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https://github.com/bluekitchen/btstack.git
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le_audio_demo_util_sink: reworked playback, PLC
This commit is contained in:
Dirk Helbig
Matthias Ringwald
parent
84a4c7a1a4
commit
c18c19debd
@ -54,6 +54,7 @@
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#include "btstack_sample_rate_compensation.h"
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#include "btstack_resample.h"
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#include "btstack_fsm.h"
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#include "hxcmod.h"
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#include "mods/mod.h"
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@ -63,22 +64,12 @@
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#include "wav_util.h"
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#endif
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//#define DEBUG_PLC
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#ifdef DEBUG_PLC
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#define printf_plc(...) { \
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printf(__VA_ARGS__); \
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log_info(__VA_ARGS__);\
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}
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#else
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#define printf_plc(...) (void)(0);
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#endif
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#define MAX_CHANNELS 2
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#define MAX_SAMPLES_PER_FRAME 480
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#define MAX_LC3_FRAME_BYTES 155
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// playback
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#define MAX_NUM_LC3_FRAMES 15
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#define MAX_NUM_LC3_FRAMES (15*2)
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#define MAX_BYTES_PER_SAMPLE 4
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#define PLAYBACK_BUFFER_SIZE (MAX_NUM_LC3_FRAMES * MAX_SAMPLES_PER_FRAME * MAX_CHANNELS * MAX_BYTES_PER_SAMPLE)
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#define PLAYBACK_START_MS (MAX_NUM_LC3_FRAMES * 20 / 3)
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@ -136,26 +127,72 @@ static uint8_t playback_buffer_storage[PLAYBACK_BUFFER_SIZE];
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static btstack_ring_buffer_t playback_buffer;
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// PLC
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static bool stream_last_packet_received[MAX_CHANNELS];
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static uint16_t stream_last_packet_sequence[MAX_CHANNELS];
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static uint16_t group_last_packet_sequence;
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static bool group_last_packet_received;
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static uint16_t plc_timeout_initial_ms;
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static uint16_t plc_timeout_subsequent_ms;
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static uint32_t le_audio_demo_sink_lc3_frames;
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static uint32_t le_audio_demo_sink_zero_frames;
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static uint32_t samples_received;
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static uint32_t samples_played;
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static uint32_t samples_dropped;
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// Audio FSM
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#define TRAN( target ) btstack_fsm_transit( &me->super, (btstack_fsm_state_handler_t)target )
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typedef struct {
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btstack_fsm_t super;
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uint32_t receive_time_ms;
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uint32_t last_receive_time_ms;
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uint32_t zero_frames;
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uint32_t have_pcm;
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uint32_t received_samples;
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} audio_processing_t;
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typedef struct {
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btstack_fsm_event_t super;
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uint16_t sequence_number;
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uint16_t size;
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uint32_t receive_time_ms;
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uint8_t *data;
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uint8_t stream;
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} data_event_t;
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typedef struct {
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btstack_fsm_event_t super;
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uint32_t time_ms;
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} time_event_t;
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audio_processing_t audio_processing;
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enum EventSignals {
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DATA_SIG = BTSTACK_FSM_USER_SIG,
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TIME_SIG
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};
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#define AUDIO_FSM_DEBUG
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#ifdef AUDIO_FSM_DEBUG
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#define ENUM_TO_TEXT(sig) [sig] = #sig
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#define audio_fsm_debug(format, ...) \
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printf( format __VA_OPT__(,) __VA_ARGS__)
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const char * const sigToString[] = {
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ENUM_TO_TEXT(BTSTACK_FSM_INIT_SIG),
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ENUM_TO_TEXT(BTSTACK_FSM_ENTRY_SIG),
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ENUM_TO_TEXT(BTSTACK_FSM_EXIT_SIG),
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ENUM_TO_TEXT(DATA_SIG),
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ENUM_TO_TEXT(TIME_SIG),
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};
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#else
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#define audio_fsm_debug(...)
