The has_FOO for pointer-valued FOO are redundant, except for arrays.
They are also a nuisance to work with. Recent commit "qapi: Start to
elide redundant has_FOO in generated C" provided the means to elide
them step by step. This is the step for qapi/audio.json.
Said commit explains the transformation in more detail. The invariant
violations mentioned there do not occur here.
Additionally, helper get_str() loses its @has_dst parameter.
Cc: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
Message-Id: <20221104160712.3005652-8-armbru@redhat.com>
improve error handling during module load, by changing:
bool module_load(const char *prefix, const char *lib_name);
void module_load_qom(const char *type);
to:
int module_load(const char *prefix, const char *name, Error **errp);
int module_load_qom(const char *type, Error **errp);
where the return value is:
-1 on module load error, and errp is set with the error
0 on module or one of its dependencies are not installed
1 on module load success
2 on module load success (module already loaded or built-in)
module_load_qom_one has been introduced in:
commit 28457744c3 ("module: qom module support"), which built on top of
module_load_one, but discarded the bool return value. Restore it.
Adapt all callers to emit errors, or ignore them, or fail hard,
as appropriate in each context.
Replace the previous emission of errors via fprintf in _some_ error
conditions with Error and error_report, so as to emit to the appropriate
target.
A memory leak is also fixed as part of the module_load changes.
audio: when attempting to load an audio module, report module load errors.
Note that still for some callers, a single issue may generate multiple
error reports, and this could be improved further.
Regarding the audio code itself, audio_add() seems to ignore errors,
and this should probably be improved.
block: when attempting to load a block module, report module load errors.
For the code paths that already use the Error API, take advantage of those
to report module load errors into the Error parameter.
For the other code paths, we currently emit the error, but this could be
improved further by adding Error parameters to all possible code paths.
console: when attempting to load a display module, report module load errors.
qdev: when creating a new qdev Device object (DeviceState), report load errors.
If a module cannot be loaded to create that device, now abort execution
(if no CONFIG_MODULE) or exit (if CONFIG_MODULE).
qom/object.c: when initializing a QOM object, or looking up class_by_name,
report module load errors.
qtest: when processing the "module_load" qtest command, report errors
in the load of the module.
Signed-off-by: Claudio Fontana <cfontana@suse.de>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Message-Id: <20220929093035.4231-4-cfontana@suse.de>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Improve readability of audio out.voices test:
If 1 is logged and set after positive test, 1 should be tested.
Signed-off-by: Helge Konetzka <hk@zapateado.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20221012114925.5084-3-hk@zapateado.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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 <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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 <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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 <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Replace a comment with a question with the answer.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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 <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-6-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>
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 <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-4-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>
Refactoring the code in audio_run_out() avoids code duplication
in the next patch. There's no functional change.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-1-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>
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.).
Signed-off-by: Brad Smith <brad@comstyle.com>
Signed-off-by: Alexandre Ratchov <alex@caoua.org>
Reviewed-by: Volker Rümelin <vr_qemu@t-online.de>
Tested-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <YxibXrWsrS3XYQM3@vm1.arverb.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
add a simple help option for -audio and -audiodev
to show the list of available drivers, and document them.
Signed-off-by: Claudio Fontana <cfontana@suse.de>
Message-Id: <20220908081441.7111-1-cfontana@suse.de>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
If you specify a known backend but it isn't compiled in, or failed to
initialize, you get a simple warning and the "none" backend as a
fallback, and QEMU runs happily:
$ qemu-system-x86_64 -audiodev id=audio,driver=dsound
audio: Unknown audio driver `dsound'
audio: warning: Using timer based audio emulation
...
Instead, QEMU should fail to start:
$ qemu-system-x86_64 -audiodev id=audio,driver=dsound
audio: Unknown audio driver `dsound'
$
Resolves:
https://bugzilla.redhat.com/show_bug.cgi?id=1983493
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220822131021.975656-1-marcandre.lureau@redhat.com>
Commit c9c847481 broken dbus audio module compilation with bad
'CONFIG_GIO' usage. Furthermore, it implied extra dependency on audio
module which aren't necessary.
The problem was that 'dbus_display' is not correctly automatically set
on MacOS, because opengl dependency wasn't taken into account.
Fixes: c9c847481 ("audio/dbus: Fix building with modules on macOS")
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220622154918.560870-1-marcandre.lureau@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
-audio is used like "-audio pa,model=sb16". It is almost as simple as
-soundhw, but it reuses the -audiodev parsing machinery and attaches an
audiodev to the newly-created device. The main 'feature' is that
it knows about adding the codec device for model=intel-hda, and adding
the audiodev to the codec device.
In the future, it could be extended to support default models or
builtin devices, just like -nic, or even a default backend. For now,
keep it simple.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Replace a config-time define with a compile time condition
define (compatible with clang and gcc) that must be declared prior to
its usage. This avoids having a global configure time define, but also
prevents from bad usage, if the config header wasn't included before.
