mirror/btstack
1
0
Fork 0
mirror of https://github.com/bluekitchen/btstack.git synced 2025-03-29 04:20:20 +00:00
Code Issues Packages Projects Releases Wiki Activity

stm32-f4discovery-cc256x: use PDM microphone

Browse Source
This commit is contained in:
Matthias Ringwald 2019-03-03 17:00:20 +01:00
parent bb92a97f89
commit e5988b8b56
1 changed files with 44 additions and 8 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

52
port/stm32-f4discovery-cc256x/src/hal_audio_f4discovery.c
View File

@ -67,10 +67,10 @@ static int32_t recording_sample_rate;
static void (*audio_recorded_callback)(const int16_t * buffer, uint16_t num_samples);
static int16_t input_buffer[INPUT_BUFFER_NUM_SAMPLES]; // single mono buffer
static uint16_t pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*8];
// 2048 16-bit samples decimation -> half/complete every 10 ms -> 160 audio samples
static uint16_t pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*8];
static int sink_pcm_samples_per_ms;
static int sink_pdm_byes_per_ms;
void BSP_AUDIO_OUT_HalfTransfer_CallBack(void){
@ -173,6 +173,8 @@ void hal_audio_sink_close(void){
}
}
#ifdef SIMULATE_SINE
// temp sine simulator
// input signal: pre-computed sine wave, 266 Hz at 16000 kHz
static const int16_t sine_int16_at_16000hz[] = {
@ -198,6 +200,7 @@ static void sco_demo_sine_wave_int16_at_8000_hz_host_endian(unsigned int num_sam
}
}
// 16 kHz samples in host endianess
static void sco_demo_sine_wave_int16_at_16000_hz_host_endian(unsigned int num_samples, int16_t * data){
unsigned int i;
@ -210,7 +213,6 @@ static void sco_demo_sine_wave_int16_at_16000_hz_host_endian(unsigned int num_sa
}
static void generate_sine(void){
log_info("generate_sine %u", INPUT_BUFFER_NUM_SAMPLES);
if (recording_sample_rate == 8000){
sco_demo_sine_wave_int16_at_8000_hz_host_endian(INPUT_BUFFER_NUM_SAMPLES, input_buffer);
} else {
@ -219,12 +221,41 @@ static void generate_sine(void){
// notify
(*audio_recorded_callback)(input_buffer, INPUT_BUFFER_NUM_SAMPLES);
}
#else
static void process_pdm(uint16_t * pdm_half_buffer){
int samples_needed = INPUT_BUFFER_NUM_SAMPLES;
int16_t * pcm_buffer = input_buffer;
while (samples_needed){
// TODO: use int16_t for pcm samples
BSP_AUDIO_IN_PDMToPCM(pdm_half_buffer, (uint16_t *) pcm_buffer);
pdm_half_buffer += sink_pdm_byes_per_ms / 2;
pcm_buffer += sink_pcm_samples_per_ms;
samples_needed -= sink_pcm_samples_per_ms;
}
// notify
(*audio_recorded_callback)(input_buffer, INPUT_BUFFER_NUM_SAMPLES);
}
#endif
void BSP_AUDIO_IN_HalfTransfer_CallBack(void){
#ifdef SIMULATE_SINE
generate_sine();
#else
process_pdm(&pdm_buffer[0]);
#endif
}
void BSP_AUDIO_IN_TransferComplete_CallBack(void){
#ifdef SIMULATE_SINE
generate_sine();
}
void BSP_AUDIO_IN_HalfTransfer_CallBack(void){
generate_sine();
#else
process_pdm(&pdm_buffer[INPUT_BUFFER_NUM_SAMPLES*4]);
#endif
}
/**
@ -238,11 +269,16 @@ void hal_audio_source_init(uint8_t channels,
void (*buffer_recorded_callback)(const int16_t * buffer, uint16_t num_samples)){
BSP_AUDIO_IN_Init(sample_rate, 16, channels);
// size of input & output of PDM filter depend on output frequency and decimation
int decimation = 64;
sink_pcm_samples_per_ms = sample_rate / 1000;
sink_pdm_byes_per_ms = sink_pcm_samples_per_ms * decimation / 8;
audio_recorded_callback = buffer_recorded_callback;
recording_sample_rate = sample_rate;
}
/**
* @brief Start stream
*/