When building with GCC9 using CFLAG -Wimplicit-fallthrough=2 we get:
audio/audio.c: In function ‘audio_pcm_init_info’:
audio/audio.c:306:14: error: this statement may fall through [-Werror=implicit-fallthrough=]
306 | sign = 1;
| ~~~~~^~~
audio/audio.c:307:5: note: here
307 | case AUDIO_FORMAT_U8:
| ^~~~
cc1: all warnings being treated as errors
Similarly to e46349414, add the missing fall through comment to
hint GCC.
Fixes: 2b9cce8c8c
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Aleksandar Markovic <amarkovic@wavecomp.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20191218192526.13845-2-philmd@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Tell the compiler to do a 32bit * 32bit -> 64bit multiplication
because period_ticks is a 64bit variable. The overflow occurs
for audio timer periods larger than 4294967us.
Fixes: be1092afa0 "audio: fix audio timer rate conversion bug"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 8893a235-66a8-8fbe-7d95-862e29da90b1@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
