Commit Graph

600 Commits

Author SHA1 Message Date
Volker Rümelin
33940dd336 audio: add pcm_ops function table for capture backend
Add a pcm_ops function table for the capture backend. This avoids
additional code in the next patches to test if the pcm_ops table
is available.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
a806f95904 audio: copy playback stream in sequential order
Change the code to copy the playback stream in sequential order.
The advantage can be seen in the next patches where the stream
copy operation effectively becomes a write through operation.

The following diagram shows the average buffer fill level and
the stream copy sequence. ### represents a timer_period sized
chunk. The rest of the buffer sizes are not to scale.

With current code:
         |--------| |#####111| |---#####|
          sw->buf    mix_buf    backend buffer
  1. clip
         |--------| |---#####| |111##222|
          sw->buf    mix_buf    backend buffer
  2. write to audio device
  333 -> |--------| |---#####| |---111##| -> 222
          sw->buf    mix_buf    backend buffer
  3a. sw device write
         |-----333| |---#####| |---111##|
          sw->buf    mix_buf    backend buffer
  3b. resample and mix
         |--------| |333#####| |---111##|
          sw->buf    mix_buf    backend buffer

With this patch:
  111 -> |--------| |---#####| |---#####|
          sw->buf    mix_buf    backend buffer
  1a: sw device write
         |-----111| |---#####| |---#####|
          sw->buf    mix_buf    backend buffer
  1b. resample and mix
         |--------| |111##222| |---#####|
          sw->buf    mix_buf    backend buffer
  2. clip
         |--------| |---111##| |222##333|
          sw->buf    mix_buf    backend buffer
  3. write to audio device
         |--------| |---111##| |---222##| -> 333
          sw->buf    mix_buf    backend buffer

The effective total playback buffer size is reduced by
timer_period.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
369829a435 jackaudio: use more jack audio buffers
The next patch reduces the effective qemu playback buffer size
by timer-period. Increase the number of jack audio buffers by
one to preserve the total effective buffer size. The size of one
jack audio buffer is 512 samples. With audio defaults that's
512 samples / 44100 samples/s = 11.6 ms and only slightly larger
than the timer-period of 10 ms.

The larger jack audio buffer increases audio dropout safety,
because the high priority jack-audio worker threads can provide
audio data for a longer period of time as with a smaller buffer
and more audio data in the mixing engine buffer that they can't
access.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20220301191311.26695-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
30ff5e24a3 paaudio: increase default latency to 46ms
This is a patch to improve the pulseaudio playback experience.
Asking pulseaudio for a playback latency of 15ms is quite
demanding. Increase this to 46ms. The total playback latency
now is 31ms larger. One of the next patches will reduce the
total playback latency again by more than 46ms.

Here is a quote from the PulseAudio Latency Control
documentation: 'For the sake of (...) drop-out safety always
make sure to pick the highest latency possible that fulfills
your needs.'

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
0ceb26af0c audio: inline function audio_pcm_sw_get_rpos_in()
Simplify code by inlining function audio_pcm_sw_get_rpos_in()
at the only call site and remove the duplicated audio_bug()
test.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
251f15496e audio: add function audio_pcm_hw_conv_in()
Add a function audio_pcm_hw_conv_in() similar to the existing
counterpart function audio_pcm_hw_clip_out(). This function reduces
the number of calls to the pcm_ops functions get_buffer_in() and
put_buffer_in(). That's one less call to get_buffer_in() and
put_buffer_in() every time the conv_buffer wraps around.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
8e56a172a1 audio: move function audio_pcm_hw_clip_out()
Move the function audio_pcm_hw_clip_out() into the correct
section 'Hard voice (playback)'.

Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00
Volker Rümelin
18404ff111 audio: replace open-coded buffer arithmetic
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>
2022-03-04 11:05:13 +01:00
Volker Rümelin
9d90ceb274 dsoundaudio: fix crackling audio recordings
Audio recordings with the DirectSound backend don't sound right.
A look a the Microsoft online documentation tells us why.

From the DirectSound Programming Guide, Capture Buffer Information:
'You can safely copy data from the buffer only up to the read
cursor.'

