SWCODEC: Tighten up coldfire assembly a little bit more. Cleanup to make differing parameters between ARM and Coldfire halfway clean. Hopefully those differences can be reconciled soon. A tiny bit of C optimizing for karaoke channel mode.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12505 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis 2007-02-27 14:25:36 +00:00
parent 8ca99d3288
commit 6fbdb912b0
3 changed files with 159 additions and 153 deletions

View file

@ -112,7 +112,7 @@ struct crossfeed_data
int32_t coefs[3]; /* 04h - Coefficients for the shelving filter */
int32_t history[4]; /* 10h - Format is x[n - 1], y[n - 1] for both channels */
int32_t delay[13][2]; /* 20h */
int index; /* 88h - Current index into the delay line */
int index; /* 88h - Current index/pointer into the delay line */
/* 8ch */
};
@ -129,13 +129,21 @@ struct eq_state
/* Include header with defines which functions are implemented in assembly
code for the target */
#ifndef SIMULATOR
#include <dsp_asm.h>
#endif
#ifndef DSP_HAVE_ASM_CROSSFEED
static void apply_crossfeed(int32_t *buf[], int count);
#endif
/* Typedefs keep things much neater in this case */
typedef int (*sample_input_fn_type)(int count, const char *src[],
int32_t *dst[]);
typedef int (*resample_fn_type)(int count, struct dsp_data *data,
int32_t *src[], int32_t *dst[]);
typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
int32_t *src[], int16_t *dst);
/* If ACF_SWITCHPARAM is no longer needed, make apply_crossfeed of type
channels_process_fn_type since it is really just that */
typedef void (*apply_crossfeed_fn_type)(ACF_SWITCHPARAM(int count,
int32_t *buf[]));
typedef void (*channels_process_fn_type)(int count, int32_t *buf[]);
/*
***************************************************************************/
@ -151,15 +159,13 @@ struct dsp_config
long gain; /* Note that this is in S8.23 format. */
/* Functions that change depending upon settings - NULL if stage is
disabled */
int (*input_samples)(int count, const char *src[], int32_t *dst[]);
int (*resample)(int count, struct dsp_data *data,
int32_t *src[], int32_t *dst[]);
void (*output_samples)(int count, struct dsp_data *data,
int32_t *src[], int16_t *dst);
sample_input_fn_type input_samples;
resample_fn_type resample;
sample_output_fn_type output_samples;
/* These will be NULL for the voice codec and is more economical that
way */
void (*apply_crossfeed)(int32_t *src[], int count);
void (*channels_process)(int count, int32_t *buf[]);
apply_crossfeed_fn_type apply_crossfeed;
channels_process_fn_type channels_process;
};
/* General DSP config */
@ -169,7 +175,14 @@ static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */
static long dither_mask IBSS_ATTR;
static long dither_bias IBSS_ATTR;
/* Crossfeed */
struct crossfeed_data crossfeed_data IBSS_ATTR; /* A */
struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */
{
#ifdef DSP_CROSSFEED_DELAY_PTR
.index = (intptr_t)crossfeed_data.delay
#else
.index = 0
#endif
};
/* Equalizer */
static struct eq_state eq_data; /* A/V */
#ifdef HAVE_SW_TONE_CONTROLS
@ -401,8 +414,7 @@ static int sample_input_gt_native_ni_stereo(
*/
static void sample_input_new_format(void)
{
static int (* const sample_input_functions[])(
int count, const char* src[], int32_t *dst[]) =
static const sample_input_fn_type sample_input_functions[] =
{
[SAMPLE_INPUT_LE_NATIVE_MONO] = sample_input_lte_native_mono,
[SAMPLE_INPUT_LE_NATIVE_I_STEREO] = sample_input_lte_native_i_stereo,
@ -539,9 +551,7 @@ static void sample_output_dithered(int count, struct dsp_data *data,
