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2023-05-05audio/pwaudio.c: Add Pipewire audio backend for QEMUDorinda Bassey
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>
2023-03-13audio/dbus: there are no sender for p2p modeMarc-André Lureau
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
2023-03-06audio: remove sw->ratioVolker Rümelin
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 Rümelin
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 Rümelin
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: make recording packet length calculation exactVolker Rümelin
Introduce the new function st_rate_frames_out() to calculate the exact number of audio output frames the resampling code can generate from a given number of audio input frames. When upsampling, this function returns the maximum number of output frames. This new function replaces the audio_frontend_frames_in() function, which calculated the average number of output frames rounded down to the nearest integer. The audio_frontend_frames_in() function was additionally used to limit the number of output frames to the resample buffer size. In audio_pcm_sw_read() the variable resample_buf.size replaces the open coded audio_frontend_frames_in() function. In audio_run_in() an additional MIN() function is necessary. After this patch the audio packet length calculation for audio recording is exact. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-12-vr_qemu@t-online.de>
2023-03-06audio: rename variables in audio_pcm_sw_read()Volker Rümelin
The audio_pcm_sw_read() function uses a few very unspecific variable names. Rename them for better readability. ret => total_out total => total_in size => buf_len samples => frames_out_max 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: <20230224190555.7409-11-vr_qemu@t-online.de>
2023-03-06audio: replace the resampling loop in audio_pcm_sw_read()Volker Rümelin
Replace the resampling loop in audio_pcm_sw_read() with the new function audio_pcm_sw_resample_in(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. 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: <20230224190555.7409-10-vr_qemu@t-online.de>
2023-03-06audio: make playback packet length calculation exactVolker Rümelin
Introduce the new function st_rate_frames_in() to calculate the exact number of audio input frames needed to get a given number of audio output frames. The exact number of frames depends only on the difference of opos - ipos and the number of output frames. When downsampling, this function returns the maximum number of input frames needed. This new function replaces the audio_frontend_frames_out() function, which calculated the average number of input frames rounded down to the nearest integer. Because audio_frontend_frames_out() also limited the number of input frames to the size of the resample buffer, st_rate_frames_in() is not a direct replacement and two additional MIN() functions are needed. One to prevent resample buffer overflows and one to limit the available bytes for the audio frontends. After this patch the audio packet length calculation for playback is exact. When upsampling, it's still possible that the audio frontends can't write the last audio frame. This will be fixed later. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de>
2023-03-06audio: remove unused noop_conv() functionVolker Rümelin
The function audio_capture_mix_and_clear() no longer uses audio_pcm_sw_write() to resample audio frames from one internal buffer to another. For this reason, the noop_conv() function is now unused. Remove it. 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: <20230224190555.7409-8-vr_qemu@t-online.de>
2023-03-06audio: don't misuse audio_pcm_sw_write()Volker Rümelin
The audio_pcm_sw_write() function is intended to convert a PCM audio stream to the internal representation, adjust the volume, and then mix it with the other audio streams with a possibly changed sample rate in mix_buf. In order for the audio_capture_mix_and_clear() function to use audio_pcm_sw_write(), it must bypass the first two tasks of audio_pcm_sw_write(). Since patch "audio: split out the resampling loop in audio_pcm_sw_write()" this is no longer necessary, because now the audio_pcm_sw_resample_out() function can be used instead of audio_pcm_sw_write(). 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: <20230224190555.7409-7-vr_qemu@t-online.de>
