mirror of
https://github.com/libretro/scummvm.git
synced 2024-12-02 23:26:44 +00:00
Patch #1721826: ARM asm versions of sound rate conversion/mixing code
svn-id: r27467
This commit is contained in:
Max Horn
parent
c7c2cb498c
commit
6498d669d0
@ -25,6 +25,8 @@ DISABLE_CRUISE = 1
|
||||
|
||||
#DISABLE_HQ_SCALERS = 1
|
||||
|
||||
USE_ARM_SOUND_ASM = 1
|
||||
|
||||
CXX = arm-wince-pe-g++
|
||||
LD = arm-wince-pe-g++
|
||||
AR = arm-wince-pe-ar cru
|
||||
|
||||
@ -16,7 +16,6 @@ MODULE_OBJS := \
|
||||
mp3.o \
|
||||
mpu401.o \
|
||||
null.o \
|
||||
rate.o \
|
||||
voc.o \
|
||||
vorbis.o \
|
||||
wave.o \
|
||||
@ -31,5 +30,14 @@ MODULE_OBJS := \
|
||||
softsynth/fluidsynth.o \
|
||||
softsynth/mt32.o \
|
||||
|
||||
ifndef USE_ARM_SOUND_ASM
|
||||
MODULE_OBJS += \
|
||||
rate.o
|
||||
else
|
||||
MODULE_OBJS += \
|
||||
rate_arm.o \
|
||||
rate_arm_asm.o
|
||||
endif
|
||||
|
||||
# Include common rules
|
||||
include $(srcdir)/rules.mk
|
||||
|
||||
428
sound/rate_arm.cpp
Normal file
428
sound/rate_arm.cpp
Normal file
@ -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
|
||||
689
sound/rate_arm_asm.s
Normal file
689
sound/rate_arm_asm.s
Normal file
@ -0,0 +1,689 @@
|
||||
@ ScummVM Scumm Interpreter
|
||||
@ Copyright (C) 2007 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: $
|
||||
@
|
||||
@ @author Robin Watts (robin@wss.co.uk)
|
||||
@
|
||||
@ This file, together with rate_arm.cpp, provides an ARM optimised version
|
||||
@ of rate.cpp. The algorithm is essentially the same as that within rate.cpp
|
||||
@ so to understand this file you should understand rate.cpp first.
|
||||
|
||||
.text
|
||||
|
||||
.global ARM_CopyRate_M
|
||||
.global ARM_CopyRate_S
|
||||
.global ARM_CopyRate_R
|
||||
.global ARM_SimpleRate_M
|
||||
.global ARM_SimpleRate_S
|
||||
.global ARM_SimpleRate_R
|
||||
.global ARM_LinearRate_M
|
||||
.global ARM_LinearRate_S
|
||||
.global ARM_LinearRate_R
|
||||
|
||||
ARM_CopyRate_M:
|
||||
@ r0 = len
|
||||
@ r1 = obuf
|
||||
@ r2 = vol_l
|
||||
@ r3 = vol_r
|
||||
@ <> = ptr
|
||||
LDR r12,[r13]
|
||||
STMFD r13!,{r4-r7,r14}
|
||||
|
||||
MOV r14,#0 @ r14= 0
|
||||
ORR r2, r2, r2, LSL #8 @ r2 = vol_l as 16 bits
|
||||
ORR r3, r3, r3, LSL #8 @ r3 = vol_r as 16 bits
|
||||
CopyRate_M_loop:
|
||||
LDRSH r5, [r12], #2 @ r5 = tmp0 = tmp1 = *ptr++
|
||||
LDRSH r6, [r1] @ r6 = obuf[0]
|
||||
LDRSH r7, [r1, #2] @ r7 = obuf[1]
|
||||
MUL r4, r2, r5 @ r4 = tmp0*vol_l
|
||||
MUL r5, r3, r5 @ r5 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r14,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r14,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r1], #2 @ Store output value
|
||||
STRH r7, [r1], #2 @ Store output value
|
||||
|
||||
SUBS r0,r0,#1 @ len--
|
||||
BGT CopyRate_M_loop @ and loop
|
||||
|
||||
LDMFD r13!,{r4-r7,PC}
|
||||
|
||||
ARM_CopyRate_S:
