Mercurial > mplayer.hg
annotate libao2/ao_alsa.c @ 29392:cd3ec59296a0
Use correct ALSA sample format for 24-bit samples packed in three bytes.
author | cladisch |
---|---|
date | Tue, 21 Jul 2009 08:20:22 +0000 |
parents | 9ccfc57f0eff |
children | f01023c524c3 |
rev | line source |
---|---|
12465 | 1 /* |
28343 | 2 * ALSA 0.9.x-1.x audio output driver |
3 * | |
4 * Copyright (C) 2004 Alex Beregszaszi | |
5 * | |
6 * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de> | |
7 * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org> | |
8 * 08/22/2002 iec958-init rewritten and merged with common init, zsolt | |
9 * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka | |
10 * 04/25/2004 printfs converted to mp_msg, Zsolt. | |
11 * | |
12 * This file is part of MPlayer. | |
13 * | |
14 * MPlayer is free software; you can redistribute it and/or modify | |
15 * it under the terms of the GNU General Public License as published by | |
16 * the Free Software Foundation; either version 2 of the License, or | |
17 * (at your option) any later version. | |
18 * | |
19 * MPlayer is distributed in the hope that it will be useful, | |
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
22 * GNU General Public License for more details. | |
23 * | |
24 * You should have received a copy of the GNU General Public License along | |
25 * with MPlayer; if not, write to the Free Software Foundation, Inc., | |
26 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. | |
27 */ | |
12465 | 28 |
29 #include <errno.h> | |
30 #include <sys/time.h> | |
31 #include <stdlib.h> | |
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32 #include <stdarg.h> |
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33 #include <ctype.h> |
12465 | 34 #include <math.h> |
35 #include <string.h> | |
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36 #include <alloca.h> |
12465 | 37 |
14123 | 38 #include "config.h" |
14328 | 39 #include "subopt-helper.h" |
14123 | 40 #include "mixer.h" |
41 #include "mp_msg.h" | |
20764 | 42 #include "help_mp.h" |
12465 | 43 |
44 #define ALSA_PCM_NEW_HW_PARAMS_API | |
45 #define ALSA_PCM_NEW_SW_PARAMS_API | |
46 | |
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47 #ifdef HAVE_SYS_ASOUNDLIB_H |
12465 | 48 #include <sys/asoundlib.h> |
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49 #elif defined(HAVE_ALSA_ASOUNDLIB_H) |
12465 | 50 #include <alsa/asoundlib.h> |
51 #else | |
52 #error "asoundlib.h is not in sys/ or alsa/ - please bugreport" | |
53 #endif | |
54 | |
55 | |
56 #include "audio_out.h" | |
57 #include "audio_out_internal.h" | |
14245 | 58 #include "libaf/af_format.h" |
12465 | 59 |
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60 static const ao_info_t info = |
12465 | 61 { |
62 "ALSA-0.9.x-1.x audio output", | |
63 "alsa", | |
64 "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>", | |
65 "under developement" | |
66 }; | |
67 | |
68 LIBAO_EXTERN(alsa) | |
69 | |
70 static snd_pcm_t *alsa_handler; | |
71 static snd_pcm_format_t alsa_format; | |
72 static snd_pcm_hw_params_t *alsa_hwparams; | |
73 static snd_pcm_sw_params_t *alsa_swparams; | |
74 | |
75 /* 16 sets buffersize to 16 * chunksize is as default 1024 | |
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76 * which seems to be good avarge for most situations |
12465 | 77 * so buffersize is 16384 frames by default */ |
78 static int alsa_fragcount = 16; | |
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79 static snd_pcm_uframes_t chunk_size = 1024; |
12465 | 80 |
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81 static size_t bytes_per_sample; |
12465 | 82 |
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83 static int ao_noblock = 0; |
12465 | 84 |
85 static int open_mode; | |
86 static int alsa_can_pause = 0; | |
87 | |
12747 | 88 #define ALSA_DEVICE_SIZE 256 |
12465 | 89 |
90 #undef BUFFERTIME | |
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91 #define SET_CHUNKSIZE |
12465 | 92 |
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93 static void alsa_error_handler(const char *file, int line, const char *function, |
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94 int err, const char *format, ...) |
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95 { |
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96 char tmp[0xc00]; |
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97 va_list va; |
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98 |
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99 va_start(va, format); |
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100 vsnprintf(tmp, sizeof tmp, format, va); |
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101 va_end(va); |
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102 tmp[sizeof tmp - 1] = '\0'; |
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103 |
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104 if (err) |
20764 | 105 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n", |
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106 file, line, function, tmp, snd_strerror(err)); |
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107 else |
20764 | 108 mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n", |
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109 file, line, function, tmp); |
