error_setg() appends newline to the formatted message.
Fixes: cb94ff5f80 ("audio: propagate Error * out of audio_init")
Signed-off-by: Michael Tokarev <mjt@tls.msk.ru>
Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org>
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>
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>
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>
Michael Tokarev
Richard Henderson
Akihiko OdakiandPaolo Bonzini
Juan Quintela
Martin KletzanderandPaolo Bonzini
Dorinda BasseyandMarc-André Lureau
Volker RümelinandMarc-André Lureau