and remove non-present symbols that are now reported by kmod_syms.awk.
Reviewed by: emaste
Sponsored by: The FreeBSD Foundation
MFC after: 1 week
Differential revision: https://reviews.freebsd.org/D32878
play the MIDI files through /dev/sequencer device with tools like
playmidi. The audio output will go through the external MIDI device
such like wavetable synthesis card.
Reviewed by: matk (a long time ago), kib
Approved by: re (kib)
Tested with: Terratec SiXPack 5.1+ + Yamaha DB50XG
MFC after: 4 weeks
problems than it solves. SYSDIR is already defined almost always and
can be used instead. Working around the one case where it isn't is
much easier than working around the fact that @ may not exist in 18
other places.
Differential Revision: https://reviews.freebsd.org/D1100
options into kern.opts.mk and change all the places where we use
src.opts.mk to pull in the options. Conditionally define SYSDIR and
use SYSDIR/conf/kern.opts.mk instead of a CURDIR path. Replace all
instances of CURDIR/../../etc with STSDIR, but only in the affected
files.
As a special compatibility hack, include bsd.owm.mk at the top of
kern.opts.mk to allow the bare build of sys/modules to work on older
systems. If the defaults ever change between 9.x, 10.x and current for
these options, however, you'll wind up with the host OS' defaults
rather than the -current defaults. This hack will be removed when
we no longer need to support this build scenario.
Reviewed by: jhb
Differential Revision: https://phabric.freebsd.org/D529
This includes:
o All directories named *ia64*
o All files named *ia64*
o All ia64-specific code guarded by __ia64__
o All ia64-specific makefile logic
o Mention of ia64 in comments and documentation
This excludes:
o Everything under contrib/
o Everything under crypto/
o sys/xen/interface
o sys/sys/elf_common.h
Discussed at: BSDcan
audio device driver is detached first and not its children. This fixes
a panic in some cases when unloading "snd_uaudio" while a USB device
is plugged. The linking order affects the order in which the module
dependencies are registered.
MFC after: 1 week
platforms, because these platforms do not implement the ISA DMA
API. Else the sound modules cannot be loaded when running these
platforms.
MFC after: 2 weeks
kernel modules that include binary-only code.
More fine-grained control is provided via MK_SOURCELESS_HOST (for native code
that runs on host CPU) and MK_SOURCELESS_UCODE (for microcode).
Reviewed by: julian, delphij, freebsd-arch
Approved by: kib (mentor)
MFC after: 2 weeks
versions derived from /usr/ports/audio/oss.
The particular headers used were taken from the
attic/drv/oss_allegro directory and are mostly identical
to the previous files.
The Maestro3 driver is now free from the GPL.
NOTE: due to lack of testers this driver is being
considered for deprecation and removal.
PR: kern/153920
Approved by: jhb (mentor)
MFC after: 2 weeks
- Huge old hdac driver was split into three independent pieces: HDA
controller driver (hdac), HDA CODEC driver (hdacc) and HDA sudio function
driver (hdaa).
- Support for multichannel recording was added. Now, as specification
defines, driver checks input associations for pins with sequence numbers
14 and 15, and if found (usually) -- works as before, mixing signals
together. If it doesn't, it configures input association as multichannel.
- Signal tracer was improved to look for cases where several DACs/ADCs in
CODEC can work with the same audio signal. If such case found, driver
registers additional playback/record stream (channel) for the pcm device.
- New controller streams reservation mechanism was implemented. That
allows to have more pcm devices then streams supported by the controller
(usually 4 in each direction). Now it limits only number of simultaneously
transferred audio streams, that is rarely reachable and properly reported
if happens.
- Codec pins and GPIO signals configuration was exported via set of
writable sysctls. Another sysctl dev.hdaa.X.reconfig allows to trigger
driver reconfiguration in run-time.
- Driver now decodes pins location and connector type names. In some cases
it allows to hint user where on the system case connectors, related to the
pcm device, are located. Number of channels supported by pcm device,
reported now (if it is not 2), should also make search easier.