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110 } |
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111 |
12465 | 112 /* to set/get/query special features/parameters */ |
113 static int control(int cmd, void *arg) | |
114 { | |
115 switch(cmd) { | |
116 case AOCONTROL_QUERY_FORMAT: | |
117 return CONTROL_TRUE; | |
118 case AOCONTROL_GET_VOLUME: | |
119 case AOCONTROL_SET_VOLUME: | |
120 { | |
121 ao_control_vol_t *vol = (ao_control_vol_t *)arg; | |
122 | |
123 int err; | |
124 snd_mixer_t *handle; | |
125 snd_mixer_elem_t *elem; | |
126 snd_mixer_selem_id_t *sid; | |
127 | |
12747 | 128 static char *mix_name = "PCM"; |
129 static char *card = "default"; | |
13434 | 130 static int mix_index = 0; |
12465 | 131 |
132 long pmin, pmax; | |
133 long get_vol, set_vol; | |
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134 float f_multi; |
12465 | 135 |
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136 if(ao_data.format == AF_FORMAT_AC3) |
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137 return CONTROL_TRUE; |
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138 |
13434 | 139 if(mixer_channel) { |
140 char *test_mix_index; | |
141 | |
142 mix_name = strdup(mixer_channel); | |
17097 | 143 if ((test_mix_index = strchr(mix_name, ','))){ |
13434 | 144 *test_mix_index = 0; |
145 test_mix_index++; | |
146 mix_index = strtol(test_mix_index, &test_mix_index, 0); | |
147 | |
148 if (*test_mix_index){ | |
149 mp_msg(MSGT_AO,MSGL_ERR, | |
20764 | 150 MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero); |
13434 | 151 mix_index = 0 ; |
152 } | |
153 } | |
154 } | |
12747 | 155 if(mixer_device) card = mixer_device; |
12465 | 156 |
157 //allocate simple id | |
158 snd_mixer_selem_id_alloca(&sid); | |
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159 |
12465 | 160 //sets simple-mixer index and name |
13434 | 161 snd_mixer_selem_id_set_index(sid, mix_index); |
12465 | 162 snd_mixer_selem_id_set_name(sid, mix_name); |
163 | |
13434 | 164 if (mixer_channel) { |
165 free(mix_name); | |
166 mix_name = NULL; | |
167 } | |
168 | |
12465 | 169 if ((err = snd_mixer_open(&handle, 0)) < 0) { |
20764 | 170 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err)); |
12465 | 171 return CONTROL_ERROR; |
172 } | |
173 | |
174 if ((err = snd_mixer_attach(handle, card)) < 0) { | |
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175 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError, |
12465 | 176 card, snd_strerror(err)); |
177 snd_mixer_close(handle); | |
178 return CONTROL_ERROR; | |
179 } | |
180 | |
181 if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) { | |
20764 | 182 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err)); |
12465 | 183 snd_mixer_close(handle); |
184 return CONTROL_ERROR; | |
185 } | |
186 err = snd_mixer_load(handle); | |
187 if (err < 0) { | |
20764 | 188 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err)); |
12465 | 189 snd_mixer_close(handle); |
190 return CONTROL_ERROR; | |
191 } | |
192 | |
193 elem = snd_mixer_find_selem(handle, sid); | |
194 if (!elem) { | |
20764 | 195 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl, |
12465 | 196 snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); |
197 snd_mixer_close(handle); | |
198 return CONTROL_ERROR; | |
199 } | |
200 | |
201 snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax); | |
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202 f_multi = (100 / (float)(pmax - pmin)); |
12465 | 203 |
204 if (cmd == AOCONTROL_SET_VOLUME) { | |
205 | |
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206 set_vol = vol->left / f_multi + pmin + 0.5; |
12465 | 207 |
208 //setting channels | |
27706 | 209 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) { |
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210 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel, |
12465 | 211 snd_strerror(err)); |
212 return CONTROL_ERROR; | |
213 } | |
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214 mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol); |
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215 |
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216 set_vol = vol->right / f_multi + pmin + 0.5; |
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217 |
27706 | 218 if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) { |
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219 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel, |
12465 | 220 snd_strerror(err)); |
221 return CONTROL_ERROR; | |
222 } | |
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223 mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", |
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224 set_vol, pmin, pmax, f_multi); |
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225 |
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226 if (snd_mixer_selem_has_playback_switch(elem)) { |
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227 int lmute = (vol->left == 0.0); |
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228 int rmute = (vol->right == 0.0); |
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229 if (snd_mixer_selem_has_playback_switch_joined(elem)) { |
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230 lmute = rmute = lmute && rmute; |
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231 } else { |
