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rockbox/lib/rbcodec/dsp/dsp_arm_v6.S

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Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2010 Michael Sevakis
*
* 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.
*
****************************************************************************/
/****************************************************************************
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
* void sample_output_mono(struct sample_io_data *this,
* struct dsp_buffer *src,
* struct dsp_buffer *dst)
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
*/
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
.section .text
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
.global sample_output_mono
.type sample_output_mono, %function
sample_output_mono:
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
@ input: r0 = this, r1 = src, r2 = dst
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
stmfd sp!, { r4, lr } @
@
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
ldr r0, [r0] @ r0 = this->outcount
ldr r3, [r2, #4] @ r3 = dst->p16out
ldr r2, [r1, #4] @ r2 = src->p32[0]
ldrb r1, [r1, #19] @ r1 = src->format.output_scale
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
@
mov r4, #1 @ r4 = 1 << (scale - 1)
mov r4, r4, lsl r1 @
subs r0, r0, #1 @ odd: end at 0; even: end at -1
mov r4, r4, lsr #1 @
beq 2f @ Zero? Only one sample!
@
1: @
ldmia r2!, { r12, r14 } @ load Mi0, Mi1
qadd r12, r12, r4 @ round, scale, saturate and
qadd r14, r14, r4 @ pack Mi0 to So0, Mi1 to So1
mov r12, r12, asr r1 @
mov r14, r14, asr r1 @
ssat r12, #16, r12 @
ssat r14, #16, r14 @
pkhbt r12, r12, r12, asl #16 @
pkhbt r14, r14, r14, asl #16 @
subs r0, r0, #2 @
stmia r3!, { r12, r14 } @ store So0, So1
bgt 1b @
@
ldmltfd sp!, { r4, pc } @ if count was even, we're done
@
2: @
ldr r12, [r2] @ round, scale, saturate
qadd r12, r12, r4 @ and pack Mi to So
mov r12, r12, asr r1 @
ssat r12, #16, r12 @
pkhbt r12, r12, r12, asl #16 @
str r12, [r3] @ store So
Some whitespace correction in dsp_*.S Change-Id: I6ee14c0adc9dd456c8a2d171952cbaedb3752428
2012-04-27 20:55:16 +00:00
@
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
ldmfd sp!, { r4, pc } @
.size sample_output_mono, .-sample_output_mono
/****************************************************************************
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
* void sample_output_stereo(struct sample_io_data *this,
* struct dsp_buffer *src,
* struct dsp_buffer *dst)
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
*/
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
.section .text
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
.global sample_output_stereo
.type sample_output_stereo, %function
sample_output_stereo:
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
@ input: r0 = this, r1 = src, r2 = dst
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
stmfd sp!, { r4-r7, lr } @
@
Fundamentally rewrite much of the audio DSP. Creates a standard buffer passing, local data passing and messaging system for processing stages. Stages can be moved to their own source files to reduce clutter and ease assimilation of new ones. dsp.c becomes dsp_core.c which supports an engine and framework for effects. Formats and change notifications are passed along with the buffer so that they arrive at the correct time at each stage in the chain regardless of the internal delays of a particular one. Removes restrictions on the number of samples that can be processed at a time and it pays attention to destination buffer size restrictions without having to limit input count, which also allows pcmbuf to remain fuller and safely set its own buffer limits as it sees fit. There is no longer a need to query input/output counts given a certain number of input samples; just give it the sizes of the source and destination buffers. Works in harmony with stages that are not deterministic in terms of sample input/output ratio (like both resamplers but most notably the timestretch). As a result it fixes quirks with timestretch hanging up with certain settings and it now operates properly throughout its full settings range. Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734 Reviewed-on: http://gerrit.rockbox.org/200 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-27 23:52:15 +00:00
ldr r0, [r0] @ r0 = this->outcount
ldr r3, [r2, #4] @ r3 = dst->p16out
ldmib r1, { r2, r4 } @ r2 = src->p32[0], r4 = src->p32[1]
ldrb r1, [r1, #19] @ r1 = src->format.output_scale
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
@
mov r5, #1 @ r5 = 1 << (scale - 1)
mov r5, r5, lsl r1 @
subs r0, r0, #1 @ odd: end at 0; even: end at -1
mov r5, r5, lsr #1 @
beq 2f @ Zero? Only one sample!
@
1: @
ldmia r2!, { r6, r7 } @ r6, r7 = Li0, Li1
ldmia r4!, { r12, r14 } @ r12, r14 = Ri0, Ri1
qadd r6, r6, r5 @ round, scale, saturate and pack
qadd r7, r7, r5 @ Li0+Ri0 to So0, Li1+Ri1 to So1
qadd r12, r12, r5 @
qadd r14, r14, r5 @
Some whitespace correction in dsp_*.S Change-Id: I6ee14c0adc9dd456c8a2d171952cbaedb3752428
2012-04-27 20:55:16 +00:00
mov r6, r6, asr r1 @
Optimized DSP sample out functions for armv6. (For stereo output) ~9% faster than SVN asm and about 4% faster than SVN asm rearranged to observe pipeline hazards. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25717 a1c6a512-1295-4272-9138-f99709370657
2010-04-25 20:04:47 +00:00
mov r7, r7, asr r1 @
mov r12, r12, asr r1 @
mov r14, r14, asr r1 @
ssat r6, #16, r6 @
ssat r12, #16, r12 @
ssat r7, #16, r7 @
ssat r14, #16, r14 @
pkhbt r6, r6, r12, asl #16 @
pkhbt r7, r7, r14, asl #16 @
subs r0, r0, #2 @
stmia r3!, { r6, r7 } @ store So0, So1
bgt 1b @
@
ldmltfd sp!, { r4-r7, pc } @ if count was even, we're done
@
2: @
ldr r6, [r2] @ r6 = Li
ldr r12, [r4] @ r12 = Ri
qadd r6, r6, r5 @ round, scale, saturate
qadd r12, r12, r5 @ and pack Li+Ri to So
mov r6, r6, asr r1 @
mov r12, r12, asr r1 @
ssat r6, #16, r6 @
ssat r12, #16, r12 @
pkhbt r6, r6, r12, asl #16 @
str r6, [r3] @ store So
@
ldmfd sp!, { r4-r7, pc } @
.size sample_output_stereo, .-sample_output_stereo
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