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2023-10-09Merge tag 'for-upstream' of https://gitlab.com/bonzini/qemu into stagingStefan Hajnoczi2-51/+37
* util/log: re-allow switching away from stderr log file * finish audio configuration rework * cleanup HVF stubs * remove more mentions of softmmu # -----BEGIN PGP SIGNATURE----- # # iQFIBAABCAAyFiEE8TM4V0tmI4mGbHaCv/vSX3jHroMFAmUi/kIUHHBib256aW5p # QHJlZGhhdC5jb20ACgkQv/vSX3jHroOXWwf/YW16QMzqdAPVHYRf9NcCneRF16El # t3lEod0q0sHhchPbh9e04aKbh+oBNeWu9sFyTl11Fwsi+DGmp/b28ziva75/4rfd # h5N9aX/z2jwPqy93IwPDu3soKXCCgTK+ywtD/5GLQwBGqxs7W2xUEEb7eCnVefHa # zwL3MOUqPICeqOnR1TNw9k3N3veF04D+rmchTwbAjAmx1f8EI+mK9VlGK9V8TUjP # 3HjpZYJluc0a92lR5VONJ7V25QfttsjLysTgpFwVAQPS6Frzatc/hWclfLYgw9vl # 2Irk83FV8gXPRl0XKNcqSDsv6h/yGP6TDFIB8QwRSRGBqIQi5aOlfBJzsQ== # =qbm7 # -----END PGP SIGNATURE----- # gpg: Signature made Sun 08 Oct 2023 15:08:50 EDT # gpg: using RSA key F13338574B662389866C7682BFFBD25F78C7AE83 # gpg: issuer "pbonzini@redhat.com" # gpg: Good signature from "Paolo Bonzini <bonzini@gnu.org>" [full] # gpg: aka "Paolo Bonzini <pbonzini@redhat.com>" [full] # Primary key fingerprint: 46F5 9FBD 57D6 12E7 BFD4 E2F7 7E15 100C CD36 69B1 # Subkey fingerprint: F133 3857 4B66 2389 866C 7682 BFFB D25F 78C7 AE83 * tag 'for-upstream' of https://gitlab.com/bonzini/qemu: (25 commits) audio, qtest: get rid of QEMU_AUDIO_DRV audio: reintroduce default audio backend for VNC audio: do not use first -audiodev as default audio device audio: extend -audio to allow creating a default backend audio: extract audio_define_default audio: disable default backends if -audio/-audiodev is used audio: error hints need a trailing \n cutils: squelch compiler warnings with custom paths configure: change $softmmu to $system system: Rename softmmu/ directory as system/ meson: Rename target_softmmu_arch -> target_system_arch meson: Rename softmmu_mods -> system_mods target/i386: Rename i386_softmmu_kvm_ss -> i386_kvm_ss semihosting: Rename softmmu_FOO_user() -> uaccess_FOO_user() gdbstub: Rename 'softmmu' -> 'system' accel: Rename accel_softmmu* -> accel_system* tcg: Correct invalid mentions of 'softmmu' by 'system-mode' fuzz: Correct invalid mentions of 'softmmu' by 'system' cpu: Correct invalid mentions of 'softmmu' by 'system-mode' travis-ci: Correct invalid mentions of 'softmmu' by 'system' ... Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>
2023-10-08audio, qtest: get rid of QEMU_AUDIO_DRVPaolo Bonzini1-12/+0
Default audio devices can now be created with "-audio". Tests for soundcards were already using "-audiodev" if they want to specify a particular backend, for the others remove the last remnants of legacy audio configuration. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: reintroduce default audio backend for VNCPaolo Bonzini2-10/+19
Make VNC use the default backend again if one is defined. The recently introduced support for disabling the VNC audio extension is still used, in case no default backend exists. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: do not use first -audiodev as default audio devicePaolo Bonzini1-18/+7
It is now possible to specify the options for the default audio device using -audio, so there is no need anymore to use a fake -audiodev option. Remove the fall back to QTAILQ_FIRST(&audio_states), instead remember the AudioState that was created from default_audiodevs and use that one. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: extend -audio to allow creating a default backendPaolo Bonzini1-1/+1
If "-audio BACKEND" is used without a model, the resulting backend will be used whenever the audiodev property is not specified. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: extract audio_define_defaultPaolo Bonzini2-7/+13
It will be used soon to define a default audio device from the command line. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: disable default backends if -audio/-audiodev is usedPaolo Bonzini2-8/+2
Match what is done for other options, for example -monitor, and also the behavior of QEMU 8.1 (see the "legacy_config" variable). Require the user to specify a backend if one is specified on the command line. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-08audio: error hints need a trailing \nPaolo Bonzini1-1/+1
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-06audio/ossaudio: Fix compiler warning with -WshadowThomas Huth1-1/+0
The "err" variable is only used twice in this code, in a very local fashion of first assigning it and then checking it in the next line. So there is no need to declare this variable a second time in the innermost block, we can re-use the variable that is declared at the beginning of the function. This fixes the compiler warning that occurs with "-Wshadow". Signed-off-by: Thomas Huth <thuth@redhat.com> Message-ID: <20231004083900.95856-1-thuth@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Signed-off-by: Markus Armbruster <armbru@redhat.com>
2023-10-03audio: forbid default audiodev backend with -nodefaultsPaolo Bonzini1-0/+16
