rockbox/apps/dsp_arm.S

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006-2007 Thom Johansen
*
* 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
/****************************************************************************
* void channels_process_sound_chan_mono(int count, int32_t *buf[])
*
* NOTE: The following code processes two samples at once. When count is odd,
* there is an additional obsolete sample processed, which will not be
* used by the calling functions.
*/
.section .icode, "ax", %progbits
.align 2
.global channels_process_sound_chan_mono
.type channels_process_sound_chan_mono, %function
channels_process_sound_chan_mono:
@ input: r0 = count, r1 = buf
stmfd sp!, {r4-r5, lr}
ldmia r1, {r2-r3} @ r4 = buf[0], r5 = buf[1]
.monoloop:
ldmia r2, {r4-r5}
ldmia r3, {r12,lr}
mov r4, r4, asr #1 @ r4 = r4/2
add r4, r4, r12, asr #1 @ r4 = r4 + r12/2 = (buf[0]+buf[1])/2
mov r5, r5, asr #1 @ r5 = r5/2
add r5, r5, lr, asr #1 @ r5 = r5 + lr/2 = (buf[0]+buf[1])/2
stmia r2!, {r4-r5}
stmia r3!, {r4-r5}
subs r0, r0, #2
bgt .monoloop
ldmfd sp!, {r4-r5, pc}
.monoend:
.size channels_process_sound_chan_mono,.monoend-channels_process_sound_chan_mono
/****************************************************************************
* void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
* NOTE: The following code processes two samples at once. When count is odd,
* there is an additional obsolete sample processed, which will not be
* used by the calling functions.
*/
.section .icode, "ax", %progbits
.align 2
.global channels_process_sound_chan_karaoke
.type channels_process_sound_chan_karaoke, %function
channels_process_sound_chan_karaoke:
@ input: r0 = count, r1 = buf
stmfd sp!, {r4-r5, lr}
ldmia r1, {r2-r3} @ r4 = buf[0], r5 = buf[1]
.karaokeloop:
ldmia r2, {r4-r5}
ldmia r3, {r12,lr}
mov r12, r12, asr #1 @ r12 = r12/2
rsb r4, r12, r4, asr #1 @ r4 = -r12 + r4/2 = (buf[0]-buf[1])/2
rsb r12, r4, #0 @ r12 = -r4
mov lr, lr, asr #1 @ lr = lr/2
rsb r5, lr, r5, asr #1 @ r5 = -lr + r5/2 = (buf[0]-buf[1])/2
rsb lr, r5, #0 @ lr = -r5
stmia r2!, {r4-r5}
stmia r3!, {r12,lr}
subs r0, r0, #2
bgt .karaokeloop
ldmfd sp!, {r4-r5, pc}
.karaokeend:
.size channels_process_sound_chan_karaoke,.karaokeend-channels_process_sound_chan_karaoke
/****************************************************************************
* void sample_output_mono(int count, struct dsp_data *data,
* const int32_t *src[], int16_t *dst)
* NOTE: The following code processes two samples at once. When count is odd,
* there is an additional obsolete sample processed, which will not be
* used by the calling functions.
*/
.section .icode, "ax", %progbits
.align 2
.global sample_output_mono
.type sample_output_mono, %function
sample_output_mono:
@ input: r0 = count, r1 = data, r2 = src, r3 = dst
stmfd sp!, {r4-r7, lr}
ldr r4, [r2] @ r4 = src[0]
ldr r5, [r1] @ lr = data->output_scale
sub r1, r5, #1 @ r1 = r5-1
mov r2, #1
mov r2, r2, asl r1 @ r2 = 1<<r1 = 1 << (scale-1)
mvn r1, #0x8000 @ r1 needed for clipping
mov r12, #0xff00
orr r12, r12, #0xff @ r12 needed for masking
.somloop:
ldmia r4!, {r6-r7}
add r6, r6, r2
mov r6, r6, asr r5 @ r6 = (r6 + 1<<(scale-1)) >> scale
mov lr, r6, asr #15
teq lr, lr, asr #31
eorne r6, r1, lr, asr #31 @ Clip (-32768...+32767)
add r7, r7, r2
mov r7, r7, asr r5 @ r7 = (r7 + 1<<(scale-1)) >> scale
mov lr, r7, asr #15
teq lr, lr, asr #31
eorne r7, r1, lr, asr #31 @ Clip (-32768...+32767)
and r6, r6, r12
orr r6, r6, r6, asl #16 @ pack first 2 halfwords into 1 word
and r7, r7, r12
orr r7, r7, r7, asl #16 @ pack last 2 halfwords into 1 word
stmia r3!, {r6-r7}
subs r0, r0, #2
bgt .somloop
ldmfd sp!, {r4-r7, pc}
.somend:
.size sample_output_mono,.somend-sample_output_mono
/****************************************************************************
* void sample_output_stereo(int count, struct dsp_data *data,
* const int32_t *src[], int16_t *dst)
* NOTE: The following code processes two samples at once. When count is odd,
* there is an additional obsolete sample processed, which will not be
* used by the calling functions.
