Move the code to generate the pa_context_new() application name
argument to a function in audio/audio.c. The new function
audio_application_name() will also be used in the jackaudio
backend.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
In current code there are no calls to pa_stream_get_latency()
or pa_stream_get_time() to receive latency or time information.
Remove the flags PA_STREAM_INTERPOLATE_TIMING and
PA_STREAM_AUTO_TIMING_UPDATE which instruct PulseAudio to
calculate this information in regular intervals.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit e50caf4a5c ("tracing: convert documentation to rST")
converted docs/devel/tracing.txt to docs/devel/tracing.rst.
We still have several references to the old file, so let's fix them
with the following command:
sed -i s/tracing.txt/tracing.rst/ $(git grep -l docs/devel/tracing.txt)
Signed-off-by: Stefano Garzarella <sgarzare@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-Id: <20210517151702.109066-2-sgarzare@redhat.com>
Signed-off-by: Thomas Huth <thuth@redhat.com>
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>
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>
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>
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>
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>
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>
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>
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>
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>
