static long alsa_mixer_max;
/** @brief Actually play sound via ALSA */
-static size_t alsa_play(void *buffer, size_t samples) {
+static size_t alsa_play(void *buffer, size_t samples, unsigned flags) {
+ /* If we're paused we just pretend. We rely on snd_pcm_writei() blocking so
+ * we have to fake up a sleep here. However it doesn't have to be all that
+ * accurate - in particular it's quite acceptable to greatly underestimate
+ * the required wait time. For 'lengthy' waits we do this by the blunt
+ * instrument of halving it. */
+ if(flags & UAUDIO_PAUSED) {
+ if(samples > 64)
+ samples /= 2;
+ const uint64_t ns = ((uint64_t)samples * 1000000000
+ / (uaudio_rate * uaudio_channels));
+ struct timespec ts[1];
+ ts->tv_sec = ns / 1000000000;
+ ts->tv_nsec = ns % 1000000000;
+ while(nanosleep(ts, ts) < 0 && errno == EINTR)
+ ;
+ return samples;
+ }
int err;
/* ALSA wants 'frames', where frame = several concurrently played samples */
const snd_pcm_uframes_t frames = samples / uaudio_channels;
switch(rc) {
case -EPIPE:
if((err = snd_pcm_prepare(alsa_pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
+ disorder_fatal(0, "error calling snd_pcm_prepare: %d", err);
return 0;
case -EAGAIN:
return 0;
default:
- fatal(0, "error calling snd_pcm_writei: %d", (int)rc);
+ disorder_fatal(0, "error calling snd_pcm_writei: %d", (int)rc);
}
}
return rc * uaudio_channels;
device,
SND_PCM_STREAM_PLAYBACK,
0)))
- fatal(0, "error from snd_pcm_open: %d", err);
+ disorder_fatal(0, "error from snd_pcm_open: %d", err);
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
if((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0)
- fatal(0, "error from snd_pcm_hw_params_any: %d", err);
+ disorder_fatal(0, "error from snd_pcm_hw_params_any: %d", err);
if((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
+ disorder_fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
int sample_format;
if(uaudio_bits == 16)
sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16;
sample_format = uaudio_signed ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
if((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams,
sample_format)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
+ disorder_fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
sample_format, err);
unsigned rate = uaudio_rate;
if((err = snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &rate, 0)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d",
+ disorder_fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d",
rate, err);
if((err = snd_pcm_hw_params_set_channels(alsa_pcm, hwparams,
uaudio_channels)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
+ disorder_fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
uaudio_channels, err);
if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0)
- fatal(0, "error calling snd_pcm_hw_params: %d", err);
+ disorder_fatal(0, "error calling snd_pcm_hw_params: %d", err);
}
-static void alsa_activate(void) {
- uaudio_thread_activate();
-}
-
-static void alsa_deactivate(void) {
- uaudio_thread_deactivate();
-}
-
static void alsa_start(uaudio_callback *callback,
void *userdata) {
if(uaudio_channels != 1 && uaudio_channels != 2)
- fatal(0, "asked for %d channels but only support 1 or 2",
+ disorder_fatal(0, "asked for %d channels but only support 1 or 2",
uaudio_channels);
if(uaudio_bits != 8 && uaudio_bits != 16)
- fatal(0, "asked for %d bits/channel but only support 8 or 16",
+ disorder_fatal(0, "asked for %d bits/channel but only support 8 or 16",
uaudio_bits);
alsa_open();
uaudio_thread_start(callback, userdata, alsa_play,
snd_mixer_selem_id_alloca(&id);
if((err = snd_mixer_open(&alsa_mixer_handle, 0)))
- fatal(0, "snd_mixer_open: %s", snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_open: %s", snd_strerror(err));
if((err = snd_mixer_attach(alsa_mixer_handle, device)))
- fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err));
if((err = snd_mixer_selem_register(alsa_mixer_handle,
0/*options*/, 0/*classp*/)))
- fatal(0, "snd_mixer_selem_register %s: %s",
+ disorder_fatal(0, "snd_mixer_selem_register %s: %s",
device, snd_strerror(err));
if((err = snd_mixer_load(alsa_mixer_handle)))
- fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err));
snd_mixer_selem_id_set_name(id, channel);
snd_mixer_selem_id_set_index(id, atoi(mixer));
if(!(alsa_mixer_elem = snd_mixer_find_selem(alsa_mixer_handle, id)))
- fatal(0, "device '%s' mixer control '%s,%s' does not exist",
- device, channel, mixer);
+ disorder_fatal(0, "device '%s' mixer control '%s,%s' does not exist",
+ device, channel, mixer);
if(!snd_mixer_selem_has_playback_volume(alsa_mixer_elem))
- fatal(0, "device '%s' mixer control '%s,%s' has no playback volume",
- device, channel, mixer);
+ disorder_fatal(0,
+ "device '%s' mixer control '%s,%s' has no playback volume",
+ device, channel, mixer);
if(snd_mixer_selem_is_playback_mono(alsa_mixer_elem)) {
alsa_mixer_left = alsa_mixer_right = SND_MIXER_SCHN_MONO;
} else {
alsa_mixer_left)
|| !snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
alsa_mixer_right))
- fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels",
- device, channel, mixer);
+ disorder_fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels",
+ device, channel, mixer);
snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
&alsa_mixer_min, &alsa_mixer_max);
alsa_mixer_left, &l))
|| (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
alsa_mixer_right, &r)))
- fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s",
+ snd_strerror(err));
*left = to_percent(l);
*right = to_percent(r);
}
if((err = snd_mixer_selem_set_playback_volume
(alsa_mixer_elem, alsa_mixer_left,
from_percent(*left > *right ? *left : *right))))
- fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s",
+ snd_strerror(err));
} else {
/* Stereo output */
if((err = snd_mixer_selem_set_playback_volume
(alsa_mixer_elem, alsa_mixer_left, from_percent(*left)))
|| (err = snd_mixer_selem_set_playback_volume
(alsa_mixer_elem, alsa_mixer_right, from_percent(*right))))
- fatal(0, "snd_mixer_selem_set_playback_volume: %s", snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s",
+ snd_strerror(err));
}
/* Read it back to see what we ended up at */
if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
alsa_mixer_left, &l))
|| (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
alsa_mixer_right, &r)))
- fatal(0, "snd_mixer_selem_get_playback_volume: %s", snd_strerror(err));
+ disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s",
+ snd_strerror(err));
*left = to_percent(l);
*right = to_percent(r);
}
.options = alsa_options,
.start = alsa_start,
.stop = alsa_stop,
- .activate = alsa_activate,
- .deactivate = alsa_deactivate,
+ .activate = uaudio_thread_activate,
+ .deactivate = uaudio_thread_deactivate,
.open_mixer = alsa_open_mixer,
.close_mixer = alsa_close_mixer,
.get_volume = alsa_get_volume,
~mdw / disorder / blobdiff