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232 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute); |
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233 } |
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234 snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute); |
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235 } |
12465 | 236 } |
237 else { | |
27706 | 238 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol); |
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239 vol->left = (get_vol - pmin) * f_multi; |
27706 | 240 snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol); |
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241 vol->right = (get_vol - pmin) * f_multi; |
12465 | 242 |
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243 mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right); |
12465 | 244 } |
245 snd_mixer_close(handle); | |
246 return CONTROL_OK; | |
247 } | |
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248 |
12465 | 249 } //end switch |
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250 return CONTROL_UNKNOWN; |
12465 | 251 } |
252 | |
14328 | 253 static void parse_device (char *dest, const char *src, int len) |
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254 { |
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255 char *tmp; |
14328 | 256 memmove(dest, src, len); |
257 dest[len] = 0; | |
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258 while ((tmp = strrchr(dest, '.'))) |
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259 tmp[0] = ','; |
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260 while ((tmp = strrchr(dest, '='))) |
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261 tmp[0] = ':'; |
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262 } |
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263 |
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264 static void print_help (void) |
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265 { |
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266 mp_msg (MSGT_AO, MSGL_FATAL, |
20764 | 267 MSGTR_AO_ALSA_CommandlineHelp); |
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268 } |
12465 | 269 |
14328 | 270 static int str_maxlen(strarg_t *str) { |
271 if (str->len > ALSA_DEVICE_SIZE) | |
272 return 0; | |
273 return 1; | |
274 } | |
275 | |
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276 static int try_open_device(const char *device, int open_mode, int try_ac3) |
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277 { |
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278 int err, len; |
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279 char *ac3_device, *args; |
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280 |
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281 if (try_ac3) { |
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282 /* to set the non-audio bit, use AES0=6 */ |
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283 len = strlen(device); |
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284 ac3_device = malloc(len + 7 + 1); |
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285 if (!ac3_device) |
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286 return -ENOMEM; |
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287 strcpy(ac3_device, device); |
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288 args = strchr(ac3_device, ':'); |
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289 if (!args) { |
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290 /* no existing parameters: add it behind device name */ |
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291 strcat(ac3_device, ":AES0=6"); |
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292 } else { |
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293 do |
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294 ++args; |
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295 while (isspace(*args)); |
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296 if (*args == '\0') { |
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297 /* ":" but no parameters */ |
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298 strcat(ac3_device, "AES0=6"); |
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299 } else if (*args != '{') { |
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300 /* a simple list of parameters: add it at the end of the list */ |
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301 strcat(ac3_device, ",AES0=6"); |
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302 } else { |
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303 /* parameters in config syntax: add it inside the { } block */ |
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304 do |
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305 --len; |
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306 while (len > 0 && isspace(ac3_device[len])); |
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307 if (ac3_device[len] == '}') |
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308 strcpy(ac3_device + len, " AES0=6}"); |
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309 } |
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310 } |
