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Diffstat (limited to 'sound/rate_arm.cpp')
| -rw-r--r-- | sound/rate_arm.cpp | 428 |
1 files changed, 428 insertions, 0 deletions
diff --git a/sound/rate_arm.cpp b/sound/rate_arm.cpp new file mode 100644 index 0000000000..c4758b2d15 --- /dev/null +++ b/sound/rate_arm.cpp @@ -0,0 +1,428 @@ +/* ScummVM - Scumm Interpreter + * Copyright (C) 2001-2006 The ScummVM project + * + * 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 "common/stdafx.h" +#include "sound/audiostream.h" +#include "sound/rate.h" +#include "sound/mixer.h" +#include "common/util.h" + +namespace Audio { + +/** + * The precision of the fractional computations used by the rate converter. + * Normally you should never have to modify this value. + */ +#define FRAC_BITS 16 + +/** + * 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" 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) +{ + 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 */ + 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) */ + st_sample_t ilast[2]; + + /** fractional position in the output stream */ + long opos_frac; + + /** fractional position increment in the output stream */ + long opos_inc_frac; + + st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE]; +} LinearRateDetails; + +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 = 0; + lr.opos = 1; + + /* increment */ + incr = (inrate << FRAC_BITS) / outrate; + + lr.opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1); + lr.opos_inc = incr >> FRAC_BITS; + + lr.ilast[0] = lr.ilast[1] = 0; + 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" 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_sample_t *ptr; + 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 |