Don't call pa_stream_peek before the recording stream is ready.
Information to reproduce the problem.
Start and stop Audacity in the guest several times because the
problem is racy.
libvirt log file:
-audiodev pa,id=audio0,server=localhost,out.latency=30000,
out.mixing-engine=off,in.mixing-engine=off \
-sandbox on,obsolete=deny,elevateprivileges=deny,spawn=deny,
resourcecontrol=deny \
-msg timestamp=on
: Domain id=4 is tainted: custom-argv
char device redirected to /dev/pts/1 (label charserial0)
audio: Device pcspk: audiodev default parameter is deprecated,
please specify audiodev=audio0
audio: Device hda: audiodev default parameter is deprecated,
please specify audiodev=audio0
pulseaudio: pa_stream_peek failed
pulseaudio: Reason: Bad state
pulseaudio: pa_stream_peek failed
pulseaudio: Reason: Bad state
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
There is no guarantee a single call to pa_stream_peek every
timer_period microseconds can read a recording stream faster
than the data gets produced at the source. Let qpa_read try to
drain the recording stream.
To reproduce the problem:
Start qemu with -audiodev pa,id=audio0,in.mixing-engine=off
On the host connect the qemu recording stream to the monitor of
a hardware output device. While the problem can also be seen
with a hardware input device, it's obvious with the monitor of
a hardware output device.
In the guest start audio recording with audacity and notice the
slow recording data rate.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Every call to pa_stream_peek which returns a data length > 0
should have a corresponding pa_stream_drop. A call to qpa_read
does not necessarily call pa_stream_drop immediately after a
call to pa_stream_peek. Test in qpa_fini_in if a last
pa_stream_drop is needed.
This prevents following messages in the libvirt log file after
a recording stream gets closed and a new one opened.
pulseaudio: pa_stream_drop failed
pulseaudio: Reason: Bad state
pulseaudio: pa_stream_drop failed
pulseaudio: Reason: Bad state
To reproduce start qemu with
-audiodev pa,id=audio0,in.mixing-engine=off
and in the guest start and stop Audacity several times.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With current code audio recording with all audio backends
except PulseAudio and DirectSound is broken. The generic audio
recording buffer management forgot to update the current read
position after a read.
Fixes: ff095e5231 "audio: api for mixeng code free backends"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
Message-id: 2fc947cf-7b42-de68-3f11-cbcf1c096be9@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Which currently only means removing some checks. Old code won't require
more than two channels, but new code will need it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 7e53be1f97e939ed3bb729ef39e76b775643118a.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples). But if we want to add support for surround sound,
this no longer holds true.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This can be used to identify stream in tools like pavucontrol when one
creates multiple -audiodevs or runs multiple qemu instances.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 2d6e337c474ac84172d0809e6959c26b21d48120.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Connection name was previously erroneously set to the server socket
path, while connection names were simply "qemu". After this patch, the
connection name will be the vm name (falling back to "qemu" if not
specified), while stream names will be the audiodev's id.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3d139426031a400a68d440608ba5e43f0e116cd8.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This way we no longer need vararg functions, improving compile time
error detection. Also now it's possible to check actually what commands
are supported, without needing to manually update ctl_caps.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 2b08b3773569c5be055d0a0fb2f29ff64e79f0f4.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit removes the ad-hoc rate-limiting code from noaudio and
wavaudio, and replaces them with a (slightly modified) code from
spiceaudio. This way multiple write calls (for example when the
circular buffer wraps around) do not cause problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: fd0fe5b95b13fa26d09ae77a72f99d0ea411de14.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before
initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the
initial size of the STSampleBuffer. The next commit tries to fix this
inconsistency.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: a78caeb2eeb6348ecb45bb2c81709570ef8ac5b3.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Unfortunately, changes introduced in af2041ed2d "audio: audiodev=
parameters no longer optional when -audiodev present" breaks backward
compatibility. This patch changes the error into a deprecation warning.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 02d4328c33455742d01e0b62395013e95293c3ba.1566847960.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The code used sizeof(AudiodevAlsaPerDirectionOptions) instead of the
appropriate per direction options for the audio backend. If the size of
the actual audiodev's per direction options are larger than alsa's, it
could cause a buffer overflow.
