Mercurial > audlegacy
annotate src/audacious/output.c @ 4779:aa79a84627b9
g_basename is deprecated, replace with g_path_get_basename.
author | Tony Vroon <chainsaw@gentoo.org> |
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date | Fri, 26 Sep 2008 21:02:49 +0100 |
parents | f084f639e962 |
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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 "main.h" | |
34 #include "input.h" | |
2328 | 35 #include "playback.h" |
2313 | 36 |
37 #include "playlist.h" | |
2717 | 38 #include "configdb.h" |
2542 | 39 |
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40 #include "flow.h" |
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41 |
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42 #include "pluginenum.h" |
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43 |
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44 #include "effect.h" |
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45 #include "volumecontrol.h" |
3559 | 46 #include "visualization.h" |
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47 |
4234 | 48 #include "libSAD.h" |
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49 #include "util.h" |
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50 #include "equalizer_flow.h" |
4234 | 51 |
2542 | 52 #include <math.h> |
53 | |
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54 #ifdef USE_SAMPLERATE |
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55 # include "src_flow.h" |
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56 #endif |
2313 | 57 |
58 OutputPluginData op_data = { | |
59 NULL, | |
60 NULL | |
61 }; | |
62 | |
63 OutputPluginState op_state = { | |
64 0, | |
65 0, | |
66 0 | |
67 }; | |
68 | |
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69 static gint decoder_srate = 0; |
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70 static gboolean bypass_dsp = FALSE; |
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71 |
2313 | 72 OutputPlugin psuedo_output_plugin = { |
3549 | 73 .description = "XMMS reverse compatibility output plugin", |
74 .get_volume = output_get_volume, | |
75 .set_volume = output_set_volume, | |
76 | |
77 .open_audio = output_open_audio, | |
78 .write_audio = output_write_audio, | |
79 .close_audio = output_close_audio, | |
2313 | 80 |
3549 | 81 .flush = output_flush, |
82 .pause = output_pause, | |
83 | |
84 .buffer_free = output_buffer_free, | |
85 .buffer_playing = output_buffer_playing, | |
86 .output_time = get_output_time, | |
87 .written_time = get_written_time, | |
2313 | 88 }; |
89 | |
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90 static void apply_replaygain_info (ReplayGainInfo *rg_info); |
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91 |
2313 | 92 OutputPlugin * |
93 get_current_output_plugin(void) | |
94 { | |
95 return op_data.current_output_plugin; | |
96 } | |
97 | |
98 void | |
99 set_current_output_plugin(gint i) | |
100 { | |
101 gboolean playing; | |
102 OutputPlugin *op = get_current_output_plugin(); | |
103 | |
104 GList *node = g_list_nth(op_data.output_list, i); | |
105 if (!node) { | |
106 op_data.current_output_plugin = NULL; | |
107 return; | |
108 } | |
109 | |
110 op_data.current_output_plugin = node->data; | |
111 | |
112 playing = playback_get_playing(); | |
113 | |
114 if (playing == TRUE) | |
115 { | |
116 guint time, pos; | |
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117 PlaylistEntry *entry; |
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118 |
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119 plugin_set_current((Plugin *)op); |
2313 | 120 |
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121 /* don't stop yet, get the seek time and playlist position first */ |
2313 | 122 pos = playlist_get_position(playlist_get_active()); |
123 time = op->output_time(); | |
124 | |
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125 /* reset the audio system */ |
2313 | 126 mainwin_stop_pushed(); |
127 op->close_audio(); | |
128 | |
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129 g_usleep(300000); |
2313 | 130 |
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131 /* wait for the playback thread to come online */ |
2313 | 132 while (playback_get_playing()) |
133 g_message("waiting for audio system shutdown..."); | |
134 | |
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135 /* wait for the playback thread to come online */ |
2313 | 136 playlist_set_position(playlist_get_active(), pos); |
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137 entry = playlist_get_entry_to_play(playlist_get_active()); |
2313 | 138 playback_play_file(entry); |
139 | |
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140 while (!playback_get_playing()) |
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141 { |
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142 gtk_main_iteration(); |
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143 g_message("waiting for audio system startup..."); |
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144 } |
2313 | 145 |
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146 /* and signal a reseek */ |
2313 | 147 if (playlist_get_current_length(playlist_get_active()) > -1 && |
148 time <= (playlist_get_current_length(playlist_get_active()))) | |
149 playback_seek(time / 1000); | |
150 } | |
151 } | |
152 | |
153 GList * | |
154 get_output_list(void) | |
155 { | |
156 return op_data.output_list; | |
157 } | |
158 | |
159 void | |
160 output_about(gint i) | |
161 { | |
162 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
163 if (out && out->about) | |
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164 { |
