audio: add pcm_ops function table for capture backend

Add a pcm_ops function table for the capture backend. This avoids
additional code in the next patches to test if the pcm_ops table
is available.

audio: add pcm_ops function table for capture backend

Add a pcm_ops function table for the capture backend. This avoids
additional code in the next patches to test if the pcm_ops table
is available.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-8-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: copy playback stream in sequential order

Change the code to copy the playback stream in sequential order.
The advantage can be seen in the next patches where the stream
copy operation effecti

audio: copy playback stream in sequential order

Change the code to copy the playback stream in sequential order.
The advantage can be seen in the next patches where the stream
copy operation effectively becomes a write through operation.

The following diagram shows the average buffer fill level and
the stream copy sequence. ### represents a timer_period sized
chunk. The rest of the buffer sizes are not to scale.

With current code:
|--------| |#####111| |---#####|
sw->buf mix_buf backend buffer
1. clip
|--------| |---#####| |111##222|
sw->buf mix_buf backend buffer
2. write to audio device
333 -> |--------| |---#####| |---111##| -> 222
sw->buf mix_buf backend buffer
3a. sw device write
|-----333| |---#####| |---111##|
sw->buf mix_buf backend buffer
3b. resample and mix
|--------| |333#####| |---111##|
sw->buf mix_buf backend buffer

With this patch:
111 -> |--------| |---#####| |---#####|
sw->buf mix_buf backend buffer
1a: sw device write
|-----111| |---#####| |---#####|
sw->buf mix_buf backend buffer
1b. resample and mix
|--------| |111##222| |---#####|
sw->buf mix_buf backend buffer
2. clip
|--------| |---111##| |222##333|
sw->buf mix_buf backend buffer
3. write to audio device
|--------| |---111##| |---222##| -> 333
sw->buf mix_buf backend buffer

The effective total playback buffer size is reduced by
timer_period.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-7-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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jackaudio: use more jack audio buffers

The next patch reduces the effective qemu playback buffer size
by timer-period. Increase the number of jack audio buffers by
one to preserve the total effectiv

jackaudio: use more jack audio buffers

The next patch reduces the effective qemu playback buffer size
by timer-period. Increase the number of jack audio buffers by
one to preserve the total effective buffer size. The size of one
jack audio buffer is 512 samples. With audio defaults that's
512 samples / 44100 samples/s = 11.6 ms and only slightly larger
than the timer-period of 10 ms.

The larger jack audio buffer increases audio dropout safety,
because the high priority jack-audio worker threads can provide
audio data for a longer period of time as with a smaller buffer
and more audio data in the mixing engine buffer that they can't
access.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20220301191311.26695-6-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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paaudio: increase default latency to 46ms

This is a patch to improve the pulseaudio playback experience.
Asking pulseaudio for a playback latency of 15ms is quite
demanding. Increase this to 46ms. T

paaudio: increase default latency to 46ms

This is a patch to improve the pulseaudio playback experience.
Asking pulseaudio for a playback latency of 15ms is quite
demanding. Increase this to 46ms. The total playback latency
now is 31ms larger. One of the next patches will reduce the
total playback latency again by more than 46ms.

Here is a quote from the PulseAudio Latency Control
documentation: 'For the sake of (...) drop-out safety always
make sure to pick the highest latency possible that fulfills
your needs.'

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: inline function audio_pcm_sw_get_rpos_in()

Simplify code by inlining function audio_pcm_sw_get_rpos_in()
at the only call site and remove the duplicated audio_bug()
test.

Signed-off-by: Volk

audio: inline function audio_pcm_sw_get_rpos_in()

Simplify code by inlining function audio_pcm_sw_get_rpos_in()
at the only call site and remove the duplicated audio_bug()
test.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: add function audio_pcm_hw_conv_in()

Add a function audio_pcm_hw_conv_in() similar to the existing
counterpart function audio_pcm_hw_clip_out(). This function reduces
the number of calls to th

audio: add function audio_pcm_hw_conv_in()

Add a function audio_pcm_hw_conv_in() similar to the existing
counterpart function audio_pcm_hw_clip_out(). This function reduces
the number of calls to the pcm_ops functions get_buffer_in() and
put_buffer_in(). That's one less call to get_buffer_in() and
put_buffer_in() every time the conv_buffer wraps around.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: move function audio_pcm_hw_clip_out()

Move the function audio_pcm_hw_clip_out() into the correct
section 'Hard voice (playback)'.

Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed

audio: move function audio_pcm_hw_clip_out()

Move the function audio_pcm_hw_clip_out() into the correct
section 'Hard voice (playback)'.

Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220301191311.26695-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: replace open-coded buffer arithmetic

Replace open-coded buffer arithmetic with the new function
audio_ring_posb(). That's the position in backward direction
of a given point at a given distan

audio: replace open-coded buffer arithmetic

Replace open-coded buffer arithmetic with the new function
audio_ring_posb(). That's the position in backward direction
of a given point at a given distance.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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dsoundaudio: fix crackling audio recordings

Audio recordings with the DirectSound backend don't sound right.
A look a the Microsoft online documentation tells us why.

From the DirectSound Programmi

dsoundaudio: fix crackling audio recordings

Audio recordings with the DirectSound backend don't sound right.
A look a the Microsoft online documentation tells us why.

From the DirectSound Programming Guide, Capture Buffer Information:
'You can safely copy data from the buffer only up to the read
cursor.'

Change the code to read up to the read cursor instead of the
capture cursor.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20211226154017.6067-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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jackaudio: use ifdefs to hide unavailable functions

On Windows the jack_set_thread_creator() function and on MacOS the
pthread_setname_np() function with a thread pointer paramater is
not available.

jackaudio: use ifdefs to hide unavailable functions

On Windows the jack_set_thread_creator() function and on MacOS the
pthread_setname_np() function with a thread pointer paramater is
not available. Use #ifdefs to remove the jack_set_thread_creator()
function call and the qjack_thread_creator() function in both
cases.

The qjack_thread_creator() function just sets the name of the
created thread for debugging purposes and isn't really necessary.

From the jack_set_thread_creator() documentation:
(...)

No normal application/client should consider calling this. (...)

Resolves: https://gitlab.com/qemu-project/qemu/-/issues/785
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Christian Schoenebeck <qemu_oss@crudebyte.com>
Message-Id: <20211226154017.6067-1-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

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audio: add "dbus" audio backend

Add a new -audio backend that accepts D-Bus clients/listeners to handle
playback & recording, to be exported via the -display dbus.

Example usage:
-audiodev dbus,in.

audio: add "dbus" audio backend

Add a new -audio backend that accepts D-Bus clients/listeners to handle
playback & recording, to be exported via the -display dbus.

Example usage:
-audiodev dbus,in.mixing-engine=off,out.mixing-engine=off,id=dbus
-display dbus,audiodev=dbus

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Acked-by: Gerd Hoffmann <kraxel@redhat.com>

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configure, meson: move audio driver detection to Meson

This brings a change that makes audio drivers more similar to all
other modules. All drivers are built by default, while
--audio-drv-list only

configure, meson: move audio driver detection to Meson

This brings a change that makes audio drivers more similar to all
other modules. All drivers are built by default, while
--audio-drv-list only governs the default choice of the audio driver.

Meson options are added to disable the drivers, and the next patches
will fix the help messages and command line options, and especially
make the non-default drivers available via -audiodev.

Cc: Gerd Hoffman <kraxel@redhat.com>
Cc: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20211007130630.632028-4-pbonzini@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>

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audio: remove CONFIG_AUDIO_WIN_INT

Ever since winwaveaudio was removed in 2015, CONFIG_AUDIO_WIN_INT
is only set if dsound is in use, so use CONFIG_AUDIO_DSOUND directly.

Cc: Gerd Hoffman <kraxel@r

audio: remove CONFIG_AUDIO_WIN_INT

Ever since winwaveaudio was removed in 2015, CONFIG_AUDIO_WIN_INT
is only set if dsound is in use, so use CONFIG_AUDIO_DSOUND directly.

Cc: Gerd Hoffman <kraxel@redhat.com>
Cc: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20211007130630.632028-3-pbonzini@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>

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