Change the code to read up to the read cursor instead of the
capture cursor.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20211226154017.6067-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-01-13 10:47:52 +01:00
Volker Rümelin
ead789eb46 jackaudio: use ifdefs to hide unavailable functions
On Windows the jack_set_thread_creator() function and on MacOS the
pthread_setname_np() function with a thread pointer paramater is
not available. Use #ifdefs to remove the jack_set_thread_creator()
function call and the qjack_thread_creator() function in both
cases.

The qjack_thread_creator() function just sets the name of the
created thread for debugging purposes and isn't really necessary.

From the jack_set_thread_creator() documentation:
(...)

No normal application/client should consider calling this. (...)

Resolves: https://gitlab.com/qemu-project/qemu/-/issues/785
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20211226154017.6067-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-01-13 10:47:52 +01:00
Marc-André Lureau
739362d420 audio: add "dbus" audio backend
Add a new -audio backend that accepts D-Bus clients/listeners to handle
playback & recording, to be exported via the -display dbus.

Example usage:
-audiodev dbus,in.mixing-engine=off,out.mixing-engine=off,id=dbus
-display dbus,audiodev=dbus

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Acked-by: Gerd Hoffmann <kraxel@redhat.com>
2021-12-21 10:50:22 +04:00
Paolo Bonzini
87430d5b13 configure, meson: move audio driver detection to Meson
This brings a change that makes audio drivers more similar to all
other modules.  All drivers are built by default, while
--audio-drv-list only governs the default choice of the audio driver.

Meson options are added to disable the drivers, and the next patches
will fix the help messages and command line options, and especially
make the non-default drivers available via -audiodev.

Cc: Gerd Hoffman <kraxel@redhat.com>
Cc: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20211007130630.632028-4-pbonzini@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2021-10-14 09:50:56 +02:00
Paolo Bonzini
7e1fbe7963 audio: remove CONFIG_AUDIO_WIN_INT
Ever since winwaveaudio was removed in 2015, CONFIG_AUDIO_WIN_INT
is only set if dsound is in use, so use CONFIG_AUDIO_DSOUND directly.

Cc: Gerd Hoffman <kraxel@redhat.com>
Cc: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20211007130630.632028-3-pbonzini@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2021-10-14 09:50:56 +02:00
Dr. David Alan Gilbert
da77adbaf6 audio: Never send migration section
The audio migration vmstate is empty, and always has been; we can't
just remove it though because an old qemu might send it us.
Changes with -audiodev now mean it's sometimes created when it didn't
used to be, and can confuse migration to old qemu.

Change it so that vmstate_audio is never sent; if it's received it
should still be accepted, and old qemu's shouldn't be too upset if it's
missing.

Signed-off-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Tested-by: Daniel P. Berrangé <berrange@redhat.com>
Message-Id: <20210809170956.78536-1-dgilbert@redhat.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-08-10 10:55:57 +02:00
Gerd Hoffmann
f6b12dfd80 modules: add audio module annotations
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Jose R. Ziviani <jziviani@suse.de>
Message-Id: <20210624103836.2382472-9-kraxel@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2021-07-09 18:20:27 +02:00
Akihiko Odaki
eb1a35e47a coreaudio: Lock only the buffer
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>
2021-06-23 14:16:48 +02:00
Akihiko Odaki
986bdbc6a2 coreaudio: Fix output stream format settings
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>
2021-06-17 12:00:26 +02:00
Akihiko Odaki
0c29b786e6 audio: Fix format specifications of debug logs
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-id: 20210616141411.53892-1-akihiko.odaki@gmail.com
Message-Id: <20210616141411.53892-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-06-17 11:56:57 +02:00
Volker Rümelin
2833d697b9 jackaudio: avoid that the client name contains the word (NULL)
Currently with jackaudio client name and qemu guest name unset,
the JACK client names are out-(NULL) and in-(NULL). These names
are user visible in the patch bay. Replace the function call to
qemu_get_vm_name() with a call to audio_application_name() which
replaces NULL with "qemu" to have more descriptive names.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-06-17 11:54:09 +02:00
Volker Rümelin
37a54d054f audio: move code to audio/audio.c
Move the code to generate the pa_context_new() application name
argument to a function in audio/audio.c. The new function
audio_application_name() will also be used in the jackaudio
backend.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-06-17 11:54:09 +02:00
Volker Rümelin
50db82d84c paaudio: remove unused stream flags
In current code there are no calls to pa_stream_get_latency()
or pa_stream_get_time() to receive latency or time information.

