Handle output IO messages in the transmit (TX) virtqueue.
It allocates a VirtIOSoundPCMBuffer for each IO message and copies the
data buffer to it. When the IO buffer is written to the host's sound
card, the guest will be notified that it has been consumed.
The lifetime of an IO message is:
1. Guest sends IO message to TX virtqueue.
2. QEMU adds it to the appropriate stream's IO buffer queue.
3. Sometime later, the host audio backend calls the output callback,
virtio_snd_pcm_out_cb(), which is defined with an AUD_open_out()
call. The callback gets an available number of bytes the backend can
receive. Then it writes data from the IO buffer queue to the backend.
If at any time a buffer is exhausted, it is returned to the guest as
completed.
4. If the guest releases the stream, its buffer queue is flushed by
attempting to write any leftover data to the audio backend and
releasing all IO messages back to the guest. This is how according to
the spec the guest knows the release was successful.
Based-on: 5a2f350eec
Signed-off-by: Igor Skalkin <Igor.Skalkin@opensynergy.com>
Signed-off-by: Anton Yakovlev <Anton.Yakovlev@opensynergy.com>
Signed-off-by: Manos Pitsidianakis <manos.pitsidianakis@linaro.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Tested-by: Alex Bennée <alex.bennee@linaro.org>
Message-Id: <b7c6fc458c763d09a4abbcb620ae9b220afa5b8f.1698062525.git.manos.pitsidianakis@linaro.org>
Acked-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
This patch adds a PCI wrapper device for the virtio-sound device.
It is necessary to instantiate a virtio-snd device in a guest.
All sound logic will be added to the virtio-snd device in the following
commits.
To add this device with a guest, you'll need a >=5.13 kernel compiled
with CONFIG_SND_VIRTIO=y, which at the time of writing most distros have
off by default.
Use with following flags in the invocation:
Pulseaudio:
-audio driver=pa,model=virtio
or
-audio driver=pa,model=virtio,server=/run/user/1000/pulse/native
sdl:
-audio driver=sdl,model=virtio
coreaudio (macos/darwin):
-audio driver=coreaudio,model=virtio
etc.
Based-on: 5a2f350eec
Signed-off-by: Igor Skalkin <Igor.Skalkin@opensynergy.com>
Signed-off-by: Anton Yakovlev <Anton.Yakovlev@opensynergy.com>
Signed-off-by: Manos Pitsidianakis <manos.pitsidianakis@linaro.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Tested-by: Alex Bennée <alex.bennee@linaro.org>
Message-Id: <b223598d59f56ead6a6d8d9bb6801e17489ddaa4.1698062525.git.manos.pitsidianakis@linaro.org>
Acked-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
Reset the current sample counter when writing the Channel Sample
Count Register. The Linux ens1370 driver and the AROS sb128
driver expect the current sample counter counts down from sample
count to 0 after a write to the Channel Sample Count Register.
Currently the current sample counter starts from 0 after a reset
or the last count when the counter was stopped.
The current sample counter is used to raise an interrupt whenever
a complete buffer was transferred. When the counter starts with a
value lower than the reload value, the interrupt triggeres before
the buffer was completly transferred. This may lead to corrupted
audio streams.
Tested-by: Rene Engel <ReneEngel80@emailn.de>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Tested-by: BALATON Zoltan <balaton@eik.bme.hu>
Message-Id: <20230917065813.6692-1-vr_qemu@t-online.de>
Pull request q800 20231008
add support for booting:
- MacOS 7.1 - 8.1, with or without virtual memory enabled
- A/UX 3.0.1
- NetBSD 9.3
- Linux (via EMILE)
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# gpg: Signature made Sun 08 Oct 2023 02:22:42 EDT
# gpg: using RSA key CD2F75DDC8E3A4DC2E4F5173F30C38BD3F2FBE3C
# gpg: issuer "laurent@vivier.eu"
# gpg: Good signature from "Laurent Vivier <lvivier@redhat.com>" [full]
# gpg: aka "Laurent Vivier <laurent@vivier.eu>" [full]
# gpg: aka "Laurent Vivier (Red Hat) <lvivier@redhat.com>" [full]
# Primary key fingerprint: CD2F 75DD C8E3 A4DC 2E4F 5173 F30C 38BD 3F2F BE3C
* tag 'q800-for-8.2-pull-request' of https://github.com/vivier/qemu-m68k:
mac_via: extend timer calibration hack to work with A/UX
q800: add alias for MacOS toolbox ROM at 0x40000000
q800: add ESCC alias at 0xc000
mac_via: always clear ADB interrupt when switching to A/UX mode
mac_via: implement ADB_STATE_IDLE state if shift register in input mode
mac_via: workaround NetBSD ADB bus enumeration issue
mac_via: work around underflow in TimeDBRA timing loop in SETUPTIMEK
swim: update IWM/ISM register block decoding
swim: split into separate IWM and ISM register blocks
swim: add trace events for IWM and ISM registers
q800: add easc bool machine class property to switch between ASC and EASC
q800: add Apple Sound Chip (ASC) audio to machine
asc: generate silence if FIFO empty but engine still running
audio: add Apple Sound Chip (ASC) emulation
q800: allow accesses to RAM area even if less memory is available
q800: add IOSB subsystem
q800: implement additional machine id bits on VIA1 port A
q800: add machine id register
q800: add djMEMC memory controller
q800-glue.c: convert to Resettable interface
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
MacOS (un)helpfully leaves the FIFO engine running even when all the samples have
been written to the hardware, and expects the FIFO status flags and IRQ to be
updated continuously.
