Commit 90320051ea ("spiceaudio: add a pcm_ops buffer_get_free
function") caused to emit messages saying "Resetting rate control"
frequently when the guest generates no frames.
audio_rate_peek_bytes() resets the rate control when frames < 0 ||
frames > 65536 where frames is the rate-limited number of frames.
Resetting when frames < 0 is sensible as the number simply doesn't make
sense.
There is a problem when frames > 65536. It implies the guest stopped
generating frames for a while so it makes sense to reset the rate
control when the guest resumed generating frames. However, the
commit mentioned earlier broke this assumption by letting spiceaudio
call audio_rate_peek_bytes() whether the guest is generating frames or
not.
Reset the rate control in audio_rate_add_bytes(), which is called only
when actually adding frames, according to the previous call to
audio_rate_peek_bytes() to avoid frequent rate control resets even when
the guest generates no frame.
Signed-off-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Message-Id: <20250317-rate-v1-1-da9df062747c@daynix.com>
Quoting Volker Rümelin: "try-poll=on tells the ALSA backend to try to
use an event loop instead of the audio timer. This works most of the
time. But the poll event handler in the ALSA backend has a bug. For
example, if the guest can't provide enough audio frames in time, the
ALSA buffer is only partly full and the event handler will be called
again and again on every iteration of the main loop. This increases
the processor load and the guest has less processor time to provide
new audio frames in time. I have two examples where a guest can't
recover from this situation and the guest seems to hang."
One reproducer I've found is booting MorphOS demo iso on
qemu-system-ppc -machine pegasos2 -audio alsa which should play a
startup sound but instead it freezes. Even when it does not hang it
plays choppy sound. Volker suggested using command line to set
try-poll=off saying: "The try-poll=off arguments are typically
necessary, because the alsa backend has a design issue with
try-poll=on. If the guest can't provide enough audio frames, it's
really unhelpful to ask for new audio frames on every main loop
iteration until the guest can provide enough audio frames. Timer based
playback doesn't have that problem."
But users cannot easily find this option and having a non-working
default is really unhelpful so to make life easier just set it to
false by default which works until the issue with the alsa backend can
be fixed.
Signed-off-by: BALATON Zoltan <balaton@eik.bme.hu>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
[ Marc-André - Updated QAPI and CLI doc ]
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20250316002046.D066A4E6004@zero.eik.bme.hu>
Commit ed2a4a7941 ("audio: proper support for float samples in
mixeng") added support for float audio samples. As there were no
audio frontend devices with float support at that time, the code
was limited to native endian float samples.
When nobody was paying attention, an audio device that supports
floating point samples crept in with commit eb9ad377bb
("virtio-sound: handle control messages and streams").
Add code for the audio subsystem to convert float samples to the
correct endianness.
The type punning code was taken from the PipeWire project.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20250515054429.7385-7-vr_qemu@t-online.de>
The buffer size calculated by AUD_get_buffer_size_out() is often
incorrect. sw->hw->samples * sw->hw->info.bytes_per_frame is the
size of the mixing engine buffer in audio frames multiplied by
the size of one frame of the audio backend. Due to resampling or
format conversion, the size of the frontend buffer can differ
significantly.
Return the correct buffer size when the mixing engine is used.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20250515054429.7385-3-vr_qemu@t-online.de>
As far as the emulated audio devices are concerned the pointer
returned by AUD_open_out() is an opaque handle. This includes
the NULL pointer. In this case, AUD_get_buffer_size_out() should
return a sensible buffer size instead of triggering a segmentation
fault. All other public AUD_*_out() and audio_*_out() functions
handle this case.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20250515054429.7385-2-vr_qemu@t-online.de>
The general expectation is that header files should follow the same
file/path naming scheme as the corresponding source file. There are
various historical exceptions to this practice in QEMU, with one of
the most notable being the include/qapi/qmp/ directory. Most of the
headers there correspond to source files in qobject/.
This patch corrects most of that inconsistency by creating
include/qobject/ and moving the headers for qobject/ there.
This also fixes MAINTAINERS for include/qapi/qmp/dispatch.h:
scripts/get_maintainer.pl now reports "QAPI" instead of "No
maintainers found".
