Some clients may send headers with trailing whitespace, which can upset
subsequent parsing of the data into numbers. This patch removes such trailing
whitespace before further processing the header data.
Some Airplay devices announce themselves as using the ALAC (Apple Lossless Audio
Codec) format, while pipewire only supports the PCM codec. A look at the
Pulseaudio RAOP reveals that ALAC is supported there, but the encoding looks
exactly like what pipewire does for PCM. This patch adds support for ALAC, but
it uses the existing PCM infrastructure to send the audio data.
Reducing the latency is just papering over the issue in #2702. The
real fix is to limit the blocksize to the fragsize like what is done
in 00a234daf2
Reducing the latency then also causes regressions like #2715 so don't
do that anymore.
Fixes#2715
Instantiate the graph nodes when the samplerate is known instead of
using a fixed samplerate of 48KHz.
Warn when the convolver samplerate does not match the graph rate. We
might want to resample the IR later.
Use a reference to the location in the node where the handle of the
plugin can be found. That way we can change the handle only in the
node and have it changed everywhere else.
Add a process_playback function and use _trigger in the process_capture
to start processing. This ensure that the requested size is updated
before calling the process function.
Add methods activate() that is called before first call to run() when
stream starts and deactivate() that is called after last call to run()
when stream stops. This makes it possible for aec-plugins to reset their
state between streams.
When a port name contains a ':' we will try to split it and use the part
before the colon as the node name, which will then fail.
If we can't find a node name after splitting, try again by assuming the
colon is part of the port name.
Fixes control port names such as "Ratio (1:n)" in #2685
Make the receiving state machine more pronounced by explicitly storing
the state in the client. Furthermore, always consume the message content
if there is one and not only if the content type is "application/octet-stream",
but do not try to do it at once - like previously, instead only as the
socket becomes readable. The body is currently dropped, but it could
easily be collected in e.g. a `pw_array` should the need ever arise.
See #2673
Previously, the state used to receive messages from the remote
end was not reset when the client connected, which could
lead to issues if the same client is reused for multiple
connections.
Previously, the content had to be a null-terminated byte
sequence because the sending function used `strlen()` to
determine its length. However, `rtsp_do_auth_setup()` needs
to send a non-textual byte sequence, and it only worked so
far because it did not happen to have any zero bytes in it.
Add a "content_length" parameter and change the type of
"content" to facilitate sending arbitrary byte sequences.
The commit cited below mistakenly removed the set_rlimit call from under
`if (impl->use_rtkit)`, saying it doesn't have an rtkit implementation.
However, this function does call rtkit, so it has to be called in the
rtkit flow, otherwise pipewire fails to set the realtime priority,
printing the following error message:
mod.rt: RTKit error: org.freedesktop.DBus.Error.AccessDenied
mod.rt: could not make thread #### realtime using RTKit: Permission denied
Fixes: 5ae1c03d77 ("module-rt: small fixes")
When a client writes more then requested, let the requested field go
negative so that it is taken into account the next time we ask for more
data.
Also the requested field follows the difference in the write pointer
caused by seeks.
See #2626Fixes#2674
A client can sometimes send more data than we requested. PulseAudio
keeps the extra data around, it just asks for more data when it consumed
some of it.
PipeWire however always tries to keep tlength worth of data, as
specified in the PulseAudio docs... Keep track of how much extra data
has been sent and keep this around as well. Make sure we flush this
extra data as well.
Fixes#2626
Move the latency fraction calculation to fix_ functions so that the
new latency rate can be used when creating the streams.
Actually set the requested record attributes on the stream instead
of modifying the defaults.
See #2671
3f6fe392 uses headers from /usr/include/lv2/atom/. but this leads compilation
failure for some distros (i.e. ubuntu 18.04) as they have some
different location for these headers. One can find these headers
at /usr/include/lv2/lv2plug.in/ns/ext/atom/ (for ubuntu 18.04)
instead /usr/include/lv2/atom/. So guard them with __has_include.
and mention other possibilities.
Fixes#2670
