1 /*****************************************************************************
2 * linear.c : linear interpolation resampler
3 *****************************************************************************
4 * Copyright (C) 2002, 2006 the VideoLAN team
7 * Authors: Gildas Bazin <gbazin@netcourrier.com>
8 * Sigmund Augdal Helberg <dnumgis@videolan.org>
10 * This program is free software; you can redistribute it and/or modify
11 * it under the terms of the GNU General Public License as published by
12 * the Free Software Foundation; either version 2 of the License, or
13 * (at your option) any later version.
15 * This program is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
18 * GNU General Public License for more details.
20 * You should have received a copy of the GNU General Public License
21 * along with this program; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
23 *****************************************************************************/
25 /*****************************************************************************
27 *****************************************************************************/
33 #include <vlc_common.h>
34 #include <vlc_plugin.h>
36 #include <vlc_filter.h>
37 #include <vlc_block.h>
40 /*****************************************************************************
42 *****************************************************************************/
43 static int OpenFilter ( vlc_object_t * );
44 static void CloseFilter( vlc_object_t * );
45 static block_t *Resample( filter_t *, block_t * );
48 typedef float sample_t;
49 # define VLC_CODEC_NATIVE VLC_CODEC_FL32
51 typedef int32_t sample_t;
52 # define VLC_CODEC_NATIVE VLC_CODEC_FI32
55 /*****************************************************************************
57 *****************************************************************************/
60 sample_t *p_prev_sample; /* this filter introduces a 1 sample delay */
62 unsigned int i_remainder; /* remainder of previous sample */
67 /*****************************************************************************
69 *****************************************************************************/
71 set_description( N_("Audio filter for linear interpolation resampling") )
72 set_category( CAT_AUDIO )
73 set_subcategory( SUBCAT_AUDIO_MISC )
74 set_capability( "audio filter2", 5 )
75 set_callbacks( OpenFilter, CloseFilter )
78 /*****************************************************************************
79 * Resample: convert a buffer
80 *****************************************************************************/
81 static block_t *Resample( filter_t *p_filter, block_t *p_in_buf )
83 if( !p_in_buf || !p_in_buf->i_nb_samples )
86 block_Release( p_in_buf );
90 filter_sys_t *p_sys = p_filter->p_sys;
91 unsigned i_nb_channels = p_filter->fmt_in.audio.i_channels;
92 sample_t *p_prev_sample = p_sys->p_prev_sample;
94 /* Check if we really need to run the resampler */
95 if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
97 if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) )
99 p_in_buf = block_Realloc( p_in_buf,
100 sizeof(sample_t) * i_nb_channels,
101 p_in_buf->i_buffer );
105 memcpy( p_in_buf->p_buffer, p_prev_sample,
106 i_nb_channels * sizeof(sample_t) );
111 unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
112 p_filter->fmt_out.audio.i_bitspersample / 8;
114 size_t i_out_size = i_bytes_per_frame * (1 + (p_in_buf->i_nb_samples *
115 p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
116 block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
120 sample_t *p_out = (sample_t *)p_out_buf->p_buffer;
122 unsigned i_in_nb = p_in_buf->i_nb_samples;
124 const sample_t *p_in = (sample_t *)p_in_buf->p_buffer;
126 /* Take care of the previous input sample (if any) */
127 if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
129 p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
130 p_sys->i_remainder = 0;
131 date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
135 while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
137 for( unsigned i = 0; i < i_nb_channels ; i++ )
139 p_out[i] = p_prev_sample[i];
141 p_out[i] += (p_in[i] - p_prev_sample[i])
143 p_out[i] += (int64_t)(p_in[i] - p_prev_sample[i])
145 * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
147 p_out += i_nb_channels;
150 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
152 p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
155 /* Take care of the current input samples (minus last one) */
156 for( unsigned i_in = 0; i_in < i_in_nb - 1; i_in++ )
158 while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
160 for( unsigned i = 0; i < i_nb_channels ; i++ )
164 p_out[i] += (p_in[i + i_nb_channels] - p_in[i])
166 p_out[i] += (int64_t)(p_in[i + i_nb_channels] - p_in[i])
168 * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
170 p_out += i_nb_channels;
173 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
176 p_in += i_nb_channels;
177 p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
180 /* Backup the last input sample for next time */
181 memcpy( p_prev_sample, p_in, i_nb_channels * sizeof(sample_t) );
183 p_out_buf->i_nb_samples = i_out;
184 p_out_buf->i_pts = p_in_buf->i_pts;
186 if( p_in_buf->i_pts !=
187 date_Get( &p_sys->end_date ) )
189 date_Set( &p_sys->end_date, p_in_buf->i_pts );
192 p_out_buf->i_length = date_Increment( &p_sys->end_date,
193 p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
195 p_out_buf->i_buffer = p_out_buf->i_nb_samples *
196 i_nb_channels * sizeof(sample_t);
198 block_Release( p_in_buf );
202 /*****************************************************************************
204 *****************************************************************************/
205 static int OpenFilter( vlc_object_t *p_this )
207 filter_t *p_filter = (filter_t *)p_this;
209 int i_out_rate = p_filter->fmt_out.audio.i_rate;
211 if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
212 p_filter->fmt_in.i_codec != VLC_CODEC_NATIVE )
217 /* Allocate the memory needed to store the module's structure */
218 p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
222 p_sys->p_prev_sample = malloc(
223 p_filter->fmt_in.audio.i_channels * sizeof(sample_t) );
224 if( p_sys->p_prev_sample == NULL )
229 date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
230 p_sys->i_remainder = 0;
232 p_filter->pf_audio_filter = Resample;
234 msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
235 (char *)&p_filter->fmt_in.i_codec,
236 p_filter->fmt_in.audio.i_rate,
237 p_filter->fmt_in.audio.i_channels,
238 (char *)&p_filter->fmt_out.i_codec,
239 p_filter->fmt_out.audio.i_rate,
240 p_filter->fmt_out.audio.i_channels);
242 p_filter->fmt_out = p_filter->fmt_in;
243 p_filter->fmt_out.audio.i_rate = i_out_rate;
248 /*****************************************************************************
249 * CloseFilter : deallocate data structures
250 *****************************************************************************/
251 static void CloseFilter( vlc_object_t *p_this )
253 filter_t *p_filter = (filter_t *)p_this;
254 free( p_filter->p_sys->p_prev_sample );
255 free( p_filter->p_sys );