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.
seems not enough to verify its consistencies.
- Define AC97_MIXER_SIZE as SOUND_MIXER_NRDEVICES (25), since we
don't need more than that. Stop doing wild and random guess about
its size since we're stricly bound to it.
sysctl_handle_int is not sizeof the int type you want to export.
The type must always be an int or an unsigned int.
Remove the instances where a sizeof(variable) is passed to stop
people accidently cut and pasting these examples.
In a few places this was sysctl_handle_int was being used on 64 bit
types, which would truncate the value to be exported. In these
cases use sysctl_handle_quad to export them and change the format
to Q so that sysctl(1) can still print them.
and should only be applied on certain specific card / vendor, hence the
addition of ac97_getsubvendor().
- Fix low volume issue on several MSI laptops through ALC655 quirk.
Reported/Tested by: Christian Mueller
<raptor-freebsd-multimedia@xpls.de>
MFC after: 1 week
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.
commit.
1) sys/dev/sound/pcm/sound.h
sys/dev/sound/pcm/channel.c
* Be more specific: SD_F_SOFTVOL -> SD_F_SOFTPCMVOL
2) sys/dev/sound/pcm/mixer.[ch]
* Implement
mix_setparentchild()
mix_setrealdev()
mix_getparent()
mix_getchild()
The purpose of these functions is implement relative volume
adjustment, such as to tie two or more mixer device into a
single logical device. Usefull for the upcoming HDA driver
and few AC97 codec (such as AD1981B) where the master volume
"vol" need to be implemented using this logical manner.
3) sys/dev/sound/pcm/ac97_patch.[ch]
* Patch for AD1981B codec to enable (automuting) headphone jack sense.
4) sys/dev/sound/pcm/ac97.c
* Implement proper logical master volume for AD9181B codec
through various mix_set{parentchild,realdev}(). Tie both
"ogain" (headphone volume) and "phone" (speaker/lineout) to
a logical "vol".
5) sys/dev/sound/pcm/usb/uaudio_pcm.c
* ditto, for "vol" -> { "pcm" }.
MFC after: 1 month
- Added new codec id for CX20468-21 and VIA1617A.
Submitted by: Chen Lihong <lihong.chen@gmail.com>
- Re-enable SOUND_MIXER_IGAIN, but set the default level as 0 (mute)
Suggested by: luigi
mixer.c:
- Set default value for SOUND_MIXER_IGAIN as 0 (mute) to avoid
feedback problems on some laptops (was disabled by jhb during
ac97.c revision 1.42).
Approved by: netchild (mentor)
* Added codec id for CMI9761.
* feeder_volume *whitelist* through ac97_fix_volume()
sys/dev/sound/pcm/ac97.h:
* Added AC97_F_SOFTVOL definition.
sys/dev/sound/pcm/channel.c:
* Slight changes for chn_setvolume() to conform with OSS.
* FEEDER_VOLUME is now part of feeder building process.
sys/dev/sound/pcm/mixer.c:
* General spl* cleanup. It doesn't serve any purpose anymore.
* Main hook for feeder_volume.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
This mostly to help CT4730, but apparently it does help other
cards too (especially via8233x). This probably need further test
and confirmation from other people with ac97 cards other than via
/ es137x.
* Aggresive dac power wake up call, again, to help CT4730 (and
probably others).
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Tested by: multimedia@
Affects to people WITH an AD1888 codec, the system will output to the port
labeled "speaker" instead of microphone. System will work the same in
multiple operating systems.
If people are currently using their systems with this codec they will need
to swap their output ports.
I have _not_ checked audio input or line input (basically, I have checked
nothing other than line-out).
I believe this is an appropriate change, it makes us consistent with
documentation, and other operating systems. Furthermore, this feature
(playing) is the vast majority of sound activities, so if this makes is
right for playback and wrong for recording... playback is more important,
and we can fix recoding in the future without worries of screwing people
again in the future (since we'll be "right" on the playback).
Submitted by: David Cross
o AD1980 hook.
o ac97_fix_auxout.
and:
o Associate AC97_MIX_AUXOUT with SOUND_MIXER_OGAIN rather than
SOUND_MIXER_MONITOR.
o Add ac97_fix_tone to remove tone controls from mixer if invalid.
Attempt to determine what function of AUX_OUT is: "True line level
out", "Headphone out", or "4-Channel out" and frig OSS mixer label
accordingly.
Addresses problem raised by Randy Bush on -multimedia of not being
able to hear audio on ich2 m/b which was eventually found to be
because the mixer monitor value was 0. On this h/w the label
"monitor" should now be presented as the marginally more intuitive
"ogain".