1 /*****************************************************************************
2 * output.c : internal management of output streams for the audio output
3 *****************************************************************************
4 * Copyright (C) 2002-2004 the VideoLAN team
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 /*****************************************************************************
26 *****************************************************************************/
31 #include <vlc_common.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) );
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" );
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 var_SetInteger( p_aout, "audio-channels", AOUT_VAR_CHAN_LEFT );
117 var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
120 else if ( p_aout->output.output.i_physical_channels ==
121 (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)
122 && (p_aout->output.output.i_original_channels &
123 (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
125 /* Stereo - create the audio-channels variable. */
126 var_Create( p_aout, "audio-channels",
127 VLC_VAR_INTEGER | VLC_VAR_HASCHOICE );
128 text.psz_string = _("Audio Channels");
129 var_Change( p_aout, "audio-channels", VLC_VAR_SETTEXT, &text, NULL );
131 if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DOLBYSTEREO )
133 val.i_int = AOUT_VAR_CHAN_DOLBYS;
134 text.psz_string = _("Dolby Surround");
138 val.i_int = AOUT_VAR_CHAN_STEREO;
139 text.psz_string = _("Stereo");
141 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
142 val.i_int = AOUT_VAR_CHAN_LEFT; text.psz_string = _("Left");
143 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
144 val.i_int = AOUT_VAR_CHAN_RIGHT; text.psz_string = _("Right");
145 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
146 val.i_int = AOUT_VAR_CHAN_RSTEREO; text.psz_string=_("Reverse stereo");
147 var_Change( p_aout, "audio-channels", VLC_VAR_ADDCHOICE, &val, &text );
148 if ( p_aout->output.output.i_original_channels & AOUT_CHAN_DUALMONO )
150 /* Go directly to the left channel. */
151 p_aout->output.output.i_original_channels = AOUT_CHAN_LEFT;
152 var_SetInteger( p_aout, "audio-channels", AOUT_VAR_CHAN_LEFT );
154 var_AddCallback( p_aout, "audio-channels", aout_ChannelsRestart,
157 var_TriggerCallback( p_aout, "intf-change" );
159 aout_FormatPrepare( &p_aout->output.output );
161 aout_lock_output_fifo( p_aout );
164 aout_FifoInit( p_aout, &p_aout->output.fifo,
165 p_aout->output.output.i_rate );
167 aout_unlock_output_fifo( p_aout );
169 aout_FormatPrint( p_aout, "output", &p_aout->output.output );
171 /* Calculate the resulting mixer output format. */
172 p_aout->mixer_format = p_aout->output.output;
173 if ( !AOUT_FMT_NON_LINEAR(&p_aout->output.output) )
175 /* Non-S/PDIF mixer only deals with float32 or fixed32. */
176 p_aout->mixer_format.i_format
177 = HAVE_FPU ? VLC_CODEC_FL32 : VLC_CODEC_FI32;
178 aout_FormatPrepare( &p_aout->mixer_format );
182 p_aout->mixer_format.i_format = p_format->i_format;
185 aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format );
187 /* Create filters. */
188 p_aout->output.i_nb_filters = 0;
189 if ( aout_FiltersCreatePipeline( p_aout, p_aout->output.pp_filters,
190 &p_aout->output.i_nb_filters,
191 &p_aout->mixer_format,
192 &p_aout->output.output ) < 0 )
194 msg_Err( p_aout, "couldn't create audio output pipeline" );
195 module_unneed( p_aout, p_aout->output.p_module );
196 p_aout->output.p_module = NULL;
200 /* Prepare hints for the buffer allocator. */
201 p_aout->mixer_allocation.b_alloc = true;
205 /*****************************************************************************
206 * aout_OutputDelete : delete the output
207 *****************************************************************************
208 * This function is entered with the mixer lock.