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#endif
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static btstack_fsm_state_t audio_processing_initial( audio_processing_t * const me, btstack_fsm_event_t const * const e );
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static btstack_fsm_state_t audio_processing_waiting( audio_processing_t * const me, btstack_fsm_event_t const * const e );
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static btstack_fsm_state_t audio_processing_streaming( audio_processing_t * const me, btstack_fsm_event_t const * const e );
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static bool audio_processing_is_streaming( audio_processing_t * const me );
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static btstack_timer_source_t next_packet_timer;
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// lc3 decoder
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static bool le_audio_demo_lc3plus_decoder_requested = false;
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static const btstack_lc3_decoder_t * lc3_decoder;
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static int16_t pcm[MAX_CHANNELS * MAX_SAMPLES_PER_FRAME];
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static bool have_pcm[MAX_CHANNELS];
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static btstack_lc3_decoder_google_t google_decoder_contexts[MAX_CHANNELS];
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#ifdef HAVE_LC3PLUS
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@ -173,12 +210,18 @@ static void le_audio_connection_sink_playback(int16_t * buffer, uint16_t num_sam
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if (playback_active == false){
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if (btstack_ring_buffer_bytes_available(&playback_buffer) >= playback_start_threshold_bytes) {
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log_info("Playback started");
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printf("Playback started\n");
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playback_active = true;
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}
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} else {
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if (bytes_needed > btstack_ring_buffer_bytes_available(&playback_buffer)) {
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log_info("Playback underrun");
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printf("Playback Underrun\n");
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if( audio_processing_is_streaming( &audio_processing ) ) {
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log_info("Playback underrun");
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printf("Playback Underrun\n");
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} else {
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log_info("Playback stopped");
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printf("Playback stopped\n");
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}
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// empty buffer
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uint32_t bytes_read;
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btstack_ring_buffer_read(&playback_buffer, (uint8_t *) buffer, bytes_needed, &bytes_read);
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@ -195,114 +238,6 @@ static void le_audio_connection_sink_playback(int16_t * buffer, uint16_t num_sam
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}
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}
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static void store_samples_in_ringbuffer(void){
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// check if we have all channels
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uint8_t channel;
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for (channel = 0; channel < le_audio_demo_sink_num_channels; channel++){
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if (have_pcm[channel] == false) return;
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}
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#ifdef HAVE_POSIX_FILE_IO
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// write wav samples
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wav_writer_write_int16(le_audio_demo_sink_num_channels * le_audio_demo_sink_num_samples_per_frame, pcm);
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#endif
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// count for samplerate compensation
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le_audio_demo_sink_received_samples += le_audio_demo_sink_num_samples_per_frame;
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// store samples in playback buffer
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samples_received += le_audio_demo_sink_num_samples_per_frame;
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uint32_t resampled_frames = btstack_resample_block(&resample_instance, pcm, le_audio_demo_sink_num_samples_per_frame, pcm_resample);
