-audio is used like "-audio pa,model=sb16". It is almost as simple as
-soundhw, but it reuses the -audiodev parsing machinery and attaches an
audiodev to the newly-created device. The main 'feature' is that
it knows about adding the codec device for model=intel-hda, and adding
the audiodev to the codec device.
In the future, it could be extended to support default models or
builtin devices, just like -nic, or even a default backend. For now,
keep it simple.
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
g_new(T, n) is neater than g_malloc(sizeof(T) * n). It's also safer,
for two reasons. One, it catches multiplication overflowing size_t.
Two, it returns T * rather than void *, which lets the compiler catch
more type errors.
This commit only touches allocations with size arguments of the form
sizeof(T).
Patch created mechanically with:
$ spatch --in-place --sp-file scripts/coccinelle/use-g_new-etc.cocci \
--macro-file scripts/cocci-macro-file.h FILES...
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Reviewed-by: Cédric Le Goater <clg@kaod.org>
Reviewed-by: Alex Bennée <alex.bennee@linaro.org>
Acked-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Message-Id: <20220315144156.1595462-4-armbru@redhat.com>
Reviewed-by: Pavel Dovgalyuk <Pavel.Dovgalyuk@ispras.ru>
Commit ff095e5231 "audio: api for mixeng code free backends"
introduced another FIFO for the audio subsystem with exactly the
same size as the mixing-engine FIFO. Most audio backends use
this generic FIFO. The generic FIFO used together with the
mixing-engine FIFO doubles the audio FIFO size, because that's
just two independent FIFOs connected together in series.
For audio playback this nearly doubles the playback latency.
This patch restores the effective mixing-engine playback buffer
size to a pre v4.2.0 size by only accepting the amount of
samples for the mixing-engine queue which the downstream queue
accepts.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
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>
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>
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>
The audio migration vmstate is empty, and always has been; we can't
just remove it though because an old qemu might send it us.
Changes with -audiodev now mean it's sometimes created when it didn't
used to be, and can confuse migration to old qemu.
Change it so that vmstate_audio is never sent; if it's received it
should still be accepted, and old qemu's shouldn't be too upset if it's
missing.
Signed-off-by: Dr. David Alan Gilbert <dgilbert@redhat.com>
Reviewed-by: Daniel P. Berrangé <berrange@redhat.com>
Tested-by: Daniel P. Berrangé <berrange@redhat.com>
Message-Id: <20210809170956.78536-1-dgilbert@redhat.com>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Move the code to generate the pa_context_new() application name
argument to a function in audio/audio.c. The new function
audio_application_name() will also be used in the jackaudio
backend.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20210517194604.2545-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Currently there is a crackling noise with SDL2 audio playback.
Commit bcf19777df: "audio/sdlaudio: Allow audio playback with
SDL2" already mentioned the crackling noise.
Add an out.buffer-count option to give users a chance to select
sane settings for glitch free audio playback. The idea was taken
from the coreaudio backend.
The in.buffer-count option will be used with one of the next
patches.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de
Message-Id: <20210110100239.27588-3-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Handle the spice special case in audio_init instead.
With the qemu_spice_audio_init() symbol dependency being
gone we can build spiceaudio as module.
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Message-id: 20200916084117.21828-2-kraxel@redhat.com
The function audio_generic_write should work exactly like
audio_pcm_hw_run_out. It's a very similar function working on a
different buffer.
This patch significantly reduces the number of drop-outs with
the DirectSound backend. To hear the difference start qemu with
-audiodev dsound,id=audio0,out.mixing-engine=off and play a
song in the guest with and without this patch.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-6-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch removes unnecessary calls to the pcm_ops function
put_buffer_in(). No audio backend needs this call if the
returned length of pcm_ops function get_buffer_in() is zero.
For the DirectSound backend this prevents a call to
dsound_unlock_in() without a preceding call to dsound_lock_in().
While Windows doesn't complain it seems wrong anyway.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-5-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The playback rate with the spiceaudio backend is currently too
fast if there's no spice client connected or the spice client
can't play audio. Rate limit the audio playback stream in all
cases. To calculate the rate correctly the limiter has to know
the maximum buffer size.
Fixes: 8c198ff065 ("spiceaudio: port to the new audio backend api")
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch allows the audio backends get_buffer_out() functions
to drop audio data and mitigates a bug reported on the qemu-devel
mailing list.
https://lists.nongnu.org/archive/html/qemu-devel/2020-09/msg03832.html
The new rules for the variables buf and size returned by
get_buffer_out() are:
size == 0: Downstream playback buffer is full. Retry later.
size > 0, buf != NULL: Copy size bytes to buf for playback.
size > 0, buf == NULL: Drop size bytes.
The audio playback rate with spiceaudio for the no audio case is
too fast, but that's what we had before commit fb35c2cec5
"audio/dsound: fix invalid parameters error". The complete fix
comes with the next patch.
