We can produce data whenever the io area status != HAVE_DATA. We
don't need to look for NEED_DATA.
Also recycling buffer happens whenever the status != HAVE_DATA.
If the target node is set to 0, remove the autoconnect flag. This makes
the session manager disable stream autoconnect and some other program
needs to connect the stream to a sink or node.
Use the channelmap from the file, if available.
Add option to specify/override the channel map for playback.
Use the DSP media subtype to describe DSP formats. DSP formats
don't include the rate, channels and channel position in the
format and must use the rate and duration from the position io. This
makes it possible to later change the samplerate dynamically without
having to renegotiate the graph.
The same goes for the video DSP format, which uses the io_video_size
from the io_position to get the size/stride. Set this up in the node
based on the defaults from the context.
Make it possible to define defaults in the daemon config file, such
as samplerate, quantum, video size and framerate. This is stored in
the context and used for the DSP formats.
This is more in line with wayland and it allows us to create new
interfaces in modules without having to add anything to the type
enum. It also removes some lookups to map type_id to readable
name in debug.
Send suspend to the node when suspending. This is usually the same
as puse for all nodes.
Implement negotiation when we Start audioadapter. This makes it
easier that to track the ports that are negotiated for now.
Use Suspend to clear the audioadapter negotiation.
Add flags to the rate match io area
Add flag to activate/deactivate rate match
Set active flag in rate match when slaved
Update rate before starting resample
When we get worken up with a callback, mark ourselves as a master
because we then need to avoid running the converter again in the
process callback. After we perform the process callback, unmark
ourself as master and wait for the next cycle.
This fixes switch from master to slave for sources.
start with a completely filled resampler so that the first
input byte immediately gives an output sample. When then have
n_taps/2 leading (almost) 0 samples.
Also make the passthrough resampler act like the real resampler
by introducing an n_taps/2 delay.
Move some things around. Move the duration of the current cycle
to the clock. Also add the estimated next timeout to the clock.
Add a generic media specific counter to the clock.
Clean up the position_bar info. We can do with only a double beat
value and make the signature in floats.
Flesh out the io_position info. This has now the information needed
to convert a raw clock time into a stream time. It basically has
the same kind of features as GStreamer segments such as looping,
variable rate playback etc.. It also contains the state of the
timeline (paused/playing) and it can be used to update the position
and state from clients.
There is also extended information in the position field that
clients can update when they can.
Plugins basically only update the clock info they get (and use
the position info to check if they are slaved or not).
Before each cycle, check if there is a pending position update and
apply it.
clean_convert() removes the internally negotiated formats but
it does not set the format (or buffers) of the externally visible
ports. Therefore, don't clear the buffers_set flags.
Instead, clear the buffers_set flag when we explicitly reconfigure
the ports, when we also clear the format.
Also clear the port buffers when we set a NULL format.
Fixes#178