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uint32_t bytes_to_store = resampled_frames * le_audio_demo_sink_num_channels * 2;
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if (btstack_ring_buffer_bytes_free(&playback_buffer) >= bytes_to_store) {
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btstack_ring_buffer_write(&playback_buffer, (uint8_t *) pcm_resample, bytes_to_store);
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log_info("Samples in playback buffer %5u", btstack_ring_buffer_bytes_available(&playback_buffer) / (le_audio_demo_sink_num_channels * 2));
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} else {
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printf("Samples dropped\n");
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samples_dropped += le_audio_demo_sink_num_samples_per_frame;
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}
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memset(have_pcm, 0, sizeof(have_pcm));
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}
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static void plc_do(uint8_t stream_index) {
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// inject packet
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uint8_t tmp_BEC_detect;
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uint8_t BFI = 1;
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uint8_t i;
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for (i = 0; i < le_audio_demo_sink_num_channels_per_stream; i++){
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uint8_t effective_channel = stream_index + i;
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(void) lc3_decoder->decode_signed_16(decoder_contexts[effective_channel], NULL, BFI,
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&pcm[effective_channel], le_audio_demo_sink_num_channels,
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&tmp_BEC_detect);
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have_pcm[i] = true;
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}
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// and store in ringbuffer when PCM for all channels is available
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store_samples_in_ringbuffer();
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}
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//
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// Perform PLC for packets missing in previous intervals
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//
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// assumptions:
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// - packet sequence number is monotonic increasing
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// - if packet with seq nr x is received, all packets with smaller seq number are either received or missed
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static void plc_check(uint16_t packet_sequence_number) {
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while (group_last_packet_sequence != packet_sequence_number){
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uint8_t i;
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for (i=0;i<le_audio_demo_sink_num_streams;i++){
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// deal with first packet missing. inject silent samples, pcm buffer is memset to zero at start
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if (stream_last_packet_received[i] == false){
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printf_plc("- ISO #%u, very first packet missing\n", i);
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have_pcm[i] = true;
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store_samples_in_ringbuffer();
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stream_last_packet_received[i] = true;
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stream_last_packet_sequence[i] = group_last_packet_sequence;
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continue;
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}
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// missing packet if group sequence counter is higher than stream sequence counter
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if (btstack_time16_delta(group_last_packet_sequence, stream_last_packet_sequence[i]) > 0) {
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printf_plc("- ISO #%u, PLC for %u\n", i, group_last_packet_sequence);
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#ifndef DEBUG_PLC
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log_info("PLC for packet 0x%04x, stream #%u", group_last_packet_sequence, i);
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#endif
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plc_do(i);
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btstack_assert((stream_last_packet_sequence[i] + 1) == group_last_packet_sequence);
