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.

1 files changed, 16 insertions, 12 deletions

diff --git a/sys/dev/sound/pci/aureal.c b/sys/dev/sound/pci/aureal.c
index 6b2c6442aab9..37e1c4584723 100644
--- a/sys/dev/sound/pci/aureal.c
+++ b/sys/dev/sound/pci/aureal.c
@@ -24,6 +24,10 @@
  * 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/pci/aureal.h>
@@ -38,19 +42,19 @@ SND_DECLARE_FILE("$FreeBSD$");
 
 /* channel interface */
 static u_int32_t au_playfmt[] = {
-	AFMT_U8,
-	AFMT_STEREO | AFMT_U8,
-	AFMT_S16_LE,
-	AFMT_STEREO | AFMT_S16_LE,
+	SND_FORMAT(AFMT_U8, 1, 0),
+	SND_FORMAT(AFMT_U8, 2, 0),
+	SND_FORMAT(AFMT_S16_LE, 1, 0),
+	SND_FORMAT(AFMT_S16_LE, 2, 0),
 	0
 };
 static struct pcmchan_caps au_playcaps = {4000, 48000, au_playfmt, 0};
 
 static u_int32_t au_recfmt[] = {
-	AFMT_U8,
-	AFMT_STEREO | AFMT_U8,
-	AFMT_S16_LE,
-	AFMT_STEREO | AFMT_S16_LE,
+	SND_FORMAT(AFMT_U8, 1, 0),
+	SND_FORMAT(AFMT_U8, 2, 0),
+	SND_FORMAT(AFMT_S16_LE, 1, 0),
+	SND_FORMAT(AFMT_S16_LE, 2, 0),
 	0
 };
 static struct pcmchan_caps au_reccaps = {4000, 48000, au_recfmt, 0};
@@ -167,7 +171,7 @@ au_wrcd(kobj_t obj, void *arg, int regno, u_int32_t data)
 static kobj_method_t au_ac97_methods[] = {
     	KOBJMETHOD(ac97_read,		au_rdcd),
     	KOBJMETHOD(ac97_write,		au_wrcd),
-	{ 0, 0 }
+	KOBJMETHOD_END
 };
 AC97_DECLARE(au_ac97);
 
@@ -242,13 +246,13 @@ static void
 au_prepareoutput(struct au_chinfo *ch, u_int32_t format)
 {
 	struct au_info *au = ch->parent;
-	int i, stereo = (format & AFMT_STEREO)? 1 : 0;
+	int i, stereo = (AFMT_CHANNEL(format) > 1)? 1 : 0;
 	u_int32_t baseaddr = sndbuf_getbufaddr(ch->buffer);
 
 	au_wr(au, 0, 0x1061c, 0, 4);
 	au_wr(au, 0, 0x10620, 0, 4);
 	au_wr(au, 0, 0x10624, 0, 4);
-	switch(format & ~AFMT_STEREO) {
+	switch(AFMT_ENCODING(format)) {
 		case 1:
 			i=0xb000;
 			break;
@@ -382,7 +386,7 @@ static kobj_method_t auchan_methods[] = {
     	KOBJMETHOD(channel_trigger,		auchan_trigger),
     	KOBJMETHOD(channel_getptr,		auchan_getptr),
     	KOBJMETHOD(channel_getcaps,		auchan_getcaps),
-	{ 0, 0 }
+	KOBJMETHOD_END
 };
 CHANNEL_DECLARE(auchan);