When HFP HF support is enabled in native backend, peer HFP HF profile connection
is preferred over same peer HSP HS profile connection if peer supports both
profiles.
Enforce the preference by rejecting HSP HS profile connections from such peer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
Native backend implements HFP AG but not HFP HF yet, therefore headset=auto
functionality is still needed if HFP HF is required.
To make headset=auto work again, drop both HFP AG and HSP AG roles while
performing handover from native backend when oFono is detected running.
While at it, restore profile description to Headset Head Unit (HSP/HFP)
to note that HFP may be still provided via oFono backend.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
Change default backend from 'auto' to 'native' so that in the usual
install pulseaudio uses the native backend with HFP_HF handling.
set default to false unless the backend is the native one, in which
case the default becomes true.
Additionally set default value of enable_native_hfp_hf to false unless
the backend is the native one, in which case the default becomes
true. so that we only bind the HFP_HF end point in the native case
(leaving it free for ofono in the ofono backend or auto case)
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
HFP 1.6 requires a stateful negotiation of AT commands. The prior
version got away with initialising HFP simply by replying 'OK' to
every negotiation attempt. This one actually tries to parse the state
and make sure the negotiation occurs correctly
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
---
v4:
- Update for PA 11.0
- Finally sort out CIND negotiaton for complex headsets
v3:
- remove internal debugging
- added comment for t->config being not null for hfp
- removed unused returns from hfp_rfcomm_handle()
- remove rfcomm comment
- use pa_startswith
- simplify negotiation
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
When all headsets supported both HSP and HFP, life was good and we
only needed to implement HSP in the native backend. Unfortunately
some headsets have started supporting HFP only. Unfortuantely, we
can't simply switch to HFP only because that might break older HSP
only headsets meaning we need to support both HSP and HFP separately.
This patch separates them from a joint profile to being two separate
ones. The older one retains the headset_head_unit name, meaning any
saved parameters will still select this (keeping us backward
compatible). It also introduces a new headset_handsfree.
For headsets that support both HSP and HFP, the two profiles will
become separately visible and selectable. This will only matter once
we start adding features to HFP that HSP can't support (like wideband
audio).
Signed-off-by: <James.Bottomley@HansenPartnership.com>
---
v6:
- merge profile switching fixes patch from Rodrigo Araujo
v5:
- rename option to enable_native_hfp_hf
- don't call profile_done for HFP_HF unless it was initialised
v3:
- Update for PA 11.0
v2:
- fold in review feedback
- add global disable option for not registering HFP
v3:
- change parameter to enable_profile_hfp
- update device_supports_profile to be aware of hfp/hsp exclusivity
- change parameter to enable_profile_hfp_hf
bluetooth: separate HSP and HFP (to me merged with this patch)
Hi.
First, just to say that your patches are going great. Finally I can use
the microphone of my HFP only headset (a version of a Bluedio T2+).
So far, I've only encontered one problem: the auto_switch option of
module_bluetooth_policy stops working. Dug through the code and I think
you missed a few spots were you have to hangle the new headset_handsfree
profile in module_bluetooth_policy.c
Applying the following after applying your v5 patches fixed the issue
for me, now when I start making a VOIP call the profile switches to
headset_handsfree and the mic works automatically, and when the call
finishes it reverts back to a2dp.
Thanks and best regards.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
The PA_BLUETOOTH_PROFILE names should mirror the PA_BLUETOOTH_UUID
names using profile_function instead of randomly made up names. Fix
this with the transformation:
PA_BLUETOOTH_PROFILE_HEADSET_HEAD_UNIT -> PA_BLUETOOTH_PROFILE_HSP_HS
PA_BLUETOOTH_PROFILE_HEADSET_AUDIO_GATEWAY -> PA_BLUETOOTH_PROFILE_HFP_AG
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
---
v4: update for PA 11.0
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
When the source-output of a virtual source with volume sharing disabled is moved,
the source output volume is reset to 100%. This patch fixes the problem by
applying the virtual source volume to the source-output after the move.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/509>
The encoding and decoding pipeline are essentially identical: both push
data in via an appsrc, route it through a codec-specific (opaque)
element, and finally pull data out of an appsink. The code already makes
it impossible to have an encoding and decoding pipeline simultaneously
set up in `gst_info`, and converting `bool for_encoding` to a tri-state
(encode, decode, or both) would be messy; particularly when encoding and
decoding could possibly differ in format.