However, alsa has three fields in per direction options: a string, an
uint32 and a bool. Oss has the same fields, coreaudio has a single
uint32, paaudio has a string and an uint32, all other backends only use
the common options, so currently no per direction options struct should
be larger than alsa's.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-Id: <7808bc816ba7da8b8de8a214713444d85f7af3c6.1566847960.git.DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
They just called audio_pcm_sw_read/write anyway, so it makes no sense
to have them too. (The noaudio's read is the only exception, but it
should work with the generic code too.)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 92ddc98133bc4b687c6e4608b9321e7b64c0e496.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Pulseaudio normally assumes that when the server wants it, the client
can generate the audio samples and send it right away. Unfortunately
this is not the case with QEMU -- it's up to the emulated system when
does it generate the samples. Buffering the samples and sending them
from a background thread is just a workaround, that doesn't work too
well. Instead enable pa's compatibility support and let pa worry about
the details.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: aa4e3613122ccbaa62b1feb4e427260731f7477c.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
audio_run is called manually by alsa and oss backends when polling.
In this case only the requesting backend should be run, not all of them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 10221fcea2028fa18d95cf531526ffe3b1d9b21a.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 303222477df6f7373217e0df768635fab5855745.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Unless we disable stream moving, pulseaudio can easily move the stream
on connect, effectively ignoring the source/sink specified by the user.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: c245929463e6e46a48b2875a150815e2ccba11b4.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This means you should probably stop using -soundhw (as it doesn't allow
you to specify any options) and add the device manually with -device.
The exception is pcspk, it's currently not possible to manually add it.
To use it with audiodev, use something like this:
-audiodev id=foo,... -global isa-pcspk.audiodev=foo -soundhw pcspk
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 9072b955acffda13976bca7b61f86d7f708c9269.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Have a pool of refcounted connections per server, so if the user creates
multiple audiodevs to the same pa server, it will use a single connection. (It
will still create different streams, so the user can manage those streams
separately in pulseaudio.)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: d43218f327c62cdbd16ea0c922612025fbc4805e.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Finally add audiodev= options to audio frontends so users can specify
which backend to use when multiple backends exist. Not specifying an
audiodev= option currently causes the first audiodev to be used, this is
fixed in the next commit.
Example usage: -audiodev pa,id=foo -device AC97,audiodev=foo
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: d64db52dda2d0e9d97bc5ab1dd9adf724280fea1.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Audio functions no longer access glob_audio_state, instead they get an
AudioState as a parameter. This is required in order to support
multiple backends.
glob_audio_state is also gone, and replaced with a tailq so we can store
more than one states.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 67aef54f9e729a7160fe95c465351115e392164b.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Remove glob_audio_state from functions, where possible without breaking
the API. This means that most static functions in audio.c now take an
AudioState pointer instead of implicitly using glob_audio_state. Also
included a pointer in SWVoice*, HWVoice* structs, so that functions
dealing them can know the audio state without having to pass it around
separately.
This is required in order to support multiple simultaneous audio
backends (added in a later commit).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: b5e241f24e795267b145bcde7c6a72dd5e6037ea.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
sysemu/sysemu.h is a rather unfocused dumping ground for stuff related
to the system-emulator. Evidence:
* It's included widely: in my "build everything" tree, changing
sysemu/sysemu.h still triggers a recompile of some 1100 out of 6600
objects (not counting tests and objects that don't depend on
qemu/osdep.h, down from 5400 due to the previous two commits).
* It pulls in more than a dozen additional headers.
Split stuff related to run state management into its own header
sysemu/runstate.h.
Touching sysemu/sysemu.h now recompiles some 850 objects. qemu/uuid.h
also drops from 1100 to 850, and qapi/qapi-types-run-state.h from 4400
to 4200. Touching new sysemu/runstate.h recompiles some 500 objects.
Since I'm touching MAINTAINERS to add sysemu/runstate.h anyway, also
add qemu/main-loop.h.
Suggested-by: Paolo Bonzini <pbonzini@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-Id: <20190812052359.30071-30-armbru@redhat.com>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
[Unbreak OS-X build]
In my "build everything" tree, changing hw/hw.h triggers a recompile
of some 2600 out of 6600 objects (not counting tests and objects that
don't depend on qemu/osdep.h).
