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https://github.com/bluekitchen/btstack.git
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pts/avdtp_sink: update audio processing
This commit is contained in:
Milanka Ringwald
parent
0b322400fd
commit
26e89fe09a
@ -69,6 +69,7 @@ AVDTP += \
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a2dp_source.c \
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a2dp_sink.c \
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btstack_ring_buffer.c \
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btstack_resample.c \
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AVRCP += \
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avrcp_media_item_iterator.c \
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@ -1,4 +1,4 @@
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use port/libusb/a2dp_example:
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use avdtp_sink_test:
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A2DP/SNK/AS/BV-01-I : (wait)
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A2DP/SNK/CC/BV-01-I : (wait)
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@ -44,9 +44,10 @@
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#include "btstack.h"
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#include "wav_util.h"
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#include "btstack_resample.h"
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#include "btstack_ring_buffer.h"
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#ifdef HAVE_PORTAUDIO
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#include "btstack_ring_buffer.h"
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#include <portaudio.h>
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#endif
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@ -84,7 +85,6 @@ static btstack_ring_buffer_t ring_buffer;
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#define FRAMES_PER_BUFFER 128
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#define PREBUFFER_BYTES (PREBUFFER_MS*SAMPLE_RATE/1000*BYTES_PER_FRAME)
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static PaStream * stream;
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static uint8_t ring_buffer_storage[2*PREBUFFER_BYTES];
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#endif
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// WAV File
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@ -98,6 +98,23 @@ static FILE * sbc_file;
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static char * sbc_filename = "avdtp_sink.sbc";
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#endif
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static int media_initialized = 0;
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static btstack_resample_t resample_instance;
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// ring buffer for SBC Frames
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// below 30: add samples, 30-40: fine, above 40: drop samples
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#define OPTIMAL_FRAMES_MIN 30
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#define OPTIMAL_FRAMES_MAX 40
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#define ADDITIONAL_FRAMES 20
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static uint8_t sbc_frame_storage[(OPTIMAL_FRAMES_MAX + ADDITIONAL_FRAMES) * MAX_SBC_FRAME_SIZE];
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static btstack_ring_buffer_t sbc_frame_ring_buffer;
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static unsigned int sbc_frame_size;
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// rest buffer for not fully used sbc frames, with additional frames for resampling
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static uint8_t decoded_audio_storage[(128+16) * BYTES_PER_FRAME];
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static btstack_ring_buffer_t decoded_audio_ring_buffer;
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static int audio_stream_started;
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typedef struct {
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// bitmaps
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uint8_t sampling_frequency_bitmap;
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@ -123,13 +140,15 @@ typedef struct {
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} avdtp_media_codec_configuration_sbc_t;
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// mac 2011: static const char * device_addr_string = "04:0C:CE:E4:85:D3";
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// pts:
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static const char * device_addr_string = "00:1B:DC:07:32:EF";
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// mac 2013: static const char * device_addr_string = "84:38:35:65:d1:15";
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// phone 2013: static const char * device_addr_string = "D8:BB:2C:DF:F0:F2";
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// minijambox: static const char * device_addr_string = "00:21:3C:AC:F7:38";
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// head phones: static const char * device_addr_string = "00:18:09:28:50:18";
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// bt dongle: static const char * device_addr_string = "00:15:83:5F:9D:46";
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// PTS:
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static const char * device_addr_string = "00:1B:DC:08:E2:5C";
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static bd_addr_t device_addr;
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static uint8_t is_cmd_triggered_localy = 0;
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@ -146,6 +165,9 @@ static uint8_t local_seid;
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static uint8_t remote_seid;
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// static avdtp_sep_t sep;
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static int16_t * request_buffer;
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static int request_frames;
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static uint16_t remote_configuration_bitmap;
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static avdtp_capabilities_t remote_configuration;
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@ -196,180 +218,154 @@ static int portaudio_callback( const void *inputBuffer, void *outputBuffer,
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}
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#endif
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static int media_processing_init(avdtp_media_codec_configuration_sbc_t configuration){
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if (is_media_initialized) return 0;
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#ifdef DECODE_SBC
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btstack_sbc_decoder_init(&state, mode, handle_pcm_data, NULL);
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#endif
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#ifdef STORE_SBC_TO_WAV_FILE
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static void media_processing_start(void){
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if (!media_initialized) return;
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// setup audio playback
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const btstack_audio_sink_t * audio = btstack_audio_sink_get_instance();
