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osmocom-analog/src/libsound/sound_alsa.c

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/* Sound device access
*
* (C) 2016 by Andreas Eversberg <jolly@eversberg.eu>
* All Rights Reserved
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdlib.h>
#include <stdint.h>
#include <math.h>
#include <alsa/asoundlib.h>
#include "../common/sample.h"
#include "../common/debug.h"
#include "../common/sender.h"
typedef struct sound {
snd_pcm_t *phandle, *chandle;
int pchannels, cchannels;
double spl_deviation; /* how much deviation is one sample step */
double paging_phaseshift; /* phase to shift every sample */
double paging_phase; /* current phase */
double rx_frequency[2]; /* rx frequency of radio connected to channel */
dispmeasparam_t *dmp[2];
} sound_t;
static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
{
snd_pcm_hw_params_t *hw_params = NULL;
int rc;
unsigned int rrate;
rc = snd_pcm_hw_params_malloc(&hw_params);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to allocate hw_params! (%s)\n", snd_strerror(rc));
goto error;
}
rc = snd_pcm_hw_params_any(handle, hw_params);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot initialize hardware parameter structure (%s)\n", snd_strerror(rc));
goto error;
}
rc = snd_pcm_hw_params_set_rate_resample(handle, hw_params, 0);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set real hardware rate (%s)\n", snd_strerror(rc));
goto error;
}
rc = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set access to interleaved (%s)\n", snd_strerror(rc));
goto error;
}
rc = snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample format (%s)\n", snd_strerror(rc));
goto error;
}
rrate = samplerate;
rc = snd_pcm_hw_params_set_rate_near(handle, hw_params, &rrate, 0);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample rate (%s)\n", snd_strerror(rc));
goto error;
}
if ((int)rrate != samplerate) {
PDEBUG(DSOUND, DEBUG_ERROR, "Rate doesn't match (requested %dHz, get %dHz)\n", samplerate, rrate);
rc = -EIO;
goto error;
}
*channels = 1;
rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
if (rc < 0) {
*channels = 2;
rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set channel count to 1 nor 2 (%s)\n", snd_strerror(rc));
goto error;
}
}
rc = snd_pcm_hw_params(handle, hw_params);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot set parameters (%s)\n", snd_strerror(rc));
goto error;
}
snd_pcm_hw_params_free(hw_params);
return 0;
error:
if (hw_params) {
snd_pcm_hw_params_free(hw_params);
}
return rc;
}
static int sound_prepare(sound_t *sound)
{
int rc;
rc = snd_pcm_prepare(sound->phandle);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
}
rc = snd_pcm_prepare(sound->chandle);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
}
return 0;
}
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation)
{
sound_t *sound;
int rc;
if (channels < 1 || channels > 2) {
PDEBUG(DSOUND, DEBUG_ERROR, "Cannot use more than two channels with the same sound card!\n");
return NULL;
}
sound = calloc(1, sizeof(sound_t));
if (!sound) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to alloc memory!\n");
return NULL;
}
sound->spl_deviation = max_deviation / 32767.0;
sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
rc = snd_pcm_open(&sound->phandle, audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for playback! (%s)\n", audiodev, snd_strerror(rc));
goto error;
}
rc = snd_pcm_open(&sound->chandle, audiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for capture! (%s)\n", audiodev, snd_strerror(rc));
goto error;
}
rc = set_hw_params(sound->phandle, samplerate, &sound->pchannels);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set playback hw params\n");
goto error;
}
if (sound->pchannels < channels) {
PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
goto error;
}
PDEBUG(DSOUND, DEBUG_DEBUG, "Playback with %d channels.\n", sound->pchannels);
rc = set_hw_params(sound->chandle, samplerate, &sound->cchannels);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set capture hw params\n");
goto error;
}
if (sound->cchannels < channels) {
PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
goto error;
}
PDEBUG(DSOUND, DEBUG_DEBUG, "Capture with %d channels.\n", sound->cchannels);
rc = sound_prepare(sound);
if (rc < 0)
goto error;
if (rx_frequency) {
sender_t *sender;
int i;
for (i = 0; i < channels; i++) {
sound->rx_frequency[i] = rx_frequency[i];
sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
if (!sender)
continue;
sound->dmp[i] = display_measurements_add(sender, "RX Level", "%.1f dB", DISPLAY_MEAS_PEAK, DISPLAY_MEAS_LEFT, -96.0, 0.0, -INFINITY);
}
}
return sound;
error:
sound_close(sound);
return NULL;
}
/* start streaming */
int sound_start(void *inst)
{