The previous commits have left only the declaration of hw_error() in
hw/hw.h. This permits dropping most of its inclusions. Touching it
now recompiles less than 200 objects.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Alistair Francis <alistair.francis@wdc.com>
Message-Id: <20190812052359.30071-19-armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Tested-by: Philippe Mathieu-Daudé <philmd@redhat.com>
In my "build everything" tree, changing migration/vmstate.h triggers a
recompile of some 2700 out of 6600 objects (not counting tests and
objects that don't depend on qemu/osdep.h).
hw/hw.h supposedly includes it for convenience. Several other headers
include it just to get VMStateDescription. The previous commit made
that unnecessary.
Include migration/vmstate.h only where it's still needed. Touching it
now recompiles only some 1600 objects.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Alistair Francis <alistair.francis@wdc.com>
Message-Id: <20190812052359.30071-16-armbru@redhat.com>
Tested-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Several people have reported to have bag microphone lag with the PA
backend. While I cannot reproduce the problem here, it seems that their
PA somehow decides to buffer the microphone input for way too long,
causing this delay. This patch sets an upper limit to the amount of
data PA should hold. This fixes the problem reliably on their side,
while having no adverse effects on mine.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190615153852.99040-1-martin@schrodt.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
No header includes qemu-common.h after this commit, as prescribed by
qemu-common.h's file comment.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-Id: <20190523143508.25387-5-armbru@redhat.com>
[Rebased with conflicts resolved automatically, except for
include/hw/arm/xlnx-zynqmp.h hw/arm/nrf51_soc.c hw/arm/msf2-soc.c
block/qcow2-refcount.c block/qcow2-cluster.c block/qcow2-cache.c
target/arm/cpu.h target/lm32/cpu.h target/m68k/cpu.h target/mips/cpu.h
target/moxie/cpu.h target/nios2/cpu.h target/openrisc/cpu.h
target/riscv/cpu.h target/tilegx/cpu.h target/tricore/cpu.h
target/unicore32/cpu.h target/xtensa/cpu.h; bsd-user/main.c and
net/tap-bsd.c fixed up]
Currently the default audio timer frequency is 10000Hz instead of
a period of 10000us. Also the audiodev timer-period property gets
converted like a frequency. Only handling of the legacy
QEMU_AUDIO_TIMER_PERIOD environment variable is correct because
it's actually a frequency.
With this patch the property timer-period is really a timer period
and QEMU_AUDIO_TIMER_PERIOD remains a frequency.
Fixes: 71830221fb "-audiodev command line option basic implementation."
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
Message-id: 90b95e4f-39ef-2b01-da6a-857ebaee1ec5@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
We spell out sub/dir/ in sub/dir/trace-events' comments pointing to
source files. That's because when trace-events got split up, the
comments were moved verbatim.
Delete the sub/dir/ part from these comments. Gets rid of several
misspellings.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20190314180929.27722-3-armbru@redhat.com
Message-Id: <20190314180929.27722-3-armbru@redhat.com>
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
The current code does not specify the metrics of the buffers for the
input device. This makes PulseAudio choose very bad defaults, which
causes input to be unusable: Audio put in gets out 30 seconds later.
This patch fixes that and makes the latency configurable as well.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190315084653.120020-4-martin@schrodt.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The latency of a connection to the PulseAudio server is determined by
the tlength parameter. This was hardcoded to 10ms, which is a bit too
tight on my machine, causing audio on host and guest to malfunction.
A setting of 15ms works fine here. To allow tweaking, I also made the
setting configurable via the new -audiodev config. This allows to squeeze out better timings in scenarios where the emulation allows it.
I also removed setting of the minreq parameter to (seemingly arbitrary) half the latency, since it showed worse audio quality during my tests. Allowing PulseAudio to request smaller chunks helped.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190315084653.120020-3-martin@schrodt.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Audiodev configuration allows to set the length of the buffered data.
The setting was ignored and a constant value used instead.
This patch makes the code apply the setting properly, and uses the
previous default if nothing is supplied.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190315084653.120020-2-martin@schrodt.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Audio drivers now get an Audiodev * as config paramters, instead of the
global audio_option structs. There is some code in audio/audio_legacy.c
that converts the old environment variables to audiodev options (this
way backends do not have to worry about legacy options). It also
contains a replacement of -audio-help, which prints out the equivalent
-audiodev based config of the currently specified environment variables.
Note that backends are not updated and still rely on environment
variables.
