git.sesse.net Git - mlt/blob - src/modules/avformat/filter_avresample.c

   1 /*
   2  * filter_avresample.c -- adjust audio sample frequency
   3  * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
   4  * Author: Charles Yates <charles.yates@pandora.be>
   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
  24 #include <stdio.h>
  25 #include <stdlib.h>
  26 #include <string.h>
  27
  28 // ffmpeg Header files
  29 #include <libavformat/avformat.h>
  30
  31 /** Get the audio.
  32 */
  33
  34 static int resample_get_audio( mlt_frame frame, void **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         mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
  46
  47         // Get the resample information
  48         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  49         int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
  50
  51         // Obtain the resample context if it exists
  52         ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
  53
  54         // Used to return number of channels in the source
  55         int channels_avail = *channels;
  56
  57         // Loop variable
  58         int i;
  59
  60         // If no resample frequency is specified, default to requested value
  61         if ( output_rate == 0 )
  62                 output_rate = *frequency;
  63
  64         // Get the producer's audio
  65         *format = mlt_audio_s16;
  66         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
  67
  68         // Duplicate channels as necessary
  69         if ( channels_avail < *channels )
  70         {
  71                 int size = *channels * *samples * sizeof( int16_t );
  72                 int16_t *new_buffer = mlt_pool_alloc( size );
  73                 int j, k = 0;
  74
  75                 // Duplicate the existing channels
  76                 for ( i = 0; i < *samples; i++ )
  77                 {
  78                         for ( j = 0; j < *channels; j++ )
  79                         {
  80                                 new_buffer[ ( i * *channels ) + j ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + k ];
  81                                 k = ( k + 1 ) % channels_avail;
  82                         }
  83                 }
  84
  85                 // Update the audio buffer now - destroys the old
  86                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
  87
  88                 *buffer = new_buffer;
  89         }
  90         else if ( channels_avail == 6 && *channels == 2 )
  91         {
  92                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
  93                 int size = *channels * *samples * sizeof( int16_t );
  94                 int16_t *new_buffer = mlt_pool_alloc( size );
  95
  96                 // Drop all but the first *channels
  97                 for ( i = 0; i < *samples; i++ )
  98                 {
  99                         new_buffer[ ( i * *channels ) + 0 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 2 ];
 100                         new_buffer[ ( i * *channels ) + 1 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 3 ];
 101                 }
 102
 103                 // Update the audio buffer now - destroys the old
 104                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
 105
 106                 *buffer = new_buffer;
 107         }
 108
 109         // Return now if no work to do
 110         if ( output_rate != *frequency )
 111         {
 112                 // Will store number of samples created
 113                 int used = 0;
 114
 115                 // Create a resampler if nececessary
 116                 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
 117                 {
 118                         // Create the resampler
 119 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
 120                         resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
 121                                 SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
 122 #else
 123                         resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
 124 #endif
 125
 126                         // And store it on properties
 127                         mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
 128
 129                         // And remember what it was created for
 130                         mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
 131                 }
 132
 133                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
 134
 135                 // Resample the audio
 136                 used = audio_resample( resample, sample_buffer, *buffer, *samples );
 137                 int size = used * *channels * sizeof( int16_t );
 138
 139                 // Resize if necessary
 140                 if ( used > *samples )
 141                 {
 142                         *buffer = mlt_pool_realloc( *buffer, size );
 143                         mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
 144                 }
 145
 146                 // Copy samples
 147                 memcpy( *buffer, sample_buffer, size );
 148
 149                 // Update output variables
 150                 *samples = used;
 151                 *frequency = output_rate;
 152         }
 153         else
 154         {
 155                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
 156         }
 157
 158         return 0;
 159 }
 160
 161 /** Filter processing.
 162 */
 163
 164 static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
 165 {
 166         // Only call this if we have a means to get audio
 167         if ( mlt_frame_is_test_audio( frame ) == 0 )
 168         {
 169                 // Push the filter on to the stack
 170                 mlt_frame_push_audio( frame, filter );
 171
 172                 // Assign our get_audio method
 173                 mlt_frame_push_audio( frame, resample_get_audio );
 174         }
 175
 176         return frame;
 177 }
 178
 179 /** Constructor for the filter.
 180 */
 181
 182 mlt_filter filter_avresample_init( char *arg )
 183 {
 184         // Create a filter
 185         mlt_filter filter = mlt_filter_new( );
 186
 187         // Initialise if successful
 188         if ( filter != NULL )
 189         {
 190                 // Calculate size of the buffer
 191                 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
 192
 193                 // Allocate the buffer
 194                 int16_t *buffer = mlt_pool_alloc( size );
 195
 196                 // Assign the process method
 197                 filter->process = filter_process;
 198
 199                 // Deal with argument
 200                 if ( arg != NULL )
 201                         mlt_properties_set( MLT_FILTER_PROPERTIES( filter ), "frequency", arg );
 202
 203                 // Default to 2 channel output
 204                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( filter ), "channels", 2 );
 205
 206                 // Store the buffer
 207                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( filter ), "buffer", buffer, size, mlt_pool_release, NULL );
 208         }
 209
 210         return filter;
 211 }