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-rw-r--r--sound/module.mk10
-rw-r--r--sound/rate_arm.cpp428
-rw-r--r--sound/rate_arm_asm.s689
3 files changed, 1126 insertions, 1 deletions
diff --git a/sound/module.mk b/sound/module.mk
index a204dbadb9..bcb822b86d 100644
--- a/sound/module.mk
+++ b/sound/module.mk
@@ -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
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
diff --git a/sound/rate_arm_asm.s b/sound/rate_arm_asm.s
new file mode 100644
index 0000000000..fa9c56976d
--- /dev/null
+++ b/sound/rate_arm_asm.s
@@ -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
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-26 05:09:33 +0000