Remove the flags PA_STREAM_INTERPOLATE_TIMING and
PA_STREAM_AUTO_TIMING_UPDATE which instruct PulseAudio to
calculate this information in regular intervals.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-06-17 11:54:09 +02:00
Volker Rümelin
243011896a alsaaudio: remove #ifdef DEBUG to avoid bit rot
Merge the #ifdef DEBUG code with the if statement a few lines
above to avoid bit rot.

Suggested-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-06-17 11:54:09 +02:00
Stefano Garzarella
d0fb9657a3 docs: fix references to docs/devel/tracing.rst
Commit e50caf4a5c ("tracing: convert documentation to rST")
converted docs/devel/tracing.txt to docs/devel/tracing.rst.

We still have several references to the old file, so let's fix them
with the following command:

  sed -i s/tracing.txt/tracing.rst/ $(git grep -l docs/devel/tracing.txt)

Signed-off-by: Stefano Garzarella <sgarzare@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-Id: <20210517151702.109066-2-sgarzare@redhat.com>
Signed-off-by: Thomas Huth <thuth@redhat.com>
2021-06-02 06:51:09 +02:00
Akihiko Odaki
3ba6e3f688 coreaudio: Handle output device change
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>
2021-03-16 07:17:50 +01:00
Akihiko Odaki
7d6948cd98 coreaudio: Extract device operations
This change prepare to support dynamic device changes, which requires to
perform device initialization/deinitialization multiple times.

Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20210311151512.22096-2-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-03-16 07:17:50 +01:00
Akihiko Odaki
c960070c36 coreaudio: Drop support for macOS older than 10.6
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>
2021-03-16 07:17:50 +01:00
Philippe Mathieu-Daudé
538f049704 sysemu: Let VMChangeStateHandler take boolean 'running' argument
The 'running' argument from VMChangeStateHandler does not require
other value than 0 / 1. Make it a plain boolean.

Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Acked-by: David Gibson <david@gibson.dropbear.id.au>
Message-Id: <20210111152020.1422021-3-philmd@redhat.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2021-03-09 23:13:57 +01:00
Zhang Han
8abf3feb4d audio: space prohibited between function name and parenthesis'('
Delete spaces between function name and open parenthesis'('

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-8-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
289db3c5a2 audio: Suspect code indent for conditional statements
Fix code indent.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-7-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
dea7d84fcf audio: Don't use '%#' in format strings
Use '0x' prefix instead of '%#'

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-6-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
c60840c758 audio: Fix lines over 90 characters
Fix the line width of code.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-5-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
dcf10e4095 audio: foo* bar" should be "foo *bar".
transfer "foo* " to "foo *"

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-4-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
3c8de96c07 audio: Add spaces around operator/delete redundant spaces
Fix problems about spaces:
-operator needs spaces around it, add them.
-somespaces are redundant, remove them.

Signed-off-by: Zhang Han <zhanghan64@huawei.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20210115012431.79533-1-zhanghan64@huawei.com
Message-Id: <20210115012431.79533-3-zhanghan64@huawei.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:49:26 +01:00
Zhang Han
6c6886bd01 audio: Add braces for statements/fix braces' position
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>
2021-01-15 11:49:26 +01:00
Volker Rümelin
2d96a00587 dsoundaudio: fix log message
There is a mismatch between message and used argument. Change
the argument from frequency to format.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-23-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
1157506161 dsoundaudio: enable f32 audio sample format
Enable the f32 audio sample format for the DirectSound backend.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-22-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
3c18e43179 dsoundaudio: rename dsound_open()
Rename dsound_open() to dsound_set_cooperative_level(). The
only task of that function is to set the cooperative level for
DirectSound.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-21-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
401dcf0540 dsoundaudio: replace GetForegroundWindow()
GetForegroundWindow() doesn't necessarily return the own window
handle. It just returns a handle to the currently active window
and can even return NULL. At the time dsound_open() gets called
the active window is most likely the shell window and not the
QEMU window.

