Commit Graph

43 Commits

Author SHA1 Message Date
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
Ariff Abdullah
082f63835c Flush remaining malloc() cleanups (M_NOWAIT -> M_WAITOK). 2007-06-17 06:10:43 +00:00
Ariff Abdullah
bdfbdcec6a Filter/compress the amount of channel trigger. This should reduce
much of lock/unlock contentions within the interrupt handler. Most
of these drivers only need PCMTRIG_START or STOP (ABORT).

Discussed with:		scottl
2007-06-11 00:49:46 +00:00
Ariff Abdullah
79462204f1 Fix broken binary issues with latest gcc 4.x due to bitfield signess
mishaps for emu10k1 [1] and few other places.

Reported/Submitted/Tested by:	Ed Schouten <ed@fxq.nl> [1]
2007-05-27 20:12:51 +00:00
Ariff Abdullah
2e334adf6a sndbuf_alloc() now accept dmaflags argument which will be forwarded to
internal bus_dmammem_alloc() for greater flexibility on setting up DMA /
page attributes.
2007-04-18 18:26:41 +00:00
Alexander Leidinger
0b989078d7 MFp4 (114068):
Use bus_get_dma_tag() to obtain the parent DMA tag to make the drivers
        a little bit more non-ia32/amd64 friendly.

        There is no man page for bus_get_dma_tag, so this is modelled after
        rev. 1.62 of src/sys/dev/sound/pci/es137x.c by marius.

        Inspired by:	commit by marius
2007-02-23 13:47:34 +00:00
Joel Dahl
a0afd24d9c Clean up the BSD license to match the preferred license in
/usr/share/examples/etc/bsd-style-copyright.  I've fixed a
few minor wording and formatting differences.

Approved by:	luigi, Hannu Savolainen <hannu@opensound.com>
2007-02-02 13:39:20 +00:00
Ariff Abdullah
d8f1a170d9 Fix broken capabilites. There are possible calculation errors within
ess_calcspeed8() and ess_calcspeed9() that need to be fixed as well
(TODO).

Reported by:	[1] Claude Buisson <cbuisson at nerim.net>
MFC after:	3 days

[1] http://lists.freebsd.org/pipermail/freebsd-multimedia/2006-January/003566.html
2006-01-16 20:01:33 +00:00
Alexander Leidinger
7a7689dea4 - Fixup the locking.
- Don't mark MPSAFE (yet).

Submitted by:	Ariff Abdullah <skywizard@MyBSD.org.my>
2005-07-31 13:51:04 +00:00
Warner Losh
098ca2bda9 Start each of the license/copyright comments with /*-, minor shuffle of lines 2005-01-06 01:43:34 +00:00
Pyun YongHyeon
eba1cb6e3e Audio drivers failed to detect failure condition and attempted to
assign DMA address to the wrong address. It can cause system lockup
or other mysterious errors. Since most sound cards requires low DMA
address(BUS_SPACE_MAXADDR_24BIT) sndbuf_alloc() would fail when the
audio driver is loaded after long running of operations.

Approved by:	jake (mentor)
Reviewed by:	truckman, matk
2004-10-13 05:45:16 +00:00
Nate Lawson
4a25d7ffe2 * Remove the acpi attachment from the es1888. It has an identify method
that conjures up the device node so it isn't true PNP.  Noticed by jhb@.

* Add an attachment for esscontrol since it too uses ISA_PNP_PROBE.

* Move an attachment from snd_mss to snd_pnpmss.  The latter is the real
  PNP user.
2004-10-12 01:56:03 +00:00
Seigo Tanimura
0739ea1de2 Rename the sound device drivers:
- `sound'
  The generic sound driver, always required.

- `snd_*'
  Device-dependent drivers, named after the sound module names.
  Configure accordingly to your hardware.

In addition, rename the `snd_pcm' module to `sound' in order to sync
with the driver names.

Suggested by:	cg
2004-07-16 04:00:08 +00:00
Brian Feldman
8fb9a995cf The newpcm headers currently #define away INTR_MPSAFE and INTR_TYPE_AV
because they bogusly check for defined(INTR_MPSAFE) -- something which
never was a #define.  Correct the definitions.

