All these were copy pasted and would lead to assertion when chained with
rtpmux. This commit rewrite the negotiation with downstream. This also
drop the fallback to ancient names if the pad is unlinked. This was
completly arbitrary decision that made no sense.
https://bugzilla.gnome.org/show_bug.cgi?id=796809
Pass through closed caption data when deinterlacing. When two
deinterlaced frames are created for the same interlaced frame (e.g.
fields=all), the second of the two frames will have no closed caption
data.
Also fixed memory leaks related to timecode meta pass-through.
https://bugzilla.gnome.org/show_bug.cgi?id=796876
This causes rtspsrc to send a teardown and wait on
PAUSED->READY transition, with a configurable delay.
Otherwise, typically teardown never gets sent in
playbin / uridecodebin where the transition back to NULL
happens too quickly.
The timeout is set to 100ms default.
https://bugzilla.gnome.org/show_bug.cgi?id=751994
Just remove the code. It's not doing anything useful anyways. The modified
caps are the result of a caps query, so either not used afterwards of a
reference to some internal caps of another element that should not be
modified.
https://bugzilla.gnome.org/show_bug.cgi?id=796837
When it is trivial to pass-through a timecode, by only removing the
"interlaced" flag, do pass-through. Otherwise, double the fps_n and
adjust the "frames" field.
https://bugzilla.gnome.org/show_bug.cgi?id=796818
When handling input with timestamps that repeat, sometimes
splitmuxsink would get confused and ignore a keyframe.
The logic in question is a holdover from before the cmd queue
moved the file cutting to the multiqueue output side and made
it deterministic, so it's no longer needed on the input
here.
https://bugzilla.gnome.org/show_bug.cgi?id=796773
This reverts commit 3ac5430311.
There's no need to make a freshly created event writable,
and the other half of this patch was already fixed
and pushed in f2f15a1
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.
We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
* When receiving a segment in TIME, use that seqnum
* Only reset the stored sequence number when doing HARD reset
(and not when we get a FLUSH event from upstream)
This patch aims at fixing the recent regressions in the adaptive test
suite.
All segment pushing in push mode is now done with
gst_qtdemux_check_send_pending_segment(), which is idempotent and
handles both edit lists cases and cases where the upstream TIME segments
have to be sent directly.
Fragmented files that start with a non-zero tfdt are also taken into
account, but their handling has been vastly simplified: now they are
handled as implicit default seeks so there is no need to extend the
GstSegment formulas as was being done before.
qtdemux->segment.duration is no longer modified when
upstream_format_is_time, respecting in this way the durations provided
by dashdemux and fixing bugs in reverse playback tests where mangled
durations appeared in the emitted segments.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Upstream driving elements such as dashdemux often do reverse playback by
feeding qtdemux with the fragments containing the requested playback
range in reverse order.
But the requested playback range stop may be somewhere in the
middle of a fragment. In that case, a naive pts >= segment.stop
condition may declare end of segment prematurely when demuxing this
first fragment.
This used not to happen because there were places in moov parsing where
segment.stop was overwritten to GST_CLOCK_TIME_NONE even if
upstream_format_is_time -- resulting in this case in a segment with rate
< 0 and stop == -1 and hence not triggering the EOS check, but that was
likely an accident.
This patch modifies the EOS check to take this case into account, not
sending EOS when upstream_format_is_time if rate < 0.
This fixes adaptive.dash.playback.seek_end_live.DASHIF_livestream_testpic_2s
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Sample table based segment event (genereted by qtdemux) could break
presentation timeline. For example, qtdemux should not modify upstream
time format segment (e.g., adaptivedemux use case)
https://bugzilla.gnome.org/show_bug.cgi?id=796480
This field is actually only informatory and the user can potentially
choose something else. EME tests in WebKit testsuite actually doesn't
take it into and force another encryption system to be used, and expects
to be given the occasion to do so.
This basically also reverts 3e063703b3.
Instead of always keeping a safe segment (start=0) event from the beginning,
delay the creation of this event to when we really know the timestamp of the
first sample. This is important to properly start fragmented streams that
we might join in the middle or to play isolated fragment files that might
have an advanced tfdt.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Fragmented files often use elst.duration=0 which before
ee78825eae was wrongly interpreted as
having no frames.
Since that issue has now been fixed, there is no reason to disable edit
lists in fragmented files. This commit enables them, therefore producing
correct stream time for files containing edit lists.
https://bugzilla.gnome.org/show_bug.cgi?id=793058
Since ca068865c3 the duration of the first
frame is not used for estimating the frame rate.
