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.
LOCK the Latency param we get from the peer so that we don't remove it
when we update our own port latency. Also don't remove our port latency
when we get an update from the peer.
This essentially keeps the update/clear of the upstream and downstrem
latencies separate and makes it easier to implement the latency
logic in the pw-stream.
When a filter receives a Latency event on a port, it can simply update
the other port latency and none of the peer latencies are removed.
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
libcamera says that cameras should default to manual focus mode. This
means that unless pipewire clients specifically change this control,
users with an autofocus-capable camera are left with an out-of-focus
image. This patch sets the autofocus mode to continuous and enables
auto-exposure (as the default for this is unspecified).
Testing with an imx708 on Raspberry Pi OS on a Raspberry Pi 4, before
this patch the image was generally out of focus in Firefox/webrtc, after
this patch autofocus works correctly.
When a linked node needs to be resynced we actually never clear the
flag or reset the dll. Move the code around so that it still does
the reset of the flag and dll without actually doing the resync in
the ringbuffer when it is a linked node.
When we do_prepare, always reset the dll. We already set the alsa_sync
field but that is only used by followers to resync in some cases.
When reseting the dll, we also reset the next_time and base_time values,
we however need to do this before calculating the error in update_time
when we are the driver in IRQ mode or else we get some crazy error
that distorts the rate estimation.
Because we do the processing of the graph in the playback process
function, only do graph reset and reconfigure from the playback state
change so that we don't have process and state change at the same time
and crash.
Make one function to update the Latencies on all streams. We need to do
this because if one of the process or latency params change, it
influences the latency params on all streams.
Interrupts are disabled in alsa_irq_wakeup_event -> playback_ready method
to not produce another wakeups when waiting for a new data. Interrupts are
enabled again when a new data arrives in a method spa_alsa_write.
In rare cases, when there is multiple streams providing data and one of
them is disconnected, a new data fails to be delivered and the spa_alsa_write
is not called. Not providing data produces underrun and alsa-pcm invokes
recovery process. Recovery process starts a new playback, but without interrupts
enabled is graph not triggered and new data are not delivered (to enable
interrupts). Recovery process keeps running in loop.
Now the interrupts are enabled again after the recovery and the starvation
should not occur.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
When stream is paused, internal delay buffers were cleared, but some
data could stay in stream output queue. Without a flush, these data where
played in front of a new data.
Patch was inspired by 64d6ff4184 fixing the
same issue in a filter-chain module.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
Combine stream selects the biggest latency from all output streams and sent
the latency upstream. To select the biggest latency, each stream needs to have
the sample rate and the quantum size set.
The combine stream recalculates the latency in the latency changed callback
or during data processing.
Stream sets the sample rate and the quantum size in a copy_position call
which is normally called during processing the output data or when state
changes to streaming.
Before this change, it wasn't guarantee the copy_position was called for
each stream already and latency in the combine stream was selected from
random stream.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
The hooks were previously used to unlock the loop but now that the
lock is handled inside the loop itself and we don't unlock before the
blocking read anymore, we should also not call the hooks.
The blocking invoke function is not meant to be called with any of the
loop context locks acquired in order to avoid a deadlock. Make this (and
other blocking risks) clear in the documentation.
See #4472
Make one buffersrc for each input and configure it to mono. Try to guess
the channel position from the port name or use the config option and
fall back to FC (MONO) if unspecified.
Make one buffersink for each output and place a format converter in
front of it. Configure the converter to produce 1 channel with a layout
guessed from the port name or from the config.
With this we can use channelsplit and amerge to create multichannel
streams for avfilter plugins.
When _probe() is called, take a ref to the newly created devices instead
if sinking the floating ref, since gst_clear_object() is called when
core is disconnected. Otherwise the devices will be freed before the
caller gets them.
Fixes the following assert in the caller:
g_object_is_floating: assertion 'G_IS_OBJECT (object)' failed
Or sometimes a segfault with the backtrace:
0 g_type_check_instance_is_fundamentally_a (type_instance=type_instance@entry=0x116c1b0, fundamental_type=fundamental_type@entry=80) at /usr/src/debug/glib-2.0/2.84.0/gobject/gtype.c:3918
1 0xb6d40cc6 in g_object_is_floating (_object=0x116c1b0) at /usr/src/debug/glib-2.0/2.84.0/gobject/gobject.c:3843
2 0xb6bc4c74 in gst_device_provider_get_devices (provider=0x109ba00) at /usr/src/debug/gstreamer1.0/1.24.12/gst/gstdeviceprovider.c:426
A signed value doesn't really make sense in this context, so let's keep
it unsigned so the semantics are clear. This does break the interface,
but should be okay since it's not in a release yet.
When the Bluetooth earphones claim to support AAC `MPEG-2 AAC LC` type only,
PipeWire encodes audio using MPEG-4 FDK-AAC profile which enables unsupported
Perceptual Noise Substitution (PNS) encoder feature.
Use AOT_MP2_AAC_LC pseudo-profile which is AOT_AAC_LC with PNS disabled.
We should prefer the value of the follower when fixating to the
PortConfig format.
To make this actually work we need to be able to check if the value is
within the configured ranges. Implement the check for all types by
simply comparing the memory. This should then work also for checking
arrays, such as channel positions.
config.h needs to be consistently included before any standard headers
if we ever want to set feature test macros (like _GNU_SOURCE or whatever)
inside. It can lead to hard-to-debug issues without that.
It can also be problematic just for our own HAVE_* that it may define
if it's not consistently made available before our own headers. Just
always include it first, before everything.
We already did this in many files, just not consistently.
