Every emulated audio device has a way to enable audio playback. Don't
start playback until the guest enables the audio device to keep the
Core Audio device run state in sync with hw->enabled.
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20201213130528.5863-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
While the variable once was used to fake audio settings, since
commit ed2a4a7941 "audio: proper support for float samples in
mixeng" this is no longer true. Rename the variable to obt_as.
This is the same naming scheme as in audio/sdlaudio.c
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20201213130528.5863-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This change registers a bottom handler to close the JACK client
connection when a server shutdown signal is received. Without this
libjack2 attempts to "clean up" old clients and causes a use after free
segfault.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20201108063351.35804-2-geoff@hostfission.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Used for files which (with CONFIG_SPICE=y) depend on spice header files
to pick up some enum, but which do not depend on on the actual spice
shared library.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20201014121120.13482-6-kraxel@redhat.com
cur_mon really needs to be coroutine-local as soon as we move monitor
command handlers to coroutines and let them yield. As a first step, just
remove all direct accesses to cur_mon so that we can implement this in
the getter function later.
Signed-off-by: Kevin Wolf <kwolf@redhat.com>
Message-Id: <20201005155855.256490-4-kwolf@redhat.com>
Reviewed-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
clang's C11 atomic_fetch_*() functions only take a C11 atomic type
pointer argument. QEMU uses direct types (int, etc) and this causes a
compiler error when a QEMU code calls these functions in a source file
that also included <stdatomic.h> via a system header file:
$ CC=clang CXX=clang++ ./configure ... && make
../util/async.c:79:17: error: address argument to atomic operation must be a pointer to _Atomic type ('unsigned int *' invalid)
Avoid using atomic_*() names in QEMU's atomic.h since that namespace is
used by <stdatomic.h>. Prefix QEMU's APIs with 'q' so that atomic.h
and <stdatomic.h> can co-exist. I checked /usr/include on my machine and
searched GitHub for existing "qatomic_" users but there seem to be none.
This patch was generated using:
$ git grep -h -o '\<atomic\(64\)\?_[a-z0-9_]\+' include/qemu/atomic.h | \
sort -u >/tmp/changed_identifiers
$ for identifier in $(</tmp/changed_identifiers); do
sed -i "s%\<$identifier\>%q$identifier%g" \
$(git grep -I -l "\<$identifier\>")
done
I manually fixed line-wrap issues and misaligned rST tables.
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Acked-by: Paolo Bonzini <pbonzini@redhat.com>
Message-Id: <20200923105646.47864-1-stefanha@redhat.com>
Handle the spice special case in audio_init instead.
With the qemu_spice_audio_init() symbol dependency being
gone we can build spiceaudio as module.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20200916084117.21828-2-kraxel@redhat.com
Run the downstream playback queue even if there are no samples
in the mixing engine buffer. The downstream queue may still have
queued samples.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-7-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The function audio_generic_write should work exactly like
audio_pcm_hw_run_out. It's a very similar function working on a
different buffer.
This patch significantly reduces the number of drop-outs with
the DirectSound backend. To hear the difference start qemu with
-audiodev dsound,id=audio0,out.mixing-engine=off and play a
song in the guest with and without this patch.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-6-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch removes unnecessary calls to the pcm_ops function
put_buffer_in(). No audio backend needs this call if the
returned length of pcm_ops function get_buffer_in() is zero.
For the DirectSound backend this prevents a call to
dsound_unlock_in() without a preceding call to dsound_lock_in().
While Windows doesn't complain it seems wrong anyway.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The function audio_generic_read should work exactly like
audio_pcm_hw_run_in. It's a very similar function working
on a different buffer.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The playback rate with the spiceaudio backend is currently too
fast if there's no spice client connected or the spice client
can't play audio. Rate limit the audio playback stream in all
cases. To calculate the rate correctly the limiter has to know
the maximum buffer size.
