git.sesse.net Git - mlt/blob - src/modules/core/filter_mono.c

   1 /*
   2  * filter_mono.c -- mix all channels to a mono signal across n channels
   3  * Copyright (C) 2003-2009 Ushodaya Enterprises Limited
   4  * Author: Dan Dennedy <dan@dennedy.org>
   5  *
   6  * This library is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2.1 of the License, or (at your option) any later version.
  10  *
  11  * This library 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 GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General Public
  17  * License along with this library; if not, write to the Free Software
  18  * Foundation, 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
  28 /** Get the audio.
  29 */
  30
  31 static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  32 {
  33         // Get the properties of the a frame
  34         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
  35         int channels_out = mlt_properties_get_int( properties, "mono.channels" );
  36         int i, j, size;
  37
  38         // Get the producer's audio
  39         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  40
  41         if ( channels_out == -1 )
  42                 channels_out = *channels;
  43         size = mlt_audio_format_size( *format, *samples, channels_out );
  44
  45         switch ( *format )
  46         {
  47                 case mlt_audio_s16:
  48                 {
  49                         int16_t *new_buffer = mlt_pool_alloc( size );
  50                         for ( i = 0; i < *samples; i++ )
  51                         {
  52                                 int16_t mixdown = 0;
  53                                 for ( j = 0; j < *channels; j++ )
  54                                         mixdown += ((int16_t*) *buffer)[ ( i * *channels ) + j ] / *channels;
  55                                 for ( j = 0; j < channels_out; j++ )
  56                                         new_buffer[ ( i * channels_out ) + j ] = mixdown;
  57                         }
  58                         *buffer = new_buffer;
  59                         break;
  60                 }
  61                 case mlt_audio_s32:
  62                 {
  63                         int32_t *new_buffer = mlt_pool_alloc( size );
  64                         for ( i = 0; i < *samples; i++ )
  65                         {
  66                                 int32_t mixdown = 0;
  67                                 for ( j = 0; j < *channels; j++ )
  68                                         mixdown += ((int32_t*) *buffer)[ ( j * *channels ) + i ] / *channels;
  69                                 for ( j = 0; j < channels_out; j++ )
  70                                         new_buffer[ ( j * *samples ) + i ] = mixdown;
  71                         }
  72                         *buffer = new_buffer;
  73                         break;
  74                 }
  75                 case mlt_audio_float:
  76                 {
  77                         float *new_buffer = mlt_pool_alloc( size );
  78                         for ( i = 0; i < *samples; i++ )
  79                         {
  80                                 float mixdown = 0;
  81                                 for ( j = 0; j < *channels; j++ )
  82                                         mixdown += ((float*) *buffer)[ ( j * *channels ) + i ] / *channels;
  83                                 for ( j = 0; j < channels_out; j++ )
  84                                         new_buffer[ ( j * *samples ) + i ] = mixdown;
  85                         }
  86                         *buffer = new_buffer;
  87                         break;
  88                 }
  89                 default:
  90                         mlt_log_error( NULL, "[filter mono] Invalid audio format\n" );
  91                         break;
  92         }
  93         if ( size > *samples * channels_out )
  94         {
  95                 mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
  96                 *channels = channels_out;
  97         }
  98
  99         return 0;
 100 }
 101
 102 /** Filter processing.
 103 */
 104
 105 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 106 {
 107         mlt_properties properties = MLT_FILTER_PROPERTIES( this );
 108         mlt_properties frame_props = MLT_FRAME_PROPERTIES( frame );
 109
 110         // Propogate the parameters
 111         mlt_properties_set_int( frame_props, "mono.channels", mlt_properties_get_int( properties, "channels" ) );
 112
 113         // Override the get_audio method
 114         mlt_frame_push_audio( frame, filter_get_audio );
 115
 116         return frame;
 117 }
 118
 119 /** Constructor for the filter.
 120 */
 121
 122 mlt_filter filter_mono_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 123 {
 124         mlt_filter this = mlt_filter_new( );
 125         if ( this != NULL )
 126         {
 127                 this->process = filter_process;
 128                 if ( arg != NULL )
 129                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", atoi( arg ) );
 130                 else
 131                         mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", -1 );
 132         }
 133         return this;
 134 }