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 <framework/mlt_filter.h>
  22 #include <framework/mlt_frame.h>
  23
  24 #include <stdio.h>
  25 #include <stdlib.h>
  26 #include <samplerate.h>
  27
  28 #define BUFFER_LEN 20480
  29 #define RESAMPLE_TYPE SRC_SINC_FASTEST
  30
  31 /** Get the audio.
  32 */
  33
  34 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  35 {
  36         // Get the properties of the frame
  37         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
  38
  39         // Get the filter service
  40         mlt_filter filter = mlt_frame_pop_audio( frame );
  41
  42         // Get the filter properties
  43         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  44
  45         // Get the resample information
  46         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  47         SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
  48         float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
  49         float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
  50         int channels_avail = *channels;
  51         SRC_DATA data;
  52         int i;
  53
  54         // If no resample frequency is specified, default to requested value
  55         if ( output_rate == 0 )
  56                 output_rate = *frequency;
  57
  58         // Get the producer's audio
  59         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  60
  61         // Duplicate channels as necessary
  62         if ( channels_avail < *channels )
  63         {
  64                 int size = *channels * *samples * sizeof( int16_t );
  65                 int16_t *new_buffer = mlt_pool_alloc( size );
  66                 int j, k = 0;
  67
  68                 // Duplicate the existing channels
  69                 for ( i = 0; i < *samples; i++ )
  70                 {
  71                         for ( j = 0; j < *channels; j++ )
  72                         {
  73                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
  74                                 k = ( k + 1 ) % channels_avail;
  75                         }
  76                 }
  77
  78                 // Update the audio buffer now - destroys the old
  79                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  80
  81                 *buffer = new_buffer;
  82         }
  83         else if ( channels_avail == 6 && *channels == 2 )
  84         {
  85                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
  86                 int size = *channels * *samples * sizeof( int16_t );
  87                 int16_t *new_buffer = mlt_pool_alloc( size );
  88
  89                 // Drop all but the first *channels
  90                 for ( i = 0; i < *samples; i++ )
  91                 {
  92                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
  93                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
  94                 }
  95
  96                 // Update the audio buffer now - destroys the old
  97                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  98
  99                 *buffer = new_buffer;
 100         }
 101
 102         // Return now if no work to do
 103         if ( output_rate != *frequency )
 104         {
 105                 float *p = input_buffer;
 106                 float *end = p + *samples * *channels;
 107                 int16_t *q = *buffer;
 108
 109                 // Convert to floating point
 110                 while( p != end )
 111                         *p ++ = ( float )( *q ++ ) / 32768.0;
 112
 113                 // Resample
 114                 data.data_in = input_buffer;
 115                 data.data_out = output_buffer;
 116                 data.src_ratio = ( float ) output_rate / ( float ) *frequency;
 117                 data.input_frames = *samples;
 118                 data.output_frames = BUFFER_LEN / *channels;
 119                 data.end_of_input = 0;
 120                 i = src_process( state, &data );
 121                 if ( i == 0 )
 122                 {
 123                         if ( data.output_frames_gen > *samples )
 124                         {
 125                                 *buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
 126                                 mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
 127                         }
 128
 129                         *samples = data.output_frames_gen;
 130                         *frequency = output_rate;
 131
 132                         p = output_buffer;
 133                         q = *buffer;
 134                         end = p + *samples * *channels;
 135
 136                         // Convert from floating back to signed 16bit
 137                         while( p != end )
 138                         {
 139                                 if ( *p > 1.0 )
 140                                         *p = 1.0;
 141                                 if ( *p < -1.0 )
 142                                         *p = -1.0;
 143                                 if ( *p > 0 )
 144                                         *q ++ = 32767 * *p ++;
 145                                 else
 146                                         *q ++ = 32768 * *p ++;
 147                         }
 148                 }
 149                 else
 150                         fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
 151         }
 152
 153         return 0;
 154 }
 155
 156 /** Filter processing.
 157 */
 158
 159 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 160 {
 161         if ( mlt_frame_is_test_audio( frame ) == 0 )
 162         {
 163                 mlt_frame_push_audio( frame, this );
 164                 mlt_frame_push_audio( frame, resample_get_audio );
 165         }
 166
 167         return frame;
 168 }
 169
 170 /** Constructor for the filter.
 171 */
 172
 173 mlt_filter filter_resample_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 174 {
 175         mlt_filter this = mlt_filter_new( );
 176         if ( this != NULL )
 177         {
 178                 int error;
 179                 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
 180                 if ( error == 0 )
 181                 {
 182                         void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 183                         void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 184                         this->process = filter_process;
 185                         if ( arg != NULL )
 186                                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
 187                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
 188                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
 189                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 190                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 191                 }
 192                 else
 193                 {
 194                         fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
 195                 }
 196         }
 197         return this;
 198 }