summaryrefslogtreecommitdiffstats
path: root/audio
Commit message (Collapse)AuthorAgeFilesLines
* module: add Error arguments to module_load and module_load_qomClaudio Fontana2022-11-061-6/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 28457744c345 ("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>
* module: rename module_load_one to module_loadClaudio Fontana2022-11-061-1/+1
| | | | | | | | Signed-off-by: Claudio Fontana <cfontana@suse.de> Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org> Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Message-Id: <20220929093035.4231-3-cfontana@suse.de> Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* audio: improve out.voices testHelge Konetzka2022-10-121-1/+1
| | | | | | | | | | 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>
* audio: fix in.voices testHelge Konetzka2022-10-121-1/+1
| | | | | | | | | | | | Calling qemu with valid -audiodev ...,in.voices=0 results in an obsolete warning: audio: Bogus number of capture voices 0, setting to 0 This patch fixes the in.voices test. Signed-off-by: Helge Konetzka <hk@zapateado.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20221012114925.5084-2-hk@zapateado.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: prevent an integer overflow in resampling codeVolker Rümelin2022-10-111-5/+6
| | | | | | | | | | | | | | | | | | | | | | | 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>
* audio: fix sw->buf size for audio recordingVolker Rümelin2022-10-112-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | 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>
* audio: refactor audio_get_avail()Volker Rümelin2022-10-111-5/+19
| | | | | | | | | | | 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>
* audio: rename audio_sw_bytes_free()Volker Rümelin2022-10-111-6/+14
| | | | | | | | | | | 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>
* audio: swap audio_rate_get_bytes() function parametersVolker Rümelin2022-10-116-8/+8
| | | | | | | | | | | 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>
* spiceaudio: update commentVolker Rümelin2022-10-111-1/+4
| | | | | | | | | 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>
* spiceaudio: add a pcm_ops buffer_get_free functionVolker Rümelin2022-10-111-2/+10
| | | | | | | | | | | | | | 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>
* audio: add more audio rate control functionsVolker Rümelin2022-10-112-11/+26
| | | | | | | | | | | | | | 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>
* alsaaudio: reduce playback latencyVolker Rümelin2022-10-111-1/+37
| | | | | | | | | | | 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>
* audio: run downstream playback queue unconditionallyVolker Rümelin2022-10-111-4/+4
| | | | | | | | | | | | | | | 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>
* audio: fix GUS audio playback with out.mixing-engine=offVolker Rümelin2022-10-111-1/+2
| | | | | | | | | | | | | | | | | | | 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>
* audio: refactor code in audio_run_out()Volker Rümelin2022-10-111-9/+8Star
| | | | | | | | | | 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>
* audio: remove abort() in audio_bug()Volker Rümelin2022-09-271-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | 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>
* Revert "audio: Log context for audio bug"Volker Rümelin2022-09-272-24/+28
| | | | | | | | | | | This reverts commit 8e30d39bade3010387177ca23dbc2244352ed4a3. Revert commit 8e30d39bad "audio: Log context for audio bug" to make error propagation work again. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20220917131626.7521-1-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: Add sndio backendAlexandre Ratchov2022-09-274-0/+569
| | | | | | | | | | | | 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>
* audio: add help option for -audio and -audiodevClaudio Fontana2022-09-192-0/+20
| | | | | | | | | 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>
* audio: exit(1) if audio backend failed to be found or initializedMarc-André Lureau2022-09-022-4/+12
| | | | | | | | | | | | | | | | | | | | | | | 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>
* audio/dbus: fix buildingMarc-André Lureau2022-07-081-1/+1
| | | | | | | | | | | | | | 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>
* introduce -audio as a replacement for -soundhwPaolo Bonzini2022-05-142-1/+8
| | | | | | | | | | | | | | -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>
* Remove qemu-common.h include from most unitsMarc-André Lureau2022-04-061-1/+0Star
| | | | | | Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20220323155743.1585078-33-marcandre.lureau@redhat.com> Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* include: move qemu_get_vm_name() to sysemu.hMarc-André Lureau2022-04-061-1/+1
| | | | | | Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20220323155743.1585078-26-marcandre.lureau@redhat.com> Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* Replace config-time define HOST_WORDS_BIGENDIANMarc-André Lureau2022-04-062-2/+2
| | | | | | | | | | | | | | | | | | | 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>
* Replace GCC_FMT_ATTR with G_GNUC_PRINTFMarc-André Lureau2022-03-227-12/+12
| | | | | | | | 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>
* Use g_new() & friends where that makes obvious senseMarkus Armbruster2022-03-215-11/+11
| | | | | | | | | | | | | | | | | | | | | | | 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>
* audio/mixeng: Do not declare unused variablesAkihiko Odaki2022-03-181-4/+4
| | | | | | | | | | 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>
* audio: Rename coreaudio extension to use Objective-C compilerPhilippe Mathieu-Daudé2022-03-152-1/+1
| | | | | | | | | | | | | | | | | | | | | | 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>
* coreaudio: Always return 0 in handle_voice_changeAkihiko Odaki2022-03-151-4/+2Star
| | | | | | | | | | | | 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>
* audio: Log context for audio bugAkihiko Odaki2022-03-152-28/+24Star
| | | | | | | | | | 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>
* audio/dbus: Fix building with modules on macOSPhilippe Mathieu-Daudé2022-03-151-1/+1
| | | | | | | | | | | | | | 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>
* audio/coreaudio: Remove a deprecation warning on macOS 12Philippe Mathieu-Daudé2022-03-151-6/+11
| | | | | | | | | | | | | | | | | | | | | | | 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>
* coreaudio: Notify error in coreaudio_init_outAkihiko Odaki2022-03-041-0/+2
| | | | | | | | | | | | 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>
* sdlaudio: fix samples vs. frames mix-upVolker Rümelin2022-03-041-5/+2Star
| | | | | | | | | 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>
* paaudio: fix samples vs. frames mix-upVolker Rümelin2022-03-041-10/+4Star
| | | | | | | | | | | 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>
* ossaudio: reduce effective playback buffer sizeVolker Rümelin2022-03-041-3/+4
| | | | | | | | | | | 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>
* dsoundaudio: reduce effective playback buffer sizeVolker Rümelin2022-03-041-13/+17
| | | | | | | | | | 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>
* paaudio: reduce effective playback buffer sizeVolker Rümelin2022-03-041-9/+24
| | | | | | | | | | 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>
* audio: restore mixing-engine playback buffer sizeVolker Rümelin2022-03-049-18/+80
| | | | | | | | | | | | | | | | | | | | | 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>
* Revert "audio: fix wavcapture segfault"Volker Rümelin2022-03-041-2/+2
| | | | | | | | | | | 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 <vr_qemu@t-online.de> Message-Id: <20220301191311.26695-9-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: add pcm_ops function table for capture backendVolker Rümelin2022-03-041-0/+2
| | | | | | | | | | 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>
* audio: copy playback stream in sequential orderVolker Rümelin2022-03-041-15/+9Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* jackaudio: use more jack audio buffersVolker Rümelin2022-03-041-2/+2
| | | | | | | | | | | | | | | | | | | | 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>
* paaudio: increase default latency to 46msVolker Rümelin2022-03-041-1/+1
| | | | | | | | | | | | | | | | | 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>
* audio: inline function audio_pcm_sw_get_rpos_in()Volker Rümelin2022-03-041-18/+5Star
| | | | | | | | | | 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>
* audio: add function audio_pcm_hw_conv_in()Volker Rümelin2022-03-041-6/+19
| | | | | | | | | | | | 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>
* audio: move function audio_pcm_hw_clip_out()Volker Rümelin2022-03-041-19/+19
| | | | | | | | | | 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>
* audio: replace open-coded buffer arithmeticVolker Rümelin2022-03-044-30/+29Star
| | | | | | | | | | | Replace open-coded buffer arithmetic with the new function audio_ring_posb(). That's the position in backward direction of a given point at a given distance. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com> Message-Id: <20220301191311.26695-1-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2024-06-01 02:25:58 +0200