*/
.section .icode, "ax", %progbits
.align 2
.global sample_output_stereo
.type sample_output_stereo, %function
sample_output_stereo:
@ input: r0 = count, r1 = data, r2 = src, r3 = dst
stmfd sp!, {r4-r10, lr}
ldmia r2, {r4-r5} @ r4 = src[0], r5 = src[1]
ldr r6, [r1] @ r6 = data->output_scale
sub r1, r6, #1 @ r1 = r6-1
mov r2, #1
mov r2, r2, asl r1 @ r2 = 1<<r1 = 1 << (scale-1)
mvn r1, #0x8000 @ r1 needed for clipping
mov r12, #0xff00
orr r12, r12, #0xff @ r12 needed for masking
.sosloop:
ldmia r4!, {r7-r8}
add r7, r7, r2
mov r7, r7, asr r6 @ r7 = (r7 + 1<<(scale-1)) >> scale
mov lr, r7, asr #15
teq lr, lr, asr #31
eorne r7, r1, lr, asr #31 @ Clip (-32768...+32767)
add r8, r8, r2
mov r8, r8, asr r6 @ r8 = (r8 + 1<<(scale-1)) >> scale
mov lr, r8, asr #15
teq lr, lr, asr #31
eorne r8, r1, lr, asr #31 @ Clip (-32768...+32767)
ldmia r5!, {r9-r10}
add r9, r9, r2
mov r9, r9, asr r6 @ r9 = (r9 + 1<<(scale-1)) >> scale
mov lr, r9, asr #15
teq lr, lr, asr #31
eorne r9, r1, lr, asr #31 @ Clip (-32768...+32767)
add r10, r10, r2
mov r10, r10, asr r6 @ r10 = (r10 + 1<<(scale-1)) >> scale
mov lr, r10, asr #15
teq lr, lr, asr #31
eorne r10, r1, lr, asr #31 @ Clip (-32768...+32767)
and r7, r7, r12
orr r9, r7, r9, asl #16 @ pack first 2 halfwords into 1 word
and r8, r8, r12
orr r10, r8, r10, asl #16 @ pack last 2 halfwords into 1 word
stmia r3!, {r9-r10}
subs r0, r0, #2
bgt .sosloop
ldmfd sp!, {r4-r10, pc}
.sosend:
.size sample_output_stereo,.sosend-sample_output_stereo
/****************************************************************************
* void apply_crossfeed(int count, int32_t* src[])
*/
.section .text
.global apply_crossfeed
apply_crossfeed:
@ unfortunately, we ended up in a bit of a register squeeze here, and need
@ to keep the count on the stack :/
stmdb sp!, { r4-r11, lr } @ stack modified regs
ldmia r1, { r2-r3 } @ r2 = src[0], r3 = src[1]
ldr r1, =crossfeed_data
ldmia r1!, { r4-r11 } @ load direct gain and filter data
add r12, r1, #13*4*2 @ calculate end of delay
stmdb sp!, { r0, r12 } @ stack count and end of delay adr
ldr r0, [r1, #13*4*2] @ fetch current delay line address
/* Register usage in loop:
* r0 = &delay[index][0], r1 = accumulator high, r2 = src[0], r3 = src[1],
* r4 = direct gain, r5-r7 = b0, b1, a1 (filter coefs),
* r8-r11 = filter history, r12 = temp, r14 = accumulator low
*/
.cfloop:
smull r14, r1, r6, r8 @ acc = b1*dr[n - 1]
smlal r14, r1, r7, r9 @ acc += a1*y_l[n - 1]
ldr r8, [r0, #4] @ r8 = dr[n]
smlal r14, r1, r5, r8 @ acc += b0*dr[n]
mov r9, r1, lsl #1 @ fix format for filter history
ldr r12, [r2] @ load left input
smlal r14, r1, r4, r12 @ acc += gain*x_l[n]
mov r1, r1, lsl #1 @ fix format
str r1, [r2], #4 @ save result
smull r14, r1, r6, r10 @ acc = b1*dl[n - 1]
smlal r14, r1, r7, r11 @ acc += a1*y_r[n - 1]
ldr r10, [r0] @ r10 = dl[n]
str r12, [r0], #4 @ save left input to delay line
smlal r14, r1, r5, r10 @ acc += b0*dl[n]
mov r11, r1, lsl #1 @ fix format for filter history
ldr r12, [r3] @ load right input
smlal r14, r1, r4, r12 @ acc += gain*x_r[n]
str r12, [r0], #4 @ save right input to delay line
mov r1, r1, lsl #1 @ fix format
str r1, [r3], #4 @ save result
ldr r12, [sp, #4] @ fetch delay line end addr from stack
cmp r0, r12 @ need to wrap to start of delay?
