Add struct spa_error_location that holds information about some parsing
context such as the line and column number, error and line fragment
with the error.
Make spa_json_get_error() fill in the spa_error_location instead. Add
some error codes to the error state and use this to add a parsing reason
to the location.
Add a debug function to log the error location in a nice way. Also
add a FILE based debug context to log to any FILE.
Replace pw_properties_check_string() with
pw_properties_update_string_checked() and add
pw_properties_new_string_checked(). The check string behaviour can still
be done by setting props to NULL but the main purpose is to be able to
avoid parsing the json file twice in the future.
When using the old pw_properties_update_string(), log a warning to the
log when we fail to parse the complete string.
Use the new checked functions and the debug functions to report about
parsing errors in the tools and conf parsing.
This gives errors like:
```
> pw-loopback --playback-props '{ foo = [ f : g ] }'
error: syntax error in --playback-props: Invalid array separator
line: 1 | { foo = [ f : g ] }
col: 14 | ^
```
Check for JSON parse errors, and log error messages as appropriate.
It's mostly enough to do this where the input is parsed for the first
time, e.g. via pw_properties_new_string, as that already validates the
JSON syntax.
Call _activate() in the aec spa plugin before both capture and sink
streams are started and call _deactivate() after both capture and sink
streams are stopped, to make sure _run() is not called while the spa
plugin is inactive.
Make a new function to set the rtp stream in the error state.
When we fail to start the stream, set the error state. Otherwise (like
when we try to use an invalid interface name) the socket create will
fail but the stream will still try to send data to the invalid socket.
Rework some things so that we can actually completely close and reopen
the driver when suspending. Leaving the device opened but not streaming
causes trouble when ffado-mixer changes the settings in between.
Do this by first doing a probe to get the ports and streams and then
link those to the actual buffers when we start streaming.
Follow the graph rate and quantum so that we open with the right
settings, this makes pw-metadata work on the quantum and rate. It
requires a driver reopen so it might be slower.
Try to handle some errors better.
See #3558
As part of LibreSSL's cleanup of removed ENGINE support, <openssl/x509.h> is no longer included in <openssl/engine.h>. <openssl/evp.h> was transiently included in <openssl/x509.h>, compared to <openssl/rand.h> for OpenSSL.
Parse the audio rate from the caps and set this as the stream node.rate
property. This will try to switch the graph to this rate and avoids
resampling.
Rework some of the property handling.
Add pw_conf_section_update_props_rules() that will not only update the
properties of a section but wil also apply rules in section.rules and
match against the context properties.
Use this by default when using pw_context_conf_update_props().
Add a new method to get a string name of the VM type. Place the
cpu.vm.name in the context properties.
This makes it possible to deprecate the vm.overrides with something more
flexible based on rules. Update the conf files and docs to refect this.
For capture streams, place the time of the cycle when we received this
buffer in the new buffer time field.
Without this (and without timestamps in an optional header metadata)
it's impossible to determine when a buffer was created. With this field
and the pw_stream_get_time_n() and pw_stream_get_nsec() we can derive
the mapping to ticks and delay in the graph.
Always copy the timing info when capturing, not only when there is a
buffer to dequeue.
We can update the rate_queued and rate_size with defaults when we don't
have rate matching. We can then also directly use the rate_size field
to update the buffer requested size.
Add a new extension that can create a server on a user provided socket
with user provided security properties.
This is mainly used in flatpaks that want to create and bind a pipewire
socket with specific permissions for the flatpak app.
The flatpak will also provide an fd that will be closed when the server
can be removed.
Remove some of the hardcoded values (the absolute smallest buffer size,
independent on the sample rate) to a config option. Set it to the
default value of 4, like what it was before.
Change the hardcoded absolute lowest limit to 1 because anything else
does not make sense.
Enforce the quantum-floor when calculating the final graph quantum.
Fixes#3908
Add a monitor.passthrough option. This will pass all latency information
directly between the port and its monitor ports.
This is interesting when the adapter (and audioconvert) is used with a
null-audio-sink that simply forwards the data to a real sink/souce. In
that case, we want the sink/source latency to be passed unmodified.
Set the monitor.passthrough on the pulseaudio null-sink because
a passthrough virtual sink is the most likely use case for this.
Add some monitor.passthrough default config where it makes sense.
Fixes#3888
When the midi message does not fit in the current cycle, save it in a
buffer and send it in the next cycle.
Incomplete midi messages will be received in the next cycle and need to
be sent out at timestamp 0 of the cycle.
Detect dropped and incomplete messages and give a warning.
Try to close/open the ffado device when streaming_start() fails. This
seems to be the case after suspend/resume or a cable disconnect.
Unfortunately, ffado seems to be stuck in an unrecoverable state,
though, to make it really work...
Clean up port buffers when the filter is destroyed.
