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>
Reviewed-by: Eric Blake <eblake@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>
audio_init() should not use hw_error(), because dumping CPU registers
is unhelpful there, and aborting is wrong, because it can be called
called from an audio device's realize() method.
The two uses of hw_error() come from commit 0d9acba:
* When qemu_new_timer() fails. It couldn't fail back then, and it
can't fail now. Drop the unreachable error handling.
* When no_audio_driver can't be initialized. It couldn't fail back
then, and it can't fail now. Replace the error handling by an
assertion.
Cc: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@pond.sub.org>
Reviewed-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>
Variable "conf" going out of scope leaks the storage
it points to in line 856.
Signed-off-by: Gonglei <arei.gonglei@huawei.com>
Message-Id: <1435021270-7768-1-git-send-email-arei.gonglei@huawei.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Enabling this option just creates a playback buffer with the specified settings,
and then ignores it. It's probably some outdated hack to set audio formats on
windows. (The first created stream dictates all other streams settings, at least
on some Windows versions). Setting DAC_FIXED_SETTINGS should have the same
effect as setting (the now removed) primary buffer.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
According to MSDN this may happen when the window is not in the foreground, but
the default is 1 since a long time (which means no retries), so it should be ok.
I've found no problems during testing it on Windows 7 and wine, so this was
probably only the case with some old Windows versions.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Setting QEMU_AUDIO_LOG_TO_MONITOR=1 can crash qemu (if qemu tries to log
to the monitor before it's being initialized), and also nothing else in
qemu logs to the monitor.
This log to monitor feature was the last thing that used the default_mon
variable, so I removed it too (as using it can cause problems).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
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>
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>
DirectSound should be a superior choice on Windows.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
ESD is no longer developed and replaced by PulseAudio.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The function cannot fail, so the check is superfluous.
Signed-off-by: Fam Zheng <famz@redhat.com>
Message-id: 1433400324-7358-11-git-send-email-famz@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
The function cannot fail, so the check is superfluous.
Signed-off-by: Fam Zheng <famz@redhat.com>
Message-id: 1433400324-7358-10-git-send-email-famz@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
When stopping an audio voice, call the audio backend's fini
method before calling audio_pcm_hw_free_resources_ rather than
afterwards. This allows backends which use helper threads (like
pulseaudio) to terminate those threads before the conv_buf or
mix_buf are freed and avoids race conditions where the helper
may access a NULL pointer or freed memory.
Cc: qemu-stable@nongnu.org
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Message-id: 1418406239-9838-1-git-send-email-peter.maydell@linaro.org
Replace fprintf(stderr,...) with error_report() in files audio/*.
The trailing "\n"s of the @fmt argument have been removed
because @fmt of error_report() should not contain newline.
Signed-off-by: Le Tan <tamlokveer@gmail.com>
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
After previous Peter patch, they are redundant. This way we don't
assign them except when needed. Once there, there were lots of case
where the ".fields" indentation was wrong:
.fields = (VMStateField []) {
and
.fields = (VMStateField []) {
Change all the combinations to:
.fields = (VMStateField[]){
The biggest problem (appart from aesthetics) was that checkpatch complained
when we copy&pasted the code from one place to another.
Signed-off-by: Juan Quintela <quintela@redhat.com>
Reviewed-by: Peter Maydell <peter.maydell@linaro.org>
Current Makefile system allows using foo.o-cflags variables to store
object-specific CFLAGS. Convert some usages of old syntax
(using QEMU_CFLAGS += construct) to the new syntax.
Do not touch multifile modules for now, as build system isn't ready for this.
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
This is more then plenty to keep audio card fifos filles / emptied.
This drops host cpu-load for audio playback inside a linux vm from
13% to 9%.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Check whenever the device path (/dev/dsp by default) exists and qemu is
allowed to access it. Return NULL if it isn't, so ossaudio will not
be used on systems wihtout oss support (increasinly common on modern
linux systems).
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Modern Linux's no longer support /dev/dsp so enabling it by
default causes audio failures on newer Linux distros.