Fixes: 8c198ff065 ("spiceaudio: port to the new audio backend api")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch allows the audio backends get_buffer_out() functions
to drop audio data and mitigates a bug reported on the qemu-devel
mailing list.
https://lists.nongnu.org/archive/html/qemu-devel/2020-09/msg03832.html
The new rules for the variables buf and size returned by
get_buffer_out() are:
size == 0: Downstream playback buffer is full. Retry later.
size > 0, buf != NULL: Copy size bytes to buf for playback.
size > 0, buf == NULL: Drop size bytes.
The audio playback rate with spiceaudio for the no audio case is
too fast, but that's what we had before commit fb35c2cec5
"audio/dsound: fix invalid parameters error". The complete fix
comes with the next patch.
Reported-by: Qi Zhou <atmgnd@outlook.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With the next patch all audio backends put_buffer_out() functions
have to handle the buf == NULL case, provided the get_buffer_out()
function may return buf = NULL and size > 0.
It turns out that all audio backends get_buffer_out() functions
either can't return buf = NULL or return buf = NULL and size = 0
at the same time. The only exception is the spiceaudio backend
where size may be uninitialized.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
I found that there are many spelling errors in the comments of qemu,
so I used the spellcheck tool to check the spelling errors
and finally found some spelling errors in the folder.
Signed-off-by: zhaolichang <zhaolichang@huawei.com>
Reviewed-by: Alex Bennee <alex.bennee@linaro.org>
Message-Id: <20200917075029.313-2-zhaolichang@huawei.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
Tracked down with the help of scripts/cleanup-trace-events.pl.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-id: 20200806141334.3646302-4-armbru@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Meson doesn't enjoy the same flexibility we have with Make in choosing
the include path. In particular the tracing headers are using
$(build_root)/$(<D).
In order to keep the include directives unchanged,
the simplest solution is to generate headers with patterns like
"trace/trace-audio.h" and place forwarding headers in the source tree
such that for example "audio/trace.h" includes "trace/trace-audio.h".
This patch is too ugly to be applied to the Makefiles now. It's only
a way to separate the changes to the tracing header files from the
Meson rewrite of the tracing logic.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
In function oss_read() a read error currently does not exit the
read loop. With no data to read the variable pos will quickly
underflow and a subsequent successful read overwrites memory
outside the buffer. This patch adds the missing break statement
to the error path of the function.
To reproduce start qemu with -audiodev oss,id=audio0 and in the
guest start audio recording. After some time this will trigger
an exception.
Fixes: 3ba4066d08 "ossaudio: port to the new audio backend api"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200707180836.5435-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
See recent commit "error: Document Error API usage rules" for
rationale.
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Vladimir Sementsov-Ogievskiy <vsementsov@virtuozzo.com>
Message-Id: <20200707160613.848843-18-armbru@redhat.com>
Instead of checking for the audodev state in each code path, centralize
the check into the initialize function itself to make it safe to call it
at any time.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-7-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
When the guest closes the audio device we must start dropping input
samples from JACK and zeroing the output buffer samples. Failure to do
so causes sound artifacts during operations such as guest OS reboot, and
causes a hang of the input pipeline breaking it until QEMU is restated.
Closing and reconnecting to JACK was tested during these enable/disable
calls which works well for Linux guests, however Windows re-opens the
audio hardware repeatedly even when doing simple tasks like playing a
system sounds. As such it was decided it is better to feed silence to
JACK while the device is disabled.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-6-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This fixes a hang when there is a communications issue with the JACK
server. Simply closing the connection is enough to completely clean up
and as such we do not need to remove the ports first. As JACK uses a
socket based protocol that relies on the `select` call, if there is a
communication breakdown with the server the client library waits
forever for a response to the unregister request.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-5-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Initial code for JACK did not support audio input and as such this
boolean was set to let QEMU know, however JACK ended up including input
support making this invalid. Further investigation shows it was invalid
to set it in the first instance anyway due to a failure on my part
understand properly what this was for when the audodev was initially
developed.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-4-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
JACK does not provide us with the configured buffer size until after
activiation which was overriding this minimum value. JACK itself doesn't
have this minimum limitation, but the QEMU virtual hardware and as such
it must be enforced, failure to do so results in audio discontinuities.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-2-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The buffer is the captured input to pass to backends.
