2 * This file is part of DisOrder.
3 * Copyright (C) 2009 Richard Kettlewell
5 * This program is free software: you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation, either version 3 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program. If not, see <http://www.gnu.org/licenses/>.
18 /** @file lib/uaudio-alsa.c
19 * @brief Support for ALSA backend */
22 #if HAVE_ALSA_ASOUNDLIB_H
24 #include <alsa/asoundlib.h>
29 #include "configuration.h"
31 /** @brief The current PCM handle */
32 static snd_pcm_t *alsa_pcm;
34 static const char *const alsa_options[] = {
41 /** @brief Mixer handle */
42 snd_mixer_t *alsa_mixer_handle;
44 /** @brief Mixer control */
45 static snd_mixer_elem_t *alsa_mixer_elem;
47 /** @brief Left channel */
48 static snd_mixer_selem_channel_id_t alsa_mixer_left;
50 /** @brief Right channel */
51 static snd_mixer_selem_channel_id_t alsa_mixer_right;
53 /** @brief Minimum level */
54 static long alsa_mixer_min;
56 /** @brief Maximum level */
57 static long alsa_mixer_max;
59 /** @brief Actually play sound via ALSA */
60 static size_t alsa_play(void *buffer, size_t samples, unsigned flags) {
61 /* If we're paused we just pretend. We rely on snd_pcm_writei() blocking so
62 * we have to fake up a sleep here. However it doesn't have to be all that
63 * accurate - in particular it's quite acceptable to greatly underestimate
64 * the required wait time. For 'lengthy' waits we do this by the blunt
65 * instrument of halving it. */
66 if(flags & UAUDIO_PAUSED) {
69 const uint64_t ns = ((uint64_t)samples * 1000000000
70 / (uaudio_rate * uaudio_channels));
71 struct timespec ts[1];
72 ts->tv_sec = ns / 1000000000;
73 ts->tv_nsec = ns % 1000000000;
74 while(nanosleep(ts, ts) < 0 && errno == EINTR)
79 /* ALSA wants 'frames', where frame = several concurrently played samples */
80 const snd_pcm_uframes_t frames = samples / uaudio_channels;
82 snd_pcm_sframes_t rc = snd_pcm_writei(alsa_pcm, buffer, frames);
86 if((err = snd_pcm_prepare(alsa_pcm)))
87 disorder_fatal(0, "error calling snd_pcm_prepare: %d", err);
92 disorder_fatal(0, "error calling snd_pcm_writei: %d", (int)rc);
95 return rc * uaudio_channels;
98 /** @brief Open the ALSA sound device */
99 static void alsa_open(void) {
100 const char *device = uaudio_get("device", "default");
103 if((err = snd_pcm_open(&alsa_pcm,
105 SND_PCM_STREAM_PLAYBACK,
107 disorder_fatal(0, "error from snd_pcm_open: %d", err);
108 snd_pcm_hw_params_t *hwparams;
109 snd_pcm_hw_params_alloca(&hwparams);
110 if((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0)
111 disorder_fatal(0, "error from snd_pcm_hw_params_any: %d", err);
112 if((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams,
113 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
114 disorder_fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
116 if(uaudio_bits == 16)
117 sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16;
119 sample_format = uaudio_signed ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8;
120 if((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams,
122 disorder_fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
124 unsigned rate = uaudio_rate;
125 if((err = snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &rate, 0)) < 0)
126 disorder_fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d",
128 if((err = snd_pcm_hw_params_set_channels(alsa_pcm, hwparams,
129 uaudio_channels)) < 0)
130 disorder_fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
131 uaudio_channels, err);
132 if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0)
133 disorder_fatal(0, "error calling snd_pcm_hw_params: %d", err);
137 static void alsa_start(uaudio_callback *callback,
139 if(uaudio_channels != 1 && uaudio_channels != 2)
140 disorder_fatal(0, "asked for %d channels but only support 1 or 2",
142 if(uaudio_bits != 8 && uaudio_bits != 16)
143 disorder_fatal(0, "asked for %d bits/channel but only support 8 or 16",
146 uaudio_thread_start(callback, userdata, alsa_play,
147 32 / uaudio_sample_size,
148 4096 / uaudio_sample_size,
152 static void alsa_stop(void) {
153 uaudio_thread_stop();
154 snd_pcm_close(alsa_pcm);