Signed-off-by: Anthony Liguori <aliguori@amazon.com>
Tested-by: Andreas Färber <afaerber@suse.de>
Message-id: 1383497154-9271-1-git-send-email-aliguori@amazon.com
Fix error: ‘inline’ is not at beginning of declaration
[-Werror=old-style-declaration]
Signed-off-by: Alex Bligh <alex@alex.org.uk>
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
Now that we no longer have MIN_REARM_TIMER_NS a bug in the audio subsys has
clearly shown it self by trying to make a timer fire every nano second.
Note we have a similar problem in 1.6, 1.5 and older but there
MIN_REARM_TIMER_NS limits the wakeups caused by audio being active to
4000 times / second. This still causes a host cpu load of 50 % for simply
playing audio, where as with this patch git master is at 13%, so we should
backport this to 1.5 and 1.6 too.
Note this will not apply to 1.5 and 1.6 as is.
Cc: qemu-stable@nongnu.org
Signed-off-by: Hans de Goede <hdegoede@redhat.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>
Using macros instead of static functions for dolog and for ldebug
simplifies the code and can also reduce the total code size.
GCC_ATTR was only used in audio_int.h, so it is now unused and
the definition can be removed from compiler.h.
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
Remove the OSS support for OpenBSD. The OSS API has not been usable
for quite some time.
Signed-off-by: Brad Smith <brad@comstyle.com>
Reviewed-by: Laszlo Ersek <lersek@redhat.com>
Reviewed-by: Andreas Färber <afaerber@suse.de>
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
Some source files #include the same header more than
once for no good reason. Remove second #includes in
such cases.
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
sw->name already uses the correct g_free to free the allocated memory.
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
Touching char/char.h basically causes the whole of QEMU to
be rebuilt. Avoid this, it is usually unnecessary.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Various header files rely on qemu-char.h including qemu-config.h or
main-loop.h, but they really do not need qemu-char.h at all (particularly
interesting is the case of the block layer!). Clean this up, and also
add missing inclusions of qemu-char.h itself.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
This reverts commit 456a84d156.
This patch wasn't submitted to the list and did not get Acked by other
copyright holders in the file.
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
This reverts commit 72bc6f1bf7.
This patch wasn't submitted to the list and did not get Acked by other
copyright holders in the file.
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
smatch report:
audio/audio_template.h:416 AUD_open_out(18) warn:
variable dereferenced before check 'as' (see line 414)
Moving the ldebug statement after the statement which checks 'as'
fixes that warning.
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: malc <av1474@comtv.ru>
Winwave audio backend has problem with pausing and restart audio out.
Unlike other backends, Winwave pausing API does not flush audio buffer.
As a result, the previous audio data are played in front of
user expected sound when user restart audio.
So changes it to waveOutReset()
Signed-off-by: Munkyu Im <munkyu.im@samsung.com>
Signed-off-by: malc <av1474@comtv.ru>
Not only clean up enabled voices but any registered one. Backends like
pulsaudio rely on unconditional fini handler invocations.
This fixes "Memory pool destroyed but not all memory blocks freed!"
warnings on VM shutdowns when pa is used and lockups of QEMU on shutdown
as it got stuck on some pa-internal synchronization point.
Signed-off-by: Jan Kiszka <jan.kiszka@siemens.com>
Signed-off-by: malc <av1474@comtv.ru>
Split IN_T into BSIZE and ITYPE, to avoid expansion if the OS has
defined macros for the intX_t and uintX_t types. The IN_T constant is
then defined in mixeng_template.h so it can be used by the
functions/macros on this header file.
This change has been tested successfully under Debian Linux and NetBSD
6.0BETA.
Cc: Vassili Karpov (malc) <av1474@comtv.ru>
Signed-off-by: Roger Pau Monne <roger.pau@citrix.com>
Signed-off-by: malc <av1474@comtv.ru>
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>
Use Spice server volume control API when available.
Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
If the audio backend is capable of volume control, don't apply
software volume (mixeng_volume ()), but instead, rely on backend
volume control. This will allow guest to have full range volume
control.
Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
Add a new PCM control operation to update the stream volume on the
audio backend. The argument given is a SWVoiceOut/SWVoiceIn.
v4:
- verified other backends didn't fail/assert on this new control
they randomly return 0 or -1, but we ignore return value.
Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
Static code analysers expect these comments for case statements without
a break statement.
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: malc <av1474@comtv.ru>
accidently->accidentally
annother->another
choosen->chosen
consideres->considers
decriptor->descriptor
developement->development
paramter->parameter
preceed->precede
preceeding->preceding
priviledge->privilege
propogation->propagation
substraction->subtraction
throught->through
upto->up to
usefull->useful
Fix also grammar in posix-aio-compat.c
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: Stefan Hajnoczi <stefanha@linux.vnet.ibm.com>
The variable is assigned a value which is never used,
so remove variable and assignment.
Signed-off-by: Stefan Weil <sw@weilnetz.de>
Signed-off-by: malc <av1474@comtv.ru>
QEMUFile * is only intended for migration nowadays. Using it for
anything else just adds pain and a layer of buffers for no good
reason.
Signed-off-by: Juan Quintela <quintela@redhat.com>
CC: malc <av1474@comtv.ru>
Signed-off-by: malc <av1474@comtv.ru>
QEMUFile * is only intended for migration nowadays. Using it for
anything else just adds pain and a layer of buffers for no good
reason.
Signed-off-by: Juan Quintela <quintela@redhat.com>
CC: malc <av1474@comtv.ru>
Signed-off-by: malc <av1474@comtv.ru>
Today, when notifying a VM state change with vm_state_notify(),
we pass a VMSTOP macro as the 'reason' argument. This is not ideal
because the VMSTOP macros tell why qemu stopped and not exactly
what the current VM state is.
One example to demonstrate this problem is that vm_start() calls
vm_state_notify() with reason=0, which turns out to be VMSTOP_USER.
This commit fixes that by replacing the VMSTOP macros with a proper
state type called RunState.
Signed-off-by: Luiz Capitulino <lcapitulino@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>
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>
Fix an integer overflow that can happen for signed 32 bit types
when using FLOAT_MIXENG. (Note that at the moment this is only true
when using the MacOSX coreaudio audio driver.)
Signed-off-by: Juha Riihim?ki <juha.riihimaki@nokia.com>
[Peter Maydell: Removed unnecessary casts]
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Signed-off-by: malc <av1474@comtv.ru>
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>
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>
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>
Add support for the spice audio interface. With this patch applied
audio can be forwarded over the network from/to the spice client. Both
recording and playback is supported.
The driver is first in the driver list, but the can_be_default flag is
set only in case spice is active. So if you have the spice protocol
enabled the spice audio driver is the default one, otherwise whatever
comes first after spice in the list. Overriding the default using
QEMU_AUDIO_DRV works in any case.
[ v2: audio codestyle: add spaces before open parenthesis ]
[ v2: add const to silence array ]
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Cc: malc <av1474@comtv.ru>
Signed-off-by: malc <av1474@comtv.ru>
snd_pcm_start() starts the capture process and ensures that the events
are delivered to the poll handler. Without the call, capture can be started
only when there is simultaneous playback running.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: malc <av1474@comtv.ru>
Playback control function did not disable polling when playback stops.
Caused busy spinning of the main loop due to unprocessed events.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: malc <av1474@comtv.ru>
Since version 4.4.x, gcc supports additional format attributes.
__attribute__ ((format (gnu_printf, 1, 2)))
should be used instead of
__attribute__ ((format (printf, 1, 2))
because QEMU always uses standard format strings (even with mingw32).
The patch replaces format attribute printf / __printf__ by macro
GCC_FMT_ATTR which uses gnu_printf if supported.
It also removes an #ifdef __GNUC__ (not needed any longer).
Cc: Blue Swirl <blauwirbel@gmail.com>
Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
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>
By moving the definition of GCC_ATTR and GCC_FMT_ATTR
from audio_int.h to qemu-common.h these macros are
now generally available for further patches which add
the gcc format attribute.
Newer gcc versions support format gnu_printf which is
better suited for use in QEMU than format printf
(QEMU always uses standard format strings (even with mingw32)).
V2: Use correct operator '==' (instead of '=')
Cc: Blue Swirl <blauwirbel@gmail.com>
Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
When available, we'd like to be able to access the DeviceState
when registering a savevm. For buses with a get_dev_path()
function, this will allow us to create more unique savevm
id strings.
