2 * filter_avresample.c -- adjust audio sample frequency
3 * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
4 * Author: Charles Yates <charles.yates@pandora.be>
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.
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.
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
21 #include <framework/mlt_filter.h>
22 #include <framework/mlt_frame.h>
23 #include <framework/mlt_log.h>
29 // ffmpeg Header files
30 #include <libavformat/avformat.h>
31 #include <libavutil/samplefmt.h>
33 #if defined(FFUDIV) || (LIBAVCODEC_VERSION_INT < ((54<<16)+(26<<8)+0))
35 #define MAX_AUDIO_FRAME_SIZE (192000) // 1 second of 48khz 32bit audio
41 static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
43 // Get the filter service
44 mlt_filter filter = mlt_frame_pop_audio( frame );
46 // Get the filter properties
47 mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
49 mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
51 // Get the resample information
52 int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
53 int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
55 // Obtain the resample context if it exists
56 ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
58 // If no resample frequency is specified, default to requested value
59 if ( output_rate == 0 )
60 output_rate = *frequency;
62 // Get the producer's audio
63 int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
64 if ( error ) return error;
66 // Return now if no work to do
67 if ( output_rate != *frequency )
69 // Will store number of samples created
72 mlt_log_debug( MLT_FILTER_SERVICE(filter), "channels %d samples %d frequency %d -> %d\n",
73 *channels, *samples, *frequency, output_rate );
75 // Do not convert to s16 unless we need to change the rate
76 if ( *format != mlt_audio_s16 )
78 *format = mlt_audio_s16;
79 mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
82 // Create a resampler if nececessary
83 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
85 // Create the resampler
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 );
89 // And store it on properties
90 mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
92 // And remember what it was created for
93 mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
96 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
99 used = audio_resample( resample, sample_buffer, *buffer, *samples );
100 int size = used * *channels * sizeof( int16_t );
102 // Resize if necessary
103 if ( used > *samples )
105 *buffer = mlt_pool_realloc( *buffer, size );
106 mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
110 memcpy( *buffer, sample_buffer, size );
112 // Update output variables
114 *frequency = output_rate;
118 mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
124 /** Filter processing.
127 static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
129 // Only call this if we have a means to get audio
130 if ( mlt_frame_is_test_audio( frame ) == 0 )
132 // Push the filter on to the stack
133 mlt_frame_push_audio( frame, filter );
135 // Assign our get_audio method
136 mlt_frame_push_audio( frame, resample_get_audio );
142 /** Constructor for the filter.
145 mlt_filter filter_avresample_init( char *arg )
148 mlt_filter filter = mlt_filter_new( );
150 // Initialise if successful
151 if ( filter != NULL )
153 // Calculate size of the buffer
154 int size = MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
156 // Allocate the buffer
157 int16_t *buffer = mlt_pool_alloc( size );
159 // Assign the process method
160 filter->process = filter_process;
162 // Deal with argument
164 mlt_properties_set( MLT_FILTER_PROPERTIES( filter ), "frequency", arg );
166 // Default to 2 channel output
167 mlt_properties_set_int( MLT_FILTER_PROPERTIES( filter ), "channels", 2 );
170 mlt_properties_set_data( MLT_FILTER_PROPERTIES( filter ), "buffer", buffer, size, mlt_pool_release, NULL );
176 #endif // defined(FFUDIV) || (LIBAVCODEC_VERSION_INT < ((54<<16)+(26<<8)+0))