Age | Commit message (Collapse) | Author |
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Delete spaces between function name and open parenthesis'('
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-8-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Fix code indent.
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-7-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Use '0x' prefix instead of '%#'
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-6-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Fix the line width of code.
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-5-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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transfer "foo* " to "foo *"
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-4-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Fix problems about spaces:
-operator needs spaces around it, add them.
-somespaces are redundant, remove them.
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-3-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Fix problems about braces:
-braces are necessary for all arms of if/for/while statements
-else should follow close brace '}'
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-2-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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There is a mismatch between message and used argument. Change
the argument from frequency to format.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-23-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Enable the f32 audio sample format for the DirectSound backend.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-22-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Rename dsound_open() to dsound_set_cooperative_level(). The
only task of that function is to set the cooperative level for
DirectSound.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-21-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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GetForegroundWindow() doesn't necessarily return the own window
handle. It just returns a handle to the currently active window
and can even return NULL. At the time dsound_open() gets called
the active window is most likely the shell window and not the
QEMU window.
Replace GetForegroundWindow() with GetDesktopWindow() which
always returns a valid window handle, and at the same time
replace the DirectSound buffer flag DSBCAPS_STICKYFOCUS with
DSBCAPS_GLOBALFOCUS where Windows only expects a valid window
handle for DirectSound function SetCooperativeLevel(). The
Microsoft online docs for IDirectSound::SetCooperativeLevel
recommend this in the remarks.
This fixes a bug where you can't hear sound from the guest.
To reproduce start qemu with -machine pcspk-audiodev=audio0
-device intel-hda -device hda-duplex,audiodev=audio0
-audiodev dsound,id=audio0,out.mixing-engine=off
from a shell and start audio playback with the hda device in the
guest. The guest will be silent. To hear guest audio you have to
activate the shell window once.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-20-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Tell PulseAudio to send recorded audio data in smaller chunks
than timer_period, so there's a good chance that qemu can read
recorded audio data every time it looks for new data.
PulseAudio tries to send buffer updates at a fragsize / 2 rate.
With fragsize = timer_period / 2 * 3 the update rate is 75% of
timer_period. The lower limit for the recording buffer size
maxlength is fragsize * 2.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-19-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Currently with the playback buffer attribute minreq = -1 and flag
PA_STREAM_EARLY_REQUESTS PulseAudio uses minreq = tlength / 4.
To improve audio playback with larger PulseAudio server side
buffers, limit minreq to a maximum of 75% of audio timer_rate.
That way there is a good chance qemu receives a stream buffer
size update before it tries to write data to the playback stream.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-18-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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The audio buffer size in audio/paaudio.c is typically larger
than expected. Just comment the bugs in qpa_init_in() and
qpa_init_out() for now. Fixing these bugs may break glitch free
audio playback with fine tuned user audio settings.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-17-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Commit baea032ec7 "audio/paaudio: fix ignored buffer_length setting"
added code to handle buffer_length defaults. This was unnecessary
because the audio_buffer_* functions in audio/audio.c already handle
this. Remove the unneeded code.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-16-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Don't call pa_stream_writable_size() in qpa_get_buffer_out()
before the playback stream is ready. This prevents a lot of the
following pulseaudio error messages.
pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state
To reproduce start qemu with
-parallel none -device gus,audiodev=audio0 -audiodev pa,id=audio0
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-15-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Don't call pa_stream_writable_size() in qpa_write() before the
playback stream is ready. This prevents a lot of the following
pulseaudio error messages.
pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state
To reproduce start qemu with
-parallel none -device gus,audiodev=audio0
-audiodev pa,id=audio0,out.mixing-engine=off
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-14-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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The pulseaudio backend currently converts, clips and copies audio
playback samples in the mixing-engine sample buffer multiple
times.
In qpa_get_buffer_out() the function pa_stream_begin_write()
returns a rather large buffer and this allows audio_pcm_hw_run_out()
in audio/audio.c to copy all samples in the mixing-engine buffer
to the pulse audio buffer. Immediately after copying, qpa_write()
notices with a call to pa_stream_writable_size() that pulse audio
only needs a smaller part of the copied samples and ignores the
rest. This copy and ignore process happens several times for each
audio sample.
To fix this behaviour, call pa_stream_writable_size() in
qpa_get_buffer_out() to limit the number of samples
audio_pcm_hw_run_out() will convert. With this change the
pulseaudio pcm_ops functions put_buffer_out and write are no
longer identical and a separate qpa_put_buffer_out is needed.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-13-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Commit 73ad33ef7b "audio: remove plive" forgot to remove this code.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Enable the SDL2 backend options -audiodev sdl,out.mixing-
engine=off,in.mixing-engine=off.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Break the unnecessary dependency of the generic buffer management
code on mixing-engine. This is required for the next patch.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Add audio recording functions. SDL 2.0.5 or later is required to
use the recording functions. Playback continues to work with
earlier SDL 2.0 versions.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Split off pcm_ops function run_buffer_in from get_buffer_in and
call run_buffer_in before get_buffer_in.
The next patch only needs the generic buffer management part
from audio_generic_get_buffer_in().
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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With the modern audio functions it's possible to add new
features like audio recording.
As a side effect this patch fixes a bug where SDL2 can't be used
on Windows. This bug was reported on the qemu-devel mailing list at
https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg04043.html
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Fill the remaining sample buffer with silence. To fill it with
zeroes is wrong for unsigned samples because this is silence
with a DC bias.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Always fill the remaining audio callback buffer with silence.
SDL 2.0 doesn't initialize the audio callback buffer. This was
an incompatible change compared to SDL 1.2. For reference read
the SDL 1.2 to 2.0 migration guide.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Every emulated audio device has a way to enable audio playback. Don't
start playback until the guest enables the audio device. This patch
keeps the SDL2 device pause state in sync with hw->enabled.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Tested-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Currently there is a crackling noise with SDL2 audio playback.
Commit bcf19777df: "audio/sdlaudio: Allow audio playback with
SDL2" already mentioned the crackling noise.
Add an out.buffer-count option to give users a chance to select
sane settings for glitch free audio playback. The idea was taken
from the coreaudio backend.
The in.buffer-count option will be used with one of the next
patches.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Move the property types and property macros implemented in
qdev-properties-system.c to a new qdev-properties-system.h
header.
Signed-off-by: Eduardo Habkost <ehabkost@redhat.com>
Reviewed-by: Igor Mammedov <imammedo@redhat.com>
Message-Id: <20201211220529.2290218-16-ehabkost@redhat.com>
Signed-off-by: Eduardo Habkost <ehabkost@redhat.com>
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Check whenever we actually found the spiceaudio driver
before flipping the can_be_default field.
Fixes: f0c4555edfdd ("audio: remove qemu_spice_audio_init()")
Buglink: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=977301
Reported-by: dann frazier <dann.frazier@canonical.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20201215081151.20095-1-kraxel@redhat.com>
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This code (introduced in commit 1d14ffa97ea, Oct 2005)
is likely unused since years. Time to remove it. If
the condition is true, simply call abort().
Suggested-by: Gerd Hoffmann <gerd@kraxel.org>
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20201210223506.263709-1-philmd@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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The previous commit removed the last call site of
audio_is_cleaning_up(). Remove the now unused function.
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20201213130528.5863-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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Always stop audio playback and remove the playback callback when
QEMU exits.
On shut down the function coreaudio_fini_out() destroys the
coreaudio mutex but fails to stop audio playback and to remove the
audio playback callback, because function audio_is_cleaning_up()
always returns true when called from coreaudio_fini_out(). Now
there is a time window from pthread_mutex_destroy() to program
exit where Core Audio may call the audio playback callback which
tries to lock the destroyed coreaudio mutex. This leads to the
following error.
coreaudio: Could not lock voice for audioDeviceIOProc
Reason: Invalid argument
This bug was reported on the qemu-discuss mailing list.
https://lists.nongnu.org/archive/html/qemu-discuss/2020-10/msg00018.html
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20201213130528.5863-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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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>
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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>
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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>
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Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20201019075224.14803-6-kraxel@redhat.com
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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
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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>
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staging
Pull request
This includes the atomic_ -> qatomic_ rename that touches many files and is
prone to conflicts.
# gpg: Signature made Wed 23 Sep 2020 17:08:43 BST
# gpg: using RSA key 8695A8BFD3F97CDAAC35775A9CA4ABB381AB73C8
# gpg: Good signature from "Stefan Hajnoczi <stefanha@redhat.com>" [full]
# gpg: aka "Stefan Hajnoczi <stefanha@gmail.com>" [full]
# Primary key fingerprint: 8695 A8BF D3F9 7CDA AC35 775A 9CA4 ABB3 81AB 73C8
* remotes/stefanha/tags/block-pull-request:
qemu/atomic.h: rename atomic_ to qatomic_
tests: add test-fdmon-epoll
fdmon-poll: reset npfd when upgrading to fdmon-epoll
gitmodules: add qemu.org vbootrom submodule
gitmodules: switch to qemu.org meson mirror
gitmodules: switch to qemu.org qboot mirror
docs/system: clarify deprecation schedule
virtio-crypto: don't modify elem->in/out_sg
virtio-blk: undo destructive iov_discard_*() operations
util/iov: add iov_discard_undo()
virtio: add vhost-user-fs-ccw device
libvhost-user: handle endianness as mandated by the spec
MAINTAINERS: add Stefan Hajnoczi as block/nvme.c maintainer
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
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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>
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Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20200916084117.21828-3-kraxel@redhat.com
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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
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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>
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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>
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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>
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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>
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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>
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