Failed default audio devices were removed from the list but not freed,
and that made LeakSanitizer sad. Free default audio devices as they are
consumed.
Signed-off-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-ID: <20231120112804.9736-1-akihiko.odaki@daynix.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Print a debug message as is done for other unsupported audio formats
to give the user the chance to understand their mistake.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Daniel P. Berrangé <berrange@redhat.com>
We can have more than one audio backend.
void audio_init_audiodevs(void)
{
AudiodevListEntry *e;
QSIMPLEQ_FOREACH(e, &audiodevs, next) {
audio_init(e->dev, &error_fatal);
}
}
Reviewed-by: Stefan Berger <stefanb@linux.ibm.com>
Signed-off-by: Juan Quintela <quintela@redhat.com>
Message-ID: <20231020090731.28701-12-quintela@redhat.com>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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
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
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
