Commit Graph

419 Commits

Author SHA1 Message Date
Paolo Bonzini 478e943f51 meson: convert audio directory to Meson
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2020-08-21 06:30:21 -04:00
Paolo Bonzini 243af0225a trace: switch position of headers to what Meson requires
Meson doesn't enjoy the same flexibility we have with Make in choosing
the include path.  In particular the tracing headers are using
$(build_root)/$(<D).

In order to keep the include directives unchanged,
the simplest solution is to generate headers with patterns like
"trace/trace-audio.h" and place forwarding headers in the source tree
such that for example "audio/trace.h" includes "trace/trace-audio.h".

This patch is too ugly to be applied to the Makefiles now.  It's only
a way to separate the changes to the tracing header files from the
Meson rewrite of the tracing logic.

Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2020-08-21 06:18:24 -04:00
Volker Rümelin 4f50d4a48e ossaudio: fix out of bounds write
In function oss_read() a read error currently does not exit the
read loop. With no data to read the variable pos will quickly
underflow and a subsequent successful read overwrites memory
outside the buffer. This patch adds the missing break statement
to the error path of the function.

To reproduce start qemu with -audiodev oss,id=audio0 and in the
guest start audio recording. After some time this will trigger
an exception.

Fixes: 3ba4066d08 "ossaudio: port to the new audio backend api"

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200707180836.5435-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-07-13 11:38:40 +02:00
Markus Armbruster 012d4c96e2 qapi: Make visitor functions taking Error ** return bool, not void
See recent commit "error: Document Error API usage rules" for
rationale.

Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Vladimir Sementsov-Ogievskiy <vsementsov@virtuozzo.com>
Message-Id: <20200707160613.848843-18-armbru@redhat.com>
2020-07-10 15:18:08 +02:00
Geoffrey McRae bc81e6e56e audio/jack: simplify the re-init code path
Instead of checking for the audodev state in each code path, centralize
the check into the initialize function itself to make it safe to call it
at any time.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-7-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Geoffrey McRae 81e0efb2e5 audio/jack: honour the enable state of the audio device
When the guest closes the audio device we must start dropping input
samples from JACK and zeroing the output buffer samples. Failure to do
so causes sound artifacts during operations such as guest OS reboot, and
causes a hang of the input pipeline breaking it until QEMU is restated.

Closing and reconnecting to JACK was tested during these enable/disable
calls which works well for Linux guests, however Windows re-opens the
audio hardware repeatedly even when doing simple tasks like playing a
system sounds. As such it was decided it is better to feed silence to
JACK while the device is disabled.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-6-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Geoffrey McRae de82640843 audio/jack: do not remove ports when finishing
This fixes a hang when there is a communications issue with the JACK
server. Simply closing the connection is enough to completely clean up
and as such we do not need to remove the ports first. As JACK uses a
socket based protocol that relies on the `select` call, if there is a
communication breakdown with the server the client library waits
forever for a response to the unregister request.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-5-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Geoffrey McRae f8f0f218d4 audio/jack: remove invalid set of input support bool
Initial code for JACK did not support audio input and as such this
boolean was set to let QEMU know, however JACK ended up including input
support making this invalid. Further investigation shows it was invalid
to set it in the first instance anyway due to a failure on my part
understand properly what this was for when the audodev was initially
developed.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-4-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Geoffrey McRae 2f33ee0808 audio/jack: remove unused stopped state
Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-3-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Geoffrey McRae 36963ed116 audio/jack: fix invalid minimum buffer size check
JACK does not provide us with the configured buffer size until after
activiation which was overriding this minimum value. JACK itself doesn't
have this minimum limitation, but the QEMU virtual hardware and as such
it must be enforced, failure to do so results in audio discontinuities.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-id: 20200613040518.38172-2-geoff@hostfission.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-06-17 14:44:51 +02:00
Philippe Mathieu-Daudé 57a878ed4f audio: Let capture_callback handler use const buffer argument
The buffer is the captured input to pass to backends.
As we should not modify it, mark the argument const.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-3-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-26 08:29:39 +02:00
Philippe Mathieu-Daudé e709d2ac47 audio: Let audio_sample_to_uint64() use const samples argument
The samples are the input to convert to u64. As we should
not modify them, mark the argument const.

Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20200505132603.8575-2-f4bug@amsat.org>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-26 08:29:30 +02:00
Bruce Rogers cbaf25d1f5 audio: fix wavcapture segfault
Commit 571a8c522e caused the HMP wavcapture command to segfault when
processing audio data in audio_pcm_sw_write(), where a NULL
sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is
valid before dereferincing it. A similar fix is also made in the
parallel function audio_pcm_sw_read().

Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and
volume_*)
Signed-off-by: Bruce Rogers <brogers@suse.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20200521172931.121903-1-brogers@suse.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-26 07:55:23 +02:00
Volker Rümelin 9c61fcc89a audio/mixeng: fix clang 10+ warning
The code in CONV_NATURAL_FLOAT() and CLIP_NATURAL_FLOAT()
seems to use the constant 2^31-0.5 to convert float to integer
and back. But the float type lacks the required precision and
the constant used for the conversion is 2^31. This is equiva-
lent to a [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] mapping.

This patch explicitly writes down the used constant. The
compiler generated code doesn't change.

The constant 2^31 has an exact float representation and the
clang 10 compiler stops complaining about an implicit int to
float conversion with a changed value.

A few notes:
- The conversion of 1.f to INT32_MAX + 1 doesn't overflow. The
  type of the destination variable is int64_t.
- At a later stage one of the clip_* functions in
  audio/mixeng_template.h limits INT32_MAX + 1 to the integer
  range.
- The clip_natural_float_* functions in audio/mixeng.c convert
  INT32_MAX and INT32_MAX + 1 to 1.f.

Buglink: https://bugs.launchpad.net/bugs/1878627
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200523201712.23908-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-26 07:46:51 +02:00
Geoffrey McRae 2e44570321 audio/jack: add JACK client audiodev
This commit adds a new audiodev backend to allow QEMU to use JACK as
both an audio sink and source.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Message-Id: <20200512101603.E3DB73A038E@moya.office.hostfission.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-05-25 11:30:03 +02:00
Volker Rümelin 8d1439b692 dsoundaudio: dsound_get_buffer_in should honor *size
This patch prevents an underflow of variable samples in function
audio_pcm_hw_run_in(). See commit 599eac4e5a "audio:
audio_generic_get_buffer_in should honor *size". This time the
while loop in audio_pcm_hw_run_in() will terminate nevertheless,
because it seems the recording stream in Windows is always rate
limited.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-04-06 13:29:53 +02:00
Volker Rümelin 174702986c dsoundaudio: fix "Could not lock capture buffer" warning
IDirectSoundCaptureBuffer_Lock() fails on Windows when called
with len = 0. Return early from dsound_get_buffer_in() in this
case.

To reproduce the warning start a linux guest. In the guest
start Audacity and you will see a lot of "Could not lock
capture buffer" warnings.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-04-06 13:29:53 +02:00
Volker Rümelin 4ba664cb0a dsoundaudio: fix never-ending playback loop
Currently the DirectSound backend fails to stop audio playback
in dsound_enable_out(). To detect a lost buffer condition
dsound_get_status_out() incorrectly uses the error code
DSERR_BUFFERLOST instead of flag DSBSTATUS_BUFFERLOST as a mask
and returns with an error. As a result dsound_enable_out()
returns early and doesn't stop playback.

To reproduce the bug start qemu on a Windows host with
-soundhw pcspk -audiodev dsound,id=audio0. On the guest
FreeDOS 1.2 command line enter beep. The image Day 1 - F-Bird
from the QEMU Advent Calendar 2018 shows the bug as well.

