Open sound device for capture or playback only, if full duplex is not required
This commit is contained in:
parent
ce58b765f5
commit
b613123291
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@ -346,7 +346,7 @@ int main(int argc, char *argv[])
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#ifdef HAVE_ALSA
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/* init sound */
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audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
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audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
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if (!audio) {
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LOGP(DBNETZ, LOGL_ERROR, "No sound device!\n");
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goto exit;
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@ -1397,7 +1397,7 @@ int datenklo_open_audio(datenklo_t *datenklo, const char *audiodev, int buffer,
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#ifdef HAVE_ALSA
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/* init sound */
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datenklo->audio = sound_open(audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->buffer_size, 1.0, 1.0, 4000.0, 2.0);
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datenklo->audio = sound_open(SOUND_DIR_DUPLEX, audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->buffer_size, 1.0, 1.0, 4000.0, 2.0);
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if (!datenklo->audio) {
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LOGP(DDATENKLO, LOGL_ERROR, "No sound device!\n");
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return -EIO;
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@ -413,13 +413,19 @@ static int get_char()
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int soundif_open(const char *audiodev, int samplerate, int buffer_size)
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{
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enum sound_direction direction = SOUND_DIR_DUPLEX;
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if (!audiodev || !audiodev[0]) {
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LOGP(DDSP, LOGL_ERROR, "No audio device given!\n");
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return -EINVAL;
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}
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/* open audiodev */
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soundif = sound_open(audiodev, NULL, NULL, NULL, (double_amplitude) ? 2 : 1, 0.0, samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
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if (tx && !rx)
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direction = SOUND_DIR_PLAY;
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if (rx && !tx)
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direction = SOUND_DIR_REC;
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soundif = sound_open(direction, audiodev, NULL, NULL, NULL, (double_amplitude) ? 2 : 1, 0.0, samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
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if (!soundif) {
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LOGP(DDSP, LOGL_ERROR, "Failed to open sound device!\n");
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return -EIO;
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@ -451,34 +457,38 @@ void soundif_work(int buffer_size)
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int rc;
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int i;
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/* encode and write */
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count = sound_get_tosend(soundif, buffer_size);
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if (count < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to get number of samples in buffer (rc = %d)!\n", count);
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return;
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}
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if (count) {
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dcf77_encode(dcf77, samples[0], count);
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if (double_amplitude) {
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for (i = 0; i < count; i++)
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samples[1][i] = -samples[0][i];
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}
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rc = sound_write(soundif, samples, NULL, count, NULL, NULL, (double_amplitude) ? 2 : 1);
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if (rc < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to write TX data to audio device (rc = %d)\n", rc);
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if (tx) {
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/* encode and write */
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count = sound_get_tosend(soundif, buffer_size);
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if (count < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to get number of samples in buffer (rc = %d)!\n", count);
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return;
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}
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if (count) {
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dcf77_encode(dcf77, samples[0], count);
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if (double_amplitude) {
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for (i = 0; i < count; i++)
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samples[1][i] = -samples[0][i];
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}
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rc = sound_write(soundif, samples, NULL, count, NULL, NULL, (double_amplitude) ? 2 : 1);
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if (rc < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to write TX data to audio device (rc = %d)\n", rc);
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return;
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}
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}
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}
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/* read */
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count = sound_read(soundif, samples, buffer_size, 1, rf_level_db);
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if (count < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to read from audio device (rc = %d)!\n", count);
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return;
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}
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if (rx) {
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/* read */
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count = sound_read(soundif, samples, buffer_size, 1, rf_level_db);
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if (count < 0) {
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LOGP(DDSP, LOGL_ERROR, "Failed to read from audio device (rc = %d)!\n", count);
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return;
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}
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/* decode */
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dcf77_decode(dcf77, samples[0], count);
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/* decode */
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dcf77_decode(dcf77, samples[0], count);
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}
