The virtio-sound device is currently not migratable. QEMU crashes
on the source machine at some point during the migration with a
segmentation fault.
Even with this bug fixed, the virtio-sound device doesn't migrate
the state of the audio streams. For example, running streams leave
the device on the destination machine in a broken condition.
Mark the device as unmigratable until these issues have been fixed.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20231204072837.6058-1-vr_qemu@t-online.de>
Commit 6f189a08c1 ("ui/gtk-egl: Check EGLSurface before doing
scanout") introduced a regression when QEMU is running with a
virtio-gpu-gl-device on a host under X11. After the guest has
initialized the virtio-gpu-gl-device, the guest screen only
shows "Display output is not active.".
Commit 6f189a08c1 moved all function calls in
gd_egl_scanout_texture() to a code path which is only called
once after gd_egl_init() succeeds in gd_egl_scanout_texture().
Move all function calls in gd_egl_scanout_texture() back to
the regular code path so they get always called if one of the
gd_egl_init() calls was successful.
Fixes: 6f189a08c1 ("ui/gtk-egl: Check EGLSurface before doing scanout")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20231111104020.26183-1-vr_qemu@t-online.de>
Commit b7639b7dd0 ("hw/audio: Simplify hda audio init") inverted
the sense of hda codec property mixer during initialization.
Change the code so that mixer=on enables the hda mixer emulation
and mixer=off disables the hda mixer emulation.
With this change audio playback and recording streams don't start
muted by default.
Fixes: b7639b7dd0 ("hw/audio: Simplify hda audio init")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20231105172552.8405-2-vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
After a relatively short time, there is an multiplication overflow
when multiplying (now - buft_start) with hda_bytes_per_second().
While the uptime now - buft_start only overflows after 2**63 ns
= 292.27 years, this happens hda_bytes_per_second() times faster
with the multiplication. At 44100 samples/s * 2 channels
* 2 bytes/channel = 176400 bytes/s that is 14.52 hours. After the
multiplication overflow the affected audio stream stalls.
Replace the multiplication and following division with muldiv64()
to prevent a multiplication overflow.
Fixes: 280c1e1cdb ("audio/hda: create millisecond timers that handle IO")
Reported-by: M_O_Bz <m_o_bz@163.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20231105172552.8405-1-vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
The virtio sound device is currently an unclassified PCI device.
~> sudo lspci -s '00:02.0' -v -nn | head -n 2
00:02.0 Unclassified device [00ff]:
Red Hat, Inc. Device [1af4:1059] (rev 01)
Subsystem: Red Hat, Inc. Device [1af4:1100]
Set the correct PCI class code to change the device to a
multimedia audio controller.
~> sudo lspci -s '00:02.0' -v -nn | head -n 2
00:02.0 Multimedia audio controller [0401]:
Red Hat, Inc. Device [1af4:1059] (rev 01)
Subsystem: Red Hat, Inc. Device [1af4:1100]
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20231107185034.6434-1-vr_qemu@t-online.de>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Reviewed-by: Manos Pitsidianakis <manos.pitsidianakis@linaro.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
Signed-off-by: Michael S. Tsirkin <mst@redhat.com>
Reset the current sample counter when writing the Channel Sample
Count Register. The Linux ens1370 driver and the AROS sb128
driver expect the current sample counter counts down from sample
count to 0 after a write to the Channel Sample Count Register.
Currently the current sample counter starts from 0 after a reset
or the last count when the counter was stopped.
The current sample counter is used to raise an interrupt whenever
a complete buffer was transferred. When the counter starts with a
value lower than the reload value, the interrupt triggeres before
the buffer was completly transferred. This may lead to corrupted
audio streams.
Tested-by: Rene Engel <ReneEngel80@emailn.de>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Tested-by: BALATON Zoltan <balaton@eik.bme.hu>
Message-Id: <20230917065813.6692-1-vr_qemu@t-online.de>
Windows sends an extra left control key up/down input event for
every right alt key up/down input event for keyboards with
international layout. Since commit 830473455f ("ui/sdl2: fix
handling of AltGr key on Windows") QEMU uses a Windows low level
keyboard hook procedure to reliably filter out the special left
control key and to grab the keyboard on Windows.
