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2023-10-03audio: commonize voice initializationPaolo Bonzini1-1/+8
Move some mostly irrelevant code out of audio_init. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-05-05audio/pwaudio.c: Add Pipewire audio backend for QEMUDorinda Bassey1-0/+4
This commit adds a new audiodev backend to allow QEMU to use Pipewire as both an audio sink and source. This backend is available on most systems Add Pipewire entry points for QEMU Pipewire audio backend Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops() qpw_write function returns the current state of the stream to pwaudio and Writes some data to the server for playback streams using pipewire spa_ringbuffer implementation. qpw_read function returns the current state of the stream to pwaudio and reads some data from the server for capture streams using pipewire spa_ringbuffer implementation. These functions qpw_write and qpw_read are called during playback and capture. Added some functions that convert pw audio formats to QEMU audio format and vice versa which would be needed in the pipewire audio sink and source functions qpw_init_in() & qpw_init_out(). These methods that implement playback and recording will create streams for playback and capture that will start processing and will result in the on_process callbacks to be called. Built a connection to the Pipewire sound system server in the qpw_audio_init() method. Signed-off-by: Dorinda Bassey <dbassey@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230417105654.32328-1-dbassey@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
2023-03-06audio: remove sw->ratioVolker Rümelin1-21/+9
Simplify the resample buffer size calculation. For audio playback we have sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq; samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); For audio recording we have sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq; samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); With hw = sw->hw this becomes in both cases samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); Now that sw->ratio is no longer needed, remove sw->ratio. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de>
2023-03-06audio/audio_template: substitute sw->hw with hwVolker Rümelin1-3/+3
Substitute sw->hw with hw in the audio_pcm_sw_alloc_resources_* functions. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-14-vr_qemu@t-online.de>
2023-03-06audio: handle leftover audio frame from upsamplingVolker Rümelin1-0/+6
Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de>
2023-03-06audio: change type and name of the resample bufferVolker Rümelin1-4/+6
Change the type of the resample buffer from struct st_sample * to STSampleBuffer. Also change the name from buf to resample_buf for better readability. The new variables resample_buf.size and resample_buf.pos will be used after the next patches. There is no functional change. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-2-vr_qemu@t-online.de>
2023-03-06audio: change type of mix_buf and conv_bufVolker Rümelin1-8/+11
Change the type of mix_buf in struct HWVoiceOut and conv_buf in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer. However, a buffer pointer is still needed. For this reason in struct STSampleBuffer samples[] is changed to *buffer. This is a preparation for the next patch. The next patch will add this line, which is not possible with the current struct STSampleBuffer definition. + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; There are no functional changes. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-1-vr_qemu@t-online.de>
2023-03-06audio/audio_template: use g_new0() to replace audio_calloc()Volker Rümelin1-17/+12
Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument >= 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-8-vr_qemu@t-online.de>
2023-03-06audio/audio_template: use g_malloc0() to replace audio_calloc()Volker Rümelin1-10/+8
Use g_malloc0() as a direct replacement for audio_calloc(). Since the type of the parameter n_bytes of the function g_malloc0() is unsigned, the type of the variables voice_size_out and voice_size_in has been changed to size_t. This means that the function argument no longer has to be checked for negative values. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-7-vr_qemu@t-online.de>
2023-03-06audio/mixeng: use g_new0() instead of audio_calloc()Volker Rümelin1-5/+1
Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument of 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-5-vr_qemu@t-online.de>
2023-03-06audio: rename hardware store to backendVolker Rümelin1-2/+2
Use a consistent friendly name for the HWVoiceOut and HWVoiceIn structures. Reviewed-by: Thomas Huth <thuth@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230121094735.11644-3-vr_qemu@t-online.de>
2023-03-06audio: don't show unnecessary error messagesVolker Rümelin1-1/+1
Let the audio_pcm_create_voice_pair_* functions handle error reporting. This avoids an additional error message in case the guest selected an unimplemented sample rate. Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-2-vr_qemu@t-online.de>
2023-03-06audio: log unimplemented audio device sample ratesVolker Rümelin1-0/+13
Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-01-30qapi, audio: Make introspection reflect build configuration more closelyDaniel P. Berrangé1-0/+20
