tsparse leaked input buffers quite badly:
GST_TRACERS=leaks GST_DEBUG=GST_TRACER:9 gst-launch-1.0 audiotestsrc num-buffers=3 ! avenc_aac ! mpegtsmux ! tsparse ! fakesink
The input_done vfunc was passed the input buffer, which it had to
consume. For this reason, the base class takes a reference on the buffer
if and only if input_done is not NULL.
Before 34af8ed66a, input_done was used in
tsparse to pass on the input buffer on the "src" pad. That commit
changed the code to packetize for that pad as well and removed the use
of input_done.
Afterwards, 0d2e908523 set input_done
again in order to handle automatic alignment of the output buffers to
the input buffers. However, it ignored the provided buffer and did not
even unref it, causing a leak.
Since no code makes use of the buffer provided with input_done, just
remove the argument in order to simplify things a bit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1274>
We might have to drain already queued input based on the old segment
before forwarding the new segment event. The new segment is only
forwarded after a discont as otherwise we might cause unnecessary
timestamp jumps as we output buffers timestamped based on sample counts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
If the input has a miss-placed filler zero byte (e.g. a filler without a 4
bytes start code on the next NAL), we would endup using the same timestamp
twice. Ask the base class to read the timestamp from the buffer were the NAL
actually starts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>
This will stop stripping four bytes start code. This was fixed and broken
again as it was causing the a timestamp shift. We now call
gst_base_parse_set_ts_at_offset() with the offset of the first NAL to ensure
that fixing a moderatly broken input stream won't affect the timestamps. We
also fixes the unit test, removing a comment about the stripping behaviour not
being correct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>
The volatile is not needed here and causes compiler warnings
with newer GLib versions.
gstautoconvert.c: In function ‘gst_auto_convert_dispose’ (and elsewhere):
glib/gatomic.h:108:3: warning: initialization discards ‘volatile’ qualifier from pointer target type [-Wdiscarded-qualifiers]
gstautoconvert.c:224:24: note: in expansion of macro ‘g_atomic_pointer_get’
224 | GList *factories = g_atomic_pointer_get (&autoconvert->factories);
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1237>
Otherwise we may endup pushing incomplete caps, which cause a renegotiation.
Note that this has the effect that caps are no longer pushed twice in presence
of valid framerate in the headers.
Otherwise we may endup pushing incomplete caps. Note that this has the side
effect that caps are no longer pushed twice in presence of VUI with valid
framerate.
There is some code to fixup broken stream that uses the SEI location,
this code is meant to locate SUFFIX SEI only. This should prevent
unwanted side effect if SUFFIX SEI is used.
Waiting for the next NAL increases the latency. If alignment=nal/au
has been negotiated, assumes the the buffer contains a complete
NAL and don't expect a second start-code. This way, nal -> nal,
au -> au and au -> nal no longer introduce latency.
As a side effect, the collect_pad() function was not able to poke at the
following NAL. This call is now moved before processing the NAL, so
it's looking at the current NAL before it's ingested into the parser
state in order to dermin if the end of an AU has been reached. The AUD
injection state as been adapted to support this.
This change will break pipelines if alignment=nal is used without respecting the
alignment. Effectively, the parser will no longer fix the broken aligment
which will result in parser error and the termination of the pipeline. Such
issue existed in tsdemux element and might exist in any forks of that code.
Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1193
Waiting for the next NAL increases the latency. If alignment=nal/au
has been negotiated, assumes that the buffer contains a complete
NAL and don't expect a second start-code. This way, nal -> nal,
au -> au and au -> nal no longer introduce latency.
As a side effect, the collect_pad() function was not able to poke at the
following NAL. This call is now moved before processing the NAL, so
it's looking at the current NAL before it's ingested into the parser
state in order to dermin if the end of an AU has been reached. The AUD
injection state as been adapted to support this.
This change will break pipelines if alignment=nal is used without respecting the
alignment. Effectively, the parser will no longer fix the broken aligment
which will result in parser error and the termination of the pipeline. Such
issue existed in tsdemux element and might exist in any forks of that code.
Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1193
Until now, any streams in tsmux had to be present when the element
started its first buffer. Now they can appear at any point during the
stream, or even disappear and reappear later using the same PID.
Per specification in 2.14.2 "For PES packetization, no specific data
alignment constraints apply". So we should not advertise NAL
alignment.
