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_SetBool( p_aout, "intf-change", true );
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 );
199 /* Prepare hints for the buffer allocator. */
200 p_aout->mixer_allocation.b_alloc = true;
201 p_aout->mixer_allocation.i_bytes_per_sec
202 = p_aout->mixer_format.i_bytes_per_frame
203 * p_aout->mixer_format.i_rate
204 / p_aout->mixer_format.i_frame_length;
206 aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters,
207 p_aout->output.i_nb_filters,
208 &p_aout->mixer_allocation );
210 p_aout->output.b_error = 0;
214 /*****************************************************************************
215 * aout_OutputDelete : delete the output
216 *****************************************************************************
217 * This function is entered with the mixer lock.
218 *****************************************************************************/
219 void aout_OutputDelete( aout_instance_t * p_aout )
221 if ( p_aout->output.b_error )
226 module_unneed( p_aout, p_aout->output.p_module );
228 aout_FiltersDestroyPipeline( p_aout, p_aout->output.pp_filters,
229 p_aout->output.i_nb_filters );
231 aout_lock_output_fifo( p_aout );
232 aout_FifoDestroy( p_aout, &p_aout->output.fifo );
233 aout_unlock_output_fifo( p_aout );
235 p_aout->output.b_error = true;
238 /*****************************************************************************
239 * aout_OutputPlay : play a buffer
240 *****************************************************************************
241 * This function is entered with the mixer lock.
242 *****************************************************************************/
243 void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
245 aout_FiltersPlay( p_aout->output.pp_filters, p_aout->output.i_nb_filters,
250 if( p_buffer->i_buffer == 0 )
252 block_Release( p_buffer );
256 aout_lock_output_fifo( p_aout );
257 aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer );
258 p_aout->output.pf_play( p_aout );
259 aout_unlock_output_fifo( p_aout );
262 /*****************************************************************************
263 * aout_OutputNextBuffer : give the audio output plug-in the right buffer
264 *****************************************************************************
265 * If b_can_sleek is 1, the aout core functions won't try to resample
266 * new buffers to catch up - that is we suppose that the output plug-in can
267 * compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
268 * This function is entered with no lock at all :-).
269 *****************************************************************************/
270 aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
274 aout_buffer_t * p_buffer;
276 aout_lock_output_fifo( p_aout );
278 p_buffer = p_aout->output.fifo.p_first;
280 /* Drop the audio sample if the audio output is really late.
281 * In the case of b_can_sleek, we don't use a resampler so we need to be
282 * a lot more severe. */
283 while ( p_buffer && p_buffer->i_pts <
284 (b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE )
286 msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
287 "trashing %"PRId64"us", mdate() - p_buffer->i_pts,
288 p_buffer->i_length );
289 p_buffer = p_buffer->p_next;
290 aout_BufferFree( p_aout->output.fifo.p_first );
291 p_aout->output.fifo.p_first = p_buffer;
294 if ( p_buffer == NULL )
296 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
298 #if 0 /* This is bad because the audio output might just be trying to fill
299 * in its internal buffers. And anyway, it's up to the audio output
300 * to deal with this kind of starvation. */
302 /* Set date to 0, to allow the mixer to send a new buffer ASAP */
303 aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
304 if ( !p_aout->output.b_starving )
306 "audio output is starving (no input), playing silence" );
307 p_aout->output.b_starving = 1;
310 aout_unlock_output_fifo( p_aout );
314 /* Here we suppose that all buffers have the same duration - this is
315 * generally true, and anyway if it's wrong it won't be a disaster.
317 if ( p_buffer->i_pts > start_date + p_buffer->i_length )
319 * + AOUT_PTS_TOLERANCE )
320 * There is no reason to want that, it just worsen the scheduling of
321 * an audio sample after an output starvation (ie. on start or on resume)
325 const mtime_t i_delta = p_buffer->i_pts - start_date;
326 aout_unlock_output_fifo( p_aout );
328 if ( !p_aout->output.b_starving )
329 msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
330 "playing silence", i_delta );
331 p_aout->output.b_starving = 1;
335 p_aout->output.b_starving = 0;
337 p_aout->output.fifo.p_first = p_buffer->p_next;
338 if ( p_buffer->p_next == NULL )
340 p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
344 ( (p_buffer->i_pts - start_date > AOUT_PTS_TOLERANCE)
345 || (start_date - p_buffer->i_pts > AOUT_PTS_TOLERANCE) ) )
347 /* Try to compensate the drift by doing some resampling. */
349 mtime_t difference = start_date - p_buffer->i_pts;
350 msg_Warn( p_aout, "output date isn't PTS date, requesting "
351 "resampling (%"PRId64")", difference );
353 aout_FifoMoveDates( p_aout, &p_aout->output.fifo, difference );
354 aout_unlock_output_fifo( p_aout );
356 aout_lock_input_fifos( p_aout );
357 for ( i = 0; i < p_aout->i_nb_inputs; i++ )
359 aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->mixer.fifo;
361 aout_FifoMoveDates( p_aout, p_fifo, difference );
363 aout_unlock_input_fifos( p_aout );
366 aout_unlock_output_fifo( p_aout );