2 * filter_resample.c -- adjust audio sample frequency
3 * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
4 * Author: Dan Dennedy <dan@dennedy.org>
6 * This program is free software; you can redistribute it and/or modify
7 * it under the terms of the GNU General Public License as published by
8 * the Free Software Foundation; either version 2 of the License, or
9 * (at your option) any later version.
11 * This program 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
14 * GNU General Public License for more details.
16 * You should have received a copy of the GNU General Public License
17 * along with this program; if not, write to the Free Software Foundation,
18 * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
21 #include "filter_resample.h"
23 #include <framework/mlt_frame.h>
27 #include <samplerate.h>
28 #define __USE_ISOC99 1
31 #define BUFFER_LEN 20480
32 #define RESAMPLE_TYPE SRC_SINC_FASTEST
37 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
39 // Get the properties of the frame
40 mlt_properties properties = mlt_frame_properties( frame );
42 // Get the filter service
43 mlt_filter filter = mlt_frame_pop_audio( frame );
45 // Get the filter properties
46 mlt_properties filter_properties = mlt_filter_properties( filter );
48 // Get the resample information
49 int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
50 SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
51 float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
52 float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
53 int channels_avail = *channels;
57 // If no resample frequency is specified, default to requested value
58 if ( output_rate == 0 )
59 output_rate = *frequency;
61 // Restore the original get_audio
62 frame->get_audio = mlt_frame_pop_audio( frame );
64 // Get the producer's audio
65 mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
67 // Duplicate channels as necessary
68 if ( channels_avail < *channels )
70 int size = *channels * *samples * sizeof( int16_t );
71 int16_t *new_buffer = mlt_pool_alloc( size );
74 // Duplicate the existing channels
75 for ( i = 0; i < *samples; i++ )
77 for ( j = 0; j < *channels; j++ )
79 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
80 k = ( k + 1 ) % channels_avail;
84 // Update the audio buffer now - destroys the old
85 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
89 else if ( channels_avail == 6 && *channels == 2 )
91 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
92 int size = *channels * *samples * sizeof( int16_t );
93 int16_t *new_buffer = mlt_pool_alloc( size );
95 // Drop all but the first *channels
96 for ( i = 0; i < *samples; i++ )
98 new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
99 new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
102 // Update the audio buffer now - destroys the old
103 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
105 *buffer = new_buffer;
108 // Return now if no work to do
109 if ( output_rate != *frequency )
111 float *p = input_buffer;
112 float *end = p + *samples * *channels;
113 int16_t *q = *buffer;
115 // Convert to floating point
117 *p ++ = ( float )( *q ++ ) / 32768.0;
120 data.data_in = input_buffer;
121 data.data_out = output_buffer;
122 data.src_ratio = ( float ) output_rate / ( float ) *frequency;
123 data.input_frames = *samples;
124 data.output_frames = BUFFER_LEN / *channels;
125 data.end_of_input = 0;
126 i = src_process( state, &data );
129 if ( data.output_frames_gen > *samples )
131 *buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
132 mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
135 *samples = data.output_frames_gen;
136 *frequency = output_rate;
140 end = p + *samples * *channels;
142 // Convert from floating back to signed 16bit
150 *q ++ = 32767 * *p ++;
152 *q ++ = 32768 * *p ++;
156 fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
162 /** Filter processing.
165 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
167 if ( frame->get_audio != NULL )
169 mlt_frame_push_audio( frame, frame->get_audio );
170 mlt_frame_push_audio( frame, this );
171 frame->get_audio = resample_get_audio;
177 /** Constructor for the filter.
180 mlt_filter filter_resample_init( char *arg )
182 mlt_filter this = mlt_filter_new( );
186 SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
189 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
190 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
191 this->process = filter_process;
193 mlt_properties_set_int( mlt_filter_properties( this ), "frequency", atoi( arg ) );
194 mlt_properties_set_int( mlt_filter_properties( this ), "channels", 2 );
195 mlt_properties_set_data( mlt_filter_properties( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
196 mlt_properties_set_data( mlt_filter_properties( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
197 mlt_properties_set_data( mlt_filter_properties( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
201 fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );