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 // Restore the original get_audio
64 frame
->get_audio
= mlt_frame_pop_audio( frame
);
66 // Get the producer's audio
67 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, &channels_avail
, samples
);
69 // Duplicate channels as necessary
70 if ( channels_avail
< *channels
)
72 int size
= *channels
* *samples
* sizeof( int16_t );
73 int16_t *new_buffer
= mlt_pool_alloc( size
);
76 // Duplicate the existing channels
77 for ( i
= 0; i
< *samples
; i
++ )
79 for ( j
= 0; j
< *channels
; j
++ )
81 new_buffer
[ ( i
* *channels
) + j
] = (*buffer
)[ ( i
* channels_avail
) + k
];
82 k
= ( k
+ 1 ) % channels_avail
;
86 // Update the audio buffer now - destroys the old
87 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
91 else if ( channels_avail
== 6 && *channels
== 2 )
93 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
94 int size
= *channels
* *samples
* sizeof( int16_t );
95 int16_t *new_buffer
= mlt_pool_alloc( size
);
97 // Drop all but the first *channels
98 for ( i
= 0; i
< *samples
; i
++ )
100 new_buffer
[ ( i
* *channels
) + 0 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
101 new_buffer
[ ( i
* *channels
) + 1 ] = (*buffer
)[ ( i
* channels_avail
) + 3 ];
104 // Update the audio buffer now - destroys the old
105 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
107 *buffer
= new_buffer
;
110 // Return now if no work to do
111 if ( output_rate
!= *frequency
)
113 // Will store number of samples created
116 // Create a resampler if nececessary
117 if ( resample
== NULL
|| *frequency
!= mlt_properties_get_int( filter_properties
, "last_frequency" ) )
119 // Create the resampler
120 resample
= audio_resample_init( *channels
, *channels
, output_rate
, *frequency
);
122 // And store it on properties
123 mlt_properties_set_data( filter_properties
, "audio_resample", resample
, 0, ( mlt_destructor
)audio_resample_close
, NULL
);
125 // And remember what it was created for
126 mlt_properties_set_int( filter_properties
, "last_frequency", *frequency
);
129 // Resample the audio
130 used
= audio_resample( resample
, sample_buffer
, *buffer
, *samples
);
132 // Resize if necessary
133 if ( used
> *samples
)
135 *buffer
= mlt_pool_realloc( *buffer
, *samples
* *channels
* sizeof( int16_t ) );
136 mlt_properties_set_data( properties
, "audio", *buffer
, *channels
* used
* sizeof( int16_t ), mlt_pool_release
, NULL
);
140 memcpy( *buffer
, sample_buffer
, *channels
* used
* sizeof( int16_t ) );
142 // Update output variables
144 *frequency
= output_rate
;
150 /** Filter processing.
153 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
155 // Only call this if we have a means to get audio
156 if ( frame
->get_audio
!= NULL
)
158 // Push the original method on to the stack
159 mlt_frame_push_audio( frame
, frame
->get_audio
);
161 // Push the filter on to the stack
162 mlt_frame_push_audio( frame
, this );
164 // Assign our get_audio method
165 frame
->get_audio
= 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
);