There is an additional problem in that not all audio backends guarantee an
all-zero output when there is no FIFO data available, in particular the Windows
dsound backend which re-uses its internal circular buffer causing the last played
sound to loop indefinitely.
Whilst this is effectively a bug in the Windows dsound backend, work around it
for now using a simple heuristic: if the FIFO remains empty for half a cycle
(~23ms) then continuously fill the generated buffer with empty silence.
Signed-off-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Laurent Vivier <laurent@vivier.eu>
Message-ID: <20231004083806.757242-9-mark.cave-ayland@ilande.co.uk>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
The Apple Sound Chip was primarily used by the Macintosh II to generate sound
in hardware which was previously handled by the toolbox ROM with software
interrupts.
Implement both the standard ASC and also the enhanced ASC (EASC) functionality
which is used in the Quadra 800.
Note that whilst real ASC hardware uses AUDIO_FORMAT_S8, this implementation uses
AUDIO_FORMAT_U8 instead because AUDIO_FORMAT_S8 is rarely used and not supported
by some audio backends like PulseAudio and DirectSound when played directly with
-audiodev out.mixing-engine=off.
Co-developed-by: Laurent Vivier <laurent@vivier.eu>
Co-developed-by: Volker Rümelin <vr_qemu@t-online.de>
Signed-off-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Message-ID: <20231004083806.757242-8-mark.cave-ayland@ilande.co.uk>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
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>
We use the user_ss[] array to hold the user emulation sources,
and the softmmu_ss[] array to hold the system emulation ones.
Hold the latter in the 'system_ss[]' array for parity with user
emulation.
Mechanical change doing:
$ sed -i -e s/softmmu_ss/system_ss/g $(git grep -l softmmu_ss)
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Message-Id: <20230613133347.82210-10-philmd@linaro.org>
Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
isa_get_dma() returns a DMA channel handler from an ISABus.
To emphasize this, rename it as isa_bus_get_dma().
Mechanical change using:
$ sed -i -e 's/isa_get_dma/isa_bus_get_dma/g' \
$(git grep -l isa_get_dma)
Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-Id: <20230215161641.32663-2-philmd@linaro.org>
PCIDeviceClass and PCIDevice are defined in pci.h. Many users of the
header don't actually need them. Similar structs live in their own
headers: PCIBusClass and PCIBus in pci_bus.h, PCIBridge in
pci_bridge.h, PCIHostBridgeClass and PCIHostState in pci_host.h,
PCIExpressHost in pcie_host.h, and PCIERootPortClass, PCIEPort, and
PCIESlot in pcie_port.h.
Move PCIDeviceClass and PCIDeviceClass to new pci_device.h, along with
the code that needs them. Adjust include directives.
This also enables the next commit.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-Id: <20221222100330.380143-6-armbru@redhat.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
Currently the intel-hda device has a reset method which manually
resets all the codecs by calling device_legacy_reset() on them. This
means they get reset twice, once because child devices on a qbus get
reset before the parent device's reset method is called, and then
again because we're manually resetting them.
Drop the manual reset call, and ensure that codecs are still reset
when the guest does a reset via ICH6_GCTL_RESET by using
device_cold_reset() (which resets all the devices on the qbus as well
as the device itself) instead of a direct call to the reset function.
This is a slight ordering change because the (only) codec reset now
happens before the controller registers etc are reset, rather than
once before and then once after, but the codec reset function
hda_audio_reset() doesn't care.
This lets us drop a use of device_legacy_reset(), which is
deprecated.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Message-Id: <20221014142632.2092404-2-peter.maydell@linaro.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
-audio is used like "-audio pa,model=sb16". It is almost as simple as
-soundhw, but it reuses the -audiodev parsing machinery and attaches an
audiodev to the newly-created device. The main 'feature' is that
it knows about adding the codec device for model=intel-hda, and adding
the audiodev to the codec device.
In the future, it could be extended to support default models or
builtin devices, just like -nic, or even a default backend. For now,
keep it simple.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
The usefulness of enabling a dozen soundcards is dubious. Simplify the
code by allowing a single instance of -soundhw, with no support for
parsing either comma-separated values or 'soundhw all'.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
The pcspk device is the only user of the init_isa function, and the only
-soundhw option which does not create a new device (it hacks into the
PCSpkState by hand). Remove it, since it was deprecated.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
g_new(T, n) is neater than g_malloc(sizeof(T) * n). It's also safer,
for two reasons. One, it catches multiplication overflowing size_t.
Two, it returns T * rather than void *, which lets the compiler catch
more type errors.
This commit only touches allocations with size arguments of the form
sizeof(T).
Patch created mechanically with:
$ spatch --in-place --sp-file scripts/coccinelle/use-g_new-etc.cocci \
--macro-file scripts/cocci-macro-file.h FILES...
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Cédric Le Goater <clg@kaod.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Acked-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Message-Id: <20220315144156.1595462-4-armbru@redhat.com>
Reviewed-by: Pavel Dovgalyuk <Pavel.Dovgalyuk@ispras.ru>