This can help to make some code independent from qemu too.
gcc supports __BYTE_ORDER__ from about 4.6 and clang from 3.2.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
[ For the s390x parts I'm involved in ]
Acked-by: Halil Pasic <pasic@linux.ibm.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Message-Id: <20220323155743.1585078-7-marcandre.lureau@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
One less qemu-specific macro. It also helps to make some headers/units
only depend on glib, and thus moved in standalone projects eventually.
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Richard W.M. Jones <rjones@redhat.com>
g_new(T, n) is neater than g_malloc(sizeof(T) * n). It's also safer,
for two reasons. One, it catches multiplication overflowing size_t.
Two, it returns T * rather than void *, which lets the compiler catch
more type errors.
This commit only touches allocations with size arguments of the form
sizeof(T).
Patch created mechanically with:
$ spatch --in-place --sp-file scripts/coccinelle/use-g_new-etc.cocci \
--macro-file scripts/cocci-macro-file.h FILES...
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Cédric Le Goater <clg@kaod.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Acked-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Message-Id: <20220315144156.1595462-4-armbru@redhat.com>
Reviewed-by: Pavel Dovgalyuk <Pavel.Dovgalyuk@ispras.ru>
The unused variables when FLOAT_MIXENG is defined caused warnings on
Apple clang version 13.1.6 (clang-1316.0.21.2).
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20220316061053.60587-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The coreaudio library includes Objective-C declarations (using the
caret '^' symbol to declare block references [*]). When building
with a C compiler we get:
[175/839] Compiling C object libcommon.fa.p/audio_coreaudio.c.o
In file included from /Library/Developer/CommandLineTools/SDKs/MacOSX12.sdk/System/Library/Frameworks/CoreAudio.framework/Headers/CoreAudio.h:18,
from ../../audio/coreaudio.c:26:
/Library/Developer/CommandLineTools/SDKs/MacOSX12.sdk/System/Library/Frameworks/CoreAudio.framework/Headers/AudioHardware.h:162:2: error: expected identifier or '(' before '^' token
162 | (^AudioObjectPropertyListenerBlock)( UInt32 inNumberAddresses,
| ^
FAILED: libcommon.fa.p/audio_coreaudio.c.o
Rename the file to use the Objective-C default extension (.m) so
meson calls the correct compiler.
[*] https://developer.apple.com/library/archive/documentation/Cocoa/Conceptual/ProgrammingWithObjectiveC/WorkingwithBlocks/WorkingwithBlocks.html
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
handle_voice_change() is a CoreAudio callback function as of CoreAudio type
AudioObjectPropertyListenerProc, and for the latter MacOSX.sdk/System/
Library/Frameworks/CoreAudio.framework/Headers/AudioHardware.h
says "The return value is currently unused and should always be 0.".
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20220306123410.61063-1-akihiko.odaki@gmail.com>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Without this change audio_bug aborts when the bug condition is met,
which discards following useful logs. Call abort after such logs.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20220306063202.27331-1-akihiko.odaki@gmail.com>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
When configuring QEMU with --enable-modules we get on macOS:
--- stderr ---
Dependency ui-dbus cannot be satisfied
ui-dbus depends on pixman and opengl, so add these dependencies
to audio-dbus.
Fixes: 739362d420 ("audio: add "dbus" audio backend")
Reviewed-by: Li Zhang <lizhang@suse.de>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
When building on macOS 12 we get:
audio/coreaudio.c:50:5: error: 'kAudioObjectPropertyElementMaster' is deprecated: first deprecated in macOS 12.0 [-Werror,-Wdeprecated-declarations]
kAudioObjectPropertyElementMaster
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
kAudioObjectPropertyElementMain
/Library/Developer/CommandLineTools/SDKs/MacOSX.sdk/System/Library/Frameworks/CoreAudio.framework/Headers/AudioHardwareBase.h:208:5: note: 'kAudioObjectPropertyElementMaster' has been explicitly marked deprecated here
kAudioObjectPropertyElementMaster API_DEPRECATED_WITH_REPLACEMENT("kAudioObjectPropertyElementMain", macos(10.0, 12.0), ios(2.0, 15.0), watchos(1.0, 8.0), tvos(9.0, 15.0)) = kAudioObjectPropertyElementMain
^
Replace by kAudioObjectPropertyElementMain, redefining it to
kAudioObjectPropertyElementMaster if not available.
Suggested-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Suggested-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Suggested-by: Roman Bolshakov <roman@roolebo.dev>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Tested-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Signed-off-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Otherwise, the audio subsystem tries to use the voice and
eventually aborts due to the maximum number of samples in the
buffer is not set.
Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20220226115953.60335-1-akihiko.odaki@gmail.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Fix the same samples vs. frames mix-up that the previous commit
fixed for the PulseAudio backend.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-15-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>
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 <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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 <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-11-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>
This reverts commit cbaf25d1f5.
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 <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Add a pcm_ops function table for the capture backend. This avoids
additional code in the next patches to test if the pcm_ops table
is available.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
Simplify code by inlining function audio_pcm_sw_get_rpos_in()
at the only call site and remove the duplicated audio_bug()
test.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
Move the function audio_pcm_hw_clip_out() into the correct
section 'Hard voice (playback)'.
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
