The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.
On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
Add support for generation of 10/12/14/16 bit bayer test pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
filter the error message and don't forward it as there might be a
following candidate decoder that can be used.
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
store the latency message and handle it after decoder is accepted.
This is to avoid the selection lock failure if decodebin3 needs to
handle latency message for candidate decoders when sending sticky event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
Send sticky events to the new created decoder after it switches
to PAUSED state. It it fails, just skip this decoder and try the
next one until finding one that works. Otherwise remove this
failing stream after trying all decoders and no one can work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
Generating the source element is done when urisourcebin is doing the READY to
PAUSED state change, so it is reasonable to set the new source element to that
state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Finally it makes more sense to have an element in READY when attempting to query
information from it (such as SCHEDULING queries or probing live-ness).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3856>
self->eos was never reset after streamsynchronizer has sent EOS
(except on explicit flush or switching back to PAUSED).
As a result, synchronization was broken if new streams were pushed later
as gst_stream_synchronizer_wait() does not wait if self->eos is set.
Fix this by reseting self->eos on STREAM_START as that means a new
stream is being sent upstream and so a new EOS will follow later on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4749>
While decodebin3 could handle changes in inputs (ex: changing codecs), there was
still one limitation which was when changing between sources which had
non-intersecting stream types (ex: switching from a video-only source to a
audio-only source). While the decoder *could* change to the proper codec ... it
would carry on using a `DecodebinOutputStream` associated to that stream
type (and therefore with pads with the wrong name).
In order to handle this:
* We notify the `MultiQueueSlot` of the change in `GstStreamType` if it already
had an associated inputstream (ex: the one associated with the static sink
pad)
* We detect such changes on the output of multiqueue as soon as
possible (i.e. when we get the GST_EVENT_STREAM_START for the new stream type)
by discarding the associated output.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1669
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4703>
When reconfigure_output_stream entry missing decoder path,
requested_selection should been update with what is really
active/selected immdiately with SELECTION_LOCK hold. So
use an optional message return from reconfigure_output_stream
and post it after release SELECTION_LOCK. This can make sure
other thread call to check_slot_reconfiguration will got
a correct requested_selection.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4599>
decodebin3 will do its best to figure out whether a parsebin is required to
process the incoming stream.
The problem is that for push-based stream it could happen that the stream would
not provide any caps, resulting in nothing being linked internally.
Furthermore, there is the possibility that a stream *with* caps would not be
using a TIME segment, which is required for multiqueue to properly work.
In order to fix those two issues, we force the usage of parsebin on push-based
streams:
* When the pad is linked, if upstream can't provide any caps
* When we get a non-TIME segment
Fixes#2521
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4492>
Fix compiler warnings about not using the return value when
freeing the GString segment with g_string_free(.., FALSE):
ignoring return value of ‘g_string_free_and_steal’ declared with attribute ‘warn_unused_result’
which we get with newer GLib versions. These were all harmless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4273>
This is a follow-up of the previous commit that enabled support for redirection.
The problem is that the urisourcebin that emitted the error redirection never
produced any pads, and therefore was never linked to decodebin3. This resulted
in the code waiting for that (output) item to finally switch over ... which will
never happen.
The fix is done by removing it early if it was never connected to decodebin3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4252>
The goal of parsebin is to figure out which elements to link together in order
to provide elementary streams given any random input.
The problem is that deciding whether a given stream should still have more
elements plugged in or not was dependent on ... the presence of compatible
decoders (sic).
Instead of that, if we can't plug anymore elements on a given stream *and* it is
detected as being an elementary stream, expose it.
Fixes#2118
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4231>
If sticky events are present on parsebin source pads, we propagate them to the
multiqueue source pads. Those will be propagated on the new urisourcebin source
pads like in the other code paths.
This ensures that STREAM_START event are present on new source pads. If CAPS
event are also present (not guaranteed), they will also be available.
Fixes#2384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4203>
Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3857>
The goal of the "global" group-id is to fix new inputs that do not come from the
same "source" as others. In order to ensure all "current" streams have the same
group-id we distribute the first valid group-id to all streams.
This commit fixes two issues with that:
* When inputs are unlinked they weren't always properly resetted (it would only
work if parsebin is used, which is no longer the default in
uridecodebin3/playbin3).
* When computing the global group-id, take into account unset
group-id (i.e. GST_GROUP_ID_INVALID).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1698
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3712>
The number of expected pads was:
* Defaulting to 1
* Or being overriden by GST_MESSAGE_STREAMS_SELECTED
This fails if upstream isn't a selectable source and has multiple streams, and
would therefore cause failures with multi-stream gapless playback
Fixes#1672
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
It is quite possible to have the blocking probe called from different streaming
threads when all expected pads are present.
* Notify all waiters by using g_cond_broadcast instead of g_cond_signal
* Properly remove the probe after waiting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3658>
Using the "GstBin" flags to check if an adaptive demuxer is streams-aware isn't
a good idea since it prevents using elements which aren't bins.
Instead we see if a collection was posted by the demuxer by the time a pad is
added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3601>
gst-launch-1.0 audiotestsrc ! udpsink host=127.0.0.1
gst-launch-1.0 udpsrc ! audioconvert ! autoaudiosink
would crash with a floating point exception when clipping the input
buffer owing to a division by zero because no caps event was received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3469>
Make sure that group-id of a given play item are made consistent from the
start (sources) and all the way through the output.
This ensures that we can reliably detect that we have switched to the next play
item on the output of decodebin3 (and we can therefore properly free/release it)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When shutting down, we want to remove the urisourcebin blocking probes ... but
we also want to propagate a GST_FLOW_FLUSHING upstream (and not
GST_FLOW_NOT_LINKED) to make the upstream task gracefully stop instead of
posting an error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When `is_selection_done` is called, it checks that all the requested streams are
present in the active stream list ...
... except there could very well be a (about to be removed) stream from the
previous selection present.
Therefore filter the list of streams we add to the message by the streams which
are actually requested.
Fixes issues when switching between different stream types (ex: video-only to
audio-only).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.
In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
This was the intention from the start, just took me a few years *cough* to
actually implement it properly.
Gapless is handled by re-using as much as possible the same decoders and sinks
if present, and only pre-rolling switching at the sources level (with buffering
if/when needed).
In order to enable "gapless" playback, the "next" uri should be set at any time
between the moment the `about-to-finish` signal is emitted and the moment the
current play item is done. Previously this could only be done with the signal
emission.
This new implementation also allows "Instantaneous URI switching". This allows a
much faster way of switching playback entries while re-using as many elements as
possible. To enable this set `instant-uri` property to TRUE, the default being
FALSE.
API: instant-uri properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
