This commit ports functionality from the `rtpsrc` to make the `ristsrc`
work with dynamic payload types.
It adds two properties:
- `caps`
- `encoding-name`
These can be used to make the `ristsrc` receive other payload types than
the MPEG TS one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5422>
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
This is consistent with the librtmp-based old rtmp plugin and ffmpeg.
While some servers require a valid flash-version, others are failing
with a too long or any flash-version at all.
By changing to the same default as in the old plugin and in ffmpeg,
GStreamer will at least behave the same and will work and fail with the
same servers without setting a flash-version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5293>
There is currently no way for applications to know if the stream has
been properly terminated by the server or if the network connection
was disconnected as EOS is sent in both cases.
Adding a property so connection errors can be reported as errors
allowing applications to distinguish between both scenarios.
Fix#2828
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5115>
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.
stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.
Just ignore the second pad. Nothing useful can be done with it anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4603>
This new property allows setting of PES stream number for AAC audio
and AVC video streams.
The stream number is subject to the following constraints:
1. it must be between 0 and 15 for video
2. it must be between 0 and 31 for audio
Currently the PES stream number is hard-coded to zero for these
stream types.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4822>
Add support for 10/12/14/16 bit depths . This consists of multiple parts.
First is the parsing of caps, which pulls out the bitness and endianness
from the video/x-bayer format.
Second, gst_bayer2rgb_split_and_upsample_horiz() is split into two similar
functions, one for 8bit bayer handling and another for 16bit bayer handling.
The content is basically identical, except one uses 8bpp and the other 16bpp
inputs and outputs, and they each use different ORC code to match. The 16bpp
variant also handles endian swapping. There is now a wrapper called
gst_bayer2rgb_split_and_upsample_horiz() which selects the correct function
based on bpp from the parser.
Third, gst_bayer2rgb_process() is extended to handle both 8bit and 16bit
bayer data. Yet again there are matching ORC functions to handle the 16bit
data. This time however the 16bit handling of data is slightly special. The
ORC is not able to emit opcodes for 'x2 mergelq', so the trick here is to
store the BG and GR longs into separate 'dtmp' temporary buffer, and then
do one more ORC post-processing step, compensate for the less-than-16bpp
bitness using left shift, and reorder them into the destination frame
using 'mergelq' .
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr16le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add comments regarding which LINE()s point to which data in the
temporary buffer and a large comment explaining how the buffer
is processed. This will hopefully be useful to someone, as the
code is not obvious. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Instead of passing a single element of GstBayer2RGB structure into the
gst_bayer2rgb_split_and_upsample_horiz(), pass the entire pointer and
let the funciton pick out whatever it needs out of the structure. This
is a preparatory patch. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Pass all three parameters used by the LINE() macro to the LINE() macro
and unroll the code for readability. Add more comments regarding which
of these LINE()s point to which data in the temporary buffer to make
the code less confusing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The j variable is used as an iterator further down in this code, but
here it can be just inlined in the macro parameters to make the code
easier to read. This is done in preparation for further changes. No
functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The bayer2rgb process implemented doesn't support in-place tranform.
This element doesn't implement a "transform_ip" vmethod of
GstBaseTransform it will revert to using the "tranform" vmethod.
It's misleading to set it to TRUE, here. Change this to FALSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add support for conversion to 10/12/14/16 bit bayer 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 num-buffers=1 ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
filesink location=/tmp/bayer12.raw
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The proxy callback for the notify::last-message was emiting the signal
again on the child, which caused an infinit loop. We could swap the child
and the user data to signal to the bin instead, but it was found that proxying
this signal was not very useful. Typical use case it to set silent=0 and use
deep-notify feature. Proxying that signal just duplicate that output which
isn't very useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4766>
adjust log level from GST_ERROR to GST_WARNING when h264 caps have
codec_data but no avc format or have no codec data or stream-format.
Because theses are not real errors, it is easy to mislead if print error
logs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4675>
We should behave similarly to video parsers so we can use:
- accept-template as we can also accept caps with missing fields.
- accept-intersect to do quick check with the pad template caps as it is
enough. Users should have figured the appropriate full caps on a
previous caps query
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4341>
The framerate should only be replaced (and corrected for alternating field)
when it is parsed from the bitstream. Otherwise, the upstream framerate
from caps should be trusted and assumed correct.
Related to gst-plugins-bad!2020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4259>
There may be garbage or some bits before a SOI comes in some problematic
mjpeg streams. For example, some network error may cause the EOI marker
of the previous frame lost, and when the new frame's SOI comes, we still
use the state of the last frame, which will generate errors.
