📄 fixed32tos16.c
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/***************************************************************************** * fixed32tos16.c : converter from fixed32 to signed 16 bits integer ***************************************************************************** * Copyright (C) 2002 VideoLAN * $Id: fixed32tos16.c 10101 2005-03-02 16:47:31Z robux4 $ * * Authors: Jean-Paul Saman <jpsaman@wxs.nl> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA. *****************************************************************************//***************************************************************************** * Preamble *****************************************************************************/#include <stdlib.h> /* malloc(), free() */#include <string.h>#include <vlc/vlc.h>#include "audio_output.h"#include "aout_internal.h"/***************************************************************************** * Local prototypes *****************************************************************************/static int Create ( vlc_object_t * );static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *, aout_buffer_t * );/***************************************************************************** * Module descriptor *****************************************************************************/vlc_module_begin(); set_category( CAT_AUDIO ); set_subcategory( SUBCAT_AUDIO_MISC ); set_description( _("audio filter for fixed32->s16 conversion") ); set_capability( "audio filter", 10 ); set_callbacks( Create, NULL );vlc_module_end();/***************************************************************************** * Create: allocate trivial mixer ***************************************************************************** * This function allocates and initializes a Crop vout method. *****************************************************************************/static int Create( vlc_object_t *p_this ){ aout_filter_t * p_filter = (aout_filter_t *)p_this; if ( p_filter->input.i_format != VLC_FOURCC('f','i','3','2') || p_filter->output.i_format != AOUT_FMT_S16_NE ) { return -1; } if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) ) { return -1; } p_filter->pf_do_work = DoWork; p_filter->b_in_place = 1; return 0;}/***************************************************************************** * support routines borrowed from mpg321 (file: mad.c), which is distributed * under GPL license * * mpg321 was written by Joe Drew <drew@debian.org>, and based upon 'plaympeg' * from the smpeg sources, which was written by various people from Loki Software * (http://www.lokigames.com). * * It also incorporates some source from mad, written by Robert Leslie *****************************************************************************//* The following two routines and data structure are from the ever-brilliant Rob Leslie.*/#define VLC_F_FRACBITS 28# if VLC_F_FRACBITS == 28# define VLC_F(x) ((vlc_fixed_t) (x##L))# endif# define VLC_F_ONE VLC_F(0x10000000)struct audio_dither { vlc_fixed_t error[3]; vlc_fixed_t random;};/******************************************************************** * NAME: prng() * DESCRIPTION: 32-bit pseudo-random number generator ********************************************************************/static inline unsigned long prng(unsigned long state){ return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;}/******************************************************************** * NAME: mpg321_s24_to_s16_pcm() * DESCRIPTION: generic linear sample quantize and dither routine ********************************************************************/static inline int16_t mpg321_s24_to_s16_pcm(unsigned int bits, vlc_fixed_t sample, struct audio_dither *dither){ unsigned int scalebits; vlc_fixed_t output, mask, random; enum { MIN = -VLC_F_ONE, MAX = VLC_F_ONE - 1 }; /* noise shape */ sample += dither->error[0] - dither->error[1] + dither->error[2]; dither->error[2] = dither->error[1]; dither->error[1] = dither->error[0] / 2; /* bias */ output = sample + (1L << (VLC_F_FRACBITS + 1 - bits - 1)); scalebits = VLC_F_FRACBITS + 1 - bits; mask = (1L << scalebits) - 1; /* dither */ random = prng(dither->random); output += (random & mask) - (dither->random & mask); dither->random = random; /* clip */ if (output > MAX) { output = MAX; if (sample > MAX) sample = MAX; } else if (output < MIN) { output = MIN; if (sample < MIN) sample = MIN; } /* quantize */ output &= ~mask; /* error feedback */ dither->error[0] = sample - output; /* scale */ return output >> scalebits;}/***************************************************************************** * s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample. *****************************************************************************/static inline int16_t s24_to_s16_pcm(vlc_fixed_t sample){ /* round */ sample += (1L << (VLC_F_FRACBITS - 16)); /* clip */ if (sample >= VLC_F_ONE) sample = VLC_F_ONE - 1; else if (sample < -VLC_F_ONE) sample = -VLC_F_ONE; /* quantize */ return (sample >> (VLC_F_FRACBITS + 1 - 16));}/***************************************************************************** * DoWork: convert a buffer *****************************************************************************/static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter, aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf ){ int i; vlc_fixed_t * p_in = (vlc_fixed_t *)p_in_buf->p_buffer; int16_t * p_out = (int16_t *)p_out_buf->p_buffer;#if 0 static struct audio_dither dither;#endif for ( i = p_in_buf->i_nb_samples * aout_FormatNbChannels( &p_filter->input ) ; i-- ; ) {#if 0 /* Accurate scaling */ *p_out++ = mpg321_s24_to_s16_pcm(16, *p_in++, &dither);#endif /* Fast Scaling */ *p_out++ = s24_to_s16_pcm(*p_in++); } p_out_buf->i_nb_samples = p_in_buf->i_nb_samples; p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes / 2;}
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