*/
static void sample_output_new_format(void)
{
static void (* const sample_output_functions[])(
int count, struct dsp_data *data,
int32_t *src[], int16_t *dst) =
static const sample_output_fn_type sample_output_functions[] =
{
sample_output_mono,
sample_output_stereo,
@ -695,42 +705,13 @@ void dsp_dither_enable(bool enable)
switch_dsp(old_dsp);
}
/**
* dsp_set_crossfeed(bool enable)
*
* !DSPPARAMSYNC
* needs syncing with changes to the following dsp parameters:
* * dsp->stereo_mode (A)
*/
void dsp_set_crossfeed(bool enable)
{
crossfeed_enabled = enable;
audio_dsp->apply_crossfeed =
(enable && audio_dsp->data.num_channels > 1)
? apply_crossfeed : NULL;
}
void dsp_set_crossfeed_direct_gain(int gain)
{
crossfeed_data.gain = get_replaygain_int(gain * -10) << 7;
}
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
long g1 = get_replaygain_int(lf_gain * -10) << 3;
long g2 = get_replaygain_int(hf_gain * -10) << 3;
filter_shelf_coefs(0xffffffff/NATIVE_FREQUENCY*cutoff, g1, g2,
crossfeed_data.coefs);
}
/* Applies crossfeed to the stereo signal in src.
* Crossfeed is a process where listening over speakers is simulated. This
* is good for old hard panned stereo records, which might be quite fatiguing
* to listen to on headphones with no crossfeed.
*/
#ifndef DSP_HAVE_ASM_CROSSFEED
static void apply_crossfeed(int32_t *buf[], int count)
static void apply_crossfeed(int count, int32_t *buf[])
{
int32_t *hist_l = &crossfeed_data.history[0];
int32_t *hist_r = &crossfeed_data.history[2];
@ -775,7 +756,36 @@ static void apply_crossfeed(int32_t *buf[], int count)
/* Write back local copies of data we've modified */
crossfeed_data.index = di;
}
#endif
#endif /* DSP_HAVE_ASM_CROSSFEED */
/**
* dsp_set_crossfeed(bool enable)
*
* !DSPPARAMSYNC
* needs syncing with changes to the following dsp parameters:
* * dsp->stereo_mode (A)
*/
void dsp_set_crossfeed(bool enable)
{
crossfeed_enabled = enable;
audio_dsp->apply_crossfeed =
(enable && audio_dsp->data.num_channels > 1)
? apply_crossfeed : NULL;
}
void dsp_set_crossfeed_direct_gain(int gain)
{
crossfeed_data.gain = get_replaygain_int(gain * -10) << 7;
}
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
long g1 = get_replaygain_int(lf_gain * -10) << 3;
long g2 = get_replaygain_int(hf_gain * -10) << 3;
filter_shelf_coefs(0xffffffff/NATIVE_FREQUENCY*cutoff, g1, g2,
crossfeed_data.coefs);
}
/* Combine all gains to a global gain. */
static void set_gain(struct dsp_config *dsp)
@ -1056,10 +1066,9 @@ static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
do
{
int32_t l = *sl/2;
int32_t r = *sr/2;
*sl++ = l - r;
*sr++ = r - l;
int32_t ch = *sl/2 - *sr/2;
*sl++ = ch;
*sr++ = -ch;
}
while (--count > 0);
}
@ -1067,8 +1076,7 @@ static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
void channels_set(int value)
{
static void (* const channels_process_functions[])(
int count, int32_t *buf[]) =
static const channels_process_fn_type channels_process_functions[] =
{
/* SOUND_CHAN_STEREO = All-purpose index for no channel processing */
[SOUND_CHAN_STEREO] = NULL,
@ -1118,7 +1126,7 @@ int dsp_process(char *dst, const char *src[], int count)
if ((samples = resample(samples, tmp)) <= 0)
break; /* I'm pretty sure we're downsampling here */
if (dsp->apply_crossfeed)
dsp->apply_crossfeed(tmp, samples);
dsp->apply_crossfeed(ACF_SWITCHPARAM(samples, tmp));
/* TODO: EQ and tone controls need separate structs for audio and voice
* DSP processing thanks to filter history. isn't really audible now, but
* might be the day we start handling voice more delicately.

