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249 lines
6.8 KiB
C++
249 lines
6.8 KiB
C++
/* ScummVM - Scumm Interpreter
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* Copyright (C) 2001-2003 The ScummVM project
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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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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* 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
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*
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* $Header$
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*
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*/
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/*
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* The code in this file is based on code with Copyright 1998 Fabrice Bellard
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* Fabrice original code is part of SoX (http://sox.sourceforge.net).
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* Max Horn adapted that code to the needs of ScummVM and rewrote it partial,
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* in the process removing any use of floating point arithmetic. Various other
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* improvments over the original code were made.
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*/
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#include "stdafx.h"
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#include "sound/rate.h"
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#include "sound/audiostream.h"
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/**
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* The precision of the fractional computations used by the rate converter.
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* Normally you should never have to modify this value.
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*/
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#define FRAC_BITS 16
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/**
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* The size of the intermediate input cache. Bigger values may increase
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* performance, but only until some point (depends largely on cache size,
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* target processor and various other factors), at which it will decrease
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* again.
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*/
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#define INTERMEDIATE_BUFFER_SIZE 512
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/**
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* Audio rate converter based on simple linear Interpolation.
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*
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* The use of fractional increment allows us to use no buffer. It
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* avoid the problems at the end of the buffer we had with the old
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* method which stored a possibly big buffer of size
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* lcm(in_rate,out_rate).
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*
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* Limited to sampling frequency <= 65535 Hz.
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*/
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template<bool stereo, bool reverseStereo>
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class LinearRateConverter : public RateConverter {
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protected:
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st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
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const st_sample_t *inPtr;
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int inLen;
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/** fractional position of the output stream in input stream unit */
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unsigned long opos, opos_frac;
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/** fractional position increment in the output stream */
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unsigned long opos_inc, opos_inc_frac;
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/** position in the input stream (integer) */
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unsigned long ipos;
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/** last sample(s) in the input stream (left/right channel) */
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st_sample_t ilast[2];
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/** current sample(s) in the input stream (left/right channel) */
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st_sample_t icur[2];
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public:
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LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
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int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
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int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
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return (ST_SUCCESS);
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}
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};
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/*
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* Prepare processing.
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*/
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template<bool stereo, bool reverseStereo>
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LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
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unsigned long incr;
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if (inrate == outrate) {
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error("Input and Output rates must be different to use rate effect");
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}
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if (inrate >= 65536 || outrate >= 65536) {
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error("rate effect can only handle rates < 65536");
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}
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opos_frac = 0;
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opos = 1;
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/* increment */
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incr = (inrate << FRAC_BITS) / outrate;
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opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1);
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opos_inc = incr >> FRAC_BITS;
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ipos = 0;
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ilast[0] = ilast[1] = 0;
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icur[0] = icur[1] = 0;
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inLen = 0;
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}
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/*
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* Processed signed long samples from ibuf to obuf.
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* Return number of samples processed.
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*/
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template<bool stereo, bool reverseStereo>
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int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol)
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{
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st_sample_t *ostart, *oend;
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st_sample_t out[2], tmpOut;
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const int numChannels = stereo ? 2 : 1;
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int i;
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ostart = obuf;
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oend = obuf + osamp * 2;
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while (obuf < oend) {
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// read enough input samples so that ipos > opos
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while (ipos <= opos) {
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// Check if we have to refill the buffer
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if (inLen == 0) {
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inPtr = inBuf;
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inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf));
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if (inLen <= 0)
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goto the_end;
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}
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for (i = 0; i < numChannels; i++) {
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ilast[i] = icur[i];
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icur[i] = *inPtr++;
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inLen--;
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}
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ipos++;
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}
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// Loop as long as the outpos trails behind, and as long as there is
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// still space in the output buffer.
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while (ipos > opos) {
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// interpolate
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tmpOut = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
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// adjust volume
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out[0] = out[1] = (st_sample_t)((tmpOut * vol) >> 8);
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if (stereo) {
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// interpolate
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tmpOut = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
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// adjust volume
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out[reverseStereo ? 0 : 1] = (st_sample_t)((tmpOut * vol) >> 8);
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}
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// output left channel
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clampedAdd(*obuf++, out[0]);
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// output right channel
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clampedAdd(*obuf++, out[1]);
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// Increment output position
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unsigned long tmp = opos_frac + opos_inc_frac;
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opos += opos_inc + (tmp >> FRAC_BITS);
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opos_frac = tmp & ((1UL << FRAC_BITS) - 1);
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// Abort if we reached the end of the output buffer
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if (obuf >= oend)
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goto the_end;
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}
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}
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the_end:
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return (ST_SUCCESS);
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}
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#pragma mark -
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/**
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* Simple audio rate converter for the case that the inrate equals the outrate.
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*/
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template<bool stereo, bool reverseStereo>
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class CopyRateConverter : public RateConverter {
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public:
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virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
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int16 tmp[2];
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st_size_t len = osamp;
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assert(input.isStereo() == stereo);
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while (!input.eos() && len--) {
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tmp[0] = tmp[1] = (input.read() * vol) >> 8;
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if (stereo)
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tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
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clampedAdd(*obuf++, tmp[0]);
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clampedAdd(*obuf++, tmp[1]);
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}
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return (ST_SUCCESS);
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}
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virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
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return (ST_SUCCESS);
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}
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};
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#pragma mark -
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/**
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* Create and return a RateConverter object for the specified input and output rates.
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*/
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RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
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if (inrate != outrate) {
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if (stereo) {
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if (reverseStereo)
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return new LinearRateConverter<true, true>(inrate, outrate);
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else
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return new LinearRateConverter<true, false>(inrate, outrate);
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} else
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return new LinearRateConverter<false, false>(inrate, outrate);
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//return new ResampleRateConverter(inrate, outrate, 1);
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} else {
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if (stereo) {
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if (reverseStereo)
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return new CopyRateConverter<true, true>();
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else
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return new CopyRateConverter<true, false>();
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} else
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return new CopyRateConverter<false, false>();
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}
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}
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