| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2009, 2013 Richard Kettlewell |
| 4 | * |
| 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. |
| 9 | * |
| 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. |
| 14 | * |
| 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/>. |
| 17 | */ |
| 18 | /** @file lib/uaudio-alsa.c |
| 19 | * @brief Support for ALSA backend */ |
| 20 | #include "common.h" |
| 21 | |
| 22 | #if HAVE_ALSA_ASOUNDLIB_H |
| 23 | |
| 24 | #include <alsa/asoundlib.h> |
| 25 | |
| 26 | #include "mem.h" |
| 27 | #include "log.h" |
| 28 | #include "uaudio.h" |
| 29 | #include "configuration.h" |
| 30 | |
| 31 | /** @brief The current PCM handle */ |
| 32 | static snd_pcm_t *alsa_pcm; |
| 33 | |
| 34 | static const char *const alsa_options[] = { |
| 35 | "device", |
| 36 | "mixer-control", |
| 37 | "mixer-channel", |
| 38 | NULL |
| 39 | }; |
| 40 | |
| 41 | /** @brief Mixer handle */ |
| 42 | snd_mixer_t *alsa_mixer_handle; |
| 43 | |
| 44 | /** @brief Mixer control */ |
| 45 | static snd_mixer_elem_t *alsa_mixer_elem; |
| 46 | |
| 47 | /** @brief Left channel */ |
| 48 | static snd_mixer_selem_channel_id_t alsa_mixer_left; |
| 49 | |
| 50 | /** @brief Right channel */ |
| 51 | static snd_mixer_selem_channel_id_t alsa_mixer_right; |
| 52 | |
| 53 | /** @brief Minimum level */ |
| 54 | static long alsa_mixer_min; |
| 55 | |
| 56 | /** @brief Maximum level */ |
| 57 | static long alsa_mixer_max; |
| 58 | |
| 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) { |
| 67 | if(samples > 64) |
| 68 | samples /= 2; |
| 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) |
| 75 | ; |
| 76 | return samples; |
| 77 | } |
| 78 | int err; |
| 79 | /* ALSA wants 'frames', where frame = several concurrently played samples */ |
| 80 | const snd_pcm_uframes_t frames = samples / uaudio_channels; |
| 81 | |
| 82 | snd_pcm_sframes_t rc = snd_pcm_writei(alsa_pcm, buffer, frames); |
| 83 | if(rc < 0) { |
| 84 | switch(rc) { |
| 85 | case -EPIPE: |
| 86 | if((err = snd_pcm_prepare(alsa_pcm))) |
| 87 | disorder_fatal(0, "error calling snd_pcm_prepare: %d", err); |
| 88 | return 0; |
| 89 | case -EAGAIN: |
| 90 | return 0; |
| 91 | default: |
| 92 | disorder_fatal(0, "error calling snd_pcm_writei: %d", (int)rc); |
| 93 | } |
| 94 | } |
| 95 | return rc * uaudio_channels; |
| 96 | } |
| 97 | |
| 98 | /** @brief Open the ALSA sound device */ |
| 99 | static void alsa_open(void) { |
| 100 | const char *device = uaudio_get("device", "default"); |
| 101 | int err; |
| 102 | |
| 103 | if((err = snd_pcm_open(&alsa_pcm, |
| 104 | device, |
| 105 | SND_PCM_STREAM_PLAYBACK, |
| 106 | 0))) |
| 107 | disorder_fatal(0, "error from snd_pcm_open: %d", err); |
| 108 | /* Hardware parameters */ |
| 109 | snd_pcm_hw_params_t *hwparams; |
| 110 | snd_pcm_hw_params_alloca(&hwparams); |
| 111 | if((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0) |
| 112 | disorder_fatal(0, "error from snd_pcm_hw_params_any: %d", err); |
| 113 | if((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams, |
| 114 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) |
| 115 | disorder_fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); |
| 116 | int sample_format; |
| 117 | if(uaudio_bits == 16) |
| 118 | sample_format = uaudio_signed ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U16; |
| 119 | else |
| 120 | sample_format = uaudio_signed ? SND_PCM_FORMAT_S8 : SND_PCM_FORMAT_U8; |
| 121 | if((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams, |
| 122 | sample_format)) < 0) |
| 123 | disorder_fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", |
| 124 | sample_format, err); |
| 125 | unsigned rate = uaudio_rate; |
| 126 | if((err = snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &rate, 0)) < 0) |