Currently the -audiodev accepts any audiodev type regardless of what is built in to QEMU. An error only occurs later at runtime when a sound device tries to use the audio backend. With this change QEMU will immediately reject -audiodev args that are not compiled into the binary. The QMP schema will also be introspectable to identify what is compiled in. This also helps to avoid compiling code that is not required in the binary. Note: When building the audiodevs as modules, the patch only compiles out code for modules that we don't build at all. Signed-off-by: Daniel P. Berrangé <berrange@redhat.com> [thuth: Rebase, take sndio and dbus devices into account] Message-Id: <20230123083957.20349-3-thuth@redhat.com> Signed-off-by: Thomas Huth <thuth@redhat.com>
2022-10-11audio: fix sw->buf size for audio recordingVolker Rümelin1-0/+4
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>
2022-09-27Revert "audio: Log context for audio bug"Volker Rümelin1-12/+15
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>
2022-09-27audio: Add sndio backendAlexandre Ratchov1-0/+2
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>
2022-03-15audio: Log context for audio bugAkihiko Odaki1-15/+12
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>
2021-12-21audio: add "dbus" audio backendMarc-André Lureau1-0/+2
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>
2021-01-15audio: space prohibited between function name and parenthesis'('Zhang Han1-6/+6
Delete spaces between function name and open parenthesis'(' Signed-off-by: Zhang Han <zhanghan64@huawei.com> Message-id: 20210115012431.79533-1-zhanghan64@huawei.com Message-Id: <20210115012431.79533-8-zhanghan64@huawei.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15audio: Add spaces around operator/delete redundant spacesZhang Han1-1/+1
Fix problems about spaces: -operator needs spaces around it, add them. -somespaces are redundant, remove them. Signed-off-by: Zhang Han <zhanghan64@huawei.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com> Message-id: 20210115012431.79533-1-zhanghan64@huawei.com Message-Id: <20210115012431.79533-3-zhanghan64@huawei.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2021-01-15audio: Add braces for statements/fix braces' positionZhang Han1-8/+4
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>
2021-01-15sdlaudio: add -audiodev sdl,out.buffer-count optionVolker Rümelin1-1/+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>
2020-05-25audio/jack: add JACK client audiodevGeoffrey McRae1-0/+2
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>
2020-02-06audio: proper support for float samples in mixengKővágó, Zoltán1-16/+25
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>
2020-02-06coreaudio: fix coreaudio playbackVolker Rümelin1-0/+7
There are reports that since commit 2ceb8240fa "coreaudio: port to the new audio backend api" audio playback with CoreAudio is broken. This patch reverts some parts the commit. Because of changes in the audio subsystem the audio clip function in v4.1.0 of coreaudio.c had to be moved to mixeng.c and the generic buffer management code needed a hint about the size of the float type. This patch is based on a patch from Zoltán Kővágó found at https://lists.nongnu.org/archive/html/qemu-devel/2020-01/msg02142.html. Fixes: 2ceb8240fa "coreaudio: port to the new audio backend api" Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 20200202140641.4737-1-vr_qemu@t-online.de Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18audio: make mixeng optionalKővágó, Zoltán1-8/+16
Implementation of the previously added mixing-engine option. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: c05bc258889ed289e8ee1bdbcc5e84174ec221e7.1570996490.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-10-18audio: fix parameter dereference before NULL checkKővágó, Zoltán1-2/+5
This should fix Coverity issues CID 1405305 and 1405301. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 0eadcc88b8421bb86ce2d68ac70517f920c3ad6c.1568157545.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23audio: split ctl_* functions into enable_* and volume_*Kővágó, Zoltán1-1/+0
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>
2019-09-23audio: unify input and output mixeng buffer managementKővágó, Zoltán1-11/+8
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the initial size of the STSampleBuffer. The next commit tries to fix this inconsistency. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: a78caeb2eeb6348ecb45bb2c81709570ef8ac5b3.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-23audio: api for mixeng code free backendsKővágó, Zoltán1-0/+1
This will make it possible to skip mixeng with audio playback and recording, allowing us to free ourselves from the limitations of the current mixeng (stereo, int64 samples only). In this case, HW and SW voices will be essentially the same, for every SW voice we will create a HW voice, since we can no longer mix multiple voices together. Some backends expect us to call a function when we have data ready write()/read() style, while others provide a buffer and expects us to directly write/read it, so for optimal performance audio_pcm_ops provide methods for both cases. Previously backends asked mixeng for more data in run_out/run_it, now instead mixeng or the frontends will call the backends, so that's why two sets of functions required. audio.c contains glue code between the two styles, so backends only ever have to implement one style and frontends are free to call whichever is more convenient for them. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-21audio: use size_t where makes senseKővágó, Zoltán1-7/+7
