Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
/*-
|
|
|
|
* Copyright (c) 2008-2009 Ariff Abdullah <ariff@FreeBSD.org>
|
|
|
|
* All rights reserved.
|
|
|
|
*
|
|
|
|
* Redistribution and use in source and binary forms, with or without
|
|
|
|
* modification, are permitted provided that the following conditions
|
|
|
|
* are met:
|
|
|
|
* 1. Redistributions of source code must retain the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer.
|
|
|
|
* 2. Redistributions in binary form must reproduce the above copyright
|
|
|
|
* notice, this list of conditions and the following disclaimer in the
|
|
|
|
* documentation and/or other materials provided with the distribution.
|
|
|
|
*
|
|
|
|
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
|
|
|
|
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
|
|
|
|
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
|
|
|
|
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
|
|
|
|
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
|
|
|
|
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
|
|
|
|
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
|
|
|
|
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
|
|
|
|
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
|
|
|
|
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
|
|
|
|
* SUCH DAMAGE.
|
|
|
|
*/
|
|
|
|
|
|
|
|
/*
|
|
|
|
* feeder_eq: Parametric (compile time) Software Equalizer. Though accidental,
|
|
|
|
* it proves good enough for educational and general consumption.
|
|
|
|
*
|
|
|
|
* "Cookbook formulae for audio EQ biquad filter coefficients"
|
|
|
|
* by Robert Bristow-Johnson <rbj@audioimagination.com>
|
|
|
|
* - http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
|
|
|
|
*/
|
|
|
|
|
|
|
|
#ifdef _KERNEL
|
|
|
|
#ifdef HAVE_KERNEL_OPTION_HEADERS
|
|
|
|
#include "opt_snd.h"
|
|
|
|
#endif
|
|
|
|
#include <dev/sound/pcm/sound.h>
|
|
|
|
#include <dev/sound/pcm/pcm.h>
|
|
|
|
#include "feeder_if.h"
|
|
|
|
|
|
|
|
#define SND_USE_FXDIV
|
|
|
|
#include "snd_fxdiv_gen.h"
|
|
|
|
|
|
|
|
SND_DECLARE_FILE("$FreeBSD$");
|
|
|
|
#endif
|
|
|
|
|
|
|
|
#include "feeder_eq_gen.h"
|
|
|
|
|
|
|
|
#define FEEDEQ_LEVELS \
|
|
|
|
(((FEEDEQ_GAIN_MAX - FEEDEQ_GAIN_MIN) * \
|
|
|
|
(FEEDEQ_GAIN_DIV / FEEDEQ_GAIN_STEP)) + 1)
|
|
|
|
|
|
|
|
#define FEEDEQ_L2GAIN(v) \
|
|
|
|
((int)min(((v) * FEEDEQ_LEVELS) / 100, FEEDEQ_LEVELS - 1))
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_IPART(x) (abs(x) >> FEEDEQ_GAIN_SHIFT)
|
|
|
|
#define FEEDEQ_PREAMP_FPART(x) (abs(x) & FEEDEQ_GAIN_FMASK)
|
|
|
|
#define FEEDEQ_PREAMP_SIGNVAL(x) ((x) < 0 ? -1 : 1)
|
|
|
|
#define FEEDEQ_PREAMP_SIGNMARK(x) (((x) < 0) ? '-' : '+')
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_IMIN -192
|
|
|
|
#define FEEDEQ_PREAMP_IMAX 192
|
|
|
|
#define FEEDEQ_PREAMP_FMIN 0
|
|
|
|
#define FEEDEQ_PREAMP_FMAX 9
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_INVALID INT_MAX
|
|
|
|
|
|
|
|
#define FEEDEQ_IF2PREAMP(i, f) \
|
|
|
|
((abs(i) << FEEDEQ_GAIN_SHIFT) | \
|
|
|
|
(((abs(f) / FEEDEQ_GAIN_STEP) * FEEDEQ_GAIN_STEP) & \
|
|
|
|
FEEDEQ_GAIN_FMASK))
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_MIN \
|
|
|
|
(FEEDEQ_PREAMP_SIGNVAL(FEEDEQ_GAIN_MIN) * \
|
|
|
|
FEEDEQ_IF2PREAMP(FEEDEQ_GAIN_MIN, 0))
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_MAX \
|
|
|
|
(FEEDEQ_PREAMP_SIGNVAL(FEEDEQ_GAIN_MAX) * \
|
|
|
|
FEEDEQ_IF2PREAMP(FEEDEQ_GAIN_MAX, 0))
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP_DEFAULT FEEDEQ_IF2PREAMP(0, 0)
|
|
|
|
|
|
|
|
#define FEEDEQ_PREAMP2IDX(v) \
|
|
|
|
((int32_t)((FEEDEQ_GAIN_MAX * (FEEDEQ_GAIN_DIV / \
|
|
|
|
FEEDEQ_GAIN_STEP)) + (FEEDEQ_PREAMP_SIGNVAL(v) * \
|
|
|
|
FEEDEQ_PREAMP_IPART(v) * (FEEDEQ_GAIN_DIV / \
|
|
|
|
FEEDEQ_GAIN_STEP)) + (FEEDEQ_PREAMP_SIGNVAL(v) * \
|
|
|
|
(FEEDEQ_PREAMP_FPART(v) / FEEDEQ_GAIN_STEP))))
|
|
|
|
|
|
|
|
static int feeder_eq_exact_rate = 0;
|
|
|
|
|
|
|
|
#ifdef _KERNEL
|
2010-06-15 07:06:54 +00:00
|
|
|
static char feeder_eq_presets[] = FEEDER_EQ_PRESETS;
|
Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .
Summary of changes includes:
1 Volume Per-Channel (vpc). Provides private / standalone volume control
unique per-stream pcm channel without touching master volume / pcm.
Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
instead of /dev/mixer. Special "bypass" mode is enabled through
/dev/mixer which will automatically detect if the adjustment is made
through /dev/mixer and forward its request to this private volume
controller. Changes to this volume object will not interfere with
other channels.
Requirements:
- SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
require specific application modifications (preferred).
- No modifications required for using bypass mode, so applications
like mplayer or xmms should work out of the box.
Kernel hints:
- hint.pcm.%d.vpc (0 = disable vpc).
Kernel sysctls:
- hw.snd.vpc_mixer_bypass (default: 1). Enable or disable /dev/mixer
bypass mode.
- hw.snd.vpc_autoreset (default: 1). By default, closing/opening
/dev/dsp will reset the volume back to 0 db gain/attenuation.
Setting this to 0 will preserve its settings across device
closing/opening.
- hw.snd.vpc_reset (default: 0). Panic/reset button to reset all
volume settings back to 0 db.
- hw.snd.vpc_0db (default: 45). 0 db relative to linear mixer value.
2 High quality fixed-point Bandlimited SINC sampling rate converter,
based on Julius O'Smith's Digital Audio Resampling -
http://ccrma.stanford.edu/~jos/resample/. It includes a filter design
script written in awk (the clumsiest joke I've ever written)
- 100% 32bit fixed-point, 64bit accumulator.
- Possibly among the fastest (if not fastest) of its kind.
- Resampling quality is tunable, either runtime or during kernel
compilation (FEEDER_RATE_PRESETS).
- Quality can be further customized during kernel compilation by
defining FEEDER_RATE_PRESETS in /etc/make.conf.
Kernel sysctls:
- hw.snd.feeder_rate_quality.
0 - Zero-order Hold (ZOH). Fastest, bad quality.
1 - Linear Interpolation (LINEAR). Slightly slower than ZOH,
better quality but still does not eliminate aliasing.
2 - (and above) - Sinc Interpolation(SINC). Best quality. SINC
quality always start from 2 and above.
Rough quality comparisons:
- http://people.freebsd.org/~ariff/z_comparison/
3 Bit-perfect mode. Bypasses all feeder/dsp effects. Pure sound will be
directly fed into the hardware.
4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.
5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
vchans in order to make digital format pass through. It also makes
vchans more dynamic by choosing a better format/rate among all the
concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
becomes sort of optional.
6 Exclusive Stream, with special open() mode O_EXCL. This will "mute"
other concurrent vchan streams and only allow a single channel with
O_EXCL set to keep producing sound.
Other Changes:
* most feeder_* stuffs are compilable in userland. Let's not
speculate whether we should go all out for it (save that for
FreeBSD 16.0-RELEASE).
* kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
* pull out channel mixing logic out of vchan.c and create its own
feeder_mixer for world justice.
* various refactoring here and there, for good or bad.
* activation of few more OSSv4 ioctls() (see [1] above).
* opt_snd.h for possible compile time configuration:
(mostly for debugging purposes, don't try these at home)
SND_DEBUG
SND_DIAGNOSTIC
SND_FEEDER_MULTIFORMAT
SND_FEEDER_FULL_MULTIFORMAT
SND_FEEDER_RATE_HP
SND_PCM_64
SND_OLDSTEREO
Manual page updates are on the way.
