The Volume property on org.bluez.MediaTransport1 is required to utilize
Absolute Volume, but it will only become availabe if the peer device
supports the feature. This happens asynchronously somewhere after the
transport itself has been acquired, after which the callbacks are
attached and software volume is reset.
To prevent race conditions availability of the property is also checked
on startup through a "Get" call.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
Write the current volume to the `Volume` DBus property to keep the
volume on the remote in sync. Without this the remote device shows the
wrong volume, and any attempts to change it will cause an unexpected
jump when the local volume has also been adjusted.
Thanks to prior investments to improve volume synchronization, setting
up callbacks and sending initial volume to the peer for HFP/HSP
implementing this feature is as easy as unconditionally assigning a
valid function to `set_source_volume`. `source_setup_volume_callback`
is already responsible for attaching a `SOURCE_VOLUME_CHANGED` hook and
sending initial (restored) volume to the peer (signifying support for
Absolute Volume - if not derived from the presence of FEATURE_CATEGORY_2
on the profile yet).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
Like the previous commit this handles `Volume` property changes but
applies them to an A2DP sink instead of source stream. As mentioned in
the AVRCP spec v1.6.2 §5.8 the rendering device (A2DP sink) is
responsible for performing volume attenuation meaning PulseAudio should
pass through audio as-is without performing any attenuation in SW.
Setting a valid pointer to `set_sink_volume` and returning `true` from
`should_attenuate_volume` attaches a hardware callback to `pa_sink` such
that no volume attenuation is performed anymore.
In addition to receiving volume change notifications it is also possible
to control remote volume by writing a new value to the DBus property.
This is especially useful when playing back to in-ear audio devices
which usually lack physical buttons to adjust the final volume on the
sink.
While software volume (used before this patch) is generally fine it is
annoying to crank it up all the way to 100% when a previous connection
to a different device left saved volume on the peer at a low volume.
Providing this bidirectional synchronization is most natural to users
who wish to use physical controls on their headphones, are used to this
from their smartphone, or aforementioned volume mismatches where both PA
as source and the peer as sink/rendering device are performing
attenutation.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
The A2DP spec mandates that the audio rendering device - the device
receiving audio, in our case a `pa_source` - is responsible for
performing attenuation:
AVRCP v1.6.2, §5.8:
The SetAbsoluteVolume command is used to set an absolute volume to be used by the rendering device.
BlueZ models this call as a change of the `Volume` property on the
`org.bluez.MediaTransport1` interface. Supporting Absolute Volume is
optional but BlueZ unconditionally reports feature category 2 in its
profile, mandating support. Hence remote devices (ie. a phone) playing
back audio to a machine running PulseAudio assume volume is to be
changed through SetAbsoluteVolume, without performing any local
attenuation.
Future changes will implement this feature the other way around: setting
an initial value for the `Volume` property as well as propagating
`pa_source` volume changes back to the peer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
In the case, where the latency is larger than the maximum block size,
module-null-sink will request multiples of the maximum block size from
the sink input instead of limiting the requested amount of data to the
the configured latency.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/554>
This reverts commit 96369919e5.
The commit was originally for the issue of Headphone can't output
sound, that was because the Headphone and Lineout share the 1st alsa
mixer and DAC, but this commit introduced a new issue of the speaker
is not muted after switching to headphone.
A recent merged kernel commit (f48652bbe3ae@linux) could fix the 1st
issue, so we could revert the fix of the 1st issue from PA, then the
2nd issue is fixed automatically.
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/553>
The variable card_name in sink_input_preferred_sink_changed_cb and
source_output_preferred_source_changed_cb could be used uninitialized,
which leads to invalid database entries.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/543>
The loopback message may be called after the sink input is already destroyed which causes
a crash. Also memory is leaked because the message object is not correctly freed.
This patch fixes the problems by adding a "dead" flag to the message structure and freeing
the message object on exit.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/541>
When fallback mapping is selected all subsequent profile sets containing
selected mapping are ignored. When there are only e.g. fallback input mappings
available, admitted profile set will only contain one profile with selected
first input fallback mapping and no outputs, and rest of profiles will only
contain outputs and no inputs. When there are only fallback input and output
mappings, there will be no profiles admitted at all.
Fix this by making sure that selected first fallback input or output mapping
is actually allowed to exist in all probed profile sets.
Note while this change allows selected fallback mappings to be found in duplex
configuraitons, probing fallbacks still can fail if there is more than one input
fallback and first one (selected) does not work in duplex configurations.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/539>
This patch fixes the following error:
$ pacat --file-format=ogg -r test.ogg
Failed to open audio file.
