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198 lines
5.8 KiB
198 lines
5.8 KiB
/*
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* filter_avresample.c -- adjust audio sample frequency
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* Copyright (C) 2003-2004 Ushodaya Enterprises Limited
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* Author: Charles Yates <charles.yates@pandora.be>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library 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 GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "filter_avresample.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 <string.h>
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// ffmpeg Header files
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#include <avformat.h>
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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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int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
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// Obtain the resample context if it exists
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ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
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// Used to return number of channels in the source
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int channels_avail = *channels;
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// Loop variable
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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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// Will store number of samples created
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int used = 0;
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// Create a resampler if nececessary
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if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
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{
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// Create the resampler
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resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
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// And store it on properties
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mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
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// And remember what it was created for
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mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
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}
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// Resample the audio
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used = audio_resample( resample, sample_buffer, *buffer, *samples );
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// Resize if necessary
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if ( used > *samples )
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{
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*buffer = mlt_pool_realloc( *buffer, *samples * *channels * sizeof( int16_t ) );
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mlt_properties_set_data( properties, "audio", *buffer, *channels * used * sizeof( int16_t ), mlt_pool_release, NULL );
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}
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// Copy samples
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memcpy( *buffer, sample_buffer, *channels * used * sizeof( int16_t ) );
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// Update output variables
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*samples = used;
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*frequency = 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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// Only call this if we have a means to get audio
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if ( mlt_frame_is_test_audio( frame ) == 0 )
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{
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// Push the filter on to the stack
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mlt_frame_push_audio( frame, this );
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// Assign our get_audio method
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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_avresample_init( char *arg )
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{
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// Create a filter
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mlt_filter this = mlt_filter_new( );
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// Initialise if successful
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if ( this != NULL )
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{
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// Calculate size of the buffer
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int size = /*AVCODEC_MAX_AUDIO_FRAME_SIZE*/ 192000 * sizeof( int16_t );
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// Allocate the buffer
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int16_t *buffer = mlt_pool_alloc( size );
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// Assign the process method
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this->process = filter_process;
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// Deal with argument
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if ( arg != NULL )
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mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
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// Default to 2 channel output
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mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
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// Store the buffer
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mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );
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}
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return this;
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}
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