Tested by: joel, Olivier SMEDTS <olivier at gid0 d org>, too many
unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
|
|
|
SYSCTL_STRING(_hw_snd, OID_AUTO, feeder_eq_presets, CTLFLAG_RD,
|
|
|
|
&feeder_eq_presets, 0, "compile-time eq presets");
|
|
|
|
|
|
|
|
TUNABLE_INT("hw.snd.feeder_eq_exact_rate", &feeder_eq_exact_rate);
|
|
|
|
SYSCTL_INT(_hw_snd, OID_AUTO, feeder_eq_exact_rate, CTLFLAG_RW,
|
|
|
|
&feeder_eq_exact_rate, 0, "force exact rate validation");
|
|
|
|
#endif
|
|
|
|
|
|
|
|
struct feed_eq_info;
|
|
|
|
|
|
|
|
typedef void (*feed_eq_t)(struct feed_eq_info *, uint8_t *, uint32_t);
|
|
|
|
|
|
|
|
struct feed_eq_tone {
|
|
|
|
intpcm_t o1[SND_CHN_MAX];
|
|
|
|
intpcm_t o2[SND_CHN_MAX];
|
|
|
|
intpcm_t i1[SND_CHN_MAX];
|
|
|
|
intpcm_t i2[SND_CHN_MAX];
|
|
|
|
int gain;
|
|
|
|
};
|
|
|
|
|
|
|
|
struct feed_eq_info {
|
|
|
|
struct feed_eq_tone treble;
|
|
|
|
struct feed_eq_tone bass;
|
|
|
|
struct feed_eq_coeff *coeff;
|
|
|
|
feed_eq_t biquad;
|
|
|
|
uint32_t channels;
|
|
|
|
uint32_t rate;
|
|
|
|
uint32_t align;
|
|
|
|
int32_t preamp;
|
|
|
|
int state;
|
|
|
|
};
|
|
|
|
|
|
|
|
#if !defined(_KERNEL) && defined(FEEDEQ_ERR_CLIP)
|
|
|
|
#define FEEDEQ_ERR_CLIP_CHECK(t, v) do { \
|
|
|
|
if ((v) < PCM_S32_MIN || (v) > PCM_S32_MAX) \
|
|
|
|
errx(1, "\n\n%s(): ["#t"] Sample clipping: %jd\n", \
|
|
|
|
__func__, (intmax_t)(v)); \
|
|
|
|
} while (0)
|
|
|
|
#else
|
|
|
|
#define FEEDEQ_ERR_CLIP_CHECK(...)
|
|
|
|
#endif
|
|
|
|
|
|
|
|
#define FEEDEQ_CLAMP(v) (((v) > PCM_S32_MAX) ? PCM_S32_MAX : \
|
|
|
|
(((v) < PCM_S32_MIN) ? PCM_S32_MIN : \
|
|
|
|
(v)))
|
|
|
|
|
|
|
|
#define FEEDEQ_DECLARE(SIGN, BIT, ENDIAN) \
|
|
|
|
static void \
|
|
|
|
feed_eq_biquad_##SIGN##BIT##ENDIAN(struct feed_eq_info *info, \
|
|
|
|
uint8_t *dst, uint32_t count) \
|
|
|
|
{ \
|
|
|
|
struct feed_eq_coeff_tone *treble, *bass; \
|
|
|
|
intpcm64_t w; \
|
|
|
|
intpcm_t v; \
|
|
|
|
uint32_t i, j; \
|
|
|
|
int32_t pmul, pshift; \
|
|
|
|
\
|
|
|
|
pmul = feed_eq_preamp[info->preamp].mul; \
|
|
|
|
pshift = feed_eq_preamp[info->preamp].shift; \
|
|
|
|
\
|
|
|
|
if (info->state == FEEDEQ_DISABLE) { \
|
|
|
|
j = count * info->channels; \
|
|
|
|
dst += j * PCM_##BIT##_BPS; \
|
|
|
|
do { \
|
|
|
|
dst -= PCM_##BIT##_BPS; \
|
|
|
|
v = _PCM_READ_##SIGN##BIT##_##ENDIAN(dst); \
|
|
|
|
v = ((intpcm64_t)pmul * v) >> pshift; \
|
|
|
|
_PCM_WRITE_##SIGN##BIT##_##ENDIAN(dst, v); \
|
|
|
|
} while (--j != 0); \
|
|
|
|
\
|
|
|
|
return; \
|
|
|
|
} \
|
|
|
|
\
|
|
|
|
treble = &(info->coeff[info->treble.gain].treble); \
|
|
|
|
bass = &(info->coeff[info->bass.gain].bass); \
|
|
|
|
\
|
|
|
|
do { \
|
|
|
|
i = 0; \
|
|
|
|
j = info->channels; \
|
|
|
|
do { \
|
|
|
|
v = _PCM_READ_##SIGN##BIT##_##ENDIAN(dst); \
|
|
|
|
v <<= 32 - BIT; \
|
|
|
|
v = ((intpcm64_t)pmul * v) >> pshift; \
|
|
|
|
\
|
|
|
|
w = (intpcm64_t)v * treble->b0; \
|
|
|
|
w += (intpcm64_t)info->treble.i1[i] * treble->b1; \
|
|
|
|
w += (intpcm64_t)info->treble.i2[i] * treble->b2; \
|
|
|
|
w -= (intpcm64_t)info->treble.o1[i] * treble->a1; \
|
|
|
|
w -= (intpcm64_t)info->treble.o2[i] * treble->a2; \
|
|
|
|
info->treble.i2[i] = info->treble.i1[i]; \
|
|
|
|
info->treble.i1[i] = v; \
|
|
|
|
info->treble.o2[i] = info->treble.o1[i]; \
|
|
|
|
w >>= FEEDEQ_COEFF_SHIFT; \
|
|
|
|
FEEDEQ_ERR_CLIP_CHECK(treble, w); \
|
|
|
|
v = FEEDEQ_CLAMP(w); \
|
|
|
|
info->treble.o1[i] = v; \
|
|
|
|
\
|
|
|
|
w = (intpcm64_t)v * bass->b0; \
|
|
|
|
w += (intpcm64_t)info->bass.i1[i] * bass->b1; \
|
|
|
|
w += (intpcm64_t)info->bass.i2[i] * bass->b2; \
|
|
|
|
w -= (intpcm64_t)info->bass.o1[i] * bass->a1; \
|
|
|
|
w -= (intpcm64_t)info->bass.o2[i] * bass->a2; \
|
|
|
|
info->bass.i2[i] = info->bass.i1[i]; \
|
|
|
|
info->bass.i1[i] = v; \
|
|
|
|
info->bass.o2[i] = info->bass.o1[i]; \
|
|
|
|
w >>= FEEDEQ_COEFF_SHIFT; \
|
|
|
|
FEEDEQ_ERR_CLIP_CHECK(bass, w); \
|
|
|
|
v = FEEDEQ_CLAMP(w); \
|
|
|
|
info->bass.o1[i] = v; \
|
|
|
|
\
|
|
|
|
v >>= 32 - BIT; \
|
|
|
|
_PCM_WRITE_##SIGN##BIT##_##ENDIAN(dst, v); \
|
|
|
|
dst += PCM_##BIT##_BPS; \
|
|
|
|
i++; \
|
|
|
|
} while (--j != 0); \
|
|
|
|
} while (--count != 0); \
|
|
|
|
}
|
|
|
|
|
|
|
|
#if BYTE_ORDER == LITTLE_ENDIAN || defined(SND_FEEDER_MULTIFORMAT)
|
|
|
|
FEEDEQ_DECLARE(S, 16, LE)
|
|
|
|
FEEDEQ_DECLARE(S, 32, LE)
|
|
|
|
#endif
|
|
|
|
#if BYTE_ORDER == BIG_ENDIAN || defined(SND_FEEDER_MULTIFORMAT)
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FEEDEQ_DECLARE(S, 16, BE)
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FEEDEQ_DECLARE(S, 32, BE)
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#endif
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#ifdef SND_FEEDER_MULTIFORMAT
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FEEDEQ_DECLARE(S, 8, NE)
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FEEDEQ_DECLARE(S, 24, LE)
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FEEDEQ_DECLARE(S, 24, BE)
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FEEDEQ_DECLARE(U, 8, NE)
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FEEDEQ_DECLARE(U, 16, LE)
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FEEDEQ_DECLARE(U, 24, LE)
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FEEDEQ_DECLARE(U, 32, LE)
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FEEDEQ_DECLARE(U, 16, BE)
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FEEDEQ_DECLARE(U, 24, BE)
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FEEDEQ_DECLARE(U, 32, BE)
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#endif
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#define FEEDEQ_ENTRY(SIGN, BIT, ENDIAN) \
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{ \
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AFMT_##SIGN##BIT##_##ENDIAN, \
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feed_eq_biquad_##SIGN##BIT##ENDIAN \
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}
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static const struct {
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uint32_t format;
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feed_eq_t biquad;
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} feed_eq_biquad_tab[] = {
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#if BYTE_ORDER == LITTLE_ENDIAN || defined(SND_FEEDER_MULTIFORMAT)
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FEEDEQ_ENTRY(S, 16, LE),
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FEEDEQ_ENTRY(S, 32, LE),
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#endif
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#if BYTE_ORDER == BIG_ENDIAN || defined(SND_FEEDER_MULTIFORMAT)
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FEEDEQ_ENTRY(S, 16, BE),
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FEEDEQ_ENTRY(S, 32, BE),
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#endif
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#ifdef SND_FEEDER_MULTIFORMAT
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FEEDEQ_ENTRY(S, 8, NE),
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FEEDEQ_ENTRY(S, 24, LE),
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FEEDEQ_ENTRY(S, 24, BE),
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FEEDEQ_ENTRY(U, 8, NE),
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FEEDEQ_ENTRY(U, 16, LE),
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FEEDEQ_ENTRY(U, 24, LE),
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FEEDEQ_ENTRY(U, 32, LE),
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FEEDEQ_ENTRY(U, 16, BE),
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FEEDEQ_ENTRY(U, 24, BE),
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FEEDEQ_ENTRY(U, 32, BE)
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#endif
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};
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#define FEEDEQ_BIQUAD_TAB_SIZE \
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((int32_t)(sizeof(feed_eq_biquad_tab) / sizeof(feed_eq_biquad_tab[0])))
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static struct feed_eq_coeff *
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feed_eq_coeff_rate(uint32_t rate)
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{
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uint32_t spd, threshold;
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int i;
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if (rate < FEEDEQ_RATE_MIN || rate > FEEDEQ_RATE_MAX)
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return (NULL);
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/*
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* Not all rates are supported. Choose the best rate that we can to
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* allow 'sloppy' conversion. Good enough for naive listeners.
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*/
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for (i = 0; i < FEEDEQ_TAB_SIZE; i++) {
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spd = feed_eq_tab[i].rate;
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threshold = spd + ((i < (FEEDEQ_TAB_SIZE - 1) &&
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feed_eq_tab[i + 1].rate > spd) ?
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((feed_eq_tab[i + 1].rate - spd) >> 1) : 0);
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if (rate == spd ||
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(feeder_eq_exact_rate == 0 && rate <= threshold))
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return (feed_eq_tab[i].coeff);
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}
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return (NULL);
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}
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int
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feeder_eq_validrate(uint32_t rate)
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{
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if (feed_eq_coeff_rate(rate) != NULL)
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return (1);
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return (0);
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}
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static void
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feed_eq_reset(struct feed_eq_info *info)
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{
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uint32_t i;
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for (i = 0; i < info->channels; i++) {
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info->treble.i1[i] = 0;
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info->treble.i2[i] = 0;
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info->treble.o1[i] = 0;
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info->treble.o2[i] = 0;
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info->bass.i1[i] = 0;
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info->bass.i2[i] = 0;
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info->bass.o1[i] = 0;
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info->bass.o2[i] = 0;
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}
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}
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static int
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feed_eq_setup(struct feed_eq_info *info)
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{
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info->coeff = feed_eq_coeff_rate(info->rate);
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if (info->coeff == NULL)
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return (EINVAL);
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feed_eq_reset(info);
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return (0);
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}
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static int
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feed_eq_init(struct pcm_feeder *f)
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{
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struct feed_eq_info *info;
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feed_eq_t biquad_op;
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int i;
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if (f->desc->in != f->desc->out)
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return (EINVAL);
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biquad_op = NULL;
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for (i = 0; i < FEEDEQ_BIQUAD_TAB_SIZE && biquad_op == NULL; i++) {
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if (AFMT_ENCODING(f->desc->in) == feed_eq_biquad_tab[i].format)
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biquad_op = feed_eq_biquad_tab[i].biquad;
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}
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if (biquad_op == NULL)
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return (EINVAL);
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info = malloc(sizeof(*info), M_DEVBUF, M_NOWAIT | M_ZERO);
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if (info == NULL)
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return (ENOMEM);
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info->channels = AFMT_CHANNEL(f->desc->in);
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info->align = info->channels * AFMT_BPS(f->desc->in);
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info->rate = FEEDEQ_RATE_MIN;
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info->treble.gain = FEEDEQ_L2GAIN(50);
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info->bass.gain = FEEDEQ_L2GAIN(50);
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info->preamp = FEEDEQ_PREAMP2IDX(FEEDEQ_PREAMP_DEFAULT);
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info->state = FEEDEQ_UNKNOWN;
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info->biquad = biquad_op;
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f->data = info;
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return (feed_eq_setup(info));
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}
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static int
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feed_eq_set(struct pcm_feeder *f, int what, int value)
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|
{
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struct feed_eq_info *info;
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info = f->data;
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switch (what) {
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case FEEDEQ_CHANNELS:
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if (value < SND_CHN_MIN || value > SND_CHN_MAX)
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|
return (EINVAL);
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info->channels = (uint32_t)value;
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info->align = info->channels * AFMT_BPS(f->desc->in);
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|
feed_eq_reset(info);
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|
break;
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|
case FEEDEQ_RATE:
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|
if (feeder_eq_validrate(value) == 0)
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|
return (EINVAL);
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|
info->rate = (uint32_t)value;
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|
|
if (info->state == FEEDEQ_UNKNOWN)
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|
info->state = FEEDEQ_ENABLE;
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|
return (feed_eq_setup(info));
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|
|
break;
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|
case FEEDEQ_TREBLE:
|
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|
case FEEDEQ_BASS:
|
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|
|
if (value < 0 || value > 100)
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|
|
return (EINVAL);
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|
if (what == FEEDEQ_TREBLE)
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|
info->treble.gain = FEEDEQ_L2GAIN(value);
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|
else
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|
info->bass.gain = FEEDEQ_L2GAIN(value);
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|
|
break;
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|
case FEEDEQ_PREAMP:
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|
if (value < FEEDEQ_PREAMP_MIN || value > FEEDEQ_PREAMP_MAX)
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|
return (EINVAL);
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|
info->preamp = FEEDEQ_PREAMP2IDX(value);
|
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|
|
break;
|
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|
|
case FEEDEQ_STATE:
|
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|
|
if (!(value == FEEDEQ_BYPASS || value == FEEDEQ_ENABLE ||
|
|
|
|
value == FEEDEQ_DISABLE))
|
|
|
|
return (EINVAL);
|
|
|
|
info->state = value;
|
|
|
|
feed_eq_reset(info);
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
return (EINVAL);
|
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|
|
break;
|
|
|
|
}
|
|
|
|
|
|
|
|
return (0);
|
|
|
|
}
|
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|
|
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|
|
static int
|
|
|
|
feed_eq_free(struct pcm_feeder *f)
|
|
|
|
{
|
|
|
|
struct feed_eq_info *info;
|
|
|
|
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|
|
info = f->data;
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|
|
if (info != NULL)
|
|
|
|
free(info, M_DEVBUF);
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|
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|
|
f->data = NULL;
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|
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|
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|
|
return (0);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
feed_eq_feed(struct pcm_feeder *f, struct pcm_channel *c, uint8_t *b,
|
|
|
|
uint32_t count, void *source)
|
|
|
|
{
|
|
|
|
struct feed_eq_info *info;
|
|
|
|
uint32_t j;
|
|
|
|
uint8_t *dst;
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|
|
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|
|
info = f->data;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* 3 major states:
|
|
|
|
* FEEDEQ_BYPASS - Bypass entirely, nothing happened.
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|
|
* FEEDEQ_ENABLE - Preamp+biquad filtering.
|
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|
|
* FEEDEQ_DISABLE - Preamp only.
|
|
|
|
*/
|
|
|
|
if (info->state == FEEDEQ_BYPASS)
|
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|
|
return (FEEDER_FEED(f->source, c, b, count, source));
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|
dst = b;
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|
|
count = SND_FXROUND(count, info->align);
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|
|
|
|
|
|
do {
|
|
|
|
if (count < info->align)
|
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|
|
break;
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|
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|
j = SND_FXDIV(FEEDER_FEED(f->source, c, dst, count, source),
|
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|
|
info->align);
|
|
|
|
if (j == 0)
|
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|
|
break;
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|
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|
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|
|
info->biquad(info, dst, j);
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|
|
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|
|
j *= info->align;
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|
|
dst += j;
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|
|
count -= j;
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|
|
|
|
|
|
} while (count != 0);
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|
|
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|
|
return (dst - b);
|
|
|
|
}
|
|
|
|
|
|
|
|
static struct pcm_feederdesc feeder_eq_desc[] = {
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|
|
{ FEEDER_EQ, 0, 0, 0, 0 },
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|
|
{ 0, 0, 0, 0, 0 }
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|
|
};
|
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|
|
static kobj_method_t feeder_eq_methods[] = {
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|
|
KOBJMETHOD(feeder_init, feed_eq_init),
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|
|
KOBJMETHOD(feeder_free, feed_eq_free),
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|
|
KOBJMETHOD(feeder_set, feed_eq_set),
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|
|
KOBJMETHOD(feeder_feed, feed_eq_feed),
|
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|
|
KOBJMETHOD_END
|
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|
|
};
|
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|
|
|
|
|
|
FEEDER_DECLARE(feeder_eq, NULL);
|
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|
|
|
|
|
|
static int32_t
|
|
|
|
feed_eq_scan_preamp_arg(const char *s)
|
|
|
|
{
|
|
|
|
int r, i, f;
|
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|
|
size_t len;
|
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|
|
char buf[32];
|
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|
|
|
|
|
|
bzero(buf, sizeof(buf));
|
|
|
|
|
|
|
|
/* XXX kind of ugly, but works for now.. */
|
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|
|
|
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|
|
r = sscanf(s, "%d.%d", &i, &f);
|
|
|
|
|
|
|
|
if (r == 1 && !(i < FEEDEQ_PREAMP_IMIN || i > FEEDEQ_PREAMP_IMAX)) {
|
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|
|
snprintf(buf, sizeof(buf), "%c%d",
|
|
|
|
FEEDEQ_PREAMP_SIGNMARK(i), abs(i));
|
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|
|
f = 0;
|
|
|
|
} else if (r == 2 &&
|
|
|
|
!(i < FEEDEQ_PREAMP_IMIN || i > FEEDEQ_PREAMP_IMAX ||
|
|
|
|
f < FEEDEQ_PREAMP_FMIN || f > FEEDEQ_PREAMP_FMAX))
|
|
|
|
snprintf(buf, sizeof(buf), "%c%d.%d",
|
|
|
|
FEEDEQ_PREAMP_SIGNMARK(i), abs(i), f);
|
|
|
|
else
|
|
|
|
return (FEEDEQ_PREAMP_INVALID);
|
|
|
|
|
|
|
|
len = strlen(s);
|
|
|
|
if (len > 2 && strcasecmp(s + len - 2, "dB") == 0)
|
|
|
|
strlcat(buf, "dB", sizeof(buf));
|
|
|
|
|
|
|
|
if (i == 0 && *s == '-')
|
|
|
|
*buf = '-';
|
|
|
|
|
|
|
|
if (strcasecmp(buf + ((*s >= '0' && *s <= '9') ? 1 : 0), s) != 0)
|
|
|
|
return (FEEDEQ_PREAMP_INVALID);
|
|
|
|
|
|
|
|
while ((f / FEEDEQ_GAIN_DIV) > 0)
|
|
|
|
f /= FEEDEQ_GAIN_DIV;
|
|
|
|
|
|
|
|
return (((i < 0 || *buf == '-') ? -1 : 1) * FEEDEQ_IF2PREAMP(i, f));
|
|
|
|
}
|
|
|
|
|
|
|
|
#ifdef _KERNEL
|
|
|
|
static int
|
|
|
|
sysctl_dev_pcm_eq(SYSCTL_HANDLER_ARGS)
|
|
|
|
{
|
|
|
|
struct snddev_info *d;
|
|
|
|
struct pcm_channel *c;
|
|
|
|
struct pcm_feeder *f;
|
|
|
|
int err, val, oval;
|
|
|
|
|
|
|
|
d = oidp->oid_arg1;
|
|
|
|
if (!PCM_REGISTERED(d))
|
|
|
|
return (ENODEV);
|
|
|
|
|
|
|
|
PCM_LOCK(d);
|
|
|
|
PCM_WAIT(d);
|
|
|
|
if (d->flags & SD_F_EQ_BYPASSED)
|
|
|
|
val = 2;
|
|
|
|
else if (d->flags & SD_F_EQ_ENABLED)
|
|
|
|
val = 1;
|
|
|
|
else
|
|
|
|
val = 0;
|
|
|
|
PCM_ACQUIRE(d);
|
|
|
|
PCM_UNLOCK(d);
|
|
|
|
|
|
|
|
oval = val;
|
|
|
|
err = sysctl_handle_int(oidp, &val, 0, req);
|
|
|
|
|
|
|
|
if (err == 0 && req->newptr != NULL && val != oval) {
|
|
|
|
if (!(val == 0 || val == 1 || val == 2)) {
|
|
|
|
PCM_RELEASE_QUICK(d);
|
|
|
|
return (EINVAL);
|
|
|
|
}
|
|
|
|
|
|
|
|
PCM_LOCK(d);
|
|
|
|
|
|
|
|
d->flags &= ~(SD_F_EQ_ENABLED | SD_F_EQ_BYPASSED);
|
|
|
|
if (val == 2) {
|
|
|
|
val = FEEDEQ_BYPASS;
|
|
|
|
d->flags |= SD_F_EQ_BYPASSED;
|
|
|
|
} else if (val == 1) {
|
|
|
|
val = FEEDEQ_ENABLE;
|
|
|
|
d->flags |= SD_F_EQ_ENABLED;
|
|
|
|
} else
|
|
|
|
val = FEEDEQ_DISABLE;
|
|
|
|
|
|
|
|
CHN_FOREACH(c, d, channels.pcm.busy) {
|
|
|
|
CHN_LOCK(c);
|
|
|
|
f = chn_findfeeder(c, FEEDER_EQ);
|
|
|
|
if (f != NULL)
|
|
|
|
(void)FEEDER_SET(f, FEEDEQ_STATE, val);
|
|
|
|
CHN_UNLOCK(c);
|
|
|
|
}
|
|
|
|
|
|
|
|
PCM_RELEASE(d);
|
|
|
|
PCM_UNLOCK(d);
|
|
|
|
} else
|
|
|
|
PCM_RELEASE_QUICK(d);
|
|
|
|
|
|
|
|
return (err);
|
|
|
|
}
|
|
|
|
|
|
|
|
static int
|
|
|
|
sysctl_dev_pcm_eq_preamp(SYSCTL_HANDLER_ARGS)
|
|
|
|
{
|
|
|
|
struct snddev_info *d;
|
|
|
|
struct pcm_channel *c;
|
|
|
|
struct pcm_feeder *f;
|
|
|
|
int err, val, oval;
|
|
|
|
char buf[32];
|
|
|
|
|
|
|
|
d = oidp->oid_arg1;
|
|
|
|
if (!PCM_REGISTERED(d))
|
|
|
|
return (ENODEV);
|
|
|
|
|
|
|
|
PCM_LOCK(d);
|
|
|
|
PCM_WAIT(d);
|
|
|
|
val = d->eqpreamp;
|
|
|
|
bzero(buf, sizeof(buf));
|
|
|
|
(void)snprintf(buf, sizeof(buf), "%c%d.%ddB",
|
|
|
|
FEEDEQ_PREAMP_SIGNMARK(val), FEEDEQ_PREAMP_IPART(val),
|
|
|
|
FEEDEQ_PREAMP_FPART(val));
|
|
|
|
PCM_ACQUIRE(d);
|
|
|
|
PCM_UNLOCK(d);
|
|
|
|
|
|
|
|
oval = val;
|
|
|
|
err = sysctl_handle_string(oidp, buf, sizeof(buf), req);
|
|
|
|
|
|
|
|
if (err == 0 && req->newptr != NULL) {
|
|
|
|
val = feed_eq_scan_preamp_arg(buf);
|
|
|
|
if (val == FEEDEQ_PREAMP_INVALID) {
|
|
|
|
PCM_RELEASE_QUICK(d);
|
|
|
|
return (EINVAL);
|
|
|
|
}
|
|
|
|
|
|
|
|
PCM_LOCK(d);
|
|
|
|
|
|
|
|
if (val != oval) {
|
|
|
|
if (val < FEEDEQ_PREAMP_MIN)
|
|
|
|
val = FEEDEQ_PREAMP_MIN;
|
|
|
|
else if (val > FEEDEQ_PREAMP_MAX)
|
|
|
|
val = FEEDEQ_PREAMP_MAX;
|
|
|
|
|
|
|
|
d->eqpreamp = val;
|
|
|
|
|
|
|
|
CHN_FOREACH(c, d, channels.pcm.busy) {
|
|
|
|
CHN_LOCK(c);
|
|
|
|
f = chn_findfeeder(c, FEEDER_EQ);
|
|
|
|
if (f != NULL)
|
|
|
|
(void)FEEDER_SET(f, FEEDEQ_PREAMP, val);
|
|
|
|
CHN_UNLOCK(c);
|
|
|
|
}
|
|
|
|
|
|
|
|
}
|
|
|
|
|
|
|
|
PCM_RELEASE(d);
|
|
|
|
PCM_UNLOCK(d);
|
|
|
|
} else
|
|
|
|
PCM_RELEASE_QUICK(d);
|
|
|
|
|
|
|
|
return (err);
|
|
|
|
}
|
|
|
|
|
|
|
|
void
|
|
|
|
feeder_eq_initsys(device_t dev)
|
|
|
|
{
|
|
|
|
struct snddev_info *d;
|
|
|
|
const char *preamp;
|
|
|
|
char buf[64];
|
|
|
|
|
|
|
|
d = device_get_softc(dev);
|
|
|
|
|
|
|
|
if (!(resource_string_value(device_get_name(dev), device_get_unit(dev),
|
|
|
|
"eq_preamp", &preamp) == 0 &&
|
|
|
|
(d->eqpreamp = feed_eq_scan_preamp_arg(preamp)) !=
|
|
|
|
FEEDEQ_PREAMP_INVALID))
|
|
|
|
d->eqpreamp = FEEDEQ_PREAMP_DEFAULT;
|
|
|
|
|
|
|
|
if (d->eqpreamp < FEEDEQ_PREAMP_MIN)
|
|
|
|
d->eqpreamp = FEEDEQ_PREAMP_MIN;
|
|
|
|
else if (d->eqpreamp > FEEDEQ_PREAMP_MAX)
|
|
|
|
d->eqpreamp = FEEDEQ_PREAMP_MAX;
|
|
|
|
|
|
|
|
SYSCTL_ADD_PROC(device_get_sysctl_ctx(dev),
|
|
|
|
SYSCTL_CHILDREN(device_get_sysctl_tree(dev)), OID_AUTO,
|
|
|
|
"eq", CTLTYPE_INT | CTLFLAG_RW, d, sizeof(d),
|
|
|
|
sysctl_dev_pcm_eq, "I",
|
|
|
|
"Bass/Treble Equalizer (0=disable, 1=enable, 2=bypass)");
|
|
|
|
|
|
|
|
bzero(buf, sizeof(buf));
|
|
|
|
|
|
|
|
(void)snprintf(buf, sizeof(buf), "Bass/Treble Equalizer Preamp "
|
|
|
|
"(-/+ %d.0dB , %d.%ddB step)",
|
|
|
|
FEEDEQ_GAIN_MAX, FEEDEQ_GAIN_STEP / FEEDEQ_GAIN_DIV,
|
|
|
|
FEEDEQ_GAIN_STEP - ((FEEDEQ_GAIN_STEP / FEEDEQ_GAIN_DIV) *
|
|
|
|
FEEDEQ_GAIN_DIV));
|
|
|
|
|
|
|
|
SYSCTL_ADD_PROC(device_get_sysctl_ctx(dev),
|
|
|
|
SYSCTL_CHILDREN(device_get_sysctl_tree(dev)), OID_AUTO,
|
|
|
|
"eq_preamp", CTLTYPE_STRING | CTLFLAG_RW, d, sizeof(d),
|
|
|
|
sysctl_dev_pcm_eq_preamp, "A", buf);
|
|
|
|
}
|
|
|
|
#endif
|