The parser does not check that POD arrays have the correct size for
their type, so the calling code must do that.
This also enumerates some of the code that cannot handle the size of the
values of an array not being the exact expected size for its type.
There is a lot of it.
Direct timestamp mode was incorrectly using over/underrun detection logic
and fill level tracking logic that is actually meant for the other mode
(referred to from now on as "constant latency mode"). Over/underruns are
tracked implicitly in the direct timestamp mode, and the absolute fill
level is not relevant in that mode, since the latency is not needed to
be constant then.
Also improve log lines and the RTP module documentation to define these
buffer modes clearly and explain their differences and use cases.
Opus and MIDI code get TODOs added, since their direct timestamp mode
implementations still may be incorrect. Fixing those will be done in
a separate commit.
When a stream has some delay, a time t1 + delay has to be read in time
t1 to play it when expected.
Decrease target_buffer by delay to start playback sooner, so sound
is played at correct time when delay is applied.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
We also need to close the SynObj fd we got, just like we close any
DmaBuf or MemFd.
Make sure we get a compiler error when we add more items to the
data type enumeration later.
Fixes#4807
It uses the onnxruntime library to parse the onnx file and construct a
neural network. It uses the label field to setup the plugin and how to
map the various tensors of the model to input, output, control and
notify ports.
Add an example config for how to use the silero VAD ONNX model with the
noise gate.
The return value is always 0, and the `impl` parameter
is not used, so ues the return value to return the boolean
result instead of an out parameter, and get rid of the
unused argument.
Instead of using a new macro with the `PW_` prefix, simply define
`SCHED_RESET_ON_FORK` to be `0` when it is not defined; as the
prefixed variant can be a bit confusing.
Reset buffers when deactivating to avoid having old data in the
ringbuffers, which also adds latency when activated again.
Clear sink_ready and capture_ready when resetting buffers to avoid
calling process() before there is new data to process.
capture and sink streams may start before playback stream so process()
may fail to dequeue a playback buffer. In that case advance the read
pointers to avoid building up latency in the ringbuffers.
The filter detects unnatural gaps (consisting of 0.0 values) and will
ramp-down or ramp-up the volume when entering/leaving those gaps.
This makes it filter out the pops and clicks you typically get when
pausing and resuming a stream.
See #4745
Now that the stream remembers the latency for us, we can only care
about the other latency.
So, if we get (output) latency on an input port/stream, we add our
own latency and then set it on the output port/stream. We do the
same for input ports.
Even if the latency didn't change, the current pw-stream
implementation will have wiped all Latency params away and we want
to put them back in all cases.
See #4731