2023-03-06audio: rename variables in audio_pcm_sw_write()Volker Rümelin
The audio_pcm_sw_write() function uses a lot of very unspecific variable names. Rename them for better readability. ret => total_in total => total_out size => buf_len hwsamples => hw->mix_buf.size samples => frames_in_max 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: <20230224190555.7409-6-vr_qemu@t-online.de>
2023-03-06audio: remove sw == NULL checkVolker Rümelin
All call sites of audio_pcm_sw_write() guarantee that sw is not NULL. Remove the unnecessary NULL check. 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: <20230224190555.7409-5-vr_qemu@t-online.de>
2023-03-06audio: replace the resampling loop in audio_pcm_sw_write()Volker Rümelin
Replace the resampling loop in audio_pcm_sw_write() with the new function audio_pcm_sw_resample_out(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. 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: <20230224190555.7409-4-vr_qemu@t-online.de>
2023-03-06audio: make the resampling code greedyVolker Rümelin
Read the maximum possible number of audio frames instead of the minimum necessary number of frames when the audio stream is downsampled and the output buffer is limited. This makes the function symmetrical to upsampling when the input buffer is limited. The maximum possible number of frames is written here. With this change it's easier to calculate the exact number of audio frames the resample function will read or write. These two functions will be introduced later. 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: <20230224190555.7409-3-vr_qemu@t-online.de>
2023-03-06audio: change type and name of the resample bufferVolker Rümelin
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 Rümelin
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>
2023-03-06alsaaudio: reintroduce default recording settingsVolker Rümelin
Audio recording with ALSA default settings currently doesn't work. The debug log shows updates every 0.75s and 1.5s. audio: Elapsed since last alsa run (running): 0.743030 audio: Elapsed since last alsa run (running): 1.486048 audio: Elapsed since last alsa run (running): 0.743008 audio: Elapsed since last alsa run (running): 1.485878 audio: Elapsed since last alsa run (running): 1.486040 audio: Elapsed since last alsa run (running): 1.485886 The time between updates should be in the 10ms range. Audio recording with ALSA has the same timing contraints as playback. Reintroduce the default recording settings and use the same default settings for recording as for playback. The term "reintroduce" is correct because commit a93f328177 ("alsaaudio: port to -audiodev config") removed the default settings for recording. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-11-vr_qemu@t-online.de>
2023-03-06alsaaudio: change default playback settingsVolker Rümelin
The currently used default playback settings in the ALSA audio backend are a bit unfortunate. With a few emulated audio devices, audio playback does not work properly. Here is a short part of the debug log while audio is playing (elapsed time in seconds). audio: Elapsed since last alsa run (running): 0.046244 audio: Elapsed since last alsa run (running): 0.023137 audio: Elapsed since last alsa run (running): 0.023170 audio: Elapsed since last alsa run (running): 0.023650 audio: Elapsed since last alsa run (running): 0.060802 audio: Elapsed since last alsa run (running): 0.031931 For some audio devices the time of more than 23ms between updates is too long. Set the period time to 5.8ms so that the maximum time between two updates typically does not exceed 11ms. This roughly matches the 10ms period time when doing playback with the audio timer. After this patch the debug log looks like this. audio: Elapsed since last alsa run (running): 0.011919 audio: Elapsed since last alsa run (running): 0.005788 audio: Elapsed since last alsa run (running): 0.005995 audio: Elapsed since last alsa run (running): 0.011069 audio: Elapsed since last alsa run (running): 0.005901 audio: Elapsed since last alsa run (running): 0.006084 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: <20230121094735.11644-10-vr_qemu@t-online.de>
2023-03-06audio: remove audio_calloc() functionVolker Rümelin
