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 #include <framework/mlt_log.h>
  24
  25 #include <stdio.h>
  26 #include <stdlib.h>
  27 #include <string.h>
  28
  29 // ffmpeg Header files
  30 #include <libavformat/avformat.h>
  31 #if LIBAVUTIL_VERSION_INT >= ((50<<16)+(38<<8)+0)
  32 #  include <libavutil/samplefmt.h>
  33 #else
  34 #  define AV_SAMPLE_FMT_S16 SAMPLE_FMT_S16
  35 #endif
  36
  37 /** Get the audio.
  38 */
  39
  40 static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  41 {
  42         // Get the filter service
  43         mlt_filter filter = mlt_frame_pop_audio( frame );
  44
  45         // Get the filter properties
  46         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  47
  48         mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
  49
  50         // Get the resample information
  51         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  52         int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
  53
  54         // Obtain the resample context if it exists
  55         ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
  56
  57         // If no resample frequency is specified, default to requested value
  58         if ( output_rate == 0 )
  59                 output_rate = *frequency;
  60
  61         // Get the producer's audio
  62         int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  63         if ( error ) return error;
  64
  65         // Return now if no work to do
  66         if ( output_rate != *frequency )
  67         {
  68                 // Will store number of samples created
  69                 int used = 0;
  70
  71                 mlt_log_debug( MLT_FILTER_SERVICE(filter), "channels %d samples %d frequency %d -> %d\n",
  72                         *channels, *samples, *frequency, output_rate );
  73
  74                 // Do not convert to s16 unless we need to change the rate
  75                 if ( *format != mlt_audio_s16 )
  76                 {
  77                         *format = mlt_audio_s16;
  78                         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  79                 }
  80
  81                 // Create a resampler if nececessary
  82                 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
  83                 {
  84                         // Create the resampler
  85 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
  86                         resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
  87                                 AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
  88 #else
  89                         resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
  90 #endif
  91
  92                         // And store it on properties
  93                         mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
  94
  95                         // And remember what it was created for
  96                         mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
  97                 }
  98
  99                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
 100
 101                 // Resample the audio
 102                 used = audio_resample( resample, sample_buffer, *buffer, *samples );
 103                 int size = used * *channels * sizeof( int16_t );
 104
 105                 // Resize if necessary
 106                 if ( used > *samples )
 107                 {
 108                         *buffer = mlt_pool_realloc( *buffer, size );
 109                         mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
 110                 }
 111
 112                 // Copy samples
 113                 memcpy( *buffer, sample_buffer, size );
 114
 115                 // Update output variables
 116                 *samples = used;
 117                 *frequency = output_rate;
 118         }
 119         else
 120         {
 121                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
 122         }
 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         // Only call this if we have a means to get audio
 133         if ( mlt_frame_is_test_audio( frame ) == 0 )
 134         {
 135                 // Push the filter on to the stack
 136                 mlt_frame_push_audio( frame, filter );
 137
 138                 // Assign our get_audio method
 139                 mlt_frame_push_audio( frame, resample_get_audio );
 140         }
 141
 142         return frame;
 143 }
 144
 145 /** Constructor for the filter.
 146 */
 147
 148 mlt_filter filter_avresample_init( char *arg )
 149 {
 150         // Create a filter
 151         mlt_filter filter = mlt_filter_new( );
 152
 153         // Initialise if successful
 154         if ( filter != NULL )
 155         {
 156                 // Calculate size of the buffer
 157                 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
 158
 159                 // Allocate the buffer
 160                 int16_t *buffer = mlt_pool_alloc( size );
 161
 162                 // Assign the process method
 163                 filter->process = filter_process;
 164
 165                 // Deal with argument
 166                 if ( arg != NULL )
 167                         mlt_properties_set( MLT_FILTER_PROPERTIES( filter ), "frequency", arg );
 168
 169                 // Default to 2 channel output
 170                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( filter ), "channels", 2 );
 171
 172                 // Store the buffer
 173                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( filter ), "buffer", buffer, size, mlt_pool_release, NULL );
 174         }
 175
 176         return filter;
 177 }