Since commit 8f68a33ad4 we get on macOS:
Audio backends
CoreAudio support : YES
PipeWire support : NO
JACK support : YES 1.9.22
../audio/jackaudio.c:654:12: error: unused function 'qjack_thread_creator' [-Werror,-Wunused-function]
654 | static int qjack_thread_creator(jack_native_thread_t *thread,
| ^~~~~~~~~~~~~~~~~~~~
This is simply due to a missing #ifdef'ry change. Update
so we can use the new qemu_thread_set_name() exposed by
commit 46255cc2be.
Fixes: 8f68a33ad4 ("audio: make jackaudio use qemu_thread_set_name")
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20260307114923.75394-1-philmd@linaro.org>
This has greater portability than directly call pthread_setname_np,
which is only 1 out of 3 possible functions for pthreads that can
set the name.
The new API requires a trampoline function, since it can only set
the name of the current thread.
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Signed-off-by: Daniel P. Berrangé <berrange@redhat.com>
Spice server 0.15.0 was released on 2021-04-16. It is part of all our
supported distro (except CentOS 9, which doesn't include it).
It has all the new required audio APIs/interfaces.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Link: https://lore.kernel.org/r/20260211-cleanups-v1-5-e63c96572389@redhat.com
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
pulseaudio 0.9.13 was released on 2009-09-10. All our supported
distros have it.
PA_*_IS_GOOD are from 0.9.11.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Link: https://lore.kernel.org/r/20260211-cleanups-v1-4-e63c96572389@redhat.com
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Simply use the class name instead.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
audio/ is now converted to use QEMU standard trace & error reporting.
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>
audio_bug() is a bit unconventional, it is meant to be used as a
condition expression, passing the actual condition as second argument
(and __func__ as first argument).
If the condition is true, it uses AUD_log() to print to stderr, and has
some dubious recommendations printed only once.
This change:
- clears the control flow, and make the condition directly visible in
the 'if' statement.
- uses standard QEMU error_report()
- audio_bug() now captures __func__
- remove the "Save all your work and restart..." once hint
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>
Currently, when given invalid settings, QEMU will print to stderr:
A bug was just triggered in audio_mixeng_backend_open_out
Save all your work and restart without audio
I am sorry
Context:
audio: frequency=44100 nchannels=0 fmt=S16 endianness=little
Now it will use error_report() and simply report:
audio: Invalid audio settings: frequency=44100 nchannels=0 fmt=s16 endian=little
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>
Use more correct GTimer, specific for each backend, with a trace event.
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>
Replace the ldebug macro calls with proper trace events:
- audio_get_avail: traces available samples in capture stream
- audio_open_out/audio_open_in: traces stream open parameters
This provides structured tracing that can be enabled at runtime
without requiring debug builds.
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>
The endianness field used an int to represent a boolean concept, with
0 meaning little-endian and 1 meaning big-endian. This required runtime
validation to reject invalid values and made the code less readable.
Replace with a bool big_endian field that is self-documenting and
type-safe. The compiler now enforces valid values, eliminating the
need for the validation check in audio_validate_settings().
Reviewed-by: Thomas Huth <thuth@redhat.com>
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>
The audio_pcm_info structure stored three fields (bits, is_signed,
is_float) that were always derived from the AudioFormat enum. This
redundancy meant the same information was represented twice, with no
type-level guarantee that they stayed in sync.
Replace these fields with a single AudioFormat field, and add helper
functions to extract the derived properties when needed:
- audio_format_bits()
- audio_format_is_signed()
- audio_format_is_float()
This improves type safety by making AudioFormat the single source of
truth, eliminating the possibility of inconsistent state between the
format enum and its derived boolean/integer representations.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
The AUDIO_CAP macro is no longer used.
Reviewed-by: Thomas Huth <thuth@redhat.com>
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>
Replace the custom audio logging infrastructure (dolog macro and
AUD_log/AUD_vlog) with standard QEMU error reporting (error_report,
error_printf, error_vprintf).
