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[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 <framework/mlt_filter.h>
22 #include <framework/mlt_frame.h>
23
24 #include <stdio.h>
25 #include <stdlib.h>
26 #include <string.h>
27
28 // ffmpeg Header files
29 #include <avformat.h>
30
31 /** Get the audio.
32 */
33
34 static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
35 {
36         // Get the properties of the frame
37         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
38
39         // Get the filter service
40         mlt_filter filter = mlt_frame_pop_audio( frame );
41
42         // Get the filter properties
43         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
44
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 );
48
49         // Obtain the resample context if it exists
50         ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
51
52         // Used to return number of channels in the source
53         int channels_avail = *channels;
54
55         // Loop variable
56         int i;
57
58         // If no resample frequency is specified, default to requested value
59         if ( output_rate == 0 )
60                 output_rate = *frequency;
61
62         // Get the producer's audio
63         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
64
65         // Duplicate channels as necessary
66         if ( channels_avail < *channels )
67         {
68                 int size = *channels * *samples * sizeof( int16_t );
69                 int16_t *new_buffer = mlt_pool_alloc( size );
70                 int j, k = 0;
71                 
72                 // Duplicate the existing channels
73                 for ( i = 0; i < *samples; i++ )
74                 {
75                         for ( j = 0; j < *channels; j++ )
76                         {
77                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
78                                 k = ( k + 1 ) % channels_avail;
79                         }
80                 }
81                 
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 );
84                 
85                 *buffer = new_buffer;
86         }
87         else if ( channels_avail == 6 && *channels == 2 )
88         {
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 );
92                 
93                 // Drop all but the first *channels
94                 for ( i = 0; i < *samples; i++ )
95                 {
96                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
97                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
98                 }
99
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 );
102                 
103                 *buffer = new_buffer;
104         }
105
106         // Return now if no work to do
107         if ( output_rate != *frequency )
108         {
109                 // Will store number of samples created
110                 int used = 0;
111
112                 // Create a resampler if nececessary
113                 if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
114                 {
115                         // Create the resampler
116                         resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
117
118                         // And store it on properties
119                         mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
120
121                         // And remember what it was created for
122                         mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
123                 }
124
125                 // Resample the audio
126                 used = audio_resample( resample, sample_buffer, *buffer, *samples );
127
128                 // Resize if necessary
129                 if ( used > *samples )
130                 {
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 );
133                 }
134
135                 // Copy samples
136                 memcpy( *buffer, sample_buffer, *channels * used * sizeof( int16_t ) );
137
138                 // Update output variables
139                 *samples = used;
140                 *frequency = output_rate;
141         }
142
143         return 0;
144 }
145
146 /** Filter processing.
147 */
148
149 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
150 {
151         // Only call this if we have a means to get audio
152         if ( mlt_frame_is_test_audio( frame ) == 0 )
153         {
154                 // Push the filter on to the stack
155                 mlt_frame_push_audio( frame, this );
156
157                 // Assign our get_audio method
158                 mlt_frame_push_audio( frame, resample_get_audio );
159         }
160
161         return frame;
162 }
163
164 /** Constructor for the filter.
165 */
166
167 mlt_filter filter_avresample_init( char *arg )
168 {
169         // Create a filter
170         mlt_filter this = mlt_filter_new( );
171
172         // Initialise if successful
173         if ( this != NULL )
174         {
175                 // Calculate size of the buffer
176                 int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
177
178                 // Allocate the buffer
179                 int16_t *buffer = mlt_pool_alloc( size );
180
181                 // Assign the process method
182                 this->process = filter_process;
183
184                 // Deal with argument
185                 if ( arg != NULL )
186                         mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
187
188                 // Default to 2 channel output
189                 mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
190
191                 // Store the buffer
192                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );
193         }
194
195         return this;
196 }