The peer will wait some time and eventually time out the connection if
no reply is sent back. When sending `ERROR` the peer can decide to break
the RFCOMM connection immediately or continue when a command is not
critical.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
Devices for Apple's iOS uses a few extra HFP AT commands to
inform the iPhone about the headphone's battery status.
Apple documented the AT commands in the following document:
https://developer.apple.com/hardwaredrivers/BluetoothDesignGuidelines.pdf
The patch has been tested with a Bose QC35, which results
in the following communication:
D: [pulseaudio] backend-native.c: RFCOMM << AT+VGS=14
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+XAPL=009E-400C-0129,3
D: [pulseaudio] backend-native.c: RFCOMM >> +XAPL=iPhone,2
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+XEVENT=Bose SoundLink,158
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+IPHONEACCEV=2,1,4,2,0
N: [pulseaudio] backend-native.c: Battery Level: 50%
N: [pulseaudio] backend-native.c: Dock Status: undocked
D: [pulseaudio] backend-native.c: RFCOMM >> OK
[Marijn: Adapt for recent HSP/HFP code changes]
Co-authored-by: Marijn Suijten <marijns95@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
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>
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>
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>
Setting these callbacks adds the HW_{VOLUME,MUTE}_CTRL flag even when
PulseAudio is solely responsible for performing attenuation whilst only
keeping the peer posted on changes. For this case the hardware callback
is not registered at all but instead a hook is attached to catch
PA_CORE_HOOK_{SINK,SOURCE}_VOLUME_CHANGED. Only when the peer performs
attenuation (the peer is in HeadSet/HandsFree role) are the callbacks
used, without touching PA software volume at all. A future change could
potentially use software volume to compensate for the extremely coarse
16 steps of volume control in HSP and HFP, and to allow volume over
100%.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/519>
Originally written for A2DP this rework of that patch enables late-bound
hardware volume control on HFP and HSP. As per the specification the
headphones (where gain control for both speaker and microphone could
happen in hardware on the peer) are supposed to send initial values for
these before the SCO connection is created; these `AT+VG[MS]` commands
are also used to determine support for it. PA uses this information in
`add_{sink,source}` to attach hardware volume callbacks, _if_ it is
supported. Otherwise PA performs the attenuation in software.
Unfortunately headphones like the WH-1000XM3's connect to A2DP
initially and only send `AT+VGS` (microphone hardware gain is not
supported) _during_ SCO connection when the user switches to the HFP
profile afterwards; the callbacks set up dynamically in
`rfcomm_io_callback` are written after the sink and source have been
created (`add_{sink,source}`), leaving them without hardware volume
callbacks and with software volume when adjusted on the PA side. (The
headphones can still send volume updates resulting in abrupt changes if
software and peer volume differ. Furthermore the same attenuation is
applied twice - once in PA software, once on the peer).
To solve this problem we simply check whether the callbacks have been
attached whenever the peer sends a volume change, and if not attach the
callbacks to the sink/source and reset software volume.
Fixes: d510ddc7f ("bluetooth: Perform software attenuation until HF/HS reports gain control")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/528>
HF/HS hardware attenuation is optional on HFP: the peer indicates
support with the AT+BRSF command, when bit 4 is set. That does not
explicitly mandate speaker or microphone gain control; either is
dynamically detected as soon as `AT+VG[MS]=` is received. Otherwise
software attenuation is performed.
It is also optional on HSP but nothing is mentioned about feature
detection, assume it is the same as HFP: perform software attenuation
until the HF/HS peer sends an `AT+VG[MS]=` command.
When PA is a HS/HF (and the peer the AG) we attenuate both channels in
software and unconditionally keep the peer up to date with
`AT+VGM/AT+VGS` commands.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
Generalize the distinction between local and peer-attenuated volumes
into a function, paving the way for future changes where this needs to
be checked in more places and when A2DP Absolute Volume support is
added.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
Sink and source naming is more generic when dealing with audio that is
directional in the sense that it either goes to or comes from the other
device, but not necessarily a microphone or speaker. A concrete example
is the swapped meaning when the current device is in the HeadSet
profile. The incoming audio can come from any source, not necessarily a
microphone. Likewise, audio captured by the microphone of the headset is
not necessarily played back by a speaker on the AG, it is merely acting
as a sink for the data: further handling is irrelevant to the naming.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
For the upcoming A2DP AVRCP Absolute Volume feature the code in BlueZ5
has to be generic to be reusable. Move this conversion so that it
becomes possible to implement A2DP volume - which uses different values
- on top without duplicating existing callback functionality.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