As we should not modify it, mark the argument const.
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-3-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The samples are the input to convert to u64. As we should
not modify them, mark the argument const.
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-2-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit 571a8c522e caused the HMP wavcapture command to segfault when
processing audio data in audio_pcm_sw_write(), where a NULL
sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is
valid before dereferincing it. A similar fix is also made in the
parallel function audio_pcm_sw_read().
Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and
volume_*)
Signed-off-by: Bruce Rogers <brogers@suse.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20200521172931.121903-1-brogers@suse.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The code in CONV_NATURAL_FLOAT() and CLIP_NATURAL_FLOAT()
seems to use the constant 2^31-0.5 to convert float to integer
and back. But the float type lacks the required precision and
the constant used for the conversion is 2^31. This is equiva-
lent to a [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] mapping.
This patch explicitly writes down the used constant. The
compiler generated code doesn't change.
The constant 2^31 has an exact float representation and the
clang 10 compiler stops complaining about an implicit int to
float conversion with a changed value.
A few notes:
- The conversion of 1.f to INT32_MAX + 1 doesn't overflow. The
type of the destination variable is int64_t.
- At a later stage one of the clip_* functions in
audio/mixeng_template.h limits INT32_MAX + 1 to the integer
range.
- The clip_natural_float_* functions in audio/mixeng.c convert
INT32_MAX and INT32_MAX + 1 to 1.f.
Buglink: https://bugs.launchpad.net/bugs/1878627
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200523201712.23908-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit adds a new audiodev backend to allow QEMU to use JACK as
both an audio sink and source.
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-Id: <20200512101603.E3DB73A038E@moya.office.hostfission.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch prevents an underflow of variable samples in function
audio_pcm_hw_run_in(). See commit 599eac4e5a "audio:
audio_generic_get_buffer_in should honor *size". This time the
while loop in audio_pcm_hw_run_in() will terminate nevertheless,
because it seems the recording stream in Windows is always rate
limited.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
IDirectSoundCaptureBuffer_Lock() fails on Windows when called
with len = 0. Return early from dsound_get_buffer_in() in this
case.
To reproduce the warning start a linux guest. In the guest
start Audacity and you will see a lot of "Could not lock
capture buffer" warnings.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently the DirectSound backend fails to stop audio playback
in dsound_enable_out(). To detect a lost buffer condition
dsound_get_status_out() incorrectly uses the error code
DSERR_BUFFERLOST instead of flag DSBSTATUS_BUFFERLOST as a mask
and returns with an error. As a result dsound_enable_out()
returns early and doesn't stop playback.
To reproduce the bug start qemu on a Windows host with
-soundhw pcspk -audiodev dsound,id=audio0. On the guest
FreeDOS 1.2 command line enter beep. The image Day 1 - F-Bird
from the QEMU Advent Calendar 2018 shows the bug as well.
Buglink: https://bugs.launchpad.net/qemu/+bug/1699628
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The current positive limit for the saturation nonlinearity is
only correct if the type of the result has 8 bits or less.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently the internal float range of the mixing engine is
[-.5f, .5f]. PulseAudio, SDL2 and libasound use a [-1.f, 1.f]
range. This means with float samples the audio playback volume
is 6dB too low and audio recording signals will be clipped in
most cases.
To avoid another scaling factor in the conv_natural_float_* and
clip_natural_float_* functions with FLOAT_MIXENG defined this
patch changes the mixing engine float range to [-1.f, 1.f].
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Change the clip_natural_float_from_mono() function in
audio/mixeng.c to be consistent with the clip_*_from_mono()
functions in audio/mixeng_template.h.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch changes the naming scheme of the FLOAT_CONV_TO and
FLOAT_CONV_FROM macros to the scheme used in mixeng_template.h.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Fixes: 286a5d201e
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Acked-by: Paolo Bonzini <pbonzini@redhat.com>
Reviewed-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Reviewed-by: Juan Quintela <quintela@redhat.com>
Message-Id: <20200218094402.26625-3-philmd@redhat.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
This adds proper support for float samples in mixeng by adding a new
audio format for it.
Limitations: only native endianness is supported. None of the virtual
sound cards support float samples (it looks like most of them only
support 8 and 16 bit, only hda supports 32 bit), it is only used for the
audio backends (i.e. host side).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
There are reports that since commit 2ceb8240fa "coreaudio: port
to the new audio backend api" audio playback with CoreAudio is
broken. This patch reverts some parts the commit.
Because of changes in the audio subsystem the audio clip
function in v4.1.0 of coreaudio.c had to be moved to mixeng.c
and the generic buffer management code needed a hint about the
size of the float type.
This patch is based on a patch from Zoltán Kővágó found at
https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg02142.html.
Fixes: 2ceb8240fa "coreaudio: port to the new audio backend api"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200202140641.4737-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Windows (unlike wine) bails out when IDirectSoundBuffer8::Lock is called
with zero length. Also, hw->pos_emul handling was incorrect when
calling this function for the first time.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reported-by: KJ Liew <liewkj@yahoo.com>
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Message-id: fe9744216d9d421a2dbb09bcf5fa0dbd18f77ac5.1580684275.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The function generic_get_buffer_in currently ignores the *size
parameter and may return a buffer larger than *size.
As a result the variable samples in function
audio_pcm_hw_run_in may underflow. The while loop then most
likely will never termiate.
Buglink: http://bugs.debian.org/948658
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently there is no way to disable poll mode in
oss_enable_out and oss_enable_in when it was enabled before.
The enable code path always resets the poll mode state variable.
Fixes: b027a538c6 "oss: Remove unused error handling of qemu_set_fd_handler"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With audiodev parameter out.mixing-engine=off hw->mix_buf is
NULL. This patch reverts a small part of dc88e38fa7 "audio:
unify input and output mixeng buffer management".
To reproduce the problem start qemu with
-audiodev oss,id=audio0,try-mmap=on,out.mixing-engine=off
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The combined generic buffer management code and buffer run out
code in function audio_generic_put_buffer_out has a problematic
behaviour. A few hundred milliseconds after playback starts the
mixing buffer and the generic buffer are nearly full and the
following pattern can be seen.
On first call of audio_pcm_hw_run_out the buffer run code in
audio_generic_put_buffer_out writes some data to the audio
hardware but the generic buffer will fill faster and is full
when audio_pcm_hw_run_out returns. This is because emulated
audio devices can produce playback data at a higher rate than
the audio backend hardware consumes this data.
On next call of audio_pcm_hw_run_out the buffer run code in
audio_generic_put_buffer_out writes some data to the audio
hardware but no audio data is transferred to the generic buffer
because the buffer is already full.
Then the pattern repeats. For the emulated audio device this
looks like the audio timer period has doubled.
This patch splits the combined generic buffer management code
and buffer run out code and calls the buffer run out code after
buffer management code to break this pattern.
The bug report is for the wav audio backend. But the problem is
not limited to this backend. All audio backends which use the
audio_generic_put_buffer_out function show this problem.
Buglink: https://bugs.launchpad.net/qemu/+bug/1858488
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With audiodev parameter out.mixing-engine=off hw->mix_buf is
NULL. This leads to a segmentation fault in
AUD_get_buffer_size_out. This patch reverts a small part of
dc88e38fa7 "audio: unify input and output mixeng buffer
management".
To reproduce the problem start qemu with
-soundhw adlib -audiodev pa,id=audio0,out.mixing-engine=off
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
