Hi,
a better way to fix the playback glitches is to use a bigger playback
buffer on pulseaudio server side. I suggest you replace your patch with
a patch like this one:
diff --git a/audio/paaudio.c b/audio/paaudio.c
index fea6071..8bd5b91 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -554
Current code doesn't provide pulseaudio buffer attributes for
recording. Without buffer attributes pulseaudio uses a default
buffer of 4MB. 4MB is approximately 24s 16bit stereo audio
data at 44.1kHz.
This patch fixes bug #1492649
|Signed-off-by: Volker Rümelin |
---
audio/paaudio.
Hi,
Am 12.09.2015 um 01:22 schrieb Marc-André Lureau:
Current code doesn't provide pulseaudio buffer attributes for
recording. Without buffer attributes pulseaudio uses a default
buffer of 4MB. 4MB is approximately 24s 16bit stereo audio
data at 44.1kHz.
Why isn't the buffer processed as soon
Am 12.09.2015 um 16:35 schrieb Kővágó Zoltán:
2015-09-12 13:23 keltezéssel, Volker Rümelin írta:
On start up qemu opens a connection to pulseaudio in function
qpa_init_in and pulseaudio immediately starts recording to the 4MB
ringbuffer. The qemu guest, Windows 8.1 in my case, doesn't co
From: Igor Mammedov
PCI-ISA bridges that are built in PIIX/Q35 are building its own AML
using AcpiDevAmlIf interface. Now build_append_pci_bus_devices()
gained AcpiDevAmlIf interface support to get AML of devices atached
to PCI slots.
So drop ad-hoc build_q35_isa_bridge()/build_piix4_isa_bridge()
Am 18.11.22 um 15:55 schrieb Igor Mammedov:
On Fri, 18 Nov 2022 14:08:36 +0100
Igor Mammedov wrote:
On Thu, 17 Nov 2022 22:51:46 +0100
Volker Rümelin wrote:
[...]
since this patch SeaBIOS no longer detects the PS/2 keyboard. This means
there's no keyboard in SeaBIOS, GRUB or FreeDOS.
Am 17.07.23 um 14:45 schrieb marcandre.lur...@redhat.com:
From: Dongwon Kim
Setting scanout mode is better to be done very last minute
right because the mode can be reset anytime after it is set in
dpy_gl_scanout_texture by any asynchronouse dpy_refresh call,
which eventually cancels drawing of
backing_height,
backing_id, false);
}
Thanks!
Adding back this line fixes the black guest screen when running the host
under X11. For Wayland I had to add back the similar line in
ui/gtk-gl-area.c.
With best regards,
Volker
On 7/20/2023 11:53 PM, Volker Rümelin wrote:
Am 05.01.24 um 12:36 schrieb Marc-André Lureau:
> Hi
>
> On Fri, Jan 5, 2024 at 12:34 AM Volker Rümelin wrote:
>> The payload size returned by command VIRTIO_SND_R_PCM_INFO is
>> wrong. The code in process_cmd() assumes that all commands
>> return only a virtio_snd_hdr
iable on reset. Once a stream has been opened, it will only
be closed after a reset or when QEMU shuts down.
Patch 10/11 "add missing vmstate fields" resets the inuse variables on
reset.
Volker Rümelin (11):
hw/audio/virtio-sound: return correct command response size
hw/audio/virti
virtio_snd_ctrl_command to allow for additional payloads.
Reviewed-by: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 7 +--
include/hw/audio/virtio-snd.h | 1 +
2 files changed, 6 insertions(+), 2 deletions(-)
diff --git a/hw/audio/virtio-snd.c b/hw/audio
after migration.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 84 +++
include/hw/audio/virtio-snd.h | 1 +
2 files changed, 77 insertions(+), 8 deletions(-)
diff --git a/hw/audio/virtio-snd.c b/hw/audio/virtio-snd.c
index 06a27ef8d9..b0a0ff2456
immediately after allocation.
This makes it easier to remove the virtio_snd_set_pcm_params()
and virtio_snd_pcm_prepare() calls in the realisation phase and
to migrate the audio streams of the virtio sound device after
the next few patches.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c
issues. However,
the placeholder makes it possible to add code in a backwards and
forwards compatible way.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 2 ++
include/hw/audio/virtio-snd.h | 2 ++
2 files changed, 4 insertions(+)
diff --git a/hw/audio/virtio-snd.c b/hw/audio
, only one
copy of the audio stream parameters is required and these do not
need to be initialised with default values.