View file

@ -22,10 +22,22 @@
#ifndef _DSP_ASM_H
#define _DSP_ASM_H
#define ACF_SWITCHPARAM(count, buf) count, buf
#ifndef SIMULATOR
#if defined(CPU_COLDFIRE) || defined(CPU_ARM)
#define DSP_HAVE_ASM_CROSSFEED
void apply_crossfeed(int32_t *src[], int count);
#if defined(CPU_COLDFIRE)
/* ACF_SWITCHPARAM can be stripped out if all have the same parameter
order - DSP_CROSSFEED_DELAY_PTR if all use a pointer instead of index */
#define DSP_CROSSFEED_DELAY_PTR
#else
#undef ACF_SWITCHPARAM
#define ACF_SWITCHPARAM(count, buf) buf, count
#endif
void apply_crossfeed(ACF_SWITCHPARAM(int count, int32_t *buf[]));
#endif /* defined(CPU_COLDFIRE) || defined(CPU_ARM) */
#if defined (CPU_COLDFIRE)
#define DSP_HAVE_ASM_RESAMPLING
@ -45,5 +57,8 @@ void sample_output_mono(int count, struct dsp_data *data,
#define DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
void sample_output_stereo(int count, struct dsp_data *data,
int32_t *src[], int16_t *dst);
#endif
#endif /* CPU_COLDFIRE */
#endif /* SIMULATOR */
#endif /* _DSP_ASM_H */

View file

@ -8,6 +8,7 @@
* $Id$
*
* Copyright (C) 2006 Thom Johansen
* Portions Copyright (C) 2007 Michael Sevakis
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
@ -18,75 +19,63 @@
****************************************************************************/
/****************************************************************************
* void apply_crossfeed(int32_t *src[], int count)
* void apply_crossfeed(int count, int32_t *src[])
*/
.section .text
.global apply_crossfeed
apply_crossfeed:
lea.l (-44, %sp), %sp
movem.l %d2-%d7/%a2-%a6, (%sp) | save all regs
move.l (44+4, %sp), %a4
movem.l (%a4), %a4-%a5 | a4 = src[0], a5 = src[1]
move.l (44+8, %sp), %d7 | d7 = count
lea.l crossfeed_data, %a1
lea.l (8*4, %a1), %a0 | a0 = &delay[0][0]
move.l (%a1)+, %a6 | a6 = direct gain
movem.l (3*4, %a1), %d0-%d3 | fetch filter history samples
move.l (33*4, %a1), %d4 | fetch delay line index
movem.l (%a1), %a1-%a3 | load filter coefs
move.l %d4, %d5
lsl.l #3, %d5
add.l %d5, %a0 | point a0 to current delay position
| lea.l (%d4*4, %a0), %a0
| lea.l (%d4*4, %a0), %a0 | point a0 to current delay position
lea.l -44(%sp), %sp
movem.l %d2-%d7/%a2-%a6, (%sp) | save all regs
movem.l 48(%sp), %d7/%a4 | %d7 = count, %a4 = src
movem.l (%a4), %a4-%a5 | %a4 = src[0], %a5 = src[1]
lea.l crossfeed_data, %a1
move.l (%a1)+, %a6 | a6 = direct gain
movem.l 12(%a1), %d0-%d3 | fetch filter history samples
move.l 132(%a1), %a0 | fetch delay line address
movem.l (%a1), %a1-%a3 | load filter coefs
/* Register usage in loop:
* a0 = &delay[index][0], a1..a3 = b0, b1, a1 (filter coefs),
* a4 = src[0], a5 = src[1], a6 = direct gain,
* d0..d3 = history
* d4 = delay line index,
* d5,d6 = temp.
