appears to be too inaccurate that it can be used to synchronize the
playback data stream. If there is a recording endpoint associated with
the playback endpoint, use that instead. That means if the isochronous
OUT endpoint is asynchronus the USB audio driver will automatically
start recording, if possible, to get exact information about the
needed sample rate adjustments. In no recording endpoint is present,
no rate adaption will be done.
While at it fix an issue where the hardware buffer pointers don't get
reset at the first device PCM trigger.
Make some variables 32-bit to avoid problems with multithreading.
MFC after: 3 days
PR: 198444
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.
- SWAPLR quirk for (unknown, luckily it is mine) broken uaudio stick.
Fixing by rewiring is impossible without damaging it. Luckily,
we can fix it using "other" methods :) .
- Add uaudio_get_vendor(), _product() and _release() in uaudio.c
(currently used by uaudio_pcm quirk).
- Implement CHANNEL_SETFRAGMENTS().
- Drop channel locking in few places where it is about to sleep
somewhere. This should help eliminating illegal locking acquisition
where the current thread is about to sleep, and also few deadlock
cases. Dropping it right here is quite safe since it is already
protected by CHN_F_BUSY flag and other threads won't bother to touch it.
Solving other illegal locking issues are quite tricky without converting
most usbd_do_request() calls to its equivalent _async() calls,
which I intend to do it later after getting full test report from
other people with different uaudio hardwares.
- Fix memory leak issues during detach. This seems common to any drivers
(notably emu10kx, csapcm?) with bridge functions.
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
It may be the case that you may hear some unwanted noise while
playing back with 24/32 bit. This is a problem in the USB system.
Explanation from Hans Petter Selasky:
---snip---
The current USB sound driver only uses one isochronous
buffer, that is restarted when it is completed. This will lead to a short
period of time, +1ms, where no sound data is sent to the external USB device.
Depending on the load of your computer, this can be as much as 50ms. So the
USB sound driver must use 2 isochronous transfers. At the beginning one will
queue both. Then these are restarted on completion. This will result in a
constant-rate data stream to the external sound device, a minimum sound
buffer equal to the size of the isochronous buffer, and possibly the sound
will reach your ears with less delay. Little delay is a result of constant
data rate. Currently only my USB driver will support that. If one tries that
with the USB driver in *BSD, then it will crash at the first moment one gets
a buffer underrun.
---snip---
Submitted by: Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp>
Mono-recording still not tested by: julian
this buffer anyway so the constraint that it had to be DMA capable only
caused pain when devices failed to aquire the memory. Use a regular
malloc instead with sndbuf_setup.
Approved by: tanimura (mentor)
One of a set of patches submitted by Kazuhito HONDA
to make the usb audio driver a lot more capable.
PR: 75274
Submitted by: Kazuhito HONDA (kazuhito at ph dot noda dot tus dot ac dot jp)
Obtained from: NetBSD (indirectly)
MFC after: 2 weeks
- `sound'
The generic sound driver, always required.
- `snd_*'
Device-dependent drivers, named after the sound module names.
Configure accordingly to your hardware.
In addition, rename the `snd_pcm' module to `sound' in order to sync
with the driver names.
Suggested by: cg
Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma. At the moment, this is used for the
asynchronous busdma_swi and callback mechanism. Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg. dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create(). The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.
sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms. The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.
If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.
Reviewed by: tmm, gibbs
This driver actually works slightly better on -stable than on -current
(the system locks on detach on -current), so it should be MFC'd somewhat
sooner.
This driver currently points out a difficulty in the sound device framework.
The PCM unregister routine is allowed to refuse the detach if the device is
in use. In the case of a USB device, however, this unregistration is much more
mandatory in nature, since the device is *actually* gone when this call is
made. The sound subsystem really should not refuse an unregistration and
should take its own steps to reject further I/O. As a result, if you detach
a USB sound device while it is in use, you can expect a panic shortly
thereafter.
This device cannot currently record audio. Some routines are unwritten as
of yet in uaudio.c to support recording.
This device hangs my -current box on detach. I don't know why. This does
not happen on my -stable machine.
Obtained from: Hiroyuki Aizu
MFC after: 2 weeks