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311 err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK, |
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312 open_mode); |
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313 free(ac3_device); |
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314 } |
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315 if (!try_ac3 || err < 0) |
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316 err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK, |
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317 open_mode); |
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318 return err; |
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319 } |
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320 |
12465 | 321 /* |
322 open & setup audio device | |
323 return: 1=success 0=fail | |
324 */ | |
325 static int init(int rate_hz, int channels, int format, int flags) | |
326 { | |
327 int err; | |
14328 | 328 int block; |
329 strarg_t device; | |
12465 | 330 snd_pcm_uframes_t bufsize; |
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331 snd_pcm_uframes_t boundary; |
14328 | 332 opt_t subopts[] = { |
333 {"block", OPT_ARG_BOOL, &block, NULL}, | |
334 {"device", OPT_ARG_STR, &device, (opt_test_f)str_maxlen}, | |
335 {NULL} | |
336 }; | |
337 | |
12747 | 338 char alsa_device[ALSA_DEVICE_SIZE + 1]; |
339 // make sure alsa_device is null-terminated even when using strncpy etc. | |
340 memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1); | |
12465 | 341 |
14249 | 342 mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz, |
343 channels, format); | |
12465 | 344 alsa_handler = NULL; |
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345 #if SND_LIB_VERSION >= 0x010005 |
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346 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version()); |
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347 #else |
12465 | 348 mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR); |
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349 #endif |
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350 |
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351 snd_lib_error_set_handler(alsa_error_handler); |
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352 |
12465 | 353 ao_data.samplerate = rate_hz; |
354 ao_data.format = format; | |
355 ao_data.channels = channels; | |
356 | |
357 switch (format) | |
358 { | |
14245 | 359 case AF_FORMAT_S8: |
12465 | 360 alsa_format = SND_PCM_FORMAT_S8; |
361 break; | |
14245 | 362 case AF_FORMAT_U8: |
12465 | 363 alsa_format = SND_PCM_FORMAT_U8; |
364 break; | |
14245 | 365 case AF_FORMAT_U16_LE: |
12465 | 366 alsa_format = SND_PCM_FORMAT_U16_LE; |
367 break; | |
14245 | 368 case AF_FORMAT_U16_BE: |
12465 | 369 alsa_format = SND_PCM_FORMAT_U16_BE; |
370 break; | |
371 #ifndef WORDS_BIGENDIAN | |
14245 | 372 case AF_FORMAT_AC3: |
12465 | 373 #endif |
14245 | 374 case AF_FORMAT_S16_LE: |
12465 | 375 alsa_format = SND_PCM_FORMAT_S16_LE; |
376 break; | |
377 #ifdef WORDS_BIGENDIAN | |
14245 | 378 case AF_FORMAT_AC3: |
12465 | 379 #endif |
14245 | 380 case AF_FORMAT_S16_BE: |
12465 | 381 alsa_format = SND_PCM_FORMAT_S16_BE; |
382 break; | |
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383 case AF_FORMAT_U32_LE: |
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384 alsa_format = SND_PCM_FORMAT_U32_LE; |
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385 break; |
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386 case AF_FORMAT_U32_BE: |
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387 alsa_format = SND_PCM_FORMAT_U32_BE; |
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388 break; |
14245 | 389 case AF_FORMAT_S32_LE: |
12465 | 390 alsa_format = SND_PCM_FORMAT_S32_LE; |
391 break; | |
14245 | 392 case AF_FORMAT_S32_BE: |
12465 | 393 alsa_format = SND_PCM_FORMAT_S32_BE; |
394 break; | |
29386 | 395 case AF_FORMAT_U24_LE: |
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396 alsa_format = SND_PCM_FORMAT_U24_3LE; |
29386 | 397 break; |
398 case AF_FORMAT_U24_BE: | |
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399 alsa_format = SND_PCM_FORMAT_U24_3BE; |
29386 | 400 break; |
401 case AF_FORMAT_S24_LE: | |
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402 alsa_format = SND_PCM_FORMAT_S24_3LE; |
29386 | 403 break; |
404 case AF_FORMAT_S24_BE: | |
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405 alsa_format = SND_PCM_FORMAT_S24_3BE; |
29386 | 406 break; |
14245 | 407 case AF_FORMAT_FLOAT_LE: |
12570 | 408 alsa_format = SND_PCM_FORMAT_FLOAT_LE; |
409 break; | |
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410 case AF_FORMAT_FLOAT_BE: |
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411 alsa_format = SND_PCM_FORMAT_FLOAT_BE; |
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412 break; |
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413 case AF_FORMAT_MU_LAW: |
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414 alsa_format = SND_PCM_FORMAT_MU_LAW; |
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415 break; |
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416 case AF_FORMAT_A_LAW: |
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417 alsa_format = SND_PCM_FORMAT_A_LAW; |
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418 break; |
12465 | 419 |
