freebsd-dev/sys/dev/sound/pcm/ac97_patch.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

118 lines
3.4 KiB
C

/*-
* Copyright (c) 2002 Orion Hodson
* 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.
*/
#ifdef HAVE_KERNEL_OPTION_HEADERS
#include "opt_snd.h"
#endif
#include <dev/sound/pcm/sound.h>
#include <dev/sound/pcm/ac97.h>
#include <dev/sound/pcm/ac97_patch.h>
SND_DECLARE_FILE("$FreeBSD$");
void ad1886_patch(struct ac97_info* codec)
{
#define AC97_AD_JACK_SPDIF 0x72
/*
* Presario700 workaround
* for Jack Sense/SPDIF Register misetting causing
* no audible output
* by Santiago Nullo 04/05/2002
*/
ac97_wrcd(codec, AC97_AD_JACK_SPDIF, 0x0010);
}
void ad198x_patch(struct ac97_info* codec)
{
switch (ac97_getsubvendor(codec)) {
case 0x11931043: /* Not for ASUS A9T (probably else too). */
break;
default:
ac97_wrcd(codec, 0x76, ac97_rdcd(codec, 0x76) | 0x0420);
break;
}
}
void ad1981b_patch(struct ac97_info* codec)
{
/*
* Enable headphone jack sensing.
*/
switch (ac97_getsubvendor(codec)) {
case 0x02d91014: /* IBM Thinkcentre */
case 0x099c103c: /* HP nx6110 */
ac97_wrcd(codec, AC97_AD_JACK_SPDIF,
ac97_rdcd(codec, AC97_AD_JACK_SPDIF) | 0x0800);
break;
default:
break;
}
}
void cmi9739_patch(struct ac97_info* codec)
{
/*
* Few laptops need extra register initialization
* to power up the internal speakers.
*/
switch (ac97_getsubvendor(codec)) {
case 0x18431043: /* ASUS W1000N */
ac97_wrcd(codec, AC97_REG_POWER, 0x000f);
ac97_wrcd(codec, AC97_MIXEXT_CLFE, 0x0000);
ac97_wrcd(codec, 0x64, 0x7110);
break;
default:
break;
}
}
void alc655_patch(struct ac97_info* codec)
{
/*
* MSI (Micro-Star International) specific EAPD quirk.
*/
switch (ac97_getsubvendor(codec)) {
case 0x00611462: /* MSI S250 */
case 0x01311462: /* MSI S270 */
case 0x01611462: /* LG K1 Express */
case 0x03511462: /* MSI L725 */
ac97_wrcd(codec, 0x7a, ac97_rdcd(codec, 0x7a) & 0xfffd);
break;
case 0x10ca1734:
/*
* Amilo Pro V2055 with ALC655 has phone out by default
* disabled (surround on), leaving us only with internal
* speakers. This should really go to mixer. We write the
* Data Flow Control reg.
*/
ac97_wrcd(codec, 0x6a, ac97_rdcd(codec, 0x6a) | 0x0001);
break;
default:
break;
}
}