|
||||
@ r0 = len
|
||||
@ r1 = obuf
|
||||
@ r2 = vol_l
|
||||
@ r3 = vol_r
|
||||
@ <> = ptr
|
||||
LDR r12,[r13]
|
||||
STMFD r13!,{r4-r7,r14}
|
||||
|
||||
MOV r14,#0 @ r14= 0
|
||||
ORR r2, r2, r2, LSL #8 @ r2 = vol_l as 16 bits
|
||||
ORR r3, r3, r3, LSL #8 @ r3 = vol_r as 16 bits
|
||||
CopyRate_S_loop:
|
||||
LDRSH r4, [r12],#2 @ r4 = tmp0 = *ptr++
|
||||
LDRSH r5, [r12],#2 @ r5 = tmp1 = *ptr++
|
||||
LDRSH r6, [r1] @ r6 = obuf[0]
|
||||
LDRSH r7, [r1,#2] @ r7 = obuf[1]
|
||||
MUL r4, r2, r4 @ r5 = tmp0*vol_l
|
||||
MUL r5, r3, r5 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r14,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r14,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r1],#2 @ Store output value
|
||||
STRH r7, [r1],#2 @ Store output value
|
||||
|
||||
SUBS r0,r0,#2 @ len -= 2
|
||||
BGT CopyRate_S_loop @ and loop
|
||||
|
||||
LDMFD r13!,{r4-r7,PC}
|
||||
|
||||
ARM_CopyRate_R:
|
||||
@ r0 = len
|
||||
@ r1 = obuf
|
||||
@ r2 = vol_l
|
||||
@ r3 = vol_r
|
||||
@ <> = ptr
|
||||
LDR r12,[r13]
|
||||
STMFD r13!,{r4-r7,r14}
|
||||
|
||||
MOV r14,#0 @ r14= 0
|
||||
ORR r2, r2, r2, LSL #8 @ r2 = vol_l as 16 bits
|
||||
ORR r3, r3, r3, LSL #8 @ r3 = vol_r as 16 bits
|
||||
CopyRate_R_loop:
|
||||
LDRSH r5, [r12],#2 @ r5 = tmp1 = *ptr++
|
||||
LDRSH r4, [r12],#2 @ r4 = tmp0 = *ptr++
|
||||
LDRSH r6, [r1] @ r6 = obuf[0]
|
||||
LDRSH r7, [r1,#2] @ r7 = obuf[1]
|
||||
MUL r4, r2, r4 @ r4 = tmp0*vol_l
|
||||
MUL r5, r3, r5 @ r5 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r14,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r14,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r1],#2 @ Store output value
|
||||
STRH r7, [r1],#2 @ Store output value
|
||||
|
||||
SUBS r0,r0,#2 @ len -= 2
|
||||
BGT CopyRate_R_loop @ and loop
|
||||
|
||||
LDMFD r13!,{r4-r7,PC}
|
||||
|
||||
ARM_SimpleRate_M:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r2,r4-r8,r10-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r2,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r2 = opos
|
||||
@ r8 = opos_inc
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE SimpleRate_M_end @ bale
|
||||
MOV r10,#0
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
SimpleRate_M_loop:
|
||||
SUBS r1, r1, #1 @ r1 = inLen -= 1
|
||||
BLT SimpleRate_M_read
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #2 @ if (r2 >= 0) { sr.inPtr++
|
||||
BGE SimpleRate_M_loop @ and loop }
|
||||
SimpleRate_M_read_return:
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp1 = *inPtr++
|
||||
LDRSH r6, [r3] @ r6 = obuf[0]
|
||||
LDRSH r7, [r3,#2] @ r7 = obuf[1]
|
||||
ADD r2, r2, r8 @ r2 = opos += opos_inc
|
||||
MUL r4, r12,r5 @ r4 = tmp0*vol_l
|
||||
MUL r5, r14,r5 @ r5 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r10,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r10,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
