Some sound cards are only adapted for Android/macOS and other
systems, without considering Linux. The hardware-reported dB
volume is incorrect (while the percentage volume is normal).
Add support for the ignore-dB option to simplify compatibility.
For example, the 3206:0798 HP SIMGOT GEW1 Sound Card reports:
numid=4,iface=MIXER,name='PCM Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=100,step=0
: values=100,100
| dBminmax-min=0.00dB,max=0.39dB
This dB value does not match actual audio perception, and the
vendor attributed this issue to non-target system compatibility.
We keep a mapping between the sndfile formats and the format we would
like to decode them to for encoded formats. Make sure we don't mix up
the sample widths between them.
Make sure we don't send encoded formats as raw.
Debug the uncompressed format name correctly.
Fixes#5155
Sink/Source pairs should not have the same link-group otherwise the
session manager will not be able to autoconnect them with a loopback or
some other internally linked stream.
Setting the current clock time when resending buffers is often wrong.
Especially for pseudo-live sources - the default mode - it discards
the original buffer time, which again is used by the base-class to
adjust the timestamps further, ultimately resulting in very wrong
timestamps.
Instead, try to calculate the delta between when we originally got the
buffer and now.
(cherry picked from commit efd1526423)
Buffer timestamps get adjusted by the base class, GstBaseSrc, even if we
take an additional ref. Arguably the base class should check if buffers
are writable (gst_buffer_make_writable()), which would trigger a buffer
copy. That is currently not the case, though, thus do so on our side.
Notes:
1. Usually a buffer copy doesn't copy the underlying memory, i.e.
copying is cheap.
2. The copy holds a reference to the copied buffer, preventing the
buffer from getting recycled as before.
(cherry picked from commit 49300d8ee0)
JACK will never return NULL from jack_port_by_id() because the id
and the port_t are the same for JACK.
In PipeWire however we use the serial number as the id and so it can
be removed and become invalid. In this case, return a dummy port
from the client that can be used for some of the basic operations
you can do on a port_t, like get the name etc.
Also make sure that port_name() doesn't return NULL in case we use the
dummy port (which has the client set to NULL).
Fixes#3512
Fix path comparison in is_socket_unix() and don't unset LISTEN_FDS since
the function that uses it is called more than once and it was not unset
when sd_listen_fds() was used.
Fixes#5140
Currently it is possible for the request completion handler (`impl::requestComplete`)
to observe `impl::source.fd` while it is being modified in `impl::stop()`.
Fix that by closing the eventfd after the camera has been stopped.
Fixes: 3e28f3e859 ("spa: libcamera: source: rework startup sequence")
(cherry picked from commit 848ac24490)
Roc-toolkit log records are captured via a callback and
written to PipeWire log with corresponding verbosity level.
The log.level config parameter limits record verbosity at
the roc-toolkit level.
Move the card->ignored check to only apply to ACTION_CHANGE, not ACTION_REMOVE. This ensures that ignored cards can still be properly removed when they are unplugged.
Patch by Lairton Lelis da Fonseca Junior (@lairton)
Remove the hard skip for IPv4 link-local addresses and add an interface
binding (matching the existing IPv6 link-local behavior).
The host needs a link-local address on the interface (ip addr add
169.254.x.x/16 dev wlan0 or via NetworkManager +ipv4.addresses).
Fixes#4830
With keepalive enabled, we need to emit state change event on acquire
similarly as we do if refcount was already positive.
Co-authored-by: Martin Geier <martin.geier@streamunlimited.com>
The stream should be streaming before the get_time call is meaningful.
Various places in the code already check this an fall back to a default
value, we just need to return an error here.
When setting up the mix matrix, don't just iterate over the first 64
(CHANNEL_BITS) positions because then we will never be able to configure
more than 64 channels in the matrix.
Instead iterate until we have filled all src and dst entries in the
matrix. For the first 64 positions we might need to check the channel
mask to get the right positions in our source matrix.
Fixes the channel mixer for >64 channels where the positions above 64
where 0 in the matrix and muted.
Fixes#5118
Add a container option to override the extension check and force a
container when saving.
