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>
This API internally checks if a requested codec can be supported on the
system. This is especially required for codecs supported via GStreamer
where the availability of a plugin decides if the said codec can be
supported.
This will be used to prevent registration of a codec which the remote
endpoint device might be able to support, but, PulseAudio can't as the
codec is not available on the system due to the absence of a plugin.
We can also prevent listing or switching to an unavailable codec.
Note that the codec negotiation happens with the bluez stack even before
a device is connected. Because of this, we need to make sure that gst_init
is called before checking for the availability of a plugin. Since
module-bluez5-device gets loaded only after a connection to the device
has been established, doing the gst_init in that or one of the bluetooth
modules is not feasible.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
For example, using the following on the command line will return the
current codec for a bluetooth device
pacmd send-message /card/bluez_card.4C_BC_98_80_01_9B/bluez get-codec
where 4C_BC_98_80_01_9B is the bluetooth device.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This exposes the currently active codec on the source or sink via the
proplist and can be seen in output of pacmd list-sinks/list-sources.
Also set it on the card. In case of a bi-directional codec, the codec
for the sink and source could be different. For example, for aptX-LL,
the codec name on card, sink and source would be aptx-ll, aptx and sbc
respectively.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
For example, using the following on the command line will return the
list of possible codecs for a bluetooth device
pacmd send-message /card/bluez_card.4C_BC_98_80_01_9B/bluez list-codecs
where 4C_BC_98_80_01_9B is the bluetooth device.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This adds a generic gstreamer codec module based on which other
bluetooth codecs viz. aptX, aptX-HD, LDAC and AAC can be supported.
The GStreamer codec plugins used here themselves depend on the native
codec implementation.
aptX/aptX-HD -> libopenaptx
LDAC -> libldac
AAC -> Fraunhofer FDK AAC
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This uses the messaging API to initiate a codec switch.
While a particular codec might be applicable only for a particular
profile, for eg. aptX can only be applicable for A2DP sink or source
and not for let's say HSP, the codec switching logic has not been
tied to the logic for switching profiles.
Codec can be switched by running the following on the command line.
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec{"ldac_hq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"ldac_mq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"ldac_sq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"aptx_hd"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"aptx"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"sbc"}
Codec name passed above is matched against pa_a2dp_codec->name. Note that
the match is case sensitive. XX_XX_XX_XX_XX_XX needs to be substituted with
the actual bluetooth device id.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
Instead of letting a codec with higher index have higher priority,
just use a lower index for high priority. This allows the for loop
iterating over the codecs to be written in a straightforward manner
and not have to iterate from the end. FWIW Pipewire does the same.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