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}
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int main(int argc, char *argv[])
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@ -307,7 +307,7 @@ int main(int argc, char *argv[])
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#ifdef HAVE_ALSA
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/* init sound */
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audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, dsp_buffer, 1.0, 1.0, 4000.0, 2.0);
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audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, dsp_buffer, 1.0, 1.0, 4000.0, 2.0);
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if (!audio) {
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LOGP(DBNETZ, LOGL_ERROR, "No sound device!\n");
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goto exit;
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@ -282,7 +282,7 @@ void jitter_save(jitter_t *jb, jitter_frame_t *jf)
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offset_timestamp = jf->timestamp - jb->window_timestamp;
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#ifdef HEAVY_DEBUG
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LOGP(DJITTER, LOGL_DEBUG, "%sFrame has offset of %.0fms in jitter buffer.\n", jb->name, (double)offset_timestamp * jb->sample_duration * 1000.0);
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LOGP(DJITTER, LOGL_DEBUG, "%s Frame has offset of %.0fms in jitter buffer.\n", jb->name, (double)offset_timestamp * jb->sample_duration * 1000.0);
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#endif
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/* measure delay */
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@ -348,7 +348,7 @@ int32_t jitter_offset(jitter_t *jb)
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jb->unlocked = true;
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/* get timestamp of chunk that is not in the past */
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while ((jf = jb->frame_list)) {
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for (jf = jb->frame_list; jf; jf = jf->next) {
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offset_timestamp = jf->timestamp - jb->window_timestamp;
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if (offset_timestamp >= 0)
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break;
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@ -446,6 +446,9 @@ copy_chunk:
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tocopy = jb->spl_len - jb->spl_pos;
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if (tocopy > len)
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tocopy = len;
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#ifdef HEAVY_DEBUG
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LOGP(DJITTER, LOGL_DEBUG, "%s loading %d samples: from valid sample buffer.\n", jb->name, tocopy);
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#endif
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/* advance jitter buffer */
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jitter_advance(jb, tocopy);
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memcpy(spl, jb->spl_buf + jb->spl_pos * sample_size, tocopy * sample_size);
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@ -469,6 +472,9 @@ copy_chunk:
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/* only process as much samples as need */
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if (offset > len)
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offset = len;
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#ifdef HEAVY_DEBUG
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LOGP(DJITTER, LOGL_DEBUG, "%s concealing %d samples: from invalid sample buffer.\n", jb->name, offset);
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#endif
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/* advance jitter buffer */
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jitter_advance(jb, offset);
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/* if there is no buffer, allocate 20ms, filled with 0 */
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jitter_reset(jb);
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return;
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}
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#ifdef HEAVY_DEBUG
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LOGP(DJITTER, LOGL_DEBUG, "%s loading new frame to sample buffer.\n", jb->name);
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#endif
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/* get data from frame */
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jitter_frame_get(jf, &decoder, &decoder_priv, &payload, &payload_len, NULL, NULL, NULL, NULL);
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/* free previous buffer */
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@ -404,7 +404,7 @@ int console_open_audio(int __attribute__((unused)) buffer_size, double __attribu
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#ifdef HAVE_ALSA
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/* open sound device for call control */
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/* use factor 1.4 of speech level for complete range of sound card */
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console.sound = sound_open(console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, buffer_size, interval, 1.4, 4000.0, 2.0);
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console.sound = sound_open(SOUND_DIR_DUPLEX, console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, buffer_size, interval, 1.4, 4000.0, 2.0);
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if (!console.sound) {
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LOGP(DSENDER, LOGL_ERROR, "No sound device!\n");
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return -EIO;
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@ -235,7 +235,7 @@ int sender_open_audio(int buffer_size, double interval)
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}
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/* open device */
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master->audio = master->audio_open(master->device, tx_f, rx_f, am, channels, paging_frequency, master->samplerate, buffer_size, interval, (master->max_deviation) ?: 1.0, master->max_modulation, master->modulation_index);
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master->audio = master->audio_open(SOUND_DIR_DUPLEX, master->device, tx_f, rx_f, am, channels, paging_frequency, master->samplerate, buffer_size, interval, (master->max_deviation) ?: 1.0, master->max_modulation, master->modulation_index);
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if (!master->audio) {
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LOGP(DSENDER, LOGL_ERROR, "No device for transceiver!\n");
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return -EIO;
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@ -44,7 +44,7 @@ typedef struct sender {
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/* audio */
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void *audio;
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char device[64]; /* audio device name (alsa or sdr) */
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void *(*audio_open)(const char *, double *, double *, int *, int, double, int, int, double, double, double, double);
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void *(*audio_open)(int, const char *, double *, double *, int *, int, double, int, int, double, double, double, double);
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int (*audio_start)(void *);
