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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.
199 lines
4.7 KiB
C
199 lines
4.7 KiB
C
/*-
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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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#ifdef HAVE_KERNEL_OPTION_HEADERS
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#include "opt_snd.h"
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#endif
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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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} while (0)
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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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