Signed-off-by: Alex Williamson <alex.williamson@redhat.com>
Signed-off-by: Anthony Liguori <aliguori@us.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>
Commits 376253ec..731b0364 introduced global variable cur_mon, which
points to the "default monitor" (if any), except during execution of
monitor_read() or monitor_control_read() it points to the monitor from
which we're reading instead (the "current monitor"). Monitor command
handlers run within monitor_read() or monitor_control_read().
Default monitor and current monitor are really separate things, and
squashing them together is confusing and error-prone.
For instance, usb_host_scan() can run both in "info usbhost" and
periodically via usb_host_auto_check(). It prints to cur_mon, which
is what we want in the former case: the monitor executing "info
usbhost". But since that's the default monitor in the latter case, it
periodically spams the default monitor there.
A few places use cur_mon to log stuff to the default monitor. If we
ever log something while cur_mon points to current monitor instead of
default monitor, the log temporarily "jumps" to another monitor.
Whether that can or cannot happen isn't always obvious.
Maybe logging to the default monitor (which may not even exist) is a
bad idea, and we should log to stderr or a logfile instead. But
that's outside the scope of this commit.
Change cur_mon to point to the current monitor. Create new
default_mon to point to the default monitor. Update users of cur_mon
accordingly.
This fixes the periodical spamming of the default monitor by
usb_host_scan(). It also stops "log jumping", should that problem
exist.
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
Turns out on those versions of FreeBSD (>= 7.x) that know OSS_GETVERSION
the ioctl doesn't actually work yet (except in the Linuxolator), so if
building on FreeBSD assume the sound drivers are new enough if the ioctl
returns the errno it does currently on FreeBSD.
(Rev 2 after private discussion with malc.)
Signed-off-by: Juergen Lock <nox@jelal.kn-bremen.de>
Signed-off-by: malc <av1474@comtv.ru>
Previous patch introduced subtle regression, in cases when
OSS_GETVERSION fails the code wasn't falling back to
SNDCTL_DSP_SETFRAGMENT.
Signed-off-by: malc <av1474@comtv.ru>
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>
Additional argument (whether to try poll mode) is only passed with
VOICE_ENABLE command.
Thanks to Markus Armbruster for noticing the potential breakage.
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>
a. Use SNDCTL_DSP_POLICY instead of SNDCTL_DSP_SETFRAGMENT
b. Add ability to open device in exclusive mode, thus bypassing vmix
Signed-off-by: malc <av1474@comtv.ru>
Problem: Our file sys-queue.h is a copy of the BSD file, but there are
some additions and it's not entirely compatible. Because of that, there have
been conflicts with system headers on BSD systems. Some hacks have been
introduced in the commits 15cc923584,
f40d753718,
96555a96d7 and
3990d09adf but the fixes were fragile.
Solution: Avoid the conflict entirely by renaming the functions and the
file. Revert the previous hacks.
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
audio/esdaudio.c: In function 'qesd_thread_out':
audio/esdaudio.c:136: error: format '%d' expects type 'int', but
argument 3 has type 'ssize_t'
audio/esdaudio.c: In function 'qesd_thread_in':
audio/esdaudio.c:366: error: format '%d' expects type 'int', but
argument 3 has type 'ssize_t'
Signed-off-by: Kirill A. Shutemov <kirill@shutemov.name>
Signed-off-by: malc <av1474@comtv.ru>
Dsound currently does not compile due to the typos in the code. This
patch makes it compile again.{PATCH}
Signed-off-by: Alex Ivanov <void@aleksoft.net>
Signed-off-by: malc <av1474@comtv.ru>
Both input and output streams may be in SND_PCM_STATE_SUSPENDED
after the host is suspended and resumed, meaning "Hardware is
suspended". snd_pcm_readi() and snd_pcm_writei() will return
-ESTRPIPE if called while the stream is in this state.
Call snd_pcm_resume() to enable audio output and capture after
host resume.
Signed-off-by: Bjørn Mork <bjorn@mork.no>
Signed-off-by: malc <av1474@comtv.ru>
Generate CONFIG_AUDIO_DRIVERS. Order is important here, because the
first driver in the list is the one used by default.
