Mercurial > audlegacy
annotate src/audacious/output.c @ 4287:19c4a20d11a1
cleanup
author | Eugene Zagidullin <e.asphyx@gmail.com> |
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date | Sat, 16 Feb 2008 02:38:07 +0300 |
parents | af324ff0a983 |
children | 5b277773870e |
rev | line source |
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2313 | 1 /* Audacious - Cross-platform multimedia player |
4234 | 2 * Copyright (C) 2005-2008 Audacious team |
2313 | 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 | |
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26 /* #define AUD_DEBUG */ |
4234 | 27 |
2313 | 28 #ifdef HAVE_CONFIG_H |
29 # include "config.h" | |
30 #endif | |
31 | |
32 #include "output.h" | |
33 #include "iir.h" | |
34 #include "main.h" | |
35 #include "input.h" | |
2328 | 36 #include "playback.h" |
2313 | 37 |
38 #include "playlist.h" | |
2717 | 39 #include "configdb.h" |
2542 | 40 |
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41 #include "flow.h" |
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42 |
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43 #include "pluginenum.h" |
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44 |
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45 #include "effect.h" |
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46 #include "volumecontrol.h" |
3559 | 47 #include "visualization.h" |
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48 |
4234 | 49 #include "libSAD.h" |
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50 #include "util.h" |
4234 | 51 |
2542 | 52 #include <math.h> |
53 | |
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54 #ifdef USE_SRC |
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55 # include "src_flow.h" |
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56 #endif |
2313 | 57 |
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58 #define FMT_FRACBITS(a) ( (a) == FMT_FIXED32 ? __AUDACIOUS_ASSUMED_MAD_F_FRACBITS__ : 0 ) |
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59 |
2313 | 60 OutputPluginData op_data = { |
61 NULL, | |
62 NULL | |
63 }; | |
64 | |
65 OutputPluginState op_state = { | |
66 0, | |
67 0, | |
68 0 | |
69 }; | |
70 | |
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71 static gint decoder_srate = 0; |
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72 static gboolean bypass_dsp = FALSE; |
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73 |
2313 | 74 OutputPlugin psuedo_output_plugin = { |
3549 | 75 .description = "XMMS reverse compatibility output plugin", |
76 .get_volume = output_get_volume, | |
77 .set_volume = output_set_volume, | |
78 | |
79 .open_audio = output_open_audio, | |
80 .write_audio = output_write_audio, | |
81 .close_audio = output_close_audio, | |
2313 | 82 |
3549 | 83 .flush = output_flush, |
84 .pause = output_pause, | |
85 | |
86 .buffer_free = output_buffer_free, | |
87 .buffer_playing = output_buffer_playing, | |
88 .output_time = get_output_time, | |
89 .written_time = get_written_time, | |
2313 | 90 }; |
91 | |
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92 static void apply_replaygain_info (ReplayGainInfo *rg_info); |
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93 |
2313 | 94 OutputPlugin * |
95 get_current_output_plugin(void) | |
96 { | |
97 return op_data.current_output_plugin; | |
98 } | |
99 | |
100 void | |
101 set_current_output_plugin(gint i) | |
102 { | |
103 gboolean playing; | |
104 OutputPlugin *op = get_current_output_plugin(); | |
105 | |
106 GList *node = g_list_nth(op_data.output_list, i); | |
107 if (!node) { | |
108 op_data.current_output_plugin = NULL; | |
109 return; | |
110 } | |
111 | |
112 op_data.current_output_plugin = node->data; | |
113 | |
114 playing = playback_get_playing(); | |
115 | |
116 if (playing == TRUE) | |
117 { | |
118 guint time, pos; | |
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119 PlaylistEntry *entry; |
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120 |
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121 plugin_set_current((Plugin *)op); |
2313 | 122 |
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123 /* don't stop yet, get the seek time and playlist position first */ |
2313 | 124 pos = playlist_get_position(playlist_get_active()); |
125 time = op->output_time(); | |
126 | |
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127 /* reset the audio system */ |
2313 | 128 mainwin_stop_pushed(); |
129 op->close_audio(); | |
130 | |
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131 g_usleep(300000); |
2313 | 132 |
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133 /* wait for the playback thread to come online */ |
2313 | 134 while (playback_get_playing()) |
135 g_message("waiting for audio system shutdown..."); | |
136 | |
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137 /* wait for the playback thread to come online */ |
2313 | 138 playlist_set_position(playlist_get_active(), pos); |
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139 entry = playlist_get_entry_to_play(playlist_get_active()); |
2313 | 140 playback_play_file(entry); |
141 | |
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142 while (!playback_get_playing()) |
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143 { |
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144 gtk_main_iteration(); |
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145 g_message("waiting for audio system startup..."); |
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146 } |
2313 | 147 |
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148 /* and signal a reseek */ |
2313 | 149 if (playlist_get_current_length(playlist_get_active()) > -1 && |
150 time <= (playlist_get_current_length(playlist_get_active()))) | |
151 playback_seek(time / 1000); | |
152 } | |
153 } | |
154 | |
155 GList * | |
156 get_output_list(void) | |
157 { | |
158 return op_data.output_list; | |
159 } | |
160 | |
161 void | |
162 output_about(gint i) | |
163 { | |
164 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
165 if (out && out->about) | |
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166 { |
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167 plugin_set_current((Plugin *)out); |
2313 | 168 out->about(); |
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169 } |
2313 | 170 } |
171 | |
172 void | |