209 *****************************************************************************/
210 void aout_OutputDelete( aout_instance_t * p_aout )
212 if( p_aout->output.p_module == NULL )
214 module_unneed( p_aout, p_aout->output.p_module );
215 p_aout->output.p_module = NULL;
217 aout_FiltersDestroyPipeline( p_aout, p_aout->output.pp_filters,
218 p_aout->output.i_nb_filters );
220 aout_lock_output_fifo( p_aout );
221 aout_FifoDestroy( p_aout, &p_aout->output.fifo );
222 aout_unlock_output_fifo( p_aout );
225 /*****************************************************************************
226 * aout_OutputPlay : play a buffer
227 *****************************************************************************
228 * This function is entered with the mixer lock.
229 *****************************************************************************/
230 void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
232 aout_FiltersPlay( p_aout->output.pp_filters, p_aout->output.i_nb_filters,
237 if( p_buffer->i_buffer == 0 )
239 block_Release( p_buffer );
243 aout_lock_output_fifo( p_aout );
244 aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer );
245 p_aout->output.pf_play( p_aout );
246 aout_unlock_output_fifo( p_aout );
249 /*****************************************************************************
250 * aout_OutputNextBuffer : give the audio output plug-in the right buffer
251 *****************************************************************************
252 * If b_can_sleek is 1, the aout core functions won't try to resample
253 * new buffers to catch up - that is we suppose that the output plug-in can
254 * compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
255 * This function is entered with no lock at all :-).
256 *****************************************************************************/
257 aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
261 aout_buffer_t * p_buffer;
262 mtime_t now = mdate();
264 aout_lock_output_fifo( p_aout );
266 p_buffer = p_aout->output.fifo.p_first;
268 /* Drop the audio sample if the audio output is really late.
269 * In the case of b_can_sleek, we don't use a resampler so we need to be
270 * a lot more severe. */
271 while ( p_buffer && p_buffer->i_pts <
272 (b_can_sleek ? start_date : now) - AOUT_PTS_TOLERANCE )
274 msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
275 "trashing %"PRId64"us", now - p_buffer->i_pts,
276 p_buffer->i_length );
277 p_buffer = p_buffer->p_next;
278 aout_BufferFree( p_aout->output.fifo.p_first );
279 p_aout->output.fifo.p_first = p_buffer;
282 if ( p_buffer == NULL )
284 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
286 #if 0 /* This is bad because the audio output might just be trying to fill
287 * in its internal buffers. And anyway, it's up to the audio output
288 * to deal with this kind of starvation. */
290 /* Set date to 0, to allow the mixer to send a new buffer ASAP */
291 aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
292 if ( !p_aout->output.b_starving )
294 "audio output is starving (no input), playing silence" );
295 p_aout->output.b_starving = 1;
298 aout_unlock_output_fifo( p_aout );
302 /* Here we suppose that all buffers have the same duration - this is
303 * generally true, and anyway if it's wrong it won't be a disaster.
305 if ( p_buffer->i_pts > start_date + p_buffer->i_length )
307 * + AOUT_PTS_TOLERANCE )
308 * There is no reason to want that, it just worsen the scheduling of
309 * an audio sample after an output starvation (ie. on start or on resume)
313 const mtime_t i_delta = p_buffer->i_pts - start_date;
314 aout_unlock_output_fifo( p_aout );
316 if ( !p_aout->output.b_starving )
317 msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
318 "playing silence", i_delta );
319 p_aout->output.b_starving = 1;
323 p_aout->output.b_starving = 0;
325 p_aout->output.fifo.p_first = p_buffer->p_next;
326 if ( p_buffer->p_next == NULL )
328 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
333 mtime_t difference = start_date - p_buffer->i_pts;
335 if( difference > AOUT_PTS_TOLERANCE
336 || difference < -AOUT_PTS_TOLERANCE )
338 /* Try to compensate the drift by doing some resampling. */
339 msg_Warn( p_aout, "output date isn't PTS date, requesting "
340 "resampling (%"PRId64")", difference );
342 aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
343 aout_unlock_output_fifo( p_aout );
345 aout_lock_input_fifos( p_aout );
346 aout_fifo_t *p_fifo = &p_aout->pp_inputs[0]->mixer.fifo;
347 aout_FifoMoveDates( p_aout, p_fifo, difference );
348 aout_unlock_input_fifos( p_aout );
352 aout_unlock_output_fifo( p_aout );