hardware imposes strict limitations on hard buffer and block sizes.
Previous code set soft buffer to be no smaller then hard buffer. On some
cards with fixed 64K physical buffer that caused up to 800ms play latency.
New code allows to set soft buffer size down to just two blocks of the hard
buffer and to not write more then that size ahead to the hardware buffer.
As result of that change I was able to reduce full practically measured
record-playback loop delay in those conditions down to only about 115ms
with theoretical playback latency of only about 50ms.
New code works fine for both vchans and direct cases. In both cases sound(4)
tries to follow hw.snd.latency_profile and hw.snd.latency values and
application-requested buffer and block sizes as much as limitation of two
hardware blocks allows.
Reviewed by: silence on multimedia@
also does this for sound drivers it's probably not necessary for all
combinations of controllers and drivers. However, given that our sound
drivers completely lack bus_dmamap_sync(9) calls this at least serves
as a workaround when enabling use of the IOMMU streaming buffers on
sparc64 and generally for arm and mips.
MFC after: 2 weeks
by zero of the second argument 'from'.
- Prefer u_int32_t over unsigned int to make its intention more clearer.
- Move the function to a header file and make it a static inline function.
Pointed out by: Andrew Reilly (areilly at bigpond dot net dot au)[1]
MFC after: 3 days
On top of that, LLVM+Clang mis-compiles this code because of its register
allocator bug.
Analyzed by: Andrew Reilly (areilly at bigpond dot net dot au)
Reviewed by: ariff, rdivacky
MFC after: 3 days
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.
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...
- Rework the entire pcm_channel structure:
* Remove rarely used link placeholder, instead, make each pcm_channel
as head/link of each own/each other. Unlock - Lock sequence due to
sleep malloc has been reduced.
* Implement "busy" queue which will contain list of busy/active
channels. This greatly reduce locking contention for example while
servicing interrupt for hardware with many channels or when virtual
channels reach its 256 peak channels.
- So I heard you like v chan ... O RLY?
Welcome to Virtual **Record** Channels (vrec, rec vchans, vchans for
recording, Rec-Chan, you decide), the ultimate solutions for your
nagging O_RDWR full-duplex wannabe (note: flash plugins) monopolizing
single record channel causing EBUSY. Vrec works exactly like Vchans
(or, should I rename it to "Vplay" :) , except that it operates on the
opposite direction (recording). Up to 256 vrecs (like vchans) are
possible.
Notes:
* Relocate dev.pcm.%d.{vchans,vchanformat,vchanrate} to each of its
respective node/direction:
dev.pcm.%d.play.* for "play" (cdev = dsp%d.vp%d)
dev.pcm.%d.rec.* for "record" (cdev = dsp%d.vr%d)
* Don't expect that it will magically give you ability to split
"recording source" (eg: 1 channel for cdrom, 1 channel for mic,
etc). Just admit that you only have a *single* recording source /
channel. Please bug your hardware vendor instead :)
- Bump maxautovchans from 4 to 16. For a full-fledged multimedia
desktop/workstation with too many soundservers installed (esound,
artsd, jackd, pulse/polypaudio, ding-dong pling plong mudkip fuh fuh,
etc), 4 seems inadequate. There will be no memory penalty here, since
virtual channels are allocate only by demand.
- Nuke/Rework the entire statically created cdev entries. Everything is
clonable through snd own clone manager which designed to withstand many
kind of abusive devfs droids such as:
* while : ; do /bin/test -e /dev/dsp ; done
* jot 16777216 0 | while read x ; do ls /dev/dsp0.$x ; done
* hundreds (could be thousands) concurrent threads/process opening
"/dev/dsp" (previously, this might result EBUSY even with just
3 contesting threads/procs).
o Reusable clone objects (instead of creating new one like there's no
tomorrow) after certain expiration deadline. The clone allocator will
decide whether to reuse, share, or creating new clone.
o Automatic garbage collector.
- Dynamic unit magic allocator. Maximum attached soundcards can be tuned
using tunable "hw.snd.maxunit" (Default to 512). Minimum is 16, and
maximum is 2048.
- ..other fixes, mostly related to concurrency issues.
joel@ will do the manpage updates on sound(4).
Have fun.
buffer resizing, etc.) that was here since eon. Free all (unmanaged)
allocated buffer through sndbuf_destroy() in case we forgot to call
sndbuf_free(). For a managed buffer (mostly hw specific managed buffer),
either provide CHANNEL_FREE() method with appropriate return value to
invoke semi-automatic sndbuf_free() or simply do it on their own. If
everything is failed, sndbuf_destroy() will come to the rescue as a
final measure.
MFC after: 3 days
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.