The SDL2 version 2.0.16 introduced its own Windows low level
keyboard hook procedure to grab the keyboard. Windows calls this
callback before the QEMU keyboard hook procedure. This disables
the special left control key filter when the keyboard is grabbed.
To fix the problem, disable the SDL2 Windows low level keyboard
hook procedure.
Reported-by: Bernhard Beschow <shentey@gmail.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Tested-by: Bernhard Beschow <shentey@gmail.com>
Message-Id: <20230418062823.5683-1-vr_qemu@t-online.de>
Simplify the resample buffer size calculation.
For audio playback we have
sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq;
samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
This can be simplified to
samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq);
For audio recording we have
sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq;
samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
This can be simplified to
samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq);
With hw = sw->hw this becomes in both cases
samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
Now that sw->ratio is no longer needed, remove sw->ratio.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de>
Upsampling may leave one remaining audio frame in the input
buffer. The emulated audio playback devices are currently
resposible to write this audio frame again in the next write
cycle. Push that task down to audio_pcm_sw_write.
This is another step towards an audio callback interface that
guarantees that when audio frontends are told they can write
n audio frames, they can actually do so.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de>
Introduce the new function st_rate_frames_out() to calculate the
exact number of audio output frames the resampling code can
generate from a given number of audio input frames. When upsampling,
this function returns the maximum number of output frames.
This new function replaces the audio_frontend_frames_in()
function, which calculated the average number of output frames
rounded down to the nearest integer. The audio_frontend_frames_in()
function was additionally used to limit the number of output frames
to the resample buffer size. In audio_pcm_sw_read() the variable
resample_buf.size replaces the open coded audio_frontend_frames_in()
function. In audio_run_in() an additional MIN() function is
necessary.
After this patch the audio packet length calculation for audio
recording is exact.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-12-vr_qemu@t-online.de>
Replace the resampling loop in audio_pcm_sw_read() with the new
function audio_pcm_sw_resample_in(). Unlike the old resample
loop the new function will try to consume input frames even if
the output buffer is full. This is necessary when downsampling
to avoid reading less audio frames than calculated in advance.
The loop was unrolled to avoid complicated loop control conditions
in this case.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-10-vr_qemu@t-online.de>
Introduce the new function st_rate_frames_in() to calculate the
exact number of audio input frames needed to get a given number
of audio output frames. The exact number of frames depends only
on the difference of opos - ipos and the number of output frames.
When downsampling, this function returns the maximum number of
input frames needed.
This new function replaces the audio_frontend_frames_out() function,
which calculated the average number of input frames rounded down
to the nearest integer. Because audio_frontend_frames_out() also
limited the number of input frames to the size of the resample
buffer, st_rate_frames_in() is not a direct replacement and two
additional MIN() functions are needed. One to prevent resample
buffer overflows and one to limit the available bytes for the audio
frontends.
After this patch the audio packet length calculation for playback is
exact. When upsampling, it's still possible that the audio frontends
can't write the last audio frame. This will be fixed later.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de>
The audio_pcm_sw_write() function is intended to convert a
PCM audio stream to the internal representation, adjust the
volume, and then mix it with the other audio streams with a
possibly changed sample rate in mix_buf. In order for the
audio_capture_mix_and_clear() function to use audio_pcm_sw_write(),
it must bypass the first two tasks of audio_pcm_sw_write().
Since patch "audio: split out the resampling loop in
audio_pcm_sw_write()" this is no longer necessary, because now
the audio_pcm_sw_resample_out() function can be used instead of
audio_pcm_sw_write().
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-7-vr_qemu@t-online.de>
Replace the resampling loop in audio_pcm_sw_write() with the new
function audio_pcm_sw_resample_out(). Unlike the old resample
loop the new function will try to consume input frames even if
the output buffer is full. This is necessary when downsampling
to avoid reading less audio frames than calculated in advance.
