Mercurial > audlegacy
annotate src/audacious/output.c @ 3957:fed07be6b708
every other menu shows on button press..
| author | Tomasz Mon <desowin@gmail.com> |
|---|---|
| date | Fri, 16 Nov 2007 15:20:07 +0100 |
| parents | bce5ac1f368b |
| children | 81b4e186765f |
| rev | line source |
|---|---|
| 2313 | 1 /* Audacious - Cross-platform multimedia player |
| 2 * Copyright (C) 2005-2007 Audacious team | |
| 3 * | |
| 4 * Based on BMP: | |
| 5 * Copyright (C) 2003-2004 BMP development team. | |
| 6 * | |
| 7 * Based on XMMS: | |
| 8 * Copyright (C) 1998-2003 XMMS development team. | |
| 9 * | |
| 10 * This program is free software; you can redistribute it and/or modify | |
| 11 * it under the terms of the GNU General Public License as published by | |
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12 * the Free Software Foundation; under version 3 of the License. |
| 2313 | 13 * |
| 14 * This program is distributed in the hope that it will be useful, | |
| 15 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
| 17 * GNU General Public License for more details. | |
| 18 * | |
| 19 * You should have received a copy of the GNU General Public License | |
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20 * along with this program. If not, see <http://www.gnu.org/licenses>. |
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21 * |
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22 * The Audacious team does not consider modular code linking to |
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23 * Audacious or using our public API to be a derived work. |
| 2313 | 24 */ |
| 25 | |
| 26 #ifdef HAVE_CONFIG_H | |
| 27 # include "config.h" | |
| 28 #endif | |
| 29 | |
| 30 #include "output.h" | |
| 31 #include "iir.h" | |
| 32 #include "main.h" | |
| 33 #include "input.h" | |
| 2328 | 34 #include "playback.h" |
| 2313 | 35 |
| 36 #include "playlist.h" | |
| 2717 | 37 #include "configdb.h" |
| 2542 | 38 |
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39 #include "flow.h" |
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40 |
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41 #include "effect.h" |
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42 #include "volumecontrol.h" |
| 3559 | 43 #include "visualization.h" |
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44 |
| 2542 | 45 #include <math.h> |
| 46 | |
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47 #ifdef USE_SRC |
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48 #include <samplerate.h> |
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49 #endif |
| 2313 | 50 |
| 51 OutputPluginData op_data = { | |
| 52 NULL, | |
| 53 NULL | |
| 54 }; | |
| 55 | |
| 56 OutputPluginState op_state = { | |
| 57 0, | |
| 58 0, | |
| 59 0 | |
| 60 }; | |
| 61 | |
| 62 OutputPlugin psuedo_output_plugin = { | |
| 3549 | 63 .description = "XMMS reverse compatibility output plugin", |
| 64 .get_volume = output_get_volume, | |
| 65 .set_volume = output_set_volume, | |
| 66 | |
| 67 .open_audio = output_open_audio, | |
| 68 .write_audio = output_write_audio, | |
| 69 .close_audio = output_close_audio, | |
| 2313 | 70 |
| 3549 | 71 .flush = output_flush, |
| 72 .pause = output_pause, | |
| 73 | |
| 74 .buffer_free = output_buffer_free, | |
| 75 .buffer_playing = output_buffer_playing, | |
| 76 .output_time = get_output_time, | |
| 77 .written_time = get_written_time, | |
| 2313 | 78 }; |
| 79 | |
| 80 OutputPlugin * | |
| 81 get_current_output_plugin(void) | |
| 82 { | |
| 83 return op_data.current_output_plugin; | |
| 84 } | |
| 85 | |
| 86 void | |
| 87 set_current_output_plugin(gint i) | |
| 88 { | |
| 89 gboolean playing; | |
| 90 OutputPlugin *op = get_current_output_plugin(); | |
| 91 | |
| 92 GList *node = g_list_nth(op_data.output_list, i); | |
| 93 if (!node) { | |
| 94 op_data.current_output_plugin = NULL; | |
| 95 return; | |
| 96 } | |
| 97 | |
| 98 op_data.current_output_plugin = node->data; | |
| 99 | |
| 100 playing = playback_get_playing(); | |
| 101 | |
| 102 if (playing == TRUE) | |
| 103 { | |
| 104 guint time, pos; | |
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105 PlaylistEntry *entry; |
| 2313 | 106 |
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107 /* don't stop yet, get the seek time and playlist position first */ |
| 2313 | 108 pos = playlist_get_position(playlist_get_active()); |
| 109 time = op->output_time(); | |
| 110 | |
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111 /* reset the audio system */ |
| 2313 | 112 mainwin_stop_pushed(); |
| 113 op->close_audio(); | |
| 114 | |
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115 g_usleep(300000); |
| 2313 | 116 |
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117 /* wait for the playback thread to come online */ |
| 2313 | 118 while (playback_get_playing()) |
| 119 g_message("waiting for audio system shutdown..."); | |
| 120 | |
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121 /* wait for the playback thread to come online */ |
| 2313 | 122 playlist_set_position(playlist_get_active(), pos); |
