ARM DSP: Add assembly custom sound channel processing. 13% to 14% faster than currently-used default C code on ARMv4.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25949 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
parent
68da06f3dc
commit
565a863dd5
2 changed files with 77 additions and 9 deletions
|
@ -22,10 +22,6 @@
|
|||
|
||||
/****************************************************************************
|
||||
* 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
|
||||
|
@ -63,13 +59,84 @@ channels_process_sound_chan_mono:
|
|||
@
|
||||
ldmfd sp!, { r4, pc } @
|
||||
.size channels_process_sound_chan_mono, \
|
||||
.-channels_process_sound_chan_mono
|
||||
|
||||
.-channels_process_sound_chan_mono
|
||||
|
||||
/****************************************************************************
|
||||
* void channels_process_sound_chan_custom(int count, int32_t *buf[])
|
||||
*/
|
||||
.section .icode, "ax", %progbits
|
||||
.align 2
|
||||
.global channels_process_sound_chan_custom
|
||||
.type channels_process_sound_chan_custom, %function
|
||||
channels_process_sound_chan_custom:
|
||||
stmfd sp!, { r4-r10, lr }
|
||||
|
||||
ldr r3, =dsp_sw_gain
|
||||
ldr r4, =dsp_sw_cross
|
||||
|
||||
ldmia r1, { r1, r2 } @ r1 = buf[0], r2 = buf[1]
|
||||
ldr r3, [r3] @ r3 = dsp_sw_gain
|
||||
ldr r4, [r4] @ r4 = dsp_sw_cross
|
||||
|
||||
subs r0, r0, #1
|
||||
beq .custom_single_sample @ Zero? Only one sample!
|
||||
|
||||
.custom_loop:
|
||||
ldmia r1, { r5, r6 } @ r5 = Li0, r6 = Li1
|
||||
ldmia r2, { r7, r8 } @ r7 = Ri0, r8 = Ri1
|
||||
|
||||
subs r0, r0, #2
|
||||
|
||||
smull r9, r10, r5, r3 @ Lc0 = Li0*gain
|
||||
smull r12, r14, r7, r3 @ Rc0 = Ri0*gain
|
||||
smlal r9, r10, r7, r4 @ Lc0 += Ri0*cross
|
||||
smlal r12, r14, r5, r4 @ Rc0 += Li0*cross
|
||||
|
||||
mov r9, r9, lsr #31 @ Convert to s0.31
|
||||
mov r12, r12, lsr #31
|
||||
orr r5, r9, r10, asl #1
|
||||
orr r7, r12, r14, asl #1
|
||||
|
||||
smull r9, r10, r6, r3 @ Lc1 = Li1*gain
|
||||
smull r12, r14, r8, r3 @ Rc1 = Ri1*gain
|
||||
smlal r9, r10, r8, r4 @ Lc1 += Ri1*cross
|
||||
smlal r12, r14, r6, r4 @ Rc1 += Li1*cross
|
||||
|
||||
mov r9, r9, lsr #31 @ Convert to s0.31
|
||||
mov r12, r12, lsr #31
|
||||
orr r6, r9, r10, asl #1
|
||||
orr r8, r12, r14, asl #1
|
||||
|
||||
stmia r1!, { r5, r6 } @ Store Lc0, Lc1
|
||||
stmia r2!, { r7, r8 } @ Store Rc0, Rc1
|
||||
|
||||
bgt .custom_loop
|
||||
|
||||
ldmltfd sp!, { r4-r10, pc } @ < 0? even count
|
||||
|
||||
.custom_single_sample:
|
||||
ldr r5, [r1] @ handle odd sample
|
||||
ldr r7, [r2]
|
||||
|
||||
smull r9, r10, r5, r3 @ Lc0 = Li0*gain
|
||||
smull r12, r14, r7, r3 @ Rc0 = Ri0*gain
|
||||
smlal r9, r10, r7, r4 @ Lc0 += Ri0*cross
|
||||
smlal r12, r14, r5, r4 @ Rc0 += Li0*cross
|
||||
|
||||
mov r9, r9, lsr #31 @ Convert to s0.31
|
||||
mov r12, r12, lsr #31
|
||||
orr r5, r9, r10, asl #1
|
||||
orr r7, r12, r14, asl #1
|
||||
|
||||
str r5, [r1] @ Store Lc0
|
||||
str r7, [r2] @ Store Rc0
|
||||
|
||||
ldmfd sp!, { r4-r10, pc }
|
||||
.size channels_process_sound_chan_custom, \
|
||||
.-channels_process_sound_chan_custom
|
||||
|
||||
/****************************************************************************
|
||||
* 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
|
||||
|
|
|
@ -30,6 +30,7 @@
|
|||
#define DSP_HAVE_ASM_RESAMPLING
|
||||
#define DSP_HAVE_ASM_CROSSFEED
|
||||
#define DSP_HAVE_ASM_SOUND_CHAN_MONO
|
||||
#define DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
|
||||
#define DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
|
||||
#define DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
|
||||
#define DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
|
||||
|
|
Loading…
Reference in a new issue