$ parecord sep.flac
Failed to open audio file.
libsndfile errors out if a WAV or OGG file is set to have anything but
SF_ENDIAN_FILE:
f4d1646e5c
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/526>
A few headsets have issues if HFP HF profile connection is attempted before
HSP HS profile connection is closed. Looks like this could happen because
bluez bluetoothd alows to make simultaneous HSP HS and HFP HF peer connections.
One of affected headsets is WH-1000XM2
Until we find out how to prevent simultaneous HSP HS and HFP HF connections,
when native backend has HFP HF profile enabled (this is the default) do disable
HSP HS completely unless user explicitly request it via discovery modarg.
Do this by adding module-bluetooth-discover arg enable_native_hsp_hs,
default to inverse of enable_native_hfp_hf.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/538>
For mSBC to work correctly the following must be set correctly
- codec object
- transport write method
- transport setsockopt method
Use helper method to set all three simultaneously.
Static configuration structure may be cleaner solution.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
HFP Audio Connection SCO configuration is negotiated symmetrically in both
directions, and USB HCI SCO packet framing is also symmetric in both directions.
This means that packet size will be the same for reads and writes over HFP SCO
socket.
HFP profile specification states that valid speech data shall exist on the
Synchronous Connection in both directions after the Audio Connection is
established.
This guarantees that an incoming packet will arrive shortly after SCO connection
is established. Use it's size to fix write MTU in case kernel value is wrong.
Discussion here https://lore.kernel.org/patchwork/patch/1303411/
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
The HFP protocol supports the ability to negotiate codecs if that is
supported by both AG and HF. This patch adds advertising of codec
negotiation support and the ability to negotiate a codec change. The
only currently supported extra codec (as of HF 1.7.1) is mSBC. mSBC
requires that the transmission be done over an eSCO link with
Transparent Data. The linux kernel ensures the former, but we have to
manually set the socket to transparent data.
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
Adding processing support for the mSBC codec is somewhat problematic,
because, although it is a SBC codec, the a2dp handling can't simply be
reused because the codec is used on an eSCO link with transparent
data, meaning the transmission unit has to be 48 bytes (fragmenting
the codec packets) and reassembly and boundary detection is required
to be done by the implementation. Therefore we have to implement
separate render and push routines for msbc that do this fragmentation.
Fragmentation is done by emulating circular buffers. The receive
(push) buffer is easy, since the mSBC packet size is 60, simply have a
buffer of this size in the sbc_info area where the fragments are
reassembled. Once we have a full 60 bytes, decode and restart from
zero. The send (render) buffer is more problematic, since the
transmit must be done from contiguous memory. This means that the
buffer must be the lowest common multiple of the transmission unit and
the packet size. This value is 240 since 240/48 == 5 and 240/60 == 4.
So the buffer pointers are reset at 240 which is a whole number of
both rendered packets and eSCO transmission units.
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
When an application sets a device for a newly created stream, we treat
that as a temporary setting, and don't save it as the preferred device
for future streams. The handling for this was broken, however: if the
stream already had a preferred device saved in the stream-restore
database, that was unset.
This was a regression introduced in
bc0e728320 and
70bbbcdc84. These commits tried to detect
in subscribe_callback() when the preferred device is cleared, but as a
side effect the preferred device started to get cleared from the
database also when a stream was created with a device set by the
application.
There's no way for subscribe_callback() to distinguish the different
cases of the preferred device being NULL. This problem is solved by
using the PREFERRED_SINK/SOURCE_CHANGED hooks. The hooks are only called
when the preferred device really changes.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1063
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/535>
The hooks are fired when the preferred device changes. This is useful
for module-stream-restore.
I added new set_preferred_sink/source() functions for firing the hooks.
The functions also log the preferred device changes.
There was already pa_sink_input_set_preferred_sink(), but that had a
side effect of moving the stream, so I needed a new function. Since it
can be confusing when the two similarly named functions should be
called, I added a comment for pa_sink_input_set_preferred_sink() that
explains the different situations.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/535>
Bluez prepends newly registered profile to a list of supported profiles,
and new peer profile connections are attempted in reverse order of profile
registration.
Currently native backend would register HFP AG profile before HSP AG profile.
When peer supports both HFP HF and HSP HS profiles, this registration order
causes extra HSP HS connection attempt before native backend would reject it
to make sure peer is reconnected with HFP HF profile.
Reorder HSP AG profile registration before HFP AG to make sure peer supporting
both profiles connects with HFP HF profile first.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/534>