173 output_configure(gint i) | |
174 { | |
175 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
176 if (out && out->configure) | |
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177 { |
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178 plugin_set_current((Plugin *)out); |
2313 | 179 out->configure(); |
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180 } |
2313 | 181 } |
182 | |
183 void | |
184 output_get_volume(gint * l, gint * r) | |
185 { | |
186 *l = *r = -1; | |
187 | |
188 if (!op_data.current_output_plugin) | |
189 return; | |
190 | |
191 if (!op_data.current_output_plugin->get_volume) | |
192 return; | |
193 | |
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194 if (cfg.software_volume_control) |
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195 volumecontrol_get_volume_state(l, r); |
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196 else |
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197 { |
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198 plugin_set_current((Plugin *)op_data.current_output_plugin); |
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199 op_data.current_output_plugin->get_volume(l, r); |
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200 } |
2313 | 201 } |
202 | |
203 void | |
204 output_set_volume(gint l, gint r) | |
205 { | |
206 if (!op_data.current_output_plugin) | |
207 return; | |
208 | |
209 if (!op_data.current_output_plugin->set_volume) | |
210 return; | |
211 | |
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212 if (cfg.software_volume_control) |
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213 volumecontrol_set_volume_state(l, r); |
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214 else |
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215 { |
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216 plugin_set_current((Plugin *)op_data.current_output_plugin); |
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217 op_data.current_output_plugin->set_volume(l, r); |
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218 } |
2313 | 219 } |
220 | |
221 void | |
222 output_set_eq(gboolean active, gfloat pre, gfloat * bands) | |
223 { | |
224 int i; | |
225 preamp[0] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre; | |
226 preamp[1] = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre; | |
227 | |
228 for (i = 0; i < 10; ++i) | |
229 { | |
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230 set_gain(i, 0, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]); |
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231 set_gain(i, 1, 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]); |
2313 | 232 } |
233 } | |
234 | |
235 /* called by input plugin to peek at the output plugin's write progress */ | |
236 gint | |
237 get_written_time(void) | |
238 { | |
239 OutputPlugin *op = get_current_output_plugin(); | |
240 | |
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241 plugin_set_current((Plugin *)op); |
2313 | 242 return op->written_time(); |
243 } | |
244 | |
245 /* called by input plugin to peek at the output plugin's output progress */ | |
246 gint | |
247 get_output_time(void) | |
248 { | |
249 OutputPlugin *op = get_current_output_plugin(); | |
250 | |
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251 plugin_set_current((Plugin *)op); |
2313 | 252 return op->output_time(); |
253 } | |
254 | |
4234 | 255 static SAD_dither_t *sad_state_to_float = NULL; |
256 static SAD_dither_t *sad_state_from_float = NULL; | |
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257 static float *sad_float_buf = NULL; |
4234 | 258 static void *sad_out_buf = NULL; |
4278 | 259 static gsize sad_float_buf_length = 0; |
260 static gsize sad_out_buf_length = 0; | |
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261 static ReplayGainInfo replay_gain_info = { |
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262 .track_gain = 0.0, |
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263 .track_peak = 0.0, |
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264 .album_gain = 0.0, |
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265 .album_peak = 0.0, |
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266 }; |
4234 | 267 |
268 static void | |
269 freeSAD() | |
270 { | |
271 if (sad_state_from_float != NULL) {SAD_dither_free(sad_state_from_float); sad_state_from_float = NULL;} | |
272 if (sad_state_to_float != NULL) {SAD_dither_free(sad_state_to_float); sad_state_to_float = NULL;} | |
273 } | |
274 | |
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275 static gboolean |
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276 reopen_audio(AFormat fmt, gint rate, gint nch) |
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277 { |
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278 OutputPlugin *op = get_current_output_plugin(); |
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279 |
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280 if (op == NULL) |
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281 return FALSE; |
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282 |
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283 /* Is our output port already open? */ |
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284 if ((op_state.rate != 0 && op_state.nch != 0) && |
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285 (op_state.rate == rate && op_state.nch == nch && op_state.fmt == fmt)) |
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286 { |
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287 /* Yes, and it's the correct sampling rate. Reset the counter and go. */ |
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288 AUDDBG("flushing output instead of reopening\n"); |
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289 plugin_set_current((Plugin *)op); |
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290 op->flush(0); |
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291 return TRUE; |
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292 } |
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293 else if (op_state.rate != 0 && op_state.nch != 0) |
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294 { |
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295 plugin_set_current((Plugin *)op); |
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296 op->close_audio(); |