This bug was introduced at the same moment the alignment field was introduced
10 years ago. The plan was that alignment=none (or no alignment field) was to
be used for mpegtsdemux, but no one noticed the error. The reason is that at
the same moment, everything dealing with H264 started defaulting to AU
alignment.
https://bugzilla.gnome.org/show_bug.cgi?id=606662#c22
This patch will have a side effect that a parser is now needed after the
tsdemux element. The following pipeline will not negotiate anymore as the
mpegtsmux element requires alignment={nal,au}.
... ! tsdemux ! mpegtsmux ! ...
As a side effect, anyone that forked from tsdemux should updated their code to
fix this bug.
The comparision was not testing anything meaninful. This fixes the comparision
so we now update the caps whenever the value differ. This was detected by
coverity.
CID 1461291
We need to do this without holding the lock as the `g_async_queue_pop`
waits on the loop thread to deliver the stats. The loop thread might
attempt to take the lock as well, leading to a deadlock.
Taking a reference to the connection should be enough to keep this
safe.
Add new property "update-timecode" to allow updating timecode
in picture timing SEI depending on timecode meta. Since the picture
timing SEI message requires proper VUI setting but we don't support
re-writing SPS, this might not work for some streams
The rtpbin sends signals for all SSRCs. Don't send an EOS when the SSRC
does not match the stream SSRC.
This avoids problems when an SSRC from another receiver times out.
In some scenarios the fakevideosink shouldn't advertize the overlay-composition
meta for instance, so that overlay elements perform subtitles blending
themselves.
According to the specification, the adaptation field length must be 183 if
there is no payload data and < 183 if the packet contains an adaptation
field and payload data.
Unfortunately some payloaders always set the flag for payload data, even if
the adaptation field length is 183.
Don't return with an error in this case. Clear the payload data flag
instead and parse the adaptation field as usual. This avoids visual
artefacts for such streams.
For interlaced video:
* set the interlace mode in the src caps
* double the height from SPS in the caps.
* set field latency, instead of frame latency.
Fix#778
Add new property to signalling that there is no incoming data
from peer. This can be useful if users want to stop the streaming
when the connection is alive but no packet is arriving.
Both 2 and 4 are supported version of AAC ADTS format stream.
So we need to set correct version to help negotiation
especially for non-autopluggable pipeline.
Some mpeg-ts (HLS, DVB, ...) streams out there have completely broken
PCR streams on which we can't reliably recover correct timestamps.
For those, provide a property that will ignore the program PCR stream
(by faking that it's not present (0x1fff)).
Initially the case "only codec_data is different" was addressed in
https://bugzilla.gnome.org/show_bug.cgi?id=705333 in order for
unusual bitstreams to be handled. That's the case where sps and pps
are placed in bitstream. When sps/pps are signalled only via caps
by upstream, however, the updated codec_data is mandatory for decoder
and therefore we shouldn't ignore them.
According to following two specs, add support for AC4 in tsdemux.
1. ETSI TS 103 190-2 V1.2.1 (2018-02) : Annex D (normative): AC-4 in MPEG-2 transport streams
2. ETSI EN 300 468 V1.16.1 (2019-08) : Annex D (normative):Service information implementation of AC-3, EnhancedAC-3, and AC-4 audio in DVB systems
- Display caps of the pad we actually tried to link.
- Use the template caps as the filter is likely to not have any caps set
yet.
- Log pad name as well.
The approach is quite simple and doesn't take all use cases into account,
it only implements support when we are using the internal timecode we
create ourself.
Also the way we compute the sought frame count is naive, but it works
for simple cases.
Filter operates on raw data so don't allow decodebin to produce
encoded data if one is defined.
My use case here is keeping the video stream untouched but apply a filter
on the audio one, while keeping the same audio format.
In case the application has to deal with fussy servers. User agent
sniffing is so last decade.
Adds a property to set the Flash version on both the sink and the src.
The default stays the same (IIRC, Flash plugin for Linux from 2009).
The former uses a thread-safe way of getting statistics from the
connection without having to protect the fields with a lock.
The latter produces a zeroed statistics structure for use when no
connection exists.
Apply outgoing sizes only after writing the chunk to the peer. This is
important particularly for the set chunk size and allows exposing it
without threading issues.
Move output chunking from gst_rtmp_connection_queue_message into
gst_rtmp_connection_start_write, which effectively moves it from the
streaming thread into the loop thread.
This allows us to handle the outgoing chunk-size message (which is
generated by changing the future chunk-size property) properly, which
could come from any other thread.