Reported-by: Qi Zhou <atmgnd@outlook.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200920171729.15861-2-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Commit 571a8c522e caused the HMP wavcapture command to segfault when
processing audio data in audio_pcm_sw_write(), where a NULL
sw->hw->pcm_ops is dereferenced. This fix checks that the pointer is
valid before dereferincing it. A similar fix is also made in the
parallel function audio_pcm_sw_read().
Fixes: 571a8c522e (audio: split ctl_* functions into enable_* and
volume_*)
Signed-off-by: Bruce Rogers <brogers@suse.com>
Reviewed-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Message-id: 20200521172931.121903-1-brogers@suse.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This patch prevents an underflow of variable samples in function
audio_pcm_hw_run_in(). See commit 599eac4e5a "audio:
audio_generic_get_buffer_in should honor *size". This time the
while loop in audio_pcm_hw_run_in() will terminate nevertheless,
because it seems the recording stream in Windows is always rate
limited.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 20200405075017.9901-3-vr_qemu@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This adds proper support for float samples in mixeng by adding a new
audio format for it.
Limitations: only native endianness is supported. None of the virtual
sound cards support float samples (it looks like most of them only
support 8 and 16 bit, only hda supports 32 bit), it is only used for the
audio backends (i.e. host side).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The function generic_get_buffer_in currently ignores the *size
parameter and may return a buffer larger than *size.
As a result the variable samples in function
audio_pcm_hw_run_in may underflow. The while loop then most
likely will never termiate.
Buglink: http://bugs.debian.org/948658
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-9-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
The combined generic buffer management code and buffer run out
code in function audio_generic_put_buffer_out has a problematic
behaviour. A few hundred milliseconds after playback starts the
mixing buffer and the generic buffer are nearly full and the
following pattern can be seen.
On first call of audio_pcm_hw_run_out the buffer run code in
audio_generic_put_buffer_out writes some data to the audio
hardware but the generic buffer will fill faster and is full
when audio_pcm_hw_run_out returns. This is because emulated
audio devices can produce playback data at a higher rate than
the audio backend hardware consumes this data.
On next call of audio_pcm_hw_run_out the buffer run code in
audio_generic_put_buffer_out writes some data to the audio
hardware but no audio data is transferred to the generic buffer
because the buffer is already full.
Then the pattern repeats. For the emulated audio device this
looks like the audio timer period has doubled.
This patch splits the combined generic buffer management code
and buffer run out code and calls the buffer run out code after
buffer management code to break this pattern.
The bug report is for the wav audio backend. But the problem is
not limited to this backend. All audio backends which use the
audio_generic_put_buffer_out function show this problem.
Buglink: https://bugs.launchpad.net/qemu/+bug/1858488
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With audiodev parameter out.mixing-engine=off hw->mix_buf is
NULL. This leads to a segmentation fault in
AUD_get_buffer_size_out. This patch reverts a small part of
dc88e38fa7 "audio: unify input and output mixeng buffer
management".
To reproduce the problem start qemu with
-soundhw adlib -audiodev pa,id=audio0,out.mixing-engine=off
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-4-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
It seems the function audio_generic_read started as a copy of
function audio_generic_write and some necessary changes were
forgotten. Fix the mixed up source and destination pointers and
rename misnamed variables.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20200123074943.6699-2-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
When building with GCC9 using CFLAG -Wimplicit-fallthrough=2 we get:
audio/audio.c: In function ‘audio_pcm_init_info’:
audio/audio.c:306:14: error: this statement may fall through [-Werror=implicit-fallthrough=]
306 | sign = 1;
| ~~~~~^~~
audio/audio.c:307:5: note: here
307 | case AUDIO_FORMAT_U8:
| ^~~~
cc1: all warnings being treated as errors
Similarly to e46349414, add the missing fall through comment to
hint GCC.
Fixes: 2b9cce8c8c
Reviewed-by: Richard Henderson <richard.henderson@linaro.org>
Signed-off-by: Philippe Mathieu-Daudé <philmd@redhat.com>
Reviewed-by: Aleksandar Markovic <amarkovic@wavecomp.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Message-Id: <20191218192526.13845-2-philmd@redhat.com>
Signed-off-by: Paolo Bonzini <pbonzini@redhat.com>
Tell the compiler to do a 32bit * 32bit -> 64bit multiplication
because period_ticks is a 64bit variable. The overflow occurs
for audio timer periods larger than 4294967us.
Fixes: be1092afa0 "audio: fix audio timer rate conversion bug"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-id: 8893a235-66a8-8fbe-7d95-862e29da90b1@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
With current code audio recording with all audio backends
except PulseAudio and DirectSound is broken. The generic audio
recording buffer management forgot to update the current read
position after a read.
Fixes: ff095e5231 "audio: api for mixeng code free backends"
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
Message-id: 2fc947cf-7b42-de68-3f11-cbcf1c096be9@t-online.de
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