|
||||
SUBS r11,r11,#1 @ len--
|
||||
BGT SimpleRate_M_loop @ and loop
|
||||
SimpleRate_M_end:
|
||||
LDR r14,[r13,#8] @ r14 = sr
|
||||
ADD r13,r13,#12 @ Skip over r0-r2 on stack
|
||||
STMIA r14,{r0,r1,r2} @ Store back updated values
|
||||
LDMFD r13!,{r4-r8,r10-r11,PC}
|
||||
SimpleRate_M_read:
|
||||
LDR r0, [r13,#4*2] @ r0 = sr
|
||||
ADD r0, r0, #16 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 3+8+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #1 @ r1 = inLen-1
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT SimpleRate_M_end
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #2 @ if (r2 >= 0) { sr.inPtr++
|
||||
BGE SimpleRate_M_loop @ and loop }
|
||||
B SimpleRate_M_read_return
|
||||
|
||||
|
||||
ARM_SimpleRate_S:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r2,r4-r8,r10-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r2,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r2 = opos
|
||||
@ r8 = opos_inc
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE SimpleRate_S_end @ bale
|
||||
MOV r10,#0
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
SimpleRate_S_loop:
|
||||
SUBS r1, r1, #2 @ r1 = inLen -= 2
|
||||
BLT SimpleRate_S_read
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #4 @ if (r2 >= 0) { sr.inPtr += 2
|
||||
BGE SimpleRate_S_loop @ and loop }
|
||||
SimpleRate_S_read_return:
|
||||
LDRSH r4, [r0],#2 @ r4 = tmp0 = *inPtr++
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp1 = *inPtr++
|
||||
LDRSH r6, [r3] @ r6 = obuf[0]
|
||||
LDRSH r7, [r3,#2] @ r7 = obuf[1]
|
||||
ADD r2, r2, r8 @ r2 = opos += opos_inc
|
||||
MUL r4, r12,r4 @ r5 = tmp0*vol_l
|
||||
MUL r5, r14,r5 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r10,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r10,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
|
||||
SUBS r11,r11,#1 @ osamp--
|
||||
BGT SimpleRate_S_loop @ and loop
|
||||
SimpleRate_S_end:
|
||||
LDR r14,[r13,#8] @ r14 = sr
|
||||
ADD r13,r13,#12 @ skip over r0-r2 on stack
|
||||
STMIA r14,{r0,r1,r2} @ store back updated values
|
||||
LDMFD r13!,{r4-r8,r10-r11,PC}
|
||||
SimpleRate_S_read:
|
||||
LDR r0, [r13,#4*2] @ r0 = sr
|
||||
ADD r0, r0, #16 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 3+8+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #2 @ r1 = inLen-2
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT SimpleRate_S_end
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #4 @ if (r2 >= 0) { sr.inPtr += 2
|
||||
BGE SimpleRate_S_loop @ and loop }
|
||||
B SimpleRate_S_read_return
|
||||
|
||||
|
||||
|
||||
ARM_SimpleRate_R:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r2,r4-r8,r10-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r2,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r2 = opos
