Make `client_queue_subscribe_event()` take the facility and type
separately, and calculate the mask itself, so that the caller
does not need to be concerned with that.
Make a function that can initialize raw audio info from a dict and fill
in the defaults. We can use this in many of the modules when the audio
format is parsed.
Use the helper instead of duplicating the same code.
Also add some helpers to parse a json array of uint32_t
Move some functions to convert between type name and id.
This gets the next key and value from an object. This function is better
because it will skip key/value pairs that don't fit in the array to hold
the key.
The previous code patter would stop parsing the object as soon as a key
larger than the available space was found.
Add spa_json_begin_array/object to replace
spa_json_init+spa_json_begin_array/object
This function is better because it does not waste a useless spa_json
structure as an iterator. The relaxed versions also error out when the
container is mismatched because parsing a mismatched container is not
going to give any results anyway.
Intercept the Output Latency paran and parse it for later.
Use the computed latency as the ProcessLatency and expose this
as the ProcessLatency param and the updated Input latency.
Accept updates to ProcessLatency to modify the latency, which then also
updates the Input Latency param.
See #4270
Improve the Latency reporting, we always report Input and Output latency
pairs.
Keep ProcessLatency on the capture and playback streams. The capture
stream process latency is reported as input latency and the playback
process latency as output latency.
Setting ProcessLatency on the capture stream (Sink), for example, will
propagate the added latency upstream. This would then instruct players
to send the audio earlier to compensate for the delay.
See #4270
Instead of binding the socket to 0.0.0.0 with port 0, make some config
options source.ip and source.port to configure this. This makes it
possible to bind to other interfaces or to use a fixed port for the
data messages.
Fixes#4144
Our current AES67 sender setup requires that that PTP driver drive the
entire graph. This adds support for allowing the AES67 RTP sink to be
driven by an arbitrary driver, while still using the PTP driver for
sending data on the network.
When aes67.driver-group is specified a pw_filter is created with no
ports, node.always-process = true and node.group set to the
aes67.driver-group. When set to PTP, this gives us process callbacks at
the PTP rate which we use to get the current PTP time in the RTP sender
by interpolating the clock snapshots from the pw-filter.
Implementation ideas from Wim Taymans. Co-authored with Sanchayan Maity.
For a detailed reference, refer the following papers by Fons Adriaensen.
- Using a DLL to filter time
(https://kokkinizita.linuxaudio.org/papers/usingdll.pdf)
- Controlling adaptive resampling
(http://kokkinizita.linuxaudio.org/papers/adapt-resamp.pdf)
We allow a quantum of jitter in the write timestamp. The previous value
of 32 seems to be empirically determined, using the actual quantum
allows us to reason about this better.
When the combine-stream module is used as a source and the input streams
are overlapping, mix the samples instead of overwriting the previous
samples.
See #3710
Avoids need for additional configuration to allow disabling send (which
in turn was needed to avoid errors when a network interface is not
available on start).
In circumstances where the network interface is not ready yet,
creating a send socket will fail. This may be ok if we only
intend to use rtp-sap as a listener, therefore add an option for it.
Only set use the graph rate and duration when the ffado.sample-rate
and ffado.period-size properties are set to 0. Othersize use the
configure values.
Without this patch, it would just ignore the settings and always use the
graph rate.
Just disable the data socket when it errors out but stop the follower
when the setup socket is in error.
This makes shutdown work properly when the setup socket is stopped,
which is what actually happens eventually.
Unloading the module on stream errors is a bit too much because a
suspend can clear the stream error again (or the error might not be
fatal)
This can happen for example when negotiation fails on some stream ports
(wireplumber tries to link the midi ports to audio ports) and it's
better to not completely fail on that.
Fixes#4121
dlopen() does not set errno on failure, rather you're supposed to call
dlerror() to get the latest error. dlerror() return a string so
instead return -ENOENT from weakjack_load_by_path().
Depending on errno weakjack_load() could think it successfully loaded
the library, and later module-jack-tunnel would crash because it call
a NULL function pointer.
