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 // Get the producer's audio
62 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, &channels_avail
, samples
);
64 // Duplicate channels as necessary
65 if ( channels_avail
< *channels
)
67 int size
= *channels
* *samples
* sizeof( int16_t );
68 int16_t *new_buffer
= mlt_pool_alloc( size
);
71 // Duplicate the existing channels
72 for ( i
= 0; i
< *samples
; i
++ )
74 for ( j
= 0; j
< *channels
; j
++ )
76 new_buffer
[ ( i
* *channels
) + j
] = (*buffer
)[ ( i
* channels_avail
) + k
];
77 k
= ( k
+ 1 ) % channels_avail
;
81 // Update the audio buffer now - destroys the old
82 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
86 else if ( channels_avail
== 6 && *channels
== 2 )
88 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
89 int size
= *channels
* *samples
* sizeof( int16_t );
90 int16_t *new_buffer
= mlt_pool_alloc( size
);
92 // Drop all but the first *channels
93 for ( i
= 0; i
< *samples
; i
++ )
95 new_buffer
[ ( i
* *channels
) + 0 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
96 new_buffer
[ ( i
* *channels
) + 1 ] = (*buffer
)[ ( i
* channels_avail
) + 3 ];
99 // Update the audio buffer now - destroys the old
100 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
102 *buffer
= new_buffer
;
105 // Return now if no work to do
106 if ( output_rate
!= *frequency
)
108 float *p
= input_buffer
;
109 float *end
= p
+ *samples
* *channels
;
110 int16_t *q
= *buffer
;
112 // Convert to floating point
114 *p
++ = ( float )( *q
++ ) / 32768.0;
117 data
.data_in
= input_buffer
;
118 data
.data_out
= output_buffer
;
119 data
.src_ratio
= ( float ) output_rate
/ ( float ) *frequency
;
120 data
.input_frames
= *samples
;
121 data
.output_frames
= BUFFER_LEN
/ *channels
;
122 data
.end_of_input
= 0;
123 i
= src_process( state
, &data
);
126 if ( data
.output_frames_gen
> *samples
)
128 *buffer
= mlt_pool_realloc( *buffer
, data
.output_frames_gen
* *channels
* sizeof( int16_t ) );
129 mlt_properties_set_data( properties
, "audio", *buffer
, *channels
* data
.output_frames_gen
* 2, mlt_pool_release
, NULL
);
132 *samples
= data
.output_frames_gen
;
133 *frequency
= output_rate
;
137 end
= p
+ *samples
* *channels
;
139 // Convert from floating back to signed 16bit
147 *q
++ = 32767 * *p
++;
149 *q
++ = 32768 * *p
++;
153 fprintf( stderr
, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i
), *frequency
, *samples
, output_rate
);
159 /** Filter processing.
162 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
164 if ( mlt_frame_is_test_audio( frame
) != 0 )
166 mlt_frame_push_audio( frame
, this );
167 mlt_frame_push_audio( frame
, resample_get_audio
);
173 /** Constructor for the filter.
176 mlt_filter
filter_resample_init( char *arg
)
178 mlt_filter
this = mlt_filter_new( );
182 SRC_STATE
*state
= src_new( RESAMPLE_TYPE
, 2 /* channels */, &error
);
185 void *input_buffer
= mlt_pool_alloc( BUFFER_LEN
);
186 void *output_buffer
= mlt_pool_alloc( BUFFER_LEN
);
187 this->process
= filter_process
;
189 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg
) );
190 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
191 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state
, 0, (mlt_destructor
)src_delete
, NULL
);
192 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
193 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
197 fprintf( stderr
, "filter_resample_init: %s\n", src_strerror( error
) );