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
  32 /** Get the audio.
  33 */
  34
  35 static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
  36 {
  37         // Get the filter service
  38         mlt_filter filter = mlt_frame_pop_audio( frame );
  39
  40         // Get the filter properties
  41         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
  42
  43         mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
  44
  45         // Get the resample information
  46         int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
  47         int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
  48
  49         // Obtain the resample context if it exists
  50         ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
  51
  52         // If no resample frequency is specified, default to requested value
  53         if ( output_rate == 0 )
  54                 output_rate = *frequency;
  55
  56         // Get the producer's audio
  57         int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  58         if ( error ) return error;
  59
  60         // Return now if no work to do
  61         if ( output_rate != *frequency )
  62         {
  63                 // Will store number of samples created
  64                 int used = 0;
  65
  66                 mlt_log_debug( MLT_FILTER_SERVICE(filter), "channels %d samples %d frequency %d -> %d\n",
  67                         *channels, *samples, *frequency, output_rate );
  68
  69                 // Do not convert to s16 unless we need to change the rate
  70                 if ( *format != mlt_audio_s16 )
  71                 {
  72                         *format = mlt_audio_s16;
  73                         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
  74                 }
  75
  76                 // Create a resampler if nececessary
  77                 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
  78                 {
  79                         // Create the resampler
  80 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
  81                         resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
  82                                 SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
  83 #else
  84                         resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
  85 #endif
  86
  87                         // And store it on properties
  88                         mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
  89
  90                         // And remember what it was created for
  91                         mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
  92                 }
  93
  94                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
  95
  96                 // Resample the audio
  97                 used = audio_resample( resample, sample_buffer, *buffer, *samples );
  98                 int size = used * *channels * sizeof( int16_t );
  99
 100                 // Resize if necessary
 101                 if ( used > *samples )
 102                 {
 103                         *buffer = mlt_pool_realloc( *buffer, size );
 104                         mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
 105                 }
 106
 107                 // Copy samples
 108                 memcpy( *buffer, sample_buffer, size );
 109
 110                 // Update output variables
 111                 *samples = used;
 112                 *frequency = output_rate;
 113         }
 114         else
 115         {
 116                 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
 117         }
 118
 119         return error;
 120 }
 121
 122 /** Filter processing.
 123 */
 124
 125 static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
 126 {
 127         // Only call this if we have a means to get audio
 128         if ( mlt_frame_is_test_audio( frame ) == 0 )
 129         {
 130                 // Push the filter on to the stack
 131                 mlt_frame_push_audio( frame, filter );
 132
 133                 // Assign our get_audio method
 134                 mlt_frame_push_audio( frame, resample_get_audio );
 135         }
 136
 137         return frame;
 138 }
 139
 140 /** Constructor for the filter.
 141 */
 142
 143 mlt_filter filter_avresample_init( char *arg )
 144 {
 145         // Create a filter
 146         mlt_filter filter = mlt_filter_new( );
 147
 148         // Initialise if successful
 149         if ( filter != NULL )
 150         {
 151                 // Calculate size of the buffer
 152                 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
 153
 154                 // Allocate the buffer
 155                 int16_t *buffer = mlt_pool_alloc( size );
 156
 157                 // Assign the process method
 158                 filter->process = filter_process;
 159
 160                 // Deal with argument
 161                 if ( arg != NULL )
 162                         mlt_properties_set( MLT_FILTER_PROPERTIES( filter ), "frequency", arg );
 163
 164                 // Default to 2 channel output
 165                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( filter ), "channels", 2 );
 166
 167                 // Store the buffer
 168                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( filter ), "buffer", buffer, size, mlt_pool_release, NULL );
 169         }
 170
 171         return filter;
 172 }