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165 plugin_set_current((Plugin *)out); |
2313 | 166 out->about(); |
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167 } |
2313 | 168 } |
169 | |
170 void | |
171 output_configure(gint i) | |
172 { | |
173 OutputPlugin *out = g_list_nth(op_data.output_list, i)->data; | |
174 if (out && out->configure) | |
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175 { |
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176 plugin_set_current((Plugin *)out); |
2313 | 177 out->configure(); |
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178 } |
2313 | 179 } |
180 | |
181 void | |
182 output_get_volume(gint * l, gint * r) | |
183 { | |
184 *l = *r = -1; | |
185 | |
186 if (!op_data.current_output_plugin) | |
187 return; | |
188 | |
189 if (!op_data.current_output_plugin->get_volume) | |
190 return; | |
191 | |
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192 if (cfg.software_volume_control) |
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193 volumecontrol_get_volume_state(l, r); |
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194 else |
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195 { |
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196 plugin_set_current((Plugin *)op_data.current_output_plugin); |
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197 op_data.current_output_plugin->get_volume(l, r); |
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198 } |
2313 | 199 } |
200 | |
201 void | |
202 output_set_volume(gint l, gint r) | |
203 { | |
204 if (!op_data.current_output_plugin) | |
205 return; | |
206 | |
207 if (!op_data.current_output_plugin->set_volume) | |
208 return; | |
209 | |
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210 if (cfg.software_volume_control) |
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211 volumecontrol_set_volume_state(l, r); |
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212 else |
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213 { |
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214 plugin_set_current((Plugin *)op_data.current_output_plugin); |
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215 op_data.current_output_plugin->set_volume(l, r); |
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216 } |
2313 | 217 } |
218 | |
219 void | |
220 output_set_eq(gboolean active, gfloat pre, gfloat * bands) | |
221 { | |
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222 AUDDBG("preamp: %f, bands: %f:%f:%f:%f:%f:%f:%f:%f:%f:%f\n", pre, bands[0], bands[1], bands[2], bands[3], bands[4], |
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223 bands[5], bands[6], bands[7], bands[8], bands[9]); |
2313 | 224 |
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225 equalizer_flow_set_bands(pre, bands); |
2313 | 226 } |
227 | |
228 /* called by input plugin to peek at the output plugin's write progress */ | |
229 gint | |
230 get_written_time(void) | |
231 { | |
232 OutputPlugin *op = get_current_output_plugin(); | |
233 | |
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234 plugin_set_current((Plugin *)op); |
2313 | 235 return op->written_time(); |
236 } | |
237 | |
238 /* called by input plugin to peek at the output plugin's output progress */ | |
239 gint | |
240 get_output_time(void) | |
241 { | |
242 OutputPlugin *op = get_current_output_plugin(); | |
243 | |
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244 plugin_set_current((Plugin *)op); |
2313 | 245 return op->output_time(); |
246 } | |
247 | |
4234 | 248 static SAD_dither_t *sad_state_to_float = NULL; |
249 static SAD_dither_t *sad_state_from_float = NULL; | |
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250 static float *sad_float_buf = NULL; |
4234 | 251 static void *sad_out_buf = NULL; |
4278 | 252 static gsize sad_float_buf_length = 0; |
253 static gsize sad_out_buf_length = 0; | |
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254 static ReplayGainInfo replay_gain_info = { |
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255 .track_gain = 0.0, |
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256 .track_peak = 0.0, |
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257 .album_gain = 0.0, |
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258 .album_peak = 0.0, |
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259 }; |
4234 | 260 |
261 static void | |
262 freeSAD() | |
263 { | |
264 if (sad_state_from_float != NULL) {SAD_dither_free(sad_state_from_float); sad_state_from_float = NULL;} | |
265 if (sad_state_to_float != NULL) {SAD_dither_free(sad_state_to_float); sad_state_to_float = NULL;} | |
266 } | |
267 | |
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268 static gboolean |
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269 reopen_audio(AFormat fmt, gint rate, gint nch) |
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270 { |
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271 OutputPlugin *op = get_current_output_plugin(); |
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272 |
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273 if (op == NULL) |
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274 return FALSE; |
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275 |
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276 /* Is our output port already open? */ |
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277 if ((op_state.rate != 0 && op_state.nch != 0) && |
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278 (op_state.rate == rate && op_state.nch == nch && op_state.fmt == fmt)) |
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279 { |
