Encoders and some decoders have additional internal latency that needs
to be accounted for.
This mostly matters for AAC (~40ms), as the other BT codecs have much
lower delays (~5ms).
Airpods don't follow the specification and set multiple bits in AAC
object type, including the ELD bit, but actually want AAC-LC. So check
the AOT in the right order.
When calculating the adjusted max quantum based off of max_latency, the
first multiplication can overflow uint32_t, leading to the quantum being
wrongfully clamped down.
Signed-off-by: Martin Louazel <martin.louazel@streamunlimited.com>
When in A2DP sink role and remote end switches codec, BlueZ nowadays
appears sometimes emit first SetConfiguration (creating new transport),
and then ClearConfiguration (removing old transport).
Handle this case: emit profiles_changed event always when transports
come/go.
Redefine profiles_changed() to take bitmask of profiles whose connection
status changed, so we don't need to emit two remove+add events.
Commit 88bb0bd7cc ("alsa: Allow to augment ucm port properties") adds
icon properties for some common port names, so that GUIs can show a
relevant icon to help disambiguate devices. However, these still do not
show up in pavucontrol, because it shows icons based on mappings'
properties. Add the relevant property to mappings as well.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/839
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
While trying to figure out device subsets that have aren't internally
contain conflicting devices, we walk through all possible subsets and
check each set if it satisfies ConflictingDevices/SupportedDevices
listed in UCM configuration. For a better user experience, we want to
skip subsets that are fully included in another valid subset we will
also generate.
The iterate_device_subsets() function that achieves the former is
intentionally in iterative form to avoid a stack overflow, since it will
walk through 2^n sets. However, the iterate_maximal_device_subsets()
function that skips incomplete sets is in recursive form, as I had
assumed tail-call optimization would take care of the potential problem.
Convert iterate_maximal_device_subsets() to an iterative form, because
the recursion seems to trigger a segfault with more than 16 devices on
PulseAudio. It doesn't seem to happen on PipeWire, but it's better to
not leave it to luck.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/838
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Because the max plugin is marked as SUPPORTS_NULL, the input and output
pointers can be NULL. Handle these cases. Also reindent with tabs (not 7
spaces).
When we have no thread running the loop, we need to flush the queues
from the invoking thread. Make sure that when multiple threads attempt
this that we serialize the flushing because the flushing code is not
thread safe.
When the IR is 0 length, make sure we copy the DSP to the convolver
because we will use it to clear memory.
Pass the dsp to the head and tail convolvers functions because they
might be NULL and we need the dsp to clear memory.
Fixes#4433
We increment the refcount of the loop when we hand it out. We use this
refcount to do some load balancing. In some cases we also decrease the
refcount again when the release_loop() functions is called but we only do
this in some modules. The result is that in most cases the refcount
just keeps on going up and this is misleading.
Change the refcount to a last_used timestamp and prefer to use loops
that have not been used in a while.
Fixes#4436
Interpolate buffer level to current playback position, and change its
definition so it directly corresponds to the total buffer latency. This
is also a bit simpler.
Now that BlueZ supports delay reporting in A2DP sink role, implement
that.
Report value that gives the total latency between packet reception and
audio rendering.
Also make Latency parameter in media-source to be not just a dummy
value.
The alsa/acp code already supports getting a user-friendly monitor name
using the EDID-Like Data (ELD) information available from cards that follow
the Intel HDA specification.
This patch adds support for also parsing the SAD fields of the ELD, and
exposing the results as a "iec958.codecs.detected" property on the
corresponding node, which should make it possible to provide more
user-friendly configuration UIs and defaults.
The default value will take effect if the session manager does not set a
different value.
Brief example:
test@test:~/checkouts/pipewire$ pw-dump | grep -E "(iec958.codecs.detected|iec958.codecs)\":"
"iec958.codecs": "[\"PCM\",\"AC3\",\"EAC3\",\"TrueHD\"]",
"iec958.codecs.detected": "[\"PCM\",\"AC3\",\"EAC3\",\"TrueHD\"]",
<after powering on my receiver>
test@test:~/checkouts/pipewire$ pw-dump | grep -E "(iec958.codecs.detected|iec958.codecs)\":"
"iec958.codecs": "[\"PCM\",\"DTS\",\"AC3\",\"EAC3\",\"TrueHD\",\"DTS-HD\"]",
"iec958.codecs.detected": "[\"PCM\",\"DTS\",\"AC3\",\"EAC3\",\"TrueHD\",\"DTS-HD\"]",
Big thanks to Pauli Virtanen <pav@iki.fi>, who also wrote large paths of the
code for this patch.
When not using PTP as the driver, it is possible that packet receive and
the process() callback are out of sync, meaning that the target buffer
fill level might be off by upto one ptime's worth of samples
occasionally. This would make the DLL hunt for the target rate, and
cause a constantly varying delay.
Accounting for the delta between the packet receive time and the
process() time allows us to eliminate this jitter, resulting in much
more consistent rate matching.
Without this for continuous frame intervals, the default is set to 25
fps even if that is outside the min/max bounds (e.g. defined by the
peer).
Clip the default with the min/max bound to avoid this.
v4l2 frame intervals instead of frame rates, which is basically the
inverse. For the most part, this is already handled correctly and
numerator and denominator are swapped accordingly.
However, the minimim frame interval is the maximum frame rate so those
need to be swapped as well.
Without this, the minimum frame rate is larger than the maximum frame
rate for v4l2 devices that define a continuous frame interval.
Use the pipewire ticks again as the clock source.
This was disabled because the v4l2 sources created bad ticks for the
graph. Now that this is improved we can enable the ticks again.
This has the advantage that simple audio playback does not drift
anymore. The only remaining problem would be timestamp drift or
discontinuities, which we don't handle here yet.
Some clocks (v4l2) don't process exactly process buffers at the given
rate/duration so mark this in the clock flags.
We need to use the nsec field in the clock to derive ticks in pw-stream
in that case to get a good clock.
