This prevents later file-descriptor shortage when opening files on
demand, which can't be reflected to the application in a sane manner.
The real fix is to open socket files not on libc level but on VFS level
only effectively consume one libc file descriptor for one socket.
* Fix GIC model to support priority and cpu target settings correspondingly
* Fix semantic of SGIR register for GICv2
* Minor GIC model IRQ state fix
* Introduce synchronization for VirtIO and GIC models
* Enable multiple CPUs in test run-script for ARMv8
Fix#3926
* Introduce hypervisor-stack per CPU
* Introduce host world context per CPU
* Mark EL2 translation table memory as inner shareable
* The VMID is not bound to a single VCPU, but to the Vm_session as a whole
* Set affinity of the VCPU accordingly
* Add VMPIDR to VM state
Ref #3926
Instead of calling core to run/pause a VCPU, go directly to the kernel.
Apart from the performance win, it would otherwise involve a more complex
protocol, when a VCPU on another core has to be removed from the scheduler.
Core's entrypoint handling those request runs on the boot-cpu only.
Ref #3926
To enable the interaction of a VMM with the kernel directly,
a hidden RPC gets introduced. It allows a kernel-specific
base-library implementation of the Vm_session::Client to request
a kernel-specific capability to address a VCPU, e.g., to
run/stop it.
Ref #3926
Now, the USB connection is established on backend initialization and
terminated on backend exit triggered by high-level libusb code.
Thanks to Peter for the patch.
- unlink shared memory files
- lower maximum number of socket pool sockets to reduce chance of file
descriptor exhaustion
- fix a build dependency which caused sporadic parallel build errors
Fixes#3910
With this commit, the alignment of anonymous 'mmap()' allocations can be
configured like this:
<config>
<libc>
<mmap align_log2="21"/>
</libc>
</config>
Fixes#3907
This plugin gives access to the Audio_out session by roughly
implementing a OSS pseudo-device. It merely wrapps the session and does
not provide any resampling or re-coding.
Fixes#3891.
In the same vein as the terminal and block I/O controls, the sound
controls are implemented via poperty files and match the OSS
API ([1] features a nice overview while [2] is v3 and [3] gives
in-depth information on the current v4.x API we eventually might want
to implement).
[1] https://wiki.freebsd.org/RyanBeasley/ioctlref/
[2] http://www.opensound.com/pguide/oss.pdf
[3] http://manuals.opensound.com/developer/
The controls currently implemented are the ones used by the cmus OSS
output plugin, which was the driving factor behind the implementation.
It uses the obsolete (v3) API and does not check if the requested
parameter was actually set, which should be done according to the
official OSS documentation.
At the moment it is not possible to set or rather change any
parameters. In case the requested setting differs from the parameters
of the underlying Audio_out session - in contrast to the suggestion in
the OSS manual - we do not silently adjust the parameters returned
to the callee but outright fail the I/O control operation.
The following list contains all currently handled I/O controls.
* SNDCTL_DSP_CHANNELS sets the number of channels. We return the
available channels here and return ENOTSUP if it differs from
the requested number of channels.
* SNDCTL_DSP_GETOSPACE returns amount of playback data that can
be written without blocking. For now it amounts the space left
in the Audio_out packet-stream.
* SNDCTL_DSP_POST forces playback to start. We do nothing and return
success.
* SNDCTL_DSP_RESET is supposed to reset the device when it is
active before any parameters are changed. We do nothing and return
success.
* SNDCTL_DSP_SAMPLESIZE sets the sample size. We return the
sample size of the underlying Audio_out session and return ENOTSUP
if it differs from the requested number of channels.
* SNDCTL_DSP_SETFRAGMENT sets the buffer size hint. We ignore the
hint and return success.
* SNDCTL_DSP_SPEED sets the samplerate. For now, we always return
the rate of the underlying Audio_out session and return ENOTSUP
if it differs from the requested one.
This commit serves as a starting point for further implementing the
OSS API by exploring more users, e.g. as VirtualBox/Qt5/SDL2 audio
backend or a more sophisticated progam like sndiod.
Issue #3891.
At least on some PIT-based platforms (x86_32 + pistachio/okl4/sel4), we run
into trouble with the reworked timeout framework that now proccesses all
pending timeouts before calling their handlers. This order change leads to a
higher rate of handling of short periodic timeouts in the timer driver which
can cause lower prioritized components to starve. Especially, if submitting
signals (from timer to client) isn't cheap (as is the case on qemu + pistachio
for example).
Issue #3884
The driver is faily simple and does not support fancy features like
TCP checksum offloading or vlan filtering, but it is fully capable of
running every Genode network based scenario I've tried. Its currently
known to work on virt_qemu arm platforms and x86_64.
Fix#3825