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")
Add 35 sec timeout for PLAY_SAMPLE streams to start streaming, similar
to what we do with normal streams, and fail playback if they don't
start.
This avoids pending sample playback using up resources indefinitely if
streams fail to start for some reason, e.g. session manager is not
linking them.
If we get EPROTO, we likely have missed on some messages from the
server, and our state is now out of sync.
It's likely we can't recover (e.g. if error is due to fd limit hit), so
just drop the server connection in this case, similarly as if we got
EPIPE.
By setting the hci handle (e.g. hci0) of the desired adapter, the BIG
config will only applied on that adapter. In case no "adapter" entry in
the config is given, it will be applied on all adapters.
Signed-off-by: Alexander Sarmanow <a.sarmanow@televic.com>
Commit 2942bae034 introduced parsing of
"SupportedFeatures" which uses a third DBusMessageIter pointer.
*** stack smashing detected ***: terminated
==389050==
==389050== Process terminating with default action of signal 6 (SIGABRT)
==389050== at 0x4F57B2C: __pthread_kill_implementation (pthread_kill.c:44)
==389050== by 0x4F57B2C: __pthread_kill_internal (pthread_kill.c:78)
==389050== by 0x4F57B2C: pthread_kill@@GLIBC_2.34 (pthread_kill.c:89)
==389050== by 0x4EFE27D: raise (raise.c:26)
==389050== by 0x4EE18FE: abort (abort.c:79)
==389050== by 0x4EE27B5: __libc_message_impl.cold (libc_fatal.c:134)
==389050== by 0x4FEFC48: __fortify_fail (fortify_fail.c:24)
==389050== by 0x4FF0ED3: __stack_chk_fail (stack_chk_fail.c:24)
==389050== by 0xBC1D1A1: remote_endpoint_update_props (bluez5-dbus.c:3137)
==389050== by 0xB53609F: ???
==389050== by 0x1DF: ???
==389050== by 0x61C17BF: ??? (in /usr/lib/x86_64-linux-gnu/libdbus-1.so.3.32.4)
==389050== by 0x1DF: ???
==389050== by 0xC5ED113: ???
Replace magic numbers (0x02, 0x05) with named constants BT_CODEC_CVSD
and BT_CODEC_MSBC for better code readability. Also remove redundant
zero initialization of num_caps field since the buffer is already
memset to zero.
The sco_offload_btcodec() function now returns void and only skips
offload setup when using the default datapath, simplifying the logic
and removing the need for explicit feature flag checks.
Add support for configuring the SCO hardware offload data path for
HFP/HSP profiles using the Bluetooth SIG-specified procedure. This
enables vendor-specific SCO offload integrations.
Changes:
- Add `bluez5.hw-offload-datapath` configuration property (default: 0)
- Implement `sco_offload_btcodec()` to set BT_CODEC socket option
- Add `SPA_BT_FEATURE_HW_OFFLOAD` quirk feature flag
- Apply offload configuration when creating SCO sockets if quirk enabled
- Document new property in pipewire-props.7.md
The datapath ID is configurable via device parameters and only applied
when the hardware offload feature flag is set in quirks, allowing
platform-specific SCO offload implementations.
The simple formats contain some common mappings for other extensions such
as mp3.
Makes pw-record test.mp3 actually write an mp3 instead of a wav file.
Clamp the control values to their min/max.
Remove the default_control array, we can just restore the default with
the port_set_control_value(). Do this when no value has been set on the
port when we set up the graph. The avantage is that we calculate the
default, min and max correctly when they depend on the graph rate.
Fixes#5088
Since commit a1f33a99df changed buffer handling to create new GstBuffers
instead of reusing pool buffers, the video crop metadata was silently
lost. The code used gst_buffer_get_video_crop_meta() which returns NULL
on a fresh buffer, so the crop values from PipeWire were never applied.
Change to gst_buffer_add_video_crop_meta() to actually attach the
metadata to the buffer.
Also remove the now-obsolete call in gst_pipewire_pool_wrap_buffer.
This was discovered when using the XDG Desktop Portal's RemoteDesktop
interface: the full desktop was being delivered instead of just the
selected window, because the crop region metadata was not being
propagated to the GStreamer buffer.
Fixes: a1f33a99df ("gst: dequeue a shared buffer instead of original
pool buffer"), from merge request !1258
CC: @jameshilliard
When the graph has no inputs and the channels is set to 0, don't create
a capture stream. Likewise, don't create a playback stream when there
are no graph outputs and the output channels is 0.
You can use this to make a sine source or a null sink.
There is no reason to fail when there is no input or output port.
We can simply run the graph with what there is. Even if there is no
input or output at all, running one instance of the plugins is
possible.
Add a busy builtin plugin that has no ports and keeps the CPU IDLE or
busy for the give percent.
The filter graph has, after parsing, a default number of input and
output ports. This is based on the description or the first/last element
input and output ports. Pass this information in the properties when
we emit the info.
Don't use the number of configured input/output ports as the default
number of channels in filter-chain because this is only determined after
activating the graph. Instead, use the default input/output channels.
The result is that when you load filter-chain without any channel
layout, it will default to the number of input/outputs of the graph
instead of 0. This allows for the node to be visible in the pulseaudio
API.
Fixes#5084
Don't rely on the out_rate vs n_phases to decide on when to use the
inter resampler because the out_rate can change when we activate the
adaptive resampler.
Instead use a boolean that we can set at the start.
If we have in and out rates with a very small GCD, we might end up with
a lot of phases. Limit the number of phases to 1024 and switch to
interpolating mode. 1024 phases is enough to accurately interpolate
from.
Together with the MAX_TAPS limit we will never create a filter
size that overflows 32 bits.
Fixes#5073
The video-play-fixate example will downgrade the stream to MemFd one
modifier at a time. Sometimes it's useful to test with not downgrading;
to avoid having to depend on actual DRM devices (real or virtual), fake
them by using memfd and mapping them in the sink.
A DMA buffer from a DRM device are typically accessed using API related
to a DRM device, e.g. Vulkan or EGL. To create such a context for using
with a PipeWire stream that passed DRM device DMA buffers applications
have so far usually guessed or made use of the same context as the
stream content will be presented. This has mostly been the Wayland
EGL/Vulkan context, and while this has most of the time worked, it's
somewhat by accident, and for reliable operation, PipeWire must be aware
of what DRM device a DMA buffer should be accessed using.
To address this, introduce device ID negotation, allowing sources and
sinks to negotiate what DRM device is supported, and what formats and
modifiers are supported by them.
This will allow applications to stop relying on luck or the windowing
system to figure out how to access the DMA buffers. It also paves the
way for being able to use multiple GPUs for different video streams,
depending on what the sources and sinks support.
Our AVX optimizations are really AVX2 so rename the files and functions
and use the right HAVE_AVX2 and cpu flags to compile and select the
right functions.
Fixes#5072
Add heuristic to resync streams if controller packet completion times
for different streams differ by too much. This likely indicates
controller has lost sync between the streams, and we have to reset
playback.
There's no way to do this properly. The ISO-over-HCI transport is badly
specified in Bluetooth Core Specification. Many controllers have broken
implementation of the current send timestamp read command, so packets
have no identifiers which ISO interval they belong to.
Controllers try to reconstruct the right interval based on
manufacturer-specific heuristics probably based on packet arrival times.
Kernel + USB introduce timing jitter, and playback sometimes desyncs and
packet from some streams are persistently sent some multiple of the SDU
interval off from the intended timing.
Try to determine this from packet completion latencies. This is
somewhat manufacturer specific, tested on Intel & Realtek, hopefully
works on others too.