Replace GetForegroundWindow() with GetDesktopWindow() which
always returns a valid window handle, and at the same time
replace the DirectSound buffer flag DSBCAPS_STICKYFOCUS with
DSBCAPS_GLOBALFOCUS where Windows only expects a valid window
handle for DirectSound function SetCooperativeLevel(). The
Microsoft online docs for IDirectSound::SetCooperativeLevel
recommend this in the remarks.

This fixes a bug where you can't hear sound from the guest.

To reproduce start qemu with -machine pcspk-audiodev=audio0
-device intel-hda -device hda-duplex,audiodev=audio0
-audiodev dsound,id=audio0,out.mixing-engine=off
from a shell and start audio playback with the hda device in the
guest. The guest will be silent. To hear guest audio you have to
activate the shell window once.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-20-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
d9a8b27a7e paaudio: send recorded data in smaller chunks
Tell PulseAudio to send recorded audio data in smaller chunks
than timer_period, so there's a good chance that qemu can read
recorded audio data every time it looks for new data.

PulseAudio tries to send buffer updates at a fragsize / 2 rate.
With fragsize = timer_period / 2 * 3 the update rate is 75% of
timer_period. The lower limit for the recording buffer size
maxlength is fragsize * 2.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-19-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
00413ed9c2 paaudio: limit minreq to 75% of audio timer_rate
Currently with the playback buffer attribute minreq = -1 and flag
PA_STREAM_EARLY_REQUESTS PulseAudio uses minreq = tlength / 4.
To improve audio playback with larger PulseAudio server side
buffers, limit minreq to a maximum of 75% of audio timer_rate.
That way there is a good chance qemu receives a stream buffer
size update before it tries to write data to the playback stream.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-18-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
cffd2fdf2c paaudio: comment bugs in functions qpa_init_*
The audio buffer size in audio/paaudio.c is typically larger
than expected. Just comment the bugs in qpa_init_in() and
qpa_init_out() for now. Fixing these bugs may break glitch free
audio playback with fine tuned user audio settings.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-17-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
521ce71425 paaudio: remove unneeded code
Commit baea032ec7 "audio/paaudio: fix ignored buffer_length setting"
added code to handle buffer_length defaults. This was unnecessary
because the audio_buffer_* functions in audio/audio.c already handle
this. Remove the unneeded code.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-16-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
7007cd3fc8 paaudio: wait until the playback stream is ready
Don't call pa_stream_writable_size() in qpa_get_buffer_out()
before the playback stream is ready. This prevents a lot of the
following pulseaudio error messages.

pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state

To reproduce start qemu with
-parallel none -device gus,audiodev=audio0 -audiodev pa,id=audio0

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-15-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
e270c54826 paaudio: wait for PA_STREAM_READY in qpa_write()
Don't call pa_stream_writable_size() in qpa_write() before the
playback stream is ready. This prevents a lot of the following
pulseaudio error messages.

pulseaudio: pa_stream_writable_size failed
pulseaudio: Reason: Bad state

To reproduce start qemu with
-parallel none -device gus,audiodev=audio0
-audiodev pa,id=audio0,out.mixing-engine=off

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-14-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
bea29e9f2e paaudio: avoid to clip samples multiple times
The pulseaudio backend currently converts, clips and copies audio
playback samples in the mixing-engine sample buffer multiple
times.

In qpa_get_buffer_out() the function pa_stream_begin_write()
returns a rather large buffer and this allows audio_pcm_hw_run_out()
in audio/audio.c to copy all samples in the mixing-engine buffer
to the pulse audio buffer. Immediately after copying, qpa_write()
notices with a call to pa_stream_writable_size() that pulse audio
only needs a smaller part of the copied samples and ignores the
rest. This copy and ignore process happens several times for each
audio sample.

To fix this behaviour, call pa_stream_writable_size() in
qpa_get_buffer_out() to limit the number of samples
audio_pcm_hw_run_out() will convert. With this change the
pulseaudio pcm_ops functions put_buffer_out and write are no
longer identical and a separate qpa_put_buffer_out is needed.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-13-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
6fb0cd5054 audio: remove remaining unused plive code
Commit 73ad33ef7b "audio: remove plive" forgot to remove this code.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
bd37ede4eb sdlaudio: enable (in|out).mixing-engine=off
Enable the SDL2 backend options -audiodev sdl,out.mixing-
engine=off,in.mixing-engine=off.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
1d8549ad5e audio: break generic buffer dependency on mixing-engine
Break the unnecessary dependency of the generic buffer management
code on mixing-engine. This is required for the next patch.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
c2031dea89 sdlaudio: add recording functions
Add audio recording functions. SDL 2.0.5 or later is required to
use the recording functions. Playback continues to work with
earlier SDL 2.0 versions.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
a2893c8303 audio: split pcm_ops function get_buffer_in
Split off pcm_ops function run_buffer_in from get_buffer_in and
call run_buffer_in before get_buffer_in.

