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scummvm/sound/rate_arm.cpp

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/* ScummVM - Graphic Adventure Engine
*
* ScummVM is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*
*/
/*
* The code in this file, together with the rate_arm_asm.s file offers
* an ARM optimised version of the code in rate.cpp. The operation of this
* code should be identical to that of rate.cpp, but faster. The heavy
* lifting is done in the assembler file.
*
* To be as portable as possible we implement the core routines with C
* linkage in assembly, and implement the C++ routines that call into
* the C here. The C++ symbol mangling varies wildly between compilers,
* so this is the simplest way to ensure that the C/C++ combination should
* work on as many ARM based platforms as possible.
*
* Essentially the algorithm herein is the same as that in rate.cpp, so
* anyone seeking to understand this should attempt to understand that
* first. That code was based in turn on code with Copyright 1998 Fabrice
* Bellard - part of SoX (http://sox.sourceforge.net).
* Max Horn adapted that code to the needs of ScummVM and partially rewrote
* it, in the process removing any use of floating point arithmetic. Various
* other improvments over the original code were made.
*/
#include "sound/audiostream.h"
#include "sound/rate.h"
#include "sound/mixer.h"
#include "common/util.h"
//#define DEBUG_RATECONV
namespace Audio {
/**
* The precision of the fractional computations used by the rate converter.
* Normally you should never have to modify this value.
* This stuff is defined in common/frac.h, but we redefine it here as the
* ARM routine we call doesn't respect those definitions.
*/
#define FRAC_BITS 16
#define FRAC_ONE (1<<FRAC_BITS)
/**
* The size of the intermediate input cache. Bigger values may increase
* performance, but only until some point (depends largely on cache size,
* target processor and various other factors), at which it will decrease
* again.
*/
#define INTERMEDIATE_BUFFER_SIZE 512
/**
* Audio rate converter based on simple resampling. Used when no
* interpolation is required.
*
* Limited to sampling frequency <= 65535 Hz.
*/
typedef struct {
const st_sample_t *inPtr;
int inLen;
/** position of how far output is ahead of input */
/** Holds what would have been opos-ipos */
long opos;
/** fractional position increment in the output stream */
long opos_inc;
st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
} SimpleRateDetails;
template<bool stereo, bool reverseStereo>
class SimpleRateConverter : public RateConverter {
protected:
SimpleRateDetails sr;
public:
SimpleRateConverter(st_rate_t inrate, st_rate_t outrate);
int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
/*
* Prepare processing.
*/
template<bool stereo, bool reverseStereo>
SimpleRateConverter<stereo, reverseStereo>::SimpleRateConverter(st_rate_t inrate, st_rate_t outrate) {
if (inrate == outrate) {
error("Input and Output rates must be different to use rate effect");
}
if ((inrate % outrate) != 0) {
error("Input rate must be a multiple of Output rate to use rate effect");
}
if (inrate >= 65536 || outrate >= 65536) {
error("rate effect can only handle rates < 65536");
}
sr.opos = 1;
/* increment */
sr.opos_inc = inrate / outrate;
sr.inLen = 0;
}
extern "C" {
#ifndef IPHONE
#define ARM_SimpleRate_M _ARM_SimpleRate_M
#define ARM_SimpleRate_S _ARM_SimpleRate_S
#define ARM_SimpleRate_R _ARM_SimpleRate_R
#endif
}
extern "C" void ARM_SimpleRate_M(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
SimpleRateDetails *sr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
extern "C" void ARM_SimpleRate_S(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
SimpleRateDetails *sr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
extern "C" void ARM_SimpleRate_R(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
SimpleRateDetails *sr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
extern "C" int SimpleRate_readFudge(Audio::AudioStream &input,
int16 *a, int b)
{
#ifdef DEBUG_RATECONV
fprintf(stderr, "Reading ptr=%x n%d\n", a, b);
fflush(stderr);
#endif
return input.readBuffer(a, b);
}
template<bool stereo, bool reverseStereo>
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) {
#ifdef DEBUG_RATECONV
fprintf(stderr, "Simple st=%d rev=%d\n", stereo, reverseStereo);
fflush(stderr);
#endif
if (!stereo) {
ARM_SimpleRate_M(input,
&SimpleRate_readFudge,
&sr,
obuf, osamp, vol_l, vol_r);
} else if (reverseStereo) {
ARM_SimpleRate_R(input,
&SimpleRate_readFudge,
&sr,
obuf, osamp, vol_l, vol_r);
} else {
ARM_SimpleRate_S(input,
&SimpleRate_readFudge,
&sr,
obuf, osamp, vol_l, vol_r);
}
return (ST_SUCCESS);
}
/**
* Audio rate converter based on simple linear Interpolation.
*
* The use of fractional increment allows us to use no buffer. It
* avoid the problems at the end of the buffer we had with the old
* method which stored a possibly big buffer of size
* lcm(in_rate,out_rate).
*
* Limited to sampling frequency <= 65535 Hz.
