Commit Graph

30 Commits

Author SHA1 Message Date
Andriy Gapon
a9bdb5d3ad sound/pcm: use non-const string as a value with SYSCTL_STRING
Although the sysctls are marked with CTLFLAG_RD and the values will stay
immutable, current sysctl implementation stores value pointer in
void* type, which means that const qualifier is discarded anyway
and some newer compilers complaint about that.
We can't use de-const trick in sysctl implementation, because in that
case we could miss an opposite situation where a const value is used
with CTLFLAG_RW sysctl.

Complaint from:	gcc 4.4, clang
MFC after:	2 weeks
2010-06-15 07:06:54 +00:00
Ariff Abdullah
df41a638f7 - Do aggresive saturation on various polynomial interpolators.
This dramatically pushing 99.9% interpolations and quantizations
  error _below_ -180dB on 32bit dynamic range, resulting extremely
  high quality conversion.
- Use BSPLINE interpolator for filter oversampling factor greater or
  equal than 64 (log2 6).

Approved by:	re (kib)
2009-07-14 18:53:34 +00:00
Ariff Abdullah
5c663ce9cf Rearrange shift operation to increase interpolation accuracy,
further reducing conversion artifacts and better worst case SNR.

Approved by:	re (kib)
2009-07-09 22:21:18 +00:00
Ariff Abdullah
96831ae59b - Increase dynamic range of filter coefficients from 28bit to 30bit.
This cause dramatic effect in overall precision and conversion quality
  by pushing down most aliasing artifacts around -180 dB.

  Spectrogram analysis/comparison:

  	http://people.freebsd.org/~ariff/z_comparison/z_28vs30/

- Guard against possible 64bit overflow during accumulation process by
  slightly normalize and saturate sample and coefficient multiplication,
  possible during extreme 32bit downsampling (eg. 380KHz -> 8KHz) with
  custom preset that require more than ~7000 taps filter (which is
  overkill).

- Add knobs through FEEDER_RATE_PRESETS to set dynamic range of filter
  coefficients/accumulator and prefered polynomial interpolator:

  	COEFFICIENT_BIT:X
	(where 1 <= X <= 30, default: 30)

	ACCUMULATOR_BIT:X
	(where 32 <= X <=64, default: 58)

	INTERPOLATOR:I
	(where I = ZOH, LINEAR, QUADRATIC, HERMITE, BSPLINE,
 	           OPT32X, OPT16X, OPT8X, OPT4X, OPT2X)

Approved by:	re (kib)
2009-07-05 18:15:06 +00:00
Ariff Abdullah
f3bf5def20 Slight comment fix. 2009-06-24 02:01:16 +00:00
Ariff Abdullah
eedb75f16b Remap type of polynomial interpolators for better polyphase
coefficients quality:
- Linear interpolator for oversampling factor larger and equal
  than 4096 (log2 = 12).
- Quadratic interpolator for oversampling factor larger and equal
  than 256 (log2 = 8).

Default oversampling factor (128 ~ log2 = 7) will use OPT32X, which
provides better accuracy.
2009-06-15 04:05:38 +00:00
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
e4e61333ff Last (again ?!?) major commit for RELENG_7, featuring total Giant
eradication in/from userland path, countless locking fixes, etc.

- General sleep call through msleep(9) has been converted to condvar(9)
  with better consistencies.
- Heavily guard every possible "slow path" entries (open(), close(),
  few ioctl()s, sysctls), but once it entering "fast path" (io, interrupt
  started), they are free to fly on their own.
- Rearrange locking sequences, resulting better concurrency and
  serialization. Large part doesn't even need locking at all, and will be
  removed in future. Less clutter, except in few places due to lock
  ordering.
- Anonymous mixer object creation/deletion to simplify mixer handling
  beyond typical mixer ioctls.
  Submitted by:		chibis (with modifications)
- Add few mix_[get|set|..] functions to avoid calling mixer_ioctl()
  directly using cryptic arguments.
- Locking fixes to avoid possible deadlock with (still under Giant) USB.
- Better simplex/duplex device handling.
- Recover mmap() functionality for recording, which has been lost
  since 2.2.x - 3.x (the introduction of newpcm). Full-duplex mmap still
  doesn't work (due to VM/page design), but people still can mmap
  both by opening each direction separately. mmaped playback is guarantee
  to work either way.
- New sysctl: "hw.snd.compat_linux_mmap" to allow PROT_EXEC page
  mapping, due to recent changes in linux compatibility layer which
  require it. All linux applications that using sound + mmap() (mostly games)
  require this to be enabled. Disabled by default.
- Other goodies.. too many, that will increase releng7 shareholder value
  and make users of releng6 (and below) cry ;)

* This commit should be atomic. If anything goes wrong (not counting problem
  originated from elsewhere), I will not hesitate to revert everything back
  within 12 hours. This substantial changes itself not a rocket science
  and the process has begun for almost 2 years, and lots of incremental
  changes are already in place during that period of time.
* Some issues does occur in snd_emu10kx (note the 'x') due to various
  internal locking issues and it is currently being worked on by chibis.

Tested by:	chibis (Yuriy Tsibizov), joel, Alexandre Vieira,
          	many innocent souls...
2007-06-16 03:37:28 +00:00
David Malone
041b706b2f Despite several examples in the kernel, the third argument of
sysctl_handle_int is not sizeof the int type you want to export.
The type must always be an int or an unsigned int.

Remove the instances where a sizeof(variable) is passed to stop
people accidently cut and pasting these examples.

In a few places this was sysctl_handle_int was being used on 64 bit
types, which would truncate the value to be exported.  In these
cases use sysctl_handle_quad to export them and change the format
to Q so that sysctl(1) can still print them.
2007-06-04 18:25:08 +00:00
Joel Dahl
e9577a5cd9 Separate license from comments.
Approved by:	ariff
2007-06-02 13:07:44 +00:00
Ariff Abdullah
1324d98beb [stage: 6/9]
- Disable stray buffer management, since sample size aligned buffering
  are pretty much guaranteed through out the entire feeder_* chain
  processes.
- Few style(9) cleanups.
2007-03-16 17:16:56 +00:00
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