For this purpose, stream->first_duration was initialized with the
duration of the first frame. In fragmented files, this was previously
done by peeking the first moof, but that can only be done in pull mode.
Fortunately, we don't really need to do that, at least with the current
design: When we are estimating the frame rate we already have the
sample table, regardless of the scheduling mode and whether the file is
fragmented or not, so we can obtain first_duration there much more
reliably.
This fixes frame rate estimation for fragmented files in push mode.
https://bugzilla.gnome.org/show_bug.cgi?id=796384
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.
This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.
https://bugzilla.gnome.org/show_bug.cgi?id=783754
This is a straightforward translation of 5dd39d8, can be trivially
checked by running:
gst-launch-1.0 -v videotestsrc ! video/x-raw, colorimetry=2:4:7:1 ! \
matroskamux ! matroskademux ! fakesink
and verifying that the colorimetry is correctly preserved.
https://bugzilla.gnome.org/show_bug.cgi?id=796344
The code before copied GstStructure twice. The first time inside
gst_value_set_structure and the second time in g_value_array_append.
Optimized version does no copies, just transfers ownership to
GValueArray. It takes advantage of the fact that array has already
enough elements preallocated and the memory is zero initialized.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
Whenever got new moov or new stream-start,
demux will try to expose new pad by following rule.
Comparing stream-id in the current moov with previous one, then
* If matched stream-id is found from previous one,
reuse existing pad (most common case)
* Otherwise, expose new pad with new stream-start
* No more used stream will be freed
https://bugzilla.gnome.org/show_bug.cgi?id=684790
Whenever demux got moov, demux will create new stream. Only exception is
duplicated track-id in a moov box. In that case the first stream
will be accepted. This patch is pre-work for rework of moov handling.
https://bugzilla.gnome.org/show_bug.cgi?id=684790
Supports CEA 608 and CEA 708 CC streams
Also supports usage in "Robust Prefill" mode if the incoming caption
stream is constant (i.e. there is one incoming CC buffer for each
video frame).
https://bugzilla.gnome.org/show_bug.cgi?id=606643
ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.
rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:
* The protection packets don't make it as far as the depayloader,
which means it will mark buffers as DISCONT every time the previous
packets were protected
* While we could work around the previous issue by looking at
the protection packets ignored and dropped in rtpptdemux, we
would still mark buffers as DISCONT when a FEC packet was lost,
as we cannot know that it was indeed a FEC packet, even though
this should have no impact on the decoding of the stream
With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.
https://bugzilla.gnome.org/show_bug.cgi?id=794909
This reverts commit af273b4de9.
While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.
Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=793964
After a CAPS event, in theory a new stream can start and it might start
with the FLAC headers again. We can't detect FLAC headers in the middle
of the stream, so we drain the parser to be able to detect either FLAC
headers after the CAPS event or the continuation of the previous stream.
This fixes for example
gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \
audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \
concat name=c ! rtpgstpay ! udpsink host=127.0.0.1 port=5000
gst-launch-1.0 udpsrc multicast-group=127.0.0.1 port=5000 \
caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \
rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin
gst_qt_mux_can_renegotiate () gets called everywhere following
that pattern:
return gst_qt_mux_can_renegotiate (ref(self));
This means the reference must be released both in the success
and failure cases, it was only done in the success case.
It can happen during teardown that the reference context becomes NULL.
In that case, trying to send the fragment-opened-closed message would
lead to a crash.
Corrupted files could potentially have multiple cdat/cdt2 atoms in
a sample entry, which is unclear how to handle.
Ignore repeated ones.
CID #1434162
CID #1434159
The code responsible for creating retransmitted buffers
assumed the stored buffer had been created with
rtp_buffer_new_allocate when copying the extension data,
which isn't necessarily the case, for example when
the rtp buffers come from a udpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=794958
Similar to the get-session and get-internal-session signals,
we expose a get-storage signal in addition to the
get-internal-storage signal to give access to the actual
element for applications that need to set properties on the
element, in particular "size-time"
https://bugzilla.gnome.org/show_bug.cgi?id=794910
With this the muxer is not set to NULL after each segment but instead
only flush events are sent to it to reset the EOS state.
