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
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
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
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
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-by: Volker
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
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
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
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
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
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
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
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
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
en) {
+ 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) {
+ fprintf(stderr, "%s: to write %d,
Am 04.01.23 um 13:35 schrieb Thomas Huth:
The windows jobs (especially the 32-bit job) recently started to
hit the timeout limit. Bump it a little bit to ease the situation
(80 minutes is quite long already - OTOH, these jobs do not have to
wait for a job from the container stage to finish, so
Am 28.12.22 um 14:52 schrieb Christian Schoenebeck:
On Monday, December 26, 2022 4:08:37 PM CET Volker Rümelin wrote:
Am 21.12.22 um 12:03 schrieb Christian Schoenebeck:
On Sunday, December 18, 2022 6:15:38 PM CET Volker Rümelin wrote:
The currently used default playback settings in the ALSA
Am 28.12.22 um 14:52 schrieb Christian Schoenebeck:
On Monday, December 26, 2022 4:08:37 PM CET Volker Rümelin wrote:
Am 21.12.22 um 12:03 schrieb Christian Schoenebeck:
On Sunday, December 18, 2022 6:15:38 PM CET Volker Rümelin wrote:
The currently used default playback settings in the ALSA
://lists.nongnu.org/archive/html/qemu-devel/2022-10/msg03987.html
I would support a patch to limit the VRA and VRM modes to the highest
supported rate of 48kHz. This is a technical limit for single data rate.
Before Volker Rümelin fixed it in 12f4abf6a245 and 0cbc8bd4694f, an adversary
could leverage
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5f50dfa0bf..0cc982e61f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -913,17 +913,14 @@ static void *alsa_audio_init(Audiodev *dev)
alsa_init_per_direction(aopts->in);
alsa_init_per_direction(aopts->out);
-
Am 21.12.22 um 12:03 schrieb Christian Schoenebeck:
On Sunday, December 18, 2022 6:15:38 PM CET 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
Am 18.12.22 um 21:26 schrieb Christian Schoenebeck:
On Sunday, December 18, 2022 6:15:29 PM CET Volker Rümelin wrote:
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
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 Rümelin
---
audio/audio_template.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/audio
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 | 23 ++-
1 file changed, 6 insertions(+), 17 deleti
since last alsa run (running): 0.006084
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/alsaaudio.c
+++ b/audio/alsaaudio.c
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.
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git
Use a consistent friendly name for the HWVoiceOut and HWVoiceIn
structures.
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..ac744d3484 100644
Now that the last call site of audio_calloc() was removed, remove
the unused audio_calloc() function.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 20
audio/audio_int.h | 1 -
2 files changed, 21 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index
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
---
au
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.
Signed-off-by: Volker Rümelin
---
audio/alsaaudio.c | 6 +-
1 file changed, 1 insertion(+)
Remove the unused #define AUDIO_STRINGIFY. It was last used before
commit 470bcabd8f ("audio: Replace AUDIO_FUNC with __func__").
Signed-off-by: Volker Rümelin
---
audio/audio_int.h | 3 ---
1 file changed, 3 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index
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.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 5 -
audio/audio_template.h | 6 +-
audio/mixeng.c | 7
Use g_malloc0() as a direct replacement for audio_calloc().
Signed-off-by: Volker Rümelin
---
audio/audio_template.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index d343a1dcb3..5f51ef26b2 100644
--- a/audio
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.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 1 +
audio/audio_template.h | 13 +
2 files changed, 14
- 11:
Audio playback and recording with the ALSA audio backend currently
doesn't work well with the default audio settings.
Volker Rümelin (11):
audio: log unimplemented audio device sample rates
audio: don't show unnecessary error messages
audio: rename hardware store to backend
audio
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. OVMF
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
Am 04.11.22 um 18:33 schrieb Thomas Huth:
On 26/10/2022 21.34, Volker Rümelin wrote:
Am 25.10.22 um 09:44 schrieb Marc-André Lureau:
Hi
On Tue, Oct 25, 2022 at 12:31 AM Volker Rümelin
wrote:
Am 24.10.22 um 10:13 schrieb Marc-André Lureau:
Hi
On Mon, Oct 24, 2022 at 9:28 AM Volker Rümelin
Am 25.10.22 um 09:44 schrieb Marc-André Lureau:
Hi
On Tue, Oct 25, 2022 at 12:31 AM Volker Rümelin wrote:
Am 24.10.22 um 10:13 schrieb Marc-André Lureau:
Hi
On Mon, Oct 24, 2022 at 9:28 AM Volker Rümelin
wrote:
Test if the audio subsystem can handle extreme up- and down-
sampling
Am 24.10.22 um 10:13 schrieb Marc-André Lureau:
Hi
On Mon, Oct 24, 2022 at 9:28 AM Volker Rümelin
wrote:
Test if the audio subsystem can handle extreme up- and down-
sampling ratios like 44100/1 and 1/44100. For some time these
used to trigger QEMU aborts. The test was taken
Am 21.10.22 um 18:59 schrieb Amarjargal Gundjalam:
The TABs should be replaced with spaces, to make sure that we have a
consistent coding style with an indentation of 4 spaces everywhere.
