at r230551.
Also while there, make sense polling use reported for each node separately
instead of reporting accumulated total status.
Submitted by: Barbara <barbara.freebsd@gmail.com> (1)
MFC after: 3 days
names to shorten them. PulseAudio reported to have problems with names
longer then 63 chars and at least in XMMS long names are inconvinient.
Reported by: hselasky
MFC after: 3 days
another process is in open() or stat() for the device node, then
close() from the owning process does not result in cdevsw close
method call. This fixes the pemanent "Device busy" seen.
Changed the sndstat_lock from mutex to sx. This allows to extend
the region covered by the lock, to include the uiomove() call in
sndstat_read() and bufptr increment. This fixes the "panic:
sbuf_put_byte called with finished or corrupt sbuf" seen.
In collaboration with: kib
MFC after: 1 week
MOD_SHUTDOWN is not an end of existence, and there is a life after it.
In particular, code previously called on MOD_SHUTDOWN grabbed lock and
deallocated unit numbering. That caused infinite wait loop if snd_uaudio
tried to destroy its PCM device after that point.
MFC after: 3 days
Code should just use the devtoname() function to obtain the name of a
character device. Also add const keywords to pieces of code that need it
to build properly.
MFC after: 2 weeks
The cs driver requires a table with firmware values. An
alternative firmware is available in a similar Open Sound
System driver. This is actually a partial revert of
Revision 77504.
Special thanks to joel@ for patiently testing several
replacement attempts.
The csa driver and the complete sound system are now free
of the GPL.
Tested by: joel
Approved by: jhb (mentor)
MFC after: 3 weeks
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@
Remove unneeded temporary variable (data) to better match the OSS code.
Remove some unused constants and type definitions.
Tested by: joel
Approved by: jhb (mentor)
MFC after: 3 weeks
- add "+HP" in case of headphones redirection;
- add device type for analog devices, if all pins have the same.
As result now it may look like "Analog 5.1+HP/2.0" or "Front Analog Mic".
I hope it will be more useful than long and confusing.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
selection in snd_hda(4) driver.
Now driver tracks jack presence detection status for every CODEC pin. For
playback associations, when configured, that information, same as before,
can be used to automatically redirect audio to headphones. Also same as
before, these events are used to track digital display connection status
and fetch ELD. Now in addition to that driver uses that information to
automatically switch recording source of the mixer to the connected input.
When there are devices with no jack detection and with one both connected,
last ones will have the precedence. As result, on most laptops after boot
internal microphone should be automatically selected. But if external one
(for example, headset) connected, it will be selected automatically.
When external mic disconnected, internal one will be selected again.
Automatic recording source selection is enabled by default now to make
recording work out of the box without touching mixer. But it can be
disabled or limited only to attach time using hint.pcm.X.rec.autosrc loader
tunables or dev.pcm.X.rec.autosrc sysctls.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
(HDMI and HBR bits set) and needed (AC3 format used with 8 channels).
This should allow DTS-HD/TrueHD pass-through with rates above 6.144Mbps.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
widgets. I am not sure if S/PDIF supports 32bit samples, but my Marantz
SR4001 doesn't, producing only single clicks on playback start/stop.
Because HDA controller requires 32bit alignment for all samples above 16bit,
we can't handle this situation in regular way and have to set 32bit format
in sound(4) for anything above 16bit. To workaround the problem, prefer
to setup hardware to use 24/20bit samples when 32bit format requested. Add
dev.pcm.X.play.32bit and dev.pcm.X.rec.32bit sysctls to control what format
really use for 32bit samples.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
maximal from 64K to 256K.
We usually don't need 750 sound interrupts per second (1.3ms latency)
when playing 192K/24/8 stream. 187 should be better. On usual 48K/16/2
it is just enough for hw.snd.latency=9 at hw.snd.latency_profile=1 with
23 and 6 interrupts per second.
MFC after: 2 weeks
Sponsored by: iXsystems, Inc.
Previous code was relatively dumb. During CODEC probe it was tracing signals
and statically binding amplifier controls to the OSS mixer controls. To set
volume it just set all bound amplifier controls proportionally to mixer
level, not looking on their hierarchy and amplification levels/offsets.