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123 entry = playlist_get_entry_to_play(playlist_get_active()); |
| 2313 | 124 playback_play_file(entry); |
| 125 | |
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126 while (!playback_get_playing()) |
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127 { |
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128 gtk_main_iteration(); |
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129 g_message("waiting for audio system startup..."); |
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130 } |
| 2313 | 131 |
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132 /* and signal a reseek */ |
| 2313 | 133 if (playlist_get_current_length(playlist_get_active()) > -1 && |
| 134 time <= (playlist_get_current_length(playlist_get_active()))) | |
| 135 playback_seek(time / 1000); | |
| 136 } | |
| 137 } | |
| 138 | |
| 139 GList * | |
| 140 get_output_list(void) | |
| 141 { | |
| 142 return op_data.output_list; | |
| 143 } | |
| 144 | |
| 145 void | |
| 146 output_about(gint i) | |
| 147 { | |
| 148 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
| 149 if (out && out->about) | |
| 150 out->about(); | |
| 151 } | |
| 152 | |
| 153 void | |
| 154 output_configure(gint i) | |
| 155 { | |
| 156 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
| 157 if (out && out->configure) | |
| 158 out->configure(); | |
| 159 } | |
| 160 | |
| 161 void | |
| 162 output_get_volume(gint * l, gint * r) | |
| 163 { | |
| 164 *l = *r = -1; | |
| 165 | |
| 166 if (!op_data.current_output_plugin) | |
| 167 return; | |
| 168 | |
| 169 if (!op_data.current_output_plugin->get_volume) | |
| 170 return; | |
| 171 | |
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172 if (cfg.software_volume_control) |
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173 volumecontrol_get_volume_state(l, r); |
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174 else |
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175 op_data.current_output_plugin->get_volume(l, r); |
| 2313 | 176 } |
| 177 | |
| 178 void | |
| 179 output_set_volume(gint l, gint r) | |
| 180 { | |
| 181 if (!op_data.current_output_plugin) | |
| 182 return; | |
| 183 | |
| 184 if (!op_data.current_output_plugin->set_volume) | |
| 185 return; | |
| 186 | |
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187 if (cfg.software_volume_control) |
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188 volumecontrol_set_volume_state(l, r); |
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189 else |
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190 op_data.current_output_plugin->set_volume(l, r); |
| 2313 | 191 } |
| 192 | |
| 193 void | |
| 194 output_set_eq(gboolean active, gfloat pre, gfloat * bands) | |
| 195 { | |
| 196 int i; | |
| 197 preamp[0] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre; | |
| 198 preamp[1] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre; | |
| 199 | |
| 200 for (i = 0; i < 10; ++i) | |
| 201 { | |
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202 set_gain(i, 0, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]); |
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203 set_gain(i, 1, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]); |
| 2313 | 204 } |
| 205 } | |
| 206 | |
| 207 /* called by input plugin to peek at the output plugin's write progress */ | |
| 208 gint | |
| 209 get_written_time(void) | |
| 210 { | |
| 211 OutputPlugin *op = get_current_output_plugin(); | |
| 212 | |
| 213 return op->written_time(); | |
| 214 } | |
| 215 | |
| 216 /* called by input plugin to peek at the output plugin's output progress */ | |
| 217 gint | |
| 218 get_output_time(void) | |
| 219 { | |
| 220 OutputPlugin *op = get_current_output_plugin(); | |
| 221 | |
| 222 return op->output_time(); | |
| 223 } | |
| 224 | |
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225 #ifdef USE_SRC |
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226 |
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227 static SRC_STATE *src_state; |
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228 static SRC_DATA src_data; |
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229 static int overSamplingFs = 96000; |
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230 static int converter_type = SRC_SINC_BEST_QUALITY; |
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231 static int srcError = 0; |
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232 |
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233 static float *srcIn = NULL, *srcOut = NULL; |
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234 static short int *wOut = NULL; |
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235 static int lengthOfSrcIn = 0; |
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236 static int lengthOfSrcOut = 0; |
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237 |
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238 static void freeSRC() |
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239 { |