Also note that (due to moving try-poll from global to backend specific
option) currently ALSA and OSS will always try poll mode, regardless of
environment variables or -audiodev options.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: e99a7cbdac0d13512743880660b2032024703e4c.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum. This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
At the end of the while-loop, either "samples" or "sdl->live" is zero, so
now that we've removed the semaphore code, the content of the while-loop
is always only executed once. Thus we can remove the while-loop now to
get rid of one indentation level here.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1549336101-17623-3-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The semaphore code was only working with SDL1.2 - with SDL2, it causes
a deadlock. Since we've removed support for SDL1.2 recently, we can
now completely remove the semaphore code from sdlaudio.c.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 1549336101-17623-2-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
audio_calloc uses g_malloc0 which never returns in case of
memory failure.
Signed-off-by: Frediano Ziglio <fziglio@redhat.com>
Message-id: 20190225154335.11397-2-fziglio@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Instead of using lot of low level function and manually allocate
the temporary string in audio_process_options use more high
level GLib function. The function is not used in hot path but to
read some initial setting.
Signed-off-by: Frediano Ziglio <fziglio@redhat.com>
Message-id: 20190225154335.11397-1-fziglio@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Kill off a pile of monitor_printf's and cur_mon usage.
The only one left in wavcapture.c is the info case.
Signed-off-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Reviewed-by: Gerd Hoffmann <kraxel@gmail.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Message-Id: <20170320173840.3626-3-dgilbert@redhat.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
Only print a message about the failed driver initialization in case it
was the driver explicitly requested by the user via QEMU_AUDIO_DRV=$drv.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20190124112055.547-6-kraxel@redhat.com
Check whenever the pulseaudio daemon pidfile is present before trying to
initialize the pulseaudio backend. Just return NULL if that is not the
case, so qemu will check the next backend in line.
In case the user explicitly configured a non-default pulseaudio server
skip the check.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20190124112055.547-5-kraxel@redhat.com
Files requiring AudioState already include "audio_int.h".
To clean "qemu/typedefs.h", move the declaration to "audio_int.h"
(removing the forward declaration).
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Signed-off-by: Thomas Huth <thuth@redhat.com>
Both GCC v4.8 and Clang v3.4 support the -Waddress option, so we do
not need the compiler version check here anymore.
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Signed-off-by: Thomas Huth <thuth@redhat.com>
The rate of pulseaudio absorbing the audio stream is used to control the
the rate of the guests audio stream. When the emulated hardware uses
small chunks (like intel-hda does) we need small chunks on the audio
backend side too, otherwise that feedback loop doesn't work very well.
Cc: Max Ehrlich <maxehr@umiacs.umd.edu>
Cc: Martin Schrodt <martin@schrodt.org>
Buglink: https://bugs.launchpad.net/bugs/1795527
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20181109142032.1628-1-kraxel@redhat.com
Make audio_driver_lookup() try load the module in case it doesn't find
the driver in the registry. Also load all modules for -audio-help, so
the help output includes the help text for modular audio drivers.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20180306074053.22856-3-kraxel@redhat.com
Add registry for audio drivers, using the existing audio_driver struct.
Make all drivers register themself. The old list of audio_driver struct
pointers is now a list of audio driver names, specifying the priority
(aka probe order) in case no driver is explicitly asked for.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20180306074053.22856-2-kraxel@redhat.com
This avoids a name clash for CONFIG_SDL, which is used by both sdl video
support and sdl audio support. It also more clear that this is a audio
driver configuration.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20180301100547.18962-13-kraxel@redhat.com
This cleanup makes the number of objects depending on qapi/error.h
drop from 1910 (out of 4743) to 1612 in my "build everything" tree.
While there, separate #include from file comment with a blank line,
and drop a useless comment on why qemu/osdep.h is included first.
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-Id: <20180201111846.21846-5-armbru@redhat.com>
[Semantic conflict with commit 34e304e975 resolved, OSX breakage fixed]
Apparently we don't use __MSC_VER as a compiler anymore and we always
require a C99 compiler (which means we always have __func__) so we don't
need a special AUDIO_FUNC macro. We can just replace AUDIO_FUNC with
__func__ instead.
Checkpatch failures were manually fixed.