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stream_last_packet_sequence[i] = group_last_packet_sequence;
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}
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}
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group_last_packet_sequence++;
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}
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}
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static void plc_timeout(btstack_timer_source_t * timer) {
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// Restart timer. This will loose sync with ISO interval, but if we stop caring if we loose that many packets
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btstack_run_loop_set_timer(timer, plc_timeout_subsequent_ms);
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btstack_run_loop_set_timer_handler(timer, plc_timeout);
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btstack_run_loop_add_timer(timer);
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switch (le_audio_demo_sink_type){
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case HCI_ISO_TYPE_CIS:
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// assume no packet received in iso interval => FT packets missed
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printf_plc("PLC: timeout cis, group %u, FT %u", group_last_packet_sequence, le_audio_demo_sink_flush_timeout);
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plc_check(group_last_packet_sequence + le_audio_demo_sink_flush_timeout);
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break;
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case HCI_ISO_TYPE_BIS:
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// assume PTO not used => 1 packet missed
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plc_check(group_last_packet_sequence + 1);
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break;
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default:
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btstack_unreachable();
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break;
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}
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}
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void le_audio_demo_util_sink_init(const char * filename_wav){
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le_audio_demo_sink_filename_wav = filename_wav;
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le_audio_demo_util_sink_state = LE_AUDIO_SINK_INIT;
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}
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void le_audio_demo_util_sink_enable_lc3plus(bool enable){
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le_audio_demo_lc3plus_decoder_requested = enable;
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}
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@ -348,13 +283,6 @@ void le_audio_demo_util_sink_configure_general(uint8_t num_streams, uint8_t num_
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le_audio_demo_sink_lc3_frames = 0;
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group_last_packet_received = false;
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uint8_t i;
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for (i=0;i<MAX_CHANNELS;i++){
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stream_last_packet_received[i] = false;
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have_pcm[i] = false;
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}
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le_audio_demo_sink_num_samples_per_frame = btstack_lc3_samples_per_frame(le_audio_demo_sink_sampling_frequency_hz, le_audio_demo_sink_frame_duration);
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// switch to lc3plus if requested and possible
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@ -403,15 +331,6 @@ void le_audio_demo_util_sink_configure_unicast(uint8_t num_streams, uint8_t num_
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playback_start_threshold_bytes = playback_start_samples * num_streams * num_channels_per_stream * 2;
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printf("Playback: start %u ms (%u samples, %u bytes)\n", playback_start_ms, playback_start_samples, playback_start_threshold_bytes);
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// set subsequent plc timeout: FT * ISO Interval
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plc_timeout_subsequent_ms = flush_timeout * iso_interval_1250us * 5 / 4;
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// set initial plc timeout:FT * ISO Interval + 4 ms
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plc_timeout_initial_ms = plc_timeout_subsequent_ms + 4;
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printf("PLC: initial timeout %u ms\n", plc_timeout_initial_ms);
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printf("PLC: subsequent timeout %u ms\n", plc_timeout_subsequent_ms);