158 /** @brief Convert a level to a percentage */
159 static int to_percent(long n) {
160 return (n - alsa_mixer_min) * 100 / (alsa_mixer_max - alsa_mixer_min);
163 /** @brief Convert a percentage to a level */
164 static int from_percent(int n) {
165 return alsa_mixer_min + n * (alsa_mixer_max - alsa_mixer_min) / 100;
168 static void alsa_open_mixer(void) {
170 snd_mixer_selem_id_t *id;
171 const char *device = uaudio_get("device", "default");
172 const char *mixer = uaudio_get("mixer-control", "0");
173 const char *channel = uaudio_get("mixer-channel", "PCM");
175 snd_mixer_selem_id_alloca(&id);
176 if((err = snd_mixer_open(&alsa_mixer_handle, 0)))
177 disorder_fatal(0, "snd_mixer_open: %s", snd_strerror(err));
178 if((err = snd_mixer_attach(alsa_mixer_handle, device)))
179 disorder_fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err));
180 if((err = snd_mixer_selem_register(alsa_mixer_handle,
181 0/*options*/, 0/*classp*/)))
182 disorder_fatal(0, "snd_mixer_selem_register %s: %s",
183 device, snd_strerror(err));
184 if((err = snd_mixer_load(alsa_mixer_handle)))
185 disorder_fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err));
186 snd_mixer_selem_id_set_name(id, channel);
187 snd_mixer_selem_id_set_index(id, atoi(mixer));
188 if(!(alsa_mixer_elem = snd_mixer_find_selem(alsa_mixer_handle, id)))
189 disorder_fatal(0, "device '%s' mixer control '%s,%s' does not exist",
190 device, channel, mixer);
191 if(!snd_mixer_selem_has_playback_volume(alsa_mixer_elem))
193 "device '%s' mixer control '%s,%s' has no playback volume",
194 device, channel, mixer);
195 if(snd_mixer_selem_is_playback_mono(alsa_mixer_elem)) {
196 alsa_mixer_left = alsa_mixer_right = SND_MIXER_SCHN_MONO;
198 alsa_mixer_left = SND_MIXER_SCHN_FRONT_LEFT;
199 alsa_mixer_right = SND_MIXER_SCHN_FRONT_RIGHT;
201 if(!snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
203 || !snd_mixer_selem_has_playback_channel(alsa_mixer_elem,
205 disorder_fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels",
206 device, channel, mixer);
207 snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem,
208 &alsa_mixer_min, &alsa_mixer_max);
212 static void alsa_close_mixer(void) {
213 /* TODO alsa_mixer_elem */
214 if(alsa_mixer_handle)
215 snd_mixer_close(alsa_mixer_handle);
218 static void alsa_get_volume(int *left, int *right) {
222 if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
223 alsa_mixer_left, &l))
224 || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
225 alsa_mixer_right, &r)))
226 disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s",
228 *left = to_percent(l);
229 *right = to_percent(r);
232 static void alsa_set_volume(int *left, int *right) {
237 if(alsa_mixer_left == alsa_mixer_right) {
238 /* Mono output - just use the loudest */
239 if((err = snd_mixer_selem_set_playback_volume
240 (alsa_mixer_elem, alsa_mixer_left,
241 from_percent(*left > *right ? *left : *right))))
242 disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s",
246 if((err = snd_mixer_selem_set_playback_volume
247 (alsa_mixer_elem, alsa_mixer_left, from_percent(*left)))
248 || (err = snd_mixer_selem_set_playback_volume
249 (alsa_mixer_elem, alsa_mixer_right, from_percent(*right))))
250 disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s",
253 /* Read it back to see what we ended up at */
254 if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
255 alsa_mixer_left, &l))
256 || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem,
257 alsa_mixer_right, &r)))
258 disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s",
260 *left = to_percent(l);
261 *right = to_percent(r);
264 static void alsa_configure(void) {
265 uaudio_set("device", config->device);
266 uaudio_set("mixer-control", config->mixer);
267 uaudio_set("mixer-channel", config->channel);
270 const struct uaudio uaudio_alsa = {
272 .options = alsa_options,
275 .activate = uaudio_thread_activate,
276 .deactivate = uaudio_thread_deactivate,
277 .open_mixer = alsa_open_mixer,
278 .close_mixer = alsa_close_mixer,
279 .get_volume = alsa_get_volume,
280 .set_volume = alsa_set_volume,
281 .configure = alsa_configure