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if (audio){
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audio->start_stream();
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}
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audio_stream_started = 1;
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}
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static void media_processing_pause(void){
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if (!media_initialized) return;
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// stop audio playback
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audio_stream_started = 0;
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const btstack_audio_sink_t * audio = btstack_audio_sink_get_instance();
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if (audio){
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audio->stop_stream();
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}
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}
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static void playback_handler(int16_t * buffer, uint16_t num_audio_frames){
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#ifdef STORE_TO_WAV_FILE
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int wav_samples = num_audio_frames * NUM_CHANNELS;
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int16_t * wav_buffer = buffer;
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#endif
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// called from lower-layer but guaranteed to be on main thread
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if (sbc_frame_size == 0){
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memset(buffer, 0, num_audio_frames * BYTES_PER_FRAME);
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return;
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}
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// first fill from resampled audio
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uint32_t bytes_read;
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btstack_ring_buffer_read(&decoded_audio_ring_buffer, (uint8_t *) buffer, num_audio_frames * BYTES_PER_FRAME, &bytes_read);
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buffer += bytes_read / NUM_CHANNELS;
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num_audio_frames -= bytes_read / BYTES_PER_FRAME;
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// then start decoding sbc frames using request_* globals
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request_buffer = buffer;
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request_frames = num_audio_frames;
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while (request_frames && btstack_ring_buffer_bytes_available(&sbc_frame_ring_buffer) >= sbc_frame_size){
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// decode frame
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uint8_t sbc_frame[MAX_SBC_FRAME_SIZE];
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btstack_ring_buffer_read(&sbc_frame_ring_buffer, sbc_frame, sbc_frame_size, &bytes_read);
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btstack_sbc_decoder_process_data(&state, 0, sbc_frame, sbc_frame_size);
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}
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#ifdef STORE_TO_WAV_FILE
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audio_frame_count += num_audio_frames;
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wav_writer_write_int16(wav_samples, wav_buffer);
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#endif
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}
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static int media_processing_init(avdtp_media_codec_configuration_sbc_t configuration){
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if (media_initialized) return 0;
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btstack_sbc_decoder_init(&state, mode, handle_pcm_data, NULL);
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#ifdef STORE_TO_WAV_FILE
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wav_writer_open(wav_filename, configuration.num_channels, configuration.sampling_frequency);
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#endif
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#ifdef STORE_SBC_TO_SBC_FILE
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#ifdef STORE_TO_SBC_FILE
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sbc_file = fopen(sbc_filename, "wb");
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#endif
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#ifdef HAVE_PORTAUDIO
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// int frames_per_buffer = configuration.frames_per_buffer;
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PaError err;
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PaStreamParameters outputParameters;
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const PaDeviceInfo *deviceInfo;
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btstack_ring_buffer_init(&sbc_frame_ring_buffer, sbc_frame_storage, sizeof(sbc_frame_storage));
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btstack_ring_buffer_init(&decoded_audio_ring_buffer, decoded_audio_storage, sizeof(decoded_audio_storage));
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btstack_resample_init(&resample_instance, configuration.num_channels);
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/* -- initialize PortAudio -- */
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err = Pa_Initialize();
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if (err != paNoError){
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printf("Error initializing portaudio: \"%s\"\n", Pa_GetErrorText(err));
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return err;
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}
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/* -- setup input and output -- */
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outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
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outputParameters.channelCount = configuration.num_channels;
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outputParameters.sampleFormat = PA_SAMPLE_TYPE;
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outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultHighOutputLatency;
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outputParameters.hostApiSpecificStreamInfo = NULL;
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deviceInfo = Pa_GetDeviceInfo( outputParameters.device );
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printf("PortAudio: Output device: %s\n", deviceInfo->name);
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log_info("PortAudio: Output device: %s", deviceInfo->name);
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/* -- setup stream -- */
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err = Pa_OpenStream(
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&stream,
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NULL, /* &inputParameters */
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&outputParameters,
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configuration.sampling_frequency,
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0,
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paClipOff, /* we won't output out of range samples so don't bother clipping them */
