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mirror of https://git.FreeBSD.org/src.git synced 2024-12-18 10:35:55 +00:00
freebsd/sys/dev/sound/unit.c
Ariff Abdullah 90da2b2859 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

199 lines
4.7 KiB
C

/*-
* Copyright (c) 2007 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.
*
* $FreeBSD$
*/
#include <sys/param.h>
#include <sys/systm.h>
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/unit.h>
/*
* Unit magic allocator for sound driver.
*
* 'u' = Unit of attached soundcards
* 'd' = Device type
* 'c' = Channel number
*
* eg: dsp0.p1 - u=0, d=p, c=1
* dsp1.vp0 - u=1, d=vp, c=0
* dsp0.10 - u=0, d=clone, c=allocated clone (see further explanation)
*
* Maximum unit of soundcards can be tuned through "hw.snd.maxunit", which
* is between SND_UNIT_UMIN (16) and SND_UNIT_UMAX (2048). By design,
* maximum allowable allocated channel is 256, with exception for clone
* devices which doesn't have any notion of channel numbering. The use of
* channel numbering in a clone device is simply to provide uniqueness among
* allocated clones. This also means that the maximum allowable clonable
* device is largely dependant and dynamically tuned depending on
* hw.snd.maxunit.
*/
/* Default width */
static int snd_u_shift = 9; /* 0 - 0x1ff : 512 distinct soundcards */
static int snd_d_shift = 5; /* 0 - 0x1f : 32 distinct device types */
static int snd_c_shift = 10; /* 0 - 0x3ff : 1024 distinct channels
(256 limit "by design",
except for clone devices) */
static int snd_unit_initialized = 0;
#ifdef SND_DIAGNOSTIC
#define SND_UNIT_ASSERT() do { \
if (snd_unit_initialized == 0) \
panic("%s(): Uninitialized sound unit!", __func__); \
} while (0)
#else
#define SND_UNIT_ASSERT() KASSERT(snd_unit_initialized != 0, \
("%s(): Uninitialized sound unit!", \
__func__))
#endif
#define MKMASK(x) ((1 << snd_##x##_shift) - 1)
int
snd_max_u(void)
{
SND_UNIT_ASSERT();
return (MKMASK(u));
}
int
snd_max_d(void)
{
SND_UNIT_ASSERT();
return (MKMASK(d));
}
int
snd_max_c(void)
{
SND_UNIT_ASSERT();
return (MKMASK(c));
}
int
snd_unit2u(int unit)
{
SND_UNIT_ASSERT();
return ((unit >> (snd_c_shift + snd_d_shift)) & MKMASK(u));
}
int
snd_unit2d(int unit)
{
SND_UNIT_ASSERT();
return ((unit >> snd_c_shift) & MKMASK(d));
}
int
snd_unit2c(int unit)
{
SND_UNIT_ASSERT();
return (unit & MKMASK(c));
}
int
snd_u2unit(int u)
{
SND_UNIT_ASSERT();
return ((u & MKMASK(u)) << (snd_c_shift + snd_d_shift));
}
int
snd_d2unit(int d)
{
SND_UNIT_ASSERT();
return ((d & MKMASK(d)) << snd_c_shift);
}
int
snd_c2unit(int c)
{
SND_UNIT_ASSERT();
return (c & MKMASK(c));
}
int
snd_mkunit(int u, int d, int c)
{
SND_UNIT_ASSERT();
return ((c & MKMASK(c)) | ((d & MKMASK(d)) << snd_c_shift) |
((u & MKMASK(u)) << (snd_c_shift + snd_d_shift)));
}
/*
* This *must* be called first before any of the functions above!!!
*/
void
snd_unit_init(void)
{
int i;
if (snd_unit_initialized != 0)
return;
snd_unit_initialized = 1;
if (getenv_int("hw.snd.maxunit", &i) != 0) {
if (i < SND_UNIT_UMIN)
i = SND_UNIT_UMIN;
else if (i > SND_UNIT_UMAX)
i = SND_UNIT_UMAX;
else
i = roundup2(i, 2);
for (snd_u_shift = 0; (i >> (snd_u_shift + 1)) != 0;
snd_u_shift++)
;
/*
* Make room for channels/clones allocation unit
* to fit within 24bit MAXMINOR limit.
*/
snd_c_shift = 24 - snd_u_shift - snd_d_shift;
}
if (bootverbose != 0)
printf("%s() u=0x%08x [%d] d=0x%08x [%d] c=0x%08x [%d]\n",
__func__, SND_U_MASK, snd_max_u() + 1,
SND_D_MASK, snd_max_d() + 1, SND_C_MASK, snd_max_c() + 1);
}