+++ /dev/null
-/*
- * filter_avresample.c -- adjust audio sample frequency
- * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
- * Author: Charles Yates <charles.yates@pandora.be>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- */
-
-#include <framework/mlt_filter.h>
-#include <framework/mlt_frame.h>
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-// ffmpeg Header files
-#include <avformat.h>
-
-/** Get the audio.
-*/
-
-static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
-{
- // Get the properties of the frame
- mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
-
- // Get the filter service
- mlt_filter filter = mlt_frame_pop_audio( frame );
-
- // Get the filter properties
- mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
-
- // Get the resample information
- int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
- int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
-
- // Obtain the resample context if it exists
- ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
-
- // Used to return number of channels in the source
- int channels_avail = *channels;
-
- // Loop variable
- int i;
-
- // If no resample frequency is specified, default to requested value
- if ( output_rate == 0 )
- output_rate = *frequency;
-
- // Get the producer's audio
- mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
-
- // Duplicate channels as necessary
- if ( channels_avail < *channels )
- {
- int size = *channels * *samples * sizeof( int16_t );
- int16_t *new_buffer = mlt_pool_alloc( size );
- int j, k = 0;
-
- // Duplicate the existing channels
- for ( i = 0; i < *samples; i++ )
- {
- for ( j = 0; j < *channels; j++ )
- {
- new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
- k = ( k + 1 ) % channels_avail;
- }
- }
-
- // Update the audio buffer now - destroys the old
- mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-
- *buffer = new_buffer;
- }
- else if ( channels_avail == 6 && *channels == 2 )
- {
- // Nasty hack for ac3 5.1 audio - may be a cause of failure?
- int size = *channels * *samples * sizeof( int16_t );
- int16_t *new_buffer = mlt_pool_alloc( size );
-
- // Drop all but the first *channels
- for ( i = 0; i < *samples; i++ )
- {
- new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
- new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
- }
-
- // Update the audio buffer now - destroys the old
- mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-
- *buffer = new_buffer;
- }
-
- // Return now if no work to do
- if ( output_rate != *frequency )
- {
- // Will store number of samples created
- int used = 0;
-
- // Create a resampler if nececessary
- if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
- {
- // Create the resampler
- resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
-
- // And store it on properties
- mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
-
- // And remember what it was created for
- mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
- }
-
- // Resample the audio
- used = audio_resample( resample, sample_buffer, *buffer, *samples );
-
- // Resize if necessary
- if ( used > *samples )
- {
- *buffer = mlt_pool_realloc( *buffer, *samples * *channels * sizeof( int16_t ) );
- mlt_properties_set_data( properties, "audio", *buffer, *channels * used * sizeof( int16_t ), mlt_pool_release, NULL );
- }
-
- // Copy samples
- memcpy( *buffer, sample_buffer, *channels * used * sizeof( int16_t ) );
-
- // Update output variables
- *samples = used;
- *frequency = output_rate;
- }
-
- return 0;
-}
-
-/** Filter processing.
-*/
-
-static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
-{
- // Only call this if we have a means to get audio
- if ( mlt_frame_is_test_audio( frame ) == 0 )
- {
- // Push the filter on to the stack
- mlt_frame_push_audio( frame, this );
-
- // Assign our get_audio method
- mlt_frame_push_audio( frame, resample_get_audio );
- }
-
- return frame;
-}
-
-/** Constructor for the filter.
-*/
-
-mlt_filter filter_avresample_init( char *arg )
-{
- // Create a filter
- mlt_filter this = mlt_filter_new( );
-
- // Initialise if successful
- if ( this != NULL )
- {
- // Calculate size of the buffer
- int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
-
- // Allocate the buffer
- int16_t *buffer = mlt_pool_alloc( size );
-
- // Assign the process method
- this->process = filter_process;
-
- // Deal with argument
- if ( arg != NULL )
- mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
-
- // Default to 2 channel output
- mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
-
- // Store the buffer
- mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );
- }
-
- return this;
-}
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