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* audio/alsaaudio: Remove superfluous semicolonsPhilippe Mathieu-Daudé2020-02-181-2/+2
| | | | | | | | | | Fixes: 286a5d201e4 Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com> Acked-by: Paolo Bonzini <pbonzini@redhat.com> Reviewed-by: Dr. David Alan Gilbert <dgilbert@redhat.com> Reviewed-by: Juan Quintela <quintela@redhat.com> Message-Id: <20200218094402.26625-3-philmd@redhat.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
* audio: proper support for float samples in mixengKővágó, Zoltán2020-02-061-0/+17
| | | | | | | | | | | | | | | This adds proper support for float samples in mixeng by adding a new audio format for it. Limitations: only native endianness is supported. None of the virtual sound cards support float samples (it looks like most of them only support 8 and 16 bit, only hda supports 32 bit), it is only used for the audio backends (i.e. host side). Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Acked-by: Markus Armbruster <armbru@redhat.com> Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: fix bug 1858488Volker Rümelin2020-01-311-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: basic support for multichannel audioKővágó, Zoltán2019-10-181-7/+0Star
| | | | | | | | | Which currently only means removing some checks. Old code won't require more than two channels, but new code will need it. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 7e53be1f97e939ed3bb729ef39e76b775643118a.1570996490.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: replace shift in audio_pcm_info with bytes_per_frameKővágó, Zoltán2019-10-181-5/+6
| | | | | | | | | | | The bit shifting trick worked because the number of bytes per frame was always a power-of-two (since QEMU only supports mono, stereo and 8, 16 and 32 bit samples). But if we want to add support for surround sound, this no longer holds true. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: split ctl_* functions into enable_* and volume_*Kővágó, Zoltán2019-09-231-37/+25Star
| | | | | | | | | | This way we no longer need vararg functions, improving compile time error detection. Also now it's possible to check actually what commands are supported, without needing to manually update ctl_caps. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 2b08b3773569c5be055d0a0fb2f29ff64e79f0f4.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* alsaaudio: port to the new audio backend apiKővágó, Zoltán2019-09-231-222/+80Star
| | | | | | Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: ab9768e73dfe7b7305bd6a51629846e0d77622a5.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: use size_t where makes senseKővágó, Zoltán2019-08-211-13/+13
| | | | | | Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: c5193e687fc6cc0f60cb3e90fe69ddf2027d0df1.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove read and write pcm_opsKővágó, Zoltán2019-08-211-12/+0Star
| | | | | | | | | | They just called audio_pcm_sw_read/write anyway, so it makes no sense to have them too. (The noaudio's read is the only exception, but it should work with the generic code too.) Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 92ddc98133bc4b687c6e4608b9321e7b64c0e496.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: do not run each backend in audio_runKővágó, Zoltán2019-08-211-2/+5
| | | | | | | | | | audio_run is called manually by alsa and oss backends when polling. In this case only the requesting backend should be run, not all of them. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-id: 10221fcea2028fa18d95cf531526ffe3b1d9b21a.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: remove audio_MIN, audio_MAXKővágó, Zoltán2019-08-211-3/+3
| | | | | | | | | | There's already a MIN and MAX macro in include/qemu/osdep.h, use them instead. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-id: 303222477df6f7373217e0df768635fab5855745.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* Include qemu/module.h where needed, drop it from qemu-common.hMarkus Armbruster2019-06-121-1/+2
| | | | | | | | | Signed-off-by: Markus Armbruster <armbru@redhat.com> Message-Id: <20190523143508.25387-4-armbru@redhat.com> [Rebased with conflicts resolved automatically, except for hw/usb/dev-hub.c hw/misc/exynos4210_rng.c hw/misc/bcm2835_rng.c hw/misc/aspeed_scu.c hw/display/virtio-vga.c hw/arm/stm32f205_soc.c; ui/cocoa.m fixed up]
* alsaaudio: port to -audiodev configKővágó, Zoltán2019-03-111-233/+100Star
| | | | | | Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 663d2c918a11ef44d4042e56c796d6dbf40be70c.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: -audiodev command line option basic implementationKővágó, Zoltán2019-03-111-1/+1