This change removes a swath of code and removes the possibility of
misusing `enc_` or `dec_` in the wrong place (ie. after copying a bit of
code and forgetting to rename one or two). When bidirectional codecs
come online a second codec instance (`gst_info`) can simply be created
and controlled independently.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/487>
LDAC encoder already supports S16, S24, S32 and F32LE. Using FLOAT32LE
for the sample format would avoid the additional call for conversion to
pa_sconv_s32le_from_float32ne. perf tool shows this as being the function
called frequently after encode. So, just avoid this by using sample format
as F32LE.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/486>
Make the code ever so slightly more generic by not using appsrc and
appsink in codec-specific logic when assigning caps specific to the raw
(PCM) format provided by or returned to PA.
Note that caps have to be set (= event) after starting, can't send
events in flushing state.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/484>
SBC codec decrements bitpool value by fixed amount each time it is asked to
reduce output bitrate. This results in reduced audio quality with SBC codec.
Implement increase_encoder_bitrate for SBC codec by adding 1 to bitpool value
each time encoder bitrate needs to be increased to restore SBC audio quality.
While at it, remove bitpool decrement limit to use connection agreed value
instead as we will be able to restore quality later.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/474>
Bluetooth thread may ask encoder to reduce bitrate if writing is not keeping up
with inputs or writing to bluetooth socket takes too much time.
Assuming conditions leading to reduced bitrate are intermittent, allow periodic
attempts to increase encoder bitrate, by default at most twice per second.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/481>
Similar to the situation/comment in `endpoint_release` BlueZ does not
request any reply to `ClearConfiguration()` either; sending one results
in the same "0 matched rules" warning from dbus-daemon:
dbus-daemon[1309]: [system] Rejected send message, 0 matched rules; type="method_return", sender=":1.71" (uid=1000 pid=87548 comm="../build/src/daemon/pulseaudio -vvvv -n -F ../buil") interface="(unset)" member="(unset)" error name="(unset)" requested_reply="0" destination=":1.3" (uid=0 pid=1308 comm="/usr/lib/bluetooth/bluetoothd -d ")
Solve this by only creating a return message when an (othwise empty)
reply is solicited for, just like in `endpoint_release`.
Unfortunately we also have to make sure to not send any error back if no
reply is requested, but fortunately an argument parsing error here is
extremely unlikely.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/472>
We move the codec specific bits to their own respective files and now
make the codec specific initialisation use a GstBin, which the generic
GStreamer module now uses in the pipeline.
It is job of the codec specific function to add elements in the GstBin
and link the added elements in the bin. It should also set up the ghost
pads as a GstBin has no pads of it's own and without which the bin
cannot be linked to the appsrc/appsink.
Also, we now only initialise either the encoding or the decoding
pipeline and not both. The codec init API already gets passed the
for_encoding flag. We pass and use the same to codec specific init
functions.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
As we now support codecs other than SBC, we might have codec which does
not have an encode or a decode capability. Specifically, in the case of
LDAC there isn't a known decoder implementation available. For such a
case, we should not register the corresponding endpoint.
In case of LDAC, as decoding cannot be supported, we should not register
a sink endpoint or vice versa in the other scenario.
To do this, we check if encode_buffer or decode_buffer entry for a codec
has been set in pa_a2dp_codec and accordingly prevent or allow it's
registration.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
When it comes to codecs provided via GStreamer, we register all codecs
if GStreamer option is enabled for bluez5 via meson. However, the
GStreamer plugin required for the codec might not be present on the
system. This results in the codec being available for registration with
the bluez stack or selection by the user, but, trying to use the said
codec then fails.
To prevent the above, we now use the can_be_supported codec API to check
if the codec is usable and if not, we do not register the said codec and
also prevent users from switching to it.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