The loop was unrolled to avoid complicated loop control conditions
in this case.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-4-vr_qemu@t-online.de>
Read the maximum possible number of audio frames instead of the
minimum necessary number of frames when the audio stream is
downsampled and the output buffer is limited. This makes the
function symmetrical to upsampling when the input buffer is
limited. The maximum possible number of frames is written here.
With this change it's easier to calculate the exact number of
audio frames the resample function will read or write. These two
functions will be introduced later.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-3-vr_qemu@t-online.de>
Change the type of the resample buffer from struct st_sample *
to STSampleBuffer. Also change the name from buf to resample_buf
for better readability.
The new variables resample_buf.size and resample_buf.pos will be
used after the next patches. There is no functional change.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-2-vr_qemu@t-online.de>
Change the type of mix_buf in struct HWVoiceOut and conv_buf
in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer.
However, a buffer pointer is still needed. For this reason in
struct STSampleBuffer samples[] is changed to *buffer.
This is a preparation for the next patch. The next patch will
add this line, which is not possible with the current struct
STSampleBuffer definition.
+ sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
There are no functional changes.
Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-1-vr_qemu@t-online.de>
Audio recording with ALSA default settings currently doesn't
work. The debug log shows updates every 0.75s and 1.5s.
audio: Elapsed since last alsa run (running): 0.743030
audio: Elapsed since last alsa run (running): 1.486048
audio: Elapsed since last alsa run (running): 0.743008
audio: Elapsed since last alsa run (running): 1.485878
audio: Elapsed since last alsa run (running): 1.486040
audio: Elapsed since last alsa run (running): 1.485886
The time between updates should be in the 10ms range. Audio
recording with ALSA has the same timing contraints as playback.
Reintroduce the default recording settings and use the same
default settings for recording as for playback.
The term "reintroduce" is correct because commit a93f328177
("alsaaudio: port to -audiodev config") removed the default
settings for recording.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20230121094735.11644-11-vr_qemu@t-online.de>
The currently used default playback settings in the ALSA audio
backend are a bit unfortunate. With a few emulated audio devices,
audio playback does not work properly. Here is a short part of
the debug log while audio is playing (elapsed time in seconds).
audio: Elapsed since last alsa run (running): 0.046244
audio: Elapsed since last alsa run (running): 0.023137
audio: Elapsed since last alsa run (running): 0.023170
audio: Elapsed since last alsa run (running): 0.023650
audio: Elapsed since last alsa run (running): 0.060802
audio: Elapsed since last alsa run (running): 0.031931
For some audio devices the time of more than 23ms between updates
is too long.
Set the period time to 5.8ms so that the maximum time between
two updates typically does not exceed 11ms. This roughly matches
the 10ms period time when doing playback with the audio timer.
After this patch the debug log looks like this.
audio: Elapsed since last alsa run (running): 0.011919
audio: Elapsed since last alsa run (running): 0.005788
audio: Elapsed since last alsa run (running): 0.005995
audio: Elapsed since last alsa run (running): 0.011069
audio: Elapsed since last alsa run (running): 0.005901
audio: Elapsed since last alsa run (running): 0.006084
Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20230121094735.11644-10-vr_qemu@t-online.de>
Use g_malloc0() as a direct replacement for audio_calloc().
Since the type of the parameter n_bytes of the function g_malloc0()
is unsigned, the type of the variables voice_size_out and
voice_size_in has been changed to size_t. This means that the
function argument no longer has to be checked for negative values.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20230121094735.11644-7-vr_qemu@t-online.de>
Some emulated audio devices allow guests to select very low
sample rates that the audio subsystem doesn't support. The lowest
supported sample rate depends on the audio backend used and in
most cases can be changed with various -audiodev arguments. Until
now, the audio_bug function emits an error message similar to the
following error message
A bug was just triggered in audio_calloc
Save all your work and restart without audio
I am sorry
Context:
audio_pcm_sw_alloc_resources_out passed invalid arguments to
audio_calloc
nmemb=0 size=16 (len=0)
audio: Could not allocate buffer for `ac97.po' (0 samples)
and the audio subsystem continues without sound for the affected
device.
The fact that the selected sample rate is not supported is not a
guest error. Instead of displaying an error message, the missing
audio support is now logged. Simply continuing without sound is
correct, since the audio stream won't transport anything
reasonable at such high resample ratios anyway.
The AUD_open_* functions return NULL like before. The opened
audio device will not be registered in the audio subsystem and
consequently the audio frontend callback functions will not be
called. The AUD_read and AUD_write functions return early in this
case. This is necessary because, for example, the Sound Blaster 16
emulation calls AUD_write from the DMA callback function.
Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
There are corner cases where rate->opos can overflow. For
example, if QEMU is started with -audiodev pa,id=audio0,
out.frequency=11025 -device ich9-intel-hda -device hda-duplex,
audiodev=audio0 and the guest plays audio with a sampling
frequency of 44100Hz, rate->opos will overflow after 27.05h
and the audio stream will be silent for a long time.
To prevent a rate->opos and also a rate->ipos overflow, both
are wrapped around after a short time. The wrap around point
rate->ipos >= 0x10001 is an arbitrarily selected value and can
be any small value, 0 and 1 included.
The comment that an ipos overflow will result in an infinite
loop has been removed, because in this case the resampling code
only generates no more output samples and the audio stream stalls.
However, there is no infinite loop.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220923183640.8314-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The calculation of the buffer size needed to store audio samples
after resampling is wrong for audio recording. For audio recording
sw->ratio is calculated as
sw->ratio = frontend sample rate / backend sample rate.
From this follows
frontend samples = frontend sample rate / backend sample rate
* backend samples
frontend samples = sw->ratio * backend samples
In 2 of 3 places in the audio recording code where sw->ratio
is used in a calculation to get the number of frontend frames,
the calculation is wrong. Fix this. The 3rd formula in
audio_pcm_sw_read() is correct.
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/71
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The next patch needs two new rate control functions. The first
one returns the bytes needed at call time to maintain the
selected rate. The second one adjusts the bytes actually sent.
Split the audio_rate_get_bytes() function into these two
functions and reintroduce audio_rate_get_bytes().
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Run the downstream playback queue even if the emulated audio
device didn't write new samples. There still may be buffered
audio samples downstream.
This is for the -audiodev out.mixing-engine=off case. Commit
a8a98cfd42 ("audio: run downstream playback queue uncondition-
ally") fixed the out.mixing-engine=on case.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Fix GUS audio playback with out.mixing-engine=off.
The GUS audio device needs to know the amount of samples to
produce in advance.
To reproduce start qemu with
-parallel none -device gus,audiodev=audio0
-audiodev pa,id=audio0,out.mixing-engine=off
and start the cartoon.exe demo in a FreeDOS guest. The demo file
is available on the download page of the GUSemu32 author.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit ab32b78cd1 "audio: Simplify audio_bug() removing old code"
introduced abort() in audio_bug() for regular builds.
audio_bug() was never meant to abort QEMU for the following
reasons.
- There's code in audio_bug() that expects audio_bug() gets
called more than once with error condition true. The variable
'shown' is only 0 on first error.
- All call sites test the return code of audio_bug(), print
an error context message and handle the errror.
- The abort() in audio_bug() enables a class of guest-triggered
aborts similar to the Launchpad Bug #1910603 at
https://bugs.launchpad.net/bugs/1910603.
Fixes: ab32b78cd1 "audio: Simplify audio_bug() removing old code"
Buglink: https://bugs.launchpad.net/bugs/1910603
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220917131626.7521-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The png_destroy_write_struct() function frees all memory used by
libpng. Don't use the glib auto cleanup mechanism to free the
memory allocated by libpng again. For the pixman image, use only the
auto cleanup mechanism and remove the qemu_pixman_image_unref()
function call to prevent another double free.
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/1210
Fixes: 9a0a119a38 ("Added parameter to take screenshot with screendump as PNG")
Tested-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220919061956.30929-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Since about 2018 virglrenderer (commit fa835b0f88 "vrend: don't
hardcode context version") tries to open the highest available GL
context version. This is done by creating the known GL context
versions from the highest to the lowest until (*create_gl_context)
returns a context != NULL.
