Get the clock pointer using the io_changed stream event.
and update the clock before triggering the process
The clock needs to be updated in the data loop thread
and before triggering the process so move the calls to
`pw_stream_trigger_process` from gstreamer thread context
into the data loop thread context by invoking a callback
and update the clock inside the data loop callback
before the trigger
Sometimes you want to use the convolver as a delay without adding
latency so add a latency option to tweak the automatic latency
reporting. Use this in the upmix rear delay filters.
Collect the latency of the graph in filter-chain. We do this by first
inspecting the LATENCY ports on the plugins and us the notify value as
the latency on the node.
We then walk the graph from source to sink and for each node take the
max latency of all linked upstream peer nodes. We end up with the max
latency of the graph and emit this in the graph properties.
We then listen for the graph latency property and use that to update the
process_latency of the filter-chain, which will then update the latency
on the filter-chain ports.
Fixes#4678
Make a function to create a filter. This is a pod that has all valid
defaults fixated and the invalid ones left unfixated.
Use this filter is a first attempt to negotiate a link format. The
effect is that a format will be chosen first that matches all the valid
defaults as much as possible instead of negotiating to the first thing
that matches.
Suppose we have a higher priority port with the format:
foo/bar
key: { default:1024, min:1, max:2048 }
And another port with two params:
foo/bar
key: 512
rate: 2/1
foo/bar
key: 1024
rate: 30/1
By first trying key: 1024 we negotiate to the more specific second property
with the higher rate.
Reset the ERR counters when pressing the c key. This only makes the
current pw-top start counting from 0 again, it does not change anything
globally on the server. It's still usefull because you can use it to
let it count the number of new errors since the keypress.
Destroy the sources from the io handler immediately when there is an
error so that we don't end up in endless error wakeups.
Schedule the free from the main loop and make sure only one can ever
run.
The manager is actually not supposed to decide much about the number of
audio and midi ports. It should just suggest a default when connecting
driver doesn't know.
Add a audio.ports parameters to manager and driver to suggest/ask for
the amount of audio ports. Let the audio.position/audio.channels be a
specification of the channel mask in case it matches the requested
channels, otherwise use AUX channels for the ports.
This means that we must derive the mode (sink/source/audio/midi) from
the ports that are negotiated in the manager and the driver, so delay
this until after negotiation.
Make sure all the possible modes work. For midi only streams, we can't
wait for the session manager to perform a PortConfig so do that
ourselves. Make sure we only use a source trigger when we have a sink.
Fixes#4666
Keep a position info for the stream it was set and then use the position
info for the stream that is driving the graph. Otherwise we might use a
destroyed position info.
Make a new port_info structure that holds the link port information for
input and output. We can use this in some places and remove some
redundant code.
We can also pass a reference to this port_info as the work-queue
object, which makes it more natural to find the associated port info in
the various work queue callbacks.
Move the private pw_context_find_format to the link implementation,
where is it actually used.
Rework the format negotiation code to use an array of 2 port_info
structures with the two ports to negotiate. The negiotiation will
always use the first port_info as higher priority.
Make sure a driver port has a lower priority than the other port. We want
to negotiate the source/sink to something close that what is
provided/requested by the client. The client can always adapt to the
driver port format fields by giving a "don't care" value for the format
property (either unspecified or with an out of range default value).
In case of encoded video we get n_planes as 0 from the video info so
passing that as n_datas is failing during the buffer negotiation. Make
sure to use an appropriate value based on whether we have raw video or
not.
Co-authored-by: Taruntej Kanakamalla <taruntej@asymptotic.io>
Add support for FairPlay SAP v2.5 (encryption type 5) type devices such as Apple Home Pod Minis.
Apparently only these devices require the `POST /feedback` heartbeat, so fix that.
When PW source is used with something like Camera and the camera is
disconnected, all buffers are removed and stream will be paused.
When using PW sink with source, the sink side pipeline can go to EOS.
This again results in all the buffers being removed and stream being
paused on the source side. PW source side pipeline can also crash if
the sink was in the middle of frame copying a buffer to render which
got removed.
Handle this scenario by sending a flush-start event at the start of
buffer removal and flush-stop at the end followed by an end of stream
or pipeline error depending on user selection.
Initialize the byte array with bytes instead of a string because the 0
byte at the end of the string does not fit in the array and causes a
compiler warning.
For a pipeline like below, we might want to dynamically switch the audio
source.
gst-launch-1.0 -e pipewiresrc autoconnect=false ! queue ! audioconvert ! autoaudiosink
On switching to a different audio source, any one of driver, quantum
or clock rate might change which changes the return `result` value of
gst_pipewire_clock_get_internal_time.
This can result in the basesrc create function incorrectly waiting in
gst_clock_id_wait. We post clock lost message to fix this. In the case
of gst-launch, it will set the pipeline to PAUSED and then PLAYING to
to force a new clock and a new base_time distribution.
Without the clock lost message, the following can be seen
before re-linking to a different source
0:00:30.887602864 79499 0x7fffe8000d40 DEBUG GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:17.024565416 now 0:00:17.024109144 diff (time-now) 456272
after re-linking to a different source
0:00:45.790843245 79499 0x7fffe8000d40 DEBUG GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:31.927694059 now 0:00:17.066883864 diff (time-now) 14860810195
With the clock lost message, the following can be seen
before re-linking to a different source
0:01:09.336533552 89461 0x7fffe8000d40 DEBUG GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:00:58.198536772 now 0:00:58.197444926 diff (time-now) 1091846
after re-linking to a different source
0:01:21.659827958 89461 0x7fffe8000d40 DEBUG GST_CLOCK gstsystemclock.c:1158:gst_system_clock_id_wait_jitter_unlocked:<pipewireclock0> entry 0x7fffd803fad0 time 0:28:24.853517646 now 0:28:24.853527204 diff (time-now) -9558
Note the difference in `time` and `now` fields of the above log message.
This is easy to reproduce by using a pipewiresink as the audio source
with a pipeline like below, as one of the sources during switching.
gst-launch-1.0 -e audiotestsrc wave=ticks ! audioconvert ! audio/x-raw,format=F32LE,rate=48000,channels=1 !
pipewiresink stream-properties="props,media.class=Audio/Source,node.description=pwsink" client-name=pwsink
Applications need to handle the GST_MESSAGE_CLOCK_LOST message in their
bus handlers.
Let's make sure we own the memory in buffers, so that we can be
resilient to the PW link going away. This currently maintains the status
quo of copying data into the pipewirepool for sending to the remote end,
but moves the allocation of buffers so that ownership is maintained by
the sink in all cases.
There are some tricky corners, especially with bufferpool vs. buffers
param negotiation -- bufferpool parameters can be negotiated in
GStreamer before the link even comes up, so we try to adapt the buffers
param to use the negotiated value. For now, that is more brittle than
tying those two aspects together. We can revisit this if we can find a
way to tie pipeline state and link state more closely.
Co-authored-by: Arun Raghavan <arun@asymptotic.io>
A number of changes for correctness.
1) We expose the actualy min and max values we support in the
allocation query.
2) We don't support max_buffers as 0, as unlimited buffers is not an
option
3) In ParamBuffers, we request the max_buffers from bufferpool config,
as we cannot dynamically allocate buffers
We need to make sure the memory sizes are correctly initialised so the
meta makes sense, and we don't copy the meta from the input buffer as
that doesn't make sense given we have our own meta already.