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280 /* Yes, and it's the correct sampling rate. Reset the counter and go. */ |
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281 AUDDBG("flushing output instead of reopening\n"); |
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282 plugin_set_current((Plugin *)op); |
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283 op->flush(0); |
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284 return TRUE; |
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285 } |
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286 else if (op_state.rate != 0 && op_state.nch != 0) |
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287 { |
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288 plugin_set_current((Plugin *)op); |
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289 op->close_audio(); |
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290 op_state.fmt = 0; |
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291 op_state.rate = 0; |
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292 op_state.nch = 0; |
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293 } |
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294 |
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295 plugin_set_current((Plugin *)op); |
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296 gint ret = op->open_audio(fmt, rate, nch); |
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297 |
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298 if (ret == 1) /* Success? */ |
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299 { |
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300 AUDDBG("opened audio: fmt=%d, rate=%d, nch=%d\n", fmt, rate, nch); |
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301 op_state.fmt = fmt; |
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302 op_state.rate = rate; |
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303 op_state.nch = nch; |
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304 |
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305 return TRUE; |
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306 } else { |
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307 return FALSE; |
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308 } |
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309 } |
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310 |
2313 | 311 gint |
312 output_open_audio(AFormat fmt, gint rate, gint nch) | |
313 { | |
314 gint ret; | |
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315 AUDDBG("requested: fmt=%d, rate=%d, nch=%d\n", fmt, rate, nch); |
4234 | 316 |
317 AFormat output_fmt; | |
318 int bit_depth; | |
319 SAD_buffer_format input_sad_fmt; | |
320 SAD_buffer_format output_sad_fmt; | |
321 | |
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322 decoder_srate = rate; |
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323 bypass_dsp = cfg.bypass_dsp; |
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324 |
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325 if (bypass_dsp) { |
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326 AUDDBG("trying to open audio in native format\n"); |
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327 bypass_dsp = reopen_audio(fmt, rate, nch); |
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328 AUDDBG("opening in native fmt %s\n", bypass_dsp ? "succeeded" : "failed"); |
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329 } |
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330 |
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331 if (bypass_dsp) { |
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332 return TRUE; |
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333 } else { |
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334 #ifdef USE_SAMPLERATE |
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335 rate = src_flow_init(rate, nch); /* returns sample rate unchanged if resampling switched off */ |
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336 #endif |
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337 |
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338 bit_depth = cfg.output_bit_depth; |
4234 | 339 |
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340 AUDDBG("bit depth: %d\n", bit_depth); |
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341 output_fmt = (bit_depth == 24) ? FMT_S24_NE : FMT_S16_NE; |
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342 |
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343 freeSAD(); |
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344 |
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345 AUDDBG("initializing dithering engine for 2 stage conversion: fmt%d --> float -->fmt%d\n", fmt, output_fmt); |
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346 input_sad_fmt.sample_format = sadfmt_from_afmt(fmt); |
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347 if (input_sad_fmt.sample_format < 0) return FALSE; |
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348 input_sad_fmt.fracbits = FMT_FRACBITS(fmt); |
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349 input_sad_fmt.channels = nch; |
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350 input_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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351 input_sad_fmt.samplerate = 0; |
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352 |
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353 output_sad_fmt.sample_format = SAD_SAMPLE_FLOAT; |
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354 output_sad_fmt.fracbits = 0; |