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le_audio_demo_util_sink_configure_general(num_streams, num_channels_per_stream, sampling_frequency_hz,
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frame_duration, octets_per_frame, iso_interval_1250us);
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}
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@ -428,15 +347,6 @@ void le_audio_demo_util_sink_configure_broadcast(uint8_t num_streams, uint8_t nu
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playback_start_threshold_bytes = playback_start_samples * num_streams * num_channels_per_stream * 2;
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printf("Playback: start %u ms (%u samples, %u bytes)\n", playback_start_ms, playback_start_samples, playback_start_threshold_bytes);
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// set subsequent plc timeout: ISO Interval
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plc_timeout_subsequent_ms = iso_interval_1250us * 5 / 4;
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// set initial plc timeout: ISO Interval + 4 ms
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plc_timeout_initial_ms = plc_timeout_subsequent_ms + 4;
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printf("PLC: initial timeout %u ms\n", plc_timeout_initial_ms);
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printf("PLC: subsequent timeout %u ms\n", plc_timeout_subsequent_ms);
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le_audio_demo_util_sink_configure_general(num_streams, num_channels_per_stream, sampling_frequency_hz, frame_duration, octets_per_frame, iso_interval_1250us);
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}
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@ -451,7 +361,7 @@ void le_audio_demo_util_sink_count(uint8_t stream_index, uint8_t *packet, uint16
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time_stamp = little_endian_read_32(packet, offset);
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offset += 4;
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}
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(void)time_stamp;
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uint32_t receive_time_ms = btstack_run_loop_get_time_ms();
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uint16_t packet_sequence_number = little_endian_read_16(packet, offset);
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@ -482,8 +392,233 @@ void le_audio_demo_util_sink_count(uint8_t stream_index, uint8_t *packet, uint16
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memcpy(&last_packet_prefix[le_audio_demo_sink_num_streams*PACKET_PREFIX_LEN], &packet[offset], PACKET_PREFIX_LEN);
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}
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void le_audio_demo_util_sink_receive(uint8_t stream_index, uint8_t *packet, uint16_t size) {
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static btstack_fsm_state_t audio_processing_initial( audio_processing_t * const me, btstack_fsm_event_t const * const e ) {
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audio_fsm_debug("%s\n", __FUNCTION__ );
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return TRAN(audio_processing_waiting);
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}
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static btstack_fsm_state_t audio_processing_waiting( audio_processing_t * const me, btstack_fsm_event_t const * const e ) {
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audio_fsm_debug("%s - %s\n", __FUNCTION__, sigToString[e->sig]);
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btstack_fsm_state_t status;
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switch(e->sig) {
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case BTSTACK_FSM_ENTRY_SIG: {
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status = BTSTACK_FSM_HANDLED_STATUS;
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break;
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}
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case BTSTACK_FSM_EXIT_SIG: {
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btstack_ring_buffer_init(&playback_buffer, playback_buffer_storage, PLAYBACK_BUFFER_SIZE);
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btstack_sample_rate_compensation_init(&sample_rate_compensation, me->last_receive_time_ms,
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le_audio_demo_sink_sampling_frequency_hz, FLOAT_TO_Q15(1.f));
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me->zero_frames = 0;
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me->received_samples = 0;
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btstack_resample_init( &resample_instance, le_audio_demo_sink_num_channels );
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me->have_pcm = 0;
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status = BTSTACK_FSM_HANDLED_STATUS;
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break;