| | | | | | | | | | | | | | | | | | | | Audio drivers now get an Audiodev * as config paramters, instead of the global audio_option structs. There is some code in audio/audio_legacy.c that converts the old environment variables to audiodev options (this way backends do not have to worry about legacy options). It also contains a replacement of -audio-help, which prints out the equivalent -audiodev based config of the currently specified environment variables. Note that backends are not updated and still rely on environment variables. Also note that (due to moving try-poll from global to backend specific option) currently ALSA and OSS will always try poll mode, regardless of environment variables or -audiodev options. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: e99a7cbdac0d13512743880660b2032024703e4c.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: use qapi AudioFormat instead of audfmt_eKővágó, Zoltán2019-03-111-25/+28
| | | | | | | | | | | | | | I had to include an enum for audio sampling formats into qapi, but that meant duplicating the audfmt_e enum. This patch replaces audfmt_e and associated values with the qapi generated AudioFormat enum. This patch is mostly a search-and-replace, except for switches where the qapi generated AUDIO_FORMAT_MAX caused problems. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Thomas Huth <thuth@redhat.com> Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio/alsaaudio: Remove compiler check around pragmaThomas Huth2018-12-121-2/+0Star
| | | | | | | | Both GCC v4.8 and Clang v3.4 support the -Waddress option, so we do not need the compiler version check here anymore. Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Thomas Huth <thuth@redhat.com>
* audio: add driver registryGerd Hoffmann2018-03-121-1/+7
| | | | | | | | | | | Add registry for audio drivers, using the existing audio_driver struct. Make all drivers register themself. The old list of audio_driver struct pointers is now a list of audio driver names, specifying the priority (aka probe order) in case no driver is explicitly asked for. Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-id: 20180306074053.22856-2-kraxel@redhat.com
* audio: Replace AUDIO_FUNC with __func__Alistair Francis2018-02-061-2/+2
| | | | | | | | | | | | | | | | | | Apparently we don't use __MSC_VER as a compiler anymore and we always require a C99 compiler (which means we always have __func__) so we don't need a special AUDIO_FUNC macro. We can just replace AUDIO_FUNC with __func__ instead. Checkpatch failures were manually fixed. Signed-off-by: Alistair Francis <alistair.francis@xilinx.com> Cc: Gerd Hoffmann <kraxel@redhat.com> Reviewed-by: Thomas Huth <thuth@redhat.com> Reviewed-by: Eric Blake <eblake@redhat.com> Reviewed-by: Gerd Hoffmann <kraxel@redhat.com> Signed-off-by: Markus Armbruster <armbru@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org> Message-Id: <20180203084315.20497-2-armbru@redhat.com>
* audio: Clean up includesPeter Maydell2016-02-021-0/+1
| | | | | | | | | | | Clean up includes so that osdep.h is included first and headers which it implies are not included manually. This commit was created with scripts/clean-includes. Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Message-id: 1453138432-8324-1-git-send-email-peter.maydell@linaro.org Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* alsaaudio: use trace events instead of verboseKővágó, Zoltán2015-06-151-46/+14Star
| | | | | Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* alsaaudio: do not use global variablesKővágó, Zoltán2015-06-151-68/+82
| | | | | Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* audio: expose drv_opaque to init_out and init_inKővágó, Zoltán2015-06-151-2/+3
| | | | | | | | | | | Currently the opaque pointer returned by audio_driver's init is only exposed to the driver's fini, but not to audio_pcm_ops. This way if someone wants to share a variable with the driver and the pcm, he must use global variables. This patch fixes it by adding a third parameter to audio_pcm_op's init_out and init_in. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* alsaaudio: Remove unused error handling of qemu_set_fd_handlerFam Zheng2015-06-121-14/+2Star
| | | | | | | | The function cannot fail, so the check is superfluous. Signed-off-by: Fam Zheng <famz@redhat.com> Message-id: 1433400324-7358-10-git-send-email-famz@redhat.com Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
* audio: Drop superfluous conditionals around g_free()Markus Armbruster2014-06-131-8/+4Star
| | | | | Signed-off-by: Markus Armbruster <armbru@redhat.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
* misc: move include files to include/qemu/Paolo Bonzini2012-12-191-1/+1
| | | | Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* janitor: do not include qemu-char everywherePaolo Bonzini2012-12-191-1/+0Star
| | | | | | | Touching char/char.h basically causes the whole of QEMU to be rebuilt. Avoid this, it is usually unnecessary. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* janitor: do not rely on indirect inclusions of or from qemu-char.hPaolo Bonzini2012-12-191-0/+1
| | | | | | | | | Various header files rely on qemu-char.h including qemu-config.h or main-loop.h, but they really do not need qemu-char.h at all (particularly interesting is the case of the block layer!). Clean this up, and also add missing inclusions of qemu-char.h itself. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
