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* 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: fix sw->buf size for audio recordingVolker Rümelin2022-10-111-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | 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 Rümelin2022-10-111-1/+1
| | | | | | | | | | | 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>
* audio: add more audio rate control functionsVolker Rümelin2022-10-111-11/+24
| | | | | | | | | | | | | | 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>
* 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 Rümelin2022-09-271-12/+13
| | | | | | | | | | | 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 Ratchov2022-09-271-0/+1
| | | | | | | | | | | | 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 Fontana2022-09-191-0/+19
| | | | | | | | | 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 Lureau2022-09-021-3/+11
| | | | | | | | | | | | | | | | | | | | | | | 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>
* introduce -audio as a replacement for -soundhwPaolo Bonzini2022-05-141-1/+7
| | | | | | | | | | | | | | -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>
* 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>
* Use g_new() & friends where that makes obvious senseMarkus Armbruster2022-03-211-2/+2
| | | | | | | | | | | | | | | | | | | | | | | 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: Log context for audio bugAkihiko Odaki2022-03-151-13/+12Star
| | | | | | | | | | 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: restore mixing-engine playback buffer sizeVolker Rümelin2022-03-041-17/+52
| | | | | | | | | | | | | | | | | | | | | 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>
* 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 Rümelin2022-03-041-18/+7Star
| | | | | | | | | | | 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>
* audio: add "dbus" audio backendMarc-André Lureau2021-12-211-0/+1
| | | | | | | | | | | | Add a new -audio backend that accepts D-Bus clients/listeners to handle playback & recording, to be exported via the -display dbus. Example usage: -audiodev dbus,in.mixing-engine=off,out.mixing-engine=off,id=dbus -display dbus,audiodev=dbus Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Acked-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: Never send migration sectionDr. David Alan Gilbert2021-08-101-0/+10
| | | | | | | | | | | | | | | | | The audio migration vmstate is empty, and always has been; we can't just remove it though because an old qemu might send it us. Changes with -audiodev now mean it's sometimes created when it didn't used to be, and can confuse migration to old qemu. Change it so that vmstate_audio is never sent; if it's received it should still be accepted, and old qemu's shouldn't be too upset if it's missing. Signed-off-by: Dr. David Alan Gilbert <dgilbert@redhat.com> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Tested-by: Daniel P. Berrangé <berrange@redhat.com> Message-Id: <20210809170956.78536-1-dgilbert@redhat.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: Fix format specifications of debug logsAkihiko Odaki2021-06-171-3/+3
| | | | | | | Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com> Message-id: 20210616141411.53892-1-akihiko.odaki@gmail.com Message-Id: <20210616141411.53892-1-akihiko.odaki@gmail.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: move code to audio/audio.cVolker Rümelin2021-06-171-0/+9
| | | | | | | | | | | Move the code to generate the pa_context_new() application name argument to a function in audio/audio.c. The new function audio_application_name() will also be used in the jackaudio backend. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20210517194604.2545-3-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* sysemu: Let VMChangeStateHandler take boolean 'running' argumentPhilippe Mathieu-Daudé2021-03-091-1/+1
| | | | | | | | | | | The 'running' argument from VMChangeStateHandler does not require other value than 0 / 1. Make it a plain boolean. Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com> Reviewed-by: Alex Bennée <alex.bennee@linaro.org> Acked-by: David Gibson <david@gibson.dropbear.id.au> Message-Id: <20210111152020.1422021-3-philmd@redhat.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
* audio: Add braces for statements/fix braces' positionZhang Han2021-01-151-14/+12Star
| | | | | | | | | | | Fix problems about braces: -braces are necessary for all arms of if/for/while statements -else should follow close brace '}' Signed-off-by: Zhang Han <zhanghan64@huawei.com> Message-id: 20210115012431.79533-1-zhanghan64@huawei.com Message-Id: <20210115012431.79533-2-zhanghan64@huawei.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove remaining unused plive codeVolker Rümelin2021-01-151-16/+1Star
| | | | | | | | | Commit 73ad33ef7b "audio: remove plive" forgot to remove this code. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de Message-Id: <20210110100239.27588-12-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: break generic buffer dependency on mixing-engineVolker Rümelin2021-01-151-7/+4Star
| | | | | | | | | | Break the unnecessary dependency of the generic buffer management code on mixing-engine. This is required for the next patch. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de Message-Id: <20210110100239.27588-10-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: split pcm_ops function get_buffer_inVolker Rümelin2021-01-151-4/+14
| | | | | | | | | | | | | Split off pcm_ops function run_buffer_in from get_buffer_in and call run_buffer_in before get_buffer_in. The next patch only needs the generic buffer management part from audio_generic_get_buffer_in(). Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de Message-Id: <20210110100239.27588-8-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* sdlaudio: add -audiodev sdl,out.buffer-count optionVolker Rümelin2021-01-151-1/+1
| | | | | | | | | | | | | | | | | | | Currently there is a crackling noise with SDL2 audio playback. Commit bcf19777df: "audio/sdlaudio: Allow audio playback with SDL2" already mentioned the crackling noise. Add an out.buffer-count option to give users a chance to select sane settings for glitch free audio playback. The idea was taken from the coreaudio backend. The in.buffer-count option will be used with one of the next patches. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de Message-Id: <20210110100239.27588-3-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: add sanity checkGerd Hoffmann2020-12-151-1/+3