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297 } |
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298 |
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299 plugin_set_current((Plugin *)op); |
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300 gint ret = op->open_audio(fmt, rate, nch); |
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301 |
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302 if (ret == 1) /* Success? */ |
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303 { |
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304 AUDDBG("opened audio: fmt=%d, rate=%d, nch=%d\n", fmt, rate, nch); |
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305 op_state.fmt = fmt; |
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306 op_state.rate = rate; |
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307 op_state.nch = nch; |
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308 |
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309 return TRUE; |
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310 } else { |
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311 return FALSE; |
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312 } |
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313 } |
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314 |
2313 | 315 gint |
316 output_open_audio(AFormat fmt, gint rate, gint nch) | |
317 { | |
318 gint ret; | |
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319 AUDDBG("requested: fmt=%d, rate=%d, nch=%d\n", fmt, rate, nch); |
4234 | 320 |
321 AFormat output_fmt; | |
322 int bit_depth; | |
323 SAD_buffer_format input_sad_fmt; | |
324 SAD_buffer_format output_sad_fmt; | |
325 | |
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326 decoder_srate = rate; |
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327 bypass_dsp = cfg.bypass_dsp; |
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328 |
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329 if (bypass_dsp) { |
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330 AUDDBG("trying to open audio in native format\n"); |
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331 bypass_dsp = reopen_audio(fmt, rate, nch); |
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332 AUDDBG("opening in native fmt %s\n", bypass_dsp ? "succeeded" : "failed"); |
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333 } |
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334 |
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335 if (bypass_dsp) { |
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336 return TRUE; |
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337 } else { |
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338 #ifdef USE_SRC |
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339 if(cfg.enable_src) rate = cfg.src_rate; |
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340 #endif |
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341 |
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342 bit_depth = cfg.output_bit_depth; |
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344 AUDDBG("bit depth: %d\n", bit_depth); |
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345 output_fmt = (bit_depth == 24) ? FMT_S24_NE : FMT_S16_NE; |
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346 |
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347 freeSAD(); |
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348 |
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349 AUDDBG("initializing dithering engine for 2 stage conversion: fmt%d --> float -->fmt%d\n", fmt, output_fmt); |
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350 input_sad_fmt.sample_format = sadfmt_from_afmt(fmt); |
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351 if (input_sad_fmt.sample_format < 0) return FALSE; |
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352 input_sad_fmt.fracbits = FMT_FRACBITS(fmt); |
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353 input_sad_fmt.channels = nch; |
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354 input_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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355 input_sad_fmt.samplerate = 0; |
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356 |
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357 output_sad_fmt.sample_format = SAD_SAMPLE_FLOAT; |
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358 output_sad_fmt.fracbits = 0; |
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359 output_sad_fmt.channels = nch; |
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360 output_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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361 output_sad_fmt.samplerate = 0; |
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362 |
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363 sad_state_to_float = SAD_dither_init(&input_sad_fmt, &output_sad_fmt, &ret); |
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364 if (sad_state_to_float == NULL) { |
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365 AUDDBG("ditherer init failed (decoder's native --> float)\n"); |
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366 return FALSE; |
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367 } |
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368 SAD_dither_set_dither (sad_state_to_float, FALSE); |
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369 |
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370 input_sad_fmt.sample_format = SAD_SAMPLE_FLOAT; |
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371 input_sad_fmt.fracbits = 0; |
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372 input_sad_fmt.channels = nch; |
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373 input_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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374 input_sad_fmt.samplerate = 0; |
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375 |
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376 output_sad_fmt.sample_format = sadfmt_from_afmt(output_fmt); |
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377 if (output_sad_fmt.sample_format < 0) return FALSE; |
