Otherwise, the audio subsystem tries to use the voice and
eventually aborts due to the maximum number of samples in the
buffer is not set.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20220226115953.60335-1-akihiko.odaki@gmail.com>
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>
On macOS 11.3.1, Core Audio calls AudioDeviceIOProc after calling an
internal function named HALB_Mutex::Lock(), which locks a mutex in
HALB_IOThread::Entry(void*). HALB_Mutex::Lock() is also called in
AudioObjectGetPropertyData, which is called by coreaudio driver.
Therefore, a deadlock will occur if coreaudio driver calls
AudioObjectGetPropertyData while holding a lock for a mutex and tries
to lock the same mutex in AudioDeviceIOProc.
audioDeviceIOProc, which implements AudioDeviceIOProc in coreaudio
driver, requires an exclusive access for the device configuration and
the buffer. Fortunately, a mutex is necessary only for the buffer in
audioDeviceIOProc because a change for the device configuration occurs
only before setting up AudioDeviceIOProc or after stopping the playback
with AudioDeviceStop.
With this change, the mutex owned by the driver will only be used for
the buffer, and the device configuration change will be protected with
the implicit iothread mutex.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-id: 20210622201740.38005-1-akihiko.odaki@gmail.com
Message-Id: <20210622201740.38005-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Before commit 7d6948cd98, it was coded to
retrieve the initial output stream format settings, modify the frame
rate, and set again. However, I removed a frame rate modification code by
mistake in the commit. It also assumes the initial output stream format
is consistent with what QEMU expects, but that expectation is not in the
code, which makes it harder to understand and will lead to breakage if
the initial settings change.
This change explicitly sets all of the output stream settings to solve
these problems.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20210616141721.54091-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
An output device change can occur when plugging or unplugging an
earphone.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20210311151512.22096-3-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Mac OS X 10.6 was released in 2009.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Peter Maydell <peter.maydell@linaro.org>
Message-Id: <20210311151512.22096-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Fix problems about braces:
-braces are necessary for all arms of if/for/while statements
-else should follow close brace '}'
Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-2-zhanghan64@huawei.com>
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 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>
There are reports that since commit 2ceb8240fa "coreaudio: port
to the new audio backend api" audio playback with CoreAudio is
broken. This patch reverts some parts the commit.
Because of changes in the audio subsystem the audio clip
function in v4.1.0 of coreaudio.c had to be moved to mixeng.c
and the generic buffer management code needed a hint about the
size of the float type.
This patch is based on a patch from Zoltán Kővágó found at
https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg02142.html.
Fixes: 2ceb8240fa "coreaudio: port to the new audio backend api"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200202140641.4737-1-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>
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
Since aa5cb7f5e, the chardevs are being cleaned up when leaving qemu,
before the atexit() handlers. audio_cleanup() may use the monitor to
notify of changes. For compatibility reasons, let's clean up audio
before the monitor so it keeps emitting monitor events.
The audio_atexit() function is made idempotent (so it can be called
multiple times), and renamed to audio_cleanup(). Since coreaudio
backend is using a 'isAtexit' code path, change it to check
audio_is_cleaning_up() instead, so the path is taken during normal
exit.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20160801112343.29082-3-marcandre.lureau@redhat.com>
Reviewed-by: Paolo Bonzini <pbonzini@redhat.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@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>
The AudioDeviceAddIOProc() and AudioDeviceRemoveIOProc() functions were
deprecated in OSX 10.5. Since we don't support any earlier versions of
OSX, we can simply replace them with the new APIs
AudioDeviceCreateIOProcID() and AudioDeviceRemoveIOProcID().
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1448747724-15572-6-git-send-email-peter.maydell@linaro.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Use the new-in-OSX 10.6 API AudioObjectGetPropertyData() instead
of the deprecated AudioDeviceGetProperty() and AudioDeviceSetProperty()
functions when possible.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1448747724-15572-5-git-send-email-peter.maydell@linaro.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The CoreAudio APIs AudioDeviceGetProperty and AudioDeviceSetProperty are
deprecated from OSX 10.6, so factor out our calls to them so we can
provide versions which use the replacement APIs on OSX newer than 10.5.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1448747724-15572-4-git-send-email-peter.maydell@linaro.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
If we're building for OSX 10.6 or better, use the new API
AudioObjectGetPropertyData for getting the default voice.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1448747724-15572-3-git-send-email-peter.maydell@linaro.org
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The CoreAudio function AudioHardwareGetProperty has been deprecated
starting with OSX 10.6, so factor out our call to it so we can
provide an equivalent with the new APIs when they exist.
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1448747724-15572-2-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>
OSStatus type is defined as SInt32. That's signed int on __LP64__ and
signed long otherwise.
Since it is an explicit 32-bit-width type, cast to corresponsing POSIX type
and use PRId32 format specifier. This avoids a warning on ppc64.
Cc: malc <av1474@comtv.ru>
Signed-off-by: Andreas Faerber <andreas.faerber@web.de>
Signed-off-by: malc <av1474@comtv.ru>
coreaudioVoiceOut's audioDevicePropertyBufferFrameSize is defined as UInt32
and is being used by reference for AudioDevice{Get,Set}Property().
UInt32 is unsigned int on __LP64__ but unsigned long otherwise.
Cast to POSIX type and use PRIu32 format specifier to hide the details.
This avoids a warning on ppc64.
Cc: malc <av1474@comtv.ru>
Signed-off-by: Andreas Faerber <andreas.faerber@web.de>
Signed-off-by: malc <av1474@comtv.ru>
In audio/coreaudio.c, a variable named "str" was assigned "const char" values,
which resulted in the following warnings:
-----8<-----
audio/coreaudio.c: In function ‘coreaudio_logstatus’:
audio/coreaudio.c:59: warning: initialization discards qualifiers from pointer target type
audio/coreaudio.c:63: warning: assignment discards qualifiers from pointer target type
(...)
-----8<-----
Signed-off-by: Alexandre Raymond <cerbere@gmail.com>
Acked-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Andreas Färber <andreas.faerber@web.de>
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>