Now that all callers support setting an audiodev, forbid using the default audiodev if -nodefaults is provided on the command line. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: propagate Error * out of audio_initMartin Kletzander2-21/+20
Starting from audio_driver_init, propagate errors via Error ** so that audio_init_audiodevs can simply pass &error_fatal, and AUD_register_card can signal faiure. Signed-off-by: Martin Kletzander <mkletzan@redhat.com> [Reworked the audio/audio.c parts, while keeping Martin's hw/ changes. - Paolo] Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: remove QEMU_AUDIO_* and -audio-help supportPaolo Bonzini17-689/+59
These have been deprecated for a long time, and the introduction of -audio in 7.1.0 has cemented the new way of specifying an audio backend's parameters. However, there is still a need for simple configuration of the audio backend in the desktop case; therefore, if no audiodev is passed to audio_init(), go through a bunch of simple Audiodev* structures and pick the first that can be initialized successfully. The only QEMU_AUDIO_* option that is left in, waiting for a better idea, is QEMU_AUDIO_DRV=none which is used by qtest. Remove all the parsing code, including the concept of "can_be_default" audio drivers: now that audio_prio_list[] is only used in a single place, wav can be excluded directly in that function. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: simplify flow in audio_initPaolo Bonzini1-32/+32
Merge two ifs into one. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: commonize voice initializationPaolo Bonzini2-18/+10
Move some mostly irrelevant code out of audio_init. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: return Error ** from audio_state_by_namePaolo Bonzini3-4/+7
Remove duplicate error formatting code. Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: allow returning an error from the driver initPaolo Bonzini15-27/+48
An error is already printed by audio_driver_init, but we can make it more precise if the driver can return an Error *. Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-10-03audio: Require AudioState in AUD_add_captureMartin Kletzander1-4/+2
Since all callers require a valid audiodev this function can now safely abort in case of missing AudioState. Signed-off-by: Martin Kletzander <mkletzan@redhat.com> Message-ID: <c6e87e678e914df0f59da2145c2753cdb4a16f63.1650874791.git.mkletzan@redhat.com> Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-09-26audio: remove shadowed localsPaolo Bonzini1-6/+5
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
2023-09-21audio/jackaudio: Avoid dynamic stack allocation in qjack_process()Peter Maydell1-5/+11
Avoid a dynamic stack allocation in qjack_process(). Since this function is a JACK process callback, we are not permitted to malloc() here, so we allocate a working buffer in qjack_client_init() instead. The codebase has very few VLAs, and if we can get rid of them all we can make the compiler error on new additions. This is a defensive measure against security bugs where an on-stack dynamic allocation isn't correctly size-checked (e.g. CVE-2021-3527). Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Francisco Iglesias <frasse.iglesias@gmail.com> Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Message-id: 20230818155846.1651287-3-peter.maydell@linaro.org
2023-09-21audio/jackaudio: Avoid dynamic stack allocation in qjack_client_initPeter Maydell1-2/+3
Avoid a dynamic stack allocation in qjack_client_init(), by using a g_autofree heap allocation instead. (We stick with allocate + snprintf() because the JACK API requires the name to be no more than its maximum size, so g_strdup_printf() would require an extra truncation step.) The codebase has very few VLAs, and if we can get rid of them all we can make the compiler error on new additions. This is a defensive measure against security bugs where an on-stack dynamic allocation isn't correctly size-checked (e.g. CVE-2021-3527). Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Francisco Iglesias <frasse.iglesias@gmail.com> Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Message-id: 20230818155846.1651287-2-peter.maydell@linaro.org
2023-09-08audio: spelling fixesMichael Tokarev1-1/+1
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
2023-07-17audio/pw: improve channel position codeMarc-André Lureau1-49/+26
Follow PulseAudio backend comment and code, and only implement the channels QEMU actually supports at this point, and add the same comment about limits and future mappings. Simplify a bit the code. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-13-marcandre.lureau@redhat.com>
2023-07-17audio/pw: remove wrong commentMarc-André Lureau1-1/+0
The stream is actually created connected. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-12-marcandre.lureau@redhat.com>
2023-07-17audio/pw: simplify error reporting in stream creationMarc-André Lureau1-11/+1
create_stream() now reports on all error paths. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-11-marcandre.lureau@redhat.com>
2023-07-17audio/pw: add more error reportingMarc-André Lureau1-1/+6
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-10-marcandre.lureau@redhat.com>