subeq r0, r0, #13*4*2 @ wrap back delay line ptr to start
ldr r1, [sp] @ fetch count from stack
subs r1, r1, #1 @ are we finished?
strne r1, [sp] @ nope, save count back to stack
bne .cfloop
@ save data back to struct
ldr r12, =crossfeed_data + 4*4
stmia r12, { r8-r11 } @ save filter history
str r0, [r12, #30*4] @ save delay line index
add sp, sp, #8 @ remove temp variables from stack
ldmia sp!, { r4-r11, pc }
.cfend:
.size apply_crossfeed,.cfend-apply_crossfeed
/****************************************************************************
* int dsp_downsample(int count, struct dsp_data *data,
* in32_t *src[], int32_t *dst[])
*/
.section .text
.global dsp_downsample
dsp_downsample:
stmdb sp!, { r4-r11, lr } @ stack modified regs
ldmib r1, { r5-r6 } @ r5 = num_channels,r6 = resample_data.delta
sub r5, r5, #1 @ pre-decrement num_channels for use
add r4, r1, #12 @ r4 = &resample_data.phase
mov r12, #0xff
orr r12, r12, #0xff00 @ r12 = 0xffff
.dschannel_loop:
ldr r1, [r4] @ r1 = resample_data.phase
ldr r7, [r2, r5, lsl #2] @ r7 = s = src[ch - 1]
ldr r8, [r3, r5, lsl #2] @ r8 = d = dst[ch - 1]
add r9, r4, #4 @ r9 = &last_sample[0]
ldr r10, [r9, r5, lsl #2] @ r10 = last_sample[ch - 1]
sub r11, r0, #1
ldr r14, [r7, r11, lsl #2] @ load last sample in s[] ...
str r14, [r9, r5, lsl #2] @ and write as next frame's last_sample
movs r9, r1, lsr #16 @ r9 = pos = phase >> 16
ldreq r11, [r7] @ if pos = 0, load src[0] and jump into loop
beq .dsuse_last_start
cmp r9, r0 @ if pos >= count, we're already done
bge .dsloop_skip
@ Register usage in loop:
@ r0 = count, r1 = phase, r4 = &resample_data.phase, r5 = cur_channel,
@ r6 = delta, r7 = s, r8 = d, r9 = pos, r10 = s[pos - 1], r11 = s[pos]
.dsloop:
add r9, r7, r9, lsl #2 @ r9 = &s[pos]
ldmda r9, { r10, r11 } @ r10 = s[pos - 1], r11 = s[pos]
.dsuse_last_start:
sub r11, r11, r10 @ r11 = diff = s[pos] - s[pos - 1]
@ keep frac in lower bits to take advantage of multiplier early termination
and r9, r1, r12 @ frac = phase & 0xffff
smull r9, r14, r11, r9
add r10, r10, r14, lsl #16
add r10, r10, r9, lsr #16 @ r10 = out = s[pos - 1] + frac*diff
str r10, [r8], #4 @ *d++ = out
add r1, r1, r6 @ phase += delta
mov r9, r1, lsr #16 @ pos = phase >> 16
cmp r9, r0 @ pos < count?
blt .dsloop @ yup, do more samples
.dsloop_skip:
subs r5, r5, #1
bpl .dschannel_loop @ if (--ch) >= 0, do another channel
sub r1, r1, r0, lsl #16 @ wrap phase back to start
str r1, [r4] @ store back
ldr r1, [r3] @ r1 = &dst[0]
sub r8, r8, r1 @ dst - &dst[0]
mov r0, r8, lsr #2 @ convert bytes->samples
ldmia sp!, { r4-r11, pc } @ ... and we're out
.dsend:
.size dsp_downsample,.dsend-dsp_downsample
/****************************************************************************
* int dsp_upsample(int count, struct dsp_data *dsp,
* in32_t *src[], int32_t *dst[])
*/
.section .text
.global dsp_upsample
dsp_upsample:
stmdb sp!, { r4-r11, lr } @ stack modified regs
ldmib r1, { r5-r6 } @ r5 = num_channels,r6 = resample_data.delta
sub r5, r5, #1 @ pre-decrement num_channels for use
add r4, r1, #12 @ r4 = &resample_data.phase
stmdb sp!, { r0, r4 } @ stack count and &resample_data.phase
.uschannel_loop:
ldr r12, [r4] @ r12 = resample_data.phase
mov r1, r12, ror #16 @ swap halfword positions, we'll use carry
@ to detect pos increments
ldr r7, [r2, r5, lsl #2] @ r7 = s = src[ch - 1]
ldr r8, [r3, r5, lsl #2] @ r8 = d = dst[ch - 1]
add r9, r4, #4 @ r9 = &last_sample[0]
ldr r10, [r9, r5, lsl #2] @ r10 = last_sample[ch - 1]
sub r11, r0, #1
ldr r14, [r7, r11, lsl #2] @ load last sample in s[] ...