The state variable will also make it easier to restore the audio
stream after migration.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 213
All code in virtio-snd.c runs with the BQL held. Remove the
command queue mutex and the stream queue mutexes. The qatomic
functions are also not needed.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 215 +++---
include/hw/audio/virtio-snd.h | 3
.
Don't try to handle the invalid virtio queue element with a stream
queue. Instead, push the invalid queue element back to the guest
immediately.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 100 ++
include/hw/audio/virtio-snd.h | 1
All remaining stream buffers in the stream queues must
be freed after a reset. This is the initial state of the
virtio-sound device.
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 11 +++
1 file changed, 11 insertions(+)
diff --git a/hw/audio/virtio-snd.c b/hw/audio/virtio
Split out virtio_snd_pcm_start_stop(). This is a preparation
for the next patch so that it doesn't become too big.
Reviewed-by: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
hw/audio/trace-events | 3 ++-
hw/audio/virtio-snd.c | 57 ---
2
Split out the function virtio_snd_pcm_open() from
virtio_snd_pcm_prepare(). A later patch also needs
the new function. There is no functional change.
Reviewed-by: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 57 +++
1 file
Split out the function virtio_snd_pcm_set_active() from
virtio_snd_pcm_start_stop(). A later patch also needs this
new funcion. There is no functional change.
Reviewed-by: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
hw/audio/virtio-snd.c | 21 -
1 file changed, 16
Am 21.02.24 um 22:16 schrieb Philippe Mathieu-Daudé:
> From: Bernhard Beschow
>
> Rather than distributing PC system flash handling across three files, let's
> confine it to one. Now, pc_system_firmware_init() creates, configures and
> cleans
> up the system flash which makes the code easier to u
0x37, not 0x38. In Linux, "lsusb -D ..." displays
> incomplete descriptor information when this length is incorrect.
>
> Signed-off-by: Joonas Kankaala
lsusb also misinterprets the invalid interface protocol 0x4 and uses
0x30 instead.
Reviewed-by: Volker Rümelin
> ---
>
Am 20.04.24 um 07:40 schrieb Mark Cave-Ayland:
> On 20/04/2024 02:21, Richard Henderson wrote:
>
>> On 4/19/24 12:51, Mark Cave-Ayland wrote:
>>> The various Intel CPU manuals claim that SGDT and SIDT can write
>>> either 24-bits
>>> or 32-bits depending upon the operand size, but this is incorrect
Am 26.04.24 um 21:41 schrieb Philippe Mathieu-Daudé:
> WHPX has a specific use of the CPUState::vcpu_dirty field
> (CPUState::vcpu_dirty is not used by common code).
> To make this field accel-specific, add and use a new
> @dirty variable in the AccelCPUState structure.
>
> Signed-off-by: Philippe
t->rpos += written;
if (chunk != written) {
+ fprintf(stderr, "%s: could not write %u bytes\n", __func__,
+ chunk - written);
break;
}
}
hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1));
+ if (avail != total_written)
From: Volker Rümelin
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
From: Volker Rümelin
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
From: Volker Rümelin
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
From: Volker Rümelin
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
From: Volker Rümelin
Substitute sw->hw with hw in the audio_pcm_sw_alloc_resources_*
functions.
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index a9a550a
From: Volker Rümelin
Wire up the st_rate_frames_out() function and replace
audio_frontend_frames_in() to make audio packet length
calculation exact.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 29 -
1 file changed, 8 insertions(+), 21 deletions(-)
diff --git
From: Volker Rümelin
All call sites of audio_pcm_sw_write() guarantee that sw is not
NULL. Remove the unnecessary NULL check.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 4
1 file changed, 4 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index b0a270ba85..9d6ffa500a
From: Volker Rümelin
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
From: Volker Rümelin
The function audio_capture_mix_and_clear() no longer uses
audio_pcm_sw_write() to resample audio frames from one internal
buffer to another. For this reason, the noop_conv() function is
now unused. Remove it.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 8
1
From: Volker Rümelin
Wire up the st_rate_frames_in() function and replace
audio_frontend_frames_out() to make audio packet length
calculation exact. When upsampling, it's still possible that
the audio frontends can't write the last audio frame. This will
be fixed later.