* d7 = count
* %a0 = delay_p, %a1..%a3 = b0, b1, a1 (filter coefs),
* %a4 = src[0], %a5 = src[1], %a6 = direct gain,
* %d0..%d3 = history
* %d4..%d6 = temp.
* %d7 = count
*/
.cfloop:
mac.l %a2, %d0, (4, %a0), %d0, %acc0 | acc = b1*dr[n - 1] d0 = dr[n]
mac.l %a1, %d0, %acc0 | acc += b0*dr[n]
mac.l %a3, %d1, (%a4), %d5, %acc0 | acc += a1*y_l[n - 1], load left input
move.l %acc0, %d1 | get filtered delayed sample
mac.l %a6, %d5, %acc0 | acc += gain*x_l[n]
movclr.l %acc0, %d6
move.l %d6, (%a4)+ | write result
mac.l %a2, %d0, 4(%a0), %d0, %acc0 | acc = b1*dr[n - 1] d0 = dr[n]
mac.l %a1, %d0 , %acc0 | acc += b0*dr[n]
mac.l %a3, %d1, (%a4), %d4, %acc0 | acc += a1*y_l[n - 1], load L
move.l %acc0, %d1 | get filtered delayed sample
mac.l %a6, %d4, %acc0 | acc += gain*x_l[n]
movclr.l %acc0, %d6 |
move.l %d6, (%a4)+ | write result
mac.l %a2, %d2, (%a0), %d2, %acc0 | acc = b1*dl[n - 1], d2 = dl[n]
move.l %d5, (%a0)+ | save left input to delay line
mac.l %a1, %d2, %acc0 | acc += b0*dl[n]
mac.l %a3, %d3, (%a5), %d5, %acc0 | acc += a1*y_r[n - 1], load right input
move.l %acc0, %d3 | get filtered delayed sample
mac.l %a6, %d5, %acc0 | acc += gain*x_r[n]
move.l %d5, (%a0)+ | save right input to delay line
movclr.l %acc0, %d6
move.l %d6, (%a5)+ | write result
mac.l %a2, %d2, (%a0), %d2, %acc0 | acc = b1*dl[n - 1], d2 = dl[n]
mac.l %a1, %d2 , %acc0 | acc += b0*dl[n]
mac.l %a3, %d3, (%a5), %d5, %acc0 | acc += a1*y_r[n - 1], load R
movem.l %d4-%d5, (%a0) | save left & right inputs to delay line
move.l %acc0, %d3 | get filtered delayed sample
mac.l %a6, %d5, %acc0 | acc += gain*x_r[n]
lea.l 8(%a0), %a0 | increment delay pointer
movclr.l %acc0, %d6 |
move.l %d6, (%a5)+ | write result
addq.l #1, %d4 | index++
moveq.l #13, %d6
cmp.l %d6, %d4 | wrap index to 0 if it overflows
jlt .cfnowrap
moveq.l #13*8, %d4
sub.l %d4, %a0 | wrap back delay line ptr as well
clr.l %d4
.cfnowrap:
subq.l #1, %d7
jne .cfloop
| save data back to struct
lea.l crossfeed_data + 4*4, %a1
movem.l %d0-%d3, (%a1)
move.l %d4, (30*4, %a1)
movem.l (%sp), %d2-%d7/%a2-%a6
lea.l (44, %sp), %sp
cmpa.l #crossfeed_data+136, %a0| wrap a0 if passed end
bge.b .cfwrap |
.word 0x51fb | tpf.l - trap the buffer wrap
.cfwrap:
lea.l -104(%a0), %a0 | wrap
subq.l #1, %d7 | --count < 0 ?