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240 if(src_state != NULL) |
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241 src_state = src_delete(src_state); |
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242 free(srcIn); |
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243 free(srcOut); |
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244 free(wOut); |
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245 srcIn = NULL; |
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246 srcOut = NULL; |
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247 wOut = NULL; |
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248 lengthOfSrcIn = 0; |
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249 lengthOfSrcOut = 0; |
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250 } |
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251 |
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252 #endif |
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253 |
| 2313 | 254 gint |
| 255 output_open_audio(AFormat fmt, gint rate, gint nch) | |
| 256 { | |
| 257 gint ret; | |
| 258 OutputPlugin *op; | |
| 259 | |
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260 #ifdef USE_SRC |
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261 ConfigDb *db; |
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262 gboolean src_enabled; |
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263 gint src_rate, src_type; |
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264 db = cfg_db_open(); |
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265 |
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266 if (cfg_db_get_bool(db, NULL, "enable_src", &src_enabled) == FALSE) |
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267 src_enabled = FALSE; |
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268 |
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269 if (cfg_db_get_int(db, NULL, "src_rate", &src_rate) == FALSE) |
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270 overSamplingFs = 48000; |
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271 else |
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272 overSamplingFs = src_rate; |
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273 |
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274 /* don't resample if sampling rates are the same --nenolod */ |
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275 if (rate == overSamplingFs) |
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276 src_enabled = FALSE; |
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277 |
|
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278 if (cfg_db_get_int(db, NULL, "src_type", &src_type) == FALSE) |
|
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279 converter_type = SRC_SINC_BEST_QUALITY; |
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280 else |
|
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281 converter_type = src_type; |
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282 |
|
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283 cfg_db_close(db); |
|
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284 |
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285 freeSRC(); |
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286 |
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287 if(src_enabled&& |
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288 (fmt == FMT_S16_NE||(fmt == FMT_S16_LE && G_BYTE_ORDER == G_LITTLE_ENDIAN)|| |
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289 (fmt == FMT_S16_BE && G_BYTE_ORDER == G_BIG_ENDIAN))) |
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290 { |
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291 src_state = src_new(converter_type, nch, &srcError); |
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292 if (src_state != NULL) |
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293 { |
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294 src_data.src_ratio = (float)overSamplingFs/(float)rate; |
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295 rate = overSamplingFs; |
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296 } |
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297 else |
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298 fprintf(stderr, "src_new(): %s\n\n", src_strerror(srcError)); |
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299 } |
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300 #endif |
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301 |
| 2313 | 302 op = get_current_output_plugin(); |
| 303 | |
| 304 if (op == NULL) | |
| 305 return -1; | |
| 306 | |
| 307 /* Is our output port already open? */ | |
| 308 if ((op_state.rate != 0 && op_state.nch != 0) && | |
|
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309 (op_state.rate == rate && op_state.nch == nch && op_state.fmt == fmt)) |
| 2313 | 310 { |
|
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311 /* Yes, and it's the correct sampling rate. Reset the counter and go. */ |