*/
typedef struct {
const st_sample_t *inPtr;
int inLen;
/** position of how far output is ahead of input */
/** Holds what would have been opos-ipos<<16 + opos_frac */
long opos;
/** integer position increment in the output stream */
long opos_inc;
/** current sample(s) in the input stream (left/right channel) */
st_sample_t icur[2];
/** last sample(s) in the input stream (left/right channel) */
/** Note, these are deliberately ints, not st_sample_t's */
int32 ilast[2];
st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
} LinearRateDetails;
extern "C" {
#ifndef IPHONE
#define ARM_LinearRate_M _ARM_LinearRate_M
#define ARM_LinearRate_S _ARM_LinearRate_S
#define ARM_LinearRate_R _ARM_LinearRate_R
#endif
}
extern "C" void ARM_LinearRate_M(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
LinearRateDetails *lr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
extern "C" void ARM_LinearRate_S(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
LinearRateDetails *lr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
extern "C" void ARM_LinearRate_R(AudioStream &input,
int (*fn)(Audio::AudioStream&,int16*,int),
LinearRateDetails *lr,
st_sample_t *obuf,
st_size_t osamp,
st_volume_t vol_l,
st_volume_t vol_r);
template<bool stereo, bool reverseStereo>
class LinearRateConverter : public RateConverter {
protected:
LinearRateDetails lr;
public:
LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
/*
* Prepare processing.
*/
template<bool stereo, bool reverseStereo>
LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
unsigned long incr;
if (inrate == outrate) {
error("Input and Output rates must be different to use rate effect");
}
if (inrate >= 65536 || outrate >= 65536) {
error("rate effect can only handle rates < 65536");
}
lr.opos = FRAC_ONE;
/* increment */
incr = (inrate << FRAC_BITS) / outrate;
lr.opos_inc = incr;
lr.ilast[0] = lr.ilast[1] = 32768;
lr.icur[0] = lr.icur[1] = 0;
lr.inLen = 0;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
template<bool stereo, bool reverseStereo>
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) {
#ifdef DEBUG_RATECONV
fprintf(stderr, "Linear st=%d rev=%d\n", stereo, reverseStereo);
fflush(stderr);
#endif
if (!stereo) {
ARM_LinearRate_M(input,
&SimpleRate_readFudge,
&lr,
obuf, osamp, vol_l, vol_r);
} else if (reverseStereo) {
ARM_LinearRate_R(input,
&SimpleRate_readFudge,
&lr,
obuf, osamp, vol_l, vol_r);
} else {
ARM_LinearRate_S(input,
&SimpleRate_readFudge,
&lr,
obuf, osamp, vol_l, vol_r);
}
return (ST_SUCCESS);
}
#pragma mark -
/**
* Simple audio rate converter for the case that the inrate equals the outrate.
*/
extern "C" {
#ifndef IPHONE
#define ARM_CopyRate_M _ARM_CopyRate_M
#define ARM_CopyRate_S _ARM_CopyRate_S
#define ARM_CopyRate_R _ARM_CopyRate_R
#endif
}
extern "C" void ARM_CopyRate_M(st_size_t len,
st_sample_t *obuf,
st_volume_t vol_l,
st_volume_t vol_r,
st_sample_t *_buffer);
extern "C" void ARM_CopyRate_S(st_size_t len,
st_sample_t *obuf,
st_volume_t vol_l,
st_volume_t vol_r,
st_sample_t *_buffer);
extern "C" void ARM_CopyRate_R(st_size_t len,
st_sample_t *obuf,
st_volume_t vol_l,
st_volume_t vol_r,
st_sample_t *_buffer);
template<bool stereo, bool reverseStereo>
class CopyRateConverter : public RateConverter {
st_sample_t *_buffer;
st_size_t _bufferSize;
public:
CopyRateConverter() : _buffer(0), _bufferSize(0) {}
~CopyRateConverter() {
free(_buffer);
}
virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
assert(input.isStereo() == stereo);
#ifdef DEBUG_RATECONV
fprintf(stderr, "Copy st=%d rev=%d\n", stereo, reverseStereo);
fflush(stderr);
#endif
st_size_t len;
if (stereo)
osamp *= 2;
// Reallocate temp buffer, if necessary
if (osamp > _bufferSize) {
free(_buffer);
_buffer = (st_sample_t *)malloc(osamp * 2);
_bufferSize = osamp;
}
// Read up to 'osamp' samples into our temporary buffer
len = input.readBuffer(_buffer, osamp);
if (len <= 0)
return (ST_SUCCESS);
// Mix the data into the output buffer
if (stereo && reverseStereo)
ARM_CopyRate_R(len, obuf, vol_l, vol_r, _buffer);
else if (stereo)
ARM_CopyRate_S(len, obuf, vol_l, vol_r, _buffer);
else
ARM_CopyRate_M(len, obuf, vol_l, vol_r, _buffer);
return (ST_SUCCESS);
}
virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
#pragma mark -
/**
* Create and return a RateConverter object for the specified input and output rates.
*/
RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
if (inrate != outrate) {
if ((inrate % outrate) == 0) {
if (stereo) {
if (reverseStereo)
return new SimpleRateConverter<true, true>(inrate, outrate);
else
return new SimpleRateConverter<true, false>(inrate, outrate);
} else
return new SimpleRateConverter<false, false>(inrate, outrate);
} else {
if (stereo) {
if (reverseStereo)
return new LinearRateConverter<true, true>(inrate, outrate);
else
return new LinearRateConverter<true, false>(inrate, outrate);
} else
return new LinearRateConverter<false, false>(inrate, outrate);
}
} else {
if (stereo) {
if (reverseStereo)
return new CopyRateConverter<true, true>();
else
return new CopyRateConverter<true, false>();
} else
return new CopyRateConverter<false, false>();
}
}
} // End of namespace Audio
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