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378 output_sad_fmt.fracbits = FMT_FRACBITS(output_fmt); |
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379 output_sad_fmt.channels = nch; |
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380 output_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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381 output_sad_fmt.samplerate = 0; |
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382 |
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383 sad_state_from_float = SAD_dither_init(&input_sad_fmt, &output_sad_fmt, &ret); |
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384 if (sad_state_from_float == NULL) { |
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385 SAD_dither_free(sad_state_to_float); |
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386 AUDDBG("ditherer init failed (float --> output)\n"); |
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387 return FALSE; |
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388 } |
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389 SAD_dither_set_dither (sad_state_from_float, TRUE); |
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390 |
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391 fmt = output_fmt; |
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392 |
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393 if(replay_gain_info.album_peak == 0.0 && replay_gain_info.track_peak == 0.0) { |
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394 AUDDBG("RG info isn't set yet. Filling replay_gain_info with default values.\n"); |
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395 replay_gain_info.track_gain = cfg.default_gain; |
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396 replay_gain_info.track_peak = 0.01; |
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397 replay_gain_info.album_gain = cfg.default_gain; |
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398 replay_gain_info.album_peak = 0.01; |
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399 } |
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400 apply_replaygain_info(&replay_gain_info); |
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401 |
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402 return reopen_audio(fmt, rate, nch); |
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403 } /* bypass_dsp */ |
2313 | 404 } |
405 | |
406 void | |
407 output_write_audio(gpointer ptr, gint length) | |
408 { | |
409 OutputPlugin *op = get_current_output_plugin(); | |
410 | |
411 /* Sanity check. */ | |
412 if (op == NULL) | |
413 return; | |
414 | |
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Implemented support for multiple subplugins inside a plugin (see bug #148) and PluginHeader finalization
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415 plugin_set_current((Plugin *)op); |
2313 | 416 op->write_audio(ptr, length); |
417 } | |
418 | |
419 void | |
420 output_close_audio(void) | |
421 { | |
422 OutputPlugin *op = get_current_output_plugin(); | |
423 | |
4234 | 424 freeSAD(); |
4237
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425 |
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426 AUDDBG("clearing RG settings\n"); |
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427 replay_gain_info.track_gain = 0.0; |
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428 replay_gain_info.track_peak = 0.0; |
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429 replay_gain_info.album_gain = 0.0; |
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430 replay_gain_info.album_peak = 0.0; |
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431 |
2313 | 432 /* Do not close if there are still songs to play and the user has |
433 * not requested a stop. --nenolod | |
434 */ | |
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435 Playlist *pl = playlist_get_active(); |
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436 if (ip_data.stop == FALSE && |
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437 (playlist_get_position_nolock(pl) < playlist_get_length(pl) - 1)) { |
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438 AUDDBG("leaving audio opened\n"); |
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439 return; |
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440 } |
2313 | 441 |
442 /* Sanity check. */ | |
443 if (op == NULL) | |
444 return; | |
445 | |
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446 plugin_set_current((Plugin *)op); |
2313 | 447 op->close_audio(); |
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448 AUDDBG("done\n"); |
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449 #ifdef USE_SRC |
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450 src_flow_free(); |
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451 #endif |
2313 | 452 |
453 /* Reset the op_state. */ | |
454 op_state.fmt = op_state.rate = op_state.nch = 0; | |
455 } | |
456 | |
457 void | |
458 output_flush(gint time) | |
459 { | |
460 OutputPlugin *op = get_current_output_plugin(); | |
461 | |
462 if (op == NULL) | |
463 return; | |
464 | |
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465 plugin_set_current((Plugin *)op); |
2313 | 466 op->flush(time); |
467 } | |
468 | |
469 void | |
470 output_pause(gshort paused) | |
471 { | |
472 OutputPlugin *op = get_current_output_plugin(); | |
473 | |
474 if (op == NULL) | |
475 return; | |
476 | |
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477 plugin_set_current((Plugin *)op); |
2313 | 478 op->pause(paused); |
479 } | |
480 | |
481 gint | |
482 output_buffer_free(void) | |
483 { | |
484 OutputPlugin *op = get_current_output_plugin(); | |
485 | |
486 if (op == NULL) | |
487 return 0; | |
488 | |
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489 plugin_set_current((Plugin *)op); |
2313 | 490 return op->buffer_free(); |
491 } | |
492 | |
493 gint | |
494 output_buffer_playing(void) | |
495 { | |
496 OutputPlugin *op = get_current_output_plugin(); | |
497 | |
498 if (op == NULL) | |
499 return 0; | |
500 | |
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501 plugin_set_current((Plugin *)op); |
2313 | 502 return op->buffer_playing(); |
503 } | |
504 | |
505 /* called by input plugin when data is ready */ | |
506 void | |
3709
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507 output_pass_audio(InputPlayback *playback, |