Signed-off-by: Juan Quintela <quintela@redhat.com>
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
/usr/include/alsa/pcm.h contains:
#define snd_pcm_sw_params_alloca(ptr) do { assert(ptr); *ptr = (snd_pcm_sw_params_t *) alloca(snd_pcm_sw_params_sizeof()); memset(*ptr, 0, snd_pcm_sw_params_sizeof()); } while (0)
The assert generates: "error: the address of 'sw_params' will always
evaluate as 'true'" which combined with -Werror prevents alsaaudio.o
from being built with certain versions of GCC.
Failure to initialize the audio subsystem is not handled consistently.
Where it is handled it has guest visible effects, which is wrong.
We already have a "nosound" audio driver as a last resort, so trying to
proceed without an audio backend seems pointless.
Also protect against multiple calls to AUD_init so that this can be
pushed down into individual devices.
Signed-off-by: Paul Brook <paul@codesourcery.com>
Suppress a warning
audio/dsoundaudio.c:35:1: warning: "WIN32_LEAN_AND_MEAN" redefined
<command line>:4:1: warning: this is the location of the previous definition
Signed-off-by: Alex Ivanov <void@aleksoft.net>
Refactor the monitor API and prepare it for decoupled terminals:
term_print functions are renamed to monitor_* and all monitor services
gain a new parameter (mon) that will once refer to the monitor instance
the output is supposed to appear on. However, the argument remains
unused for now. All monitor command callbacks are also extended by a mon
parameter so that command handlers are able to pass an appropriate
reference to monitor output services.
For the case that monitor outputs so far happen without clearly
identifiable context, the global variable cur_mon is introduced that
shall once provide a pointer either to the current active monitor (while
processing commands) or to the default one. On the mid or long term,
those use case will be obsoleted so that this variable can be removed
again.
Due to the broad usage of the monitor interface, this patch mostly deals
with converting users of the monitor API. A few of them are already
extended to pass 'mon' from the command handler further down to internal
functions that invoke monitor_printf.
At this chance, monitor-related prototypes are moved from console.h to
a new monitor.h. The same is done for the readline API.
Signed-off-by: Jan Kiszka <jan.kiszka@siemens.com>
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@6711 c046a42c-6fe2-441c-8c8c-71466251a162
Change from v1:
Avoid changing the existing coding style in certain files.
Signed-off-by: Stuart Brady <stuart.brady@gmail.com>
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@6120 c046a42c-6fe2-441c-8c8c-71466251a162
when compiling on NetBSD:
warning: array subscript has type 'char'
Signed-off-by: Christoph Egger <Christoph.Egger@amd.com>
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@5727 c046a42c-6fe2-441c-8c8c-71466251a162
The issue was first noticed/addressed by Laurent Vivier in his QEMU on
AIX patches.
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@5712 c046a42c-6fe2-441c-8c8c-71466251a162
This makes it easier to reuse in other parts of QEMU.
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4975 c046a42c-6fe2-441c-8c8c-71466251a162
- got recent copy from netbsd.
- deleted debug code.
- deleted some list implementations, left in only the ones which
qemu already has.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4973 c046a42c-6fe2-441c-8c8c-71466251a162
This brings latency down to acceptable levels when using dmix
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4865 c046a42c-6fe2-441c-8c8c-71466251a162
Instead of having separate option for each card and driver use
--audio-drv-list and --audio-card-list options.
Under Linux it allows to set the default(first probed) driver
to something other than OSS.
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4792 c046a42c-6fe2-441c-8c8c-71466251a162
At least on one system zero is returned in either fragsize or
fragstotal (reported by Dave Scott), this results in an audio_calloc
failing the audio_bug check and another ominous error message. Fail
early and blame the system.
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4699 c046a42c-6fe2-441c-8c8c-71466251a162
From http://repo.or.cz/w/qemu/malc.git?a=shortlog;h=refs/heads/audio :
Implicitly lock the mutex at startup of playback/capture threads, otherwise
pthread_mutex_destroy (in audio_pt_fini) fails with EBUSY.
Endianness fix.
Remove a c&p residue.
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@4296 c046a42c-6fe2-441c-8c8c-71466251a162