Signed-off-
From: Volker Rümelin
The audio_pcm_sw_write() function uses a lot of very unspecific
variable names. Rename them for better readability.
ret => total_in
total => total_out
size => buf_len
hwsamples => hw->mix_buf.size
samples => frames_in_max
Signed-off-by: Volker Rümelin
From: Volker Rümelin
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 function will later replace the audio_frontend_frames_in()
function
From: Volker Rümelin
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-
From: Volker Rümelin
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
From: Volker Rümelin
The audio_pcm_sw_read() function uses a few very unspecific
variable names. Rename them for better readability.
ret => total_out
total => total_in
size => buf_len
samples => frames_out_max
Signed-off-by: Volker Rümelin
---
audio/audio.c | 18 +-
From: Volker Rümelin
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
From: Volker Rümelin
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
Am 15.01.23 um 14:08 schrieb Volker Rümelin:
Ccing a few more people who might be interested in this patch series.
@Mark:
After this patch series, the code in your out of tree ASC audio device
(and a few in tree audio devices) could be simplified. write_audio() and
the loops calling
Am 13.01.23 um 17:21 schrieb Daniel P. Berrangé:
The audio_calloc function does various checks on the size and
nmembers parameters to detect various error conditions. There
are only 5 callers
* alsa_poll_helper: the pollfd count is small and bounded,
* audio_pcm_create_voice_pair_: allocatin
properly here. It should report:
ERROR: space prohibited between function name and open parenthesis '('
With this changed,
Reviewed-by: Volker Rümelin
QTAILQ_INSERT_TAIL(&audio_states, s, list);
QLIST_INIT (&s->card_head);
Am 16.01.23 um 10:03 schrieb Daniel P. Berrangé:
On Sun, Dec 18, 2022 at 06:15:36PM +0100, Volker Rümelin wrote:
Replace audio_calloc() with the equivalent g_new0().
With a n_structs argument >= 1, g_new0() never returns NULL.
Also remove the unnecessary NULL checks.
Signed-off-by: Vol
Am 16.01.23 um 09:58 schrieb Daniel P. Berrangé:
On Sun, Dec 18, 2022 at 06:39:00PM +0100, Volker Rümelin wrote:
Am 18.12.22 um 18:26 schrieb Philippe Mathieu-Daudé:
On 18/12/22 18:15, Volker Rümelin wrote:
Use g_malloc0() as a direct replacement for audio_calloc().
Signed-off-by: Volker
Am 19.03.23 um 21:03 schrieb Volker Rümelin:
Am 19.03.23 um 18:49 schrieb BALATON Zoltan:
Not sure this helps but I get these with DEBUG enabled in qemu/audio
on Linux host with alsa set to 44100 Hz dmix rate with default
settings without any -audiodev options with AmigaOS guest.
With
es_out 0.000 pid=8358 hw_free=0x38c hw_written=0x372
Hi Rene,
it seems your Mac uses a 48kHz sample rate, although QEMU requested a
44.1kHz sample rate. Could you add -audiodev
coreaudio,id=audio0,out.frequency=48000 to your command line and test if
the playback speed and pitch is now correct
/-/issues/1191.
I can't really help to fix this Core Audio backend issue. I don't have a
Mac. Until this bug is fixed, you will have to live with the workaround.
With best regards,
Volker
--- Ursprüngliche Nachricht ---
Von: Volker Rümelin
Datum: 28.03.2023 20:26:14
An: Rene Engel
B
Am 29.03.23 um 21:20 schrieb BALATON Zoltan:
On Tue, 28 Mar 2023, Volker Rümelin wrote:
it seems your Mac uses a 48kHz sample rate, although QEMU requested a
44.1kHz sample rate. Could you add -audiodev
coreaudio,id=audio0,out.frequency=48000 to your command line and test
if the playback
Am 28.03.23 um 15:08 schrieb Dorinda Bassey:
Hi Dorinda,
This commit adds a new audiodev backend to allow QEMU to use Pipewire as
both an audio sink and source. This backend is available on most systems
Add Pipewire entry points for QEMU Pipewire audio backend
Add wrappers for QEMU Pipewire au
code to connect to the
specified
source/sink. I would prefer the latter. PW_KEY_TARGET_OBJECT looks
promising.
