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 val
.i_int
= AOUT_VAR_CHAN_LEFT
;
116 var_Set( p_aout
, "audio-channels", val
);
118 var_AddCallback( p_aout
, "audio-channels", aout_ChannelsRestart
,
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");
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
,
160 var_Set( p_aout
, "intf-change", val
);
162 aout_FormatPrepare( &p_aout
->output
.output
);
164 aout_lock_output_fifo( p_aout
);
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
);
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
);
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;
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
)
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
,
253 if( p_buffer
->i_buffer
== 0 )
255 block_Release( p_buffer
);
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
,
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
)
309 "audio output is starving (no input), playing silence" );
310 p_aout
->output
.b_starving
= 1;
313 aout_unlock_output_fifo( p_aout
);
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)
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;
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
;
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. */
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
);
369 aout_unlock_output_fifo( p_aout
);