Merge tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu into staging

Audio patches for QEMU 8.0

Cleanups and improvements from Volker Rümelin.

# -----BEGIN PGP SIGNATURE-----
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Merge tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu into staging

Audio patches for QEMU 8.0

Cleanups and improvements from Volker Rümelin.

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# gpg: Signature made Mon 06 Mar 2023 06:50:47 GMT
# gpg: using RSA key 87A9BD933F87C606D276F62DDAE8E10975969CE5
# gpg: issuer "marcandre.lureau@redhat.com"
# gpg: Good signature from "Marc-André Lureau <marcandre.lureau@redhat.com>" [full]
# gpg: aka "Marc-André Lureau <marcandre.lureau@gmail.com>" [full]
# Primary key fingerprint: 87A9 BD93 3F87 C606 D276 F62D DAE8 E109 7596 9CE5

* tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu: (27 commits)
audio: remove sw->ratio
audio/audio_template: substitute sw->hw with hw
audio: handle leftover audio frame from upsampling
audio: make recording packet length calculation exact
audio: rename variables in audio_pcm_sw_read()
audio: replace the resampling loop in audio_pcm_sw_read()
audio: make playback packet length calculation exact
audio: remove unused noop_conv() function
audio: don't misuse audio_pcm_sw_write()
audio: rename variables in audio_pcm_sw_write()
audio: remove sw == NULL check
audio: replace the resampling loop in audio_pcm_sw_write()
audio: make the resampling code greedy
audio: change type and name of the resample buffer
audio: change type of mix_buf and conv_buf
alsaaudio: reintroduce default recording settings
alsaaudio: change default playback settings
audio: remove audio_calloc() function
audio/audio_template: use g_new0() to replace audio_calloc()
audio/audio_template: use g_malloc0() to replace audio_calloc()
...

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

show more ...

audio: make the resampling code greedy

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 i

audio: make the resampling code greedy

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>

show more ...

Merge tag 'kraxel-20221013-pull-request' of https://gitlab.com/kraxel/qemu into staging

pci: cleanup virtio ids.
audio: bugfixes and latency improvements.
misc fixes for hw/display and ui

# -----BE

Merge tag 'kraxel-20221013-pull-request' of https://gitlab.com/kraxel/qemu into staging

pci: cleanup virtio ids.
audio: bugfixes and latency improvements.
misc fixes for hw/display and ui

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# gpg: Signature made Thu 13 Oct 2022 02:51:55 EDT
# gpg: using RSA key A0328CFFB93A17A79901FE7D4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg: aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg: aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901 FE7D 4CB6 D8EE D3E8 7138

* tag 'kraxel-20221013-pull-request' of https://gitlab.com/kraxel/qemu: (26 commits)
audio: improve out.voices test
audio: fix in.voices test
gtk: Add show_menubar=on|off command line option.
qemu-edid: Restrict input parameter -d to avoid division by zero
ui/gtk: Fix the implicit mouse ungrabbing logic
pci-ids: document modern virtio-pci ids in pci.h too
pci-ids: drop list of modern virtio devices
pci-ids: drop PCI_DEVICE_ID_VIRTIO_PMEM
pci-ids: drop PCI_DEVICE_ID_VIRTIO_MEM
pci-ids: drop PCI_DEVICE_ID_VIRTIO_IOMMU
docs: add firmware feature flags
cirrus_vga: fix potential memory overflow
ui/gtk-egl: egl context needs to be unbound in the end of gd_egl_switch
ui/vnc-clipboard: fix integer underflow in vnc_client_cut_text_ext
audio: prevent an integer overflow in resampling code
audio: fix sw->buf size for audio recording
audio: refactor audio_get_avail()
audio: rename audio_sw_bytes_free()
audio: swap audio_rate_get_bytes() function parameters
spiceaudio: update comment
...

Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>

show more ...

audio: prevent an integer overflow in resampling code

There are corner cases where rate->opos can overflow. For
example, if QEMU is started with -audiodev pa,id=audio0,
out.frequency=11025 -device i

audio: prevent an integer overflow in resampling code

There are corner cases where rate->opos can overflow. For
example, if QEMU is started with -audiodev pa,id=audio0,
out.frequency=11025 -device ich9-intel-hda -device hda-duplex,
audiodev=audio0 and the guest plays audio with a sampling
frequency of 44100Hz, rate->opos will overflow after 27.05h
and the audio stream will be silent for a long time.

To prevent a rate->opos and also a rate->ipos overflow, both
are wrapped around after a short time. The wrap around point
rate->ipos >= 0x10001 is an arbitrarily selected value and can
be any small value, 0 and 1 included.

The comment that an ipos overflow will result in an infinite
loop has been removed, because in this case the resampling code
only generates no more output samples and the audio stream stalls.
However, there is no infinite loop.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20220923183640.8314-12-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

show more ...

Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190821-pull-request' into staging

audio: second batch of -audiodev support, adding support for multiple backends.

# gpg: Signature made Wed

Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190821-pull-request' into staging

audio: second batch of -audiodev support, adding support for multiple backends.

# gpg: Signature made Wed 21 Aug 2019 09:40:37 BST
# gpg: using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg: aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg: aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901 FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20190821-pull-request:
audio: fix memory leak reported by ASAN
audio: use size_t where makes sense
audio: remove read and write pcm_ops
paaudio: fix playback glitches
audio: do not run each backend in audio_run
audio: remove audio_MIN, audio_MAX
paaudio: properly disconnect streams in fini_*
paaudio: do not move stream when sink/source name is specified
audio: audiodev= parameters no longer optional when -audiodev present
paaudio: prepare for multiple audiodev
audio: add audiodev properties to frontends
audio: add audiodev property to vnc and wav_capture
audio: basic support for multi backend audio
audio: reduce glob_audio_state usage
audio: Add missing fall through comments

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

show more ...

audio: use size_t where makes sense

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: c5193e687fc6cc0f60cb3e90fe69ddf2027d0df1.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: G

audio: use size_t where makes sense

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: c5193e687fc6cc0f60cb3e90fe69ddf2027d0df1.1566168923.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

show more ...

Merge remote-tracking branch 'remotes/kraxel/tags/audio-20170717-pull-request' into staging

audio: fixes for 2.10

# gpg: Signature made Mon 17 Jul 2017 10:29:08 BST
# gpg: using RSA

Merge remote-tracking branch 'remotes/kraxel/tags/audio-20170717-pull-request' into staging

audio: fixes for 2.10

# gpg: Signature made Mon 17 Jul 2017 10:29:08 BST
# gpg: using RSA key 0x4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>"
# gpg: aka "Gerd Hoffmann <gerd@kraxel.org>"
# gpg: aka "Gerd Hoffmann (private) <kraxel@gmail.com>"
# Primary key fingerprint: A032 8CFF B93A 17A7 9901 FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20170717-pull-request:
audio/adlib: remove limitation of one adlib card
audio/fmopl: modify timer callback to give opaque and channel parameters in two arguments
audio: st_rate_flow exist a infinite loop

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

show more ...

audio: st_rate_flow exist a infinite loop

If a voice recording equipment is opened for a long time(several days)
in windows guest, rate->ipos will overflow and rate->opos will never
have a chance to

audio: st_rate_flow exist a infinite loop

If a voice recording equipment is opened for a long time(several days)
in windows guest, rate->ipos will overflow and rate->opos will never
have a chance to change. It will result to a infinite loop.

Signed-off-by: Peng Hao <peng.hao2@zte.com.cn>
Signed-off-by: Wang Yechao <wang.yechao255@zte.com.cn>
Message-id: 1500128061-20849-1-git-send-email-peng.hao2@zte.com.cn
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

show more ...

Make audio violate POSIX less

git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@5864 c046a42c-6fe2-441c-8c8c-71466251a162

rate converter fix (malc)


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@2038 c046a42c-6fe2-441c-8c8c-71466251a162

audio merge (malc)


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1601 c046a42c-6fe2-441c-8c8c-71466251a162

merged 15a_aqemu.patch audio patch (malc)


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@1584 c046a42c-6fe2-441c-8c8c-71466251a162