[melted] / src / modules / avformat / filter_avresample.c

/*
 * 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 "filter_avresample.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;
}