Add encoder support for Dirac using the Schroedinger library.
[vlc/asuraparaju-public.git] / src / audio_output / output.c
blob1bfb2d92a058ae94c8ef2ee13942acecf99ee563
1 /*****************************************************************************
2 * output.c : internal management of output streams for the audio output
3 *****************************************************************************
4 * Copyright (C) 2002-2004 the VideoLAN team
5 * $Id$
7 * Authors: Christophe Massiot <massiot@via.ecp.fr>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
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.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
25 * Preamble
26 *****************************************************************************/
27 #ifdef HAVE_CONFIG_H
28 # include "config.h"
29 #endif
31 #include <vlc_common.h>
32 #include <vlc_aout.h>
33 #include <vlc_cpu.h>
34 #include <vlc_modules.h>
36 #include "aout_internal.h"
38 /*****************************************************************************
39 * aout_OutputNew : allocate a new output and rework the filter pipeline
40 *****************************************************************************
41 * This function is entered with the mixer lock.
42 *****************************************************************************/
43 int aout_OutputNew( aout_instance_t * p_aout,
44 audio_sample_format_t * p_format )
46 /* Retrieve user defaults. */
47 int i_rate = var_InheritInteger( p_aout, "aout-rate" );
48 vlc_value_t val, text;
49 /* kludge to avoid a fpu error when rate is 0... */
50 if( i_rate == 0 ) i_rate = -1;
52 memcpy( &p_aout->output.output, p_format, sizeof(audio_sample_format_t) );
53 if ( i_rate != -1 )
54 p_aout->output.output.i_rate = i_rate;
55 aout_FormatPrepare( &p_aout->output.output );
57 /* Find the best output plug-in. */
58 p_aout->output.p_module = module_need( p_aout, "audio output", "$aout", false );
59 if ( p_aout->output.p_module == NULL )
61 msg_Err( p_aout, "no suitable audio output module" );
62 return -1;
65 if ( var_Type( p_aout, "audio-channels" ) ==
66 (VLC_VAR_INTEGER | VLC_VAR_HASCHOICE) )
68 /* The user may have selected a different channels configuration. */
69 var_Get( p_aout, "audio-channels", &val );
71 if ( val.i_int == AOUT_VAR_CHAN_RSTEREO )
73 p_aout->output.output.i_original_channels |=
74 AOUT_CHAN_REVERSESTEREO;
76 else if ( val.i_int == AOUT_VAR_CHAN_STEREO )
78 p_aout->output.output.i_original_channels =
79 AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
81 else if ( val.i_int == AOUT_VAR_CHAN_LEFT )
83 p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
85 else if ( val.i_int == AOUT_VAR_CHAN_RIGHT )
87 p_aout->output.output.i_original_channels = AOUT_CHAN_RIGHT;
89 else if ( val.i_int == AOUT_VAR_CHAN_DOLBYS )
91 p_aout->output.output.i_original_channels
92 = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_DOLBYSTEREO;
95 else if ( p_aout->output.output.i_physical_channels == AOUT_CHAN_CENTER
96 && (p_aout->output.output.i_original_channels
97 & AOUT_CHAN_PHYSMASK) == (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT) )
99 /* Mono - create the audio-channels variable. */
100 var_Create( p_aout, "audio-channels",
101 VLC_VAR_INTEGER | VLC_VAR_HASCHOICE );
102 text.psz_string = _("Audio Channels");
103 var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL );
105 val.i_int = AOUT_VAR_CHAN_STEREO; text.psz_string = _("Stereo");
106 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
107 val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left");
108 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
109 val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right");
110 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
111 if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO )
113 /* Go directly to the left channel. */
114 p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
115 val.i_int = AOUT_VAR_CHAN_LEFT;
116 var_Set( p_aout, "audio-channels", val );
118 var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
119 NULL );
121 else if ( p_aout->output.output.i_physical_channels ==