420 default: | |
14251 | 421 alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 |
12465 | 422 break; |
423 } | |
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424 |
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425 //subdevice parsing |
14328 | 426 // set defaults |
427 block = 1; | |
12465 | 428 /* switch for spdif |
429 * sets opening sequence for SPDIF | |
430 * sets also the playback and other switches 'on the fly' | |
431 * while opening the abstract alias for the spdif subdevice | |
432 * 'iec958' | |
433 */ | |
14245 | 434 if (format == AF_FORMAT_AC3) { |
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435 device.str = "iec958"; |
12747 | 436 mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels); |
12465 | 437 } |
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438 else |
14328 | 439 /* in any case for multichannel playback we should select |
440 * appropriate device | |
441 */ | |
442 switch (channels) { | |
443 case 1: | |
444 case 2: | |
445 device.str = "default"; | |
446 mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n"); | |
447 break; | |
448 case 4: | |
449 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) | |
450 // hack - use the converter plugin | |
451 device.str = "plug:surround40"; | |
452 else | |
453 device.str = "surround40"; | |
454 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n"); | |
455 break; | |
456 case 6: | |
457 if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) | |
458 device.str = "plug:surround51"; | |
459 else | |
460 device.str = "surround51"; | |
461 mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n"); | |
462 break; | |
463 default: | |
464 device.str = "default"; | |
20764 | 465 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels); |
14328 | 466 } |
467 device.len = strlen(device.str); | |
468 if (subopt_parse(ao_subdevice, subopts) != 0) { | |
469 print_help(); | |
470 return 0; | |
471 } | |
472 ao_noblock = !block; | |
473 parse_device(alsa_device, device.str, device.len); | |
12465 | 474 |
20185 | 475 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device); |
12465 | 476 |
477 //setting modes for block or nonblock-mode | |
478 if (ao_noblock) { | |
479 open_mode = SND_PCM_NONBLOCK; | |
480 } | |
481 else { | |
482 open_mode = 0; | |
483 } | |
484 | |
485 //sets buff/chunksize if its set manually | |
486 if (ao_data.buffersize) { | |
487 switch (ao_data.buffersize) | |
488 { | |
489 case 1: | |
490 alsa_fragcount = 16; | |
491 chunk_size = 512; | |
492 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n"); | |
493 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n"); | |
494 break; | |
495 case 2: | |
496 alsa_fragcount = 8; | |
497 chunk_size = 1024; | |
498 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n"); | |
499 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n"); | |
500 break; | |
501 case 3: | |
502 alsa_fragcount = 32; | |
503 chunk_size = 512; | |
504 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n"); | |
505 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n"); | |
506 break; | |
507 case 4: | |
508 alsa_fragcount = 16; | |
509 chunk_size = 1024; | |
510 mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n"); | |
511 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n"); | |
512 break; | |
513 default: | |
514 alsa_fragcount = 16; | |
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515 chunk_size = 1024; |
12465 | 516 break; |
517 } | |
518 } | |
519 | |
520 if (!alsa_handler) { | |
521 //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC | |
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522 if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0) |
12465 | 523 { |
524 if (err != -EBUSY && ao_noblock) { | |
20764 | 525 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed); |
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526 if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) { |
20764 | 527 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err)); |
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528 return 0; |
12465 | 529 } |
530 } else { | |
20764 | 531 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err)); |
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532 return 0; |
12465 | 533 } |
534 } | |
535 | |
17619
9b619133f11a
Using non-blocking writes makes sense when the program wants to do other
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diff
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|
536 if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) { |
20764 | 537 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err)); |
12465 | 538 } else { |
20743 | 539 mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n"); |
12465 | 540 } |
541 | |
542 snd_pcm_hw_params_alloca(&alsa_hwparams); | |
543 snd_pcm_sw_params_alloca(&alsa_swparams); | |
544 | |
545 // setting hw-parameters | |
546 if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0) | |
547 { | |
20764 | 548 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters, |
12465 | 549 snd_strerror(err)); |
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550 return 0; |
12465 | 551 } |
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552 |
17616
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This patch removes mmap support because it doesn't have any benefit.