Ack.
Thanks,
Dorinda.
On Mon, Mar 20, 2023 at 7:31 AM Volker Rümelin
wrote:
> diff --git a/audio/trace-events b/audio/trace-events
> index e1ab643add..e0acf9ac5
ink was already in your email.
With best regards,
Volker
Thanks,
Dorinda
On Mon, Apr 3, 2023 at 8:51 AM Volker Rümelin wrote:
Am 28.03.23 um 13:56 schrieb Dorinda Bassey:
Hi Dorinda,
> Hi Volker,
>
> Thanks for the feedback.
>
> This term is
Hi Dorinda,
This commit adds a new audiodev backend to allow QEMU to use Pipewire as
both an audio sink and source. This backend is available on most systems
Add Pipewire entry points for QEMU Pipewire audio backend
Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops()
qpw_write functio
Hi Dorinda,
This commit adds a new audiodev backend to allow QEMU to use Pipewire as
both an audio sink and source. This backend is available on most systems
Add Pipewire entry points for QEMU Pipewire audio backend
Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops()
qpw_write functio
dio/audio_template: use g_new0() to replace
audio_calloc()
Check samples for negative values. It's possible to have negative
samples due to overflows or unsigned to signed conversions. Patch 01/11
takes care of samples == 0.
Volker Rümelin (11):
audio: log unimplemented audio device samp
Remove the unused #define AUDIO_STRINGIFY. It was last used before
commit 470bcabd8f ("audio: Replace AUDIO_FUNC with __func__").
Reviewed-by: Philippe Mathieu-Daudé
Reviewed-by: Thomas Huth
Signed-off-by: Volker Rümelin
---
audio/audio_int.h | 3 ---
1 file changed, 3 deletion
fault 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
--
Replace audio_calloc() with the equivalent g_new0().
With a n_structs argument of 1, g_new0() never returns NULL.
Also remove the unnecessary NULL checks.
Reviewed-by: Richard Henderson
Signed-off-by: Volker Rümelin
---
audio/audio.c | 5 -
audio/audio_template.h | 6
Let the audio_pcm_create_voice_pair_* functions handle error
reporting. This avoids an additional error message in case
the guest selected an unimplemented sample rate.
Reviewed-by: Philippe Mathieu-Daudé
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 2 +-
1 file changed, 1
Replace audio_calloc() with the equivalent g_new0().
With a n_structs argument >= 1, g_new0() never returns NULL.
Also remove the unnecessary NULL checks.
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 29 -
1 file changed, 12 insertions(+), 17 deleti
Now that the last call site of audio_calloc() was removed, remove
the unused audio_calloc() function.
Reviewed-by: Philippe Mathieu-Daudé
Signed-off-by: Volker Rümelin
---
audio/audio.c | 20
audio/audio_int.h | 1 -
2 files changed, 21 deletions(-)
diff --git a
Replace audio_calloc() with the equivalent g_new0().
The value of the g_new0() argument count is >= 1, which means
g_new0() will never return NULL. Also remove the unnecessary
NULL check.
Reviewed-by: Philippe Mathieu-Daudé
Reviewed-by: Richard Henderson
Signed-off-by: Volker Rüme
since last alsa run (running): 0.006084
Acked-by: Christian Schoenebeck
Signed-off-by: Volker Rümelin
---
audio/alsaaudio.c | 11 ---
1 file changed, 4 insertions(+), 7 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5f50dfa0bf..0cc982e61f 100644
--- a/audio
he 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
Signed-off-by: Volker Rümelin
---
audio/audio.c | 1 +
audio/audio_
negative values.
Signed-off-by: Volker Rümelin
---
audio/audio_int.h | 4 ++--
audio/audio_template.h | 18 --
2 files changed, 10 insertions(+), 12 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 4632cdf9cc..ce2d6bf92c 100644
--- a/audio/audio_int.h
Use a consistent friendly name for the HWVoiceOut and HWVoiceIn
structures.