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355 output_sad_fmt.channels = nch; |
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356 output_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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357 output_sad_fmt.samplerate = 0; |
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358 |
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359 sad_state_to_float = SAD_dither_init(&input_sad_fmt, &output_sad_fmt, &ret); |
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360 if (sad_state_to_float == NULL) { |
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361 AUDDBG("ditherer init failed (decoder's native --> float)\n"); |
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362 return FALSE; |
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363 } |
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364 SAD_dither_set_dither (sad_state_to_float, FALSE); |
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365 |
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366 input_sad_fmt.sample_format = SAD_SAMPLE_FLOAT; |
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367 input_sad_fmt.fracbits = 0; |
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368 input_sad_fmt.channels = nch; |
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369 input_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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370 input_sad_fmt.samplerate = 0; |
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371 |
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372 output_sad_fmt.sample_format = sadfmt_from_afmt(output_fmt); |
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373 if (output_sad_fmt.sample_format < 0) return FALSE; |
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374 output_sad_fmt.fracbits = FMT_FRACBITS(output_fmt); |
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375 output_sad_fmt.channels = nch; |
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376 output_sad_fmt.channels_order = SAD_CHORDER_INTERLEAVED; |
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377 output_sad_fmt.samplerate = 0; |
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378 |
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379 sad_state_from_float = SAD_dither_init(&input_sad_fmt, &output_sad_fmt, &ret); |
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380 if (sad_state_from_float == NULL) { |
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381 SAD_dither_free(sad_state_to_float); |
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382 AUDDBG("ditherer init failed (float --> output)\n"); |
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383 return FALSE; |
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384 } |
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385 SAD_dither_set_dither (sad_state_from_float, TRUE); |
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386 |
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387 fmt = output_fmt; |
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388 |
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389 if(replay_gain_info.album_peak == 0.0 && replay_gain_info.track_peak == 0.0) { |
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390 AUDDBG("RG info isn't set yet. Filling replay_gain_info with default values.\n"); |
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|
391 replay_gain_info.track_gain = cfg.default_gain; |
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392 replay_gain_info.track_peak = 0.01; |
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393 replay_gain_info.album_gain = cfg.default_gain; |
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394 replay_gain_info.album_peak = 0.01; |
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395 } |
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396 apply_replaygain_info(&replay_gain_info); |
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397 |
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398 return reopen_audio(fmt, rate, nch); |
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399 } /* bypass_dsp */ |
2313 | 400 } |
401 | |
402 void | |
403 output_write_audio(gpointer ptr, gint length) | |
404 { | |
405 OutputPlugin *op = get_current_output_plugin(); | |
406 | |
407 /* Sanity check. */ | |
408 if (op == NULL) | |
409 return; | |
410 | |
4266
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Implemented support for multiple subplugins inside a plugin (see bug #148) and PluginHeader finalization
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|
411 plugin_set_current((Plugin *)op); |
2313 | 412 op->write_audio(ptr, length); |
413 } | |
414 | |
415 void | |
416 output_close_audio(void) | |
417 { | |
418 OutputPlugin *op = get_current_output_plugin(); | |
419 | |
4234 | 420 freeSAD(); |
4237
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421 |
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422 AUDDBG("clearing RG settings\n"); |
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423 replay_gain_info.track_gain = 0.0; |
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424 replay_gain_info.track_peak = 0.0; |
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425 replay_gain_info.album_gain = 0.0; |
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|
426 replay_gain_info.album_peak = 0.0; |
2538
719e0898ff3c
[svn] - Patch to use libsamplerate for high-quality interpolated upsampling,
nenolod
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diff
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|
427 |
4523
b3e4f5c31546
Conditional USE_SRC was changed to USE_SAMPLERATE earlier.
