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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>
2023-03-06audio: make the resampling code greedyVolker Rümelin1-8/+13
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 Rümelin3-13/+16
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 Rümelin3-64/+67
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 Rümelin1-6/+4
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 Rümelin1-7/+4
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 Rümelin2-21/+0
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 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 Rümelin2-12/+10
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 Rümelin1-5/+1
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 Rümelin3-16/+2
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 Rümelin1-3/+0
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 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 Rümelin2-0/+14
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 Armbruster1-1/+1
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 Maydell2-0/+84
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 Armbruster2-0/+84
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. Berrangé3-1/+80
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. Berrangé1-0/+12
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 Armbruster7-28/+21
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>
generated by cgit v1.1 at 2024-12-26 05:05:41 +0000