Reviewed-by: Thomas Huth
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index ee320a2e3f
Am 23.01.23 um 08:43 schrieb Philippe Mathieu-Daudé:
On 21/1/23 10:47, Volker Rümelin wrote:
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
Am 22.01.23 um 19:13 schrieb Mark Cave-Ayland:
On 15/01/2023 13:45, Volker Rümelin wrote:
Am 15.01.23 um 14:08 schrieb Volker Rümelin:
Ccing a few more people who might be interested in this patch series.
@Mark:
After this patch series, the code in your out of tree ASC audio
device (and a
Am 31.01.23 um 15:53 schrieb Marc-André Lureau:
Hi
On Sun, Jan 15, 2023 at 5:10 PM Volker Rümelin wrote:
Based-on: <3b1404eb-a7c5-f64c-3e47-1397c54c4...@t-online.de>
([PATCH 00/11] audio: more improvements)
Something didn't work with patchew
(https://patchew.org/QEMU/61bd351f-068
Hi Brad,
audio: Add sndio backend
Add a sndio backend.
sndio is the native API used by OpenBSD, although it has been ported to
other *BSD's and Linux (packages for Ubuntu, Debian, Void, Arch, etc.).
The C code is from Alexandre Ratchov and the rest of
the bits are from me.
---
audio/audio.
Hi Brad,
just a few white space and coding style issues.
+/*
+ * stop polling descriptors
+ */
+static void sndio_poll_clear(SndioVoice *self)
+{
+struct pollfd *pfd;
+int i;
+
+for (i = 0; i < self->nfds; i++) {
+pfd = &self->pfds[i];
+qemu_set_fd_handler (pfd->fd,
ptions.txt | 2 +-
qemu-options.hx | 4 ++--
scripts/meson-buildoptions.sh | 2 +-
7 files changed, 17 insertions(+), 17 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/meson.build b/meson.build
index 229eb585f7..4c44736bd4 100644
--- a/meson.build
+++ b/m
From: Marc-André Lureau
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 10 --
1 file changed, 10 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 9eb69bfd18..51cfc0b052 100644
--- a/audio/pwaudio.c
+++ b/audio/pwaudio.c
From: Marc-André Lureau
g_clear_pointer() already checks for NULL.
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 8 ++--
1 file changed, 2 insertions(+), 6 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 51cfc0b052..6ca4ef4f62
From: Marc-André Lureau
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 6ca4ef4f62..2b12b40934 100644
--- a/audio/pwaudio.c
+++ b/audio
From: Marc-André Lureau
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 85 -
1 file changed, 34 insertions(+), 51 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index d0bc4680a6..67df53948c
From: Marc-André Lureau
The stream is actually created connected.
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 1 -
1 file changed, 1 deletion(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 38905f5be2..f74d506ec6 100644
--- a/audio
+
1 file changed, 26 insertions(+), 49 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index f74d506ec6..062610a704 100644
--- a/audio/pwaudio.c
+++ b/audio/pwaudio.c
@@ -417,8 +417,8 @@ pw_to_audfmt(enum
From: Marc-André Lureau
PipeWire uses errno to report error details.
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 7 ---
1 file changed, 4 insertions(+), 3 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 2b12b40934..d0bc4680a6
From: Marc-André Lureau
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 7 ++-
1 file changed, 6 insertions(+), 1 deletion(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 67df53948c..5c706a9fde 100644
--- a/audio/pwaudio.c
+++ b/audio
From: Marc-André Lureau
create_stream() now reports on all error paths.