| 127 | disorder_fatal(0, "error from snd_pcm_hw_params_set_rate_near (%d): %d", |
| 128 | rate, err); |
| 129 | if((err = snd_pcm_hw_params_set_channels(alsa_pcm, hwparams, |
| 130 | uaudio_channels)) < 0) |
| 131 | disorder_fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", |
| 132 | uaudio_channels, err); |
| 133 | if((err = snd_pcm_hw_params(alsa_pcm, hwparams)) < 0) |
| 134 | disorder_fatal(0, "error calling snd_pcm_hw_params: %d", err); |
| 135 | /* Software parameters */ |
| 136 | snd_pcm_sw_params_t *swparams; |
| 137 | snd_pcm_sw_params_alloca(&swparams); |
| 138 | if((err = snd_pcm_sw_params_current(alsa_pcm, swparams)) < 0) |
| 139 | disorder_fatal(-err, "error calling snd_pcm_sw_params_current"); |
| 140 | /* Bump the start threshold a bit since Pulseaudio sulks with the defaults */ |
| 141 | if((err = snd_pcm_sw_params_set_start_threshold(alsa_pcm, swparams, 1024)) < 0) |
| 142 | disorder_fatal(-err, "error calling snd_pcm_sw_params_set_start_threshold"); |
| 143 | if((err = snd_pcm_sw_params(alsa_pcm, swparams)) < 0) |
| 144 | disorder_fatal(-err, "error calling snd_pcm_sw_params"); |
| 145 | } |
| 146 | |
| 147 | static void alsa_start(uaudio_callback *callback, |
| 148 | void *userdata) { |
| 149 | if(uaudio_channels != 1 && uaudio_channels != 2) |
| 150 | disorder_fatal(0, "asked for %d channels but only support 1 or 2", |
| 151 | uaudio_channels); |
| 152 | if(uaudio_bits != 8 && uaudio_bits != 16) |
| 153 | disorder_fatal(0, "asked for %d bits/channel but only support 8 or 16", |
| 154 | uaudio_bits); |
| 155 | alsa_open(); |
| 156 | uaudio_thread_start(callback, userdata, alsa_play, |
| 157 | 32 / uaudio_sample_size, |
| 158 | 4096 / uaudio_sample_size, |
| 159 | 0); |
| 160 | } |
| 161 | |
| 162 | static void alsa_stop(void) { |
| 163 | uaudio_thread_stop(); |
| 164 | snd_pcm_close(alsa_pcm); |
| 165 | alsa_pcm = 0; |
| 166 | } |
| 167 | |
| 168 | /** @brief Convert a level to a percentage */ |
| 169 | static int to_percent(long n) { |
| 170 | return (n - alsa_mixer_min) * 100 / (alsa_mixer_max - alsa_mixer_min); |
| 171 | } |
| 172 | |
| 173 | /** @brief Convert a percentage to a level */ |
| 174 | static int from_percent(int n) { |
| 175 | return alsa_mixer_min + n * (alsa_mixer_max - alsa_mixer_min) / 100; |
| 176 | } |
| 177 | |
| 178 | static void alsa_open_mixer(void) { |
| 179 | int err; |
| 180 | snd_mixer_selem_id_t *id; |
| 181 | const char *device = uaudio_get("device", "default"); |
| 182 | const char *mixer = uaudio_get("mixer-control", "0"); |
| 183 | const char *channel = uaudio_get("mixer-channel", "Master"); |
| 184 | |
| 185 | snd_mixer_selem_id_alloca(&id); |
| 186 | if((err = snd_mixer_open(&alsa_mixer_handle, 0))) |
| 187 | disorder_fatal(0, "snd_mixer_open: %s", snd_strerror(err)); |
| 188 | if((err = snd_mixer_attach(alsa_mixer_handle, device))) |
| 189 | disorder_fatal(0, "snd_mixer_attach %s: %s", device, snd_strerror(err)); |
| 190 | if((err = snd_mixer_selem_register(alsa_mixer_handle, |
| 191 | 0/*options*/, 0/*classp*/))) |
| 192 | disorder_fatal(0, "snd_mixer_selem_register %s: %s", |
| 193 | device, snd_strerror(err)); |
| 194 | if((err = snd_mixer_load(alsa_mixer_handle))) |
| 195 | disorder_fatal(0, "snd_mixer_load %s: %s", device, snd_strerror(err)); |
| 196 | snd_mixer_selem_id_set_name(id, channel); |
| 197 | snd_mixer_selem_id_set_index(id, atoi(mixer)); |
| 198 | if(!(alsa_mixer_elem = snd_mixer_find_selem(alsa_mixer_handle, id))) |
| 199 | disorder_fatal(0, "device '%s' mixer control '%s,%s' does not exist", |
| 200 | device, channel, mixer); |
| 201 | if(!snd_mixer_selem_has_playback_volume(alsa_mixer_elem)) |
| 202 | disorder_fatal(0, |
| 203 | "device '%s' mixer control '%s,%s' has no playback volume", |
| 204 | device, channel, mixer); |
