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>
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>
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>
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>
The current code does not specify the metrics of the buffers for the
input device. This makes PulseAudio choose very bad defaults, which
causes input to be unusable: Audio put in gets out 30 seconds later.
This patch fixes that and makes the latency configurable as well.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190315084653.120020-4-martin@schrodt.org
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>
Audiodev configuration allows to set the length of the buffered data.
The setting was ignored and a constant value used instead.
This patch makes the code apply the setting properly, and uses the
previous default if nothing is supplied.
Signed-off-by: Martin Schrodt <martin@schrodt.org>
Message-id: 20190315084653.120020-2-martin@schrodt.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Audio drivers now get an Audiodev * as config paramters, instead of the
global audio_option structs. There is some code in audio/audio_legacy.c
that converts the old environment variables to audiodev options (this
way backends do not have to worry about legacy options). It also
contains a replacement of -audio-help, which prints out the equivalent
-audiodev based config of the currently specified environment variables.
Note that backends are not updated and still rely on environment
variables.
Also note that (due to moving try-poll from global to backend specific
option) currently ALSA and OSS will always try poll mode, regardless of
environment variables or -audiodev options.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: e99a7cbdac0d13512743880660b2032024703e4c.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum. This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Check whenever the pulseaudio daemon pidfile is present before trying to
initialize the pulseaudio backend. Just return NULL if that is not the
case, so qemu will check the next backend in line.
In case the user explicitly configured a non-default pulseaudio server
skip the check.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20190124112055.547-5-kraxel@redhat.com
The rate of pulseaudio absorbing the audio stream is used to control the
the rate of the guests audio stream. When the emulated hardware uses
small chunks (like intel-hda does) we need small chunks on the audio
backend side too, otherwise that feedback loop doesn't work very well.
Cc: Max Ehrlich <maxehr@umiacs.umd.edu>
Cc: Martin Schrodt <martin@schrodt.org>
Buglink: https://bugs.launchpad.net/bugs/1795527
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20181109142032.1628-1-kraxel@redhat.com
Add registry for audio drivers, using the existing audio_driver struct.
Make all drivers register themself. The old list of audio_driver struct
pointers is now a list of audio driver names, specifying the priority
(aka probe order) in case no driver is explicitly asked for.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20180306074053.22856-2-kraxel@redhat.com
Apparently we don't use __MSC_VER as a compiler anymore and we always
require a C99 compiler (which means we always have __func__) so we don't
need a special AUDIO_FUNC macro. We can just replace AUDIO_FUNC with
__func__ instead.
Checkpatch failures were manually fixed.
Signed-off-by: Alistair Francis <alistair.francis@xilinx.com>
Cc: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20180203084315.20497-2-armbru@redhat.com>
The point of writing a macro embedded in a 'do { ... } while (0)'
loop (particularly if the macro has multiple statements or would
otherwise end with an 'if' statement) is so that the macro can be
used as a drop-in statement with the caller supplying the
trailing ';'. Although our coding style frowns on brace-less 'if':
if (cond)
statement;
else
something else;
that is the classic case where failure to use do/while(0) wrapping
would cause the 'else' to pair with any embedded 'if' in the macro
rather than the intended outer 'if'. But conversely, if the macro
includes an embedded ';', then the same brace-less coding style
would now have two statements, making the 'else' a syntax error
rather than pairing with the outer 'if'. Thus, even though our
coding style with required braces is not impacted, ending a macro
with ';' makes our code harder to port to projects that use
brace-less styles.
The change should have no semantic impact. I was not able to
fully compile-test all of the changes (as some of them are
examples of the ugly bit-rotting debug print statements that are
completely elided by default, and I didn't want to recompile
with the necessary -D witnesses - cleaning those up is left as a
bite-sized task for another day); I did, however, audit that for
all files touched, all callers of the changed macros DID supply
a trailing ';' at the callsite, and did not appear to be used
as part of a brace-less conditional.
Found mechanically via: $ git grep -B1 'while (0);' | grep -A1 \\\\
Signed-off-by: Eric Blake <eblake@redhat.com>
Acked-by: Cornelia Huck <cohuck@redhat.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Acked-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Message-Id: <20171201232433.25193-7-eblake@redhat.com>
Reviewed-by: Juan Quintela <quintela@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Since pulseaudio 1.0 it's possible to set the individual stream volume
rather than setting the device volume. With this, setting hardware mixer
of a emulated sound card doesn't mess up the volume configuration of the
host.
A side effect is that this limits compatible pulseaudio version to 1.0
which was released on 2011-09-27.
Signed-off-by: Peter Krempa <pkrempa@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 78853815be2069971b89b3a2e3181837064dd8f3.1462962512.git.pkrempa@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Clean up includes so that osdep.h is included first and headers
which it implies are not included manually.
This commit was created with scripts/clean-includes.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1453138432-8324-1-git-send-email-peter.maydell@linaro.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
qpa_audio_init did not clean up resources properly if the initialization
failed. This hopefully fixes it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently the opaque pointer returned by audio_driver's init is only
exposed to the driver's fini, but not to audio_pcm_ops. This way if
someone wants to share a variable with the driver and the pcm, he must
use global variables. This patch fixes it by adding a third parameter to
audio_pcm_op's init_out and init_in.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Unfortunately, pa_simple is a limited API which doesn't let us
retrieve the associated pa_stream. It is needed to control the volume
of the stream.
In v4:
- add missing braces
Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
Request reasonable buffer sizes from pulseaudio. Without this
pa_simple_write() can block quite long and lead to dropouts,
especially with guests which use small audio ring buffers.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
Limit the size of data pieces processed by the pulseaudio worker
threads. Never ever process more than 1/4 of the buffer at once.
Background: The buffer area currently processed by the pulseaudio thread
is blocked, i.e. the main thread (or iothread) can't fill in more data
there. The buffer processing time is roughly real-time due to the
pa_simple_write() call blocking when the output queue to the pulse
server is full. Thus processing big chunks at once means blocking
a large part of the buffer for a long time. This brings high latency
and can lead to dropouts.
When processing the buffer in smaller chunks the rpos handling becomes a
problem though. The thread reads hw->rpos without knowing whenever
qpa_run_out has already seen the last (small) chunk processed and
updated rpos accordingly. There is no point in reading hw->rpos though,
pa->rpos can be used instead. We just need to take care to initialize
pa->rpos before kicking the thread.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
Refactor the volume mixing, so it can be reused for capturing devices.
Additionally, it removes superfluous multiplications with the nominal
volume within the hardware voice code path.
Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
Fix a rpos coordination bug between qpa_run_out() and qpa_thread_out(),
which shows up as playback noises.
qpa_run_out()
qpa_thread_out loop N critical section 1
qpa_run_out() qpa_thread_out loop N doing pa_simple_write()
qpa_run_out() qpa_thread_out loop N doing pa_simple_write()
qpa_thread_out loop N critical section 2
qpa_thread_out loop N+1 critical section 1
qpa_run_out() qpa_thread_out loop N+1 doing pa_simple_write()
In the above scheme, "qpa_thread_out loop N+1 critical section 1" will
get the same rpos as the one used by "qpa_thread_out loop N critical
section 1". So it will be reading dead samples from the old rpos.
The rpos can only be updated back to qpa_thread_out when there is a
qpa_run_out() run between two qpa_thread_out loops.
normal sequence:
qpa_thread_out:
hw->rpos (X0) => local rpos => pa->rpos (X1)
qpa_run_out:
pa->rpos (X1) => hw->rpos (X1)
qpa_thread_out:
hw->rpos (X1) => local rpos => pa->rpos (X2)
buggy sequence:
qpa_thread_out:
hw->rpos (X0) => local rpos => pa->rpos (X1)
qpa_thread_out:
hw->rpos (X0) => local rpos => pa->rpos (X1')
Obviously qpa_run_out() shall be called at least once between any two
qpa_thread_out loops (after pa->rpos is set), in order for the new
qpa_thread_out loop to see the updated rpos.
Setting pa->live to 0 does the trick. The next loop will have to wait
for one qpa_run_out() invocation in order to get a non-zero pa->live
and proceed.
Signed-off-by: malc <av1474@comtv.ru>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
We're seeing various issues with the SDL audio backend and want to
switch to the pulseaudio backend. See e.g.
https://bugzilla.redhat.com/495964https://bugzilla.redhat.com/519540https://bugzilla.redhat.com/496627
The pulseaudio backend seems to work well, so we should allow it to be
selected as the default.
Signed-off-by: Mark McLoughlin <markmc@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
pcm_ops.run_out now takes number of live samples (which will be always
greater than zero) as a second argument, every driver was calling
audio_pcm_hw_get_live_out anyway with exception of fmod which used
audio_pcm_hw_get_live_out2 for no good reason.
Signed-off-by: malc <av1474@comtv.ru>