This make INTR_TYPE_AV finally get used instead of the lower-priority
INTR_TYPE_TTY, so it's quite possible some improvement will be had
on sound driver performance.  It would also make all the drivers
marked INTR_MPSAFE actually run without Giant (which does seem to
work for me), but:
	INTR_MPSAFE HAS BEEN REMOVED FROM EVERY SOUND DRIVER!
It needs to be re-added on a case-by-case basis since there is no one
who will vouch for which sound drivers, if any, willy actually operate
correctly without Giant, since there hasn't been testing because of
this bug disabling INTR_MPSAFE.

Found by:	"Yuriy Tsibizov" <Yuriy.Tsibizov@gfk.ru>
2004-04-14 14:57:49 +00:00
Nate Lawson
5f96beb9e0 Convert callers to the new bus_alloc_resource_any(9) API.
Submitted by:	Mark Santcroos <marks@ripe.net>
Reviewed by:	imp, dfr, bde
2004-03-17 17:50:55 +00:00
Mathew Kanner
0d8ed52ea5 Augment /dev/sndstat with the module names, if applicable.
Approved by:	  tanimura (mentor)
2004-03-06 15:52:42 +00:00
Cameron Grant
3f22597838 update my email address. 2003-09-07 16:28:03 +00:00
Scott Long
f6b1c44d1f Mega busdma API commit.
Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma.  At the moment, this is used for the
asynchronous busdma_swi and callback mechanism.  Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg.  dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create().  The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.

sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms.  The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.

If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.

Reviewed by:	tmm, gibbs
2003-07-01 15:52:06 +00:00
Yoshihiro Takahashi
3febcc57ec - Clean up ISA DMA supports.
- Rename all sndbuf_isadma* functions to sndbuf_dma* and move them into
  sys/dev/sound/isa/sndbuf_dma.c.

No response from:	sound
2003-02-07 14:05:34 +00:00
Semen Ustimenko
83190e29d3 Do not return(foo()) in void function.
Submitted by:	marius@alchemy.franken.de
MFC after:	3 days
2002-12-18 22:53:24 +00:00
Scott Long
436c9b651a Fix code that had rotted behind debugging macros.
Approved by:	cg (in principle)
MFC after:	2 weeks
2002-01-25 04:14:12 +00:00
Cameron Grant
374e1c5baa allow the hardware buffer size to be controlled with hints
release isa dma channels on unload (ad1816, ess, sb8)
2001-09-29 07:57:07 +00:00
Cameron Grant
67b1dce3bc many changes:
* add new channels to the end of the list so channels used in order of
addition

* de-globalise definition of struct snddev_info and provide accessor
functions where necessary.

* move the $FreeBSD$ tag in each .c file into a macro and allow the
/dev/sndstat handler to display these when set to maximum verbosity to aid
debugging.

* allow each device to register its own sndstat handler to reduce the amount
of groping sndstat must do in foreign structs.
2001-08-23 11:30:52 +00:00
George C A Reid
733a4ea771 Use the M_ZERO flag to malloc(9)
Reviewed by:	cg
MFC after:	1 week
2001-06-21 19:45:59 +00:00
Cameron Grant
d95502a838 use a global devclass for all drivers - i'm not entirely sure why this
worked before.

mixer, dsp and sndstat are seperate devices - give them their own cdevsws
instead of demuxing requests sent to a single cdevsw.

use the si_drv1/si_drv2 fields in dev_t structures for holding information
specific to an open instance of mixer/dsp.

nuke /dev/{dsp,dspW,audio}[0-9]* links - this functionality is now provided
using cloning.

various locking fixes.
2001-06-16 21:25:10 +00:00
George C A Reid
9de0de1dd4 Reinitialise the DSP and mixer after a resume from suspend
PR:		22372
Submitted by:	Hiroyuki Aizu <aizu@jaist.ac.jp>
Reviewed by:	cg
2001-04-08 23:02:06 +00:00
Cameron Grant
66ef8af5b0 mega-commit.
this introduces a new buffering mechanism which results in dramatic
simplification of the channel manager.

