git.sesse.net Git - mlt/blob - src/modules/resample/filter_resample.c

   1 /*
   2  * filter_resample.c -- adjust audio sample frequency
   3  * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
   4  * Author: Dan Dennedy <dan@dennedy.org>
   5  *
   6  * This program is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, write to the Free Software Foundation,
  18  * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
  19  */
  20
  21 #include "filter_resample.h"
  22
  23 #include <framework/mlt_frame.h>
  24
  25 #include <stdio.h>
  26 #include <stdlib.h>
  27 #include <samplerate.h>
  28 #define __USE_ISOC99 1
  29 #include <math.h>
  30
  31 #define BUFFER_LEN 20480
  32 #define RESAMPLE_TYPE SRC_SINC_FASTEST
  33
  34 /** Get the audio.
  35 */
  36
  37 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  38 {
  39         // Get the properties of the frame
  40         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
  41
  42         // Get the filter service
  43         mlt_filter filter = mlt_frame_pop_audio( frame );
  44
  45         // Get the filter properties
  46         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  47
  48         // Get the resample information
  49         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  50         SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
  51         float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
  52         float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
  53         int channels_avail = *channels;
  54         SRC_DATA data;
  55         int i;
  56
  57         // If no resample frequency is specified, default to requested value
  58         if ( output_rate == 0 )
  59                 output_rate = *frequency;
  60
  61         // Get the producer's audio
  62         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  63
  64         // Duplicate channels as necessary
  65         if ( channels_avail < *channels )
  66         {
  67                 int size = *channels * *samples * sizeof( int16_t );
  68                 int16_t *new_buffer = mlt_pool_alloc( size );
  69                 int j, k = 0;
  70
  71                 // Duplicate the existing channels
  72                 for ( i = 0; i < *samples; i++ )
  73                 {
  74                         for ( j = 0; j < *channels; j++ )
  75                         {
  76                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
  77                                 k = ( k + 1 ) % channels_avail;
  78                         }
  79                 }
  80
  81                 // Update the audio buffer now - destroys the old
  82                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  83
  84                 *buffer = new_buffer;
  85         }
  86         else if ( channels_avail == 6 && *channels == 2 )
  87         {
  88                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
  89                 int size = *channels * *samples * sizeof( int16_t );
  90                 int16_t *new_buffer = mlt_pool_alloc( size );
  91
  92                 // Drop all but the first *channels
  93                 for ( i = 0; i < *samples; i++ )
  94                 {
  95                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
  96                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
  97                 }
  98
  99                 // Update the audio buffer now - destroys the old
 100                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
 101
 102                 *buffer = new_buffer;
 103         }
 104
 105         // Return now if no work to do
 106         if ( output_rate != *frequency )
 107         {
 108                 float *p = input_buffer;
 109                 float *end = p + *samples * *channels;
 110                 int16_t *q = *buffer;
 111
 112                 // Convert to floating point
 113                 while( p != end )
 114                         *p ++ = ( float )( *q ++ ) / 32768.0;
 115
 116                 // Resample
 117                 data.data_in = input_buffer;
 118                 data.data_out = output_buffer;
 119                 data.src_ratio = ( float ) output_rate / ( float ) *frequency;
 120                 data.input_frames = *samples;
 121                 data.output_frames = BUFFER_LEN / *channels;
 122                 data.end_of_input = 0;
 123                 i = src_process( state, &data );
 124                 if ( i == 0 )
 125                 {
 126                         if ( data.output_frames_gen > *samples )
 127                         {
 128                                 *buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
 129                                 mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
 130                         }
 131
 132                         *samples = data.output_frames_gen;
 133                         *frequency = output_rate;
 134
 135                         p = output_buffer;
 136                         q = *buffer;
 137                         end = p + *samples * *channels;
 138
 139                         // Convert from floating back to signed 16bit
 140                         while( p != end )
 141                         {
 142                                 if ( *p > 1.0 )
 143                                         *p = 1.0;
 144                                 if ( *p < -1.0 )
 145                                         *p = -1.0;
 146                                 if ( *p > 0 )
 147                                         *q ++ = 32767 * *p ++;
 148                                 else
 149                                         *q ++ = 32768 * *p ++;
 150                         }
 151                 }
 152                 else
 153                         fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
 154         }
 155
 156         return 0;
 157 }
 158
 159 /** Filter processing.
 160 */
 161
 162 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 163 {
 164         if ( mlt_frame_is_test_audio( frame ) != 0 )
 165         {
 166                 mlt_frame_push_audio( frame, this );
 167                 mlt_frame_push_audio( frame, resample_get_audio );
 168         }
 169
 170         return frame;
 171 }
 172
 173 /** Constructor for the filter.
 174 */
 175
 176 mlt_filter filter_resample_init( char *arg )
 177 {
 178         mlt_filter this = mlt_filter_new( );
 179         if ( this != NULL )
 180         {
 181                 int error;
 182                 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
 183                 if ( error == 0 )
 184                 {
 185                         void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 186                         void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 187                         this->process = filter_process;
 188                         if ( arg != NULL )
 189                                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
 190                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
 191                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
 192                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 193                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 194                 }
 195                 else
 196                 {
 197                         fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
 198                 }
 199         }
 200         return this;
 201 }