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508 AFormat fmt, /* output format */ |
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509 gint nch, /* channels */ |
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510 gint length, /* length of sample */ |
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511 gpointer ptr, /* data */ |
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512 int *going /* 0 when time to stop */ |
2313 | 513 ) |
514 { | |
3553
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515 static Flow *postproc_flow = NULL; |
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516 static Flow *legacy_flow = NULL; |
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517 OutputPlugin *op = playback->output; |
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518 gint writeoffs; |
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519 gpointer float_ptr; |
4271 | 520 |
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521 if (!bypass_dsp) { |
2313 | 522 |
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523 if(length <= 0 || sad_state_from_float == NULL || sad_state_to_float == NULL) return; |
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524 |
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525 plugin_set_current((Plugin *)(playback->output)); |
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526 gint time = playback->output->written_time(); |
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527 |
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528 if (legacy_flow == NULL) |
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529 { |
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530 legacy_flow = flow_new(); |
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531 flow_link_element(legacy_flow, iir_flow); |
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532 flow_link_element(legacy_flow, effect_flow); |
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533 } |
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534 |
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535 if (postproc_flow == NULL) |
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536 { |
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537 postproc_flow = flow_new(); |
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538 flow_link_element(postproc_flow, vis_flow); |
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539 #ifdef USE_SRC |
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540 flow_link_element(postproc_flow, src_flow); |
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541 #endif |
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542 flow_link_element(postproc_flow, volumecontrol_flow); |
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543 } |
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544 |
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545 int frames = length / nch / FMT_SIZEOF(fmt); |
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546 int len = frames * nch * sizeof(float); |
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547 if(sad_float_buf == NULL || sad_float_buf_length < len) { |
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548 sad_float_buf_length = len; |
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549 sad_float_buf = smart_realloc(sad_float_buf, &sad_float_buf_length); |
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550 } |
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551 |
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552 SAD_dither_process_buffer(sad_state_to_float, ptr, sad_float_buf, frames); |
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553 float_ptr = sad_float_buf; |
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554 |
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555 length = flow_execute(postproc_flow, |
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556 time, |
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557 &float_ptr, |
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558 len, |
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559 FMT_FLOAT, |
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560 decoder_srate, |
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561 nch); |
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562 |
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563 frames = length / nch / sizeof(float); |
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564 len = frames * nch * FMT_SIZEOF(op_state.fmt); |
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565 if(sad_out_buf == NULL || sad_out_buf_length < len) { |
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566 sad_out_buf_length = len; |
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567 sad_out_buf = smart_realloc(sad_out_buf, &sad_out_buf_length); |
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568 } |
4234 | 569 |
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570 SAD_dither_process_buffer(sad_state_from_float, float_ptr, sad_out_buf, frames); |
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571 |
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572 length = len; |
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573 ptr = sad_out_buf; |
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574 |
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575 if (op_state.fmt == FMT_S16_NE || (op_state.fmt == FMT_S16_LE && G_BYTE_ORDER == G_LITTLE_ENDIAN) || |
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576 (op_state.fmt == FMT_S16_BE && G_BYTE_ORDER == G_BIG_ENDIAN)) { |
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577 length = flow_execute(legacy_flow, time, &ptr, length, op_state.fmt, op_state.rate, op_state.nch); |
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578 } else { |
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added passthrough mode. tested with ape and flac
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changeset
|
579 AUDDBG("legacy_flow can deal only with S16_NE streams\n"); /*FIXME*/ |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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changeset
|
580 } |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
581 } /* !bypass_dsp */ |
3546
fd4cd1945b91
Connect volumecontrol flow to the work pipeline.