Buglink: https://bugs.launchpad.net/qemu/+bug/1699628
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-04-06 13:29:53 +02:00
Volker Rümelin 194bdf5069 audio: fix saturation nonlinearity in clip_* functions
The current positive limit for the saturation nonlinearity is
only correct if the type of the result has 8 bits or less.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-03-16 10:18:07 +01:00
Volker Rümelin 4218fdd77f audio: change mixing engine float range to [-1.f, 1.f]
Currently the internal float range of the mixing engine is
[-.5f, .5f]. PulseAudio, SDL2 and libasound use a [-1.f, 1.f]
range. This means with float samples the audio playback volume
is 6dB too low and audio recording signals will be clipped in
most cases.

To avoid another scaling factor in the conv_natural_float_* and
clip_natural_float_* functions with FLOAT_MIXENG defined this
patch changes the mixing engine float range to [-1.f, 1.f].

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-03-16 10:18:07 +01:00
Volker Rümelin 33a93baeae audio: consistency changes
Change the clip_natural_float_from_mono() function in
audio/mixeng.c to be consistent with the clip_*_from_mono()
functions in audio/mixeng_template.h.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-03-16 10:18:07 +01:00
Volker Rümelin dd381319a3 audio: change naming scheme of FLOAT_CONV macros
This patch changes the naming scheme of the FLOAT_CONV_TO and
FLOAT_CONV_FROM macros to the scheme used in mixeng_template.h.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200308193321.20668-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-03-16 10:18:07 +01:00
Philippe Mathieu-Daudé 3a1bdd1583 audio/alsaaudio: Remove superfluous semicolons
Fixes: 286a5d201e
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Acked-by: Paolo Bonzini <pbonzini@redhat.com>
Reviewed-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Reviewed-by: Juan Quintela <quintela@redhat.com>
Message-Id: <20200218094402.26625-3-philmd@redhat.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2020-02-18 20:20:49 +01:00
Kővágó, Zoltán ed2a4a7941 audio: proper support for float samples in mixeng
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>
2020-02-06 14:35:57 +01:00
Volker Rümelin 180b044ffd coreaudio: fix coreaudio playback
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>
2020-02-06 14:35:04 +01:00
Kővágó, Zoltán fb35c2cec5 audio/dsound: fix invalid parameters error
Windows (unlike wine) bails out when IDirectSoundBuffer8::Lock is called
with zero length.  Also, hw->pos_emul handling was incorrect when
calling this function for the first time.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reported-by: KJ Liew <liewkj@yahoo.com>
Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Message-id: fe9744216d9d421a2dbb09bcf5fa0dbd18f77ac5.1580684275.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-02-06 14:31:20 +01:00
Volker Rümelin 599eac4e5a audio: audio_generic_get_buffer_in should honor *size
The function generic_get_buffer_in currently ignores the *size
parameter and may return a buffer larger than *size.

As a result the variable samples in function
audio_pcm_hw_run_in may underflow. The while loop then most
likely will never termiate.

Buglink: http://bugs.debian.org/948658
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:49:48 +01:00
Volker Rümelin f03cd06814 ossaudio: disable poll mode can't be reached
Currently there is no way to disable poll mode in
oss_enable_out and oss_enable_in when it was enabled before.
The enable code path always resets the poll mode state variable.

Fixes: b027a538c6 "oss: Remove unused error handling of qemu_set_fd_handler"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:57 +01:00
Volker Rümelin 3e0c1bbab5 ossaudio: prevent SIGSEGV in oss_enable_out
With audiodev parameter out.mixing-engine=off hw->mix_buf is
NULL. This patch reverts a small part of dc88e38fa7 "audio:
unify input and output mixeng buffer management".

