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 a frame
  40         mlt_properties properties = mlt_frame_properties( frame );
  41         int output_rate = mlt_properties_get_int( properties, "resample.frequency" );
  42         SRC_STATE *state = mlt_properties_get_data( properties, "resample.state", NULL );
  43         SRC_DATA data;
  44         float *input_buffer = mlt_properties_get_data( properties, "resample.input_buffer", NULL );
  45         float *output_buffer = mlt_properties_get_data( properties, "resample.output_buffer", NULL );
  46         int channels_avail = *channels;
  47         int i;
  48
  49         if ( output_rate == 0 )
  50                 output_rate = *frequency;
  51
  52         // Restore the original get_audio
  53         frame->get_audio = mlt_properties_get_data( properties, "resample.get_audio", NULL );
  54
  55         // Get the producer's audio
  56         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  57
  58         // Duplicate channels as necessary
  59         if ( channels_avail < *channels )
  60         {
  61                 int size = *channels * *samples * sizeof( int16_t );
  62                 int16_t *new_buffer = mlt_pool_alloc( size );
  63
  64                 // Duplicate the existing channels
  65                 for ( i = 0; i < *samples; i++ )
  66                 {
  67                         int j, k = 0;
  68                         for ( j = 0; j < *channels; j++ )
  69                         {
  70                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
  71                                 k = ( k + 1 ) % channels_avail;
  72                         }
  73                 }
  74
  75                 // Update the audio buffer now - destroys the old
  76                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  77
  78                 *buffer = new_buffer;
  79         }
  80         else if ( channels_avail > *channels )
  81         {
  82                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
  83                 int size = *channels * *samples * sizeof( int16_t );
  84                 int16_t *new_buffer = mlt_pool_alloc( size );
  85
  86                 // Drop all but the first *channels
  87                 for ( i = 0; i < *samples; i++ )
  88                 {
  89                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
  90                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
  91                 }
  92
  93                 // Update the audio buffer now - destroys the old
  94                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  95
  96                 *buffer = new_buffer;
  97         }
  98
  99         // Return now if no work to do
 100         if ( output_rate == *frequency )
 101                 return 0;
 102
 103         //fprintf( stderr, "resample_get_audio: input_rate %d output_rate %d\n", *frequency, output_rate );
 104
 105         // Convert to floating point
 106         for ( i = 0; i < *samples * *channels; ++i )
 107                 input_buffer[ i ] = ( float )( (*buffer)[ i ] ) / 32768;
 108
 109         // Resample
 110         data.data_in = input_buffer;
 111         data.data_out = output_buffer;
 112         data.src_ratio = ( float ) output_rate / ( float ) *frequency;
 113         data.input_frames = *samples;
 114         data.output_frames = BUFFER_LEN / *channels;
 115         data.end_of_input = 0;
 116         i = src_process( state, &data );
 117         if ( i == 0 )
 118         {
 119                 if ( data.output_frames_gen > *samples )
 120                 {
 121                         *buffer = mlt_pool_alloc( data.output_frames_gen * *channels * sizeof( int16_t ) );
 122                         mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
 123                 }
 124                 *samples = data.output_frames_gen;
 125                 *frequency = output_rate;
 126
 127                 // Convert from floating back to signed 16bit
 128                 for ( i = 0; i < *samples * *channels; ++i )
 129                 {
 130                         float sample = output_buffer[ i ];
 131                         if ( sample > 1.0 )
 132                                 sample = 1.0;
 133                         if ( sample < -1.0 )
 134                                 sample = -1.0;
 135                         if ( sample >= 0 )
 136                                 (*buffer)[ i ] = lrint( 32767.0 * sample );
 137                         else
 138                                 (*buffer)[ i ] = lrint( 32768.0 * sample );
 139                 }
 140         }
 141         else
 142                 fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
 143
 144         return 0;
 145 }
 146
 147 /** Filter processing.
 148 */
 149
 150 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 151 {
 152         if ( frame->get_audio != NULL )
 153         {
 154                 mlt_properties properties = mlt_filter_properties( this );
 155                 mlt_properties frame_props = mlt_frame_properties( frame );
 156
 157                 // Propogate the frequency property if supplied
 158                 if ( mlt_properties_get( properties, "frequency" ) != NULL )
 159                         mlt_properties_set_int( frame_props, "resample.frequency", mlt_properties_get_int( properties, "frequency" ) );
 160
 161                 // Propogate the other properties
 162                 mlt_properties_set_int( frame_props, "resample.channels", mlt_properties_get_int( properties, "channels" ) );
 163                 mlt_properties_set_data( frame_props, "resample.state", mlt_properties_get_data( properties, "state", NULL ), 0, NULL, NULL );
 164                 mlt_properties_set_data( frame_props, "resample.input_buffer", mlt_properties_get_data( properties, "input_buffer", NULL ), 0, NULL, NULL );
 165                 mlt_properties_set_data( frame_props, "resample.output_buffer", mlt_properties_get_data( properties, "output_buffer", NULL ), 0, NULL, NULL );
 166
 167                 // Backup the original get_audio (it's still needed)
 168                 mlt_properties_set_data( frame_props, "resample.get_audio", frame->get_audio, 0, NULL, NULL );
 169
 170                 // Override the get_audio method
 171                 frame->get_audio = resample_get_audio;
 172         }
 173
 174         return frame;
 175 }
 176
 177 /** Constructor for the filter.
 178 */
 179
 180 mlt_filter filter_resample_init( char *arg )
 181 {
 182         mlt_filter this = mlt_filter_new( );
 183         if ( this != NULL )
 184         {
 185                 int error;
 186                 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
 187                 if ( error == 0 )
 188                 {
 189                         void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 190                         void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 191                         this->process = filter_process;
 192                         if ( arg != NULL )
 193                                 mlt_properties_set_int( mlt_filter_properties( this ), "frequency", atoi( arg ) );
 194                         mlt_properties_set_int( mlt_filter_properties( this ), "channels", 2 );
 195                         mlt_properties_set_data( mlt_filter_properties( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
 196                         mlt_properties_set_data( mlt_filter_properties( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 197                         mlt_properties_set_data( mlt_filter_properties( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 198                 }
 199                 else
 200                 {
 201                         fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
 202                 }
 203         }
 204         return this;
 205 }