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312 op->flush(0); |
| 2313 | 313 return 1; |
| 314 } | |
| 315 else if (op_state.rate != 0 && op_state.nch != 0) | |
|
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316 op->close_audio(); |
| 2313 | 317 |
| 318 ret = op->open_audio(fmt, rate, nch); | |
| 319 | |
|
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320 if (ret == 1) /* Success? */ |
| 2313 | 321 { |
| 322 op_state.fmt = fmt; | |
| 323 op_state.rate = rate; | |
| 324 op_state.nch = nch; | |
| 325 } | |
| 326 | |
| 327 return ret; | |
| 328 } | |
| 329 | |
| 330 void | |
| 331 output_write_audio(gpointer ptr, gint length) | |
| 332 { | |
| 333 OutputPlugin *op = get_current_output_plugin(); | |
| 334 | |
| 335 /* Sanity check. */ | |
| 336 if (op == NULL) | |
| 337 return; | |
| 338 | |
| 339 op->write_audio(ptr, length); | |
| 340 } | |
| 341 | |
| 342 void | |
| 343 output_close_audio(void) | |
| 344 { | |
| 345 OutputPlugin *op = get_current_output_plugin(); | |
| 346 | |
|
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347 #ifdef USE_SRC |
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348 freeSRC(); |
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349 #endif |
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350 |
| 2313 | 351 /* Do not close if there are still songs to play and the user has |
| 352 * not requested a stop. --nenolod | |
| 353 */ | |
| 354 if (ip_data.stop == FALSE && | |
| 2673 | 355 (playlist_get_position_nolock(playlist_get_active()) < |
| 356 playlist_get_length(playlist_get_active()) - 1)) | |
| 2313 | 357 return; |
| 358 | |
| 359 /* Sanity check. */ | |
| 360 if (op == NULL) | |
| 361 return; | |
| 362 | |
| 363 op->close_audio(); | |
| 364 | |
| 365 /* Reset the op_state. */ | |
| 366 op_state.fmt = op_state.rate = op_state.nch = 0; | |
| 367 } | |
| 368 | |
| 369 void | |
| 370 output_flush(gint time) | |
| 371 { | |
| 372 OutputPlugin *op = get_current_output_plugin(); | |
| 373 | |
| 374 if (op == NULL) | |
| 375 return; | |
| 376 | |
| 377 op->flush(time); | |
| 378 } | |
| 379 | |
| 380 void | |
| 381 output_pause(gshort paused) | |
| 382 { | |
| 383 OutputPlugin *op = get_current_output_plugin(); | |
| 384 | |
| 385 if (op == NULL) | |
| 386 return; | |
| 387 | |
| 388 op->pause(paused); | |
| 389 } | |
| 390 | |
| 391 gint | |
| 392 output_buffer_free(void) | |
| 393 { | |
| 394 OutputPlugin *op = get_current_output_plugin(); | |
| 395 | |
| 396 if (op == NULL) | |
| 397 return 0; | |
| 398 | |
| 399 return op->buffer_free(); | |
| 400 } | |
| 401 | |
| 402 gint | |
| 403 output_buffer_playing(void) | |
| 404 { | |
| 405 OutputPlugin *op = get_current_output_plugin(); | |
| 406 | |
| 407 if (op == NULL) | |
| 408 return 0; | |
| 409 | |
| 410 return op->buffer_playing(); | |
| 411 } | |
| 412 | |
| 413 /* called by input plugin when data is ready */ | |
| 414 void | |
|
3709
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Add InputPlayback::pass_audio() which replaces produce_audio().
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|
415 output_pass_audio(InputPlayback *playback, |
|
3696
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
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|
416 AFormat fmt, /* output format */ |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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417 gint nch, /* channels */ |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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|
418 gint length, /* length of sample */ |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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419 gpointer ptr, /* data */ |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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|
420 int *going /* 0 when time to stop */ |
| 2313 | 421 ) |
| 422 { | |
|
3553
a140fadd741d
software volumecontrol->flow manager API
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3549
diff
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|
423 static Flow *postproc_flow = NULL; |
|
3696
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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424 OutputPlugin *op = playback->output; |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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|
425 gint writeoffs; |
|
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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426 gint time = playback->output->written_time(); |
| 2313 | 427 |
|
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|
428 if (postproc_flow == NULL) |
|
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|
429 { |
|
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|
430 postproc_flow = flow_new(); |
|
3560
154f21dcd61e
equalizer -> flow API
William Pitcock <nenolod@atheme.org>
parents:
3559
diff
changeset
|
431 flow_link_element(postproc_flow, iir_flow); |
|
3555
a73951b8cd9f
effect processing -> flow manager API / attached to postproc_flow.
