/**
* @brief Process a single stereo audio sample.
*
* Operates only on as_conf_cache, and requires no synchronization as
* long as the calling thread is always iq_thread.
*/
void audio_stream_put_samples(short left_sample,short right_sample) {
int audio_buffer_length;
sdr_thread_assert_id(&audiostream_tid);
/* FIXME: This really only applies once, at startup */
if (!audio_buffer)
allocate_audio_buffer();
// samples are delivered at 48K
// output to stream at 8K (1 in 6) or 48K (1 in 1)
// codec2 encoding works only for 8K
if(sample_count==0) {
int offset;
// use this sample and convert to a-law or PCM or codec2
if(as_conf_cache.channels==1) {
switch (as_conf_cache.encoding) {
default: /* ALAW */
case ENCODING_ALAW:
offset = audio_stream_buffer_insert + AUDIO_BUFFER_HEADER_SIZE;
audio_buffer[offset] = alaw((left_sample + right_sample) / 2);
break;
case ENCODING_PCM:
offset = audio_stream_buffer_insert * 2 + AUDIO_BUFFER_HEADER_SIZE;
/* PCM on the wire is always LE? */
audio_buffer[offset] = (left_sample/2 + right_sample/2) & 0xff;
audio_buffer[offset + 1] = (left_sample/2 + right_sample/2) >> 8;
break;
case ENCODING_CODEC2:
codec2_buffer[audio_stream_buffer_insert] = left_sample/2 + right_sample/2;
break;
}
} else {
switch (as_conf_cache.encoding) {
/**
* @brief Process a single stereo audio sample.
*
* Operates only on as_conf_cache, and requires no synchronization as
* long as the calling thread is always iq_thread.
*/
void audio_stream_put_samples(short left_sample,short right_sample) {
int audio_buffer_length;
sdr_thread_assert_id(&audiostream_tid);
/* FIXME: This really only applies once, at startup */
if (!audio_buffer) {
allocate_audio_buffer();
samples_per_frame = codec2_samples_per_frame( codec2 );
bits_per_frame = codec2_bits_per_frame( codec2 );
codec2_buffer = (short *) malloc( sizeof( short ) * samples_per_frame );
bits = (unsigned char *) malloc( sizeof( unsigned char ) * BITS_SIZE );
}
// samples are delivered at 48K or 8K depending on audiostream_conf.samplerate
// codec2 encoding works only for 8K
int offset;
// use this sample and convert to a-law or PCM or codec2
if(as_conf_cache.channels==1) {
switch (as_conf_cache.encoding) {
default: /* ALAW */
case ENCODING_ALAW:
offset = audio_stream_buffer_insert + AUDIO_BUFFER_HEADER_SIZE;
audio_buffer[offset] = alaw((left_sample + right_sample) / 2);
break;
case ENCODING_PCM:
offset = audio_stream_buffer_insert * 2 + AUDIO_BUFFER_HEADER_SIZE;
/* PCM on the wire is always LE? */
audio_buffer[offset] = (left_sample/2 + right_sample/2) & 0xff;
audio_buffer[offset + 1] = (left_sample/2 + right_sample/2) >> 8;
break;
case ENCODING_CODEC2:
codec2_buffer[audio_stream_buffer_insert] = left_sample/2 + right_sample/2;
break;
}
} else {
switch (as_conf_cache.encoding) {
void allocate_audio_buffer(){
int samplesize;
sdr_thread_assert_id(&audiostream_tid);
sem_wait(&audiostream_sem);
if (audiostream_conf.age > 0) {
/* The stream configuration has changed, and we need to update
* our copy to reflect that. */
memcpy(&as_conf_cache, &audiostream_conf, sizeof(as_conf_cache));
audiostream_conf.age = 0;
}
sem_post(&audiostream_sem);
/* From this point until the buffer allocated here is handed to the
* client or discarded, all iq_thread operations are on
* as_conf_cache, NOT audiostream_conf. */
switch (as_conf_cache.encoding) {
case ENCODING_ALAW:
/* 8 bits per sample */
samplesize = as_conf_cache.bufsize * as_conf_cache.channels;
break;
case ENCODING_PCM:
/* 16 bit per sample */
samplesize = as_conf_cache.bufsize * as_conf_cache.channels * 2;
break;
case ENCODING_CODEC2:
/* FIXME: This seems like the wrong place for this? */
codec2_count = 0;
/* Force buffer size */
as_conf_cache.bufsize = BITS_SIZE * NO_CODEC2_FRAMES;
samplesize = as_conf_cache.bufsize * as_conf_cache.channels;
break;
default:
/* No ENCODING_ALAW2! */
samplesize = 0;
}
audio_buffer = malloc(samplesize + AUDIO_BUFFER_HEADER_SIZE);
}
