#include "mem.h"
#include "log.h"
#include "uaudio.h"
+#include "configuration.h"
/** @brief The current PCM handle */
static snd_pcm_t *alsa_pcm;
static const char *const alsa_options[] = {
"device",
+ "mixer-control",
+ "mixer-channel",
NULL
};
+/** @brief Mixer handle */
+snd_mixer_t *alsa_mixer_handle;
+
+/** @brief Mixer control */
+static snd_mixer_elem_t *alsa_mixer_elem;
+
+/** @brief Left channel */
+static snd_mixer_selem_channel_id_t alsa_mixer_left;
+
+/** @brief Right channel */
+static snd_mixer_selem_channel_id_t alsa_mixer_right;
+
+/** @brief Minimum level */
+static long alsa_mixer_min;
+
+/** @brief Maximum level */
+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;
/** @brief Open the ALSA sound device */
static void alsa_open(void) {
- const char *device = uaudio_get("device");
+ const char *device = uaudio_get("device", "default");
int err;
- if(!device || !*device)
- device = "default";
if((err = snd_pcm_open(&alsa_pcm,
device,
SND_PCM_STREAM_PLAYBACK,
}
-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)
alsa_open();
uaudio_thread_start(callback, userdata, alsa_play,
32 / uaudio_sample_size,
- 4096 / uaudio_sample_size);
+ 4096 / uaudio_sample_size,
+ 0);
}
static void alsa_stop(void) {
alsa_pcm = 0;
}
+/** @brief Convert a level to a percentage */
+static int to_percent(long n) {
+ return (n - alsa_mixer_min) * 100 / (alsa_mixer_max - alsa_mixer_min);
+}
+
+/** @brief Convert a percentage to a level */
+static int from_percent(int n) {
+ return alsa_mixer_min + n * (alsa_mixer_max - alsa_mixer_min) / 100;
+}
+
+static void alsa_open_mixer(void) {
+ int err;
+ snd_mixer_selem_id_t *id;
+ const char *device = uaudio_get("device", "default");
+ const char *mixer = uaudio_get("mixer-control", "0");
+ const char *channel = uaudio_get("mixer-channel", "PCM");
+
+ snd_mixer_selem_id_alloca(&id);
+ if((err = snd_mixer_open(&alsa_mixer_handle, 0)))
+ 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));
+ if((err = snd_mixer_selem_register(alsa_mixer_handle,
+ 0/*options*/, 0/*classp*/)))
+ 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));
+ 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);
+ 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);
+ 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_SCHN_FRONT_LEFT;
+ alsa_mixer_right = SND_MIXER_SCHN_FRONT_RIGHT;
+ }
+ if(!snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
+ 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);
+ snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
+ &alsa_mixer_min, &alsa_mixer_max);
+
+}
+
+static void alsa_close_mixer(void) {
+ /* TODO alsa_mixer_elem */
+ if(alsa_mixer_handle)
+ snd_mixer_close(alsa_mixer_handle);
+}
+
+static void alsa_get_volume(int *left, int *right) {
+ long l, r;
+ int err;
+
+ 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));
+ *left = to_percent(l);
+ *right = to_percent(r);
+}
+
+static void alsa_set_volume(int *left, int *right) {
+ long l, r;
+ int err;
+
+ /* Set the volume */
+ if(alsa_mixer_left == alsa_mixer_right) {
+ /* Mono output - just use the loudest */
+ 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));
+ } 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));
+ }
+ /* 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));
+ *left = to_percent(l);
+ *right = to_percent(r);
+}
+
+static void alsa_configure(void) {
+ uaudio_set("device", config->device);
+ uaudio_set("mixer-control", config->mixer);
+ uaudio_set("mixer-channel", config->channel);
+}
+
const struct uaudio uaudio_alsa = {
.name = "alsa",
.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,
+ .set_volume = alsa_set_volume,
+ .configure = alsa_configure
};
#endif
~mdw / disorder / blobdiff