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}
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case DATA_SIG: {
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data_event_t *data_event = (data_event_t*)e;
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// nothing to do here
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if( data_event->data == NULL ) {
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status = BTSTACK_FSM_IGNORED_STATUS;
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break;
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}
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// ignore empty data at start
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if( data_event->size == 0 ) {
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status = BTSTACK_FSM_IGNORED_STATUS;
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break;
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}
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// always start at first stream
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if( data_event->stream > 0 ) {
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status = BTSTACK_FSM_IGNORED_STATUS;
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break;
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}
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me->last_receive_time_ms = data_event->receive_time_ms;
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status = TRAN(audio_processing_streaming);
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break;
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}
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default: {
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status = BTSTACK_FSM_IGNORED_STATUS;
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break;
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}
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}
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return status;
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}
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static void audio_processing_resample( audio_processing_t * const me, data_event_t *e ) {
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if( me->have_pcm != ((1<<(le_audio_demo_sink_num_streams*le_audio_demo_sink_num_channels_per_stream))-1) ) {
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return;
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}
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int16_t *data_in = pcm;
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int16_t *data_out = pcm_resample;
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#ifdef HAVE_POSIX_FILE_IO
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// write wav samples
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wav_writer_write_int16(le_audio_demo_sink_num_channels * le_audio_demo_sink_num_samples_per_frame, data_in);
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#endif
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// count for samplerate compensation
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me->received_samples += le_audio_demo_sink_num_samples_per_frame;
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// store samples in playback buffer
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samples_received += le_audio_demo_sink_num_samples_per_frame;
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uint32_t resampled_frames = btstack_resample_block(&resample_instance, data_in, le_audio_demo_sink_num_samples_per_frame, data_out);
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uint32_t bytes_to_store = resampled_frames * le_audio_demo_sink_num_channels * 2;
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if (btstack_ring_buffer_bytes_free(&playback_buffer) >= bytes_to_store) {
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btstack_ring_buffer_write(&playback_buffer, (uint8_t *)data_out, bytes_to_store);
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log_info("Samples in playback buffer %5u", btstack_ring_buffer_bytes_available(&playback_buffer) / (le_audio_demo_sink_num_channels * 2));
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} else {
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printf("Samples dropped\n");
|
||||
samples_dropped += le_audio_demo_sink_num_samples_per_frame;
|
||||
}
|
||||
e->data = NULL;
|
||||
me->have_pcm = 0;
|
||||
}
|
||||
|
||||
static btstack_fsm_state_t audio_processing_decode( audio_processing_t * const me, btstack_fsm_event_t const * const e ) {
|
||||
audio_fsm_debug("%s - %s\n", __FUNCTION__, sigToString[e->sig]);
|
||||
btstack_fsm_state_t status;
|
||||
switch(e->sig) {
|
||||
case BTSTACK_FSM_ENTRY_SIG: {
|
||||
btstack_assert( (le_audio_demo_sink_num_streams*le_audio_demo_sink_num_channels_per_stream) < (sizeof(me->have_pcm)*8));
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
case BTSTACK_FSM_EXIT_SIG: {
|
||||
const btstack_audio_sink_t * sink = btstack_audio_sink_get_instance();