The next patch only needs the generic buffer management part
from audio_generic_get_buffer_in().

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
ce31f099fb sdlaudio: replace legacy functions with modern ones
With the modern audio functions it's possible to add new
features like audio recording.

As a side effect this patch fixes a bug where SDL2 can't be used
on Windows. This bug was reported on the qemu-devel mailing list at

https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg04043.html

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
e02d178f78 sdlaudio: fill remaining sample buffer with silence
Fill the remaining sample buffer with silence. To fill it with
zeroes is wrong for unsigned samples because this is silence
with a DC bias.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
bcce2ea5f6 sdlaudio: always clear the sample buffer
Always fill the remaining audio callback buffer with silence.
SDL 2.0 doesn't initialize the audio callback buffer. This was
an incompatible change compared to SDL 1.2. For reference read
the SDL 1.2 to 2.0 migration guide.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
14cefe14bb sdlaudio: don't start playback in init routine
Every emulated audio device has a way to enable audio playback. Don't
start playback until the guest enables the audio device. This patch
keeps the SDL2 device pause state in sync with hw->enabled.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Tested-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
5a0926c23f sdlaudio: add -audiodev sdl,out.buffer-count option
Currently there is a crackling noise with SDL2 audio playback.
Commit bcf19777df: "audio/sdlaudio: Allow audio playback with
SDL2" already mentioned the crackling noise.

Add an out.buffer-count option to give users a chance to select
sane settings for glitch free audio playback. The idea was taken
from the coreaudio backend.

The in.buffer-count option will be used with one of the next
patches.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
ff69c481a2 audio: fix bit-rotted code
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Volker Rümelin
ef26632e3a sdlaudio: remove leftover SDL1.2 code
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15 11:25:22 +01:00
Eduardo Habkost
ce35e2295e qdev: Move softmmu properties to qdev-properties-system.h
Move the property types and property macros implemented in
qdev-properties-system.c to a new qdev-properties-system.h
header.

Signed-off-by: Eduardo Habkost <ehabkost@redhat.com>
Reviewed-by: Igor Mammedov <imammedo@redhat.com>
Message-Id: <20201211220529.2290218-16-ehabkost@redhat.com>
Signed-off-by: Eduardo Habkost <ehabkost@redhat.com>
2020-12-18 15:20:17 -05:00
Gerd Hoffmann
06c8c37538 audio: add sanity check
Check whenever we actually found the spiceaudio driver
before flipping the can_be_default field.

Fixes: f0c4555edf ("audio: remove qemu_spice_audio_init()")
Buglink: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=977301
Reported-by: dann frazier <dann.frazier@canonical.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20201215081151.20095-1-kraxel@redhat.com>
2020-12-15 09:28:52 +01:00
Philippe Mathieu-Daudé
ab32b78cd1 audio: Simplify audio_bug() removing old code
This code (introduced in commit 1d14ffa97e, Oct 2005)
is likely unused since years. Time to remove it.  If
the condition is true, simply call abort().

Suggested-by: Gerd Hoffmann <gerd@kraxel.org>
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20201210223506.263709-1-philmd@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-12-15 09:23:14 +01:00
Volker Rümelin
ba6371b0c3 audio: remove unused function audio_is_cleaning_up()
The previous commit removed the last call site of
audio_is_cleaning_up(). Remove the now unused function.