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>
2019-08-21audio: basic support for multi backend audioKővágó, Zoltán1-1/+1
Audio functions no longer access glob_audio_state, instead they get an AudioState as a parameter. This is required in order to support multiple backends. glob_audio_state is also gone, and replaced with a tailq so we can store more than one states. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 67aef54f9e729a7160fe95c465351115e392164b.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-08-21audio: reduce glob_audio_state usageKővágó, Zoltán1-24/+22
Remove glob_audio_state from functions, where possible without breaking the API. This means that most static functions in audio.c now take an AudioState pointer instead of implicitly using glob_audio_state. Also included a pointer in SWVoice*, HWVoice* structs, so that functions dealing them can know the audio state without having to pass it around separately. This is required in order to support multiple simultaneous audio backends (added in a later commit). Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-id: b5e241f24e795267b145bcde7c6a72dd5e6037ea.1566168923.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-03-11audio: -audiodev command line option basic implementationKővágó, Zoltán1-4/+38
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>
2018-02-06audio: Replace AUDIO_FUNC with __func__Alistair Francis1-13/+13
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>
2015-06-15audio: remove pliveKővágó, Zoltán1-41/+0
It was useless even 3 years ago, so it can probably safely go away: https://lists.nongnu.org/archive/html/qemu-devel/2012-03/msg02427.html Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2015-06-15audio: expose drv_opaque to init_out and init_inKővágó, Zoltán1-1/+1
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>
2014-12-22audio: Don't free hw resources until after hw backend is stoppedPeter Maydell1-1/+1
When stopping an audio voice, call the audio backend's fini method before calling audio_pcm_hw_free_resources_ rather than afterwards. This allows backends which use helper threads (like pulseaudio) to terminate those threads before the conv_buf or mix_buf are freed and avoids race conditions where the helper may access a NULL pointer or freed memory. Cc: qemu-stable@nongnu.org Reviewed-by: Gerd Hoffmann <kraxel@redhat.com> Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Message-id: 1418406239-9838-1-git-send-email-peter.maydell@linaro.org
2014-06-13audio: Drop superfluous conditionals around g_free()Markus Armbruster1-11/+4
Signed-off-by: Markus Armbruster <armbru@redhat.com> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2012-09-23audio: Fix warning from static code analysisStefan Weil1-3/+3
smatch report: audio/audio_template.h:416 AUD_open_out(18) warn: variable dereferenced before check 'as' (see line 414) Moving the ldebug statement after the statement which checks 'as' fixes that warning. Signed-off-by: Stefan Weil <sw@weilnetz.de> Signed-off-by: malc <av1474@comtv.ru>
2012-04-17audio: don't apply volume effect if backend has VOICE_VOLUME_CAPMarc-André Lureau1-0/+2
If the audio backend is capable of volume control, don't apply software volume (mixeng_volume ()), but instead, rely on backend volume control. This will allow guest to have full range volume control. Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com> Signed-off-by: malc <av1474@comtv.ru>
2011-08-20Use glib memory allocation and free functionsAnthony Liguori1-10/+10
qemu_malloc/qemu_free no longer exist after this commit. Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
2010-09-28audio: Fix memory size for resampling buffer in DAC casemalc1-4/+0
Signed-off-by: malc <av1474@comtv.ru>
2010-05-22Fix %lld or %llx printf format useBlue Swirl1-1/+1
Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
2009-12-03audio: fix compilation of DEBUG_PLIVEJuan Quintela1-3/+3
Signed-off-by: Juan Quintela <quintela@redhat.com> Signed-off-by: Anthony Liguori <aliguori@us.ibm.com>
2009-11-18audio: Remove conditional around sw which can not be NULLmalc1-17/+15
Noticed by Steve Grubb. Signed-off-by: malc <av1474@comtv.ru>
2009-10-15audio: remove last remnants of _tmalc1-1/+1
Signed-off-by: malc <av1474@comtv.ru>
2009-09-18audio: use muldiv64 where it makes sensemalc1-1/+1
Signed-off-by: malc <av1474@comtv.ru>
2009-09-12Fix sys-queue.h conflict for goodBlue Swirl1-6/+6
Problem: Our file sys-queue.h is a copy of the BSD file, but there are some additions and it's not entirely compatible. Because of that, there have been conflicts with system headers on BSD systems. Some hacks have been introduced in the commits 15cc9235840a22c289edbe064a9b3c19c5f49896, f40d753718c72693c5f520f0d9899f6e50395e94, 96555a96d724016e13190b28cffa3bc929ac60dc and 3990d09adf4463eca200ad964cc55643c33feb50 but the fixes were fragile. Solution: Avoid the conflict entirely by renaming the functions and the file. Revert the previous hacks. Signed-off-by: Blue Swirl <blauwirbel@gmail.com>
generated by cgit v1.1 at 2024-09-04 07:02:48 +0000