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553 err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams, |
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554 SND_PCM_ACCESS_RW_INTERLEAVED); |
12465 | 555 if (err < 0) { |
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556 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType, |
12465 | 557 snd_strerror(err)); |
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558 return 0; |
12465 | 559 } |
560 | |
561 /* workaround for nonsupported formats | |
562 sets default format to S16_LE if the given formats aren't supported */ | |
563 if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams, | |
564 alsa_format)) < 0) | |
565 { | |
566 mp_msg(MSGT_AO,MSGL_INFO, | |
20764 | 567 MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format)); |
12465 | 568 alsa_format = SND_PCM_FORMAT_S16_LE; |
14245 | 569 ao_data.format = AF_FORMAT_S16_LE; |
12465 | 570 } |
571 | |
572 if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams, | |
573 alsa_format)) < 0) | |
574 { | |
20764 | 575 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat, |
12465 | 576 snd_strerror(err)); |
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577 return 0; |
12465 | 578 } |
579 | |
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set the nearest number of channels, return(0) upon errors
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580 if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams, |
41278ab73e9b
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581 &ao_data.channels)) < 0) |
12465 | 582 { |
20764 | 583 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels, |
12465 | 584 snd_strerror(err)); |
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585 return 0; |
12465 | 586 } |
587 | |
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To avoid a bug in ALSA's rate plugin that causes spurious overruns, try
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588 /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11) |
ebebe97331af
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589 prefer our own resampler */ |
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590 #if SND_LIB_VERSION >= 0x010009 |
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591 if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams, |
ebebe97331af
To avoid a bug in ALSA's rate plugin that causes spurious overruns, try
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592 0)) < 0) |
ebebe97331af
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593 { |
20764 | 594 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling, |
17849
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595 snd_strerror(err)); |
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596 return 0; |
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To avoid a bug in ALSA's rate plugin that causes spurious overruns, try
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|
597 } |
ebebe97331af
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|
598 #endif |
ebebe97331af
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|
599 |
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600 if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, |
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601 &ao_data.samplerate, NULL)) < 0) |
12465 | 602 { |
20764 | 603 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2, |
12465 | 604 snd_strerror(err)); |
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605 return 0; |
12465 | 606 } |
607 | |
17570
401521ec0d61
This replaces the hardcoded numbers for the sample format widths with a
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608 bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8; |
401521ec0d61
This replaces the hardcoded numbers for the sample format widths with a
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609 bytes_per_sample *= ao_data.channels; |
401521ec0d61
This replaces the hardcoded numbers for the sample format widths with a
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610 ao_data.bps = ao_data.samplerate * bytes_per_sample; |
16309 | 611 |
12465 | 612 #ifdef BUFFERTIME |
613 { | |
614 int alsa_buffer_time = 500000; /* original 60 */ | |
615 int alsa_period_time; | |
616 alsa_period_time = alsa_buffer_time/4; | |
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617 if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams, |
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618 &alsa_buffer_time, NULL)) < 0) |
12465 | 619 { |
20764 | 620 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear, |
12465 | 621 snd_strerror(err)); |
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622 return 0; |
12465 | 623 } else |
624 alsa_buffer_time = err; | |
625 | |
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626 if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, |
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627 &alsa_period_time, NULL)) < 0) |
12465 | 628 /* original: alsa_buffer_time/ao_data.bps */ |
629 { | |
20764 | 630 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime, |
12465 | 631 snd_strerror(err)); |
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632 return 0; |
12465 | 633 } |
20764 | 634 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime, |
12465 | 635 alsa_buffer_time, err); |
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636 } |
12465 | 637 #endif//end SET_BUFFERTIME |
638 | |
639 #ifdef SET_CHUNKSIZE | |
640 { | |
641 //set chunksize | |
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642 if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams, |
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643 &chunk_size, NULL)) < 0) |
12465 | 644 { |
20764 | 645 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize, |
12465 | 646 chunk_size, snd_strerror(err)); |
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647 return 0; |
12465 | 648 } |
649 else { | |
17366 | 650 mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %li\n", chunk_size); |
12465 | 651 } |
652 if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams, | |
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653 &alsa_fragcount, NULL)) < 0) { |
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654 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods, |
12465 | 655 snd_strerror(err)); |
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656 return 0; |
12465 | 657 } |
658 else { | |
659 mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount); | |
660 } | |
661 } | |
662 #endif//end SET_CHUNKSIZE | |
663 | |
664 /* finally install hardware parameters */ | |
665 if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) | |
666 { | |
20764 | 667 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters, |
12465 | 668 snd_strerror(err)); |
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669 return 0; |
12465 | 670 } |
671 // end setting hw-params | |
672 | |
673 | |
674 // gets buffersize for control | |
675 if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0) | |
676 { | |
20764 | 677 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err)); |
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678 return 0; |
12465 | 679 } |
680 else { | |