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parents:
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diff
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|
428 #ifdef USE_SAMPLERATE |
4290
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429 src_flow_free(); |
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430 #endif |
2313 | 431 /* Do not close if there are still songs to play and the user has |
432 * not requested a stop. --nenolod | |
433 */ | |
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434 Playlist *pl = playlist_get_active(); |
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435 if (ip_data.stop == FALSE && |
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diff
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436 (playlist_get_position_nolock(pl) < playlist_get_length(pl) - 1)) { |
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437 AUDDBG("leaving audio opened\n"); |
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438 return; |
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439 } |
2313 | 440 |
441 /* Sanity check. */ | |
442 if (op == NULL) | |
443 return; | |
444 | |
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|
445 plugin_set_current((Plugin *)op); |
2313 | 446 op->close_audio(); |
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447 AUDDBG("done\n"); |
2313 | 448 |
449 /* Reset the op_state. */ | |
450 op_state.fmt = op_state.rate = op_state.nch = 0; | |
4299
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- 1st attempt to integrate MPlayer's equalizer (will work in 24bit mode too).
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|
451 equalizer_flow_free(); |
2313 | 452 } |
453 | |
454 void | |
455 output_flush(gint time) | |
456 { | |
457 OutputPlugin *op = get_current_output_plugin(); | |
458 | |
459 if (op == NULL) | |
460 return; | |
461 | |
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462 plugin_set_current((Plugin *)op); |
2313 | 463 op->flush(time); |
464 } | |
465 | |
466 void | |
467 output_pause(gshort paused) | |
468 { | |
469 OutputPlugin *op = get_current_output_plugin(); | |
470 | |
471 if (op == NULL) | |
472 return; | |
473 | |
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|
474 plugin_set_current((Plugin *)op); |
2313 | 475 op->pause(paused); |
476 } | |
477 | |
478 gint | |
479 output_buffer_free(void) | |
480 { | |
481 OutputPlugin *op = get_current_output_plugin(); | |
482 | |
483 if (op == NULL) | |
484 return 0; | |
485 | |
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|
486 plugin_set_current((Plugin *)op); |
2313 | 487 return op->buffer_free(); |
488 } | |
489 | |
490 gint | |
491 output_buffer_playing(void) | |
492 { | |
493 OutputPlugin *op = get_current_output_plugin(); | |
494 | |
495 if (op == NULL) | |
496 return 0; | |
497 | |
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498 plugin_set_current((Plugin *)op); |
2313 | 499 return op->buffer_playing(); |
500 } | |
501 | |
502 /* called by input plugin when data is ready */ | |
503 void | |
3709
a89a12aa4f2c
Add InputPlayback::pass_audio() which replaces produce_audio().
William Pitcock <nenolod@atheme.org>
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|
504 output_pass_audio(InputPlayback *playback, |
3696
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Use a mutex/condvar/timed wait to allow tickless operation in the output loop.
