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202 lines
6.1 KiB
202 lines
6.1 KiB
/*
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* filter_resample.c -- adjust audio sample frequency
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* Copyright (C) 2003-2004 Ushodaya Enterprises Limited
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* Author: Dan Dennedy <dan@dennedy.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*/
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#include "filter_resample.h"
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#include <framework/mlt_frame.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <samplerate.h>
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#define __USE_ISOC99 1
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#include <math.h>
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#define BUFFER_LEN 20480
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#define RESAMPLE_TYPE SRC_SINC_FASTEST
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/** Get the audio.
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*/
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static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
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{
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// Get the properties of the frame
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mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
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// Get the filter service
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mlt_filter filter = mlt_frame_pop_audio( frame );
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// Get the filter properties
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mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
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// Get the resample information
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int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
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SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
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float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
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float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
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int channels_avail = *channels;
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SRC_DATA data;
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int i;
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// If no resample frequency is specified, default to requested value
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if ( output_rate == 0 )
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output_rate = *frequency;
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// Get the producer's audio
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mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
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// Duplicate channels as necessary
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if ( channels_avail < *channels )
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{
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int size = *channels * *samples * sizeof( int16_t );
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int16_t *new_buffer = mlt_pool_alloc( size );
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int j, k = 0;
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// Duplicate the existing channels
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for ( i = 0; i < *samples; i++ )
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{
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for ( j = 0; j < *channels; j++ )
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{
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new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
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k = ( k + 1 ) % channels_avail;
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}
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}
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// Update the audio buffer now - destroys the old
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mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
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*buffer = new_buffer;
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}
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else if ( channels_avail == 6 && *channels == 2 )
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{
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// Nasty hack for ac3 5.1 audio - may be a cause of failure?
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int size = *channels * *samples * sizeof( int16_t );
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int16_t *new_buffer = mlt_pool_alloc( size );
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// Drop all but the first *channels
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for ( i = 0; i < *samples; i++ )
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{
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new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
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new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
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}
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// Update the audio buffer now - destroys the old
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mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
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*buffer = new_buffer;
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}
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// Return now if no work to do
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if ( output_rate != *frequency )
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{
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float *p = input_buffer;
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float *end = p + *samples * *channels;
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int16_t *q = *buffer;
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// Convert to floating point
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while( p != end )
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*p ++ = ( float )( *q ++ ) / 32768.0;
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// Resample
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data.data_in = input_buffer;
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data.data_out = output_buffer;
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data.src_ratio = ( float ) output_rate / ( float ) *frequency;
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data.input_frames = *samples;
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data.output_frames = BUFFER_LEN / *channels;
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data.end_of_input = 0;
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i = src_process( state, &data );
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if ( i == 0 )
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{
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if ( data.output_frames_gen > *samples )
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{
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*buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
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mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
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}
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*samples = data.output_frames_gen;
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*frequency = output_rate;
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p = output_buffer;
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q = *buffer;
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end = p + *samples * *channels;
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// Convert from floating back to signed 16bit
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while( p != end )
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{
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if ( *p > 1.0 )
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*p = 1.0;
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if ( *p < -1.0 )
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*p = -1.0;
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if ( *p > 0 )
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*q ++ = 32767 * *p ++;
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else
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*q ++ = 32768 * *p ++;
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}
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}
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else
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fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
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}
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return 0;
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}
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/** Filter processing.
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*/
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static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
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{
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if ( mlt_frame_is_test_audio( frame ) == 0 )
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{
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mlt_frame_push_audio( frame, this );
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mlt_frame_push_audio( frame, resample_get_audio );
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}
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return frame;
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}
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/** Constructor for the filter.
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*/
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mlt_filter filter_resample_init( char *arg )
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{
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mlt_filter this = mlt_filter_new( );
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if ( this != NULL )
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{
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int error;
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SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
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if ( error == 0 )
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{
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void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
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void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
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this->process = filter_process;
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if ( arg != NULL )
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mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
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mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
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mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
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mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
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mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
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}
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else
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{
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fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
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}
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}
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return this;
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}
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