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>
28 // ffmpeg Header files
29 #include <libavformat/avformat.h>
34 static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
36 // Get the properties of the frame
37 mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
39 // Get the filter service
40 mlt_filter filter = mlt_frame_pop_audio( frame );
42 // Get the filter properties
43 mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
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 );
49 // Obtain the resample context if it exists
50 ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
52 // Used to return number of channels in the source
53 int channels_avail = *channels;
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 *format = mlt_audio_s16;
64 mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
66 // Duplicate channels as necessary
67 if ( channels_avail < *channels )
69 int size = *channels * *samples * sizeof( int16_t );
70 int16_t *new_buffer = mlt_pool_alloc( size );
73 // Duplicate the existing channels
74 for ( i = 0; i < *samples; i++ )
76 for ( j = 0; j < *channels; j++ )
78 new_buffer[ ( i * *channels ) + j ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + k ];
79 k = ( k + 1 ) % channels_avail;
83 // Update the audio buffer now - destroys the old
84 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
88 else if ( channels_avail == 6 && *channels == 2 )
90 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
91 int size = *channels * *samples * sizeof( int16_t );
92 int16_t *new_buffer = mlt_pool_alloc( size );
94 // Drop all but the first *channels
95 for ( i = 0; i < *samples; i++ )
97 new_buffer[ ( i * *channels ) + 0 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 2 ];
98 new_buffer[ ( i * *channels ) + 1 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 3 ];
101 // Update the audio buffer now - destroys the old
102 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
104 *buffer = new_buffer;
107 // Return now if no work to do
108 if ( output_rate != *frequency )
110 // Will store number of samples created
113 // Create a resampler if nececessary
114 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
116 // Create the resampler
117 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
118 resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
119 SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
121 resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
124 // And store it on properties
125 mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
127 // And remember what it was created for
128 mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
131 // Resample the audio
132 used = audio_resample( resample, sample_buffer, *buffer, *samples );
133 int size = used * *channels * sizeof( int16_t );
135 // Resize if necessary
136 if ( used > *samples )
138 *buffer = mlt_pool_realloc( *buffer, size );
139 mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
143 memcpy( *buffer, sample_buffer, size );
145 // Update output variables
147 *frequency = output_rate;
153 /** Filter processing.
156 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
158 // Only call this if we have a means to get audio
159 if ( mlt_frame_is_test_audio( frame ) == 0 )
161 // Push the filter on to the stack
162 mlt_frame_push_audio( frame, this );
164 // Assign our get_audio method
165 mlt_frame_push_audio( frame, resample_get_audio );
171 /** Constructor for the filter.
174 mlt_filter filter_avresample_init( char *arg )
177 mlt_filter this = mlt_filter_new( );
179 // Initialise if successful
182 // Calculate size of the buffer
183 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
185 // Allocate the buffer
186 int16_t *buffer = mlt_pool_alloc( size );
188 // Assign the process method
189 this->process = filter_process;
191 // Deal with argument
193 mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
195 // Default to 2 channel output
196 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
199 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );