2007-05-31 18:35:24 +00:00
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/*-
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2017-11-27 14:52:40 +00:00
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* SPDX-License-Identifier: BSD-2-Clause-FreeBSD
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*
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2007-05-31 18:35:24 +00:00
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* Copyright (c) 2007 Ariff Abdullah <ariff@FreeBSD.org>
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*
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* $FreeBSD$
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*/
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#include <sys/param.h>
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#include <sys/systm.h>
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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
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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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2007-05-31 18:35:24 +00:00
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#include <dev/sound/unit.h>
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/*
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* Unit magic allocator for sound driver.
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*
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* 'u' = Unit of attached soundcards
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* 'd' = Device type
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* 'c' = Channel number
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*
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* eg: dsp0.p1 - u=0, d=p, c=1
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* dsp1.vp0 - u=1, d=vp, c=0
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* dsp0.10 - u=0, d=clone, c=allocated clone (see further explanation)
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*
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* Maximum unit of soundcards can be tuned through "hw.snd.maxunit", which
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* is between SND_UNIT_UMIN (16) and SND_UNIT_UMAX (2048). By design,
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* maximum allowable allocated channel is 256, with exception for clone
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* devices which doesn't have any notion of channel numbering. The use of
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* channel numbering in a clone device is simply to provide uniqueness among
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* allocated clones. This also means that the maximum allowable clonable
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* device is largely dependant and dynamically tuned depending on
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* hw.snd.maxunit.
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*/
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/* Default width */
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static int snd_u_shift = 9; /* 0 - 0x1ff : 512 distinct soundcards */
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static int snd_d_shift = 5; /* 0 - 0x1f : 32 distinct device types */
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static int snd_c_shift = 10; /* 0 - 0x3ff : 1024 distinct channels
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(256 limit "by design",
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except for clone devices) */
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static int snd_unit_initialized = 0;
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#ifdef SND_DIAGNOSTIC
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#define SND_UNIT_ASSERT() do { \
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if (snd_unit_initialized == 0) \
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panic("%s(): Uninitialized sound unit!", __func__); \
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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
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} while (0)
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2007-05-31 18:35:24 +00:00
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#else
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#define SND_UNIT_ASSERT() KASSERT(snd_unit_initialized != 0, \
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("%s(): Uninitialized sound unit!", \
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__func__))
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#endif
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#define MKMASK(x) ((1 << snd_##x##_shift) - 1)
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int
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snd_max_u(void)
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{
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SND_UNIT_ASSERT();
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return (MKMASK(u));
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}
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int
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snd_max_d(void)
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{
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SND_UNIT_ASSERT();
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return (MKMASK(d));
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}
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int
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snd_max_c(void)
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{
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SND_UNIT_ASSERT();
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return (MKMASK(c));
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}
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int
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snd_unit2u(int unit)
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{
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SND_UNIT_ASSERT();
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return ((unit >> (snd_c_shift + snd_d_shift)) & MKMASK(u));
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}
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int
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snd_unit2d(int unit)
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{
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SND_UNIT_ASSERT();
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return ((unit >> snd_c_shift) & MKMASK(d));
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}
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int
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snd_unit2c(int unit)
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{
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SND_UNIT_ASSERT();
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return (unit & MKMASK(c));
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}
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int
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snd_u2unit(int u)
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{
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SND_UNIT_ASSERT();
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return ((u & MKMASK(u)) << (snd_c_shift + snd_d_shift));
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}
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int
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snd_d2unit(int d)
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{
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SND_UNIT_ASSERT();
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return ((d & MKMASK(d)) << snd_c_shift);
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}
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int
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snd_c2unit(int c)
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{
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SND_UNIT_ASSERT();
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return (c & MKMASK(c));
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}
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int
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snd_mkunit(int u, int d, int c)
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{
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SND_UNIT_ASSERT();
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return ((c & MKMASK(c)) | ((d & MKMASK(d)) << snd_c_shift) |
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((u & MKMASK(u)) << (snd_c_shift + snd_d_shift)));
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}
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/*
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* This *must* be called first before any of the functions above!!!
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*/
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void
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snd_unit_init(void)
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{
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int i;
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if (snd_unit_initialized != 0)
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return;
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snd_unit_initialized = 1;
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if (getenv_int("hw.snd.maxunit", &i) != 0) {
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if (i < SND_UNIT_UMIN)
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i = SND_UNIT_UMIN;
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else if (i > SND_UNIT_UMAX)
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i = SND_UNIT_UMAX;
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else
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i = roundup2(i, 2);
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for (snd_u_shift = 0; (i >> (snd_u_shift + 1)) != 0;
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snd_u_shift++)
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;
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/*
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* Make room for channels/clones allocation unit
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* to fit within 24bit MAXMINOR limit.
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*/
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snd_c_shift = 24 - snd_u_shift - snd_d_shift;
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}
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if (bootverbose != 0)
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printf("%s() u=0x%08x [%d] d=0x%08x [%d] c=0x%08x [%d]\n",
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__func__, SND_U_MASK, snd_max_u() + 1,
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SND_D_MASK, snd_max_d() + 1, SND_C_MASK, snd_max_c() + 1);
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}
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