Commit Graph

19 Commits

Author SHA1 Message Date
Ariff Abdullah
a580b31a54 Welcome to Once-a-year Sound Mega-Commit. Enjoy numerous updates and fixes
in every sense.

General
-------

- Multichannel safe, endian safe, format safe
   * Large part of critical pcm filters such as vchan.c, feeder_rate.c,
     feeder_volume.c, feeder_fmt.c and feeder.c has been rewritten so that
     using them does not cause the pcm data to be converted to 16bit little
     endian.
   * Macrosses for accessing pcm data safely are defined within sound.h in
     the form of PCM_READ_* / PCM_WRITE_*
   * Currently, most of them are probably limited for mono/stereo handling,
     but the future addition of true multichannel will be much easier.

- Low latency operation
  * Well, this require lot more works to do not just within sound driver,
    but we're heading towards right direction. Buffer/block sizing within
    channel.c is rewritten to calculate precise allocation for various
    combination of sample/data/rate size. As a result, applying correct
    SNDCTL_DSP_POLICY value will achive expected latency behaviour simmilar
    to what commercial 4front driver do.
  * Signal handling fix. ctrl+c of "cat /dev/zero > /dev/dsp" does not
    result long delay.
  * Eliminate sound truncation if the sound data is too small.
    DIY:
      1) Download / extract
         http://people.freebsd.org/~ariff/lowlatency/shortfiles.tar.gz
      2) Do a comparison between "cat state*.au > /dev/dsp" and
         "for x in state*.au ; do cat $x > /dev/dsp ; done"
         - there should be no "perceivable" differences.
    Double close for PR kern/31445.

  CAVEAT: Low latency come with (unbearable) price especially for poorly
          written applications. Applications that trying to act smarter
	  by requesting (wrong) blocksize/blockcount will suffer the most.
	  Fixup samples/patches can be found at:
	  http://people.freebsd.org/~ariff/ports/

- Switch minimum/maximum sampling rate limit to "1" and "2016000" (48k * 42)
  due to closer compatibility with 4front driver.
  Discussed with: marcus@ (long time ago?)

- All driver specific sysctls in the form of "hw.snd.pcm%d.*" have been
  moved to their own dev sysctl nodes, notably:
  hw.snd.pcm%d.vchans -> dev.pcm.%d.vchans
  Bump __FreeBSD_version.

Driver specific
---------------

- Ditto for sysctls.

- snd_atiixp, snd_es137x, snd_via8233, snd_hda
  * Numerous cleanups and fixes.
  * _EXPERIMENTAL_ polling mode support using simple callout_* mechanisme.
   This was intended for pure debugging and latency measurement, but proven
   good enough in few unexpected and rare cases (such as problematic shared
   IRQ with GIANT devices - USB). Polling can be enabled/disabled through
   dev.pcm.0.polling. Disabled by default.

- snd_ich
  * Fix possible overflow during speed calibration. Delay final
    initialization (pcm_setstatus) after calibration finished.
    PR: kern/100169
    Tested by: Kevin Overman <oberman@es.net>
  * Inverted EAPD for few Nec VersaPro.
    PR: kern/104715
    Submitted by: KAWATA Masahiko <kawata@mta.biglobe.ne.jp>

Thanks to various people, notably Joel Dahl, Yuriy Tsibizov, Kevin Oberman,
those at #freebsd-azalia @ freenode and others for testing.

Joel Dahl will do the manpage update.
2006-11-26 12:24:06 +00:00
Alexander Leidinger
851a904af5 - Rename hw.snd.unit to hw.snd.default_unit to make the purpose more obvious.
- Enable 4 automatic vchan's by default.
- Add some comments which provide ides/questions for improvement.
- Prefix some temporary sysctl's with an underscore to denote that it is not
  an official API but a workaround until the real solution is implemented.
2006-06-18 14:14:41 +00:00
Joel Dahl
da8623eca0 Fix typos and clean up some comments.
Approved by:	ariff
2006-01-25 21:13:46 +00:00
Ariff Abdullah
14665331ab channel.c:
(1) Fix DMA alignment, based on bytes per sample.

feeder_rate.c:
	Handle strayed bytes (mostly caused by #1) better.

This DMA alignment issues are extremely hard to reproduce unless
the user happen to have a 32bit capable soundcards (ATI IXP) and
knowledgeable enough to force it to operate under pure 32bit
operations on both record and play directions.
2006-01-24 01:10:07 +00:00
Ariff Abdullah
d9bd844573 Various fixups:
feeder.h:
 feeder.c:
	- Implement scoring mechanisme to select best format for conversion.
	  This is actually part of newer format chaining procedures which
	  will be commited someday. Confusion during chaining process solved
	  by this scoring since it will try to reduce list of from/to formats
	  to a single, best format.
	  Related PR:	kern/91683
channel.c:
	- Simplify feeder building process since we have smarter format
	  chaining.

feeder_fmt.c:
	- Add few more sign conversion feeders for 24 and 32 bit format.

feeder_rate.c:
	- Force buffer / bytes allignment. Unaligned buffer may cause
	  panics during recording on pure 32bit sample format if it
	  involves feeder_rate as part of feeders chain.
	  Tested on: ATI IXP, force 32bit recording.

MFC after:	5 days
2006-01-22 15:06:49 +00:00
Ariff Abdullah
720289b2cd Update my email address, so people know where the exact /
proper / correct place to bug me.

Approved by:	netchild (mentor)
2005-11-14 18:37:59 +00:00
Ariff Abdullah
3f3c2c43b0 Added missing comma. This fixes compilation if we need to enable
RATE_ASSERT debug macro.

