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

   1 /*
   2  * filter_audiochannels.c -- convert from one audio format to another
   3  * Copyright (C) 2009-2012 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 <string.h>
  26
  27 static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  28 {
  29         // Used to return number of channels in the source
  30         int channels_avail = *channels;
  31
  32         // Get the producer's audio
  33         int error = mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  34         if ( error ) return error;
  35
  36         if ( channels_avail < *channels )
  37         {
  38                 int size = mlt_audio_format_size( *format, *samples, *channels );
  39                 int16_t *new_buffer = mlt_pool_alloc( size );
  40
  41                 // Duplicate the existing channels
  42                 if ( *format == mlt_audio_s16 )
  43                 {
  44                         int i, j, k = 0;
  45                         for ( i = 0; i < *samples; i++ )
  46                         {
  47                                 for ( j = 0; j < *channels; j++ )
  48                                 {
  49                                         new_buffer[ ( i * *channels ) + j ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + k ];
  50                                         k = ( k + 1 ) % channels_avail;
  51                                 }
  52                         }
  53                 }
  54                 else if ( *format == mlt_audio_s32le || *format == mlt_audio_f32le )
  55                 {
  56                         int32_t *p = (int32_t*) new_buffer;
  57                         int i, j, k = 0;
  58                         for ( i = 0; i < *samples; i++ )
  59                         {
  60                                 for ( j = 0; j < *channels; j++ )
  61                                 {
  62                                         p[ ( i * *channels ) + j ] = ((int32_t*)(*buffer))[ ( i * channels_avail ) + k ];
  63                                         k = ( k + 1 ) % channels_avail;
  64                                 }
  65                         }
  66                 }
  67                 else if ( *format == mlt_audio_u8 )
  68                 {
  69                         uint8_t *p = (uint8_t*) new_buffer;
  70                         int i, j, k = 0;
  71                         for ( i = 0; i < *samples; i++ )
  72                         {
  73                                 for ( j = 0; j < *channels; j++ )
  74                                 {
  75                                         p[ ( i * *channels ) + j ] = ((uint8_t*)(*buffer))[ ( i * channels_avail ) + k ];
  76                                         k = ( k + 1 ) % channels_avail;
  77                                 }
  78                         }
  79                 }
  80                 else
  81                 {
  82                         // non-interleaved - s32 or float
  83                         int size_avail = mlt_audio_format_size( *format, *samples, channels_avail );
  84                         int32_t *p = (int32_t*) new_buffer;
  85                         int i = *channels / channels_avail;
  86                         while ( i-- )
  87                         {
  88                                 memcpy( p, *buffer, size_avail );
  89                                 p += size_avail / sizeof(*p);
  90                         }
  91                         i = *channels % channels_avail;
  92                         if ( i )
  93                         {
  94                                 size_avail = mlt_audio_format_size( *format, *samples, i );
  95                                 memcpy( p, *buffer, size_avail );
  96                         }
  97                 }
  98                 // Update the audio buffer now - destroys the old
  99                 mlt_frame_set_audio( frame, new_buffer, *format, size, mlt_pool_release );
 100                 *buffer = new_buffer;
 101         }
 102         else if ( channels_avail > *channels )
 103         {
 104                 int size = mlt_audio_format_size( *format, *samples, *channels );
 105                 int16_t *new_buffer = mlt_pool_alloc( size );
 106
 107                 // Drop all but the first *channels
 108                 if ( *format == mlt_audio_s16 )
 109                 {
 110                         int i, j;
 111                         for ( i = 0; i < *samples; i++ )
 112                                 for ( j = 0; j < *channels; j++ )
 113                                         new_buffer[ ( i * *channels ) + j ]     = ((int16_t*)(*buffer))[ ( i * channels_avail ) + j ];
 114                 }
 115                 else
 116                 {
 117                         // non-interleaved
 118                         memcpy( new_buffer, *buffer, size );
 119                 }
 120                 // Update the audio buffer now - destroys the old
 121                 mlt_frame_set_audio( frame, new_buffer, *format, size, mlt_pool_release );
 122                 *buffer = new_buffer;
 123         }
 124         return error;
 125 }
 126
 127 /** Filter processing.
 128 */
 129
 130 static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
 131 {
 132         mlt_frame_push_audio( frame, filter_get_audio );
 133         return frame;
 134 }
 135
 136 /** Constructor for the filter.
 137 */
 138
 139 mlt_filter filter_audiochannels_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 140 {
 141         mlt_filter filter = mlt_filter_new();
 142         if ( filter )
 143                 filter->process = filter_process;
 144         return filter;
 145 }