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
34 static int resample_get_audio( mlt_frame frame
, int16_t **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 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, &channels_avail
, samples
);
65 // Duplicate channels as necessary
66 if ( channels_avail
< *channels
)
68 int size
= *channels
* *samples
* sizeof( int16_t );
69 int16_t *new_buffer
= mlt_pool_alloc( size
);
72 // Duplicate the existing channels
73 for ( i
= 0; i
< *samples
; i
++ )
75 for ( j
= 0; j
< *channels
; j
++ )
77 new_buffer
[ ( i
* *channels
) + j
] = (*buffer
)[ ( i
* channels_avail
) + k
];
78 k
= ( k
+ 1 ) % channels_avail
;
82 // Update the audio buffer now - destroys the old
83 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
87 else if ( channels_avail
== 6 && *channels
== 2 )
89 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
90 int size
= *channels
* *samples
* sizeof( int16_t );
91 int16_t *new_buffer
= mlt_pool_alloc( size
);
93 // Drop all but the first *channels
94 for ( i
= 0; i
< *samples
; i
++ )
96 new_buffer
[ ( i
* *channels
) + 0 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
97 new_buffer
[ ( i
* *channels
) + 1 ] = (*buffer
)[ ( i
* channels_avail
) + 3 ];
100 // Update the audio buffer now - destroys the old
101 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
103 *buffer
= new_buffer
;
106 // Return now if no work to do
107 if ( output_rate
!= *frequency
)
109 // Will store number of samples created
112 // Create a resampler if nececessary
113 if ( resample
== NULL
|| *frequency
!= mlt_properties_get_int( filter_properties
, "last_frequency" ) )
115 // Create the resampler
116 resample
= audio_resample_init( *channels
, *channels
, output_rate
, *frequency
);
118 // And store it on properties
119 mlt_properties_set_data( filter_properties
, "audio_resample", resample
, 0, ( mlt_destructor
)audio_resample_close
, NULL
);
121 // And remember what it was created for
122 mlt_properties_set_int( filter_properties
, "last_frequency", *frequency
);
125 // Resample the audio
126 used
= audio_resample( resample
, sample_buffer
, *buffer
, *samples
);
128 // Resize if necessary
129 if ( used
> *samples
)
131 *buffer
= mlt_pool_realloc( *buffer
, *samples
* *channels
* sizeof( int16_t ) );
132 mlt_properties_set_data( properties
, "audio", *buffer
, *channels
* used
* sizeof( int16_t ), mlt_pool_release
, NULL
);
136 memcpy( *buffer
, sample_buffer
, *channels
* used
* sizeof( int16_t ) );
138 // Update output variables
140 *frequency
= output_rate
;
146 /** Filter processing.
149 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
151 // Only call this if we have a means to get audio
152 if ( mlt_frame_is_test_audio( frame
) == 0 )
154 // Push the filter on to the stack
155 mlt_frame_push_audio( frame
, this );
157 // Assign our get_audio method
158 mlt_frame_push_audio( frame
, resample_get_audio
);
164 /** Constructor for the filter.
167 mlt_filter
filter_avresample_init( char *arg
)
170 mlt_filter
this = mlt_filter_new( );
172 // Initialise if successful
175 // Calculate size of the buffer
176 int size
= AVCODEC_MAX_AUDIO_FRAME_SIZE
* sizeof( int16_t );
178 // Allocate the buffer
179 int16_t *buffer
= mlt_pool_alloc( size
);
181 // Assign the process method
182 this->process
= filter_process
;
184 // Deal with argument
186 mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg
);
188 // Default to 2 channel output
189 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
192 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer
, size
, mlt_pool_release
, NULL
);