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.
- make usb2_power_mask_t 16-bit
- remove "usb2_config_sub" structure from "usb2_config". To compensate for this
"usb2_config" has a new field called "usb_mode" which select for which mode
the current xfer entry is active. Options are: a) Device mode only b) Host
mode only (default-by-zero) c) Both modes. This change was scripted using
the following sed script: "s/\.mh\././g".
- the standard packet size table in "usb_transfer.c" is now a function, hence
the code for the function uses less memory than the table itself.
Submitted by: Hans Petter Selasky
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>
the devfs clone handler to open the (invisible) devices on the fly.
The /dev entries are layed out as follows,
/dev/usbctl = master device
/dev/usb/0.1.0.5 = usb device, (<bus>.<dev>.<iface>.<endpoint>)
/dev/ugen0.1 -> usb/0.1.0.0 = ugen link to ctrl endpoint
This also removes the custom permissions model from USB. Bump
__FreeBSD_version to 800066.
Submitted by: rink (earlier version)
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.
use patches so far:
+ Envy24:
- fix: broken init data for M Audio Delta DiO 2496
- add: support for M Audio Delta 44
- add: support for M Audio Delta 1010LT
Tested by: Dominique Goncalves, dominique.goncalves at gmail.com
- add: support for Terratec EWX 2496
Tested by: Stefan Sperling, stsp at stsp.name
- add: support for M Audio Delta 66
Tested by: Richard Bown, richard.bown at blueyonder.co.uk
- add: support for M Audio Delta 1010
Tested by: Andrew Reilly, areilly at bigpond.net.au
+ Envy24HT:
- add: support for Terrasoniq TS22PCI
- fix: M-Audio Revolution 5.1 sound volume is very low
Reported by: Oliver Hartmann, ohartman at zedat.fu-berlin.de
Andrey Slusar, anrays at gmail.com
Tested by: Andrey Slusar, anrays at gmail.com
Rusu Silviu, arol.the at gmail.com
- fix: M-Audio Revolution 7.1 sound is distorted and very quiet
Reported by: Olev Hannula, hannula at gmail.com
Tested by: Olev Hannula, hannula at gmail.com
Stanislav Belansky, stanislav at icmail.ru
- fix: Terratec PHASE 22 codec is power-off due to wrong init data
Reported by: Philipp Ost, pj at smo.de
Tested by: Philipp Ost, pj at smo.de
+ SpicDS:
- fix: AK4381 produce hiss sound on 192kHz sample rate
- fix: stupid bug with volume control for AK4396
Submitted by: Konstantin Dimitrov <kosio.dimitrov@gmail.com>
solving a possible panic when snd_ai2s is loaded at boot time. Also change
the device setup to indicate to the pcm layer that the device is MPSAFE.
Submitted by: Marco Trillo
Suggestions by: Ariff Abdullah
