It turns out that downstream returning OK after EOS is a bug in
multiqueue. As we moved to queue, we no longer have this issue.
Let's keep the code clean and just assuming that downstream will
keep returning EOS and allow convergence of flow.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2247>
As the alphacombine is simplified to received matching pair of buffers,
we can't just stop streaming when we receive EOS from downstream. Due
to usage of queue, the moment we get this return value may differ.
Though, by continuing pushing, we override the last_flowret on the pad
which can make us miss that we effectively can combine all flow into
EOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2238>
This element will merge video buffers in order to use the alpha stream
luma plane as the alpha of the video stream. The implementation is zero-copy
and currently only support merging I420 stream with an I420, NV12 or GRAY8
alpha stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
The previous test was preventing the pad to be in EOS
when the segment position was greater than segment stop.
It ended up consuming all the data before getting in EOS.
Regarding GST_SEEK_FLAG_SEGMENT it seems to be
correctly handled later in the method.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2173>
The path caps describe the input caps that will select each path, don't
intersect those with the srcpad caps, which could be completely
different. Instead, when querying allowed caps for the srcpad, just
construct the union of all possible output caps from all path srcpads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2018>
When handling a caps query on the src pad, don't return the union
of input caps. Even when not active, a path element can be queried
for srcpad template caps, or for dropping paths the allowed downstream
caps is anything - as data will be dropped anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2018>
In a case of a scr different from 0, after a seek,
the src_segment.stop has been updated with the duration
not including the base_time (scr). The segment position
needs to be tested upon segment.stop + base_time (scr)
to check for an EOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2048>
In order to always insert a PCR packet right on time, we need to
check whether one is needed when outputting any packet, not only
a packet for the PCR stream. Most of the PCR packets will remain
data-carrying packets, but as a last resort we may insert stuffing
packets on the PCR stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
Change PCR / SI scheduling so that instead of checking if
the current PCR is larger than the next target time, instead
check if the PCR of the next packet would be too late, so PCR
and SI are always scheduled earlier than the target, not later.
There are still cases where PCR can be written too late though,
because we don't check before each output packet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
Especially specify the field-order in the interleaved mode. Otherwise it
might cause the negotiation to fail, because
GST_PAD_SET_ACCEPT_INTERSECT is not set on the sinkpad, and the
field-order is missing in the sink template but can be present in the
outside caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
The first time update_src_caps is called, there's no frame parsed yet,
therefore we don't know whether the file has alternate-field interlacing
mode. If we run it again after we have a frame, it might be that now we
have the SEI pic_struct parsed, and therefore we know that it's
field-based interlaced, and therefore the height must be multiplied by
two. Earlier on this was not detected as a change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2022>
Instead of relying on GstBaseParse default behaviour of computing
the duration of a parsed buffer based on the framerate passed
to gst_base_parse_set_framerate(), we instead compute the duration
ourselves, as we have more information available.
In particular, this means we now output buffers with a duration
that matches that of raw interlaced buffers when each field is
output in a separate buffer.
This fixes DTS interpolation performed by GstBaseParse, as the
previous behaviour of outputting each field with the duration of
a full frame was messing up the base class calculations.
When not enough information is available, h264parse simply falls
back to calculating the duration based on the framerate and hope
for the best as was the case previously.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1973>
1. Set the default output alignment to frame, rather than current
alignment of obu. This make it the same behaviour as h264/h265
parse, which default align to AU.
2. Set the default input alignment to byte. It can handle the "not
enough data" error while the OBU alignment can not. Also make it
conform to the comments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
The current behaviour for obu aligned output is not very precise.
Several OBUs will be output together within one gst buffer. We
should output each gst buffer just containing one OBU. This is
the same way as the h264/h265 parse do when NAL aligned.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
The current optimization when input align and out out align are
the same is not very correct. We simply copy the data from input
buffer to output buffer, but we failed to consider the dropping of
OBUs. When we need to drop some OBUs(such as filter out the OBUs
of some temporal ID), we can not do simple copy. So we need to
always copy the input OBUs into a cache.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
1. Add the mono_chrome to identify 4:0:0 chroma-format.
2. Correct the mapping between subsampling_x/y and chroma-format.
There is no 4:4:0 format definition in AV1. And 4:4:4 should
let both subsampling_x/y be equal to 0.
3. Send the chroma-format when the color space is not RGB.
Fixes: #1502
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1974>
This AV1 parse implements the conversion between alignment of obu,
tu and frame, and the conversion between stream-format of obu-stream
and annexb.
TODO:
1. May need a property of operating_point to filter the OBUs
2. May add a property to disable deep parse.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1614>
When carrying over existing GstStream to a new GstStreamCollection we need to
check whether they *actually* were being used in the previous collection.
This avoids adding unknown streams (metadata, PSI, etc...) to the collection on
updates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Otherwise, when rtpm2src cancels an inflight operation that has a queued
message stored, then the rtmp connection operation is not stopped.
If the cancellation occurs during rtmp connection start up, then
rtpm2src does not have any way of accessing the connection object as it
has not been returned yet. As a result, rtpm2src will cancel, the
connection will still be processing things and the
GMainContext/GMainLoop associated with the outstanding operation will be
destroyed. All outstanding operations and the rtmpconnection object will
therefore be leaked in this case.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1862>
Instead of waiting so that we can simply use a clocksync element as
filter, otherwise we won't know the pipeline is live as it won't
return NO_PREROLL as one would expect in that case.
Adding it right away shouldn't create any issue, both ways are fine.
A lot of content producers out there targetting "adaptive streaming" are riddled
with non-compliant PCR streams (essentially all the players out there just use
PTS/DTS and don't care about the PCR).
In order to gracefully cope with these, we detect them appropriately and any
small (< 15s) PCR resets get gracefully ignored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1785>
Currently for buffer splitting only output duration can be specified.
Allow specifying a buffer size in bytes for splitting.
Consider a use case of the below pipeline
appsrc ! rptL16pay ! capsfilter ! rtpbin ! udpsink
Maintaining MTU for RTP transfer is desirable but in a scenario
where the buffers being pushed to appsrc do not adhere to this,
an audiobuffersplit element placed between appsrc and rtpL16pay
with output buffer size specified considering the MTU can help
mitigate this.
While rtpL16pay already has a MTU setting, in case of where an
incoming buffer has a size close to MTU, for eg. with a MTU of
1280, a buffer of size 1276 bytes would be split into two buffers,
one of 1268 and other of 8 bytes considering RTP header size of
12 bytes. Putting audiobuffersplit between appsrc and rtpL16pay
can take care of this.
While buffer duration could still be used being able to specify
the size in bytes is helpful here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1578>
