make mntflushfree() return the original rpc and do the
botched clunk check on the original instead of the
current rpc.
so if we get a botched flush of a clunk, we abandon the
fid of the channel as well.
if theres an error transmitting a Tclunk or Tremove request,
we cannot assume the fid to be clunked. in case this was
a transient error, reusing the fid on further requests
will fail.
as a work arround, we zero the channels fid and allocate
a new fid before the chan is reused.
this is not correct as we essentially leak the fid
on the fileserver, but we will still be able to use
the mount.
the shift instructions does not change the zero flag
when the shift count is 0, so we cannot remove the
compare instruction in this case.
this fixes oggdec under 386.
ftrvxmtrx repots devices that use the endpoint number for
input and output of different types like:
nusb/ether: parsedesc endpoint 5[7] 07 05 81 03 08 00 09 # ep1 in intr
nusb/ether: parsedesc endpoint 5[7] 07 05 82 02 00 02 00
nusb/ether: parsedesc endpoint 5[7] 07 05 01 02 00 02 00 # ep1 out bulk
the previous change tried to work arround this but had the
concequence that only the lastly defined endpoint was
usable.
this change addresses the issue by allowing up to 32 endpoints
per device (16 output + 16 input endpoints) in devusb. the
hci driver will ignore the 4th bit and will only use the
lower 4 bits as endpoint address when talking to the usb
device.
when we encounter a conflict, we map the input endpoint
to the upper id range 16..31 and the output endpoint
to id 0..15 so two distinct endpoints are created.
nusb code assumes endpoint numbers are unique. It's true in general
case, but it becomes false once the direction bit is ignored. The
commit adds a check so that two endpoints of different types are not
merged into one with Eboth direction. It does overwrite endpoint
though, so it shouldn't be considered as a full fix.
the 802.11 spec only specifies the msb of the rate for
Beacon, Probe Response, Association Response, Reassociation Response,
Mesh Peering Open, and Mesh Peering Confirm management frames
...
The MSB of each Supported Rate octet in other
management frame types is ignored by receiving STAs.
this should make no difference but on some netgear ap's not
setting this bit seems to ignore these data rates.
driver sets wifi->rates array to tell wifi layer what
rates it supports. when we receive beacon, we determine
the minimum and maximum data rates and set wn->minrate
and wn->maxrate to point to the entries in wifi->rates.
it is the responsibility of the driver to use this
information on transmit.
to run aux/wpa at boot, we need factotum to be running. tho
factotum was started only after the network was configured.
what we do now is start factotum early, not fetching keys
from secstore. once network is available and the auth server
is known, we fetch keys from secstore using auth/secstore in
bootrc.
to pass the authserver for p9 authentication to factotum, we
write it in /net/ndb and the special _authdial() in factotum
will picks it up.
as we are using auth/secstore binary in any case, we remove
the duplicated secstore code from factotum and make it just
exec auth/secstore to fetch the keys on startup (unless -n
or -S is specified).
load x-stretched scanline and use image replication bit to let
devdraw do the y-stretching. this reduces slow RGB15 -> display
conversions as devdraw caches small numbers of converted source
scanlines (one in hour case). setting pixels should also be a
bit faster. (only 3 writes instead of 9 for x3 scaling)
remove unused memsetb() routine.
replace foo ? 1 : 0 with foo != 0
avoid double calculation of rgb components in readbyte()
handle byte aligned color components in writebyte()
which lets us avoid the read-modify-write and the shifting.
surprisingly, the branches in the loop are way less important
than avoiding the memory access.
this change makes ganes/snes playable at -3 scaling.
used to set audio device frequency thru /dev/volume tho
only ac97 driver supports this. as a quick work arround,
upsample the 32000 hz audio signal to 44100 hz (without
any interpolation).
move the sample buffer room check from audiosample() into
dspstep() so that when the buffer is full (shouldnt happen),
we wont advance dspstate so samples will not get dropped.
