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some cleanup

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svn-id: r9575
This commit is contained in:
Max Horn 2003-08-06 17:13:04 +00:00
parent 32107ae69a
commit 69ee268e7f
5 changed files with 90 additions and 78 deletions
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1
sound/mixer.cpp
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@ -27,6 +27,7 @@
#include "sound/mixer.h" #include "sound/mixer.h"
#include "sound/rate.h" #include "sound/rate.h"
#include "sound/audiostream.h"
#pragma mark - #pragma mark -

1
sound/rate.cpp
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@ -29,6 +29,7 @@
#include "stdafx.h" #include "stdafx.h"
#include "sound/rate.h" #include "sound/rate.h"
#include "sound/audiostream.h"
/** /**
* The precision of the fractional computations used by the rate converter. * The precision of the fractional computations used by the rate converter.

5
sound/rate.h
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@ -22,13 +22,12 @@
#ifndef SOUND_RATE_H #ifndef SOUND_RATE_H
#define SOUND_RATE_H #define SOUND_RATE_H
#include <stdio.h>
#include <assert.h>
#include "common/scummsys.h" #include "common/scummsys.h"
#include "common/engine.h" #include "common/engine.h"
#include "common/util.h" #include "common/util.h"
#include "sound/audiostream.h" //#include "sound/audiostream.h"
class AudioInputStream;
typedef int16 st_sample_t; typedef int16 st_sample_t;
typedef uint16 st_volume_t; typedef uint16 st_volume_t;

68
sound/resample.cpp
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@ -50,18 +50,55 @@
* Get the idea? :) * Get the idea? :)
*/ */
#include "stdafx.h"
#include <math.h> #include <math.h>
#include <stdlib.h> #include "sound/resample.h"
#include <string.h> #include "sound/audiostream.h"
#include "rate.h"
/* resample includes */
#include "resample.h"
typedef struct { /* Conversion constants */
byte priv[1024]; #define Lc 7
} eff_struct; #define Nc (1<<Lc)
typedef eff_struct *eff_t; #define La 16
#define Na (1<<La)
#define Lp (Lc+La)
#define Np (1<<Lp)
#define Amask (Na-1)
#define Pmask (Np-1)
#define MAXNWING (80<<Lc)
/* Description of constants:
*
* Nc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Nc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Nc must be a power of 2 due to the details of the current
* implementation. The default value of 128 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit precision, and quadratic interpolation
* gives about 23-bit (float) precision in filter coefficients.
*
* Lc - is log base 2 of Nc.
*
* La - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Lp - is La + Lc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. The default value of 23 is
* about right. There is a constraint that the filter window must be
* "addressable" in a int32_t, more precisely, if Nmult is the number
* of sinc zero-crossings in the right wing of the filter window, then
* (Nwing<<Lp) must be expressible in 31 bits.
*
*/
/* this Float MUST match that in filter.c */ /* this Float MUST match that in filter.c */
#define Float double/*float*/ #define Float double/*float*/
@ -284,6 +321,7 @@ int st_resample_flow(eff_t effp, AudioInputStream &input, st_sample_t *obuf, st_
long Nproc; // The number of bytes we process to generate Nout output bytes long Nproc; // The number of bytes we process to generate Nout output bytes
const long obufSize = *osamp; const long obufSize = *osamp;
/*
TODO: adjust for the changes made to AudioInputStream; add support for stereo TODO: adjust for the changes made to AudioInputStream; add support for stereo
initially, could just average the left/right channel -> bad for quality of course, initially, could just average the left/right channel -> bad for quality of course,
but easiest to implement and would get this going again. but easiest to implement and would get this going again.
@ -293,6 +331,7 @@ But better for efficiency would be to rewrite those to deal with 2 channels, too
Because esp in SrcEX/SrcUD, only very few computations depend on the input data, Because esp in SrcEX/SrcUD, only very few computations depend on the input data,
and dealing with both channels in parallel should only be a little slower than dealing and dealing with both channels in parallel should only be a little slower than dealing
with them alone with them alone
*/
// Constrain amount we actually process // Constrain amount we actually process
//fprintf(stderr,"Xp %d, Xread %d\n",r->Xp, r->Xread); //fprintf(stderr,"Xp %d, Xread %d\n",r->Xp, r->Xread);
@ -740,17 +779,6 @@ static void LpFilter(double *c, long N, double frq, double Beta, long Num) {
#pragma mark - #pragma mark -
class ResampleRateConverter : public RateConverter {
protected:
eff_struct effp;
public:
ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality);
~ResampleRateConverter();
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
};
ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) { ResampleRateConverter::ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality) {
// FIXME: quality is for now a nasty hack. // FIXME: quality is for now a nasty hack.
// Valid values are 0,1,2,3 (everything else is treated like 0 for now) // Valid values are 0,1,2,3 (everything else is treated like 0 for now)

93
sound/resample.h
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@ -1,61 +1,44 @@
/* /* ScummVM - Scumm Interpreter
* FILE: resample.h * Copyright (C) 2001-2003 The ScummVM project
* BY: Julius Smith (at CCRMA, Stanford U)
* C BY: translated from SAIL to C by Christopher Lee Fraley
* (cf0v@andrew.cmu.edu)
* DATE: 7-JUN-88
* VERS: 2.0 (17-JUN-88, 3:00pm)
*/
/*
* October 29, 1999
* Various changes, bugfixes(?), increased precision, by Stan Brooks.
* *
* This source code is distributed in the hope that it will be useful, * 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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. * 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 02111-1307, USA.
*
* $Header$
* *
*/ */
/* Conversion constants */ #ifndef SOUND_RESAMPLE_H
#define Lc 7 #define SOUND_RESAMPLE_H
#define Nc (1<<Lc)
#define La 16
#define Na (1<<La)
#define Lp (Lc+La)
#define Np (1<<Lp)
#define Amask (Na-1)
#define Pmask (Np-1)
#define MAXNWING (80<<Lc) #include "sound/rate.h"
/* Description of constants:
* typedef struct {
* Nc - is the number of look-up values available for the lowpass filter byte priv[1024];
* between the beginning of its impulse response and the "cutoff time" } eff_struct;
* of the filter. The cutoff time is defined as the reciprocal of the typedef eff_struct *eff_t;
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Nc would be the index of the /** High quality rate conversion algorithm, based on SoX (http://sox.sourceforge.net). */
* impulse-response lookup-table corresponding to the first zero- class ResampleRateConverter : public RateConverter {
* crossing of the sinc function. (The inverse first zero-crossing protected:
* time of a sinc function equals its nominal cutoff frequency in Hz.) eff_struct effp;
* Nc must be a power of 2 due to the details of the current public:
* implementation. The default value of 128 is sufficiently high that ResampleRateConverter(st_rate_t inrate, st_rate_t outrate, int quality);
* using linear interpolation to fill in between the table entries ~ResampleRateConverter();
* gives approximately 16-bit precision, and quadratic interpolation virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
* gives about 23-bit (float) precision in filter coefficients. virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
* };
* Lc - is log base 2 of Nc.
*
* La - is the number of bits devoted to linear interpolation of the #endif
* filter coefficients.
*
* Lp - is La + Lc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. The default value of 23 is
* about right. There is a constraint that the filter window must be
* "addressable" in a int32_t, more precisely, if Nmult is the number
* of sinc zero-crossings in the right wing of the filter window, then
* (Nwing<<Lp) must be expressible in 31 bits.
*
*/