Now that the last call site of audio_calloc() was removed, remove the unused audio_calloc() function. 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: <20230121094735.11644-9-vr_qemu@t-online.de>
2023-03-06audio/audio_template: use g_new0() to replace audio_calloc()Volker Rümelin
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 Rümelin
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/alsaaudio: use g_new0() instead of audio_calloc()Volker Rümelin
Replace audio_calloc() with the equivalent g_new0(). The value of the g_new0() argument count is >= 1, which means g_new0() will never return NULL. Also remove the unnecessary NULL check. Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> 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: <20230121094735.11644-6-vr_qemu@t-online.de>
2023-03-06audio/mixeng: use g_new0() instead of audio_calloc()Volker Rümelin
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: remove unused #define AUDIO_STRINGIFYVolker Rümelin
Remove the unused #define AUDIO_STRINGIFY. It was last used before commit 470bcabd8f ("audio: Replace AUDIO_FUNC with __func__"). Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> 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> Message-Id: <20230121094735.11644-4-vr_qemu@t-online.de>
2023-03-06audio: rename hardware store to backendVolker Rümelin
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 Rümelin
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 Rümelin
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>
2023-02-08Fix non-first inclusions of qemu/osdep.hMarkus Armbruster
This commit was created with scripts/clean-includes. Signed-off-by: Markus Armbruster <armbru@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Michael S. Tsirkin <mst@redhat.com> Reviewed-by: Juan Quintela <quintela@redhat.com> Message-Id: <20230202133830.2152150-18-armbru@redhat.com>
2023-02-04Merge tag 'pull-monitor-2023-02-03-v2' of https://repo.or.cz/qemu/armbru ↵Peter Maydell
into staging Monitor patches for 2023-02-03 # -----BEGIN PGP SIGNATURE----- # # iQJGBAABCAAwFiEENUvIs9frKmtoZ05fOHC0AOuRhlMFAmPeAkgSHGFybWJydUBy # ZWRoYXQuY29tAAoJEDhwtADrkYZTUagP/iZ24jXaWoFOKaO70wdQ/tdoQObWZnUV # 8xJNJYmYYbWoiq9wQXHebi/yEgBudso1lLzAnp8lsF12ybnNV1zsjyV/yumEKSNW # 3nL1NZIcuY9IDmCe97clY9nm9H2lUhjjyCG3gnjg+uC3JjlSjO/T8lbkdT+fYnkR # AInVTCPYFjSO9MIOhN0WNIY73HlAjr4zx5TEgS/D4pFj6iGq2qEniSDGMRf+/fVr # uSbIXbQlum+VAdxbGMSVf8yQPlNcFUXUpSJrbgJE272H6saQuvn5mkwD0RcYXyaI # OlfXpATDRNTsP3yYImxgr7y29Exo1HnCuC6T1n/+fwkirtMR3a7X6XjaQwFsWcrx # xxGiHQOve3r/I3DAO6A64T2ceD/XuI43LygqkkljfuoXifnJz7Lo39P9HrY0dhpC # KSld2n/Vv4xYyykvqAzpvzijwq679ILIbTplhm9gOrfrDRZjWad3uLAcYxsTXXR8 # BQbHGovcAzTOEx/0Quo3NThpAeNYPGyrPz3xBIV+XtPJGWvFsrA/s/po4qWDTmF6 # UTzPoEmznsD+DRboNOKfinCsOnpTAru4gbXevi7sfmMHQbLYN5xgsrF7WdlaxWa6 # 4QbJyNUq0O+aL0gyfVLuiZBCQ32Jaz1WvowK856Yl4jwczP5HM0ujyyM75+Kx072 # PdnMgxYYLSij # =d+wL # -----END PGP SIGNATURE----- # gpg: Signature made Sat 04 Feb 2023 06:59:20 GMT # gpg: using RSA key 354BC8B3D7EB2A6B68674E5F3870B400EB918653 # gpg: issuer "armbru@redhat.com" # gpg: Good signature from "Markus Armbruster <armbru@redhat.com>" [full] # gpg: aka "Markus Armbruster <armbru@pond.sub.org>" [full] # Primary key fingerprint: 354B C8B3 D7EB 2A6B 6867 4E5F 3870 B400 EB91 8653 * tag 'pull-monitor-2023-02-03-v2' of https://repo.or.cz/qemu/armbru: (35 commits) monitor: Rename misc.c to hmp-target.c monitor: Loosen coupling between misc.c and monitor.c slightly monitor: Move remaining QMP stuff from misc.c to qmp-cmds.c monitor: Move remaining HMP commands from misc.c to hmp-cmds.c monitor: Move target-dependent HMP commands to hmp-cmds-target.c monitor: Move monitor_putc() next to monitor_puts & external linkage monitor: Split file descriptor passing stuff off misc.c qdev: Move HMP command completion from monitor to softmmu/ acpi: Move the QMP command from monitor/ to hw/acpi/ stats: Move HMP commands from monitor/ to stats/ stats: Move QMP commands from monitor/ to stats/ runstate: Move HMP commands from monitor/ to softmmu/ tpm: Move HMP commands from monitor/ to softmmu/ virtio: Move HMP commands from monitor/ to hw/virtio/ migration: Move the QMP command from monitor/ to migration/ migration: Move HMP commands from monitor/ to migration/ net: Move hmp_info_network() to net-hmp-cmds.c net: Move HMP commands from monitor to net/ hmp: Rewrite strlist_from_comma_list() as hmp_split_at_comma() rocker: Move HMP commands from monitor to hw/net/rocker/ ... Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
