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.
This allows us to specify a static session ID, so that (Dante) receivers
can uniquely identify us as a sender. This prevents duplicate streams in
Dante Controller, for example.
Move the address:port parsing code to the net helper. Add a default
address option.
Pass the interface address to protocol-simple and use this as the
default address for listening.
This makes sure that when the user passes tcp:3400 that we don't end up
publishing 0.0.0.0:3400 but the actual address of the interface we are
listening on so that the snapcast discover can use this to notify the
snapcast server.
Fixes#4093
This provides access to GNU C library-style endian and byteswap functions.
Windows doesn't provide pre-processor defines for endianness, but
all current Windows architectures (X32, X64, ARM) are little-endian.
When spa-plugins is enabled, the gio-2.0 global dependency is
overwritten.
When bluez support is enabled, OR when gsettings is enabled, the gio-2.0
dependency is then detected as found. This means that
pipewire-module-protocol-pulse can end up enabling gsettings support
even if it has been forcibly turned off.
Rename the meson variables to ensure they are looked up separately.
32 bits are enough, and additionally this also fixes an incorrect
format string, which caused the default `audio.rate` to be
incorrectly set on some platforms, such as 32-bit arm ones.
Fixes#4080
Can be used to group ports together. Mostly because they are all from
the same stream and split into multiple ports by audioconvert/adapter.
Also useful for the alsa sequence to group client ports together.
Also interesting when pw-filter would be able to handle streams in the
future to find out what ports belong to what streams.