Resolves:https://gitlab.com/qemu-project/qemu/-/issues/370
Reviewed-by: Daniel P. Berrangé
Signed-off-by:
-by: Marc-André Lureau
Signed-off-by: Volker Rümelin
---
tests/qtest/ac97-test.c | 40 +++-
1 file changed, 39 insertions(+), 1 deletion(-)
diff --git a/tests/qtest/ac97-test.c b/tests/qtest/ac97-test.c
index 74103efdfa..ce25f1d588 100644
--- a/tests/qtest/ac97
qemu-triv...@nongnu.org
Signed-off-by: Volker Rümelin
---
ui/console.c | 2 +-
ui/gtk.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/ui/console.c b/ui/console.c
index 49da6a91df..65c117874c 100644
--- a/ui/console.c
+++ b/ui/console.c
@@ -1297,7 +1297,7 @@ static void kbd_
Am 27.09.22 um 13:54 schrieb Marc-André Lureau:
On Fri, Sep 23, 2022 at 10:48 PM Volker Rümelin
wrote:
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->
It seems there is a demand [1] for low latency playback over
SPICE. Add a pcm_ops buffer_get_free function to reduce the
playback latency. The mixing engine buffer becomes a temporary
buffer.
[1] https://lists.nongnu.org/archive/html/qemu-devel/2022-01/msg01644.html
Signed-off-by: Volker Rümelin
mment 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
---
audio/rate_template.h | 11 ++-
Swap the rate and info parameters of the audio_rate_get_bytes()
function to align the parameter order with the rest of the
audio_rate_*() functions.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 2 +-
audio/audio_int.h | 2 +-
audio/dbusaudio.c | 4 ++--
audio/noaudio.c| 4
Replace a comment with a question with the answer.
Signed-off-by: Volker Rümelin
---
audio/spiceaudio.c | 5 -
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 22892a7b9d..f52f3a8bbb 100644
--- a/audio/spiceaudio.c
+++ b/audio
Change the buffer_get_free pcm_ops function to report the free
ALSA playback buffer. The generic buffer becomes a temporary
buffer and is empty after a call to audio_run_out().
Signed-off-by: Volker Rümelin
---
audio/alsaaudio.c | 38 +-
1 file changed, 37
Split out the code in audio_get_avail() that calculates the
buffer size that the audio frontend can read. This is similar
to the code changes in audio_get_free().
Signed-off-by: Volker Rümelin
---
audio/audio.c | 24 +++-
1 file changed, 19 insertions(+), 5 deletions
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/71
Signed-off-by: Volker Rümelin
---
audio/audio.c | 2 +-
audio/audio_template.h | 4
2 files changed, 5 insertions(+), 1 deletion(-)
diff --git a/audio/audio.c b/audio/audio.c
index ba0c62b120..60c7472d37 100644
--- a/audio/audi
ot;) fixed the out.mixing-engine=on case.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 8
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 343786243d..9e55834909 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1143,6 +1143,10 @@ static void aud
Refactoring the code in audio_run_out() avoids code duplication
in the next patch. There's no functional change.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 17 -
1 file changed, 8 insertions(+), 9 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index cfa4119c05
().
Signed-off-by: Volker Rümelin
---
audio/audio.c | 35 ---
audio/audio_int.h | 2 ++
2 files changed, 26 insertions(+), 11 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 9e55834909..557538a7b7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2250,26
guest. The demo file
is available on the download page of the GUSemu32 author.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/audio/audio.c b/audio/audio.c
index 04f685fe24..343786243d 100644
--- a/audio/audio.c
+++ b/audio
-devel/2022-09/msg02347.html,
issue #71 triggers a QEMU abort. Patch 11/12 is nevertheless correct.
Volker Rümelin (12):
audio: refactor code in audio_run_out()
audio: fix GUS audio playback with out.mixing-engine=off
audio: run downstream playback queue unconditionally
alsaaudio: reduce p
Rename and refactor audio_sw_bytes_free(). This function is not
limited to calculate the free audio buffer size. The renamed
function returns the number of frames instead of bytes.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 20 ++--
1 file changed, 14 insertions(+), 6
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
Reviewed-by: Philippe Mathieu-Daudé
Signed-off-by: Volker Rümelin
---
v2:
There's no need to inititaliz
double free.