Add some more formats that are supported by libsndfile.
Add some options to list all supported formats, extensions/containers,
layouts and channel names.
Fixes#5117
Depending on the codec kind, select appropriate settings to pass
to select_config().
This allows to pass the bluez5.bap.force-target-latency property,
and so to select the same configuration.
Some PTS tests (e.g. BAP/UCL/SCC/BV-046-C or BAP/UCL/SCC/BV-077-C)
requests to select QoS from low-latency or high-reliabilty.
The bluez5.bap.force-target-latency device property allows to force it.
For other values than low-latency or high-reliabilty the QoS selection
will use both tables to find the more appropriate configuration.
A call to `release_buffer()` may happen in a gstreamer thread concurrently
with the pipewire stream emitting the `remove_buffer` event in the thread
loop, which, in pipewiresink calls `gst_pipewire_pool_remove_buffer()`, which
in turn modifies the `GstPipeWirePoolData` object. Thus a data race occurs
when accessing its members, which can lead to `pw_stream_return_buffer()`
being called with a null pointer.
Fix that by locking the thread loop before checking the conditions.
Fixes: c0a6a7ea32 ("gst: handle flush event in pipewiresink")
Only set rate_match rate to DLL correction when matching is active.
When ALSA is driver and not matching, set rate to 1.0 to indicate no rate adjustment needed.
DLL still runs for buffer level management but rate_match should not expose correction
when matching is inactive to avoid confusion during debugging.
Signed-off-by: Stanislav Ruzani <stanislav.ruzani@streamunlimited.com>
On some device combinations (MT7925 / Sony LinkBuds S) the low-latency
48 kHz QoS crackle.
It's probably better to default to high-reliability for those, until we
have proper quality vs. latency configuration in place.
Socket activation uses sd_listen_fds from libsystemd, and can only be
compiled on systems with systemd.
This is an issue for Alpine / postmarketOS, where upstream has no
systemd package, but downstream depends on upstream's pipewire package
and wants to rely on socket activation. This also prevents using
socket-activation on other non-systemd distributions, including
non-Linux.
Implement equivalent functionality without a dependency on libsystemd.
sizeof(adapter) is larger than the big_entry->adapter and so the code
would copy too much. Instead only copy the strlen of the parsed
adapter, which we checked above to be smaller than the available size.
This doesn't copy the 0 byte because the memory is assumed to be 0
filled already by the calloc.
If the address is exactly the HCI_DEV_NAME_LEN, it will result in a non-0
terminated string, which may or may not be a problem...
This can easily be overlooked if the RTP rate equals the clock rate, which
is fairly common (for example, RTP rate and clock rate both being 48 kHz).
And, if an ASRC is active, and converting the output of the RTP source
node, the resampler's delay need to be taken into the account as well.
Clear the ringbuffer in stream_stop() when processing stops to prevent old invalid packets
from being sent when processing resumes via rtp_audio_flush_packets().
This ensures a clean state when the stream restarts.
Clearing the ring buffer is important not only in the direct timestamp
mode, but also in the constant delay mode, since missed packets can lead
to gaps in the ring buffer. These gaps may have stale data inside if the
ringbuffer is not cleared after reading from it.
In corner cases where the read and write pointers are very close, it may
not be possible to read out all the wanted samples. This can for example
happen when there is a jump in the graph driver position. Currently, the
code reads the wanted number of samples out of the ring buffer regardless
of the write and read pointer positions. It does so even when the read
pointer is ahead of the write pointer (that is, an underrun occurs).
Fix this by checking the fill level and reading only the available amount
of samples if that amount is less than the wanted amount. The remaining
space in the target buffer is then filled with nullbytes.
This reverts commit dcdc19238b.
Reverting this because it caused big sync errors of ~62 ms in test setups.
Further discussions about this can be found here:
https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/2666
Followup commits modify the device delay application (by scaling it),
which is another reason why this needs to be reverted.
Previously the override was only present in `cc_flags`, meaning that
C++ source files, like `aec-webrtc.cpp`, would not have it.
Fixes: 6d74eee874 ("spa: bump channels to 128 again")