|
||||
@ r8 = opos_inc
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE SimpleRate_R_end @ bale
|
||||
MOV r10,#0
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
SimpleRate_R_loop:
|
||||
SUBS r1, r1, #2 @ r1 = inLen -= 2
|
||||
BLT SimpleRate_R_read
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #4 @ if (r2 >= 0) { sr.inPtr += 2
|
||||
BGE SimpleRate_R_loop @ and loop }
|
||||
SimpleRate_R_read_return:
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp0 = *inPtr++
|
||||
LDRSH r4, [r0],#2 @ r4 = tmp1 = *inPtr++
|
||||
LDRSH r6, [r3] @ r6 = obuf[0]
|
||||
LDRSH r7, [r3,#2] @ r7 = obuf[1]
|
||||
ADD r2, r2, r8 @ r2 = opos += opos_inc
|
||||
MUL r4, r12,r4 @ r5 = tmp0*vol_l
|
||||
MUL r5, r14,r5 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r6, r4, r6, LSL #16 @ r6 = obuf[0]<<16 + tmp0*vol_l
|
||||
RSCVS r6, r10,#1<<31 @ Clamp r6
|
||||
ADDS r7, r5, r7, LSL #16 @ r7 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r7, r10,#1<<31 @ Clamp r7
|
||||
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
|
||||
SUBS r11,r11,#1 @ osamp--
|
||||
BGT SimpleRate_R_loop @ and loop
|
||||
SimpleRate_R_end:
|
||||
LDR r14,[r13,#8] @ r14 = sr
|
||||
ADD r13,r13,#12 @ Skip over r0-r2 on stack
|
||||
STMIA r14,{r0,r1,r2} @ Store back updated values
|
||||
LDMFD r13!,{r4-r8,r10-r11,PC}
|
||||
SimpleRate_R_read:
|
||||
LDR r0, [r13,#4*2] @ r0 = sr
|
||||
ADD r0, r0, #16 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 3+8+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #2 @ r1 = inLen-2
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT SimpleRate_R_end
|
||||
SUBS r2, r2, #1 @ r2 = opos--
|
||||
ADDGE r0, r0, #4 @ if (r2 >= 0) { sr.inPtr += 2
|
||||
BGE SimpleRate_R_loop @ and loop }
|
||||
B SimpleRate_R_read_return
|
||||
|
||||
|
||||
ARM_LinearRate_M:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r1,r4-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r8 = opos
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE LinearRate_M_end @ bale
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
CMP r1,#0
|
||||
BGT LinearRate_M_part2
|
||||
|
||||
@ part1 - read input samples
|
||||
LinearRate_M_loop:
|
||||
SUBS r1, r1, #1 @ r1 = inLen -= 1
|
||||
BLT LinearRate_M_read
|
||||
LinearRate_M_read_return:
|
||||
LDR r10,[r2, #16] @ r10= icur[0,1]
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp1 = *inPtr++
|
||||
SUBS r8, r8, #1 @ r8 = opos--
|
||||
STR r10,[r2,#20] @ ilast[0,1] = icur[0,1]
|
||||
STRH r5, [r2,#16] @ icur[0] = tmp1
|
||||
BGE LinearRate_M_loop
|
||||
|
||||
@ part2 - form output samples
|
||||
LinearRate_M_part2:
|
||||
@ We are guaranteed that opos < 0 here
|
||||
LDRSH r6, [r2,#20] @ r6 = ilast[0]
|
||||
LDRSH r5, [r2,#16] @ r5 = icur[0]
|
||||
LDRH r4, [r2,#24] @ r4 = opos_frac
|
||||
LDR r10,[r2,#28] @ r10= opos_frac_inc
|
||||
MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16
|
||||
SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0]
|
||||
ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1)
|
||||
MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
|
||||
|
||||
ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac
|
||||
STRH r4,[r2,#24] @ opos_frac &= 65535
|
||||
ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS)
|
||||
|
||||
LDRSH r4, [r3] @ r4 = obuf[0]
|
||||
LDRSH r5, [r3,#2] @ r5 = obuf[1]
|
||||
MOV r6, r6, ASR #16 @ r6 = tmp0 = tmp1 >>= 16
|
||||
MUL r7, r12,r6 @ r7 = tmp0*vol_l
|
||||
MUL r6, r14,r6 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l
|
||||
MOV r4, #0
|
||||
RSCVS r7, r4, #1<<31 @ Clamp r7
|
||||
ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r6, r4, #1<<31 @ Clamp r6
|
||||
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
|
||||
LDR r5, [r2,#12] @ r5 = opos_inc
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
SUBS r11, r11,#1 @ opos--
|
||||
BLE LinearRate_M_end @ end if needed
|
||||
|
||||
ADDS r8, r8, r5 @ r8 = opos += opos_inc
|
||||
BLT LinearRate_M_part2
|
||||
B LinearRate_M_loop
|
||||
LinearRate_M_end:
|
||||
ADD r13,r13,#8
|
||||
STMIA r2,{r0,r1,r8}
|
||||
LDMFD r13!,{r4-r11,PC}
|
||||
LinearRate_M_read:
|
||||
ADD r0, r2, #32 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 2+9+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #1 @ r1 = inLen-1
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT LinearRate_M_end
|
||||
B LinearRate_M_read_return
|
||||
|
||||
ARM_LinearRate_S:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r1,r4-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r8 = opos
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE LinearRate_S_end @ bale
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
CMP r1,#0
|
||||
BGT LinearRate_S_part2
|
||||
|
||||
@ part1 - read input samples
|
||||
LinearRate_S_loop:
|
||||
SUBS r1, r1, #2 @ r1 = inLen -= 2
|
||||
BLT LinearRate_S_read
|
||||
LinearRate_S_read_return:
|
||||
LDR r10,[r2, #16] @ r10= icur[0,1]
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp0 = *inPtr++
|
||||
LDRSH r6, [r0],#2 @ r5 = tmp1 = *inPtr++
|
||||
SUBS r8, r8, #1 @ r8 = opos--
|
||||
STR r10,[r2,#20] @ ilast[0,1] = icur[0,1]
|
||||
STRH r5, [r2,#16] @ icur[0] = tmp0
|
||||
STRH r6, [r2,#16] @ icur[1] = tmp1
|
||||
BGE LinearRate_S_loop
|
||||
|
||||
@ part2 - form output samples
|
||||
LinearRate_S_part2:
|
||||
@ We are guaranteed that opos < 0 here
|
||||
LDRSH r6, [r2,#20] @ r6 = ilast[0]
|
||||
LDRSH r5, [r2,#16] @ r5 = icur[0]
|
||||
LDRH r4, [r2,#24] @ r4 = opos_frac
|
||||
MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16
|
||||
SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0]
|
||||
ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1)
|
||||
MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
|
||||
|
||||
LDRSH r7, [r2,#22] @ r6 = ilast[1]
|
||||
LDRSH r5, [r2,#18] @ r5 = icur[1]
|
||||
LDR r10,[r2,#28] @ r10= opos_frac_inc
|
||||
MOV r7, r7, LSL #16 @ r7 = ilast[1]<<16
|
||||
SUB r5, r5, r7, ASR #16 @ r5 = icur[1] - ilast[1]
|
||||
ADD r7, r7, #1<<15 @ r6 = ilast[1]+1<<(FRAC_BITS-1)
|
||||
MLA r7, r4, r5, r7 @ r6 = (icur[1]-ilast[1])*opos_frac+ilast[1]
|
||||
|
||||
ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac
|
||||
STRH r4,[r2,#24] @ opos_frac &= 65535
|
||||
ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS)
|
||||
|
||||
LDRSH r4, [r3] @ r4 = obuf[0]
|
||||
LDRSH r5, [r3,#2] @ r5 = obuf[1]
|
||||
MOV r7, r7, ASR #16 @ r7 = tmp0 >>= 16
|
||||
MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16
|
||||
MUL r7, r12,r7 @ r7 = tmp0*vol_l
|
||||
MUL r6, r14,r6 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l
|
||||
MOV r4, #0
|
||||
RSCVS r7, r4, #1<<31 @ Clamp r7
|
||||
ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r6, r4, #1<<31 @ Clamp r6
|
||||
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
|
||||
LDR r5, [r2,#12] @ r5 = opos_inc
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
SUBS r11, r11,#1 @ opos--
|
||||
BLE LinearRate_S_end @ and loop
|
||||
|
||||
ADDS r8, r8, r5 @ r8 = opos += opos_inc
|
||||
BLT LinearRate_S_part2
|
||||
B LinearRate_S_loop
|
||||
LinearRate_S_end:
|
||||
ADD r13,r13,#8
|
||||
STMIA r2,{r0,r1,r8}
|
||||
LDMFD r13!,{r4-r11,PC}
|
||||
LinearRate_S_read:
|
||||
ADD r0, r2, #32 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 2+9+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #2 @ r1 = inLen-2
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT LinearRate_S_end
|
||||
B LinearRate_S_read_return
|
||||
|
||||
ARM_LinearRate_R:
|
||||
@ r0 = AudioStream &input
|
||||
@ r1 = input.readBuffer
|
||||
@ r2 = input->sr
|
||||
@ r3 = obuf
|
||||
@ <> = osamp
|
||||
@ <> = vol_l
|
||||
@ <> = vol_r
|
||||
MOV r12,r13
|
||||
STMFD r13!,{r0-r1,r4-r11,r14}
|
||||
LDMFD r12,{r11,r12,r14} @ r11= osamp
|
||||
@ r12= vol_l
|
||||
@ r14= vol_r
|
||||
LDMIA r2,{r0,r1,r8} @ r0 = inPtr
|
||||
@ r1 = inLen
|
||||
@ r8 = opos
|
||||
CMP r11,#0 @ if (osamp <= 0)
|
||||
BLE LinearRate_R_end @ bale
|
||||
ORR r12,r12,r12,LSL #8 @ r12= vol_l as 16 bits
|
||||
ORR r14,r14,r14,LSL #8 @ r14= vol_r as 16 bits
|
||||
CMP r1,#0
|
||||
BGT LinearRate_R_part2
|
||||
|
||||
@ part1 - read input samples
|
||||
LinearRate_R_loop:
|
||||
SUBS r1, r1, #2 @ r1 = inLen -= 2
|
||||
BLT LinearRate_R_read
|
||||
LinearRate_R_read_return:
|
||||
LDR r10,[r2, #16] @ r10= icur[0,1]
|
||||
LDRSH r5, [r0],#2 @ r5 = tmp0 = *inPtr++
|
||||
LDRSH r6, [r0],#2 @ r5 = tmp1 = *inPtr++
|
||||
SUBS r8, r8, #1 @ r8 = opos--
|
||||
STR r10,[r2,#20] @ ilast[0,1] = icur[0,1]
|
||||
STRH r5, [r2,#16] @ icur[0] = tmp0
