1 /*****************************************************************************
2 * input.c : internal management of input streams for the audio output
3 *****************************************************************************
4 * Copyright (C) 2002 VideoLAN
5 * $Id: input.c,v 1.20 2002/11/11 22:27:00 gbazin Exp $
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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
22 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
27 #include <stdlib.h> /* calloc(), malloc(), free() */
36 #include "audio_output.h"
37 #include "aout_internal.h"
39 /*****************************************************************************
40 * aout_InputNew : allocate a new input and rework the filter pipeline
41 *****************************************************************************/
42 int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
44 audio_sample_format_t intermediate_format;
46 aout_FormatPrint( p_aout, "input", &p_input->input );
49 aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate );
50 p_input->p_first_byte_to_mix = NULL;
53 memcpy( &intermediate_format, &p_aout->mixer.mixer,
54 sizeof(audio_sample_format_t) );
55 intermediate_format.i_rate = p_input->input.i_rate;
56 if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
57 &p_input->i_nb_filters, &p_input->input,
58 &intermediate_format ) < 0 )
60 msg_Err( p_aout, "couldn't set an input pipeline" );
62 aout_FifoDestroy( p_aout, &p_input->fifo );
68 /* Prepare hints for the buffer allocator. */
69 p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
70 p_input->input_alloc.i_bytes_per_sec = -1;
72 if ( AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) )
74 p_input->i_nb_resamplers = 0;
78 /* Create resamplers. */
79 intermediate_format.i_rate = (p_input->input.i_rate
80 * (100 + AOUT_MAX_RESAMPLING)) / 100;
81 if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate )
84 intermediate_format.i_rate++;
86 if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
87 &p_input->i_nb_resamplers,
89 &p_aout->mixer.mixer ) < 0 )
91 msg_Err( p_aout, "couldn't set a resampler pipeline" );
93 aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
94 p_input->i_nb_filters );
95 aout_FifoDestroy( p_aout, &p_input->fifo );
101 aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
102 p_input->i_nb_resamplers,
103 &p_input->input_alloc );
105 /* Setup the initial rate of the resampler */
106 p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
107 p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
111 p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
112 p_input->input_alloc.i_bytes_per_sec = -1;
113 aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
114 p_input->i_nb_filters,
115 &p_input->input_alloc );
117 /* i_bytes_per_sec is still == -1 if no filters */
118 p_input->input_alloc.i_bytes_per_sec = __MAX(
119 p_input->input_alloc.i_bytes_per_sec,
120 p_input->input.i_bytes_per_frame
121 * p_input->input.i_rate
122 / p_input->input.i_frame_length );
123 /* Allocate in the heap, it is more convenient for the decoder. */
124 p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
126 p_input->b_error = 0;
131 /*****************************************************************************
132 * aout_InputDelete : delete an input
133 *****************************************************************************
134 * This function must be entered with the mixer lock.
135 *****************************************************************************/
136 int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
138 if ( p_input->b_error ) return 0;
140 aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
141 p_input->i_nb_filters );
142 aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
143 p_input->i_nb_resamplers );
144 aout_FifoDestroy( p_aout, &p_input->fifo );
149 /*****************************************************************************
150 * aout_InputPlay : play a buffer
151 *****************************************************************************
152 * This function must be entered with the input lock.
153 *****************************************************************************/
154 int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
155 aout_buffer_t * p_buffer )
159 /* We don't care if someone changes the start date behind our back after
160 * this. We'll deal with that when pushing the buffer, and compensate
161 * with the next incoming buffer. */
162 vlc_mutex_lock( &p_aout->input_fifos_lock );
163 start_date = aout_FifoNextStart( p_aout, &p_input->fifo );
164 vlc_mutex_unlock( &p_aout->input_fifos_lock );
166 if ( start_date != 0 && start_date < mdate() )
168 /* The decoder is _very_ late. This can only happen if the user
169 * pauses the stream (or if the decoder is buggy, which cannot
171 msg_Warn( p_aout, "computed PTS is out of range ("I64Fd"), "
172 "clearing out", mdate() - start_date );
173 vlc_mutex_lock( &p_aout->input_fifos_lock );
174 aout_FifoSet( p_aout, &p_input->fifo, 0 );
175 vlc_mutex_unlock( &p_aout->input_fifos_lock );
176 if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
177 msg_Warn( p_aout, "timing screwed, stopping resampling" );
178 p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
179 p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
180 p_input->pp_resamplers[0]->b_reinit = VLC_TRUE;
184 if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
186 /* The decoder gives us f*cked up PTS. It's its business, but we
187 * can't present it anyway, so drop the buffer. */
188 msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer",
189 mdate() - p_buffer->start_date );
190 aout_BufferFree( p_buffer );
195 if ( start_date == 0 ) start_date = p_buffer->start_date;
197 /* Run pre-filters. */
198 aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
201 /* Run the resampler if needed.
202 * We first need to calculate the output rate of this resampler. */
203 if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
204 ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
205 || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) )
207 /* Can happen in several circumstances :
208 * 1. A problem at the input (clock drift)
209 * 2. A small pause triggered by the user
210 * 3. Some delay in the output stage, causing a loss of lip
212 * Solution : resample the buffer to avoid a scratch.
214 mtime_t drift = p_buffer->start_date - start_date;
216 p_input->i_resamp_start_date = mdate();
217 p_input->i_resamp_start_drift = drift;
220 p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
222 p_input->i_resampling_type = AOUT_RESAMPLING_UP;
224 msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling",
225 drift > 0 ? drift : -drift,
226 drift > 0 ? "in advance" : "late",
227 drift > 0 ? "down" : "up");
230 if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
232 /* Resampling has been triggered previously (because of dates
233 * mismatch). We want the resampling to happen progressively so
234 * it isn't too audible to the listener. */
236 if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
238 p_input->pp_resamplers[0]->input.i_rate += 10; /* Hz */
242 p_input->pp_resamplers[0]->input.i_rate -= 10; /* Hz */
245 /* Check if everything is back to normal, in which case we can stop the
247 if( p_input->pp_resamplers[0]->input.i_rate ==
248 p_input->input.i_rate )
250 p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
251 msg_Warn( p_aout, "resampling stopped after "I64Fi" usec",
252 mdate() - p_input->i_resamp_start_date );
254 else if( abs( p_buffer->start_date - start_date ) <
255 abs( p_input->i_resamp_start_drift ) / 2 )
257 /* if we reduced the drift from half, then it is time to switch
258 * back the resampling direction. */
259 if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
260 p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
262 p_input->i_resampling_type = AOUT_RESAMPLING_UP;
263 p_input->i_resamp_start_drift = 0;
265 else if( p_input->i_resamp_start_drift &&
266 ( abs( p_buffer->start_date - start_date ) >
267 abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
269 /* If the drift is increasing and not decreasing, than something
270 * is bad. We'd better stop the resampling right now. */
271 msg_Warn( p_aout, "timing screwed, stopping resampling" );
272 p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
273 p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
278 /* Adding the start date will be managed by aout_FifoPush(). */
279 p_buffer->start_date = start_date;
280 p_buffer->end_date = start_date +
281 (p_buffer->end_date - p_buffer->start_date);
283 /* Actually run the resampler now. */
284 if ( p_aout->mixer.mixer.i_rate !=
285 p_input->pp_resamplers[0]->input.i_rate )
287 aout_FiltersPlay( p_aout, p_input->pp_resamplers,
288 p_input->i_nb_resamplers,
292 vlc_mutex_lock( &p_aout->input_fifos_lock );
293 aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
294 vlc_mutex_unlock( &p_aout->input_fifos_lock );