This does not work properly with
the current QEMU gd_gl_area_create_context() function, because
gdk_gl_context_realize() on Wayland creates a version 3.0 legacy
context if the requested GL context version can't be created.
In order for virglrenderer to find the highest available GL
context version, return NULL if the created context version is
lower than the requested version.
This fixes the following error:
QEMU started with -device virtio-vga-gl -display gtk,gl=on.
Under Wayland, the guest window remains black and the following
information can be seen on the host.
gl_version 30 - compat profile
(qemu:5978): Gdk-WARNING **: 16:19:01.533:
gdk_gl_context_set_required_version
- GL context versions less than 3.2 are not supported.
(qemu:5978): Gdk-WARNING **: 16:19:01.537:
gdk_gl_context_set_required_version -
GL context versions less than 3.2 are not supported.
(qemu:5978): Gdk-WARNING **: 16:19:01.554:
gdk_gl_context_set_required_version -
GL context versions less than 3.2 are not supported.
vrend_renderer_fill_caps: Entering with stale GL error: 1282
To reproduce this error, an OpenGL driver is required on the host
that doesn't have the latest OpenGL extensions fully implemented.
An example for this is the Intel i965 driver on a Haswell processor.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220605085131.7711-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The counterpart function for gd_gl_area_create_context() is
currently empty. Implement the gd_gl_area_destroy_context()
function to avoid GL context leaks.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220605085131.7711-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Now that the mixing buffer size no longer adds to playback
latency, fix the samples vs. frames mix-up in the mixing buffer
size calculation. This change will go largely unnoticed as long
as the user doesn't use a buffer-size smaller than timer-period.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-14-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Return the free buffer size for the mmapped case in function
oss_buffer_get_free() to reduce the effective playback buffer
size. All intermediate audio playback buffers become temporary
buffers.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-13-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit ff095e5231 "audio: api for mixeng code free backends"
introduced another FIFO for the audio subsystem with exactly the
same size as the mixing-engine FIFO. Most audio backends use
this generic FIFO. The generic FIFO used together with the
mixing-engine FIFO doubles the audio FIFO size, because that's
just two independent FIFOs connected together in series.
For audio playback this nearly doubles the playback latency.
This patch restores the effective mixing-engine playback buffer
size to a pre v4.2.0 size by only accepting the amount of
samples for the mixing-engine queue which the downstream queue
accepts.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
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>
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>
Current code creates directories with mode 0644. Even the creator
can't create files in the new directory. Set all x mode flags in
variable mask and clear all x mode flags in function open() to
preserve the current open mode.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220122140619.7514-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Quote from:
High Definition Audio Specification 1.0a, section 3.3.35
Offset 80: {IOB}SDnCTL Stream Reset (SRST): Writing a 1 causes
the corresponding stream to be reset. The Stream Descriptor
registers (except the SRST bit itself) ... are reset.
Change the code to reset the Stream Descriptor Control and Status
registers except the SRST bit.
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/757
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20211226154017.6067-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
The test if the chardev frontend is connected in
kbd_put_keysym_console() is redundant, because the call
to qemu_chr_be_can_write() in kbd_send_chars() tests
the connected condition again.
Remove the redundant test whether the chardev frontend
is connected.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210916192239.18742-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
A PS/2 keyboard has a separate command reply queue that is
independent of the key queue. This prevents that command replies
and keyboard input mix. Keyboard command replies take precedence
over queued keystrokes. A new keyboard command removes any
remaining command replies from the command reply queue.
Implement a separate keyboard command reply queue and clear the
command reply queue before command execution. This brings the
PS/2 keyboard emulation much closer to a real PS/2 keyboard.
The command reply queue is located in a few free bytes directly
in front of the scancode queue. Because the scancode queue has
a maximum length of 16 bytes there are 240 bytes available for
the command reply queue. At the moment only a maximum of 3 bytes
are required. For compatibility reasons rptr, wptr and count kept
their function. rptr is the start, wptr is the end and count is
the length of the entire keyboard queue. The new variable cwptr
is the end of the command reply queue or -1 if the queue is
empty. To write to the command reply queue, rptr is moved
backward by the number of required bytes and the command replies
are written to the buffer starting at the new rptr position.