681 ao_data.buffersize = bufsize * bytes_per_sample; | |
682 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize); | |
683 } | |
684 | |
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685 if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) { |
20764 | 686 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err)); |
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687 return 0; |
17620
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688 } else { |
dd4db8c43d92
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689 mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size); |
dd4db8c43d92
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690 } |
dd4db8c43d92
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691 ao_data.outburst = chunk_size * bytes_per_sample; |
dd4db8c43d92
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692 |
dd4db8c43d92
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693 /* setting software parameters */ |
dd4db8c43d92
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|
694 if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) { |
20764 | 695 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters, |
17620
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|
696 snd_strerror(err)); |
dd4db8c43d92
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|
697 return 0; |
dd4db8c43d92
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|
698 } |
18009
fb7888812f13
Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
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699 #if SND_LIB_VERSION >= 0x000901 |
17620
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700 if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) { |
20764 | 701 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary, |
17620
dd4db8c43d92
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|
702 snd_strerror(err)); |
dd4db8c43d92
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diff
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|
703 return 0; |
dd4db8c43d92
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cladisch
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|
704 } |
18009
fb7888812f13
Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
cladisch
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|
705 #else |
fb7888812f13
Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
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parents:
17849
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706 boundary = 0x7fffffff; |
fb7888812f13
Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
cladisch
parents:
17849
diff
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|
707 #endif |
17620
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|
708 /* start playing when one period has been written */ |
dd4db8c43d92
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709 if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) { |
20764 | 710 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold, |
17620
dd4db8c43d92
This changes the software parameters to be more compatible with the
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|
711 snd_strerror(err)); |
dd4db8c43d92
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|
712 return 0; |
dd4db8c43d92
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diff
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|
713 } |
dd4db8c43d92
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diff
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|
714 /* disable underrun reporting */ |
dd4db8c43d92
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cladisch
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|
715 if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) { |
20764 | 716 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold, |
17620
dd4db8c43d92
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17619
diff
changeset
|
717 snd_strerror(err)); |
dd4db8c43d92
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cladisch
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|
718 return 0; |
dd4db8c43d92
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cladisch
parents:
17619
diff
changeset
|
719 } |
18009
fb7888812f13
Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
cladisch
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17849
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|
720 #if SND_LIB_VERSION >= 0x000901 |
17620
dd4db8c43d92
This changes the software parameters to be more compatible with the
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|
721 /* play silence when there is an underrun */ |
dd4db8c43d92
This changes the software parameters to be more compatible with the
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|
722 if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) { |
20764 | 723 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize, |
17620
dd4db8c43d92
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parents:
17619
diff
changeset
|
724 snd_strerror(err)); |
dd4db8c43d92
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diff
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|
725 return 0; |
dd4db8c43d92
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17619
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|
726 } |
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Add workarounds for old prerelease versions of alsa-lib 0.9.0 that did
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|
727 #endif |
17620
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728 if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) { |
20764 | 729 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters, |
17620
dd4db8c43d92
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|
730 snd_strerror(err)); |
dd4db8c43d92
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|
731 return 0; |
dd4db8c43d92
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|
732 } |
dd4db8c43d92
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733 /* end setting sw-params */ |
dd4db8c43d92
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734 |
20185 | 735 mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", |
27682
25245b3e962f
Fix "format '%d' expects type 'int', but argument 6 has type 'size_t'" warning.
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|
736 ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize, |
12465 | 737 snd_pcm_format_description(alsa_format)); |
738 | |
739 } // end switch alsa_handler (spdif) | |
740 alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); | |
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741 return 1; |
12465 | 742 } // end init |
743 | |
744 | |
745 /* close audio device */ | |
746 static void uninit(int immed) | |
747 { | |
748 | |
749 if (alsa_handler) { | |
750 int err; | |
751 | |
14849
d313f591d1a4
aos should respect the immed uninit flag (quit immediatly vs waiting till file
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752 if (!immed) |
d313f591d1a4
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|
753 snd_pcm_drain(alsa_handler); |
d313f591d1a4
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diff
changeset
|
754 |
12465 | 755 if ((err = snd_pcm_close(alsa_handler)) < 0) |
756 { | |
