AudioBackend had a vm_running field which was set in
audio_vm_change_state_handler().
The state change handler "bool running" argument is true when
vm_prepare_start() calls it, and the VM runstate is either SUSPENDED or
RUNNING.
Audio hw voices shouldn't be running when the VM is suspended, but only
when running. Thus replacing the vm_running field with a call to
runstate_is_running() is both simpler and more correct.
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Set "using_dbus_display" during early_dbus_init(), so that we can try to
create the "dbus" audio backend by default from audio_prio_list.
This makes dbus audio work by default when using an audio device,
without having to setup and wire up the -audiodev manually.
The added FIXME is addressed in the following commits.
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Previous commits made this dead code.
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
AUD_init_time_stamp_{in,out} and AUD_get_elapsed_usec_{in,out} are only
used by the adlib device. The result isn't actually being used since
ADLIB_KILL_TIMERS was set some 20y ago. Let's drop this dead code now.
Drop QEMUAudioTimeStamp as well as reported by Akihiko Odaki.
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
All these files indirectly include the "qemu/bswap.h" header.
Make this inclusion explicit to avoid build errors when
refactoring unrelated headers.
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Message-ID: <20260109164742.58041-4-philmd@linaro.org>
The command is niche and better served by the host audio system.
There is no QMP equivalent, fortunately. You can capture the audio
stream via remote desktop protocols too (dbus, vnc, spice).
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Acked-by: Dr. David Alan Gilbert <dave@treblig.org>
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Message-Id: <20251022105753.1474739-1-marcandre.lureau@redhat.com>
@endianness is used as a boolean, rename for clarity.
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
For modularity/clarity reasons, move the capture API in a specific
header.
The current audio/ header license is MIT.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Use slightly better types for the job.
Fix some checkpatch issues.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
There is no clear need for this extra intermediary structure between
the audio backend and its user.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Suggested-by: Paolo Bonzini <pbonzini@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Naming is hard. But in general in QEMU, a host "backend" is the term
used to fullfill the request made by the device or frontend.
AudioBackend will become an abstract base class in a follow-up series.
Currently the frontend is QEMUSoundCard, we are going to drop that next.
Note that "audiodev" is the corresponding QAPI type name (or configuration).
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Fix some check-patch issues while at it.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
There are 2 sets of functions since the introduction of multi-channel
Volume structure: AUD_set_volume_{in,out} and audio_set_volume_{in,out}.
Use the AUD_ prefix for consistency with other audio.c functions. Rename
the stereo function with "_lr" suffix.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
The only reason it would fail to add the handler is if it's calling a
stub. But this cannot happen as audio is only supported with system qemu.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Catch and return from error early to avoid indentations and ease the
flow & return a bool for success value. All driver init() calls have
been checked to set errp on error.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Whenever NULL is returned, errp should be set.
Inline SetCooperativeLevel call to simplify code.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Use dsound_audio_fini() on error & fail if the capture failed to
initialize too.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
audio_cleanup() is already called at exit (similar to chardev)
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
QOM tree now has /audiodevs objects.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
QOM brings some conveniences for introspection, type checking, reference
counting, interfaces etc. This is only the first step to introduce QOM
in audio/ (I have more in the pipeline)
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
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>
Marc-André Lureau
Philippe Mathieu-Daudé
Akihiko Odaki
BALATON Zoltan
Volker Rümelin
Daniel P. Berrangé
Michal Privoznik
Paolo Bonzini
Peter Maydell