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https://github.com/libretro/scummvm.git
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359 lines
9.7 KiB
C++
359 lines
9.7 KiB
C++
/* ScummVM - Graphic Adventure Engine
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*
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* ScummVM is the legal property of its developers, whose names
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* are too numerous to list here. Please refer to the COPYRIGHT
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* file distributed with this source distribution.
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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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*
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* $URL$
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* $Id$
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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 "common/stdafx.h"
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#include "sound/audiostream.h"
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#include "sound/rate.h"
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#include "sound/mixer.h"
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#include "common/frac.h"
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#include "common/util.h"
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namespace Audio {
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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 resampling. Used when no
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* interpolation is required.
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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 SimpleRateConverter : 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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/** position of how far output is ahead of input */
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/** Holds what would have been opos-ipos */
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long opos;
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/** fractional position increment in the output stream */
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long opos_inc;
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public:
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SimpleRateConverter(st_rate_t inrate, st_rate_t outrate);
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int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
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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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SimpleRateConverter<stereo, reverseStereo>::SimpleRateConverter(st_rate_t inrate, st_rate_t outrate) {
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if ((inrate % outrate) != 0) {
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error("Input rate must be a multiple of output rate 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 = 1;
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/* increment */
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opos_inc = inrate / outrate;
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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 SimpleRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
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st_sample_t *ostart, *oend;
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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 opos >= 0
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do {
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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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return ST_EOF;
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}
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inLen -= (stereo ? 2 : 1);
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opos--;
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if (opos >= 0) {
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inPtr += (stereo ? 2 : 1);
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}
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} while (opos >= 0);
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st_sample_t out0, out1;
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out0 = *inPtr++;
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out1 = (stereo ? *inPtr++ : out0);
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// Increment output position
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opos += opos_inc;
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// output left channel
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clampedAdd(obuf[reverseStereo ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
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// output right channel
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clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
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obuf += 2;
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}
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return ST_SUCCESS;
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}
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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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frac_t opos;
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/** fractional position increment in the output stream */
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frac_t opos_inc;
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/** last sample(s) in the input stream (left/right channel) */
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st_sample_t ilast0, ilast1;
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/** current sample(s) in the input stream (left/right channel) */
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st_sample_t icur0, icur1;
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public:
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LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
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int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
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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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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_ONE;
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// Compute the linear interpolation increment.
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// This will overflow if inrate >= 2^16, and underflow if outrate >= 2^16.
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// Also, if the quotient of the two rate becomes too small / too big, that
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// would cause problems, but since we rarely scale from 1 to 65536 Hz or vice
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// versa, I think we can live with that limiation ;-).
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opos_inc = (inrate << FRAC_BITS) / outrate;
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ilast0 = ilast1 = 0;
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icur0 = icur1 = 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(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
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st_sample_t *ostart, *oend;
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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 opos < 0
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while (0 <= 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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return ST_EOF;
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}
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inLen -= (stereo ? 2 : 1);
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ilast0 = icur0;
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icur0 = *inPtr++;
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if (stereo) {
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ilast1 = icur1;
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icur1 = *inPtr++;
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}
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opos -= FRAC_ONE;
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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 (0 > opos && obuf < oend) {
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// interpolate
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st_sample_t out0, out1;
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const frac_t scale = (opos & FRAC_LO_MASK);
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out0 = (st_sample_t)(ilast0 + (((icur0 - ilast0) * scale + FRAC_HALF) >> FRAC_BITS));
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out1 = (stereo ?
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(st_sample_t)(ilast1 + (((icur1 - ilast1) * scale + FRAC_HALF) >> FRAC_BITS)) :
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out0);
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// output left channel
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clampedAdd(obuf[reverseStereo ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
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// output right channel
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clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
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obuf += 2;
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// Increment output position
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opos += opos_inc;
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}
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}
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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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st_sample_t *_buffer;
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st_size_t _bufferSize;
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public:
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CopyRateConverter() : _buffer(0), _bufferSize(0) {}
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~CopyRateConverter() {
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free(_buffer);
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}
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virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
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assert(input.isStereo() == stereo);
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st_sample_t *ptr;
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st_size_t len;
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if (stereo)
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osamp *= 2;
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// Reallocate temp buffer, if necessary
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if (osamp > _bufferSize) {
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free(_buffer);
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_buffer = (st_sample_t *)malloc(osamp * 2);
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_bufferSize = osamp;
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}
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// Read up to 'osamp' samples into our temporary buffer
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len = input.readBuffer(_buffer, osamp);
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// Mix the data into the output buffer
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ptr = _buffer;
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for (; len > 0; len -= (stereo ? 2 : 1)) {
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st_sample_t out0, out1;
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out0 = *ptr++;
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out1 = (stereo ? *ptr++ : out0);
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// output left channel
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clampedAdd(obuf[reverseStereo ], (out0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
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// output right channel
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clampedAdd(obuf[reverseStereo ^ 1], (out1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
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obuf += 2;
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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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template<bool stereo, bool reverseStereo>
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RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate) {
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if (inrate != outrate) {
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if ((inrate % outrate) == 0) {
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return new SimpleRateConverter<stereo, reverseStereo>(inrate, outrate);
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} else {
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return new LinearRateConverter<stereo, reverseStereo>(inrate, outrate);
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}
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} else {
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return new CopyRateConverter<stereo, reverseStereo>();
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}
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}
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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 (stereo) {
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if (reverseStereo)
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return makeRateConverter<true, true>(inrate, outrate);
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else
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return makeRateConverter<true, false>(inrate, outrate);
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} else
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return makeRateConverter<false, false>(inrate, outrate);
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}
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} // End of namespace Audio
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