These have been deprecated for a long time, and the introduction of
-audio in 7.1.0 has cemented the new way of specifying an audio backend's
parameters. However, there is still a need for simple configuration
of the audio backend in the desktop case; therefore, if no audiodev is
passed to audio_init(), go through a bunch of simple Audiodev* structures
and pick the first that can be initialized successfully.
The only QEMU_AUDIO_* option that is left in, waiting for a better idea,
is QEMU_AUDIO_DRV=none which is used by qtest.
Remove all the parsing code, including the concept of "can_be_default"
audio drivers: now that audio_prio_list[] is only used in a single place,
wav can be excluded directly in that function.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
An error is already printed by audio_driver_init, but we can make
it more precise if the driver can return an Error *.
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Swap the rate and info parameters of the audio_rate_get_bytes()
function to align the parameter order with the rest of the
audio_rate_*() functions.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-8-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>
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>
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>
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples). But if we want to add support for surround sound,
this no longer holds true.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.git.DirtY.iCE.hu@gmail.com
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>
This commit removes the ad-hoc rate-limiting code from noaudio and
wavaudio, and replaces them with a (slightly modified) code from
spiceaudio. This way multiple write calls (for example when the
circular buffer wraps around) do not cause problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: fd0fe5b95b13fa26d09ae77a72f99d0ea411de14.1568927990.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>
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
Move it to the actual users. There are some inclusions of
qemu/host-utils.h in headers, but they are all necessary.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
This patch replaces get_ticks_per_sec() calls with the macro
NANOSECONDS_PER_SECOND. Also, as there are no callers, get_ticks_per_sec()
is then removed. This replacement improves the readability and
understandability of code.
For example,
timer_mod(fdctrl->result_timer,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + (get_ticks_per_sec() / 50));
NANOSECONDS_PER_SECOND makes it obvious that qemu_clock_get_ns
matches the unit of the expression on the right side of the plus.
Signed-off-by: Rutuja Shah <rutu.shah.26@gmail.com>
Signed-off-by: Paolo Bonzini <pbonzini@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>
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 is an autogenerated patch using scripts/switch-timer-api.
Switch the entire code base to using the new timer API.
Note this patch may introduce some line length issues.
Signed-off-by: Alex Bligh <alex@alex.org.uk>
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
This was done with:
sed -i 's/qemu_get_clock\>/qemu_get_clock_ns/' \
$(git grep -l 'qemu_get_clock\>' )
sed -i 's/qemu_new_timer\>/qemu_new_timer_ns/' \
$(git grep -l 'qemu_new_timer\>' )
after checking that get_clock and new_timer never occur twice
on the same line. There were no missed occurrences; however, even
if there had been, they would have been caught by the compiler.
There was exactly one false positive in qemu_run_timers:
- current_time = qemu_get_clock (clock);
+ current_time = qemu_get_clock_ns (clock);
which is of course not in this patch.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
This will fix the return value of the function which otherwise returns too
many samples because sw->total_hw_samples_acquired isn't correctly
accounted.
Signed-off-by: Michael Walle <michael@walle.cc>
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>