Tested-by: Howard Spoelstra <hsp.cat7@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20201213130528.5863-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-12-15 09:14:17 +01:00
Volker Rümelin
ceb1165e9d coreaudio: always stop audio playback on shut down
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>
2020-12-15 09:14:17 +01:00
Volker Rümelin
53e78d1cfb coreaudio: don't start playback in init routine
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>
2020-12-15 09:14:17 +01:00
Volker Rümelin
1d47067394 coreaudio: rename misnamed variable fake_as
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>
2020-12-15 09:14:17 +01:00
Geoffrey McRae
a6e037390d audio/jack: fix use after free segfault
This change registers a bottom handler to close the JACK client
connection when a server shutdown signal is received. Without this
libjack2 attempts to "clean up" old clients and causes a use after free
segfault.

Signed-off-by: Geoffrey McRae <geoff@hostfission.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20201108063351.35804-2-geoff@hostfission.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-11-13 07:36:33 +01:00
Gerd Hoffmann
05b53636d0 spice: move add_interface() to QemuSpiceOps.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-id: 20201019075224.14803-6-kraxel@redhat.com
2020-10-21 15:46:14 +02:00
Gerd Hoffmann
d72c34cccc meson: add spice_headers dependency.
Used for files which (with CONFIG_SPICE=y) depend on spice header files
to pick up some enum, but which do not depend on on the actual spice
shared library.

Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20201014121120.13482-6-kraxel@redhat.com
2020-10-15 11:14:40 +02:00
Kevin Wolf
947e47448d monitor: Use getter/setter functions for cur_mon
cur_mon really needs to be coroutine-local as soon as we move monitor
command handlers to coroutines and let them yield. As a first step, just
remove all direct accesses to cur_mon so that we can implement this in
the getter function later.

Signed-off-by: Kevin Wolf <kwolf@redhat.com>
Message-Id: <20201005155855.256490-4-kwolf@redhat.com>
Reviewed-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
2020-10-09 07:08:19 +02:00
Peter Maydell
8c1c07929f Pull request
This includes the atomic_ -> qatomic_ rename that touches many files and is
 prone to conflicts.
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Merge remote-tracking branch 'remotes/stefanha/tags/block-pull-request' into staging

Pull request

This includes the atomic_ -> qatomic_ rename that touches many files and is
prone to conflicts.

# gpg: Signature made Wed 23 Sep 2020 17:08:43 BST
# gpg:                using RSA key 8695A8BFD3F97CDAAC35775A9CA4ABB381AB73C8
# gpg: Good signature from "Stefan Hajnoczi <stefanha@redhat.com>" [full]
# gpg:                 aka "Stefan Hajnoczi <stefanha@gmail.com>" [full]
# Primary key fingerprint: 8695 A8BF D3F9 7CDA AC35  775A 9CA4 ABB3 81AB 73C8

* remotes/stefanha/tags/block-pull-request:
  qemu/atomic.h: rename atomic_ to qatomic_
  tests: add test-fdmon-epoll
  fdmon-poll: reset npfd when upgrading to fdmon-epoll
  gitmodules: add qemu.org vbootrom submodule
  gitmodules: switch to qemu.org meson mirror
  gitmodules: switch to qemu.org qboot mirror
  docs/system: clarify deprecation schedule
  virtio-crypto: don't modify elem->in/out_sg
  virtio-blk: undo destructive iov_discard_*() operations
  util/iov: add iov_discard_undo()
  virtio: add vhost-user-fs-ccw device
  libvhost-user: handle endianness as mandated by the spec
  MAINTAINERS: add Stefan Hajnoczi as block/nvme.c maintainer

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
2020-09-24 18:48:45 +01:00
Stefan Hajnoczi
d73415a315 qemu/atomic.h: rename atomic_ to qatomic_
clang's C11 atomic_fetch_*() functions only take a C11 atomic type
pointer argument. QEMU uses direct types (int, etc) and this causes a
compiler error when a QEMU code calls these functions in a source file
that also included <stdatomic.h> via a system header file:

  $ CC=clang CXX=clang++ ./configure ... && make
  ../util/async.c:79:17: error: address argument to atomic operation must be a pointer to _Atomic type ('unsigned int *' invalid)

Avoid using atomic_*() names in QEMU's atomic.h since that namespace is
used by <stdatomic.h>. Prefix QEMU's APIs with 'q' so that atomic.h
and <stdatomic.h> can co-exist. I checked /usr/include on my machine and
searched GitHub for existing "qatomic_" users but there seem to be none.