122 (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)
123 && (p_aout->output.output.i_original_channels &
124 (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
126 /* Stereo - create the audio-channels variable. */
127 var_Create( p_aout, "audio-channels",
128 VLC_VAR_INTEGER | VLC_VAR_HASCHOICE );
129 text.psz_string = _("Audio Channels");
130 var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL );
132 if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DOLBYSTEREO )
134 val.i_int = AOUT_VAR_CHAN_DOLBYS;
135 text.psz_string = _("Dolby Surround");
137 else
139 val.i_int = AOUT_VAR_CHAN_STEREO;
140 text.psz_string = _("Stereo");
142 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
143 val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left");
144 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
145 val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right");
146 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
147 val.i_int = AOUT_VAR_CHAN_RSTEREO; text.psz_string=_("Reverse stereo");
148 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
149 if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO )
151 /* Go directly to the left channel. */
152 p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
153 val.i_int = AOUT_VAR_CHAN_LEFT;
154 var_Set( p_aout, "audio-channels", val );
156 var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
157 NULL );
159 val.b_bool = true;
160 var_Set( p_aout, "intf-change", val );
162 aout_FormatPrepare( &p_aout->output.output );
164 aout_lock_output_fifo( p_aout );
166 /* Prepare FIFO. */
167 aout_FifoInit( p_aout, &p_aout->output.fifo,
168 p_aout->output.output.i_rate );
170 aout_unlock_output_fifo( p_aout );
172 aout_FormatPrint( p_aout, "output", &p_aout->output.output );
174 /* Calculate the resulting mixer output format. */
175 p_aout->mixer_format = p_aout->output.output;
176 if ( !AOUT_FMT_NON_LINEAR(&p_aout->output.output) )
178 /* Non-S/PDIF mixer only deals with float32 or fixed32. */
179 p_aout->mixer_format.i_format
180 = HAVE_FPU ? VLC_CODEC_FL32 : VLC_CODEC_FI32;
181 aout_FormatPrepare( &p_aout->mixer_format );
183 else
185 p_aout->mixer_format.i_format = p_format->i_format;
188 aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format );
190 /* Create filters. */
191 p_aout->output.i_nb_filters = 0;
192 if ( aout_FiltersCreatePipeline( p_aout, p_aout->output.pp_filters,
193 &p_aout->output.i_nb_filters,
194 &p_aout->mixer_format,
195 &p_aout->output.output ) < 0 )
197 msg_Err( p_aout, "couldn't create audio output pipeline" );
198 module_unneed( p_aout, p_aout->output.p_module );
199 return -1;
202 /* Prepare hints for the buffer allocator. */
203 p_aout->mixer_allocation.b_alloc = true;
204 p_aout->mixer_allocation.i_bytes_per_sec
205 = p_aout->mixer_format.i_bytes_per_frame
206 * p_aout->mixer_format.i_rate
207 / p_aout->mixer_format.i_frame_length;
209 aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters,
210 p_aout->output.i_nb_filters,
211 &p_aout->mixer_allocation );
213 p_aout->output.b_error = 0;
214 return 0;
217 /*****************************************************************************
218 * aout_OutputDelete : delete the output
219 *****************************************************************************
220 * This function is entered with the mixer lock.
221 *****************************************************************************/
222 void aout_OutputDelete( aout_instance_t * p_aout )
224 if ( p_aout->output.b_error )
226 return;
229 module_unneed( p_aout, p_aout->output.p_module );
231 aout_FiltersDestroyPipeline( p_aout, p_aout->output.pp_filters,
232 p_aout->output.i_nb_filters );
234 aout_lock_output_fifo( p_aout );
235 aout_FifoDestroy( p_aout, &p_aout->output.fifo );
236 aout_unlock_output_fifo( p_aout );
238 p_aout->output.b_error = true;
241 /*****************************************************************************
242 * aout_OutputPlay : play a buffer
243 *****************************************************************************
244 * This function is entered with the mixer lock.
