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stm32-f4discovery-cc256x: fix start/stop stream audio playback

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This commit is contained in:
Matthias Ringwald 2019-03-10 18:29:53 +01:00
parent e448e9a777
commit 7dc18472b7
1 changed files with 31 additions and 19 deletions
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50
port/stm32-f4discovery-cc256x-2/port/hal_audio_f4discovery.c
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@ -49,6 +49,7 @@
static void (*audio_played_handler)(uint8_t buffer_index); static void (*audio_played_handler)(uint8_t buffer_index);
static int playback_started; static int playback_started;
static uint32_t sink_sample_rate;
// our storage // our storage
static int16_t output_buffer[NUM_OUTPUT_BUFFERS * OUTPUT_BUFFER_NUM_SAMPLES * 2]; // stereo static int16_t output_buffer[NUM_OUTPUT_BUFFERS * OUTPUT_BUFFER_NUM_SAMPLES * 2]; // stereo
@ -69,10 +70,11 @@ static void (*audio_recorded_callback)(const int16_t * buffer, uint16_t num_samp
static int16_t input_buffer[INPUT_BUFFER_NUM_SAMPLES]; // single mono buffer 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 uint32_t source_sample_rate;
static int sink_pdm_bytes_per_ms; static int source_pcm_samples_per_ms;
static int sink_pcm_samples_per_irq; static int source_pdm_bytes_per_ms;
static int sink_pdm_samples_total; static int source_pcm_samples_per_irq;
static int source_pdm_samples_total;
void BSP_AUDIO_OUT_HalfTransfer_CallBack(void){ void BSP_AUDIO_OUT_HalfTransfer_CallBack(void){
@ -121,7 +123,7 @@ void hal_audio_sink_init(uint8_t channels,
} }
audio_played_handler = buffer_played_callback; audio_played_handler = buffer_played_callback;
BSP_AUDIO_OUT_Init(OUTPUT_DEVICE_BOTH, 80, sample_rate); sink_sample_rate = sample_rate;
} }
/** /**
@ -160,6 +162,9 @@ int16_t * hal_audio_sink_get_output_buffer(uint8_t buffer_index){
*/ */
void hal_audio_sink_start(void){ void hal_audio_sink_start(void){
playback_started = 1; playback_started = 1;
BSP_AUDIO_OUT_Init(OUTPUT_DEVICE_BOTH, 80, sink_sample_rate);
// BSP_AUDIO_OUT_Play gets number bytes -> 1 frame - 16 bit/stereo = 4 bytes // BSP_AUDIO_OUT_Play gets number bytes -> 1 frame - 16 bit/stereo = 4 bytes
BSP_AUDIO_OUT_Play( (uint16_t*) output_buffer, NUM_OUTPUT_BUFFERS * OUTPUT_BUFFER_NUM_SAMPLES * 4); BSP_AUDIO_OUT_Play( (uint16_t*) output_buffer, NUM_OUTPUT_BUFFERS * OUTPUT_BUFFER_NUM_SAMPLES * 4);
} }
@ -177,7 +182,7 @@ void hal_audio_sink_stop(void){
*/ */
void hal_audio_sink_close(void){ void hal_audio_sink_close(void){
if (playback_started){ if (playback_started){
hal_audio_sink_close(); hal_audio_sink_stop();
} }
} }
@ -233,19 +238,19 @@ 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 = sink_pcm_samples_per_irq; int samples_needed = source_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_bytes_per_ms / 2; pdm_half_buffer += source_pdm_bytes_per_ms / 2;
pcm_buffer += sink_pcm_samples_per_ms; pcm_buffer += source_pcm_samples_per_ms;
samples_needed -= sink_pcm_samples_per_ms; samples_needed -= source_pcm_samples_per_ms;
} }
// notify // notify
(*audio_recorded_callback)(input_buffer, sink_pcm_samples_per_irq); (*audio_recorded_callback)(input_buffer, source_pcm_samples_per_irq);
} }
#endif #endif
@ -262,7 +267,7 @@ void BSP_AUDIO_IN_TransferComplete_CallBack(void){
#ifdef SIMULATE_SINE #ifdef SIMULATE_SINE
generate_sine(); generate_sine();
#else #else
process_pdm(&pdm_buffer[sink_pdm_samples_total/2]); process_pdm(&pdm_buffer[source_pdm_samples_total/2]);
#endif #endif
} }
@ -276,19 +281,25 @@ void hal_audio_source_init(uint8_t channels,
uint32_t sample_rate, uint32_t sample_rate,
void (*buffer_recorded_callback)(const int16_t * buffer, uint16_t num_samples)){ void (*buffer_recorded_callback)(const int16_t * buffer, uint16_t num_samples)){
BSP_AUDIO_IN_Init(sample_rate, 16, channels); source_sample_rate = sample_rate;
// Driver only supports mono recording
if (channels != 1){
log_error("F4 Discovery only has single microphone, stereo recording not supported");
return;
}
int decimation = 64; int decimation = 64;
// size of input & output of PDM filter depend on output frequency and decimation // 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 source_pcm_samples_per_irq = sample_rate / 1000 * 16; // 256@16 kHz, 128@8 kHz
sink_pcm_samples_per_ms = sample_rate / 1000; source_pcm_samples_per_ms = sample_rate / 1000;
sink_pdm_bytes_per_ms = sink_pcm_samples_per_ms * decimation / 8; source_pdm_bytes_per_ms = source_pcm_samples_per_ms * decimation / 8;
sink_pdm_samples_total = INPUT_BUFFER_NUM_SAMPLES * 8 * sample_rate / 16000; source_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); log_info("Source: PDM bytes per ms %u, PDM samples total %u - PCM samples per ms %u", source_pdm_bytes_per_ms, source_pdm_samples_total, source_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;
@ -298,7 +309,8 @@ 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, sink_pdm_samples_total); BSP_AUDIO_IN_Init(source_sample_rate, 16, 1);
BSP_AUDIO_IN_Record(pdm_buffer, source_pdm_samples_total);
recording_started = 1; recording_started = 1;
} }