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portaudio_callback, /* use callback */
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NULL );
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if (err != paNoError){
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printf("Error initializing portaudio: \"%s\"\n", Pa_GetErrorText(err));
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return err;
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// setup audio playback
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const btstack_audio_sink_t * audio = btstack_audio_sink_get_instance();
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if (audio){
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audio->init(NUM_CHANNELS, configuration.sampling_frequency, &playback_handler);
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}
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log_info("PortAudio: stream opened");
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printf("PortAudio: stream opened\n");
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#endif
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#if defined(HAVE_PORTAUDIO)
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memset(ring_buffer_storage, 0, sizeof(ring_buffer_storage));
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btstack_ring_buffer_init(&ring_buffer, ring_buffer_storage, sizeof(ring_buffer_storage));
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audio_stream_started = 0;
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audio_stream_paused = 0;
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#endif
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is_media_initialized = 1;
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media_initialized = 1;
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return 0;
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}
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static void media_processing_close(void){
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if (is_media_initialized) return;
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is_media_initialized = 0;
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if (!media_initialized) return;
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media_initialized = 0;
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audio_stream_started = 0;
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sbc_frame_size = 0;
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#ifdef STORE_SBC_TO_WAV_FILE
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#ifdef STORE_TO_WAV_FILE
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wav_writer_close();
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int total_frames_nr = state.good_frames_nr + state.bad_frames_nr + state.zero_frames_nr;
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uint32_t total_frames_nr = state.good_frames_nr + state.bad_frames_nr + state.zero_frames_nr;
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printf("WAV Writer: Decoding done. Processed totaly %d frames:\n - %d good\n - %d bad\n", total_frames_nr, state.good_frames_nr, total_frames_nr - state.good_frames_nr);
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printf("WAV Writer: Written %d frames to wav file: %s\n", frame_count, wav_filename);
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printf("WAV Writer: Decoding done. Processed %u SBC frames:\n - %d good\n - %d bad\n", total_frames_nr, state.good_frames_nr, total_frames_nr - state.good_frames_nr);
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printf("WAV Writer: Wrote %u audio frames to wav file: %s\n", audio_frame_count, wav_filename);
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#endif
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#ifdef STORE_SBC_TO_SBC_FILE
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#ifdef STORE_TO_SBC_FILE
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fclose(sbc_file);
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#endif
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#if defined(HAVE_PORTAUDIO)
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audio_stream_started = 0;
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#endif
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#ifdef HAVE_PORTAUDIO
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printf("PortAudio: Stream closed\n");
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log_info("PortAudio: Stream closed");
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PaError err = Pa_StopStream(stream);
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if (err != paNoError){
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printf("Error stopping the stream: \"%s\"\n", Pa_GetErrorText(err));
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log_error("Error stopping the stream: \"%s\"", Pa_GetErrorText(err));
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return;
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}
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err = Pa_CloseStream(stream);
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if (err != paNoError){
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printf("Error closing the stream: \"%s\"\n", Pa_GetErrorText(err));
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log_error("Error closing the stream: \"%s\"", Pa_GetErrorText(err));
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return;
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}
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err = Pa_Terminate();
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if (err != paNoError){
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printf("Error terminating portaudio: \"%s\"\n", Pa_GetErrorText(err));
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log_error("Error terminating portaudio: \"%s\"", Pa_GetErrorText(err));
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return;
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}
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#endif
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// stop audio playback
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const btstack_audio_sink_t * audio = btstack_audio_sink_get_instance();
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if (audio){
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printf("close stream\n");
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audio->close();
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}
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}
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/* @section Handle Media Data Packet
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*
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* @text Media data packets, in this case the audio data, are received through the handle_l2cap_media_data_packet callback.
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* Currently, only the SBC media codec is supported. Hence, the media data consists of the media packet header and the SBC packet.
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* The SBC data will be decoded using an SBC decoder if either HAVE_PORTAUDIO or STORE_SBC_TO_WAV_FILE directive is defined.
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* The resulting PCM frames can be then captured through a PCM data callback registered during SBC decoder setup, i.e. the
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* handle_pcm_data callback.