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505 AFormat fmt, /* output format */ |
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506 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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507 gint length, /* length of sample */ |
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508 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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|
509 int *going /* 0 when time to stop */ |
2313 | 510 ) |
511 { | |
4323
04ade98fafa3
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512 static Flow *visualization_flow = NULL; |
3553
a140fadd741d
software volumecontrol->flow manager API
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3549
diff
changeset
|
513 static Flow *postproc_flow = NULL; |
4259
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|
514 static Flow *legacy_flow = NULL; |
3696
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|
515 OutputPlugin *op = playback->output; |
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516 gint writeoffs; |
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517 gpointer float_ptr; |
4323
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518 |
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519 if (visualization_flow == NULL) |
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520 { |
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521 visualization_flow = flow_new(); |
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522 flow_link_element(visualization_flow, vis_flow); |
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523 } |
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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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changeset
|
527 |
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changeset
|
528 flow_execute(visualization_flow, time, &ptr, length, fmt, decoder_srate, nch); |
4271 | 529 |
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changeset
|
530 if (!bypass_dsp) { |
2313 | 531 |
4286
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changeset
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532 if(length <= 0 || sad_state_from_float == NULL || sad_state_to_float == NULL) return; |
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changeset
|
533 |
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diff
changeset
|
534 if (legacy_flow == NULL) |
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changeset
|
535 { |
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changeset
|
536 legacy_flow = flow_new(); |
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changeset
|
537 flow_link_element(legacy_flow, effect_flow); |
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changeset
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538 } |
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changeset
|
539 |
af324ff0a983
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diff
changeset
|
540 if (postproc_flow == NULL) |
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diff
changeset
|
541 { |
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changeset
|
542 postproc_flow = flow_new(); |
4523
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Conditional USE_SRC was changed to USE_SAMPLERATE earlier.
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diff
changeset
|
543 #ifdef USE_SAMPLERATE |
4286
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changeset
|
544 flow_link_element(postproc_flow, src_flow); |
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changeset
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545 #endif |
4299
a16edefb8836
- 1st attempt to integrate MPlayer's equalizer (will work in 24bit mode too).
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diff
changeset
|
546 flow_link_element(postproc_flow, equalizer_flow); |
4286
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changeset
|
547 flow_link_element(postproc_flow, volumecontrol_flow); |
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changeset
|
548 } |
4267
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changeset
|
549 |
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changeset
|
550 int frames = length / nch / FMT_SIZEOF(fmt); |
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changeset
|
551 int len = frames * nch * sizeof(float); |
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changeset
|
552 if(sad_float_buf == NULL || sad_float_buf_length < len) { |
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changeset
|
553 sad_float_buf_length = len; |
af324ff0a983
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changeset
|
554 sad_float_buf = smart_realloc(sad_float_buf, &sad_float_buf_length); |
af324ff0a983
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changeset
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555 } |
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changeset
|
556 |
4286
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changeset
|
557 SAD_dither_process_buffer(sad_state_to_float, ptr, sad_float_buf, frames); |
af324ff0a983
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changeset
|
558 float_ptr = sad_float_buf; |
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added passthrough mode. tested with ape and flac
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changeset
|
559 |
af324ff0a983
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changeset
|
560 length = flow_execute(postproc_flow, |
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changeset
|
561 time, |
af324ff0a983
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changeset
|
562 &float_ptr, |
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changeset
|
563 len, |
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changeset
|
564 FMT_FLOAT, |
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changeset
|
565 decoder_srate, |
af324ff0a983
added passthrough mode. tested with ape and flac
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changeset
|
566 nch); |
af324ff0a983
added passthrough mode. tested with ape and flac
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changeset
|
567 |
af324ff0a983
added passthrough mode. tested with ape and flac
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diff
changeset
|
568 frames = length / nch / sizeof(float); |
af324ff0a983
added passthrough mode. tested with ape and flac
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changeset
|
569 len = frames * nch * FMT_SIZEOF(op_state.fmt); |
af324ff0a983
added passthrough mode. tested with ape and flac
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changeset
|
570 if(sad_out_buf == NULL || sad_out_buf_length < len) { |
af324ff0a983
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changeset
|
571 sad_out_buf_length = len; |
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changeset
|
572 sad_out_buf = smart_realloc(sad_out_buf, &sad_out_buf_length); |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
573 } |
4234 | 574 |
4286
af324ff0a983
added passthrough mode. tested with ape and flac
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diff
changeset
|
575 SAD_dither_process_buffer(sad_state_from_float, float_ptr, sad_out_buf, frames); |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
576 |
af324ff0a983
added passthrough mode. tested with ape and flac
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
577 length = len; |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
578 ptr = sad_out_buf; |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
579 |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
580 if (op_state.fmt == FMT_S16_NE || (op_state.fmt == FMT_S16_LE && G_BYTE_ORDER == G_LITTLE_ENDIAN) || |
af324ff0a983
added passthrough mode. tested with ape and flac
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
581 (op_state.fmt == FMT_S16_BE && G_BYTE_ORDER == G_BIG_ENDIAN)) { |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
582 length = flow_execute(legacy_flow, time, &ptr, length, op_state.fmt, op_state.rate, op_state.nch); |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
583 } else { |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
584 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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diff
changeset
|
585 } |
af324ff0a983
added passthrough mode. tested with ape and flac
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parents:
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diff
changeset
|
586 } /* !bypass_dsp */ |
3546
fd4cd1945b91
Connect volumecontrol flow to the work pipeline.