Resolves: https://gitlab.com/qemu-project/qemu/-/issues/1210
Signed-off-by: Volker Rümelin
---
ui/console.c | 5 ++---
1 file changed, 2 insertions(+), 3 deletions(-)
diff --git a/ui/console.c b/ui/console.c
index 765892f84f..030e75bc71 100644
--- a/ui/console.c
+++ b/ui/console.c
g #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
---
audio/audio.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/audio/audio.c b
This reverts commit 8e30d39bade3010387177ca23dbc2244352ed4a3.
Revert commit 8e30d39bad "audio: Log context for audio bug"
to make error propagation work again.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 25 +
audio/audio_templ
en=0)
Aborted (core dumped)
Remove the abort() in audio_bug() to avoid this class of guest-triggered
aborts.
Volker Rümelin (2):
Revert "audio: Log context for audio bug"
audio: remove abort() in audio_bug()
audio/audio.c | 24
audio/audio_template.
Am 08.09.22 um 19:31 schrieb Arwed Meyer:
@@ -54,21 +60,15 @@ DECLARE_INSTANCE_CHECKER(MouseChardev, MOUSE_CHARDEV,
static void msmouse_chr_accept_input(Chardev *chr)
{
MouseChardev *mouse = MOUSE_CHARDEV(chr);
-int len;
+uint32_t len_out, len;
-len =
_options.txt | 4 +-
qapi/audio.json | 25 +-
qemu-options.hx | 16 +
scripts/meson-buildoptions.sh | 7 +-
10 files changed, 632 insertions(+), 5 deletions(-)
create mode 100644 audio/sndioaudio.c
Tested again on Linux.
Reviewed-by: Volker Rümelin
Tested-by: Volker Rümelin
Am 07.09.22 um 19:30 schrieb Claudio Fontana:
add a simple help option for -audio and -audiodev
to show the list of available drivers, and document them.
Signed-off-by: Claudio Fontana
---
audio/audio.c | 20
audio/audio.h | 1 +
qemu-options.hx | 10 ++
s);
+return NULL;
Hi Marc-André,
I don't understand why you move the QTAILQ_INSERT_TAIL(_states, s,
list) macro down. Without this you don't need the additional
free_audio_state() call. audio_cleanup() takes care to free the audio state.
The rest looks good. You can add my
Revi
Am 10.06.22 um 22:16 schrieb Richard Henderson:
On 6/10/22 02:20, Gerd Hoffmann wrote:
The following changes since commit
9cc1bf1ebca550f8d90f967ccd2b6d2e00e81387:
Merge tag 'pull-xen-20220609' of
https://xenbits.xen.org/git-http/people/aperard/qemu-dm into staging
(2022-06-09 08:25:17
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
---
ui/gtk-gl-area.c | 8 +++-
ui/trace-events | 1 +
2 files changed, 8 insertions(+), 1 deletion
ions fully implemented.
An example for this is the Intel i965 driver on a Haswell processor.
Signed-off-by: Volker Rümelin
---
ui/gtk-gl-area.c | 31 ++-
ui/trace-events | 1 +
2 files changed, 31 insertions(+), 1 deletion(-)
diff --git a/ui/gtk-gl-area.c b/ui/gtk-gl-ar
The first patch fixes a GL context leak.
The second patch fixes a black guest screen on Wayland with OpenGL
accelerated QEMU graphics devices. This bug doesn't seem to be related
to issues #910, #865, #671 or #298.
Volker Rümelin (2):
ui/gtk-gl-area: implement GL context destruction
ui
Am 18.03.22 um 16:26 schrieb Paolo Bonzini:
KVM support for AMX includes a new system attribute, KVM_X86_XCOMP_GUEST_SUPP.
Commit 19db68ca68 ("x86: Grant AMX permission for guest", 2022-03-15) however
did not fully consider the behavior on older kernels. First, it warns
too aggressively.
On Dienstag, 1. März 2022 20:13:02 CET Volker Rümelin wrote:
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
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
---
audio/ossaudio.c | 7 ---
1 file changed, 4 insertions(+), 3
Add the buffer_get_free pcm_ops function to reduce the effective
playback buffer size. All intermediate audio playback buffers
become temporary buffers.