After writing, cwptr is at the old rptr position. Copying cwptr
to rptr clears the command reply queue. The command reply queue
can't overflow because each new keyboard command clears the
command reply queue.
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/501
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/502
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210810133258.8231-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Extend the used ps2 buffer size to the available buffer size but
keep the maximum ps2 queue size.
The next patch needs a few bytes of the larger buffer size.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210810133258.8231-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit 584af1f1d9 ("ui/gtk: add a keyboard fifo to the VTE
consoles") changed the VTE chardev backend code to rely on the
chr_accept_input() callback function. The code expects a
chr_accept_input() call whenever qemu_chr_be_can_write() bytes
were written. It turns out this is wrong. Some chardev
frontends only call this callback after can_write was 0.
Change the code to send data until the keyboard fifo is empty
or qemu_chr_be_can_write() returns 0.
Fixes: 584af1f1d9 ("ui/gtk: add a keyboard fifo to the VTE consoles")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20210810063257.17411-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Since commit 8eb13bbbac ("ui/gtk: vte: fix sending multiple
characeters") it's very easy to lock up QEMU with the GTK ui.
If you configure a guest with a serial device and the guest
doesn't listen on this device, QEMU will lock up after
entering two characters in the serial console. That's because
current code uses a busy loop for the chardev write retries
and the busy loop doesn't terminate in this case.
To fix this problem add a fifo to the VTE consoles and use the
chr_accept_input() callback function to write the remaining
characters in the queue to the chardev.
The fifo has a size of 4096 bytes, so one can copy and paste
a fairly large URL or file path.
Fixes: 8eb13bbbac ("ui/gtk: vte: fix sending multiple characeters")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210725165039.5242-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
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>
The variable outport_present is a flag to show if the outport
subsection was loaded. Clear the outport_present flag in the
outer pre_load() function. This method is recommended in the QEMU
manual for developers in the chapter "Migration, Subsections".
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
In 2005 the author of commit daa579632d "PS2 mouse and keyboard
separation (Paul Brook)" and 0e43e99c04 "PS2 mouse and keyboard
separation (Paul Brook)" separated the PS/2 controller code and
the PS/2 keyboard and mouse code. It seems he forgot to remove
a few defines. Remove them now.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently the PS/2 controller command KBD_CCMD_MOUSE_DISABLE
doesn't disable the PS/2 mouse communication at all, and the
PS/2 controller commands KBD_CCMD_KBD_DISABLE and
KBD_CCMD_KBD_ENABLE only disable and enable the keyboard
interrupt, which is very different from what a real PS/2
controller does. A guest may notice the difference.
Mask out pending data on disabled queues to correctly disable
the PS/2 controller communication.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Add a separate queue for PS/2 controller responses. The
responses no longer get queued in the keyboard or mouse queues.
The advantage of this can be seen after the next two patches,
where the guest can disable the PS/2 communication with keyboard
and mouse and still talk to the PS/2 controller.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently there is only one flag to distinguish between two
interrupt sources and there are no available flags for more
sources. Add an internal state variable to store the interrupt
source. The next patch will introduce an additional interrupt
source. There is no functional change.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Limit the keyboard data rate to the serial link speed. Some old
DOS software relies on being able to read an incoming scan-code
more than once. After reading keyboard data from the i8042
controller, the guest software has 1ms to read the same data
again.
Use -global i8042.kbd-throttle=on to enable this feature.
To see how this patch works, start a FreeDOS 1.2 guest with the
qemu option -global i8042.kbd-throttle=on and open a text file
with the vim 7.3 32 bit for DOS executable. Then use the cursor
keys (not the cursor keys on the numeric keypad) to move through
the text. Without the kbd-throttle option enabled each keystroke
will move the cursor two positions.
Buglink: https://bugs.launchpad.net/bugs/1895363
Buglink: https://bugs.launchpad.net/bugs/1897568
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210525181441.27768-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Volker Rümelin