To reproduce the problem start qemu with
-audiodev oss,id=audio0,try-mmap=on,out.mixing-engine=off

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:03 +01:00
Volker Rümelin fdc8c5f471 audio: fix bug 1858488
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>
2020-01-31 08:48:03 +01:00
Volker Rümelin 69ac078632 audio: prevent SIGSEGV in AUD_get_buffer_size_out
With audiodev parameter out.mixing-engine=off hw->mix_buf is
NULL. This leads to a segmentation fault in
AUD_get_buffer_size_out. This patch reverts a small part of
dc88e38fa7 "audio: unify input and output mixeng buffer
management".

To reproduce the problem start qemu with
-soundhw adlib -audiodev pa,id=audio0,out.mixing-engine=off

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:03 +01:00
Volker Rümelin a76e6b8794 paaudio: remove unused variables
The unused variables were last used before commit 49ddd7e122
"paaudio: port to the new audio backend api".

Fixes: 49ddd7e122
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:03 +01:00
Volker Rümelin 4da58faa5b audio: fix audio_generic_read
It seems the function audio_generic_read started as a copy of
function audio_generic_write and some necessary changes were
forgotten. Fix the mixed up source and destination pointers and
rename misnamed variables.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:03 +01:00
Volker Rümelin d3ed099671 audio: fix audio_generic_write
The pcm_ops function put_buffer_out expects the returned pointer
of function get_buffer_out as argument. Fix this.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-31 08:48:03 +01:00
Gerd Hoffmann 7a4ede0047 audio/oss: fix buffer pos calculation
Fixes: 3ba4066d08 ("ossaudio: port to the new audio backend api")
Reported-by: ziming zhang <ezrakiez@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20200120101804.29578-1-kraxel@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
2020-01-31 08:47:55 +01:00
Philippe Mathieu-Daudé f7621fd1aa audio/audio: Add missing fall through comment
When building with GCC9 using CFLAG -Wimplicit-fallthrough=2 we get:

  audio/audio.c: In function ‘audio_pcm_init_info’:
  audio/audio.c:306:14: error: this statement may fall through [-Werror=implicit-fallthrough=]
    306 |         sign = 1;
        |         ~~~~~^~~
  audio/audio.c:307:5: note: here
    307 |     case AUDIO_FORMAT_U8:
        |     ^~~~
  cc1: all warnings being treated as errors

Similarly to e46349414, add the missing fall through comment to
hint GCC.

Fixes: 2b9cce8c8c
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Aleksandar Markovic <amarkovic@wavecomp.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20191218192526.13845-2-philmd@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2020-01-24 20:59:07 +01:00
Volker Rümelin 40ad46d3cc audio: fix integer overflow
Tell the compiler to do a 32bit * 32bit -> 64bit multiplication
because period_ticks is a 64bit variable. The overflow occurs
for audio timer periods larger than 4294967us.

Fixes: be1092afa0 "audio: fix audio timer rate conversion bug"

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 8893a235-66a8-8fbe-7d95-862e29da90b1@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-06 08:47:16 +01:00
Volker Rümelin 7c9eb86e67 paaudio: wait until the recording stream is ready
Don't call pa_stream_peek before the recording stream is ready.

Information to reproduce the problem.

Start and stop Audacity in the guest several times because the
problem is racy.

libvirt log file:
-audiodev pa,id=audio0,server=localhost,out.latency=30000,
 out.mixing-engine=off,in.mixing-engine=off \
-sandbox on,obsolete=deny,elevateprivileges=deny,spawn=deny,
 resourcecontrol=deny \
-msg timestamp=on
: Domain id=4 is tainted: custom-argv
char device redirected to /dev/pts/1 (label charserial0)
audio: Device pcspk: audiodev default parameter is deprecated,
 please specify audiodev=audio0
audio: Device hda: audiodev default parameter is deprecated,
 please specify audiodev=audio0
pulseaudio: pa_stream_peek failed
pulseaudio: Reason: Bad state
pulseaudio: pa_stream_peek failed
pulseaudio: Reason: Bad state

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-06 08:47:16 +01:00
Volker Rümelin acc3b63e1b paaudio: try to drain the recording stream
There is no guarantee a single call to pa_stream_peek every
timer_period microseconds can read a recording stream faster
than the data gets produced at the source. Let qpa_read try to
drain the recording stream.