|
||||
if( sink == NULL ) {
|
||||
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
uint32_t resampling_factor = btstack_sample_rate_compensation_update( &sample_rate_compensation, me->receive_time_ms,
|
||||
me->received_samples, sink->get_samplerate() );
|
||||
btstack_resample_set_factor(&resample_instance, resampling_factor);
|
||||
me->received_samples = 0;
|
||||
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
case DATA_SIG: {
|
||||
data_event_t *data_event = (data_event_t*)e;
|
||||
uint8_t *data_in = data_event->data;
|
||||
int16_t *data_out = pcm;
|
||||
uint16_t offset = 0;
|
||||
uint8_t BFI = 0;
|
||||
if (data_event->size != le_audio_demo_sink_num_channels_per_stream * le_audio_demo_sink_octets_per_frame) {
|
||||
// incorrect size. we assume that we received this packet on time but cannot decode it, so we use PLC
|
||||
BFI = 1;
|
||||
printf("predict audio\n");
|
||||
}
|
||||
uint8_t i;
|
||||
for (i = 0 ; i < le_audio_demo_sink_num_channels_per_stream ; i++){
|
||||
uint8_t tmp_BEC_detect;
|
||||
uint8_t effective_channel = (data_event->stream * le_audio_demo_sink_num_channels_per_stream) + i;
|
||||
(void) lc3_decoder->decode_signed_16(decoder_contexts[effective_channel], &data_in[offset], BFI,
|
||||
&data_out[effective_channel], le_audio_demo_sink_num_channels,
|
||||
&tmp_BEC_detect);
|
||||
offset += le_audio_demo_sink_octets_per_frame;
|
||||
btstack_assert( !(me->have_pcm & (1<<effective_channel)) );
|
||||
me->have_pcm |= (1<<effective_channel);
|
||||
}
|
||||
audio_processing_resample( me, data_event );
|
||||
status = TRAN(audio_processing_streaming);
|
||||
break;
|
||||
}
|
||||
default: {
|
||||
status = BTSTACK_FSM_IGNORED_STATUS;
|
||||
break;
|
||||
}
|
||||
}
|
||||
return status;
|
||||
}
|
||||
|
||||
static btstack_fsm_state_t audio_processing_streaming( audio_processing_t * const me, btstack_fsm_event_t const * const e ) {
|
||||
audio_fsm_debug("%s - %s\n", __FUNCTION__, sigToString[e->sig]);
|
||||
|
||||
btstack_fsm_state_t status;
|
||||
switch(e->sig) {
|
||||
case BTSTACK_FSM_ENTRY_SIG: {
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
case BTSTACK_FSM_EXIT_SIG: {
|
||||
me->last_receive_time_ms = me->receive_time_ms;
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
case TIME_SIG: {
|
||||
time_event_t *time = (time_event_t*)e;
|
||||
printf("time: %d - %d %d\n", time->time_ms, me->last_receive_time_ms, time->time_ms-me->last_receive_time_ms );
|
||||
// we were last called ages ago, so just start waiting again
|
||||
if( btstack_time_delta( time->time_ms, me->last_receive_time_ms ) > 100) {
|
||||
status = TRAN(audio_processing_waiting);
|
||||
break;
|
||||
}
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
case DATA_SIG: {
|
||||
data_event_t *data_event = (data_event_t*)e;
|
||||
me->receive_time_ms = data_event->receive_time_ms;
|
||||
|
||||
// done processing this data
|
||||
if( data_event->data == NULL ) {
|
||||
status = BTSTACK_FSM_HANDLED_STATUS;
|
||||
break;
|
||||
}
|
||||
|
||||
if( btstack_time_delta( data_event->receive_time_ms, me->last_receive_time_ms ) > 100) {
|
||||
status = TRAN(audio_processing_waiting);
|
||||
break;
|
||||
}
|
||||
|
||||
if( me->zero_frames > 10 ) {
|
||||
status = TRAN(audio_processing_waiting);
|
||||
break;
|
||||
}
|
||||
|
||||
// track consecutive audio frames without data
|
||||
if( data_event->size == 0 ) {
|
||||
me->zero_frames++;
|
||||
} else {
|
||||
me->zero_frames = 0;
|
||||
}
|
||||
|
||||
// will decode and/or predict missing data
|
||||
status = TRAN(audio_processing_decode);
|
||||
break;
|
||||
}
|
||||
default: {
|
||||
status = BTSTACK_FSM_IGNORED_STATUS;
|
||||
break;
|
||||
}
|
||||
}
|
||||
return status;
|
||||
}
|
||||
|
||||
static void audio_processing_constructor( audio_processing_t *me) {
|
||||
btstack_fsm_constructor(&me->super, (btstack_fsm_state_handler_t)&audio_processing_initial);
|
||||
btstack_fsm_init(&me->super, NULL);
|
||||
}
|
||||
|
||||
static void audio_processing_task( audio_processing_t *me, btstack_fsm_event_t const *e ) {
|
||||
btstack_fsm_dispatch_until(&me->super, e);
|
||||
}
|
||||
|
||||
static bool audio_processing_is_streaming( audio_processing_t *me ) {
|
||||
btstack_fsm_t *fsm = &me->super;
|
||||
time_event_t const time_event = { TIME_SIG, btstack_run_loop_get_time_ms() };
|
||||
audio_processing_task( me, &time_event.super );
|
||||
return fsm->state == (btstack_fsm_state_handler_t)&audio_processing_streaming;
|
||||
}
|
||||
|
||||
void le_audio_demo_util_sink_receive(uint8_t stream_index, uint8_t *packet, uint16_t size) {
|
||||
if (le_audio_demo_util_sink_state != LE_AUDIO_SINK_CONFIGURED) return;
|
||||
|
||||
uint16_t header = little_endian_read_16(packet, 0);
|
||||
@ -516,117 +651,31 @@ void le_audio_demo_util_sink_receive(uint8_t stream_index, uint8_t *packet, uint
|
||||
UNUSED(packet_status_flag);
|
||||
UNUSED(time_stamp);
|
||||
|
||||
// start with first packet on first stream
|
||||
if (group_last_packet_received == false){
|
||||
if (stream_index != 0){
|
||||
printf("Ignore first packet for second stream\n");