* Use glib memory allocation and free functionsAnthony Liguori2011-08-211-5/+5
| | | | | | qemu_malloc/qemu_free no longer exist after this commit. Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
* audio: split sample conversion and volume mixingMichael Walle2011-01-121-1/+1
| | | | | | | | | Refactor the volume mixing, so it can be reused for capturing devices. Additionally, it removes superfluous multiplications with the nominal volume within the hardware voice code path. Signed-off-by: Michael Walle <michael@walle.cc> Signed-off-by: malc <av1474@comtv.ru>
* alsaaudio: add endianness support for VoiceInMichael Walle2011-01-091-7/+27
| | | | | Signed-off-by: Michael Walle <michael@walle.cc> Signed-off-by: malc <av1474@comtv.ru>
* issue snd_pcm_start() when capturing audioJindrich Makovicka2010-10-171-6/+17
| | | | | | | | | snd_pcm_start() starts the capture process and ensures that the events are delivered to the poll handler. Without the call, capture can be started only when there is simultaneous playback running. Signed-off-by: Jindrich Makovicka <makovick@gmail.com> Signed-off-by: malc <av1474@comtv.ru>
* fix 100% CPU load when idle with ALSAJindrich Makovicka2010-10-171-0/+4
| | | | | | | | Playback control function did not disable polling when playback stops. Caused busy spinning of the main loop due to unprocessed events. Signed-off-by: Jindrich Makovicka <makovick@gmail.com> Signed-off-by: malc <av1474@comtv.ru>
* audio/alsa: Avoid snd_pcm_format_t vs audfmt_e mixupmalc2010-04-211-5/+6
| | | | | | Spotted by Serge Ziryukin and based on his patch, thanks. Signed-off-by: malc <av1474@comtv.ru>
* audio/alsa: Handle SND_PCM_STATE_SETUP in alsa_poll_handlermalc2010-02-281-0/+4
| | | | Signed-off-by: malc <av1474@comtv.ru>
* audio/alsa: Spelling typo (paramters)Vagrant Cascadian2010-02-281-1/+1
| | | | | | Trivial patch to fix the spelling of "parameters". Signed-off-by: malc <av1474@comtv.ru>
* oss/alsa: Do not invoke UB described in 7.15.1.1 (this time for ADC)malc2009-10-031-12/+15
| | | | Signed-off-by: malc <av1474@comtv.ru>
* alsa: Change default buffer/period sizemalc2009-10-021-1/+2
| | | | | | | Increase buffer size but do not rely on ALSA picking up default period size. Signed-off-by: malc <av1474@comtv.ru>
* oss/alsa: Do not invoke UB described in 7.15.1.1malc2009-10-021-11/+14
| | | | | | | Additional argument (whether to try poll mode) is only passed with VOICE_ENABLE command. Thanks to Markus Armbruster for noticing the potential breakage.
* audio: internal API changemalc2009-09-181-7/+2Star
| | | | | | | | | pcm_ops.run_out now takes number of live samples (which will be always greater than zero) as a second argument, every driver was calling audio_pcm_hw_get_live_out anyway with exception of fmod which used audio_pcm_hw_get_live_out2 for no good reason. Signed-off-by: malc <av1474@comtv.ru>
* alsa: use audio_pcm_hw_clip_outmalc2009-09-181-45/+43Star
| | | | Signed-off-by: malc <av1474@comtv.ru>
* alsa: Use proper value when testing returned events in alsa_poll_handlermalc2009-09-141-4/+6
| | | | Signed-off-by: malc <av1474@comtv.ru>
* alsa/oss: Remove fd transfer handlers before closing oss/alsa fd/handlemalc2009-09-141-25/+28
| | | | Signed-off-by: malc <av1474@comtv.ru>
* alsa: poll mode handlingmalc2009-09-121-21/+204
| | | | Signed-off-by: malc <av1474@comtv.ru>
* Aestheticsmalc2009-08-111-48/+67
| | | | | | | Reformat to make item borders more visible Fix cases of stray tabs and vertical misalignments Signed-off-by: malc <av1474@comtv.ru>
* use C99 initializers for all audio/*Juan Quintela2009-08-111-2/+4
| | | | Signed-off-by: Juan Quintela <quintela@redhat.com>
* use C99 initializers for audio_pcm_opsJuan Quintela2009-08-111-11/+11
| | | | Signed-off-by: Juan Quintela <quintela@redhat.com>
* Use C99 initializers for audio_optionJuan Quintela2009-08-111-31/+45
| | | | Signed-off-by: Juan Quintela <quintela@redhat.com>
* Use proper struct initializers and remove INIT_FIELD() macroJuan Quintela2009-08-111-11/+11
| | | | Signed-off-by: Juan Quintela <quintela@redhat.com>
* alsa: add host suspend/resume supportBjørn Mork2009-07-301-2/+47
| | | | | | | | | | | | | Both input and output streams may be in SND_PCM_STATE_SUSPENDED after the host is suspended and resumed, meaning "Hardware is suspended". snd_pcm_readi() and snd_pcm_writei() will return -ESTRPIPE if called while the stream is in this state. Call snd_pcm_resume() to enable audio output and capture after host resume. Signed-off-by: Bjørn Mork <bjorn@mork.no> Signed-off-by: malc <av1474@comtv.ru>
* Ignore -Waddress for alsaaudio.cmalc2009-07-171-0/+4
| | | | | | | | | | /usr/include/alsa/pcm.h contains: #define snd_pcm_sw_params_alloca(ptr) do { assert(ptr); *ptr = (snd_pcm_sw_params_t *) alloca(snd_pcm_sw_params_sizeof()); memset(*ptr, 0, snd_pcm_sw_params_sizeof()); } while (0) The assert generates: "error: the address of 'sw_params' will always evaluate as 'true'" which combined with -Werror prevents alsaaudio.o from being built with certain versions of GCC.
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