William Pitcock <nenolod@atheme.org>
parents:
3486
diff
changeset
|
587 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
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parents:
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diff
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|
588 /**** write it out ****/ |
a41fb6bc632a
- src stuff traveled to src_flow.c
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parents:
4259
diff
changeset
|
589 |
2313 | 590 writeoffs = 0; |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
changeset
|
591 while (writeoffs < length) |
ad1d7687814c
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|
592 { |
ad1d7687814c
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mf0102
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changeset
|
593 int writable = length - writeoffs; |
2313 | 594 |
2373
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mf0102
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changeset
|
595 if (writable > 2048) |
ad1d7687814c
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mf0102
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changeset
|
596 writable = 2048; |
2313 | 597 |
2813
49f3d1b43518
[svn] - code cleanups involving SRC patch and indentation
nenolod
parents:
2811
diff
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|
598 if (writable == 0) |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
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|
599 return; |
2313 | 600 |
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
|
601 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
|
602 { |
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
|
603 GTimeVal pb_abs_time; |
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
|
604 |
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
|
605 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
|
606 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
|
607 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
changeset
|
608 if (going && !*going) /* thread stopped? */ |
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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changeset
|
609 return; /* so finish */ |
2313 | 610 |
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
|
611 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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diff
changeset
|
612 return; /* yes, so finish */ |
2313 | 613 |
3680
6615978ca88c
Instead of waiting 10ms, wait half the time of the period buffer.
William Pitcock <nenolod@atheme.org>
parents:
3560
diff
changeset
|
614 /* 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:
3686
diff
changeset
|
615 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
|
616 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.
William Pitcock <nenolod@atheme.org>
parents:
3686
diff
changeset
|
617 g_mutex_unlock(playback->pb_change_mutex); |
2373
ad1d7687814c
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mf0102
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|
618 } |
2313 | 619 |
2373
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[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
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|
620 if (ip_data.stop) |
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
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diff
changeset
|
621 return; |
2313 | 622 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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diff
changeset
|
623 if (going && !*going) /* thread stopped? */ |
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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diff
changeset
|
624 return; /* so finish */ |
2313 | 625 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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diff
changeset
|
626 /* do output */ |
4266
2b7a74fce100
Implemented support for multiple subplugins inside a plugin (see bug #148) and PluginHeader finalization
stefano@zanga
parents:
4259
diff
changeset
|
627 plugin_set_current((Plugin *)op); |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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diff
changeset
|
628 op->write_audio(((guint8 *) ptr) + writeoffs, writable); |
2313 | 629 |
2373
ad1d7687814c
[svn] made strings.h for existing strings.c, cleanups
mf0102
parents:
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diff
changeset
|
630 writeoffs += writable; |
2313 | 631 } |
632 } | |
4235
2d4b4f13d10d
set_replaygain_info added to PAPI
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4234
diff
changeset
|
633 |
4237
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initial Replay Gain support
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parents:
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diff
changeset
|
634 /* called by input plugin when RG info available --asphyx */ |
4235
2d4b4f13d10d
set_replaygain_info added to PAPI
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4234
diff
changeset
|
635 void |
2d4b4f13d10d
set_replaygain_info added to PAPI
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parents:
4234
diff
changeset
|
636 output_set_replaygain_info (InputPlayback *pb, ReplayGainInfo *rg_info) |
2d4b4f13d10d
set_replaygain_info added to PAPI
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4234
diff
changeset
|
637 { |
4237
8f6956130372
initial Replay Gain support
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parents:
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diff