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*/
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static int read_media_data_header(uint8_t * packet, int size, int * offset, avdtp_media_packet_header_t * media_header);
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static int read_sbc_header(uint8_t * packet, int size, int * offset, avdtp_sbc_codec_header_t * sbc_header);
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static void handle_l2cap_media_data_packet(uint8_t seid, uint8_t *packet, uint16_t size){
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UNUSED(seid);
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int pos = 0;
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avdtp_media_packet_header_t media_header;
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if (!read_media_data_header(packet, size, &pos, &media_header)) return;
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avdtp_sbc_codec_header_t sbc_header;
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if (!read_sbc_header(packet, size, &pos, &sbc_header)) return;
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#if defined(HAVE_PORTAUDIO) || defined(STORE_SBC_TO_WAV_FILE)
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btstack_sbc_decoder_process_data(&state, 0, packet+pos, size-pos);
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#endif
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#ifdef STORE_SBC_TO_SBC_FILE
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fwrite(packet+pos, size-pos, 1, sbc_file);
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#endif
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}
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/* @section Handle PCM Data
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*
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* @text In this example, we use the [PortAudio library](http://www.portaudio.com) to play the audio stream.
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* The PCM data are bufferd in a ring buffer.
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* Aditionally, tha audio data can be stored in the avdtp_sink.wav file.
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*/
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#if defined(HAVE_PORTAUDIO) || defined(STORE_SBC_TO_WAV_FILE)
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static void handle_pcm_data(int16_t * data, int num_samples, int num_channels, int sample_rate, void * context){
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static void handle_pcm_data(int16_t * data, int num_audio_frames, int num_channels, int sample_rate, void * context){
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UNUSED(sample_rate);
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UNUSED(context);
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UNUSED(num_channels); // must be stereo == 2
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#ifdef STORE_SBC_TO_WAV_FILE
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wav_writer_write_int16(num_samples*num_channels, data);
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frame_count++;
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const btstack_audio_sink_t * audio_sink = btstack_audio_sink_get_instance();
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if (!audio_sink){
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#ifdef STORE_TO_WAV_FILE
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audio_frame_count += num_audio_frames;
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wav_writer_write_int16(num_audio_frames * NUM_CHANNELS, data);
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#endif
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#ifdef HAVE_PORTAUDIO
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// store pcm samples in ring buffer
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btstack_ring_buffer_write(&ring_buffer, (uint8_t *)data, num_samples*num_channels*2);
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if (!audio_stream_started){
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audio_stream_paused = 1;
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/* -- start stream -- */
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PaError err = Pa_StartStream(stream);
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if (err != paNoError){
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printf("Error starting the stream: \"%s\"\n", Pa_GetErrorText(err));
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return;
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}
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audio_stream_started = 1;
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return;
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}
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// resample into request buffer - add some additional space for resampling
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int16_t output_buffer[(128+16) * NUM_CHANNELS]; // 16 * 8 * 2