- Added workaround for digital mic on some Asus laptops/netbooks.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
This uses the emuxkireg.h already used in the emu10k1
snd driver. Special thanks go to Alexander Motin as
he was able to find some errors and reverse engineer
some wrong values in the emuxkireg header.
The emu10kx driver is now free from the GPL.
PR: 153901
Tested by: mav, joel
Approved by: jhb (mentor)
MFC after: 2 weeks
This brings in the emuxkireg.h from NetBSD (dev/pci) which
is used for the same purpose but is smaller. The emu10k1
is now free from the GPL.
PR: 153901
Obtained from: NetBSD
Approved by: core (mentor implicit)
MFC after: 2 weeks
Use MACHINE_CPUARCH in preference to MACHINE_ARCH. The former is the
source code location of the machine, the latter the binary output. In
general, we want to use MACHINE_CPUARCH instead of MACHINE_ARCH unless
we're tesitng for a specific target. The isn't even moot for
i386/amd64 where there's momemntum towards a MACHINE_CPUARCH == x86,
although a specific cleanup for that likely would be needed...
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.
module. These files cause manual interaction when building
ports/audio/aureal-kmod which provides a usable i386-only driver (it requires
linking against some linux object files distributed by vendor which bankrupted
back in 2000).
MFC after: 1 week
which are also likely to be irrelevant for sun4v (there's no SBus on sun4v
and only some EBus devices). While at it fix some style bugs according to
style.Makefile(5) where appropriate.
MFC after: 3 days
other changes too).
(without any real order)
1. Use device_get_nameunit for mutex naming
2. Add timer for low-latency playback
3. Move most mixer controls from sysctls to mixer(8) controls.
This is a largest part of this patch.
4. Add analog/digital switch (as a temporary sysctl)
5. Get back support for low-bitrate playback (with help of (2))
6. Change locking for exclusive I/O. Writing to non-PTR register
is almost safe and does not need to be ordered with PTR operations.
7. Disable MIDI until we get it to detach properly and fix memory
managment problems.
8. Enable multichannel playback by default. It is as stable as
single-channel mode. Multichannel recording is still an
experimental feature.
9. Multichannel options can be changed by loader tunables.
10. Add a way to disable card from a loader tunable.
11. Add new PCI IDs.
12. Debugger settings are loader tunables now.
14. Remove some unused variables.
15. Mark pcm sub-devices MPSAFE.
16. Partially revert (bus_setup_intr -> snd_setup_intr) since it need
to be done independently
Submitted by: Yuriy Tsibizov (driver maintainer)
Approved by: re (bmah)
- 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.
recording enabled some programs (audio/audacity from ports) can't
correctly enumerate all /dev/dsp device.
Note: previous commit did not enable some debugging stuff, my eyes did
misread "#undef" as "#define".
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
---snip---
New features:
1. Optional multichannel recording (32 channels on Live!, 64 channels
on Audigy).
All channels are 16bit/48000Hz/mono, format is fixed.
Half of them are copied from sound output, another half can be
used to record any data from DSP. What should be recorded is
hardcoded in DSP code. In this version it records dummy data, but
can be used to record all DSP inputs, for example..
Because there are no support of more-than-stereo sound streams
multichannell stream is presented as one 32(64)*48000 Hz 16bit mono
stream.