Approved by:	netchild (mentor)
2005-10-18 21:18:47 +00:00
Alexander Leidinger
87506547d2 Whats New:
1. Support wide range sampling rate, as low as 1hz up to int32 max
   (which is, insane) through new feeder_rate, multiple precisions
   choice (32/64 bit converter). This is indeed, quite insane, but it
   does give us more room and flexibility. Plenty sysctl options to
   adjust resampling characteristics.
2. Support 24/32 bit pcm format conversion through new, much improved,
   simplified and optimized feeder_fmt.

Changes:
1. buffer.c / dsp.c / sound.h
   * Support for 24/32 AFMT.
2. feeder_rate.c
   * New implementation of sampling rate conversion with 32/64 bit
     precision, 1 - int32max hz (which is, ridiculous, yet very
     addictive).  Much improved / smarter buffer management to not
     cause any missing samples at the end of conversion process
   * Tunable sysctls for various aspect:
       hw.snd.feeder_rate_ratemin - minimum allowable sampling rate
       (default to 4000)
       hw.snd.feeder_rate_ratemax - maximum allowable sampling rate
       (default to 1102500)
       hw.snd.feeder_rate_buffersize - conversion buffer size
       (default to 8192)
       hw.snd.feeder_rate_scaling - scaling / conversion method
       (please refer to the source for explaination). Default to
       previous implementation type.
3. feeder_fmt.c / sound.h
   * New implementation, support for 24/32bit conversion, optimized,
     and simplified. Few routines has been removed (8 to xlaw, 16 to
     8). It just doesn't make sense.
4. channel.c
   * Support for 24/32 AFMT
   * Fix wrong xruns increment, causing incorrect underruns statistic
     while using vchans.
5. vchan.c
   * Support for 24/32 AFMT
   * Proper speed / rate detection especially for fixed rate ac97.
     User can override it using kernel hint:
     hint.pcm.<unit>.vchanrate="xxxx".

Notes / Issues:
        * Virtual Channels (vchans)
          Enabling vchans can really, really help to solve overrun
          issues.  This is quite understandable, because it operates
          entirely within its own buffering system without relying on
          hardware interrupt / state. Even if you don't need vchan,
          just enable single channel can help much. Few soundcards
          (notably via8233x, sblive, possibly others) have their own
          hardware multi channel, and this is unfortunately beyond
          vchan reachability.
        * The arrival of 24/32 also come with a price. Applications
          that can do 24/32bit playback need to be recompiled (notably
          mplayer).  Use (recompiled) mplayer to experiment / test /
          debug this various format using -af format=fmt. Note that
          24bit seeking in mplayer is a little bit broken, sometimes
          can cause silence or loud static noise. Pausing / seeking
          few times can solve this problem.
          You don't have to rebuild world entirely for this. Simply
          copy /usr/src/sys/sys/soundcard.h to
          /usr/include/sys/soundcard.h would suffice. Few drivers also
          need recompilation, and this can be done via
          /usr/src/sys/modules/sound/.
          Support for 24bit hardware playback is beyond the scope of
          this changes. That would require spessific hardware driver
          changes.
        * Don't expect playing 9999999999hz is a wise decision. Be
          reasonable. The new feeder_rate implemention provide
          flexibility, not insanity. You can easily chew up your CPU
          with this kind of mind instability. Please use proper
          mosquito repellent device for this obvious cracked brain
          attempt. As for testing purposes, you can use (again)
          mplayer to generate / play with different sampling rate. Use
          something like "mplayer -af resample=192000:0:0 <files>".

Submitted by:	Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by:	multimedia@
2005-07-31 16:16:22 +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
Orion Hodson
6635978f23 Additional malloc failure checks. 2003-04-20 17:08:56 +00:00
Orion Hodson
a7576e2e4b Back out last commit, which is fine in theory, but ignores the fact
that a lock is held whilst the allocations are made (M_WAITOK -> M_NOWAIT).
2003-03-05 14:48:28 +00:00
Brian Feldman
3fbe138ca9 It seems that sound(4)'s feeder routines don't need to allocate memory
without waiting, since they are called from a system-call context only.
This appears to fix all sorts of problems with open("/dev/dsp", O_WRONLY)
randomly returning ENXIO.

Found by:	cognet
2003-02-23 20:49:45 +00:00
Orion Hodson
63679b6573 Fix comment typo.
Sync with userland test framework which now deals better with pcm feeder kobj
emulation.

Reduce max rate from 96kHz to 48kHz as userland tests found a few bad
points about 90kHz and we don't care about operating up there for now.
2003-02-06 17:32:02 +00:00
Orion Hodson
456922d5f2 o Constrain inputs to 25Hz granularity so interpolator can operate
between any pair of values in range 4-96kHz.  Thanks to Ken Marks for
discovering there were problems with the previous version.

o Use a non-recursive gcd routine.
2003-01-30 16:32:56 +00:00
Orion Hodson
4a532ff091 Re-implemention of the interpolation code used for sample rate
conversion.  The new version has improved interpolation accuracy and
maintains the timing relationship between the input and output signals
exactly.

Approved by:	cg
2003-01-20 00:54:24 +00:00
Cameron Grant
67beb5a5c8 various fixes to eliminate locking warnings
Approved by:	re
Reviewed by:	orion
2002-11-25 17:17:43 +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
Cameron Grant
4dce85407c add a new method for retrieving feeder parameters 2001-05-27 14:49:14 +00:00
Cameron Grant
60391e107d add a software sample rate conversion feeder. this uses linear
interpolation for reasonable quality whilst not using too much cpu time.
2001-04-08 20:26:22 +00:00