William Pitcock <nenolod@atheme.org>
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3553
diff
changeset
|
432 flow_link_element(postproc_flow, effect_flow); |
| 3559 | 433 flow_link_element(postproc_flow, vis_flow); |
|
3553
a140fadd741d
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|
434 flow_link_element(postproc_flow, volumecontrol_flow); |
|
a140fadd741d
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|
435 } |
|
a140fadd741d
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diff
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|
436 |
|
2538
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|
437 #ifdef USE_SRC |
|
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438 if(src_state != NULL&&length > 0) |
|
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|
439 { |
|
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|
440 int lrLength = length/2; |
|
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|
441 int overLrLength = (int)floor(lrLength*(src_data.src_ratio+1)); |
|
2648
51495131e4ae
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|
442 if(lengthOfSrcIn < lrLength) |
|
51495131e4ae
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|
443 { |
|
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|
444 lengthOfSrcIn = lrLength; |
|
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[svn] - synchronize SRC conversion patch with upstream
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|
445 free(srcIn); |
|
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|
446 srcIn = (float*)malloc(sizeof(float)*lrLength); |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
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|
447 } |
|
51495131e4ae
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|
448 if(lengthOfSrcOut < overLrLength) |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
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|
449 { |
|
51495131e4ae
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nenolod
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changeset
|
450 lengthOfSrcOut = overLrLength; |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
nenolod
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|
451 free(srcOut); |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
nenolod
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|
452 free(wOut); |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
nenolod
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|
453 srcOut = (float*)malloc(sizeof(float)*overLrLength); |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
nenolod
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|
454 wOut = (short int*)malloc(sizeof(short int)*overLrLength); |
|
51495131e4ae
[svn] - synchronize SRC conversion patch with upstream
nenolod
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diff
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|
455 } |
|
2538
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|
456 src_short_to_float_array((short int*)ptr, srcIn, lrLength); |
|
719e0898ff3c
[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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|
457 src_data.data_in = srcIn; |
|
719e0898ff3c
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|
458 src_data.data_out = srcOut; |
|
719e0898ff3c
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|
459 src_data.end_of_input = 0; |
|
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|
460 src_data.input_frames = lrLength/2; |
|
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[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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|
461 src_data.output_frames = overLrLength/2; |
| 2542 | 462 if ((srcError = src_process(src_state, &src_data)) > 0) |
|
2538
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[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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|
463 { |
|
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|
464 fprintf(stderr, "src_process(): %s\n", src_strerror(srcError)); |
|
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|
465 } |
|
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|
466 else |
|
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|
467 { |
|
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|
468 src_float_to_short_array(srcOut, wOut, src_data.output_frames_gen*2); |
|
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|
469 ptr = wOut; |
|
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470 length = src_data.output_frames_gen*4; |
|
719e0898ff3c
[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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|
471 } |
|
719e0898ff3c
[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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|
472 } |
|
719e0898ff3c
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|
473 #endif |
|
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[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
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2373
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|
474 |
|
3708
6f4068a0f291
make sndstretch work properly
William Pitcock <nenolod@atheme.org>
parents:
3696
diff
changeset
|
475 length = flow_execute(postproc_flow, time, &ptr, length, op_state.fmt, op_state.rate, op_state.nch); |
|
3546
fd4cd1945b91
Connect volumecontrol flow to the work pipeline.
William Pitcock <nenolod@atheme.org>
parents:
3486
diff
changeset
|
476 |
| 2313 | 477 writeoffs = 0; |
|
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
2328
diff
changeset
|
478 while (writeoffs < length) |
|
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
2328
diff
changeset
|
479 { |
|
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
2328
diff
changeset
|
480 int writable = length - writeoffs; |
| 2313 | 481 |
|
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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2328
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changeset
|
482 if (writable > 2048) |
|
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
2328
diff
changeset
|
483 writable = 2048; |
| 2313 | 484 |
|
2813
49f3d1b43518
[svn] - code cleanups involving SRC patch and indentation
nenolod
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2811
diff
changeset
|
485 if (writable == 0) |
|
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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486 return; |
| 2313 | 487 |
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488 while (op->buffer_free() < writable) /* wait output buf */ |
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489 { |
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490 GTimeVal pb_abs_time; |
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491 |
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492 g_get_current_time(&pb_abs_time); |
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493 g_time_val_add(&pb_abs_time, (cfg.output_buffer_size / 2) * 1000); |
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494 |
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495 if (going && !*going) /* thread stopped? */ |
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496 return; /* so finish */ |
| 2313 | 497 |
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498 if (ip_data.stop) /* has a stop been requested? */ |
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499 return; /* yes, so finish */ |
| 2313 | 500 |
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501 /* else sleep for retry */ |
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502 #ifndef GDK_WINDOWING_QUARTZ |
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503 g_mutex_lock(playback->pb_change_mutex); |
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504 g_cond_timed_wait(playback->pb_change_cond, playback->pb_change_mutex, &pb_abs_time); |
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505 g_mutex_unlock(playback->pb_change_mutex); |
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506 #else |
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507 /* Darwin threading sucks. */ |
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508 g_usleep(10000); |
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509 #endif |
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510 } |
| 2313 | 511 |
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512 if (ip_data.stop) |
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513 return; |
| 2313 | 514 |
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515 if (going && !*going) /* thread stopped? */ |
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516 return; /* so finish */ |
| 2313 | 517 |
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518 /* do output */ |
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519 op->write_audio(((guint8 *) ptr) + writeoffs, writable); |
| 2313 | 520 |
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521 writeoffs += writable; |
| 2313 | 522 } |
| 523 } |
