williamp/osmocom-analog
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Move from local to external osmo* libraries

Browse Source 
src/libdebug -> libosmocore
src/libselect -> libosmocore
src/libtimer -> libosmocore
src/libosmocc -> libosmo-cc
src/libg711 -> libosmo-cc
This commit is contained in:
Andreas Eversberg
2024-01-05 14:20:36 +01:00
parent 6cd2c3e323
commit 3158c48365
198 changed files with 4202 additions and 12564 deletions
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62
src/libsound/sound_alsa.c
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@@ -22,7 +22,7 @@
#include <math.h>
#include <alsa/asoundlib.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
#include "../liblogging/logging.h"
#ifdef HAVE_MOBILE
#include "../libmobile/sender.h"
#else
@@ -54,42 +54,42 @@ static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_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);
LOGP(DSOUND, LOGL_ERROR, "Rate doesn't match (requested %dHz, get %dHz)\n", samplerate, rrate);
rc = -EIO;
goto error;
}
@@ -100,14 +100,14 @@ static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
*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));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_ERROR, "cannot set parameters (%s)\n", snd_strerror(rc));
goto error;
}
@@ -130,51 +130,51 @@ static int dev_open(sound_t *sound)
rc_play = snd_pcm_open(&sound->phandle, sound->audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
rc_rec = snd_pcm_open(&sound->chandle, sound->audiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if (rc_play < 0 && rc_rec < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s'! (%s)\n", sound->audiodev, snd_strerror(rc_play));
PDEBUG(DSOUND, DEBUG_ERROR, "Run 'aplay -l' to get a list of available cards and devices.\n");
PDEBUG(DSOUND, DEBUG_ERROR, "Then use 'hw:<card>:<device>' for audio device.\n");
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s'! (%s)\n", sound->audiodev, snd_strerror(rc_play));
LOGP(DSOUND, LOGL_ERROR, "Run 'aplay -l' to get a list of available cards and devices.\n");
LOGP(DSOUND, LOGL_ERROR, "Then use 'hw:<card>:<device>' for audio device.\n");
return rc_play;
}
if (rc_play < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_play));
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_play));
return rc_play;
}
if (rc_rec < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_rec));
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_rec));
return rc_rec;
}
rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set playback hw params\n");
LOGP(DSOUND, LOGL_ERROR, "Failed to set playback hw params\n");
return rc;
}
if (sound->pchannels < sound->channels) {
PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
return rc;
}
PDEBUG(DSOUND, DEBUG_DEBUG, "Playback with %d channels.\n", sound->pchannels);
LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set capture hw params\n");
LOGP(DSOUND, LOGL_ERROR, "Failed to set capture hw params\n");
return rc;
}
if (sound->cchannels < sound->channels) {
PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
return -EIO;
}
PDEBUG(DSOUND, DEBUG_DEBUG, "Capture with %d channels.\n", sound->cchannels);
LOGP(DSOUND, LOGL_DEBUG, "Capture with %d channels.\n", sound->cchannels);
rc = snd_pcm_prepare(sound->phandle);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
LOGP(DSOUND, LOGL_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));
LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
}
@@ -196,13 +196,13 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
int rc;
if (channels < 1 || channels > 2) {
PDEBUG(DSOUND, DEBUG_ERROR, "Cannot use more than two channels with the same sound card!\n");
LOGP(DSOUND, LOGL_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");
LOGP(DSOUND, LOGL_ERROR, "Failed to alloc memory!\n");
return NULL;
}
@@ -234,7 +234,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
if ((env = getenv("KEEP_FRAMES"))) {
KEEP_FRAMES = atoi(env);
PDEBUG(DSOUND, DEBUG_NOTICE, "KEEP %d samples in RX buffer, to prevent corrupt read.\n", KEEP_FRAMES);
LOGP(DSOUND, LOGL_NOTICE, "KEEP %d samples in RX buffer, to prevent corrupt read.\n", KEEP_FRAMES);
}
return sound;
@@ -377,7 +377,7 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
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));
LOGP(DSOUND, LOGL_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE) {
dev_close(sound);
rc = dev_open(sound);
@@ -390,7 +390,7 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
}
if (rc != num)
PDEBUG(DSOUND, DEBUG_ERROR, "short write to audio interface, written %d bytes, got %d bytes\n", num, rc);
LOGP(DSOUND, LOGL_ERROR, "short write to audio interface, written %d bytes, got %d bytes\n", num, rc);
return rc;
}
@@ -423,7 +423,7 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf
if (rc < 0) {
if (errno == EAGAIN)
return 0;
PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
LOGP(DSOUND, LOGL_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
/* recover read */
if (rc == -EPIPE) {
dev_close(sound);
@@ -505,9 +505,9 @@ int sound_get_tosend(void *inst, int buffer_size)
rc = snd_pcm_delay(sound->phandle, &delay);
if (rc < 0) {
if (rc == -32)
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher buffer and enable real time scheduling\n");
LOGP(DSOUND, LOGL_ERROR, "Buffer underrun: Please use higher buffer and enable real time scheduling\n");
else
PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
LOGP(DSOUND, LOGL_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE) {
dev_close(sound);
rc = dev_open(sound);