Note that we also dropped the abort() call in DEBUG_AUDIO, as it is not
usually compiled with, doesn't help much, and can easily be added back
when doing development as needed.
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Replace the custom audio logging infrastructure with standard QEMU
error reporting and tracing.
qpa_conn_init() is called during audio_pa_realize() and already reports
an error through Error *.
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>
Replace the custom audio logging infrastructure with standard QEMU
error reporting and tracing.
Note the patch drops DEBUG_MISMATCH condition, and now always trace
actual audio parameters.
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>
Replace the custom audio logging infrastructure with standard QEMU
error reporting and tracing.
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>
The DirectSound audio backend uses its own logging infrastructure
(AUD_log, AUD_vlog, dolog) and the AUDIO_CAP macro. This approach is
inconsistent with the rest of QEMU and makes the output harder to
filter and configure.
Replace the custom logging with standard QEMU error reporting.
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>
The DBus audio backend already uses error_report() for error logging.
Remove the unused AUDIO_CAP macro which was left over from the old
logging infrastructure.
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>
The CoreAudio backend uses its own logging infrastructure (AUD_log,
AUD_vlog, dolog) and the AUDIO_CAP macro. This approach is inconsistent
with the rest of QEMU and makes the output harder to filter and
configure.
Replace the custom logging with standard QEMU error reporting:
- Use error_report() / error_printf() for errors
- Use warn_report() for non-fatal warnings (buffer frame size
adjustments)
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
The ALSA audio backend uses its own logging infrastructure (AUD_log,
AUD_vlog, dolog, ldebug) and a custom alsa_dump_info() debug helper.
This approach is inconsistent with the rest of QEMU and makes the
output harder to filter and configure.
Replace the custom logging with standard QEMU error reporting:
- Use error_report() / error_printf() for errors
- Use warn_report() for non-fatal warnings (invalid formats,
rejected parameters, unexpected states)
- Convert ldebug() calls and alsa_dump_info() to trace events
Remove DEBUG_ALSA and AUDIO_CAP macros which are no longer needed.
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>
Remove the separate audio_pcm_ops structure and move its function
pointers directly into AudioMixengBackendClass. This is a cleaner
QOM-style design where the PCM operations are part of the class
vtable rather than a separate indirection through hw->pcm_ops.
The HWVoiceOut and HWVoiceIn structures no longer need to store
a pcm_ops pointer, as the operations are now accessed through
the class with AUDIO_MIXENG_BACKEND_GET_CLASS().
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Move all fields from audio_driver directly into AudioMixengBackendClass,
eliminating an unnecessary extra struct. Drivers now set class
fields directly in class_init instead of creating a static audio_driver
instance.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
They are no longer used after conversion to QOM.
Also removing the drv_opaque from a few of the pcm_ops methods.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Migrate the DirectSound audio backend from the legacy driver init/fini
callbacks to proper QOM realize and finalize methods.
The dsound struct fields are now embedded directly in the AudioDsound
QOM object instead of being allocated separately as drv_opaque. This
allows accessing the backend state through proper QOM type casting
with AUDIO_DSOUND() rather than casting drv_opaque pointers.
The DirectSound and DirectSoundCapture COM objects are now managed
through the QOM lifecycle, with initialization in realize and cleanup
in finalize.
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>
Remove the legacy driver init/fini callbacks from the CoreAudio backend.
Both coreaudio_audio_init() and coreaudio_audio_fini() were no-ops that
performed no real initialization or cleanup work. Access to the
Audiodev is now through hw->s->dev instead of the drv_opaque pointer.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Mark Cave-Ayland <mark.caveayland@nutanix.com>
Reviewed-by: Akihiko Odaki <odaki@rsg.ci.i.u-tokyo.ac.jp>
Remove the legacy driver init/fini callbacks from the sndio audio backend.
Both sndio_audio_init() and sndio_audio_fini() were no-ops that
performed no real initialization or cleanup work. Access to the
Audiodev is now through hw->s->dev instead of the drv_opaque pointer.