changeset
|
638 replay_gain_info = *rg_info; |
8f6956130372
initial Replay Gain support
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parents:
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diff
changeset
|
639 apply_replaygain_info(rg_info); |
4235
2d4b4f13d10d
set_replaygain_info added to PAPI
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4234
diff
changeset
|
640 } |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
641 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
642 static void |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
643 apply_replaygain_info (ReplayGainInfo *rg_info) |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
644 { |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
645 SAD_replaygain_mode mode; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
646 SAD_replaygain_info info; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
647 gboolean rg_enabled; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
648 gboolean album_mode; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
649 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4259
diff
changeset
|
650 if(sad_state_from_float == NULL) { |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
651 AUDDBG("SAD not initialized!\n"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
652 return; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
653 } |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
654 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
655 rg_enabled = cfg.enable_replay_gain; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
656 album_mode = cfg.replay_gain_album; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
657 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
|
658 mode.hard_limit = FALSE; |
b0ca963fd965
adaptive scaler added, disabled hard limiter
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4247
diff
changeset
|
659 mode.adaptive_scaler = cfg.enable_adaptive_scaler; |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
660 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
661 if(!rg_enabled) return; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
662 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
663 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
|
664 mode.preamp = cfg.replay_gain_preamp; |
4237
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
665 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
666 info.present = TRUE; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
667 info.track_gain = rg_info->track_gain; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
668 info.track_peak = rg_info->track_peak; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
669 info.album_gain = rg_info->album_gain; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
670 info.album_peak = rg_info->album_peak; |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
671 |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
672 AUDDBG("Applying Replay Gain settings:\n"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
673 AUDDBG("* mode: %s\n", mode.mode == SAD_RG_ALBUM ? "album" : "track"); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
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diff
changeset
|
674 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
|
675 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
|
676 AUDDBG("* preamp: %+f dB\n", mode.preamp); |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
677 AUDDBG("Replay Gain info for current track:\n"); |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
678 AUDDBG("* track gain: %+f dB\n", info.track_gain); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
679 AUDDBG("* track peak: %f\n", info.track_peak); |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
680 AUDDBG("* album gain: %+f dB\n", info.album_gain); |
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
681 AUDDBG("* album peak: %f\n", info.album_peak); |
8f6956130372
initial Replay Gain support
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4235
diff
changeset
|
682 |
4267
a41fb6bc632a
- src stuff traveled to src_flow.c
Eugene Zagidullin <e.asphyx@gmail.com>
parents:
4259
diff
changeset
|
683 SAD_dither_apply_replaygain(sad_state_from_float, &info, &mode); |
4237
8f6956130372
initial Replay Gain support
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parents:
4235
diff
changeset
|
684 } |
4755
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
685 |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
686 void output_plugin_cleanup(void) |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
687 { |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
688 OutputPlugin *op = get_current_output_plugin(); |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
689 op->init(); |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
690 output_close_audio(); |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
691 printf("output plugin cleanupn\n"); |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
692 } |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
693 void output_plugin_reinit(void) |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
694 { |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
695 |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
|
696 printf("output plugin reinit \n"); |
c6f352d25d27
makeing possible to reinitialize the output plugin from outher plugins
Paula Stanciu <paula.stanciu@gmail.com>
parents:
4523
diff
changeset
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697 } |