- 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.
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.
- SMBus Controller
- SATA Controller
- HD Audio Controller
- Watchdog Controller
Thanks to Seth Heasley (seth.heasley@intel.com) for providing us code.
MFC after 3 days
infrastructure, not us. This appears to be a leftover from an older
version of the driver.
Submitted by: avg
Tested by: Anton Shterenlikht <mexas bristol.ac.uk>
MFC after: 1 week
X-MFC-Note: To stable/8 and stable/7 only
loopback.
- Change the meaning of "mix" OSS control. Now it controls loopback level,
according to comments in soundcard.h.
- Allow AD1981HD codecs to use playback mixer. Now driver should be able to
really use it.
- Fix bug in shared muters operation.
now due to unidentified synchonization problem. For 7.1 soundcards 5.1
support handled correctly via software upmix done by sound(4).
Stereo stream is no more duplicated to all ports. If you loose sound, check
you are using right connectors. Front speakers connector is usually green,
center/LFE - orange, rear - black, side - gray.
The newbus lock is responsible for protecting newbus internIal structures,
device states and devclass flags. It is necessary to hold it when all
such datas are accessed. For the other operations, softc locking should
ensure enough protection to avoid races.
Newbus lock is automatically held when virtual operations on the device
and bus are invoked when loading the driver or when the suspend/resume
take place. For other 'spourious' operations trying to access/modify
the newbus topology, newbus lock needs to be automatically acquired and
dropped.
For the moment Giant is also acquired in some key point (modules subsystem)
in order to avoid problems before the 8.0 release as module handlers could
make assumptions about it. This Giant locking should go just after
the release happens.
Please keep in mind that the public interface can be expanded in order
to provide more support, if there are really necessities at some point
and also some bugs could arise as long as the patch needs a bit of
further testing.
Bump __FreeBSD_version in order to reflect the newbus lock introduction.
Reviewed by: ed, hps, jhb, imp, mav, scottl
No answer by: ariff, thompsa, yongari
Tested by: pho,
G. Trematerra <giovanni dot trematerra at gmail dot com>,
Brandon Gooch <jamesbrandongooch at gmail dot com>
Sponsored by: Yahoo! Incorporated
Approved by: re (ksmith)
- honor parent DMA tag limitations, as man page requires,
- allow data buffer to be allocated within full 64bit address range, when
support is announced by hardware,
- add quirk, disabling 64bit addresses for broken chips, use it for MCP78.
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.
mic inputs. I have no idea what for it was made that time, but now I have
several reports that it should be removed to make microphones work. If
this quirk is still required for some systems then they should be identified
and specified explicitly.
only for mic-type inputs. This gives better chances to use it.
Change default configuration for some AD1986A codec based ASUS boards,
use it also for ASUS P5PL2 board. This makes front mic preamplifier working.
Tested by: Vadim Frolov <frolov@frolov.ck.ua>
implement CD input in hardware, while unconditional showing it confuse users.
Also it was made in the way that sometimes improper with present driver.
Add patch for ALC268 based Acer TM5320 to make headphones jack sensing work.
Default configuration defines two separate playback associations, which
current driver unable to trace properly due to order they are defined and
limited codec uniformity.
Submitted by: G. Mirov <g.mirov AT gmail.com>
Disable MSI for nVidia MCP51 controller. Enabling MSI there leads to
unexpected errors and timeouts, that should not happen even if interrupts
are not working completely.
Disable some unneeded pathes in overcomplicated input mixer to help parser
to handle the rest better. This gives mic input boost control in some
configurations and just more predictable operation in others.
nodes capabilities. Add "Analog"/"Digital" marks to the pcm device names.
I hope it will help new users easier accept concept of several PCM devices
and understand exact purposes of that devices.
with several points unappropriate for the present parser. This patch
disables input-to-output analog monitoring but instead fixes recording.
Tested by Tobias Grosser on ThinkPad T61p.
Left only parts surely required for basic troubleshooting and configuration
purposes. There is still very long output, but further shrinking makes it
less informative.
Original debugging can be enabled with hw.snd.verbose=4.
Because of using more clear and same time more functional codec parser
new driver is able to handle more codecs, use them better then before and
without most of previous quirks. All of tested codecs itself manage playback,
record, input mixing and monitoring quite fine. In all of investigated
trouble cases problem was found or in nonstandard codec usage or incorrect
codec configuration made by BIOS. Most of that cases could be fixed using
device hints, some of which are already included to the driver.
New driver supports multiple codecs per HDA bus, multiple audio function
groups per codec and multiple logical sound devices per audio function group.
So don't worry when you get several PCM devices instead of one, it is normal.
It is usual situation with powerful codecs to provide, for example, 3 PCM
devices: one for 7.1 playback and main recording, one for headset and one
for digital SPDIF I/O.
New driver implements Universal Audio Architecture (UAA) much better then
previous one. Most information about recommended codec usage now taken from
the codec configuration registers initialized by BIOS. User may alter that
configuration using device hints to reconfigure logical audio devices to
his needs in a very broad range up to the limits of the codec functionality.
New driver supports digital PCM playback and AC3 pass-through. I am not sure
about completeness of this implementation, but I have several success stories
including my own. Vchans subsystem does not support AC3 pass-through so it
had to be disabled for that devices at this moment.
New driver is ready for multichannel playback, but until our OSS is unable
to use this it will just duplicate same stereo stream into all channel
pairs.
New driver supports suspend/resume. I am unable to really test this part
myself, but I have got several success stories.
Driver has very informative verbose boot messages. So if you have any
questions or problems - enable and read them first.
Discussed on: freebsd-multimedia@
Tested by: many