Signed-off-by: Marc-André Lureau
---
audio/pwaudio.c | 12 +---
1 file changed, 1 insertion(+), 11 deletions(-)
Reviewed-by: Volker Rümelin
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index 5c706a9fde
Am 12.04.23 um 15:59 schrieb Dorinda Bassey:
> Hi Volker,
>
> It seems that for some unknown reason using audio_pcm_info_clear_buf in
> playback_process causes segmentation fault. Hence I moved the handling of
> buffer underruns from the playback process to the qpw_write process because
> that i
I'm trying to use QEMU on Windows hosts for fun and for profit. While the GTK
GUI doesn't seem to support OpenGL under Windows the SDL2 GUI does. Hence I
used the SDL2 GUI where I ran into several issues of which three are fixed in
this series, which are:
* Alt+Tab switches tasks on the host rath
Am 13.04.23 um 22:43 schrieb Bernhard Beschow:
Am 13. April 2023 17:54:34 UTC schrieb "Volker Rümelin" :
I'm trying to use QEMU on Windows hosts for fun and for profit. While the GTK
GUI doesn't seem to support OpenGL under Windows the SDL2 GUI does. Hence I
used the SDL2
Am 13.04.23 um 22:43 schrieb Bernhard Beschow:
Am 13. April 2023 17:54:34 UTC schrieb "Volker Rümelin" :
I'm trying to use QEMU on Windows hosts for fun and for profit. While the GTK
GUI doesn't seem to support OpenGL under Windows the SDL2 GUI does. Hence I
used the SDL2
Am 13.04.23 um 22:43 schrieb Bernhard Beschow:
Am 13. April 2023 17:54:34 UTC schrieb "Volker Rümelin"
:
I'm trying to use QEMU on Windows hosts for fun and for profit.
While the GTK
GUI doesn't seem to support OpenGL under Windows the SDL2 GUI does.
Hence I
used the
Hi Dorinda,
This commit adds a new audiodev backend to allow QEMU to use Pipewire as
both an audio sink and source. This backend is available on most systems
Add Pipewire entry points for QEMU Pipewire audio backend
Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops()
qpw_write functio
be switched in
the VM.
Signed-off-by: Bernhard Beschow
---
ui/sdl2.c | 3 +++
1 file changed, 3 insertions(+)
Reviewed-by: Volker Rümelin
diff --git a/ui/sdl2.c b/ui/sdl2.c
index b12dec4caf..8af8b89f1d 100644
--- a/ui/sdl2.c
+++ b/ui/sdl2.c
@@ -856,6 +856,9 @@ static void
rtion(+)
Reviewed-by: Volker Rümelin
diff --git a/ui/sdl2.c b/ui/sdl2.c
index 8af8b89f1d..00aadfae37 100644
--- a/ui/sdl2.c
+++ b/ui/sdl2.c
@@ -859,6 +859,7 @@ static void sdl2_display_init(DisplayState *ds,
DisplayOptions *o)
#ifdef SDL_HINT_ALLOW_ALT_TAB_WHILE_GRABBED
SDL_Se
hen enabled with a grabbed keyboard,
this disables the special left control key filter and handling of the
AltGr key on Windows is broken again.
This patch fixes handling of the AltGr key on Windows with international
keyboard layout.
Volker Rümelin (1):
ui/sdl2: disable SDL_HINT_GRAB_KEYBO
r when the keyboard is grabbed.
To fix the problem, disable the SDL2 Windows low level keyboard
hook procedure.
Reported-by: Bernhard Beschow
Signed-off-by: Volker Rümelin
---
ui/sdl2.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/ui/sdl2.c b/ui/sdl2.c
index 00aadfae37..9d703200bf 10
the correct QEMU branch.
Volker Rümelin (1):
ui/sdl2: disable SDL_HINT_GRAB_KEYBOARD on Windows
ui/sdl2.c | 3 +++
1 file changed, 3 insertions(+)
--
2.35.3
r when the keyboard is grabbed.
To fix the problem, disable the SDL2 Windows low level keyboard
hook procedure.
Reported-by: Bernhard Beschow
Signed-off-by: Volker Rümelin
---
ui/sdl2.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/ui/sdl2.c b/ui/sdl2.c
index 00aadfae37..9d703200bf 10
0644 audio/pwaudio.c
I think the program code is ok now.
Reviewed-by: Volker Rümelin
: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
v2:
Cosmetic changes to the comments. The promised improved commit
message appeared in commit b9ae74e2e4 ("audio: log unimplemented
audio device sample rates").
tests/qtest/ac97-test.c | 40 ++
wrap should be at least
fc5eaaf6f68d5cff76468c63984c33c4fb51506d.
Building QEMU on my Linux system works fine.
Tested-by: Volker Rümelin
abled()
I tested that QEMU builds on Windows with MSYS2 mingw64.
One unrelated note: The DLL loader doesn't find the libslirp-0.dll when
QEMU was built with libslirp from the subprojects folder and started
from the build folder. It's necessary to copy this library to a folder
in the DLL search path.
Tested-by: Volker Rümelin
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