20764 | 757 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err)); |
12465 | 758 return; |
759 } | |
760 else { | |
761 alsa_handler = NULL; | |
20185 | 762 mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n"); |
12465 | 763 } |
764 } | |
765 else { | |
20764 | 766 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined); |
12465 | 767 } |
768 } | |
769 | |
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770 static void audio_pause(void) |
12465 | 771 { |
772 int err; | |
773 | |
774 if (alsa_can_pause) { | |
775 if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) | |
776 { | |
20764 | 777 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err)); |
12465 | 778 return; |
779 } | |
780 mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n"); | |
781 } else { | |
782 if ((err = snd_pcm_drop(alsa_handler)) < 0) | |
783 { | |
20764 | 784 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err)); |
12465 | 785 return; |
786 } | |
787 } | |
788 } | |
789 | |
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790 static void audio_resume(void) |
12465 | 791 { |
792 int err; | |
793 | |
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794 if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { |
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795 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume); |
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796 while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); |
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797 } |
12465 | 798 if (alsa_can_pause) { |
799 if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) | |
800 { | |
20764 | 801 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err)); |
12465 | 802 return; |
803 } | |
804 mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n"); | |
805 } else { | |
806 if ((err = snd_pcm_prepare(alsa_handler)) < 0) | |
807 { | |
20764 | 808 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); |
12465 | 809 return; |
810 } | |
811 } | |
812 } | |
813 | |
814 /* stop playing and empty buffers (for seeking/pause) */ | |
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815 static void reset(void) |
12465 | 816 { |
817 int err; | |
818 | |
819 if ((err = snd_pcm_drop(alsa_handler)) < 0) | |
820 { | |
20764 | 821 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); |
12465 | 822 return; |
823 } | |
824 if ((err = snd_pcm_prepare(alsa_handler)) < 0) | |
825 { | |
20764 | 826 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); |
12465 | 827 return; |
828 } | |
829 return; | |
830 } | |
831 | |
832 /* | |
833 plays 'len' bytes of 'data' | |
834 returns: number of bytes played | |
835 modified last at 29.06.02 by jp | |
836 thanxs for marius <marius@rospot.com> for giving us the light ;) | |
837 */ | |
838 | |
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839 static int play(void* data, int len, int flags) |
12465 | 840 { |
841 int num_frames = len / bytes_per_sample; | |
842 snd_pcm_sframes_t res = 0; | |
843 | |
844 //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); | |
845 | |
846 if (!alsa_handler) { | |
20764 | 847 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError); |
12465 | 848 return 0; |
849 } | |
850 | |
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851 if (num_frames == 0) |
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852 return 0; |
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853 |
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854 do { |
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855 res = snd_pcm_writei(alsa_handler, data, num_frames); |
12465 | 856 |
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857 if (res == -EINTR) { |
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858 /* nothing to do */ |
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859 res = 0; |
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860 } |
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861 else if (res == -ESTRPIPE) { /* suspend */ |
20764 | 862 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume); |
12465 | 863 while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) |
864 sleep(1); | |
865 } | |
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866 if (res < 0) { |
20764 | 867 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res)); |
868 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard); | |
12465 | 869 if ((res = snd_pcm_prepare(alsa_handler)) < 0) { |
20764 | 870 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res)); |
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871 return 0; |
12465 | 872 break; |
873 } | |
874 } | |
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875 } while (res == 0); |
12465 | 876 |
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877 return res < 0 ? res : res * bytes_per_sample; |
12465 | 878 } |
879 | |
880 /* how many byes are free in the buffer */ | |
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881 static int get_space(void) |
12465 | 882 { |
883 snd_pcm_status_t *status; | |
884 int ret; | |
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885 |
12747 | 886 snd_pcm_status_alloca(&status); |
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887 |
12465 | 888 if ((ret = snd_pcm_status(alsa_handler, status)) < 0) |
889 { | |
20764 | 890 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret)); |
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891 return 0; |
12465 | 892 } |
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893 |
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894 ret = snd_pcm_status_get_avail(status) * bytes_per_sample; |
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895 if (ret > ao_data.buffersize) // Buffer underrun? |
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896 ret = ao_data.buffersize; |
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897 return ret; |
12465 | 898 } |
899 | |
900 /* delay in seconds between first and last sample in buffer */ | |
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901 static float get_delay(void) |
12465 | 902 { |
903 if (alsa_handler) { | |
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904 snd_pcm_sframes_t delay; |
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905 |
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906 if (snd_pcm_delay(alsa_handler, &delay) < 0) |
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907 return 0; |
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908 |
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909 if (delay < 0) { |
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910 /* underrun - move the application pointer forward to catch up */ |
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911 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */ |
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912 snd_pcm_forward(alsa_handler, -delay); |
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913 #endif |
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914 delay = 0; |
12465 | 915 } |
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916 return (float)delay / (float)ao_data.samplerate; |
12465 | 917 } else { |
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918 return 0; |
12465 | 919 } |
920 } |