New code is much smarter. It also traces signals during probe, but mostly
to find out possible amplification control rages in dB for each specific
signal. To set volume it retraces each affected signal again and sets
amplifiers controls recursively to reach desired amplification level in dB.
It would be nice to export values in dB to user, but unluckily our OSS mixer
API is too simple for that.
As result of this change:
- cascaded amplifiers will work together to reach maximal precision.
If some input has 0/+40dB preamplifier with 10dB step and -10/+10dB mixer
with 1dB step after it, new code will use both to provide 0/+40dB control
with 1dB step! We could even get -10/+50dB range there, but that is
intentionally blocked for now.
- different channels of multichannel associations on non-uniform CODECs
such as VIA VT1708S will have the same volume, not looking that control
ranges are different. It was not good when fronts were 12dB louder.
- for multiplexed recording, when we can record from only one source at
a time, we can now use recording amplifier controls to set different
volume levels for different inputs if they have no own controls of they
are less precise. If recording source change, amplifiers will be
reconfigured.
To improve out-of-the-box behavior, ignore default volume levels set by
sound(4) and use own, more reasonable: +20dB for mics, -10dB for analog
output volume and 0dB for the rest of controls. sound(4) defaults of 75%
mean absolutely random things for different controls of different CODECs
because of very different control ranges.
Together with further planned automatic recording source selection this
should allow users to get fine playback and recording without touching
mixer first.
Note that existing users should delete /var/db/mixer*-state and reboot
or trigger CODEC reconfiguration to get new default values.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
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
frightening "unknown" word. In most cases we don't need to know chips
to properly handle them, but having IDs in logs may simplify debugging.
MFC after: 2 weeks
Sponsored by: iXsystems, Inc.
of HDA bus. Handle that from two directions:
- Add support for "striping" (using several SDO lines), if supported.
- Account HDA bus utilization and return error on new stream allocation
attempt if remaining bandwidth is unsifficient.
Most of HDA controllers have one SDO line with 46Mbps output bandwidth.
NVIDIA GF210 has 2 lines - 92Mbps. NVIDIA GF520 has 4 lines - 184Mbps!
MFC after: 2 months
Sponsored by: iXsystems, Inc.
- Enable and handle unsolicited responses from digital display pins,
reporting connection and EDID-Like Data (ELD) validity status changes.
- Fetch ELD data, describing connected digital display device audio
capabilities. These data not really used at the moment (user is not
denied to use audio formats not supported by the device), only printed to
verbose logs. But they are useful for debugging. The fact that ELD was
received tells that HDMI link was established and video driver enabled
HDMI audio passthrough. Some old chips may not return ELD, so lack of it
is not necessary a problem.
- Add some more points to CODEC configuration sequence:
- For converter widgets, supporting more then two channels (HDMI/DP
converter widgets support 8), set number of channels to handle.
- For digital display pins (HDMI/DP) fill audio infoframe, reporting
connected device about number of channels and speakers allocation.
- For digital display pins (HDMI/DP) set mapping between channels seen
by software and channels transferred via HDMI/DisplayPort.
- Allow more audio formats, not used for analog connections because of
stereo pairs orientation, but easily applicable to HDMI/DisplayPort: 2.1,
3.0, 3.1, 4.1, 5.0, 6.0, 6.1, 7.0. That list may be filtered later using
info from ELD.
- Disable MSI interrupts for NVIDIA HDA controllers before GT520.
At this point I can successfully play audio over HDMI from NVIDIA GT210
and GT520 cards with nvidia-driver-290.10 driver to Marantz SR4001
receiver in 2.0, 2.1, 3.0, 4.0, 4.1, 5.0 and 5.1 PCM formats at 44, 48,
88 and 96KHz at 16 and 24 bits, same as do AC3/DTS passthrough.
6.0, 6.1, 7.0 and 7.1 PCM formats are not working for me, but I think
it is because of receiver age.
MFC after: 2 months
Sponsored by: iXsystems, Inc.
- 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