Signed-off-by: Alistair Francis <alistair.francis@xilinx.com>
Cc: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20180203084315.20497-2-armbru@redhat.com>
The point of writing a macro embedded in a 'do { ... } while (0)'
loop (particularly if the macro has multiple statements or would
otherwise end with an 'if' statement) is so that the macro can be
used as a drop-in statement with the caller supplying the
trailing ';'. Although our coding style frowns on brace-less 'if':
if (cond)
statement;
else
something else;
that is the classic case where failure to use do/while(0) wrapping
would cause the 'else' to pair with any embedded 'if' in the macro
rather than the intended outer 'if'. But conversely, if the macro
includes an embedded ';', then the same brace-less coding style
would now have two statements, making the 'else' a syntax error
rather than pairing with the outer 'if'. Thus, even though our
coding style with required braces is not impacted, ending a macro
with ';' makes our code harder to port to projects that use
brace-less styles.
The change should have no semantic impact. I was not able to
fully compile-test all of the changes (as some of them are
examples of the ugly bit-rotting debug print statements that are
completely elided by default, and I didn't want to recompile
with the necessary -D witnesses - cleaning those up is left as a
bite-sized task for another day); I did, however, audit that for
all files touched, all callers of the changed macros DID supply
a trailing ';' at the callsite, and did not appear to be used
as part of a brace-less conditional.
Found mechanically via: $ git grep -B1 'while (0);' | grep -A1 \\\\
Signed-off-by: Eric Blake <eblake@redhat.com>
Acked-by: Cornelia Huck <cohuck@redhat.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Acked-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Message-Id: <20171201232433.25193-7-eblake@redhat.com>
Reviewed-by: Juan Quintela <quintela@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
In trace format '#' flag of printf is forbidden. Fix it to '0x%'.
This patch is created by the following:
check that we have a problem
> find . -name trace-events | xargs grep '%#' | wc -l
56
check that there are no cases with additional printf flags before '#'
> find . -name trace-events | xargs grep "%[-+ 0'I]+#" | wc -l
0
check that there are no wrong usage of '#' and '0x' together
> find . -name trace-events | xargs grep '0x%#' | wc -l
0
fix the problem
> find . -name trace-events | xargs sed -i 's/%#/0x%/g'
[Eric Blake noted that xargs grep '%[-+ 0'I]+#' should be xargs grep
"%[-+ 0'I]+#" instead so the shell quoting is correct.
--Stefan]
Signed-off-by: Vladimir Sementsov-Ogievskiy <vsementsov@virtuozzo.com>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Message-id: 20170731160135.12101-3-vsementsov@virtuozzo.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
With the move of some docs/ to docs/devel/ on ac06724a71,
no references were updated.
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
If a voice recording equipment is opened for a long time(several days)
in windows guest, rate->ipos will overflow and rate->opos will never
have a chance to change. It will result to a infinite loop.
Signed-off-by: Peng Hao <peng.hao2@zte.com.cn>
Signed-off-by: Wang Yechao <wang.yechao255@zte.com.cn>
Message-id: 1500128061-20849-1-git-send-email-peng.hao2@zte.com.cn
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
When compiling with SDL2, the semaphore trick used in sdlaudio.c
does not work - QEMU locks up completely in this case. To avoid
the hang and get at least some audio playback up and running (it's
a little bit crackling, but better than nothing), we can use the
SDL locking functions SDL_LockAudio() and SDL_UnlockAudio() to sync
with the sound playback thread instead.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1485852398-2327-1-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch changes resetting strategy of the audio polling timer.
It does not change expiration time if the timer is already set.
This patch is needed to make this timer deterministic and to use execution
record/replay for audio devices.
audio_reset_timer is used in the function audio_vm_change_state_handler.
Therefore every time VM is stopped or restarted the timer will be reset
to new timeout. Virtual clock does not proceed while VM is stopped.
Therefore there is no need in resetting the timeout when VM restarts.
v2: updated commit message
v3: now using timer_mod_anticipate function (as suggested by Yurii Zubrytskyi)
Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170214071510.6112.76764.stgit@PASHA-ISP
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch adds recording and replaying audio data. Is saves synchronization
information for audio out and inputs from the microphone.
v2: removed unneeded whitespace change
Signed-off-by: Pavel Dovgalyuk <pavel.dovgaluk@ispras.ru>
Message-id: 20170202055054.4848.94901.stgit@PASHA-ISP.lan02.inno
[ kraxel: add qemu/error-report.h include to fix osx build failure ]
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