as several structures have changed, we take the opportunity to move their
definitions into the source files where they are used, make them private and
de-typedef them.

the sound drivers are updated to use snd_setup_intr instead of
bus_setup_intr, and to comply with the de-typedefed structures.

the ac97, mixer and channel layers have been updated with finegrained
locking, as have some drivers- not all though.  the rest will follow soon.
2001-03-24 23:10:29 +00:00
Cameron Grant
1b0dabf0c0 Add speaker volume adjusting support
Submitted by:	Tai-hwa Liang <avatar@mmlab.cse.yzu.edu.tw>
PR:		i386/21452
2001-02-27 12:44:31 +00:00
Cameron Grant
d60a6a8e0c quieten the esscontrol device 2001-02-02 16:41:06 +00:00
Cameron Grant
826737698c change irq handler slightly, get rid of superflous messages 2000-12-27 04:04:36 +00:00
Cameron Grant
350a5fafb1 update code dealing with snd_dbuf objects to do so using a functional interface
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use.  it will aim for a power of two size small
enough to generate block sizes of at most 20ms.  it will also set the
hard-blocksize taking into account rate/format conversions in use.

update drivers to implement setblocksize  correctly:
updated, tested: 	sb16, emu10k1, maestro, solo
updated, untested: 	ad1816, ess, mss, sb8, csa
not updated: 		ds1, es137x, fm801, neomagic, t4dwave, via82c686

i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
2000-12-23 03:16:13 +00:00
Cameron Grant
0f55ac6c1a kobjify.
this gives us several benefits, including:

* easier extensibility- new optional methods can be added to
  ac97/mixer/channel classes without having to fixup every driver.

* forward compatibility for drivers, provided no new mandatory methods are
  added.
2000-12-18 01:36:41 +00:00
Cameron Grant
9ec437a334 add reinit functions to mixers
unstaticize chn_start()
add reset/resetdone functions to channels
2000-10-26 20:46:58 +00:00
Cameron Grant
306f91b60b detach support
remove un-needed setdir functions
add bus_teardown_intr calls where necessary
destroy our dma tags where necessary
destroy ac97 before releasing resources
2000-09-09 19:21:04 +00:00
Cameron Grant
33dbf14a17 change mixer api slightly
change channel interface - kobj implementation coming soonish
make pcm_makelinks not panic if modular
add pcm_unregister()

these changes support newpcm kld unloading, but this is only implemented
by ds1.c
2000-09-01 20:09:24 +00:00
Cameron Grant
513693be6c rework feeder sytem to allow feeders in klds
modify driver capability reporting format to list every audio format
seperately- required for above and because we could not previously indicate
that mono was unsupported.

there should be no functional impact.
2000-08-20 22:18:56 +00:00
Cameron Grant
f314f3dad2 add module metadata. this is a hack, sound drivers will eventually present a
bus to which pcm, mixer, etc will attach.
2000-07-03 20:52:27 +00:00
Peter Wemm
cb73359d42 Unused include: #include "sbc.h" 2000-06-10 07:17:29 +00:00
Cameron Grant
35f9e4a1db handle emulated dma reads
don't try to get sample size from snd_dbuf
2000-05-26 21:15:47 +00:00
Cameron Grant
942aeab734 fix a speed bug that nobody noticed 2000-05-15 02:10:27 +00:00
Cameron Grant
674c45bd04 make drivers start at beginning of buffer when triggered - improves mmap.
not all tested.

not sure about aureal.c or csapcm.c
2000-04-17 16:57:12 +00:00
Cameron Grant
0a3eb835c2 bump the buffer size from 4k to 16k. should improve performance under load. 2000-04-01 21:28:09 +00:00
Cameron Grant
57376530f0 split up ess and sb code
rewrite ess mixer to use native registers
rewrite play/rec code to use more accurate timer when available
add code to use audio2 for playback, but disable it as no irqs are generated
2000-03-28 18:31:01 +00:00