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>
A simple assignment automatically converts a void pointer type
to any other pointer type.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20250515054429.7385-6-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>
According to its man page [1], pw_context_connect() sets errno on
failure:
Returns a Core on success or NULL with errno set on error.
It may be handy to see errno when figuring out why PipeWire
failed to connect. That leaves us with just one possible path to
reach 'fail_error' label which is then moved to that path and
also its error message is adjusted slightly.
1: https://docs.pipewire.org/group__pw__core.html#ga5994e3a54e4ec718094ca02a1234815b
Signed-off-by: Michal Privoznik <mprivozn@redhat.com>
Reviewed-by: Manos Pitsidianakis <manos.pitsidianakis@linaro.org>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-ID: <3a78811ad5b0e87816b7616ab21d2eeef00b9c52.1726647033.git.mprivozn@redhat.com>
macOS versions older than 12.0 are no longer supported.
docs/about/build-platforms.rst says:
> Support for the previous major version will be dropped 2 years after
> the new major version is released or when the vendor itself drops
> support, whichever comes first.
macOS 12.0 was released 2021:
https://www.apple.com/newsroom/2021/10/macos-monterey-is-now-available/
Signed-off-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Reviewed-by: Peter Maydell <peter.maydell@linaro.org>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Message-ID: <20240629-macos-v1-2-6e70a6b700a0@daynix.com>
Signed-off-by: Philippe Mathieu-Daudé <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 term "iothread lock" is obsolete. The APIs use Big QEMU Lock (BQL)
in their names. Update the code comments to use "BQL" instead of
"iothread lock".
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Paul Durrant <paul@xen.org>
Reviewed-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Reviewed-by: Cédric Le Goater <clg@kaod.org>
Reviewed-by: Harsh Prateek Bora <harshpb@linux.ibm.com>
Message-id: 20240102153529.486531-5-stefanha@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
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>
Failed default audio devices were removed from the list but not freed,
and that made LeakSanitizer sad. Free default audio devices as they are
consumed.
Signed-off-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-ID: <20231120112804.9736-1-akihiko.odaki@daynix.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
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>
Make VNC use the default backend again if one is defined.
The recently introduced support for disabling the VNC audio
extension is still used, in case no default backend exists.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
It is now possible to specify the options for the default audio device
using -audio, so there is no need anymore to use a fake -audiodev option.
Remove the fall back to QTAILQ_FIRST(&audio_states), instead remember the
AudioState that was created from default_audiodevs and use that one.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
If "-audio BACKEND" is used without a model, the resulting backend
will be used whenever the audiodev property is not specified.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Match what is done for other options, for example -monitor, and also
the behavior of QEMU 8.1 (see the "legacy_config" variable). Require
the user to specify a backend if one is specified on the command line.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
The "err" variable is only used twice in this code, in a very
local fashion of first assigning it and then checking it in the
next line. So there is no need to declare this variable a second
time in the innermost block, we can re-use the variable that is
declared at the beginning of the function. This fixes the compiler
warning that occurs with "-Wshadow".
Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-ID: <20231004083900.95856-1-thuth@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Now that all callers support setting an audiodev, forbid using the default
audiodev if -nodefaults is provided on the command line.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Starting from audio_driver_init, propagate errors via Error ** so that
audio_init_audiodevs can simply pass &error_fatal, and AUD_register_card
can signal faiure.
Signed-off-by: Martin Kletzander <mkletzan@redhat.com>
[Reworked the audio/audio.c parts, while keeping Martin's hw/ changes. - Paolo]
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
These have been deprecated for a long time, and the introduction of
-audio in 7.1.0 has cemented the new way of specifying an audio backend's
parameters. However, there is still a need for simple configuration
of the audio backend in the desktop case; therefore, if no audiodev is
passed to audio_init(), go through a bunch of simple Audiodev* structures
and pick the first that can be initialized successfully.
The only QEMU_AUDIO_* option that is left in, waiting for a better idea,
is QEMU_AUDIO_DRV=none which is used by qtest.
Remove all the parsing code, including the concept of "can_be_default"
audio drivers: now that audio_prio_list[] is only used in a single place,
wav can be excluded directly in that function.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
An error is already printed by audio_driver_init, but we can make
it more precise if the driver can return an Error *.
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Since all callers require a valid audiodev this function can now safely
abort in case of missing AudioState.
Signed-off-by: Martin Kletzander <mkletzan@redhat.com>
Message-ID: <c6e87e678e914df0f59da2145c2753cdb4a16f63.1650874791.git.mkletzan@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Avoid a dynamic stack allocation in qjack_process(). Since this
function is a JACK process callback, we are not permitted to malloc()
here, so we allocate a working buffer in qjack_client_init() instead.
The codebase has very few VLAs, and if we can get rid of them all we
can make the compiler error on new additions. This is a defensive
measure against security bugs where an on-stack dynamic allocation
isn't correctly size-checked (e.g. CVE-2021-3527).
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Francisco Iglesias <frasse.iglesias@gmail.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-id: 20230818155846.1651287-3-peter.maydell@linaro.org
Avoid a dynamic stack allocation in qjack_client_init(), by using
a g_autofree heap allocation instead.
(We stick with allocate + snprintf() because the JACK API requires
the name to be no more than its maximum size, so g_strdup_printf()
would require an extra truncation step.)
The codebase has very few VLAs, and if we can get rid of them all we
can make the compiler error on new additions. This is a defensive
measure against security bugs where an on-stack dynamic allocation
isn't correctly size-checked (e.g. CVE-2021-3527).
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Francisco Iglesias <frasse.iglesias@gmail.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-id: 20230818155846.1651287-2-peter.maydell@linaro.org
Follow PulseAudio backend comment and code, and only implement the
channels QEMU actually supports at this point, and add the same comment
about limits and future mappings. Simplify a bit the code.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230506163735.3481387-13-marcandre.lureau@redhat.com>