bgt.b .cfloop |
lea.l crossfeed_data+16, %a1 | save data back to struct
movem.l %d0-%d3, (%a1) | ...history
move.l %a0, 120(%a1) | ...delay_p
movem.l (%sp), %d2-%d7/%a2-%a6 | restore all regs
lea.l 44(%sp), %sp
rts
.cfend:
.size apply_crossfeed,.cfend-apply_crossfeed
/****************************************************************************
* int dsp_downsample(int count, struct dsp_data *data,
* in32_t *src[], int32_t *dst[])
@ -128,10 +117,10 @@ dsp_downsample:
lsl.l %d7, %d0 |
lsr.l #1, %d0 |
mac.l %d0, %d1, %acc0 | %acc0 += frac * diff
move.l %acc0, %d0 |
add.l %d4, %d5 | phase += delta
move.l %d5, %d6 | pos = phase >> 16
lsr.l %d7, %d6 |
movclr.l %acc0, %d0 |
move.l %d0, (%a4)+ | *d++ = %d0
cmp.l %d2, %d6 | pos < count?
blt.b .dsloop | yes? continue resampling
@ -145,7 +134,6 @@ dsp_downsample:
sub.l (%a2), %d0 |
asr.l #2, %d0 | convert bytes->samples
movem.l (%sp), %d2-%d7/%a2-%a5 | restore non-clobberables
move.l %acc1, %acc0 | clear %acc0
lea.l 40(%sp), %sp | cleanup stack
rts | buh-bye
.dsend:
@ -196,8 +184,8 @@ dsp_upsample:
.usloop_0:
lsr.l #1, %d5 | make phase into frac
mac.l %d1, %d5, %acc0 | %acc0 = diff * frac
movclr.l %acc0, %d7 | %d7 = product
lsl.l #1, %d5 | restore frac to phase
movclr.l %acc0, %d7 | %d7 = product
add.l %d0, %d7 | %d7 = last + product
move.l %d7, (%a4)+ | *d++ = %d7
add.l %d4, %d5 | phase += delta
@ -272,10 +260,10 @@ channels_process_sound_chan_custom:
move.l dsp_sw_cross, %d4 | load cross (side) gain
1:
move.l (%a0), %d1 |
mac.l %d1, %d3 , (%a1), %d2, %acc0 | L = l*gain + r*cross
mac.l %d1, %d4 , %acc1 | R = r*gain + l*cross
mac.l %d2, %d4 , %acc0 |
mac.l %d2, %d3 , %acc1 |
mac.l %d1, %d3, (%a1), %d2, %acc0 | L = l*gain + r*cross
mac.l %d1, %d4 , %acc1 | R = r*gain + l*cross
mac.l %d2, %d4 , %acc0 |
mac.l %d2, %d3 , %acc1 |
movclr.l %acc0, %d1 |
movclr.l %acc1, %d2 |
move.l %d1, (%a0)+ |
@ -306,15 +294,12 @@ channels_process_sound_chan_karaoke:
move.l #0x40000000, %d4 | %d3 = 0.5
1:
move.l (%a0), %d1 |
mac.l %d1, %d4, (%a1), %d2, %acc0 | L = l/2 - r/2
mac.l %d2, %d4, %acc1 | R = r/2 - l/2
msac.l %d1, %d4, (%a1), %d2, %acc0 | R = r/2 - l/2
mac.l %d2, %d4 , %acc0 |
movclr.l %acc0, %d1 |
movclr.l %acc1, %d2 |
move.l %d1, %d3 |
sub.l %d2, %d1 |
sub.l %d3, %d2 |
move.l %d1, (%a1)+ |
neg.l %d1 | L = -R = -(r/2 - l/2) = l/2 - r/2
move.l %d1, (%a0)+ |
move.l %d2, (%a1)+ |
subq.l #1, %d0 |
bgt.s 1b |
movem.l (%sp), %d1-%d4 | restore registers
@ -323,7 +308,6 @@ channels_process_sound_chan_karaoke:
rts
.cpkaraoke_end:
.size channels_process_sound_chan_karaoke, .cpkaraoke_end-channels_process_sound_chan_karaoke