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uint32_t resampled_frames = btstack_resample_block(&resample_instance, data, num_audio_frames, output_buffer);
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// store data in btstack_audio buffer first
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int frames_to_copy = btstack_min(resampled_frames, request_frames);
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memcpy(request_buffer, output_buffer, frames_to_copy * BYTES_PER_FRAME);
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request_frames -= frames_to_copy;
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request_buffer += frames_to_copy * NUM_CHANNELS;
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// and rest in ring buffer
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int frames_to_store = resampled_frames - frames_to_copy;
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if (frames_to_store){
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int status = btstack_ring_buffer_write(&decoded_audio_ring_buffer, (uint8_t *)&output_buffer[frames_to_copy * NUM_CHANNELS], frames_to_store * BYTES_PER_FRAME);
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if (status){
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printf("Error storing samples in PCM ring buffer!!!\n");
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}
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}
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#endif
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}
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#endif
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static int read_sbc_header(uint8_t * packet, int size, int * offset, avdtp_sbc_codec_header_t * sbc_header){
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int sbc_header_len = 12; // without crc
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@ -450,23 +446,71 @@ static void dump_sbc_configuration(avdtp_media_codec_configuration_sbc_t configu
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printf(" - bitpool_value [%d, %d] \n", configuration.min_bitpool_value, configuration.max_bitpool_value);
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}
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static void handle_l2cap_media_data_packet(uint8_t seid, uint8_t *packet, uint16_t size){
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UNUSED(seid);
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int pos = 0;
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avdtp_media_packet_header_t media_header;
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if (!read_media_data_header(packet, size, &pos, &media_header)) return;
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avdtp_sbc_codec_header_t sbc_header;
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if (!read_sbc_header(packet, size, &pos, &sbc_header)) return;
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static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packet, uint16_t size){
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if (packet_type != HCI_EVENT_PACKET) return;
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#ifdef STORE_TO_SBC_FILE
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fwrite(packet+pos, size-pos, 1, sbc_file);
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#endif
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if (hci_event_packet_get_type(packet) == HCI_EVENT_PIN_CODE_REQUEST){
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// inform about pin code request
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bd_addr_t event_addr;
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printf("Pin code request - using '0000'\n");
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hci_event_pin_code_request_get_bd_addr(packet, event_addr);
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gap_pin_code_response(event_addr, "0000");
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const btstack_audio_sink_t * audio = btstack_audio_sink_get_instance();
|
||||
// process data right away if there's no audio implementation active, e.g. on posix systems to store as .wav
|
||||
if (!audio){
|
||||
btstack_sbc_decoder_process_data(&state, 0, packet+pos, size-pos);
|
||||
return;
|
||||
}
|
||||
|
||||
if (hci_event_packet_get_type(packet) != HCI_EVENT_AVDTP_META) return;
|
||||
UNUSED(channel);
|
||||
// store sbc frame size for buffer management
|
||||
sbc_frame_size = (size-pos)/ sbc_header.num_frames;
|
||||
|
||||
int status = btstack_ring_buffer_write(&sbc_frame_ring_buffer, packet+pos, size-pos);
|
||||
if (status){
|
||||
printf("Error storing samples in SBC ring buffer!!!\n");
|
||||
}
|
||||
|
||||
// decide on audio sync drift based on number of sbc frames in queue
|
||||
int sbc_frames_in_buffer = btstack_ring_buffer_bytes_available(&sbc_frame_ring_buffer) / sbc_frame_size;
|
||||
uint32_t resampling_factor;
|
||||
|
||||
