more IPv4 address from a ranged list in CIRD notation:
ipv4_addrs_ed0="192.168.0.1/24 192.168.1.1-5/28"
In the process move alias processing into new ipv4_up/down functions to
more toward a less IPv4 centric world.
Submitted by: Philipp Wuensche <cryx dash freebsd at h3q dot com>
execute a ET_DYN binary (shared object).
This does not make much sense, but some linux scripts expect to be able to
execute /lib/ld-linux.so.2 (ldd comes to mind).
The sysctl defaults to 0.
MFC after: 3 days
currently present is minor and offers no real semantic issues, it also
doesn't make sense since an earlier lockless check has already
occurred. Also hold the mutex longer, over a manipulation of
per-process ktrace state, which requires synchronization.
MFC after: 1 month
Pointed out by: jhb
This one simply tries to simplify the logic to select the
buffer sizes. I am not sure it is necessary but the code
seems a bit more readable to me. And at least i have tried
to document how the buffer sizes are computed.
Thanks to luigi for deciphering one of the most cryptic part of
sound driver.
Submitted by: luigi
Approved by: netchild (mentor)
In SNDCTL_DSP_SETFRAGMENT, if you specify both read and
write channels, the existing code first acts on the
read channel, but as a side effect it updates the
arguments (maxfrags, fragsz) passed by the caller according
to acceptable values for the read channel, and then uses the
modified values to act on the write channel.
The problem with this approach is that, given a
(maxfrags, fragsz) user-specified value, the actual
values computed by the read and write channels may differ:
e.g. the read channel might want to allocate more fragments
than what the user specified because it has no side-effects
on the delay and it helps in case of slow readers,
whereas the write channel needs to use as few fragments
as possible to keep the audio latency low (very important
with telephony apps).
This patch stores the values computed by the read channel
into temproary variables so the write channel will use
the actual arguments of the ioctl.
This patch is very helpful with telephony apps such as asterisk.
Submitted by: luigi
Approved by: netchild (mentor)
- 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)
erratic system slowdown (beaten to a pulp) and possible panic. This
issue has bugged me for as long as I could remember, until I
realized that it is possible for register base offset to hold zero
value which is definitely a "FALSE".
Approved by: netchild (mentor)
compatible AC97 codec.
- As the driver supports so many variants, create a table ids for
ease of probing and maintenance.
Submitted by: yongari
Reviewed/Tested by: multimedia@
- From luigi:
The code to compute fragment sizes in the ich driver almost
invariably ends up using the full buffer available, no matter
how the user specifies fragment size and number.
With audio telephony (8khz, 16bit-stereo) and the 16k buffer
size this results in an unbearable 500ms delay.
This patch makes sure that we never use more than 4 fragments,
(i don't think we need more unless there are huge interrupt
servicing latencies), and obey to the requested fragment size,
so that latency is acceptable.
Based on this (and after much regression tests), I can conclude
that this driver works best with 2 fragments, thus solving various
long standing issues of ICH driver not capable to flush or play
short files perfectly.
Suggested by: luigi (the idea of smaller fragments)
- MPSAFE conversion.
Approved by: netchild (mentor)
distinct hardware playback channels. DAC configuration can be
accessed through kernel hint - hint.pcm.<unit>.dac="val" with
following possible values:
0 = Enable both DACs (default)
1 = Enable single DAC (DAC1)
2 = Enable single DAC (DAC2)
3 = Enable both DACs, swap position (DAC2 comes first instead
of DAC1)
Special case for ES1370:
Unlike ES1371,2,3/CT5880, volume for each DAC 1 and 2 can be
controlled indepedently (synth for DAC1, pcm for DAC2). It is
possible that user will confuse by this behaviour, since both
DACs are enabled by default. Thus, provide a knob through sysctl
hw.snd.pcm<unit>.single_pcm_mixer:
0 = each DACs will be controlled separately (synth/pcm).
1 = combine both DACs volume mixer controller into a single
"pcm" (default)
As a side note, fixed rate operation (provided by previous
commit) is not a mandatory if the configuration space does not
involve DAC2 (perhaps disabled by user through the above kernel
hint). Unlike DAC2, DAC1 has its own register / control space,
not affected by the speed settings of ADC.
Tested by: multimedia@
Approved by: netchild (mentor)
mask to recdev_l and recdev_r, since each have its own unique mask.
Submitted by: Watanabe Kazuhiro <CQG00620@nifty.ne.jp>
Approved by: netchild (mentor)
o Change the result of gctl(001) now that a bogus verb still requires
a valid geom,
o Insert gctl(024) to test for an appropriate error when a bogus verb
is given that does have a proper geom parameter.
verbs. Only the create verb operates on a provider. All other verbs
operate on a GPT geom. Also, the GPT entry oriented verbs require
a non-downgraded GPT.
o Have all verbs take an optional flags parameter. The flags parameter
is a string of single-letter flags. The typical use of these flags
is to enable certain behaviour in support fo the gpt(8) tool.
o Add dummy implementations for the destroy and recover verbs.
This change causes test 2 of the GPT regression test suite to fail.
The presence of a geom parameter is now required even for unknown
verbs.
If the packet is rejected from pfil(9) then continue the loop rather than
returning, this means that we can still try to send it out the remaining
interfaces but more importantly the mbuf is freed and refcount decremented on
exit.
just discard the received frame and reuse the old mbuf.
This should prevent the connection from stalling after high network traffic.
MFC after: 2 weeks
a new mbuf, just discard the received frame and reuse the old mbuf.
This should fix kernel panics on high network traffic.
Obtained from: NetBSD (joerg@)
MFC after: 2 weeks
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
kernel memory and not using sysctl. Previously, libmemstat was used
only for the live kernel via sysctl paths.
This results in netstat output becoming both more consistent between
core dumps and the live kernel, and also more information in the core
dump case than previously (i.e., mbuf cache information).
Statistics relating to sfbufs still rely on a kvm descriptor as they
are not currently exposed via libmemstat. netstat -m operating on a
core is still unable to print certain sfbuf stats available on the live
kernel.
MFC after: 1 week