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void (*audio_close)(void *);
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int (*audio_write)(void *, sample_t **, uint8_t **, int, enum paging_signal *, int *, int);
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@ -137,7 +137,7 @@ static void show_spectrum(const char *direction, double halfbandwidth, double ce
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LOGP(DSDR, LOGL_INFO, "Frequency P = %.4f MHz (Paging Frequency)\n", paging_frequency / 1e6);
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}
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void *sdr_open(const char __attribute__((__unused__)) *device, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index)
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void *sdr_open(int __attribute__((__unused__)) direction, const char __attribute__((__unused__)) *device, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index)
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{
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sdr_t *sdr;
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int threads = 1, oversample = 1; /* always use threads */
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@ -2,7 +2,7 @@
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enum paging_signal;
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int sdr_start(void *inst);
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void *sdr_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
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void *sdr_open(int direction, const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
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void sdr_close(void *inst);
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int sdr_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
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int sdr_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);
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@ -1,7 +1,13 @@
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enum paging_signal;
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void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
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enum sound_direction {
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SOUND_DIR_PLAY,
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SOUND_DIR_REC,
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SOUND_DIR_DUPLEX,
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};
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void *sound_open(int direction, const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
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int sound_start(void *inst);
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void sound_close(void *inst);
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int sound_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
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@ -32,6 +32,7 @@
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static int KEEP_FRAMES=8; /* minimum frames not to read, to prevent reading from buffer before data has been received (seems to be a bug in ALSA) */
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typedef struct sound {
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enum sound_direction direction;
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snd_pcm_t *phandle, *chandle;
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int pchannels, cchannels;
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int channels; /* required number of channels */
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@ -125,49 +126,57 @@ error:
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static int dev_open(sound_t *sound)
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{
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int rc, rc_rec, rc_play;
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int rc, rc_rec = 0, rc_play = 0;
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rc_play = snd_pcm_open(&sound->phandle, sound->paudiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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rc_rec = snd_pcm_open(&sound->chandle, sound->caudiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
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if (rc_play < 0)
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LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports playing audio.\n", sound->paudiodev, snd_strerror(rc_play));
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if (rc_rec < 0)
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LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports capturing audio.\n", sound->caudiodev, snd_strerror(rc_rec));
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if (sound->direction == SOUND_DIR_PLAY || sound->direction == SOUND_DIR_DUPLEX) {
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rc_play = snd_pcm_open(&sound->phandle, sound->paudiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (rc_play < 0)
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LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports playing audio.\n", sound->paudiodev, snd_strerror(rc_play));
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}
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if (sound->direction == SOUND_DIR_REC || sound->direction == SOUND_DIR_DUPLEX) {
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rc_rec = snd_pcm_open(&sound->chandle, sound->caudiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
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if (rc_rec < 0)
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LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports capturing audio.\n", sound->caudiodev, snd_strerror(rc_rec));
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}
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if (rc_play < 0 || rc_rec < 0)
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return (rc_play < 0) ? rc_play : rc_rec;
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rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "Failed to set playback hw params\n");
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return rc;
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}
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if (sound->pchannels < sound->channels) {
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LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
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return rc;
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}
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LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
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if (sound->direction == SOUND_DIR_PLAY || sound->direction == SOUND_DIR_DUPLEX) {
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rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "Failed to set playback hw params\n");
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return rc;
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}
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if (sound->pchannels < sound->channels) {
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LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
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return rc;
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}
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LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
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rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "Failed to set capture hw params\n");