William Pitcock <nenolod@atheme.org>
parents:
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diff
changeset
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582 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
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parents:
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583 /**** write it out ****/ |
a41fb6bc632a
- src stuff traveled to src_flow.c
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parents:
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diff
changeset
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584 |
2313 | 585 writeoffs = 0; |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
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586 while (writeoffs < length) |
ad1d7687814c
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587 { |
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mf0102
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588 int writable = length - writeoffs; |
2313 | 589 |
2373
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[svn] made strings.h for existing strings.c, cleanups
mf0102
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590 if (writable > 2048) |
ad1d7687814c
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mf0102
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changeset
|
591 writable = 2048; |
2313 | 592 |
2813
49f3d1b43518
[svn] - code cleanups involving SRC patch and indentation
nenolod
parents:
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diff
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|
593 if (writable == 0) |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
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|
594 return; |
2313 | 595 |
3486
96baf555b449
Get rid of "XMMS Multiple Effect Plugin support" hack. Effects processing is handled in produce_audio() now days.
William Pitcock <nenolod@atheme.org>
parents:
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diff
changeset
|
596 while (op->buffer_free() < writable) /* wait output buf */ |
96baf555b449
Get rid of "XMMS Multiple Effect Plugin support" hack. Effects processing is handled in produce_audio() now days.
William Pitcock <nenolod@atheme.org>
parents:
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diff
changeset
|
597 { |
3696
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
parents:
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598 GTimeVal pb_abs_time; |
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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parents:
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diff
changeset
|
599 |
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
parents:
3686
diff
changeset
|
600 g_get_current_time(&pb_abs_time); |
4230
b3e82a1eb129
Experimental patch to fix pulseaudio deficencies by sleeping 10000us.
William Pitcock <nenolod@atheme.org>
parents:
4186
diff
changeset
|
601 g_time_val_add(&pb_abs_time, 10000); |
3696
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
parents:
3686
diff
changeset
|
602 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
changeset
|
603 if (going && !*going) /* thread stopped? */ |
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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changeset
|
604 return; /* so finish */ |
2313 | 605 |
3486
96baf555b449
Get rid of "XMMS Multiple Effect Plugin support" hack. Effects processing is handled in produce_audio() now days.
William Pitcock <nenolod@atheme.org>
parents:
3437
diff
changeset
|
606 if (ip_data.stop) /* has a stop been requested? */ |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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changeset
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607 return; /* yes, so finish */ |
2313 | 608 |
3680
6615978ca88c
Instead of waiting 10ms, wait half the time of the period buffer.