2023-02-04audio: Move HMP commands from monitor/ to audio/Markus Armbruster
This moves these commands from MAINTAINERS sections "Human Monitor (HMP)" and "QMP" to "Overall Audio backends". Signed-off-by: Markus Armbruster <armbru@redhat.com> Message-Id: <20230124121946.1139465-3-armbru@redhat.com>
2023-01-30qapi, audio: Make introspection reflect build configuration more closelyDaniel P. Berrangé
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>
2023-01-30qapi, audio: add query-audiodev commandDaniel P. Berrangé
Way back in QEMU 4.0, the -audiodev command line option was introduced for configuring audio backends. This CLI option does not use QemuOpts so it is not visible for introspection in 'query-command-line-options', instead using the QAPI Audiodev type. Unfortunately there is also no QMP command that uses the Audiodev type, so it is not introspectable with 'query-qmp-schema' either. This introduces a 'query-audiodev' command that simply reflects back the list of configured -audiodev command line options. This alone is maybe not very useful by itself, but it makes Audiodev introspectable via 'query-qmp-schema', so that libvirt (and other upper layer tools) can discover the available audiodevs. Signed-off-by: Daniel P. Berrangé <berrange@redhat.com> [thuth: Update for upcoming QEMU v8.0, and use QAPI_LIST_PREPEND] Message-Id: <20230123083957.20349-2-thuth@redhat.com> Signed-off-by: Thomas Huth <thuth@redhat.com>
2022-12-13qapi audio: Elide redundant has_FOO in generated CMarkus Armbruster
The has_FOO for pointer-valued FOO are redundant, except for arrays. They are also a nuisance to work with. Recent commit "qapi: Start to elide redundant has_FOO in generated C" provided the means to elide them step by step. This is the step for qapi/audio.json. Said commit explains the transformation in more detail. The invariant violations mentioned there do not occur here. Additionally, helper get_str() loses its @has_dst parameter. Cc: Gerd Hoffmann <kraxel@redhat.com> Signed-off-by: Markus Armbruster <armbru@redhat.com> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20221104160712.3005652-8-armbru@redhat.com>
2022-11-06module: add Error arguments to module_load and module_load_qomClaudio Fontana
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>
2022-11-06module: rename module_load_one to module_loadClaudio Fontana
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>
2022-10-12audio: improve out.voices testHelge Konetzka
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>
2022-10-12audio: fix in.voices testHelge Konetzka
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>
2022-10-11audio: prevent an integer overflow in resampling codeVolker Rümelin
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>
2022-10-11audio: fix sw->buf size for audio recordingVolker Rümelin
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>
2022-10-11audio: refactor audio_get_avail()Volker Rümelin
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>
2022-10-11audio: rename audio_sw_bytes_free()Volker Rümelin
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>
2022-10-11audio: swap audio_rate_get_bytes() function parametersVolker Rümelin
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>
2022-10-11spiceaudio: update commentVolker Rümelin
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>
2022-10-11spiceaudio: add a pcm_ops buffer_get_free functionVolker Rümelin
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>
2022-10-11audio: add more audio rate control functionsVolker Rümelin
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>
2022-10-11alsaaudio: reduce playback latencyVolker Rümelin
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>
2022-10-11audio: run downstream playback queue unconditionallyVolker Rümelin
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>
2022-10-11audio: fix GUS audio playback with out.mixing-engine=offVolker Rümelin
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>
2022-10-11audio: refactor code in audio_run_out()Volker Rümelin
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>
generated by cgit v1.2.3 (git 2.26.2) at 2024-12-19 19:01:48 +0000