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example/le_audio_demo_tuil_sink: added sample rate compensation

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Dirk Helbig 2023-10-19 12:28:01 +02:00 committed by Matthias Ringwald
parent 1a6491ffa7
commit 78d87b813c
1 changed files with 36 additions and 3 deletions
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39
example/le_audio_demo_util_sink.c
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@ -37,6 +37,8 @@
#define BTSTACK_FILE__ "le_audio_demo_util_sink.c"
#include <stdio.h>
#include "le_audio_demo_util_sink.h"
#include "btstack_bool.h"
@ -49,6 +51,9 @@
#include "btstack_lc3_google.h"
#include "btstack_lc3plus_fraunhofer.h"
#include "btstack_sample_rate_compensation.h"
#include "btstack_resample.h"
#include "hxcmod.h"
#include "mods/mod.h"
@ -89,6 +94,12 @@
// SINK
static const char * le_audio_demo_sink_filename_wav;
static btstack_sample_rate_compensation_t sample_rate_compensation;
static btstack_resample_t resample_instance;
static bool sink_receive_streaming;
static int16_t pcm_resample[MAX_CHANNELS * MAX_SAMPLES_PER_FRAME * 2];
static btstack_lc3_frame_duration_t le_audio_demo_sink_frame_duration;
static hci_iso_type_t le_audio_demo_sink_type;
@ -179,10 +190,12 @@ static void store_samples_in_ringbuffer(void){
wav_writer_write_int16(le_audio_demo_sink_num_channels * le_audio_demo_sink_num_samples_per_frame, pcm);
#endif
// store samples in playback buffer
uint32_t bytes_to_store = le_audio_demo_sink_num_channels * le_audio_demo_sink_num_samples_per_frame * 2;
samples_received += le_audio_demo_sink_num_samples_per_frame;
uint32_t resampled_frames = btstack_resample_block(&resample_instance, pcm, le_audio_demo_sink_num_samples_per_frame, pcm_resample);
uint32_t bytes_to_store = resampled_frames * le_audio_demo_sink_num_channels * 2;
if (btstack_ring_buffer_bytes_free(&playback_buffer) >= bytes_to_store) {
btstack_ring_buffer_write(&playback_buffer, (uint8_t *) pcm, bytes_to_store);
btstack_ring_buffer_write(&playback_buffer, (uint8_t *) pcm_resample, bytes_to_store);
} else {
printf("Samples dropped\n");
samples_dropped += le_audio_demo_sink_num_samples_per_frame;
@ -301,6 +314,8 @@ void le_audio_demo_util_sink_configure_general(uint8_t num_streams, uint8_t num_
le_audio_demo_sink_num_streams = num_streams;
le_audio_demo_sink_num_channels_per_stream = num_channels_per_stream;
sink_receive_streaming = false;
le_audio_demo_sink_num_channels = le_audio_demo_sink_num_streams * le_audio_demo_sink_num_channels_per_stream;
btstack_assert((le_audio_demo_sink_num_channels == 1) || (le_audio_demo_sink_num_channels == 2));
@ -339,6 +354,8 @@ void le_audio_demo_util_sink_configure_general(uint8_t num_streams, uint8_t num_
// start playback
const btstack_audio_sink_t * sink = btstack_audio_sink_get_instance();
if (sink != NULL){
btstack_sample_rate_compensation_reset( &sample_rate_compensation, btstack_run_loop_get_time_ms() );
btstack_resample_init(&resample_instance, le_audio_demo_sink_num_channels_per_stream);
sink->init(le_audio_demo_sink_num_channels, le_audio_demo_sink_sampling_frequency_hz, le_audio_connection_sink_playback);
sink->start_stream();
}
@ -463,6 +480,21 @@ void le_audio_demo_util_sink_receive(uint8_t stream_index, uint8_t *packet, uint
return;
}
const btstack_audio_sink_t * sink = btstack_audio_sink_get_instance();
if( (sink != NULL) && (iso_sdu_length>0)) {
if( !sink_receive_streaming && playback_active ) {
btstack_sample_rate_compensation_init( &sample_rate_compensation, receive_time_ms,
le_audio_demo_sink_sampling_frequency_hz, FLOAT_TO_Q15(1.f) );
sink_receive_streaming = true;
}
if( sink_receive_streaming ) {
uint32_t resampling_factor = btstack_sample_rate_compensation_update( &sample_rate_compensation, receive_time_ms,
le_audio_demo_sink_num_samples_per_frame, sink->get_samplerate() );
btstack_resample_set_factor(&resample_instance, resampling_factor);
}
}
if (iso_sdu_length == 0) {
// empty packet -> generate silence
memset(pcm, 0, sizeof(pcm));
@ -520,6 +552,7 @@ void le_audio_demo_util_sink_close(void){
if (sink != NULL){
sink->stop_stream();
}
sink_receive_streaming = false;
// stop timer
btstack_run_loop_remove_timer(&next_packet_timer);
}
}