Commit Graph

215 Commits

Author SHA1 Message Date
Paolo Bonzini
525877c999 build: move rules from Makefile to */Makefile.objs
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2012-12-19 08:29:06 +01:00
Anthony Liguori
46ee77b357 Revert "audio/wavcapture: Clarify licensing"
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>
2012-11-30 09:04:47 -06:00
Anthony Liguori
d76aa45bf1 Revert "audio/audio_pt_int: Clarify licensing"
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>
2012-11-30 09:04:08 -06:00
malc
72bc6f1bf7 audio/audio_pt_int: Clarify licensing
Signed-off-by: malc <av1474@comtv.ru>
2012-11-19 22:26:13 +04:00
malc
456a84d156 audio/wavcapture: Clarify licensing
Signed-off-by: malc <av1474@comtv.ru>
2012-11-19 22:23:17 +04:00
Stefan Weil
93b6599734 audio: Fix warning from static code analysis
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>
2012-09-23 01:34:16 +04:00
munkyu.im
13ef70f64e audio/winwave: previous audio buffer should be flushed
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>
2012-08-28 19:11:28 +04:00
malc
8361710398 audio: Unbreak capturing in mixemu case
Signed-off-by: malc <av1474@comtv.ru>
2012-07-16 18:08:36 +04:00
malc
eb2aeacf98 audio/winwave: Fix typo
Signed-off-by: malc <av1474@comtv.ru>
2012-06-15 20:58:54 +04:00
Paolo Bonzini
b0b68fc671 build: move audio/ objects to nested Makefile.objs
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2012-06-07 09:21:14 +02:00
Jan Kiszka
aeb29b6459 audio: Always call fini on exit
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>
2012-05-24 19:35:27 +04:00
Roger Pau Monne
a28853871d audio: split IN_T into two separate constants
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>
2012-05-18 15:19:28 +04:00
Gerd Hoffmann
8f473dd104 fix build with pulseaudio versions older than 0.9.11
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
2012-05-04 00:47:09 +04:00
Gerd Hoffmann
d6c05bbf29 fix paaudio.c warnings
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
2012-04-25 21:04:57 +04:00
Marc-André Lureau
6e7a7f3d9b Allow controlling volume with PulseAudio backend
Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
2012-04-17 16:57:58 +04:00
Marc-André Lureau
ea9ebc2ce6 Do not use pa_simple PulseAudio API
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>
2012-04-17 16:57:58 +04:00
Marc-André Lureau
a70c99c614 audio/spice: add support for volume control
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>
2012-04-17 16:57:58 +04:00
Marc-André Lureau
c01b245623 audio: don't apply volume effect if backend has VOICE_VOLUME_CAP
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>
2012-04-17 16:57:57 +04:00
Marc-André Lureau
6c95ab94f9 audio: add VOICE_VOLUME ctl
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>
2012-04-17 16:57:57 +04:00
Stefan Weil
b4bd0b168e audio: Add some fall through comments
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>
2012-02-25 18:16:11 +04:00
Stefan Weil
e7d81004e4 Fix spelling in comments, documentation and messages
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>
2011-12-14 11:09:44 +00:00
Stefan Weil
15d4a72338 fmodaudio: Remove unused variable 'bits16' (reported by cppcheck)
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>
2011-11-18 21:55:29 +04:00
Anthony Liguori
7f67d8922e Merge remote-tracking branch 'qmp/queue/qmp' into staging 2011-09-20 15:16:00 -05:00
Juan Quintela
27acf660aa wavaudio: Use stdio instead of QEMUFile
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>
2011-09-20 17:55:52 +04:00
Juan Quintela
b04df2a440 wavcapture: Use stdio instead of QEMUFile
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>
2011-09-20 17:50:23 +04:00
Luiz Capitulino
1dfb4dd993 Replace the VMSTOP macros with a proper state type
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>
2011-09-15 16:39:32 -03:00
Anthony Liguori
7267c0947d Use glib memory allocation and free functions
qemu_malloc/qemu_free no longer exist after this commit.

Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
2011-08-20 23:01:08 -05:00
Andreas Färber
744d364418 coreaudio: Fix OSStatus format specifier
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>
2011-06-23 18:56:58 +04:00
Andreas Färber
cbc36cb05d coreaudio: Avoid formatting UInt32 type
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>
2011-06-23 18:56:50 +04:00
Alexandre Raymond
d9cbb0f3ed Fix compilation warning due to incorrectly specified type
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>
2011-06-14 03:08:56 +02:00
Alexandre Raymond
9bf0960a9a Fix compilation warning due to missing header for sigaction (followup)
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>
2011-06-08 09:04:29 +01:00
Juha Riihim?ki
578c7b2ca8 audio: fix integer overflow expression
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>
2011-06-01 00:14:07 +04:00
Stefan Weil
4ff9786c67 Fix trivial "endianness bugs"
Replace endianess -> endianness.

Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Aurelien Jarno <aurelien@aurel32.net>
2011-04-03 21:42:57 +02:00
Paolo Bonzini
7447545544 change all other clock references to use nanosecond resolution accessors
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>
2011-03-21 09:23:23 +01:00
Gerd Hoffmann
bf1064b587 pulseaudio: tweak config
Zap unused divisor field.
Raise the buffer size default.

Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
2011-01-25 19:56:53 +03:00
Gerd Hoffmann
e6d16fa439 pulseaudio: setup buffer attrs
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>
2011-01-25 19:56:47 +03:00
Gerd Hoffmann
6315633b25 pulseaudio: process 1/4 buffer max at once
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>
2011-01-25 19:56:35 +03:00
Michael Walle
00e076795f audio: split sample conversion and volume mixing
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>
2011-01-12 18:36:22 +03:00
Michael Walle
d66bddd7a4 alsaaudio: add endianness support for VoiceIn
Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
2011-01-09 03:06:08 +03:00
Michael Walle
b6c9c9401c ossaudio: add endianness support for VoiceIn
Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
2011-01-09 03:05:24 +03:00
Michael Walle
8a7d0890ac noaudio: correctly account acquired samples
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>
2011-01-04 03:53:47 +03:00
Michael Walle
85882c71a9 noaudio: fix return value for read()
Read should return bytes instead of samples.

Signed-off-by: Michael Walle <michael@walle.cc>
Signed-off-by: malc <av1474@comtv.ru>
2010-12-10 05:25:35 +03:00
Stefan Weil
ab9de3692e audio: Use GCC_FMT_ATTR (format checking)
Cc: Blue Swirl <blauwirbel@gmail.com>
Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
2010-12-04 20:51:18 +00:00
malc
39deb1e496 audio: Only use audio timer when necessary
Originally proposed by Gerd Hoffmann.

Signed-off-by: malc <av1474@comtv.ru>
Acked-by: Gerd Hoffmann <kraxel@redhat.com>
2010-11-18 14:30:31 +03:00
Gerd Hoffmann
cf2c1839a9 add copyright to spiceaudio
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
2010-11-11 17:59:25 +03:00
Gerd Hoffmann
3e31375378 spice: add audio
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>
2010-11-09 23:39:30 +03:00
Jindrich Makovicka
38cc9b607f issue snd_pcm_start() when capturing audio
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>
2010-10-18 00:39:06 +04:00
Jindrich Makovicka
22d948a2d9 fix 100% CPU load when idle with ALSA
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>
2010-10-18 00:39:02 +04:00
Stefan Weil
8b7968f7c4 Use GCC_FMT_ATTR (format checking)
Signed-off-by: Stefan Weil <weil@mail.berlios.de>
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
2010-10-03 06:34:51 +00:00
Stefan Weil
e5924d8980 Replace most gcc format attributes by macro GCC_FMT_ATTR (format checking)
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>
2010-10-03 06:34:36 +00:00