245 *****************************************************************************/
246 void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
248 aout_FiltersPlay( p_aout->output.pp_filters, p_aout->output.i_nb_filters,
249 &p_buffer );
251 if( !p_buffer )
252 return;
253 if( p_buffer->i_buffer == 0 )
255 block_Release( p_buffer );
256 return;
259 aout_lock_output_fifo( p_aout );
260 aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer );
261 p_aout->output.pf_play( p_aout );
262 aout_unlock_output_fifo( p_aout );
265 /*****************************************************************************
266 * aout_OutputNextBuffer : give the audio output plug-in the right buffer
267 *****************************************************************************
268 * If b_can_sleek is 1, the aout core functions won't try to resample
269 * new buffers to catch up - that is we suppose that the output plug-in can
270 * compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
271 * This function is entered with no lock at all :-).
272 *****************************************************************************/
273 aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
274 mtime_t start_date,
275 bool b_can_sleek )
277 aout_buffer_t * p_buffer;
279 aout_lock_output_fifo( p_aout );
281 p_buffer = p_aout->output.fifo.p_first;
283 /* Drop the audio sample if the audio output is really late.
284 * In the case of b_can_sleek, we don't use a resampler so we need to be
285 * a lot more severe. */
286 while ( p_buffer && p_buffer->i_pts <
287 (b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE )
289 msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
290 "trashing %"PRId64"us", mdate() - p_buffer->i_pts,
291 p_buffer->i_length );
292 p_buffer = p_buffer->p_next;
293 aout_BufferFree( p_aout->output.fifo.p_first );
294 p_aout->output.fifo.p_first = p_buffer;
297 if ( p_buffer == NULL )
299 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
301 #if 0 /* This is bad because the audio output might just be trying to fill
302 * in its internal buffers. And anyway, it's up to the audio output
303 * to deal with this kind of starvation. */
305 /* Set date to 0, to allow the mixer to send a new buffer ASAP */
306 aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
307 if ( !p_aout->output.b_starving )
308 msg_Dbg( p_aout,
309 "audio output is starving (no input), playing silence" );
310 p_aout->output.b_starving = 1;
311 #endif
313 aout_unlock_output_fifo( p_aout );
314 return NULL;
317 /* Here we suppose that all buffers have the same duration - this is
318 * generally true, and anyway if it's wrong it won't be a disaster.
320 if ( p_buffer->i_pts > start_date + p_buffer->i_length )
322 * + AOUT_PTS_TOLERANCE )
323 * There is no reason to want that, it just worsen the scheduling of
324 * an audio sample after an output starvation (ie. on start or on resume)
325 * --Gibalou
328 const mtime_t i_delta = p_buffer->i_pts - start_date;
329 aout_unlock_output_fifo( p_aout );
331 if ( !p_aout->output.b_starving )
332 msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
333 "playing silence", i_delta );
334 p_aout->output.b_starving = 1;
335 return NULL;
338 p_aout->output.b_starving = 0;
340 p_aout->output.fifo.p_first = p_buffer->p_next;
341 if ( p_buffer->p_next == NULL )
343 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
346 if ( !b_can_sleek &&
347 ( (p_buffer->i_pts - start_date > AOUT_PTS_TOLERANCE)
348 || (start_date - p_buffer->i_pts > AOUT_PTS_TOLERANCE) ) )
350 /* Try to compensate the drift by doing some resampling. */
351 int i;
352 mtime_t difference = start_date - p_buffer->i_pts;
353 msg_Warn( p_aout, "output date isn't PTS date, requesting "
354 "resampling (%"PRId64")", difference );
356 aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
357 aout_unlock_output_fifo( p_aout );
359 aout_lock_input_fifos( p_aout );
360 for ( i = 0; i < p_aout->i_nb_inputs; i++ )
362 aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->mixer.fifo;
364 aout_FifoMoveDates( p_aout, p_fifo, difference );
366 aout_unlock_input_fifos( p_aout );
368 else
369 aout_unlock_output_fifo( p_aout );
371 return p_buffer;