William Pitcock <nenolod@atheme.org>
parents:
3560
diff
changeset
|
609 /* else sleep for retry */ |
3696
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
parents:
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diff
changeset
|
610 g_mutex_lock(playback->pb_change_mutex); |
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
William Pitcock <nenolod@atheme.org>
parents:
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diff
changeset
|
611 g_cond_timed_wait(playback->pb_change_cond, playback->pb_change_mutex, &pb_abs_time); |
418ac922ce87
Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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parents:
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changeset
|
612 g_mutex_unlock(playback->pb_change_mutex); |
2373
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613 } |
2313 | 614 |
2373
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[svn] made strings.h for existing strings.c, cleanups
mf0102
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615 if (ip_data.stop) |
ad1d7687814c
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616 return; |
2313 | 617 |
2373
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mf0102
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618 if (going && !*going) /* thread stopped? */ |
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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changeset
|
619 return; /* so finish */ |
2313 | 620 |
2373
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mf0102
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changeset
|
621 /* do output */ |
4266
2b7a74fce100
Implemented support for multiple subplugins inside a plugin (see bug #148) and PluginHeader finalization
stefano@zanga
parents:
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changeset
|
622 plugin_set_current((Plugin *)op); |
2373
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mf0102
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changeset
|
623 op->write_audio(((guint8 *) ptr) + writeoffs, writable); |
2313 | 624 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
changeset
|
625 writeoffs += writable; |
2313 | 626 } |
627 } | |
4235
2d4b4f13d10d
set_replaygain_info added to PAPI
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4234
diff
changeset
|
628 |
4237
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initial Replay Gain support
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diff
changeset
|
629 /* called by input plugin when RG info available --asphyx */ |
4235
2d4b4f13d10d
set_replaygain_info added to PAPI
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parents:
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diff
changeset
|
630 void |
2d4b4f13d10d
set_replaygain_info added to PAPI
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parents:
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diff
changeset
|
631 output_set_replaygain_info (InputPlayback *pb, ReplayGainInfo *rg_info) |
2d4b4f13d10d
set_replaygain_info added to PAPI
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parents:
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diff
changeset
|
632 { |
4237
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diff
changeset
|
633 replay_gain_info = *rg_info; |
8f6956130372
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diff
changeset
|
634 apply_replaygain_info(rg_info); |
4235
2d4b4f13d10d
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parents:
4234
diff
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|
635 } |
4237
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diff
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|
636 |
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parents:
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diff
changeset
|
637 static void |
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4235
diff
changeset
|
638 apply_replaygain_info (ReplayGainInfo *rg_info) |
8f6956130372
initial Replay Gain support
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parents:
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|
639 { |
8f6956130372
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parents:
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diff
changeset
|
640 SAD_replaygain_mode mode; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
641 SAD_replaygain_info info; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
642 gboolean rg_enabled; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
643 gboolean album_mode; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
644 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
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parents:
4259
diff
changeset
|
645 if(sad_state_from_float == NULL) { |
4237
8f6956130372
initial Replay Gain support
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parents:
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diff
changeset
|
646 AUDDBG("SAD not initialized!\n"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
647 return; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
648 } |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
649 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
650 rg_enabled = cfg.enable_replay_gain; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
651 album_mode = cfg.replay_gain_album; |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
652 mode.clipping_prevention = cfg.enable_clipping_prevention; |
4256
b0ca963fd965
adaptive scaler added, disabled hard limiter
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4247
diff
changeset
|
653 mode.hard_limit = FALSE; |
b0ca963fd965
adaptive scaler added, disabled hard limiter
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4247
diff
changeset
|
654 mode.adaptive_scaler = cfg.enable_adaptive_scaler; |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
655 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
656 if(!rg_enabled) return; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
657 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
658 mode.mode = album_mode ? SAD_RG_ALBUM : SAD_RG_TRACK; |
4242
21008f43bb93
RG improved. It f... works.
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4240
diff
changeset
|
659 mode.preamp = cfg.replay_gain_preamp; |
4237
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
660 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
661 info.present = TRUE; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
662 info.track_gain = rg_info->track_gain; |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
663 info.track_peak = rg_info->track_peak; |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
664 info.album_gain = rg_info->album_gain; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
665 info.album_peak = rg_info->album_peak; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
666 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
667 AUDDBG("Applying Replay Gain settings:\n"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
668 AUDDBG("* mode: %s\n", mode.mode == SAD_RG_ALBUM ? "album" : "track"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
669 AUDDBG("* clipping prevention: %s\n", mode.clipping_prevention ? "yes" : "no"); |
4256
b0ca963fd965
adaptive scaler added, disabled hard limiter
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4247
diff
changeset
|
670 AUDDBG("* adaptive scaler %s\n", mode.adaptive_scaler ? "yes" : "no"); |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
671 AUDDBG("* preamp: %+f dB\n", mode.preamp); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
672 AUDDBG("Replay Gain info for current track:\n"); |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
673 AUDDBG("* track gain: %+f dB\n", info.track_gain); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
674 AUDDBG("* track peak: %f\n", info.track_peak); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
675 AUDDBG("* album gain: %+f dB\n", info.album_gain); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
676 AUDDBG("* album peak: %f\n", info.album_peak); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
677 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4259
diff
changeset
|
678 SAD_dither_apply_replaygain(sad_state_from_float, &info, &mode); |
4237
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
679 } |