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return rc;
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}
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if (sound->cchannels < sound->channels) {
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LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
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return -EIO;
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}
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LOGP(DSOUND, LOGL_DEBUG, "Capture with %d channels.\n", sound->cchannels);
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rc = snd_pcm_prepare(sound->phandle);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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rc = snd_pcm_prepare(sound->phandle);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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}
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}
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rc = snd_pcm_prepare(sound->chandle);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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if (sound->direction == SOUND_DIR_REC || sound->direction == SOUND_DIR_DUPLEX) {
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rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "Failed to set capture hw params\n");
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return rc;
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}
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if (sound->cchannels < sound->channels) {
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LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
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return -EIO;
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}
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LOGP(DSOUND, LOGL_DEBUG, "Capture with %d channels.\n", sound->cchannels);
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rc = snd_pcm_prepare(sound->chandle);
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if (rc < 0) {
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LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
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return rc;
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}
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}
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return 0;
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@ -181,7 +190,7 @@ static void dev_close(sound_t *sound)
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snd_pcm_close(sound->chandle);
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}
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void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) buffer_size, double __attribute__((unused)) interval, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
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void *sound_open(int direction, const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) buffer_size, double __attribute__((unused)) interval, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
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{
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sound_t *sound;
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const char *env;
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@ -206,6 +215,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
|
|||
} else {
|
||||
sound->caudiodev = sound->paudiodev;
|
||||
}
|
||||
sound->direction = direction;
|
||||
sound->channels = channels;
|
||||
sound->samplerate = samplerate;
|
||||
sound->spl_deviation = max_deviation / 32767.0;
|
||||
|
@ -249,6 +259,9 @@ int sound_start(void *inst)
|
|||
sound_t *sound = (sound_t *)inst;
|
||||
int16_t buff[2];
|
||||
|
||||
if (sound->direction != SOUND_DIR_REC && sound->direction != SOUND_DIR_DUPLEX)
|
||||
return -EINVAL;
|
||||
|
||||
/* trigger capturing */
|
||||
snd_pcm_readi(sound->chandle, buff, 1);
|
||||
|
||||
|
@ -319,6 +332,9 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
|
|||
int rc;
|
||||
int i, ii;
|
||||
|
||||
if (sound->direction != SOUND_DIR_PLAY && sound->direction != SOUND_DIR_DUPLEX)
|
||||
return -EINVAL;
|
||||
|
||||
if (sound->pchannels == 2) {
|
||||
/* two channels */
|
||||
#ifdef HAVE_MOBILE
|
||||
|
@ -404,6 +420,9 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf
|
|||
int in, rc;
|
||||
int i, ii;
|
||||
|
||||
if (sound->direction != SOUND_DIR_REC && sound->direction != SOUND_DIR_DUPLEX)
|
||||
return -EINVAL;
|
||||
|
||||
/* get samples in rx buffer */
|
||||
in = snd_pcm_avail(sound->chandle);
|
||||
/* if not more than KEEP_FRAMES frames available, try next time */
|
||||
|
@ -501,6 +520,9 @@ int sound_get_tosend(void *inst, int buffer_size)
|
|||
snd_pcm_sframes_t delay;
|
||||
int tosend;
|
||||
|
||||
if (sound->direction != SOUND_DIR_PLAY && sound->direction != SOUND_DIR_DUPLEX)
|
||||
return -EINVAL;
|
||||
|
||||
rc = snd_pcm_delay(sound->phandle, &delay);
|
||||
if (rc < 0) {
|
||||
if (rc == -32)
|
||||
|
|
|
@ -268,7 +268,7 @@ inval_number:
|
|||
|
||||
#ifdef HAVE_ALSA
|
||||
/* open audio device */
|
||||
sound = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
sound = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
if (!sound) {
|
||||
rc = -EIO;
|
||||
LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");
|
||||
|
|
|
@ -272,7 +272,7 @@ int main(int argc, char *argv[])
|
|||
|
||||
#ifdef HAVE_ALSA
|
||||
/* init sound */
|
||||
audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
|
||||
audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
|
||||
if (!audio) {
|
||||
LOGP(DDSP, LOGL_ERROR, "No sound device!\n");
|
||||
goto exit;
|
||||
|
|
|
@ -396,7 +396,7 @@ int main(int argc, char *argv[])
|
|||
tx_frequencies[0] = frequency;
|
||||
rx_frequencies[0] = frequency;
|
||||
am[0] = 0;
|
||||
sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
|
||||
sdr = sdr_open(0, NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
|
||||
if (!sdr)
|
||||
goto error;
|
||||
sdr_start(sdr);
|
||||
|
|
|
@ -100,7 +100,7 @@ int radio_init(radio_t *radio, int buffer_size, int samplerate, double frequency
|
|||
/* open audio device */
|
||||
radio->tx_audio_samplerate = 48000;
|
||||
radio->tx_audio_channels = (stereo) ? 2 : 1;
|
||||
radio->tx_sound = sound_open(tx_audiodev, NULL, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
radio->tx_sound = sound_open(SOUND_DIR_PLAY, tx_audiodev, NULL, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
if (!radio->tx_sound) {
|
||||
rc = -EIO;
|
||||
LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");
|
||||
|
@ -164,10 +164,7 @@ int radio_init(radio_t *radio, int buffer_size, int samplerate, double frequency
|
|||
radio->rx_audio_samplerate = 48000;
|
||||
radio->rx_audio_channels = (stereo) ? 2 : 1;
|
||||
/* check if we use same device */
|
||||
if (radio->tx_sound && !strcmp(tx_audiodev, rx_audiodev))
|
||||
radio->rx_sound = radio->tx_sound;
|
||||
else
|
||||
radio->rx_sound = sound_open(rx_audiodev, NULL, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
radio->rx_sound = sound_open(SOUND_DIR_REC, rx_audiodev, NULL, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
|
||||
if (!radio->rx_sound) {
|
||||
rc = -EIO;
|
||||
LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");
|
||||
|
|
|
@ -342,7 +342,7 @@ static void tx_bas(sample_t *sample_bas, __attribute__((__unused__)) sample_t *s
|
|||
tx_frequencies[0] = frequency;
|
||||
rx_frequencies[0] = frequency;
|
||||
am[0] = 0;
|
||||
sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
|
||||
sdr = sdr_open(0, NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
|
||||
if (!sdr)
|
||||
goto error;
|
||||
sdr_start(sdr);
|
||||
|
|
Loading…
Reference in New Issue