// nomimal factor (fixed-point 2^16) and compensation offset
|
||||
uint32_t nomimal_factor = 0x10000;
|
||||
uint32_t compensation = 0x00100;
|
||||
|
||||
if (sbc_frames_in_buffer < OPTIMAL_FRAMES_MIN){
|
||||
resampling_factor = nomimal_factor - compensation; // stretch samples
|
||||
} else if (sbc_frames_in_buffer <= OPTIMAL_FRAMES_MAX){
|
||||
resampling_factor = nomimal_factor; // nothing to do
|
||||
} else {
|
||||
resampling_factor = nomimal_factor + compensation; // compress samples
|
||||
}
|
||||
|
||||
btstack_resample_set_factor(&resample_instance, resampling_factor);
|
||||
|
||||
// start stream if enough frames buffered
|
||||
if (!audio_stream_started && sbc_frames_in_buffer >= OPTIMAL_FRAMES_MIN){
|
||||
audio_stream_started = 1;
|
||||
// setup audio playback
|
||||
if (audio){
|
||||
audio->start_stream();
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packet, uint16_t size){
|
||||
UNUSED(channel);
|
||||
UNUSED(size);
|
||||
uint8_t status;
|
||||
|
||||
if (packet_type != HCI_EVENT_PACKET) return;
|
||||
if (hci_event_packet_get_type(packet) != HCI_EVENT_AVDTP_META) return;
|
||||
|
||||
switch (packet[2]){
|
||||
case AVDTP_SUBEVENT_SIGNALING_CONNECTION_ESTABLISHED:
|
||||
avdtp_cid = avdtp_subevent_signaling_connection_established_get_avdtp_cid(packet);
|
||||
@ -477,10 +521,12 @@ static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packe
|
||||
}
|
||||
printf("AVDTP Sink connected: avdtp_cid 0x%02x.\n", avdtp_cid);
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_CONNECTION_RELEASED:
|
||||
avdtp_cid = avdtp_subevent_signaling_connection_released_get_avdtp_cid(packet);
|
||||
printf("AVDTP connection released: avdtp_cid 0x%02x.\n", avdtp_cid);
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_SEP_FOUND:
|
||||
remote_seid = avdtp_subevent_signaling_sep_found_get_remote_seid(packet);
|
||||
printf("Found sep: seid %u, in_use %d, media type %d, sep type %d (1-SNK)\n",
|
||||
@ -499,6 +545,7 @@ static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packe
|
||||
sbc_capability.max_bitpool_value = avdtp_subevent_signaling_media_codec_sbc_capability_get_max_bitpool_value(packet);
|
||||
dump_sbc_capability(sbc_capability);
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_MEDIA_TRANSPORT_CAPABILITY:
|
||||
printf("CAPABILITY - MEDIA_TRANSPORT supported on remote.\n");
|
||||
break;
|
||||
@ -537,9 +584,13 @@ static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packe
|
||||
printf(" - media %d\n", avdtp_subevent_signaling_header_compression_capability_get_media(packet));
|
||||
printf(" - recovery %d\n", avdtp_subevent_signaling_header_compression_capability_get_recovery(packet));
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_MEDIA_CODEC_OTHER_CAPABILITY:
|
||||
printf("Received non SBC codec, event not parsed.\n");
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_MEDIA_CODEC_SBC_CONFIGURATION:{
|
||||
printf("Received SBC codec configuration\n");
|
||||
printf("Received SBC codec configuration: avdtp_cid 0x%02x.\n", avdtp_cid);
|
||||
sbc_configuration.reconfigure = avdtp_subevent_signaling_media_codec_sbc_configuration_get_reconfigure(packet);
|
||||
sbc_configuration.num_channels = avdtp_subevent_signaling_media_codec_sbc_configuration_get_num_channels(packet);
|
||||
sbc_configuration.sampling_frequency = avdtp_subevent_signaling_media_codec_sbc_configuration_get_sampling_frequency(packet);
|
||||
@ -553,33 +604,32 @@ static void packet_handler(uint8_t packet_type, uint16_t channel, uint8_t *packe
|
||||
dump_sbc_configuration(sbc_configuration);
|
||||
|
||||
avdtp_sink_delay_report(avdtp_cid, remote_seid, 100);
|
||||
media_processing_close();
|
||||
media_processing_init(sbc_configuration);
|
||||
break;
|
||||
}
|
||||
|
||||
case AVDTP_SUBEVENT_STREAMING_CONNECTION_ESTABLISHED:
|
||||
printf("Streaming connection opened.\n");
|
||||
avdtp_cid = avdtp_subevent_streaming_connection_established_get_avdtp_cid(packet);
|
||||
printf("Streaming connection opened: avdtp_cid 0x%02x.\n", avdtp_cid);
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_STREAMING_CONNECTION_RELEASED:
|
||||
printf("Streaming connection released.\n");
|
||||
printf("Streaming connection released: avdtp_cid 0x%02x.\n", avdtp_cid);
|
||||
is_cmd_triggered_localy = 0;
|
||||
is_media_header_reported_once = 0;
|
||||
media_processing_close();
|
||||
break;
|
||||
case AVDTP_SUBEVENT_SIGNALING_MEDIA_CODEC_OTHER_CAPABILITY:
|
||||
printf("Received non SBC codec, event not parsed.\n");
|
||||
break;
|
||||
|
||||
case AVDTP_SUBEVENT_SIGNALING_ACCEPT:{
|
||||
switch (avdtp_subevent_signaling_accept_get_signal_identifier(packet)){
|
||||
case AVDTP_SI_START:
|
||||
printf("Stream started\n");
|
||||
media_processing_close();
|
||||
media_processing_init(sbc_configuration);
|
||||
media_processing_start();
|
||||
break;
|
||||
case AVDTP_SI_SUSPEND:
|
||||
printf("Stream paused\n");
|
||||
media_processing_close();
|
||||
media_processing_pause();
|
||||
break;
|
||||
case AVDTP_SI_ABORT:
|
||||
case AVDTP_SI_CLOSE:
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user