Channel map:
SB Live! (4.0/5.1)
offset (words) substream
0x00 Front L
0x01 Front R
0x02 Digital Front L
0x03 Digital Front R
0x04 Digital Center
0x05 Digital Sub
0x06 Headphones L
0x07 Headphones R
0x08 Rear L
0x09 Rear R
0x0A ADC (multi-rate recording) L
0x0B ADC (multi-rate recording) R
0x0C unused
0x0D unused
0x0E unused
0x0F unused
0x10 Analog Center (Live! 5.1) / dummy (Live! 4.0)
0x11 Analog Sub (Live! 5.1) / dummy (Live! 4.0)
0x12..-0x1F dummy
Audigy / Audigy 2 / Audigy 2 Value / Audigy 4
offset (words) substream
0x00 Digital Front L
0x01 Digital Front R
0x02 Digital Center
0x03 Digital Sub
0x04 Digital Side L (7.1 cards) / Headphones L (5.1 cards)
0x05 Digital Side R (7.1 cards) / Headphones R (5.1 cards)
0x06 Digital Rear L
0x07 Digital Rear R
0x08 Front L
0x09 Front R
0x0A Center
0x0B Sub
0x0C Side L
0x0D Side R
0x0E Rear L
0x0F Rear R
0x10 output to AC97 input L (muted)
0x11 output to AC97 input R (muted)
0x12 unused
0x13 unused
0x14 unused
0x15 unused
0x16 ADC (multi-rate recording) L
0x17 ADC (multi-rate recording) R
0x18 unused
0x19 unused
0x1A unused
0x1B unused
0x1C unused
0x1D unused
0x1E unused
0x1F unused
0x20..0x3F dummy
Fixes:
1. Do not assign negative values to variables used to index emu_cards
array. This array was never accessed when index is negative, but
Alexander (netchild@) told me that Coverity does not like it.
After this change emu_cards[0] should never be used to identify
valid sound card.
2. Fix off-by-one errors in interrupt manager. Add more checks there.
3. Fixes to sound buffering code now allows driver to use large playback
buffers.
4. Fix memory allocation bug when multichannel recording is not
enabled.
5. Fix interrupt timeout when recording with low bitrate (8kHz).
Hardware:
1. Add one more known Audigy ZS card to list. Add two cards with
PCI IDs betwen old known cards and new one.
Other changes:
1. Do not use ALL CAPS in messages.
Incomplete code:
1. Automute S/PDIF when S/PDIF signal is lost.
Tested on i386 only, gcc 3.4.6 & gcc41/gcc42 (syntax only).
---snip---
This commits enables a little bit of debugging output when the driver is
loaded as a module. I did a cross-build test for amd64.
The code has some style issues, this will be addressed later.
The multichannel recording part is some work in progress to allow playing
around with it until the generic sound code is better able to handle
multichannel streams.
This is supposed to fix
CID: 171187
Found by: Coverity Prevent
Submitted by: Yuriy Tsibizov <Yuriy.Tsibizov@gfk.ru>
Add support for Intel High Definition Audio Controller.
This driver make a special guarantee that "playback" works
on majority hardwares with minimal or without specific vendor
quirk.
This driver is a product of collaborative effort made by:
Stephane E. Potvin <sepotvin@videotron.ca>
Andrea Bittau <a.bittau@cs.ucl.ac.uk>
Wesley Morgan <morganw@chemikals.org>
Daniel Eischen <deischen@FreeBSD.org>
Maxime Guillaud <bsd-ports@mguillaud.net>
Ariff Abdullah <ariff@FreeBSD.org>
....and various people from freebsd-multimedia@FreeBSD.org
Refer to snd_hda(4) for features and issues.
Welcome To HDA.
Sponsored by: Defenxis Sdn. Bhd.
This driver make a special guarantee that "playback" works
on majority hardwares with minimal or without specific vendor
quirk.
This driver is a product of collaborative effort made by:
Stephane E. Potvin <sepotvin@videotron.ca>
Andrea Bittau <a.bittau@cs.ucl.ac.uk>
Wesley Morgan <morganw@chemikals.org>
Daniel Eischen <deischen@FreeBSD.org>
Maxime Guillaud <bsd-ports@mguillaud.net>
Ariff Abdullah <ariff@FreeBSD.org>
....and various people from freebsd-multimedia@FreeBSD.org
Refer to snd_hda(4) for features and issues.
Welcome To HDA.
Sponsored by: Defenxis Sdn. Bhd.