Signed-off-by: Daniel P. Berrangé <berrange@redhat.com>
Reviewed-by: Zhao Liu <zhao1.liu@intel.com>
Acked-by: Halil Pasic <pasic@linux.ibm.com> #s390x
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-ID: <20241118151235.2665921-2-armbru@redhat.com>
[Rebased]
GLib doesn't implement EXTERNAL on win32 at the moment, and disables
ANONYMOUS by default. zbus dropped support for COOKIE_SHA1 in 5.0,
making it no longer possible to connect to qemu -display dbus.
Since p2p connections are gated by existing QMP (or a D-Bus connection),
qemu -display dbus p2p can accept authentication with ANONYMOUS.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Headers in include/sysemu/ are not only related to system
*emulation*, they are also used by virtualization. Rename
as system/ which is clearer.
Files renamed manually then mechanical change using sed tool.
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Tested-by: Lei Yang <leiyang@redhat.com>
Message-Id: <20241203172445.28576-1-philmd@linaro.org>
The dbus_display1_dep is not really used since all occurrences also
request gio independently. Just list the generated sources and drop
dbus_display1_dep.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
This commit was created with scripts/clean-includes:
./scripts/clean-includes --git misc net/af-xdp.c plugins/*.c audio/pwaudio.c util/userfaultfd.c
All .c should include qemu/osdep.h first. The script performs three
related cleanups:
* Ensure .c files include qemu/osdep.h first.
* Including it in a .h is redundant, since the .c already includes
it. Drop such inclusions.
* Likewise, including headers qemu/osdep.h includes is redundant.
Drop these, too.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Reviewed-by: Zhao Liu <zhao1.liu@intel.com>
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
The Big QEMU Lock (BQL) has many names and they are confusing. The
actual QemuMutex variable is called qemu_global_mutex but it's commonly
referred to as the BQL in discussions and some code comments. The
locking APIs, however, are called qemu_mutex_lock_iothread() and
qemu_mutex_unlock_iothread().
The "iothread" name is historic and comes from when the main thread was
split into into KVM vcpu threads and the "iothread" (now called the main
loop thread). I have contributed to the confusion myself by introducing
a separate --object iothread, a separate concept unrelated to the BQL.
The "iothread" name is no longer appropriate for the BQL. Rename the
locking APIs to:
- void bql_lock(void)
- void bql_unlock(void)
- bool bql_locked(void)
There are more APIs with "iothread" in their names. Subsequent patches
will rename them. There are also comments and documentation that will be
updated in later patches.
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Reviewed-by: Paul Durrant <paul@xen.org>
Acked-by: Fabiano Rosas <farosas@suse.de>
Acked-by: David Woodhouse <dwmw@amazon.co.uk>
Reviewed-by: Cédric Le Goater <clg@kaod.org>
Acked-by: Peter Xu <peterx@redhat.com>
Acked-by: Eric Farman <farman@linux.ibm.com>
Reviewed-by: Harsh Prateek Bora <harshpb@linux.ibm.com>
Acked-by: Hyman Huang <yong.huang@smartx.com>
Reviewed-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Message-id: 20240102153529.486531-2-stefanha@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
error_setg() appends newline to the formatted message.
Fixes: cb94ff5f80 ("audio: propagate Error * out of audio_init")
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Print a debug message as is done for other unsupported audio formats
to give the user the chance to understand their mistake.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Daniel P. Berrangé <berrange@redhat.com>
We can have more than one audio backend.
void audio_init_audiodevs(void)
{
AudiodevListEntry *e;
QSIMPLEQ_FOREACH(e, &audiodevs, next) {
audio_init(e->dev, &error_fatal);
}
}
Reviewed-by: Stefan Berger <stefanb@linux.ibm.com>
Signed-off-by: Juan Quintela <quintela@redhat.com>
Message-ID: <20231020090731.28701-12-quintela@redhat.com>
Default audio devices can now be created with "-audio". Tests for
soundcards were already using "-audiodev" if they want to specify a
particular backend, for the others remove the last remnants of
legacy audio configuration.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