Signed-off-by: Volker Rümelin
---
audio/dsoundaudio.c | 30 +-
1 file changed, 17 insertions(+), 13 deletions
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
---
audio/audio.c | 23 +--
1 file changed, 5 insertions(+), 18 deletions(-)
diff --git a/audio/audio.c b/audio
wraps around.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 25 +++--
1 file changed, 19 insertions(+), 6 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index dfd32912da..f28e91853f 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -548,6 +548,24 @@ static
This reverts commit cbaf25d1f59ee13fc7542a06ea70784f2e000c04.
Since previous commit every audio backend has a pcm_ops function
table. It's no longer necessary to test if the table is available.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 4 ++--
1 file changed, 2 insertions(+), 2
Fix the same samples vs. frames mix-up that the previous commit
fixed for the PulseAudio backend.
Signed-off-by: Volker Rümelin
---
audio/sdlaudio.c | 7 ++-
1 file changed, 2 insertions(+), 5 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index e605c787ba..797b47bbdd 100644
Add the buffer_get_free pcm_ops function to reduce the effective
playback buffer size. All intermediate audio playback buffers
become temporary buffers.
Signed-off-by: Volker Rümelin
---
audio/paaudio.c | 33 -
1 file changed, 24 insertions(+), 9 deletions
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
---
audio
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
---
audio/audio.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/audio/audio.c b/audio/audio.c
index 9e2d7fb209
Move the function audio_pcm_hw_clip_out() into the correct
section 'Hard voice (playback)'.
Reviewed-by: Philippe Mathieu-Daudé
Signed-off-by: Volker Rümelin
---
audio/audio.c | 38 +++---
1 file changed, 19 insertions(+), 19 deletions(-)
diff --git a/audio
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
---
audio/audio.c | 25 +++--
audio/audio_int.h | 13 +
audio
that they can't
access.
Signed-off-by: Volker Rümelin
---
audio/jackaudio.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index 317009e936..26246c3a8b 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -483,8 +483,8 @@ static int
ueue which the downstream queue
accepts.
Signed-off-by: Volker Rümelin
---
audio/alsaaudio.c | 1 +
audio/audio.c | 69 +++
audio/audio_int.h | 7 -
audio/coreaudio.c | 3 +++
audio/jackaudio.c | 1 +
audio/noaudio.c | 1 +
audio/ossaud
bufbackend buffer
The effective total playback buffer size is reduced by
timer_period.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 24 +---
1 file changed, 9 insertions(+), 15 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 35437986d9..9e2d7fb209 1006
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
---
audio/paaudio.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git
e
Improved commit message.
v3:
[PATCH v3 01/15] audio: replace open-coded buffer arithmetic
Use kernel-doc comments to document the audio_ring_posb()
function.
Volker Rümelin (15):
audio: replace open-coded buffer arithmetic
audio: move function audio_pcm_hw_clip_out()
audio: add function audio_pc
Am 26.01.22 um 09:48 schrieb Gerd Hoffmann:
Hi,
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -266,6 +266,12 @@ static inline size_t audio_ring_dist(size_t dst, size_t
src, size_t len)
return (dst >= src) ? (dst - src) : (len - src + dst);
}
You haven't touched this
On Samstag, 22. Januar 2022 13:57:31 CET Volker Rümelin wrote:
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
---
First of all, getting rid of all those
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
---
hw/usb/dev-mtp.c | 4 ++--
1 file
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
---
audio
ueue which the downstream queue
accepts.
Signed-off-by: Volker Rümelin
---
audio/alsaaudio.c | 1 +
audio/audio.c | 69 +++
audio/audio_int.h | 7 -
audio/coreaudio.c | 3 +++
audio/jackaudio.c | 1 +
audio/noaudio.c | 1 +
audio/ossaud
Fix the same samples vs. frames mix-up that the previous commit
fixed for the PulseAudio backend.
Signed-off-by: Volker Rümelin
---
audio/sdlaudio.c | 7 ++-
1 file changed, 2 insertions(+), 5 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index e605c787ba..797b47bbdd 100644
that they can't
access.
Signed-off-by: Volker Rümelin
---
audio/jackaudio.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index 317009e936..26246c3a8b 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -483,8 +483,8 @@ static int
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
---
audio/audio.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/audio/audio.c b/audio/audio.c
index 9e2d7fb209
bufbackend buffer
The effective total playback buffer size is reduced by
timer_period.
Signed-off-by: Volker Rümelin
---
audio/audio.c | 24 +---
1 file changed, 9 insertions(+), 15 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 35437986d9..9e2d7fb209 1006
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
---
audio/paaudio.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git
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
---
audio/audio.c | 25 +++--
audio/audio_int.h | 6 ++
audio/coreaudio.c | 10
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
---
audio/audio.c | 23 +--
1 file changed, 5 insertions(+), 18 deletions(-)
diff --git a/audio/audio.c b/audio
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