For this kind of frames without EOI, if that frame already has some data
(the SOS segment is detected), we still push it as a frame with CORRUPTED
flag set. But if not, we just discard all the data before the new SOI.
Co-Authored-By: Víctor Jáquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4039>
It's only malformed data in APP when its length is less than 6 chars,
because it should have at least an id string. Otherwise, if the id string
is not handled, no warning is raised, only a debug message noticing it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3943>
When the QoS stats are reset (e.g. changing the source) the counters for
dropped + rendered frames are reset to zero which result in negative values
for their difference. This results in max-fps getting pegged at an extremely
high value.
```
fpsdisplaysink.c:373:display_current_fps:<fpsdisplaysink0> Updated max-fps to 36840705952231460864.000000
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3989>
If we know there's only one stream we care about and we
don't have to synchronise audio and video, or send RRs,
we might just as well not hook up all the RTCP bits and
use fewer threads and sockets and simplify the pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3531>
No matter if they're allocated via GSlice or malloc(). The allocator is
completely irrelevant, all local tags need to be in the primer so they
can be handled.
This didn't have any effect in practice because all local tags that
appear in the muxer are allocated via GSlice. Only from the demuxer they
might be allocated via malloc().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3699>
If a discontinuity is detected in push mode, we need to clear the cached section
observations since they might have potentially changed.
This was only done properly when operating with TIME segments (dvb, udp,
adaptive demuxers, ...) but not with BYTE segments (such as with custom app/fd
sources).
We still don't want to flush out the PCR observations, since this might be
needed for seeking in push-based BYTE sources.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1650
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3584>
This wasn't really done, and is needed in order to detect potential section
changes for sections that have got identical information (such as when switching
between streams that have the same PAT/PMT pid and subtable information).
Other checks exist in tsbase to detect if the "new" PAT/PMT really is an update or not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3530>
An end packet is only produced once for the last subtitle, so multiple
GAP events between subtitles would result only in a single end packet
and nothing else otherwise. This would potentially starve downstream
then, so instead forward the GAP events in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3534>
This reverts commit fcad4cc646.
This was wrong is so many ways.
* The memcmp was badly used (it should use == 0 to check the data is identical,
and not != 0)
* There was no boundary checks on the present stream section_data when passing
it to memcmp.
* The return value should have been TRUE (i.e. we have done all checks, none of
them failed, therefore the section has been seen before)
* stream->section_data would *always* be NULL if the section had already been
processed
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1559
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3421>
Add support for more formats so as to run the libvpx high bit depth test suite.
This means the files under CONFIG_VP9_HIGHBITDEPTH
This also allows running the yuv444p 8bit file in the regular 8 bit vp9 suite.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3356>
The gap handling was in place, but there was no event handler to trigger it.
Implement the alpha sink event handler for the gaps. This fixes handling of
valid streams which may not refresh the alpha frames for every video frames.
It will also allow a clean error if the stream was missing the initial
alpha frame, at least until we find a better way to handle these
invalid frames.
Related to #1518
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3264>
When the output alignment is smaller than the input alignment, for
example, When the output alignment is "FRAME" and the parse is likely
connecting to a decoder, the current PTS setting for AV1 frames inside
a TU is not very correct.
For example, a TU may begin with non-displayed frames and end with a
displayed frame. The current way will assign the PTS to the first
non-displayed frame, which is a decode-only frame and the PTS will be
discarded in the video decoder. While the last displayed frame has
invalid PTS, and so the video decoder needs to guess its PTS based on
the frame rate and previous frame's PTS. This is not a decent and
robust way. And more important, when the previous frames provide DTS,
the video decoder will also guess the PTS based on the previous frames'
DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a TU, let the non-displayed frames have
no PTS while set the correct PTS to the displayed one. Also, when the
AV1 stream has multi spatial layers, there are more than one displayed
frames inside one TU with the same PTS.
Note: If the input alignment is not TU aligned, we can not know the
exact PTS of this TU, and so we just clear the PTS of the decode only
frame and leave others unchanged.
We also correct all the PTS if the output is OBU aligned. All their
PTS and DTS are set to the input buffer's PTS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
When the incoming data has big alignment than the output, we do not need to
call finish_frame() and exit the current handle_frame() for each splitted
frame. We can push them all at one shot with in one handle_frame(), whcih
may improve the performance and can help us to find the edge of TU.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.
For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