This patch was generated using:

  $ git grep -h -o '\<atomic\(64\)\?_[a-z0-9_]\+' include/qemu/atomic.h | \
    sort -u >/tmp/changed_identifiers
  $ for identifier in $(</tmp/changed_identifiers); do
        sed -i "s%\<$identifier\>%q$identifier%g" \
            $(git grep -I -l "\<$identifier\>")
    done

I manually fixed line-wrap issues and misaligned rST tables.

Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Acked-by: Paolo Bonzini <pbonzini@redhat.com>
Message-Id: <20200923105646.47864-1-stefanha@redhat.com>
2020-09-23 16:07:44 +01:00
Gerd Hoffmann
5e626fa736 audio: build spiceaudio as module
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20200916084117.21828-3-kraxel@redhat.com
2020-09-23 08:36:50 +02:00
Gerd Hoffmann
f0c4555edf audio: remove qemu_spice_audio_init()
Handle the spice special case in audio_init instead.

With the qemu_spice_audio_init() symbol dependency being
gone we can build spiceaudio as module.

Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20200916084117.21828-2-kraxel@redhat.com
2020-09-23 08:36:50 +02:00
Volker Rümelin
a8a98cfd42 audio: run downstream playback queue unconditionally
Run the downstream playback queue even if there are no samples
in the mixing engine buffer. The downstream queue may still have
queued samples.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-7-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
2d8823077e audio: align audio_generic_write with audio_pcm_hw_run_out
The function audio_generic_write should work exactly like
audio_pcm_hw_run_out. It's a very similar function working on a
different buffer.

This patch significantly reduces the number of drop-outs with
the DirectSound backend. To hear the difference start qemu with
-audiodev dsound,id=audio0,out.mixing-engine=off and play a
song in the guest with and without this patch.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-6-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
ac221f45e3 audio: remove unnecessary calls to put_buffer_in
This patch removes unnecessary calls to the pcm_ops function
put_buffer_in(). No audio backend needs this call if the
returned length of pcm_ops function get_buffer_in() is zero.

For the DirectSound backend this prevents a call to
dsound_unlock_in() without a preceding call to dsound_lock_in().
While Windows doesn't complain it seems wrong anyway.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
b9896dc5be audio: align audio_generic_read with audio_pcm_hw_run_in
The function audio_generic_read should work exactly like
audio_pcm_hw_run_in. It's a very similar function working
on a different buffer.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-4-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
aec6d0dc4e audio/spiceaudio: always rate limit playback stream
The playback rate with the spiceaudio backend is currently too
fast if there's no spice client connected or the spice client
can't play audio. Rate limit the audio playback stream in all
cases. To calculate the rate correctly the limiter has to know
the maximum buffer size.

Fixes: 8c198ff065 ("spiceaudio: port to the new audio backend api")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
4c3356f965 audio/audio: fix video playback slowdown with spiceaudio
This patch allows the audio backends get_buffer_out() functions
to drop audio data and mitigates a bug reported on the qemu-devel
mailing list.

https://lists.nongnu.org/archive/html/qemu-devel/2020-09/msg03832.html

The new rules for the variables buf and size returned by
get_buffer_out() are:
size == 0: Downstream playback buffer is full. Retry later.
size > 0, buf != NULL: Copy size bytes to buf for playback.
size > 0, buf == NULL: Drop size bytes.

The audio playback rate with spiceaudio for the no audio case is
too fast, but that's what we had before commit fb35c2cec5
"audio/dsound: fix invalid parameters error". The complete fix
comes with the next patch.

Reported-by: Qi Zhou <atmgnd@outlook.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
Volker Rümelin
d4b70fa4ed audio: handle buf == NULL in put_buffer_out()
With the next patch all audio backends put_buffer_out() functions
have to handle the buf == NULL case, provided the get_buffer_out()
function may return buf = NULL and size > 0.

It turns out that all audio backends get_buffer_out() functions
either can't return buf = NULL or return buf = NULL and size = 0
at the same time. The only exception is the spiceaudio backend
where size may be uninitialized.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-1-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-09-23 08:19:42 +02:00
zhaolichang
e3a6e0daf4 qemu/: fix some comment spelling errors
I found that there are many spelling errors in the comments of qemu,
so I used the spellcheck tool to check the spelling errors
and finally found some spelling errors in the folder.

