INPUT_BUFFER_NUM_SAMPLES 256
static int recording_started; static int recording_started;
@ -70,7 +70,9 @@ static int16_t input_buffer[INPUT_BUFFER_NUM_SAMPLES]; // single mono buffer
static uint16_t pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*8]; static uint16_t pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*8];
static int sink_pcm_samples_per_ms; static int sink_pcm_samples_per_ms;
static int sink_pdm_byes_per_ms; static int sink_pdm_bytes_per_ms;
static int sink_pcm_samples_per_irq;
static int sink_pdm_samples_total;
void BSP_AUDIO_OUT_HalfTransfer_CallBack(void){ void BSP_AUDIO_OUT_HalfTransfer_CallBack(void){
@ -225,19 +227,19 @@ static void generate_sine(void){
static void process_pdm(uint16_t * pdm_half_buffer){ static void process_pdm(uint16_t * pdm_half_buffer){
int samples_needed = INPUT_BUFFER_NUM_SAMPLES; int samples_needed = sink_pcm_samples_per_irq;
int16_t * pcm_buffer = input_buffer; int16_t * pcm_buffer = input_buffer;
while (samples_needed){ while (samples_needed){
// TODO: use int16_t for pcm samples // TODO: use int16_t for pcm samples
BSP_AUDIO_IN_PDMToPCM(pdm_half_buffer, (uint16_t *) pcm_buffer); BSP_AUDIO_IN_PDMToPCM(pdm_half_buffer, (uint16_t *) pcm_buffer);
pdm_half_buffer += sink_pdm_byes_per_ms / 2; pdm_half_buffer += sink_pdm_bytes_per_ms / 2;
pcm_buffer += sink_pcm_samples_per_ms; pcm_buffer += sink_pcm_samples_per_ms;
samples_needed -= sink_pcm_samples_per_ms; samples_needed -= sink_pcm_samples_per_ms;
} }
// notify // notify
(*audio_recorded_callback)(input_buffer, INPUT_BUFFER_NUM_SAMPLES); (*audio_recorded_callback)(input_buffer, sink_pcm_samples_per_irq);
} }
#endif #endif
@ -254,7 +256,7 @@ void BSP_AUDIO_IN_TransferComplete_CallBack(void){
#ifdef SIMULATE_SINE #ifdef SIMULATE_SINE
generate_sine(); generate_sine();
#else #else
process_pdm(&pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*4]); process_pdm(&pdm_buffer[sink_pdm_samples_total/2]);
#endif #endif
} }
@ -270,10 +272,17 @@ void hal_audio_source_init(uint8_t channels,
BSP_AUDIO_IN_Init(sample_rate, 16, channels); BSP_AUDIO_IN_Init(sample_rate, 16, channels);
// size of input & output of PDM filter depend on output frequency and decimation
int decimation = 64; int decimation = 64;
// size of input & output of PDM filter depend on output frequency and decimation
sink_pcm_samples_per_irq = sample_rate / 1000 * 16; // 256@16 kHz, 128@8 kHz
sink_pcm_samples_per_ms = sample_rate / 1000; sink_pcm_samples_per_ms = sample_rate / 1000;
sink_pdm_byes_per_ms = sink_pcm_samples_per_ms * decimation / 8; sink_pdm_bytes_per_ms = sink_pcm_samples_per_ms * decimation / 8;
sink_pdm_samples_total = INPUT_BUFFER_NUM_SAMPLES * 8 * sample_rate / 16000;
log_info("Source: PDM bytes per ms %u, PDM samples total %u - PCM samples per ms %u", sink_pdm_bytes_per_ms, sink_pdm_samples_total, sink_pcm_samples_per_ms);
audio_recorded_callback = buffer_recorded_callback; audio_recorded_callback = buffer_recorded_callback;
recording_sample_rate = sample_rate; recording_sample_rate = sample_rate;
@ -283,7 +292,7 @@ void hal_audio_source_init(uint8_t channels,
* @brief Start stream * @brief Start stream
*/ */
void hal_audio_source_start(void){ void hal_audio_source_start(void){
BSP_AUDIO_IN_Record(pdm_buffer, sizeof(pdm_buffer) / 2); BSP_AUDIO_IN_Record(pdm_buffer, sink_pdm_samples_total);
recording_started = 1; recording_started = 1;
} }