| | | | | | | | | | | Check whenever we actually found the spiceaudio driver before flipping the can_be_default field. Fixes: f0c4555edfdd ("audio: remove qemu_spice_audio_init()") Buglink: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=977301 Reported-by: dann frazier <dann.frazier@canonical.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> Message-Id: <20201215081151.20095-1-kraxel@redhat.com>
* audio: Simplify audio_bug() removing old codePhilippe Mathieu-Daudé2020-12-151-18/+1Star
| | | | | | | | | | | | This code (introduced in commit 1d14ffa97ea, Oct 2005) is likely unused since years. Time to remove it. If the condition is true, simply call abort(). Suggested-by: Gerd Hoffmann <gerd@kraxel.org> Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-id: 20201210223506.263709-1-philmd@redhat.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove unused function audio_is_cleaning_up()Volker Rümelin2020-12-151-8/+0Star
| | | | | | | | | | The previous commit removed the last call site of audio_is_cleaning_up(). Remove the now unused function. Tested-by: Howard Spoelstra <hsp.cat7@gmail.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20201213130528.5863-4-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove qemu_spice_audio_init()Gerd Hoffmann2020-09-231-0/+16
| | | | | | | | | | Handle the spice special case in audio_init instead. With the qemu_spice_audio_init() symbol dependency being gone we can build spiceaudio as module. Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> Message-id: 20200916084117.21828-2-kraxel@redhat.com
* audio: run downstream playback queue unconditionallyVolker Rümelin2020-09-231-0/+3
| | | | | | | | | | Run the downstream playback queue even if there are no samples in the mixing engine buffer. The downstream queue may still have queued samples. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-7-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: align audio_generic_write with audio_pcm_hw_run_outVolker Rümelin2020-09-231-5/+27
| | | | | | | | | | | | | | | The function audio_generic_write should work exactly like audio_pcm_hw_run_out. It's a very similar function working on a different buffer. This patch significantly reduces the number of drop-outs with the DirectSound backend. To hear the difference start qemu with -audiodev dsound,id=audio0,out.mixing-engine=off and play a song in the guest with and without this patch. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-6-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove unnecessary calls to put_buffer_inVolker Rümelin2020-09-231-2/+0Star
| | | | | | | | | | | | | | This patch removes unnecessary calls to the pcm_ops function put_buffer_in(). No audio backend needs this call if the returned length of pcm_ops function get_buffer_in() is zero. For the DirectSound backend this prevents a call to dsound_unlock_in() without a preceding call to dsound_lock_in(). While Windows doesn't complain it seems wrong anyway. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-5-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: align audio_generic_read with audio_pcm_hw_run_inVolker Rümelin2020-09-231-4/+15
| | | | | | | | | | The function audio_generic_read should work exactly like audio_pcm_hw_run_in. It's a very similar function working on a different buffer. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-4-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio/spiceaudio: always rate limit playback streamVolker Rümelin2020-09-231-1/+2
| | | | | | | | | | | | | The playback rate with the spiceaudio backend is currently too fast if there's no spice client connected or the spice client can't play audio. Rate limit the audio playback stream in all cases. To calculate the rate correctly the limiter has to know the maximum buffer size. Fixes: 8c198ff065 ("spiceaudio: port to the new audio backend api") Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-3-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio/audio: fix video playback slowdown with spiceaudioVolker Rümelin2020-09-231-2/+5
| | | | | | | | | | | | | | | | | | | | | | | | This patch allows the audio backends get_buffer_out() functions to drop audio data and mitigates a bug reported on the qemu-devel mailing list. https://lists.nongnu.org/archive/html/qemu-devel/2020-09/msg03832.html The new rules for the variables buf and size returned by get_buffer_out() are: size == 0: Downstream playback buffer is full. Retry later. size > 0, buf != NULL: Copy size bytes to buf for playback. size > 0, buf == NULL: Drop size bytes. The audio playback rate with spiceaudio for the no audio case is too fast, but that's what we had before commit fb35c2cec5 "audio/dsound: fix invalid parameters error". The complete fix comes with the next patch. Reported-by: Qi Zhou <atmgnd@outlook.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200920171729.15861-2-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* qemu/: fix some comment spelling errorszhaolichang2020-09-171-1/+1
| | | | | | | | | | | I found that there are many spelling errors in the comments of qemu, so I used the spellcheck tool to check the spelling errors and finally found some spelling errors in the folder. Signed-off-by: zhaolichang <zhaolichang@huawei.com> Reviewed-by: Alex Bennee <alex.bennee@linaro.org> Message-Id: <20200917075029.313-2-zhaolichang@huawei.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
* audio: fix wavcapture segfaultBruce Rogers2020-05-261-2/+2
| | | | | | | | | | | | | | | Commit 571a8c522e caused the HMP wavcapture command to segfault when processing audio data in audio_pcm_sw_write(), where a NULL sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is valid before dereferincing it. A similar fix is also made in the parallel function audio_pcm_sw_read(). Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and volume_*) Signed-off-by: Bruce Rogers <brogers@suse.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com> Message-id: 20200521172931.121903-1-brogers@suse.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio/jack: add JACK client audiodevGeoffrey McRae2020-05-251-0/+1
| | | | | | | | | This commit adds a new audiodev backend to allow QEMU to use JACK as both an audio sink and source. Signed-off-by: Geoffrey McRae <geoff@hostfission.com> Message-Id: <20200512101603.E3DB73A038E@moya.office.hostfission.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2025-01-15 19:44:25 +0100