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 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_avresample.h"
23 #include <framework/mlt_frame.h>
29 // ffmpeg Header files
35 static int resample_get_audio( mlt_frame frame
, int16_t **buffer
, mlt_audio_format
*format
, int *frequency
, int *channels
, int *samples
)
37 // Get the properties of the frame
38 mlt_properties properties
= MLT_FRAME_PROPERTIES( frame
);
40 // Get the filter service
41 mlt_filter filter
= mlt_frame_pop_audio( frame
);
43 // Get the filter properties
44 mlt_properties filter_properties
= MLT_FILTER_PROPERTIES( filter
);
46 // Get the resample information
47 int output_rate
= mlt_properties_get_int( filter_properties
, "frequency" );
48 int16_t *sample_buffer
= mlt_properties_get_data( filter_properties
, "buffer", NULL
);
50 // Obtain the resample context if it exists
51 ReSampleContext
*resample
= mlt_properties_get_data( filter_properties
, "audio_resample", NULL
);
53 // Used to return number of channels in the source
54 int channels_avail
= *channels
;
59 // If no resample frequency is specified, default to requested value
60 if ( output_rate
== 0 )
61 output_rate
= *frequency
;
63 // Get the producer's audio
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
] = (*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 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
98 new_buffer
[ ( i
* *channels
) + 1 ] = (*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 resample
= audio_resample_init( *channels
, *channels
, output_rate
, *frequency
);
119 // And store it on properties
120 mlt_properties_set_data( filter_properties
, "audio_resample", resample
, 0, ( mlt_destructor
)audio_resample_close
, NULL
);
122 // And remember what it was created for
123 mlt_properties_set_int( filter_properties
, "last_frequency", *frequency
);
126 // Resample the audio
127 used
= audio_resample( resample
, sample_buffer
, *buffer
, *samples
);
129 // Resize if necessary
130 if ( used
> *samples
)
132 *buffer
= mlt_pool_realloc( *buffer
, *samples
* *channels
* sizeof( int16_t ) );
133 mlt_properties_set_data( properties
, "audio", *buffer
, *channels
* used
* sizeof( int16_t ), mlt_pool_release
, NULL
);
137 memcpy( *buffer
, sample_buffer
, *channels
* used
* sizeof( int16_t ) );
139 // Update output variables
141 *frequency
= output_rate
;
147 /** Filter processing.
150 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
152 // Only call this if we have a means to get audio
153 if ( mlt_frame_is_test_audio( frame
) == 0 )
155 // Push the filter on to the stack
156 mlt_frame_push_audio( frame
, this );
158 // Assign our get_audio method
159 mlt_frame_push_audio( frame
, resample_get_audio
);
165 /** Constructor for the filter.
168 mlt_filter
filter_avresample_init( char *arg
)
171 mlt_filter
this = mlt_filter_new( );
173 // Initialise if successful
176 // Calculate size of the buffer
177 int size
= AVCODEC_MAX_AUDIO_FRAME_SIZE
* sizeof( int16_t );
179 // Allocate the buffer
180 int16_t *buffer
= mlt_pool_alloc( size
);
182 // Assign the process method
183 this->process
= filter_process
;
185 // Deal with argument
187 mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg
);
189 // Default to 2 channel output
190 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
193 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer
, size
, mlt_pool_release
, NULL
);