2023-07-17audio/pw: factorize some common codeMarc-André Lureau1-51/+34
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-9-marcandre.lureau@redhat.com>
2023-07-17audio/pw: add more details on errorMarc-André Lureau1-3/+4
PipeWire uses errno to report error details. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-8-marcandre.lureau@redhat.com>
2023-07-17audio/pw: trace during init before calling pipewire APIMarc-André Lureau1-2/+3
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-7-marcandre.lureau@redhat.com>
2023-07-17audio/pw: needless check for NULLMarc-André Lureau1-6/+2
g_clear_pointer() already checks for NULL. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-6-marcandre.lureau@redhat.com>
2023-07-17audio/pw: drop needless case statementMarc-André Lureau1-10/+0
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-5-marcandre.lureau@redhat.com>
2023-07-17audio/pw: Pipewire->PipeWire case fix for user-visible textMarc-André Lureau2-6/+6
"PipeWire" is the correct case. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230506163735.3481387-4-marcandre.lureau@redhat.com>
2023-07-01audio: dbus requires pixmanMarc-André Lureau1-1/+1
Commit commit 6cc5a615 ("ui/dbus: win32 support") has broken audio/dbus compilation when pixman is not included. Fixes: https://gitlab.com/qemu-project/qemu/-/issues/1739 Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230630214156.2181558-1-marcandre.lureau@redhat.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
2023-06-27ui/dbus: win32 supportMarc-André Lureau1-8/+36
D-Bus doesn't support fd-passing on Windows (AF_UNIX doesn't have SCM_RIGHTS yet, but there are other means to share objects. I have proposed various solutions upstream, but none seem fitting enough atm). To make the "-display dbus" work on Windows, implement an alternative D-Bus interface where all the 'h' (FDs) arguments are replaced with 'ay' (WSASocketW data), and sockets are passed to the other end via WSADuplicateSocket(). Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230606115658.677673-6-marcandre.lureau@redhat.com>
2023-06-27ui/dbus: compile without gio/gunixfdlist.hMarc-André Lureau1-0/+7
D-Bus on windows doesn't support fd-passing. Let's isolate the fdlist-related code as a first step, before adding Windows support, using another mechanism. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230606115658.677673-4-marcandre.lureau@redhat.com>
2023-06-20meson: Replace softmmu_ss -> system_ssPhilippe Mathieu-Daudé1-4/+4
We use the user_ss[] array to hold the user emulation sources, and the softmmu_ss[] array to hold the system emulation ones. Hold the latter in the 'system_ss[]' array for parity with user emulation. Mechanical change doing: $ sed -i -e s/softmmu_ss/system_ss/g $(git grep -l softmmu_ss) Signed-off-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Message-Id: <20230613133347.82210-10-philmd@linaro.org> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
2023-06-02cutils: Adjust signature of parse_uint[_full]Eric Blake1-2/+2
It's already confusing that we have two very similar functions for wrapping the parse of a 64-bit unsigned value, differing mainly on whether they permit leading '-'. Adjust the signature of parse_uint() and parse_uint_full() to be like all of qemu_strto*(): put the result parameter last, use the same types (uint64_t and unsigned long long have the same width, but are not always the same type), and mark endptr const (this latter change only affects the rare caller of parse_uint). Adjust all callers in the tree. While at it, note that since cutils.c already includes: QEMU_BUILD_BUG_ON(sizeof(int64_t) != sizeof(long long)); we are guaranteed that the result of parse_uint* cannot exceed UINT64_MAX (or the build would have failed), so we can drop pre-existing dead comparisons in opts-visitor.c that were never false. Reviewed-by: Hanna Czenczek <hreitz@redhat.com> Message-Id: <20230522190441.64278-8-eblake@redhat.com> [eblake: Drop dead code spotted by Markus] Signed-off-by: Eric Blake <eblake@redhat.com>
2023-05-05audio/pwaudio.c: Add Pipewire audio backend for QEMUDorinda Bassey5-0/+931
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 Lureau2-3/+6
Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
2023-03-06audio: remove sw->ratioVolker Rümelin3-24/+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 Rümelin2-6/+34
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 Rümelin3-21/+50
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 Rümelin1-9/+9
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 Rümelin1-24/+35
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 Rümelin3-25/+58
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 Rümelin1-8/+0
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 Rümelin1-11/+18
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 Rümelin1-23/+22
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 Rümelin1-4/+0
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 Rümelin1-27/+36
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>
generated by cgit v1.1 at 2024-09-27 15:33:37 +0000