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 #include <framework/mlt_log.h>
  24
  25 #include <stdio.h>
  26 #include <stdlib.h>
  27 #include <samplerate.h>
  28 #include <string.h>
  29
  30 #define BUFFER_LEN 20480
  31 #define RESAMPLE_TYPE SRC_SINC_FASTEST
  32
  33 /** Get the audio.
  34 */
  35
  36 static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  37 {
  38         // Get the filter service
  39         mlt_filter filter = mlt_frame_pop_audio( frame );
  40
  41         // Get the filter properties
  42         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  43
  44         // Get the resample information
  45         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  46         int error = 0;
  47
  48         // If no resample frequency is specified, default to requested value
  49         if ( output_rate == 0 )
  50                 output_rate = *frequency;
  51
  52         // Get the producer's audio
  53         if ( *format != mlt_audio_s16 )
  54                 *format = mlt_audio_float;
  55         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  56
  57         // Return now if no work to do
  58         if ( output_rate != *frequency )
  59         {
  60                 // Do not convert to float unless we need to change the rate
  61                 if ( *format != mlt_audio_float )
  62                 {
  63                         *format = mlt_audio_float;
  64                         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  65                 }
  66                 float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
  67                 float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
  68                 SRC_DATA data;
  69                 data.data_in = input_buffer;
  70                 data.data_out = output_buffer;
  71                 data.src_ratio = ( float ) output_rate / ( float ) *frequency;
  72                 data.input_frames = *samples;
  73                 data.output_frames = BUFFER_LEN / *channels;
  74                 data.end_of_input = 0;
  75
  76                 SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
  77                 if ( !state || mlt_properties_get_int( filter_properties, "channels" ) != *channels )
  78                 {
  79                         // Recreate the resampler if the number of channels changed
  80                         state = src_new( RESAMPLE_TYPE, *channels, &error );
  81                         mlt_properties_set_data( filter_properties, "state", state, 0, (mlt_destructor) src_delete, NULL );
  82                         mlt_properties_set_int( filter_properties, "channels", *channels );
  83                 }
  84
  85                 // Convert to interleaved
  86                 float *q = (float*) *buffer;
  87                 float *p = input_buffer;
  88                 int s, c;
  89                 for ( s = 0; s < *samples; s++ )
  90                         for ( c = 0; c < *channels; c++ )
  91                                 *p++ = *( q + c * *samples + s );
  92
  93                 // Resample the audio
  94                 error = src_process( state, &data );
  95                 if ( !error )
  96                 {
  97                         int size = data.output_frames_gen * *channels * sizeof(float);
  98
  99                         // Resize if necessary
 100                         if ( data.output_frames_gen > *samples )
 101                         {
 102                                 *buffer = mlt_pool_realloc( *buffer, size );
 103                                 mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
 104                         }
 105
 106                         // Convert to non-interleaved
 107                         p = (float*) *buffer;
 108                         for ( c = 0; c < *channels; c++ )
 109                         {
 110                                 float *q = output_buffer + c;
 111                                 int i = data.output_frames_gen + 1;
 112                                 while ( --i  )
 113                                 {
 114                                         *p++ = *q;
 115                                         q += *channels;
 116                                 }
 117                         }
 118
 119                         // Update output variables
 120                         *samples = data.output_frames_gen;
 121                         *frequency = output_rate;
 122                 }
 123                 else
 124                         mlt_log_error( MLT_FILTER_SERVICE( filter ), "%s %d,%d,%d\n", src_strerror( error ), *frequency, *samples, output_rate );
 125         }
 126
 127         return error;
 128 }
 129
 130 /** Filter processing.
 131 */
 132
 133 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 134 {
 135         if ( mlt_frame_is_test_audio( frame ) == 0 )
 136         {
 137                 mlt_frame_push_audio( frame, this );
 138                 mlt_frame_push_audio( frame, resample_get_audio );
 139         }
 140
 141         return frame;
 142 }
 143
 144 /** Constructor for the filter.
 145 */
 146
 147 mlt_filter filter_resample_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 148 {
 149         mlt_filter this = mlt_filter_new( );
 150         if ( this != NULL )
 151         {
 152                 int error;
 153                 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
 154                 if ( error == 0 )
 155                 {
 156                         void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 157                         void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 158                         this->process = filter_process;
 159                         if ( arg != NULL )
 160                                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
 161                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
 162                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
 163                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 164                         mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 165                 }
 166                 else
 167                 {
 168                         fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
 169                 }
 170         }
 171         return this;
 172 }