To reproduce the problem:

Start qemu with -audiodev pa,id=audio0,in.mixing-engine=off

On the host connect the qemu recording stream to the monitor of
a hardware output device. While the problem can also be seen
with a hardware input device, it's obvious with the monitor of
a hardware output device.

In the guest start audio recording with audacity and notice the
slow recording data rate.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-06 08:47:16 +01:00
Volker Rümelin 4db3e634c7 paaudio: drop recording stream in qpa_fini_in
Every call to pa_stream_peek which returns a data length > 0
should have a corresponding pa_stream_drop. A call to qpa_read
does not necessarily call pa_stream_drop immediately after a
call to pa_stream_peek. Test in qpa_fini_in if a last
pa_stream_drop is needed.

This prevents following messages in the libvirt log file after
a recording stream gets closed and a new one opened.

pulseaudio: pa_stream_drop failed
pulseaudio: Reason: Bad state
pulseaudio: pa_stream_drop failed
pulseaudio: Reason: Bad state

To reproduce start qemu with
-audiodev pa,id=audio0,in.mixing-engine=off
and in the guest start and stop Audacity several times.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200104091122.13971-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-06 08:47:16 +01:00
Volker Rümelin 7ffc90f3ae audio: fix audio recording
With current code audio recording with all audio backends
except PulseAudio and DirectSound is broken. The generic audio
recording buffer management forgot to update the current read
position after a read.

Fixes: ff095e5231 "audio: api for mixeng code free backends"

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
Message-id: 2fc947cf-7b42-de68-3f11-cbcf1c096be9@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-11-20 09:11:12 +01:00
Paolo Bonzini 5608956575 audio: fix missing break
Reported by Coverity (CID 1406449).

Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2019-10-26 15:38:06 +02:00
Kővágó, Zoltán 0cf13e367a paaudio: fix channel order for usb-audio 5.1 and 7.1 streams
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 2900e462d27bd73277ae083d037c32b1b4451ee2.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán b5c7db3eef audio: basic support for multichannel audio
Which currently only means removing some checks.  Old code won't require
more than two channels, but new code will need it.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 7e53be1f97e939ed3bb729ef39e76b775643118a.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán 2b9cce8c8c audio: replace shift in audio_pcm_info with bytes_per_frame
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>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán cecc1e79bf audio: support more than two channels in volume setting
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 5d3dd2ee3baaa62805e79c3901abb7415ae32461.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán 337e8de6fb paaudio: get/put_buffer functions
This lets us avoid some buffer copying when using mixeng.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: d03d30138b9b5a9681cc90cbfbfec0a197cac88c.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán 1930616b98 audio: make mixeng optional
Implementation of the previously added mixing-engine option.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: c05bc258889ed289e8ee1bdbcc5e84174ec221e7.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 08:14:05 +02:00
Kővágó, Zoltán f47dffe8d1 audio: paaudio: ability to specify stream name
This can be used to identify stream in tools like pavucontrol when one
creates multiple -audiodevs or runs multiple qemu instances.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 2d6e337c474ac84172d0809e6959c26b21d48120.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 07:50:53 +02:00
Kővágó, Zoltán 3443ad4ed6 audio: paaudio: fix connection and stream name
Connection name was previously erroneously set to the server socket
path, while connection names were simply "qemu".  After this patch, the
connection name will be the vm name (falling back to "qemu" if not
specified), while stream names will be the audiodev's id.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 3d139426031a400a68d440608ba5e43f0e116cd8.1568157545.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18 07:50:53 +02:00