As a result, the muxer will keep stream state and e.g. mpegtsmux will
keep the packet continuity counter continuous between segments as needed
by hlssink2.
https://bugzilla.gnome.org/show_bug.cgi?id=794816
The value stored in cenc_aux_sample_count wasn't in sync with the
parsing code that followed which checks whether all entries are
valid and present.
Only write the actual sample count when we know for sure.
CID #1427087
This exposes a new property, mtu, which is used to determine the
initial size of buffers from the buffer pool. If received data
exceeds this, the element gracefully handles that in a manner similar
to what we had previously: a large memory gets filled and reallocated
at the next call to "fill".
The default size is set to 1500, which should cover most use cases.
With contributions from Mathieu Duponchelle <mathieu@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=772841
Optimize GstUdpSrc for cache performance.
Move the hot properties, which are used by the read function, to the top:
@used_socket, @addr, @cancellable, @skip_first_bytes, @timeout,
@retrieve_sender_address.
Remove the unused property @ttl.
Where needed reorder so that holes are avoided (the 64-bit @timeout)
https://bugzilla.gnome.org/show_bug.cgi?id=772841
The samples table is sorted by DTS, not PTS. As such we can only get the
correct result when using a binary search on it, if we search for the
DTS.
Also if we only ever search for the frame, where the following frame is
the first one with a PTS after the search position, we will generally
stop searching too early if frames are reordered.
In forwards playback this is not really a problem (after the decoder
reordered the frames, clipping is happening), in reverse playback
it means that we can output one or more frames too few as we stop too
early and the decoder would never receive it.
https://bugzilla.gnome.org/show_bug.cgi?id=782118
76e458a119 changed the conditions from
"queued > threshold" to "queued >= threshold", which broke hlssink2 and
resulting in too small fragments being created although keyframes would
be at *exactly* the configured threshold.
https://bugzilla.gnome.org/show_bug.cgi?id=794440
Fix compilation with MSVC. We still assume that attribute
is supported by all other relevant compilers, which seems
to be the case since we haven't had any complaints about
similar code in rtpsbcpay.
This works around a bug in various ONVIF cameras that implement the
attributes the wrong way around. They still won't work with a
backchannel but at least normal playback will work for the time being.
It restores pre-1.14 behaviour where we would fail to preroll on any SDP
that lists a recvonly stream. For 1.16 a better solution should be
found.
The problem here is that the ONVIF spec has the meaning of the two
attributes the wrong way around in the examples, compared to RFC4566.
https://bugzilla.gnome.org/show_bug.cgi?id=793715
Only up to timescale * G_MAXINT16 is possible as cluster duration, which
is already higher than our default value. Using higher values would
cause overflows and broken files.
Based on the investigation by Nicola Murino <nicola.murino@gmail.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792775
Matroska does not support changing the stream type and stream properties
after the headers were started to be written, and for example H264
codec_data changes can't be supported.
https://bugzilla.gnome.org/show_bug.cgi?id=782949
rtpulpfeccommon.c:432:27: error: format ‘%lx’ expects argument of type
‘long unsigned int’, but argument 10 has type ‘guint64 {aka long long unsigned int}’
https://bugzilla.gnome.org/show_bug.cgi?id=793732
The ulpfecenc "mux-seq" and "ssrc" properties were initially added
because the element did more than implement ULPFEC. As it was
decided that FLEXFEC would be implemented in a separate element,
both properties are now unneeded and confusing.
Change the default for the ulpfecenc multi-packet property,
as it is expected that most users of this element will be protecting video
streams.
Change the default property for the rtpredenc allow-no-red-blocks
property, as it should also be its default mode of operation.
https://bugzilla.gnome.org/show_bug.cgi?id=793843
It is expected that when connecting to a stream that has
already started, the caps will only arrive at the interval
specified on rtpgstpay, we shouldn't be warning as this is
a normal mode of operation.
https://bugzilla.gnome.org/show_bug.cgi?id=793798
We expose a set of new elements:
* ULPFEC encoder / decoder
* A storage element, which should be placed before jitterbuffers,
and is used to store packets in order to attempt reconstruction
after the jitterbuffer has sent PacketLost events
* RED encoder / decoder (RFC 2198), these are necessary to
use FEC in webrtc, as browsers will propose and expect ulpfec
packets to be wrapped in red packets
With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>
Sebastian Dröge <sebastian@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792696
Packets with these payload types will be dropped. A use case
for this is FEC, where we want FEC packets to go through the
jitterbuffer, but not be output by rtpbin.
https://bugzilla.gnome.org/show_bug.cgi?id=792696