The goal was to sync with the OSSv4 API 4Front Technologies uses in their
proprietary OSS driver. This was successful as far as possible. The part
of the API which is stable is implemented, for the rest there are some
stubs already.
New system ioctls:
- SNDCTL_SYSINFO - obtain audio system info (version, # of audio/midi/
mixer devices, etc.)
- SNDCTL_AUDIOINFO - fetch details about a specific audio device
- SNDCTL_MIXERINFO - fetch details about a specific mixer device
New audio ioctls:
- Sync groups (SNDCTL_DSP_SYNCGROUP/SNDCTL_DSP_SYNCSTART) which allow
triggered playback/recording on multiple devices (even across processes
simultaneously).
- Peak meters (SNDCTL_DSP_GETIPEAKS/SNDCTL_DSP_GETOPEAKS) - can query
audio drivers for peak levels (needs driver support, disabled for now).
- Per channel playback/recording levels -
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL. Note that these are still in name
only, just wrapping around the AC97-style mixer at the moment. The next
step is to push them down to the drivers.
Audio ioctls still under development by 4Front (for which stubs may exist
in this commit):
- SNDCTL_GETNAME, SNDCTL_{GET,SET}{SONG,LABEL}
- SNDCTL_DSP_{GET,SET}_CHNORDER
- SNDCTL_MIX_ENUMINFO, SNDCTL_MIX_EXTINFO - (might be documented enough in
the OSS releases to work on this. These ioctls cover the cool "twiddle
any knob on your card" features.)
Missing:
- SNDCTL_DSP_COOKEDMODE -- this ioctl is used to give applications direct
access to a card's buffers, bypassing the feeder architecture. It's
a toughy -- "someone" needs to decide :
(a) if this is desireable, and (b) if it's reasonably feasible.
Updates for driver writers:
So far, only two routines to the channel class (in channel_if.m) are added.
One is for fetching a list of discrete supported playback/recording rates
of a channel, and the other is for fetching peak level info (useful for
drawing peak meters). Interested parties may want to help pushing down
SNDCTL_DSP_{GET,SET}{PLAY,REC}VOL into the drivers.
To use the new stuff you need to rebuild the sound drivers or your kernel
(depending on if you use modules or not) and to install soundcard.h (a
buildworld/installworld handles this).
Sponsored by: Google SoC 2006
Submitted by: ryanb
Many thanks to: 4Front Technologies for their cooperation, explanations
and the nice license of their soundcard.h.
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@
panic() so that the buffer overflow just beyond this point is always
caught, even when the code is not compiled with INVARIANTS.
Change chn_setblocksize() buffer reallocation code to attempt to avoid
the feed_vchan16() buffer overflow by attempting to always keep the
bufsoft buffer at least as large as the bufhard buffer.
Print a diagnositic message
Danger! %s bufsoft size increasing from %d to %d after CHANNEL_SETBLOCKSIZE()
if our best attempts fail. If feed_vchan16() were to be called by
the interrupt handler while locks are dropped in chn_setblocksize()
to increase the size bufsoft to match the size of bufhard, the panic()
code in feed_vchan16() will be triggered. If the diagnostic message
is printed, it is a warning that a panic is possible if the system
were to see events in an "unlucky" order.
Change the locking code to avoid the need for MTX_RECURSIVE mutexes.
Add the MTX_DUPOK option to the channel mutexes and change the locking
sequence to always lock the parent channel before its children to avoid
the possibility of deadlock.
Actually implement locking assertions for the channel mutexes and fix
the problems found by the resulting assertion violations.
Clean up the locking code in dsp_ioctl().
Allocate the channel buffers using the malloc() M_WAITOK option instead
of M_NOWAIT so that buffer allocation won't fail. Drop locks across
the malloc() calls.
Add/modify KASSERTS() in attempt to detect problems early.
Abuse layering by adding a pointer to the snd_dbuf structure that points
back to the pcm_channel that owns it. This allows sndbuf_resize() to do
proper locking without having to change the its API, which is used by
the hardware drivers.
Don't dereference a NULL pointer when setting hw.snd.maxautovchans
if a hardware driver is not loaded. Noticed by Ryan Sommers
<ryans at gamersimpact.com>.
Tested by: Stefan Ehmann <shoesoft AT gmx.net>
Tested by: matk (Mathew Kanner)
Tested by: Gordon Bergling <gbergling AT 0xfce3.net>
* be more specific in verbose boot messages
* allow the feeder subsystem to veto pcm* attaching if there is an error
initialising the root feeder
* don't free/malloc a new tmpbuf when resizing a snd_dbuf to the same size as
it currently is
* store the feeder description in the feeder structure instead of mallocing
space for it
* 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.
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.
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.