/****************************************************************************
* void sample_output_stereo(int count, struct dsp_data *data,
* int32_t *src[], int16_t *dst)
@ -382,34 +366,33 @@ sample_output_stereo:
.sos_lineloop_start:
lea.l -12(%a0), %a5 | %a5 = at or just before last line bound
.sos_lineloop:
move.l (%a2)+, %d0 | get next 4 L samples and scale
mac.l %d0, %a1, (%a2)+, %d1, %acc0 | with saturation
mac.l %d1, %a1, (%a2)+, %d2, %acc1 |
mac.l %d2, %a1, (%a2)+, %d3, %acc2 |
mac.l %d3, %a1, %acc3 |
movclr.l %acc0, %d0 | obtain results
movclr.l %acc1, %d1 |
movclr.l %acc2, %d2 |
movclr.l %acc3, %d3 |
move.l (%a3)+, %d4 | get next 4 R samples and scale
mac.l %d4, %a1, (%a3)+, %d5, %acc0 | with saturation
mac.l %d5, %a1, (%a3)+, %d6, %acc1 |
mac.l %d6, %a1, (%a3)+, %d7, %acc2 |
mac.l %d7, %a1, %acc3 |
movclr.l %acc0, %d4 | obtain results
mac.l %d4, %a1, (%a3)+, %d5, %acc0 | with saturation
mac.l %d5, %a1, (%a3)+, %d6, %acc1 |
mac.l %d6, %a1, (%a3)+, %d7, %acc2 |
mac.l %d7, %a1, (%a2)+, %d0, %acc3 |
lea.l 16(%a4), %a4 | increment dest here, mitigate stalls
movclr.l %acc0, %d4 | obtain R results
movclr.l %acc1, %d5 |
movclr.l %acc2, %d6 |
movclr.l %acc3, %d7 |
swap %d4 | interleave most significant
move.w %d4, %d0 | 16 bits of L and R
mac.l %d0, %a1, (%a2)+, %d1, %acc0 | get next 4 L samples and scale
mac.l %d1, %a1, (%a2)+, %d2, %acc1 | with saturation
mac.l %d2, %a1, (%a2)+, %d3, %acc2 |
mac.l %d3, %a1 , %acc3 |
swap %d4 | a) interleave most significant...
swap %d5 |
move.w %d5, %d1 |
swap %d6 |
move.w %d6, %d2 |
swap %d7 |
movclr.l %acc0, %d0 | obtain L results
movclr.l %acc1, %d1 |
movclr.l %acc2, %d2 |
movclr.l %acc3, %d3 |
move.w %d4, %d0 | a) ... 16 bits of L and R
move.w %d5, %d1 |
move.w %d6, %d2 |
move.w %d7, %d3 |
movem.l %d0-%d3, (%a4) | write four stereo samples
lea.l 16(%a4), %a4 |
movem.l %d0-%d3, -16(%a4) | write four stereo samples
cmp.l %a4, %a5 |
bhi.b .sos_lineloop |
.sos_longloop_1_start:
@ -480,7 +463,8 @@ sample_output_mono:
mac.l %d0, %d5, (%a2)+, %d1, %acc0 | with saturation
mac.l %d1, %d5, (%a2)+, %d2, %acc1 |
mac.l %d2, %d5, (%a2)+, %d3, %acc2 |
mac.l %d3, %d5, %acc3 |
mac.l %d3, %d5 , %acc3 |
lea.l 16(%a3), %a3 | increment dest here, mitigate stalls
movclr.l %acc0, %d0 | obtain results
movclr.l %acc1, %d1 |
movclr.l %acc2, %d2 |
@ -497,8 +481,7 @@ sample_output_mono:
move.l %d3, %d4 |
swap %d4 |
move.w %d4, %d3 |
movem.l %d0-%d3, (%a3) | write four stereo samples
lea.l 16(%a3), %a3 |
movem.l %d0-%d3, -16(%a3) | write four stereo samples
cmp.l %a3, %a1 |
bhi.b .som_lineloop |
.som_longloop_1_start: