Fix the same samples vs. frames mix-up that the previous commit
fixed for the PulseAudio backend.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-15-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit ff095e5231 "audio: api for mixeng code free backends"
introduced another FIFO for the audio subsystem with exactly the
same size as the mixing-engine FIFO. Most audio backends use
this generic FIFO. The generic FIFO used together with the
mixing-engine FIFO doubles the audio FIFO size, because that's
just two independent FIFOs connected together in series.
For audio playback this nearly doubles the playback latency.
This patch restores the effective mixing-engine playback buffer
size to a pre v4.2.0 size by only accepting the amount of
samples for the mixing-engine queue which the downstream queue
accepts.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Replace open-coded buffer arithmetic with the new function
audio_ring_posb(). That's the position in backward direction
of a given point at a given distance.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
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>
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>
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>
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>
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>
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>
This way we no longer need vararg functions, improving compile time
error detection. Also now it's possible to check actually what commands
are supported, without needing to manually update ctl_caps.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 2b08b3773569c5be055d0a0fb2f29ff64e79f0f4.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
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>
At the end of the while-loop, either "samples" or "sdl->live" is zero, so
now that we've removed the semaphore code, the content of the while-loop
is always only executed once. Thus we can remove the while-loop now to
get rid of one indentation level here.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1549336101-17623-3-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The semaphore code was only working with SDL1.2 - with SDL2, it causes
a deadlock. Since we've removed support for SDL1.2 recently, we can
now completely remove the semaphore code from sdlaudio.c.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 1549336101-17623-2-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <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>
When compiling with SDL2, the semaphore trick used in sdlaudio.c
does not work - QEMU locks up completely in this case. To avoid
the hang and get at least some audio playback up and running (it's
a little bit crackling, but better than nothing), we can use the
SDL locking functions SDL_LockAudio() and SDL_UnlockAudio() to sync
with the sound playback thread instead.
Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1485852398-2327-1-git-send-email-thuth@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>
Since SDL uses a lot of global data, we can't create independent
instances of sdl audio backend.
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>
This patch removes all references to signal.h when qemu-common.h is included
as they become redundant.
Signed-off-by: Alexandre Raymond <cerbere@gmail.com>
Signed-off-by: Stefan Hajnoczi <stefanha@linux.vnet.ibm.com>
Remove unused 'shift' variable spotted by clang.
Also clean up aud_to_sdlfmt which used to get the value
of shift.
Signed-off-by: Serge Ziryukin <ftrvxmtrx@gmail.com>
Signed-off-by: malc <av1474@comtv.ru>
This reverts commit 4839abe78f.
The commit was badly broken, Gentoo has sdl as the default driver,
consequently 5 gentoo users have hit the breakage and were kind enough
to report, so thank you:
Claes Gyllenswrd
vekin
Chris
But above all thanks to Toralf Foerster who actually provied enough
information to pinpoint the breakage to sdlaudio.
http://bugs.gentoo.org/show_bug.cgi?id=294269
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>