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>
Remove the legacy driver init/fini callbacks from the JACK audio backend.
Both qjack_init() and qjack_fini() were no-ops that performed no real
initialization or cleanup work. Access to the Audiodev is now through
hw->s->dev instead of the drv_opaque pointer.
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>
Migrate the PipeWire audio backend from the legacy driver init/fini
callbacks to proper QOM realize and finalize methods.
The pwaudio struct fields are now embedded directly in the AudioPw
QOM object instead of being allocated separately as drv_opaque. This
allows accessing the backend state through proper QOM type casting
with AUDIO_PW() rather than casting drv_opaque pointers.
The PipeWire thread loop and context are now managed through the QOM
lifecycle, with initialization in realize and cleanup in finalize.
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>
Migrate the SDL audio backend from the legacy driver init/fini
callbacks to proper QOM realize and finalize methods.
The sdl_audio_init() function is replaced with audio_sdl_realize(),
which initializes the SDL audio subsystem before delegating to the
parent class realize method. The sdl_audio_fini() is replaced with
audio_sdl_finalize() to properly clean up the SDL audio subsystem.
Access to the Audiodev is now through hw->s->dev instead of the
drv_opaque pointer.
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>
Migrate the PulseAudio backend from the legacy driver init/fini
callbacks to proper QOM realize and finalize methods.
The paaudio struct fields are now embedded directly in the AudioPa
QOM object instead of being allocated separately as drv_opaque. This
allows accessing the backend state through proper QOM type casting
with AUDIO_PA() rather than casting drv_opaque pointers.
The PulseAudio connection is now managed through the QOM lifecycle,
with the connection reference acquired in realize and released in
finalize.
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>
Migrate the OSS audio backend from the legacy driver init/fini
callbacks to proper QOM realize method.
The oss_audio_init() function is replaced with audio_oss_realize(),
which performs the same device accessibility check before delegating
to the parent class realize method. The empty oss_audio_fini() is
removed.
Access to the Audiodev is now through hw->s->dev instead of the
drv_opaque pointer.
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>
Migrate the ALSA audio backend from the legacy driver init/fini
callbacks to proper QOM realize method.
The alsa_audio_init() function is replaced with audio_alsa_realize(),
which performs the same ALSA option initialization before delegating
to the parent class realize method. The empty alsa_audio_fini() is
removed.
The Audiodev pointer stored in ALSAVoiceOut and ALSAVoiceIn is removed
as it can now be accessed through hw->s->dev.
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>
Remove the legacy driver init/fini callbacks from the null audio backend.
Both no_audio_init() and no_audio_fini() were no-ops that performed
no real initialization or cleanup work.
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>
Remove the legacy driver init/fini callbacks from the WAV audio backend.
The wav_audio_init() just returned the Audiodev pointer and
wav_audio_fini() only logged a debug message. Access to the Audiodev
is now through hw->s->dev instead of the drv_opaque pointer.
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>
Migrate the SPICE audio backend from the legacy driver init/fini
callbacks to proper QOM realize method.
The spice_audio_init() function is replaced with spice_audio_realize(),
which validates that SPICE is enabled before delegating to the parent
class realize method. The empty spice_audio_fini() is removed as no
cleanup is needed.
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>
Migrate the D-Bus audio backend from the legacy driver init/fini
callbacks to proper QOM realize and finalize methods.
The DBusAudio struct fields are now embedded directly in the AudioDbus
QOM object instead of being allocated separately as drv_opaque. This
allows accessing the backend state through proper QOM type casting
with AUDIO_DBUS() rather than casting drv_opaque pointers.
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>
The next patches are going to remove usage of the driver callbacks in
favor of QOM realize/finalize.
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>
Allow to build the audio/ base classes without the
resampling/mixing/queuing 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>
Philippe Mathieu-Daudé
Daniel P. Berrangé
Marc-André Lureau