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Cleanup license declarations and remove dv1394d references.
[mlt] / src / modules / avformat / filter_avresample.c
1 /*
2  * filter_avresample.c -- adjust audio sample frequency
3  * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
4  * Author: Charles Yates <charles.yates@pandora.be>
5  *
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.
10  *
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.
15  *
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
19  */
20
21 #include "filter_avresample.h"
22
23 #include <framework/mlt_frame.h>
24
25 #include <stdio.h>
26 #include <stdlib.h>
27 #include <string.h>
28
29 // ffmpeg Header files
30 #include <avformat.h>
31
32 /** Get the audio.
33 */
34
35 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
36 {
37         // Get the properties of the frame
38         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
39
40         // Get the filter service
41         mlt_filter filter = mlt_frame_pop_audio( frame );
42
43         // Get the filter properties
44         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
45
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 );
49
50         // Obtain the resample context if it exists
51         ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
52
53         // Used to return number of channels in the source
54         int channels_avail = *channels;
55
56         // Loop variable
57         int i;
58
59         // If no resample frequency is specified, default to requested value
60         if ( output_rate == 0 )
61                 output_rate = *frequency;
62
63         // Get the producer's audio
64         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
65
66         // Duplicate channels as necessary
67         if ( channels_avail < *channels )
68         {
69                 int size = *channels * *samples * sizeof( int16_t );
70                 int16_t *new_buffer = mlt_pool_alloc( size );
71                 int j, k = 0;
72                 
73                 // Duplicate the existing channels
74                 for ( i = 0; i < *samples; i++ )
75                 {
76                         for ( j = 0; j < *channels; j++ )
77                         {
78                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
79                                 k = ( k + 1 ) % channels_avail;
80                         }
81                 }
82                 
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 );
85                 
86                 *buffer = new_buffer;
87         }
88         else if ( channels_avail == 6 && *channels == 2 )
89         {
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 );
93                 
94                 // Drop all but the first *channels
95                 for ( i = 0; i < *samples; i++ )
96                 {
97                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
98                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
99                 }
100
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 );
103                 
104                 *buffer = new_buffer;
105         }
106
107         // Return now if no work to do
108         if ( output_rate != *frequency )
109         {
110                 // Will store number of samples created
111                 int used = 0;
112
113                 // Create a resampler if nececessary
114                 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
115                 {
116                         // Create the resampler
117                         resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
118
119                         // And store it on properties
120                         mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
121
122                         // And remember what it was created for
123                         mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
124                 }
125
126                 // Resample the audio
127                 used = audio_resample( resample, sample_buffer, *buffer, *samples );
128
129                 // Resize if necessary
130                 if ( used > *samples )
131                 {
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 );
134                 }
135
136                 // Copy samples
137                 memcpy( *buffer, sample_buffer, *channels * used * sizeof( int16_t ) );
138
139                 // Update output variables
140                 *samples = used;
141                 *frequency = output_rate;
142         }
143
144         return 0;
145 }
146
147 /** Filter processing.
148 */
149
150 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
151 {
152         // Only call this if we have a means to get audio
153         if ( mlt_frame_is_test_audio( frame ) == 0 )
154         {
155                 // Push the filter on to the stack
156                 mlt_frame_push_audio( frame, this );
157
158                 // Assign our get_audio method
159                 mlt_frame_push_audio( frame, resample_get_audio );
160         }
161
162         return frame;
163 }
164
165 /** Constructor for the filter.
166 */
167
168 mlt_filter filter_avresample_init( char *arg )
169 {
170         // Create a filter
171         mlt_filter this = mlt_filter_new( );
172
173         // Initialise if successful
174         if ( this != NULL )
175         {
176                 // Calculate size of the buffer
177                 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
178
179                 // Allocate the buffer
180                 int16_t *buffer = mlt_pool_alloc( size );
181
182                 // Assign the process method
183                 this->process = filter_process;
184
185                 // Deal with argument
186                 if ( arg != NULL )
187                         mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
188
189                 // Default to 2 channel output
190                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
191
192                 // Store the buffer
193                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );
194         }
195
196         return this;
197 }