|
||||
STRH r6, [r2,#16] @ icur[1] = tmp1
|
||||
BGE LinearRate_R_loop
|
||||
|
||||
@ part2 - form output samples
|
||||
LinearRate_R_part2:
|
||||
@ We are guaranteed that opos < 0 here
|
||||
LDRSH r6, [r2,#20] @ r6 = ilast[0]
|
||||
LDRSH r5, [r2,#16] @ r5 = icur[0]
|
||||
LDRH r4, [r2,#24] @ r4 = opos_frac
|
||||
MOV r6, r6, LSL #16 @ r6 = ilast[0]<<16
|
||||
SUB r5, r5, r6, ASR #16 @ r5 = icur[0] - ilast[0]
|
||||
ADD r6, r6, #1<<15 @ r6 = ilast[0]+1<<(FRAC_BITS-1)
|
||||
MLA r6, r4, r5, r6 @ r6 = (icur[0]-ilast[0])*opos_frac+ilast[0]
|
||||
|
||||
LDRSH r7, [r2,#22] @ r6 = ilast[1]
|
||||
LDRSH r5, [r2,#18] @ r5 = icur[1]
|
||||
LDR r10,[r2,#28] @ r10= opos_frac_inc
|
||||
MOV r7, r7, LSL #16 @ r7 = ilast[1]<<16
|
||||
SUB r5, r5, r7, ASR #16 @ r5 = icur[1] - ilast[1]
|
||||
ADD r7, r7, #1<<15 @ r6 = ilast[1]+1<<(FRAC_BITS-1)
|
||||
MLA r7, r4, r5, r7 @ r6 = (icur[1]-ilast[1])*opos_frac+ilast[1]
|
||||
|
||||
ADD r4, r4, r10 @ r4 = tmp = opos_frac+opos_inc_frac
|
||||
STRH r4,[r2,#24] @ opos_frac &= 65535
|
||||
ADD r8, r8, r4, LSR #16 @ opos += (tmp>>FRAC_BITS)
|
||||
|
||||
LDRSH r4, [r3] @ r4 = obuf[0]
|
||||
LDRSH r5, [r3,#2] @ r5 = obuf[1]
|
||||
MOV r7, r7, ASR #16 @ r7 = tmp0 >>= 16
|
||||
MOV r6, r6, ASR #16 @ r6 = tmp1 >>= 16
|
||||
MUL r7, r12,r7 @ r7 = tmp0*vol_l
|
||||
MUL r6, r14,r6 @ r6 = tmp1*vol_r
|
||||
|
||||
ADDS r7, r7, r4, LSL #16 @ r7 = obuf[0]<<16 + tmp0*vol_l
|
||||
MOV r4, #0
|
||||
RSCVS r7, r4, #1<<31 @ Clamp r7
|
||||
ADDS r6, r6, r5, LSL #16 @ r6 = obuf[1]<<16 + tmp1*vol_r
|
||||
RSCVS r6, r4, #1<<31 @ Clamp r6
|
||||
|
||||
MOV r7, r7, LSR #16 @ Shift back to halfword
|
||||
MOV r6, r6, LSR #16 @ Shift back to halfword
|
||||
|
||||
LDR r5, [r2,#12] @ r5 = opos_inc
|
||||
STRH r6, [r3],#2 @ Store output value
|
||||
STRH r7, [r3],#2 @ Store output value
|
||||
SUBS r11, r11,#1 @ opos--
|
||||
BLE LinearRate_R_end @ and loop
|
||||
|
||||
ADDS r8, r8, r5 @ r8 = opos += opos_inc
|
||||
BLT LinearRate_R_part2
|
||||
B LinearRate_R_loop
|
||||
LinearRate_R_end:
|
||||
ADD r13,r13,#8
|
||||
STMIA r2,{r0,r1,r8}
|
||||
LDMFD r13!,{r4-r11,PC}
|
||||
LinearRate_R_read:
|
||||
ADD r0, r2, #32 @ r0 = inPtr = inBuf
|
||||
STMFD r13!,{r0,r2-r3,r12,r14}
|
||||
|
||||
MOV r1, r0 @ r1 = inBuf
|
||||
LDR r0, [r13,#4*5] @ r0 = AudioStream & input
|
||||
MOV r2, #512 @ r2 = ARRAYSIZE(inBuf)
|
||||
|
||||
@ Calling back into C++ here. WinCE is fairly easy about such things
|
||||
@ but other OS are more awkward. r9 is preserved for Symbian, and
|
||||
@ we have 2+9+5 = 16 things on the stack (an even number).
|
||||
MOV r14,PC
|
||||
LDR PC,[r13,#4*6] @ inLen = input.readBuffer(inBuf,512)
|
||||
SUBS r1, r0, #2 @ r1 = inLen-2
|
||||
LDMFD r13!,{r0,r2-r3,r12,r14}
|
||||
BLT LinearRate_R_end
|
||||
B LinearRate_R_read_return
|
||||
Loading…
Reference in New Issue
Block a user