|
||||
return;
|
||||
}
|
||||
group_last_packet_received = true;
|
||||
group_last_packet_sequence = packet_sequence_number;
|
||||
}
|
||||
|
||||
if (stream_last_packet_received[stream_index]) {
|
||||
printf_plc("ISO #%u, receive %u\n", stream_index, packet_sequence_number);
|
||||
|
||||
int16_t packet_sequence_delta = btstack_time16_delta(packet_sequence_number,
|
||||
stream_last_packet_sequence[stream_index]);
|
||||
if (packet_sequence_delta < 1) {
|
||||
// drop delayed packet that had already been generated by PLC
|
||||
printf_plc("- dropping delayed packet. Current sequence number %u, last received or generated by PLC: %u\n",
|
||||
packet_sequence_number, stream_last_packet_sequence[stream_index]);
|
||||
return;
|
||||
}
|
||||
// simple check
|
||||
if (packet_sequence_number != stream_last_packet_sequence[stream_index] + 1) {
|
||||
printf_plc("- ISO #%u, missing %u\n", stream_index, stream_last_packet_sequence[stream_index] + 1);
|
||||
}
|
||||
} else {
|
||||
printf_plc("ISO %u, first packet seq number %u\n", stream_index, packet_sequence_number);
|
||||
stream_last_packet_received[stream_index] = true;
|
||||
}
|
||||
|
||||
if (sink_receive_streaming){
|
||||
plc_check(packet_sequence_number);
|
||||
}
|
||||
|
||||
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 (iso_sdu_length == 0) {
|
||||
if (sink_receive_streaming){
|
||||
// empty packet -> generate silence
|
||||
memset(pcm, 0, sizeof(pcm));
|
||||
uint8_t i;
|
||||
for (i = 0 ; i < le_audio_demo_sink_num_channels_per_stream ; i++) {
|
||||
uint8_t effective_channel = (stream_index * le_audio_demo_sink_num_channels_per_stream) + i;
|
||||
have_pcm[effective_channel] = true;
|
||||
}
|
||||
le_audio_demo_sink_zero_frames++;
|
||||
// pause detection (1000 ms for 10 ms, 750 ms for 7.5 ms frames)
|
||||
if (le_audio_demo_sink_zero_frames > 100){
|
||||
printf("Pause detected, stopping audio\n");
|
||||
log_info("Pause detected, stopping audio");
|
||||
// pause/reset audio
|
||||
btstack_ring_buffer_init(&playback_buffer, playback_buffer_storage, PLAYBACK_BUFFER_SIZE);
|
||||
sink_receive_streaming = false;
|
||||
playback_active = false;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
// regular packet -> decode codec frame if size ok
|
||||
uint8_t BFI = 0;
|
||||
if (iso_sdu_length != le_audio_demo_sink_num_channels_per_stream * le_audio_demo_sink_octets_per_frame) {
|
||||
// incorrect size. we assume that we received this packet on time but cannot decode it, so we use PLC
|
||||
BFI = 1;
|
||||
}
|
||||
le_audio_demo_sink_zero_frames = 0;
|
||||
uint8_t i;
|
||||
for (i = 0 ; i < le_audio_demo_sink_num_channels_per_stream ; i++){
|
||||
uint8_t tmp_BEC_detect;
|
||||
uint8_t BFI = 0;
|
||||
uint8_t effective_channel = (stream_index * le_audio_demo_sink_num_channels_per_stream) + i;
|
||||
(void) lc3_decoder->decode_signed_16(decoder_contexts[effective_channel], &packet[offset], BFI,
|
||||
&pcm[effective_channel], le_audio_demo_sink_num_channels,
|
||||
&tmp_BEC_detect);
|
||||
offset += le_audio_demo_sink_octets_per_frame;
|
||||
have_pcm[effective_channel] = true;
|
||||
}
|
||||
}
|
||||
|
||||
store_samples_in_ringbuffer();
|
||||
|
||||
if( (sink != NULL) && (iso_sdu_length>0)) {
|
||||
if( sink_receive_streaming ) {
|
||||
uint32_t resampling_factor = btstack_sample_rate_compensation_update( &sample_rate_compensation, receive_time_ms,
|
||||
le_audio_demo_sink_received_samples, sink->get_samplerate() );
|
||||
btstack_resample_set_factor(&resample_instance, resampling_factor);
|
||||
le_audio_demo_sink_received_samples = 0;
|
||||
}
|
||||
}
|
||||
data_event_t const data_event = {
|
||||
.super.sig = DATA_SIG,
|
||||
.sequence_number = packet_sequence_number,
|
||||
.stream = stream_index,
|
||||
.data = &packet[offset],
|
||||
.size = iso_sdu_length,
|
||||
.receive_time_ms = receive_time_ms,
|
||||
};
|
||||
audio_fsm_debug("new data\n");
|
||||
audio_processing_task( &audio_processing, &data_event.super );
|
||||
|
||||
le_audio_demo_sink_lc3_frames++;
|
||||
|
||||
// PLC
|
||||
btstack_run_loop_remove_timer(&next_packet_timer);
|
||||
btstack_run_loop_set_timer(&next_packet_timer, plc_timeout_initial_ms);
|
||||
btstack_run_loop_set_timer_handler(&next_packet_timer, plc_timeout);
|
||||
btstack_run_loop_add_timer(&next_packet_timer);
|
||||
|
||||
if (samples_received >= le_audio_demo_sink_sampling_frequency_hz){
|
||||
printf("LC3 Frames: %4u - samples received %5u, played %5u, dropped %5u\n", le_audio_demo_sink_lc3_frames, samples_received, samples_played, samples_dropped);
|
||||
samples_received = 0;
|
||||
samples_dropped = 0;
|
||||
samples_played = 0;
|
||||
}
|
||||
}
|
||||
|
||||
stream_last_packet_sequence[stream_index] = packet_sequence_number;
|
||||
void le_audio_demo_util_sink_init(const char * filename_wav){
|
||||
le_audio_demo_sink_filename_wav = filename_wav;
|
||||
le_audio_demo_util_sink_state = LE_AUDIO_SINK_INIT;
|
||||
audio_processing_constructor( &audio_processing );
|
||||
}
|
||||
|
||||
/**
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user