str r14, [r9, r5, lsl #2] @ and write as next frame's last_sample
add r9, r7, r0, lsl #2 @ r9 = src_end = &src[count]
movs r14, r12, lsr #16 @ pos = resample_data.phase >> 16
beq .usstart_0 @ pos = 0
cmp r14, r0 @ if pos >= count, we're already done
bge .usloop_skip
add r7, r7, r14, lsl #2 @ r7 = &s[pos]
ldr r10, [r7, #-4] @ r11 = s[pos - 1]
b .usstart_0
@ Register usage in loop:
@ r0 = count, r1 = phase, r4 = &resample_data.phase, r5 = cur_channel,
@ r6 = delta, r7 = s, r8 = d, r9 = src_end, r10 = s[pos - 1], r11 = s[pos]
.usloop_1:
mov r10, r11 @ r10 = previous sample
.usstart_0:
ldr r11, [r7], #4 @ r11 = next sample
sub r0, r11, r10 @ r0 = s[pos] - s[pos - 1]
.usloop_0:
mov r4, r1, lsr #16 @ r4 = frac = phase >> 16
smull r12, r14, r4, r0
add r14, r10, r14, lsl #16
add r14, r14, r12, lsr #16 @ r14 = out = s[pos - 1] + frac*diff
str r14, [r8], #4 @ *d++ = out
adds r1, r1, r6, lsl #16 @ phase += delta << 16
bcc .usloop_0 @ if carry is set, pos is incremented
cmp r7, r9 @ if s < src_end, do another sample
blo .usloop_1
.usloop_skip:
subs r5, r5, #1
ldmia sp, { r0, r4 } @ reload count and &resample_data.phase
bpl .uschannel_loop @ if (--ch) >= 0, do another channel
mov r1, r1, ror #16 @ wrap phase back to start of next frame
str r1, [r4] @ store back
ldr r1, [r3] @ r1 = &dst[0]
sub r8, r8, r1 @ dst - &dst[0]
mov r0, r8, lsr #2 @ convert bytes->samples
add sp, sp, #8 @ adjust stack for temp variables
ldmia sp!, { r4-r11, pc } @ ... and we're out
.usend:
.size dsp_upsample,.usend-dsp_upsample
/****************************************************************************
* void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
* NOTE: The following code processes two samples at once. When count is odd,
* there is an additional obsolete sample processed, which will not be
* used by the calling functions.
*/
.section .icode, "ax", %progbits
.align 2
.global dsp_apply_gain
.type dsp_apply_gain, %function
dsp_apply_gain:
@ input: r0 = count, r1 = data, r2 = buf[]
stmfd sp!, {r4-r8, lr}
ldr r3, [r1, #4] @ r3 = data->num_channels
ldr r4, [r1, #32] @ r5 = data->gain
.dag_outerloop:
ldr r1, [r2], #4 @ r1 = buf[0] and increment index of buf[]
mov lr, r0 @ lr = r0 = count
.dag_innerloop:
ldmia r1, {r5, r6} @ load r5, r6 from r1
smull r7, r8, r5, r4 @ r5 = FRACMUL_SHL(r5, r4, 8)
mov r8, r8, asl #9
orr r5, r8, r7, lsr #23
smull r7, r8, r6, r4 @ r6 = FRACMUL_SHL(r6, r4, 8)
mov r8, r8, asl #9
orr r6, r8, r7, lsr #23
stmia r1!, {r5, r6} @ save r5, r6 to r1 and increment r1
subs lr, lr, #2
bgt .dag_innerloop @ end of inner loop
subs r3, r3, #1
bgt .dag_outerloop @ end of outer loop
ldmfd sp!, {r4-r8, pc}
.dagend:
.size dsp_apply_gain,.dagend-dsp_apply_gain