Add a new -audio backend that accepts D-Bus clients/listeners to handle
playback & recording, to be exported via the -display dbus.
Example usage:
-audiodev dbus,in.mixing-engine=off,out.mixing-engine=off,id=dbus
-display dbus,audiodev=dbus
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Acked-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>
Audio stuff is under "Miscellanea", and authorization stuff is under
"Input". Add suitable header doc comments to correct that.
Cc: Gerd Hoffmann <kraxel@redhat.com>
Cc: Daniel P. Berrange <berrange@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-Id: <20201102081550.171061-3-armbru@redhat.com>
Acked-by: Daniel P. Berrangé <berrange@redhat.com>
In commit 26ec4e53f2 and similar commits we fixed the indentation
for doc comments in our qapi json files to follow a new stricter
standard for indentation, which permits only:
@arg: description line 1
description line 2
or:
@arg:
line 1
line 2
Unfortunately since we didn't manage to get the script changes that
enforced the new style in, a variety of commits (eg df4097aeaf,
2e44570321) introduced new doc text which doesn't follow the new
stricter rules for indentation on multi-line doc comments. Bring
those into line with the new rules.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-Id: <20200810195019.25427-3-peter.maydell@linaro.org>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Signed-off-by: Markus Armbruster <armbru@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>
The review for patch ed2a4a7941 "audio: proper support for
float samples in mixeng" suggested this would be a good idea.
Acked-by: Markus Armbruster <armbru@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Tested-by: John Arbuckle <programmingkidx@gmail.com>
Message-id: 20200308193321.20668-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
This will allow us to disable mixeng when we use a decent backend.
Disabling mixeng have a few advantages:
* we no longer convert the audio output from one format to another, when
the underlying audio system would just convert it to a third format.
We no longer convert, only the underlying system, when needed.
* the underlying system probably has better resampling and sample format
converting methods anyway...
* we may support formats that the mixeng currently does not support (S24
or float samples, more than two channels)
* when using an audio server (like pulseaudio) different sound card
outputs will show up as separate streams, even if we use only one
backend
Disadvantages:
* audio capturing no longer works (wavcapture, and vnc audio extension)
* some backends only support a single playback stream or very picky
about the audio format. In this case we can't disable mixeng.
Originally thw two main use cases of the disabled option was: using
unsupported audio formats (5.1 and 7.1 audio) and having different
pulseaudio streams per audio frontend. Since we can have multiple
-audiodevs, the latter is not that important, so currently you only need
this option if you want to use 5.1 or 7.1 audio (implemented in a later
patch), otherwise it's probably better to stick to the old and tried
mixeng, since it's less picky about the backends.
The ideal solution would be to port as much as possible to gstreamer,
but this is currently out of scope:
https://wiki.qemu.org/Internships/ProjectIdeas/AudioGStreamer
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 5765186a7aadd51a72bc7d3e804307f0ee8a34ce.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>
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>
This patch adds structures into qapi to replace the existing
configuration structures used by audio backends currently. This qapi
will be the base of the -audiodev command line parameter (that replaces
the old environment variables based config).
This is not a 1:1 translation of the old options, I've tried to make
them much more consistent (e.g. almost every backend had an option to
specify buffer size, but the name was different for every backend, and
some backends required usecs, while some other required frames, samples
or bytes). Also tried to reduce the number of abbreviations used by the
config keys.
Some of the more important changes:
* use `in` and `out` instead of `ADC` and `DAC`, as the former is more
user friendly imho
* moved buffer settings into the global setting area (so it's the same
for all backends that support it. Backends that can't change buffer
size will simply ignore them). Also using usecs, as it's probably more
user friendly than samples or bytes.
* try-poll is now an alsa backend specific option (as all other backends
currently ignore it)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Markus Armbruster <armbru@redhat.com>
Message-id: 5461b514dbf3e0bc31b0abb6498a9b3a008c271e.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Marc-André Lureau
Volker Rümelin
Markus Armbruster
Peter Maydell
Geoffrey McRae
Kővágó, Zoltán
Martin Schrodt