sound_t *sound = (sound_t *)inst;
int16_t buff[2];
/* trigger capturing */
snd_pcm_readi(sound->chandle, buff, 1);
return 0;
}
void sound_close(void *inst)
{
sound_t *sound = (sound_t *)inst;
if (sound->phandle != NULL)
snd_pcm_close(sound->phandle);
if (sound->chandle != NULL)
snd_pcm_close(sound->chandle);
free(sound);
}
static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum paging_signal paging_signal, int on)
{
double phaseshift, phase;
int i;
switch (paging_signal) {
case PAGING_SIGNAL_NOTONE:
/* no tone if paging signal is on */
on = !on;
// fall through
case PAGING_SIGNAL_TONE:
/* tone if paging signal is on */
if (on) {
phaseshift = sound->paging_phaseshift;
phase = sound->paging_phase;
for (i = 0; i < length; i++) {
if (phase < 0.5)
*samples++ = 30000;
else
*samples++ = -30000;
phase += phaseshift;
if (phase >= 1.0)
phase -= 1.0;
}
sound->paging_phase = phase;
} else
memset(samples, 0, length << 1);
break;
case PAGING_SIGNAL_NEGATIVE:
/* negative signal if paging signal is on */
on = !on;
// fall through
case PAGING_SIGNAL_POSITIVE:
/* positive signal if paging signal is on */
if (on)
memset(samples, 127, length << 1);
else
memset(samples, 128, length << 1);
break;
case PAGING_SIGNAL_NONE:
break;
}
}
int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused)) **power, int num, enum paging_signal *paging_signal, int *on, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int32_t value;
int16_t buff[num << 1];
int rc;
int i, ii;
if (sound->pchannels == 2) {
/* two channels */
if (paging_signal && on && paging_signal[0] != PAGING_SIGNAL_NONE) {
int16_t paging[num << 1];
gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
buff[ii++] = paging[i];
}
} else if (channels == 2) {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
value = samples[1][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
}
} else {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
buff[ii++] = value;
}
}
} else {
/* one channel */
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
}
}
rc = snd_pcm_writei(sound->phandle, buff, num);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
if (rc != num)
PDEBUG(DSOUND, DEBUG_ERROR, "short write to audio interface, written %d bytes, got %d bytes\n", num, rc);
return rc;
}
#define KEEP_FRAMES 8 /* minimum frames not to read, due to bug in ALSA */
int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int16_t buff[num << 1];
int32_t spl;
int32_t max[2], a;
int in, rc;
int i, ii;
/* get samples in rx buffer */
in = snd_pcm_avail(sound->chandle);
/* if not more than KEEP_FRAMES frames available, try next time */
if (in <= KEEP_FRAMES)
return 0;
/* read some frames less than in buffer, because snd_pcm_readi() seems
* to corrupt last frames */
in -= KEEP_FRAMES;
if (in > num)
in = num;
rc = snd_pcm_readi(sound->chandle, buff, in);
if (rc < 0) {
if (errno == EAGAIN)
return 0;
PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
/* recover read */
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
if (rc == 0)
return rc;
if (sound->cchannels == 2) {
if (channels < 2) {
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
spl += buff[ii++];
a = (spl >= 0) ? spl : -spl;
if (i == 0 || a > max[0])
max[0] = a;
samples[0][i] = (double)spl * spl_deviation;
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
a = (spl >= 0) ? spl : -spl;
if (i == 0 || a > max[0])
max[0] = a;
samples[0][i] = (double)spl * spl_deviation;
spl = buff[ii++];
a = (spl >= 0) ? spl : -spl;
if (i == 0 || a > max[1])
max[1] = a;
samples[1][i] = (double)spl * spl_deviation;
}
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
a = (spl >= 0) ? spl : -spl;
if (i == 0 || a > max[0])
max[0] = a;
samples[0][i] = (double)spl * spl_deviation;
}
}
sender_t *sender;
for (i = 0; i < channels; i++) {
sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
if (!sender)
continue;
display_measurements_update(sound->dmp[i], log10((double)max[i] / 32768.0) * 20, 0.0);
rf_level_db[i] = 0.0;
}
return rc;
}
/*
* get playback buffer space
*
* return number of samples to be sent */
int sound_get_tosend(void *inst, int latspl)
{
sound_t *sound = (sound_t *)inst;
int rc;
snd_pcm_sframes_t delay;
int tosend;
rc = snd_pcm_delay(sound->phandle, &delay);
if (rc < 0) {
if (rc == -32)
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency and enable real time scheduling\n");
else
PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE) {
sound_prepare(sound);
sound_start(sound);
}
return rc;
}
tosend = latspl - delay;
return tosend;
}
int sound_is_stereo_capture(void *inst)
{
sound_t *sound = (sound_t *)inst;
if (sound->cchannels == 2)
return 1;
return 0;
}
int sound_is_stereo_playback(void *inst)
{
sound_t *sound = (sound_t *)inst;
if (sound->pchannels == 2)
return 1;
return 0;
}
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