Signed-off-by: zhaolichang <zhaolichang@huawei.com>
Reviewed-by: Alex Bennee <alex.bennee@linaro.org>
Message-Id: <20200917075029.313-2-zhaolichang@huawei.com>
Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2020-09-17 20:35:43 +02:00
Markus Armbruster
6ec9379870 trace-events: Delete unused trace points
Tracked down with the help of scripts/cleanup-trace-events.pl.

Signed-off-by: Markus Armbruster <armbru@redhat.com>
Message-id: 20200806141334.3646302-4-armbru@redhat.com
Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
2020-09-09 17:17:02 +01:00
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
Kővágó, Zoltán
d1670b20dc audio: fix parameter dereference before NULL check
This should fix Coverity issues CID 1405305 and 1405301.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 0eadcc88b8421bb86ce2d68ac70517f920c3ad6c.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
571a8c522e audio: split ctl_* functions into enable_* and volume_*
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>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
857271a29c audio: common rate control code for timer based outputs
This commit removes the ad-hoc rate-limiting code from noaudio and
wavaudio, and replaces them with a (slightly modified) code from
spiceaudio.  This way multiple write calls (for example when the
circular buffer wraps around) do not cause problems.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: fd0fe5b95b13fa26d09ae77a72f99d0ea411de14.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
dc88e38fa7 audio: unify input and output mixeng buffer management
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before
initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the
initial size of the STSampleBuffer.  The next commit tries to fix this
inconsistency.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: a78caeb2eeb6348ecb45bb2c81709570ef8ac5b3.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
3f5bbfc25a audio: remove remains of the old backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 497decab6d0f0fb9529bea63ec7ce0bd7b553038.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
ef3612e11b wavaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: eede77aeb9c17b379948b0b6d2ac10f45d74fa62.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
8c198ff065 spiceaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 4d3356df9ccbffee2f710b93d456443c81e3f011.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
ff71876766 sdlaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: ac1722a03fb1b530c2081f46585ce7fa80ebef6c.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
49ddd7e122 paaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 21fe8f2cf949039c8c40a0352590c593b104917d.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
3ba4066d08 ossaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 22ab335146acd8099779583edcf6ed46de836bd6.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
affc691a14 noaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 10eebdd2e1529c2bd403ef98dd9d346c6d4ca3d1.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
7fa9754ac8 dsoundaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 2ca925ab551ea832c930fc2db213a9e73d8dab7f.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
2ceb8240fa coreaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 586a1e66de5cbc6c5234f9ae556d24befb6afada.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
286a5d201e alsaaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: ab9768e73dfe7b7305bd6a51629846e0d77622a5.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
ff095e5231 audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only).  In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.

Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases.  Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required.  audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23 12:28:47 +02:00
Kővágó, Zoltán
4b3b7793e1 audio: omitting audiodev= parameter is only deprecated
Unfortunately, changes introduced in af2041ed2d "audio: audiodev=
parameters no longer optional when -audiodev present" breaks backward
compatibility.  This patch changes the error into a deprecation warning.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 02d4328c33455742d01e0b62395013e95293c3ba.1566847960.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-28 11:57:45 +02:00
Kővágó, Zoltán
725662d6db audio: fix invalid malloc size in audio_create_pdos
The code used sizeof(AudiodevAlsaPerDirectionOptions) instead of the
appropriate per direction options for the audio backend.  If the size of
the actual audiodev's per direction options are larger than alsa's, it
could cause a buffer overflow.

However, alsa has three fields in per direction options: a string, an
uint32 and a bool.  Oss has the same fields, coreaudio has a single
uint32, paaudio has a string and an uint32, all other backends only use
the common options, so currently no per direction options struct should
be larger than alsa's.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-Id: <7808bc816ba7da8b8de8a214713444d85f7af3c6.1566847960.git.DirtY.iCE.hu@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-28 11:56:56 +02:00
Kővágó, Zoltán
e76ba19a1f audio: fix memory leak reported by ASAN
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: ed35e9e72aa77c9376e9c8a8f3a5443703fe6fbe.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-21 09:13:37 +02:00
Kővágó, Zoltán
7520462bc1 audio: use size_t where makes sense
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: c5193e687fc6cc0f60cb3e90fe69ddf2027d0df1.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-21 09:13:37 +02:00