| 205 | if(snd_mixer_selem_is_playback_mono(alsa_mixer_elem)) { |
| 206 | alsa_mixer_left = alsa_mixer_right = SND_MIXER_SCHN_MONO; |
| 207 | } else { |
| 208 | alsa_mixer_left = SND_MIXER_SCHN_FRONT_LEFT; |
| 209 | alsa_mixer_right = SND_MIXER_SCHN_FRONT_RIGHT; |
| 210 | } |
| 211 | if(!snd_mixer_selem_has_playback_channel(alsa_mixer_elem, |
| 212 | alsa_mixer_left) |
| 213 | || !snd_mixer_selem_has_playback_channel(alsa_mixer_elem, |
| 214 | alsa_mixer_right)) |
| 215 | disorder_fatal(0, "device '%s' mixer control '%s,%s' lacks required playback channels", |
| 216 | device, channel, mixer); |
| 217 | snd_mixer_selem_get_playback_volume_range(alsa_mixer_elem, |
| 218 | &alsa_mixer_min, &alsa_mixer_max); |
| 219 | |
| 220 | } |
| 221 | |
| 222 | static void alsa_close_mixer(void) { |
| 223 | /* TODO alsa_mixer_elem */ |
| 224 | if(alsa_mixer_handle) |
| 225 | snd_mixer_close(alsa_mixer_handle); |
| 226 | } |
| 227 | |
| 228 | static void alsa_get_volume(int *left, int *right) { |
| 229 | long l, r; |
| 230 | int err; |
| 231 | |
| 232 | if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem, |
| 233 | alsa_mixer_left, &l)) |
| 234 | || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem, |
| 235 | alsa_mixer_right, &r))) |
| 236 | disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s", |
| 237 | snd_strerror(err)); |
| 238 | *left = to_percent(l); |
| 239 | *right = to_percent(r); |
| 240 | } |
| 241 | |
| 242 | static void alsa_set_volume(int *left, int *right) { |
| 243 | long l, r; |
| 244 | int err; |
| 245 | |
| 246 | /* Set the volume */ |
| 247 | if(alsa_mixer_left == alsa_mixer_right) { |
| 248 | /* Mono output - just use the loudest */ |
| 249 | if((err = snd_mixer_selem_set_playback_volume |
| 250 | (alsa_mixer_elem, alsa_mixer_left, |
| 251 | from_percent(*left > *right ? *left : *right)))) |
| 252 | disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s", |
| 253 | snd_strerror(err)); |
| 254 | } else { |
| 255 | /* Stereo output */ |
| 256 | if((err = snd_mixer_selem_set_playback_volume |
| 257 | (alsa_mixer_elem, alsa_mixer_left, from_percent(*left))) |
| 258 | || (err = snd_mixer_selem_set_playback_volume |
| 259 | (alsa_mixer_elem, alsa_mixer_right, from_percent(*right)))) |
| 260 | disorder_fatal(0, "snd_mixer_selem_set_playback_volume: %s", |
| 261 | snd_strerror(err)); |
| 262 | } |
| 263 | /* Read it back to see what we ended up at */ |
| 264 | if((err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem, |
| 265 | alsa_mixer_left, &l)) |
| 266 | || (err = snd_mixer_selem_get_playback_volume(alsa_mixer_elem, |
| 267 | alsa_mixer_right, &r))) |
| 268 | disorder_fatal(0, "snd_mixer_selem_get_playback_volume: %s", |
| 269 | snd_strerror(err)); |
| 270 | *left = to_percent(l); |
| 271 | *right = to_percent(r); |
| 272 | } |
| 273 | |
| 274 | static void alsa_configure(void) { |
| 275 | uaudio_set("device", config->device); |
| 276 | uaudio_set("mixer-control", config->mixer); |
| 277 | uaudio_set("mixer-channel", config->channel); |
| 278 | } |
| 279 | |
| 280 | const struct uaudio uaudio_alsa = { |
| 281 | .name = "alsa", |
| 282 | .options = alsa_options, |
| 283 | .start = alsa_start, |
| 284 | .stop = alsa_stop, |
| 285 | .activate = uaudio_thread_activate, |
| 286 | .deactivate = uaudio_thread_deactivate, |
| 287 | .open_mixer = alsa_open_mixer, |
| 288 | .close_mixer = alsa_close_mixer, |
| 289 | .get_volume = alsa_get_volume, |
| 290 | .set_volume = alsa_set_volume, |
| 291 | .configure = alsa_configure, |
| 292 | .flags = UAUDIO_API_CLIENT | UAUDIO_API_SERVER, |
| 293 | }; |
| 294 | |
| 295 | #endif |
| 296 | |
| 297 | /* |
| 298 | Local Variables: |
| 299 | c-basic-offset:2 |
| 300 | comment-column:40 |
| 301 | fill-column:79 |
| 302 | indent-tabs-mode:nil |
| 303 | End: |
| 304 | */ |