/*
* This file is part of DisOrder.
* Copyright (C) 2009 Richard Kettlewell
*
* 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 .
*/
/** @file lib/uaudio-alsa.c
* @brief Support for ALSA backend */
#include "common.h"
#if HAVE_ALSA_ASOUNDLIB_H
#include
#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, 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;
snd_pcm_sframes_t rc = snd_pcm_writei(alsa_pcm, buffer, frames);
if(rc < 0) {
switch(rc) {
case -EPIPE:
if((err = snd_pcm_prepare(alsa_pcm)))
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);
}
}
return rc * uaudio_channels;
}
/** @brief Open the ALSA sound device */
static void alsa_open(void) {
const char *device = uaudio_get("device", "default");
int err;
if((err = snd_pcm_open(&alsa_pcm,
device,
SND_PCM_STREAM_PLAYBACK,
0)))
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);
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);
int sample_format;
if(uaudio_bits == 16)
sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16;
else
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",
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",
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",
uaudio_channels, err);
if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0)
fatal(0, "error calling snd_pcm_hw_params: %d", err);
}
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",
uaudio_channels);
if(uaudio_bits != 8 && uaudio_bits != 16)
fatal(0, "asked for %d bits/channel but only support 8 or 16",
uaudio_bits);
alsa_open();
uaudio_thread_start(callback, userdata, alsa_play,
32 / uaudio_sample_size,
4096 / uaudio_sample_size,
0);
}
static void alsa_stop(void) {
uaudio_thread_stop();
snd_pcm_close(alsa_pcm);
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 = 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
/*
Local Variables:
c-basic-offset:2
comment-column:40
fill-column:79
indent-tabs-mode:nil
End:
*/