This patch allows matroskademux to parse a second Tracks element,
erroring out if the tracks are not compatible (different number, type or
codec) and emitting new caps and tag events should they have changed.
https://bugzilla.gnome.org/show_bug.cgi?id=793333
This splits gst_matroska_demux_add_stream() into:
* gst_matroska_demux_parse_stream(): will read the Matroska bytestream
and fill a GstMatroskaTrackContext.
* gst_matroska_demux_parse_tracks(): will check there are no repeated
tracks.
* gst_matroska_demux_add_stream(): creates and sets up the pad for the
track.
https://bugzilla.gnome.org/show_bug.cgi?id=793333
This is necessary for MSE, where a new MSE initialization segment may be
appended at any point. These MSE initialization segments consist of an
entire WebM file until the first Cluster element (not included). [1]
Note that track definitions are ignored on successive headers, they must
match, but this is not checked by matroskademux (look for
`(!demux->tracks_parsed)` in the code).
Source pads are not altered when the new headers are read.
This patch has been splitted from the original patch from eocanha in [2].
[1] https://www.w3.org/TR/mse-byte-stream-format-webm/
[2] https://bug334082.bugzilla-attachments.gnome.org/attachment.cgi?id=362212https://bugzilla.gnome.org/show_bug.cgi?id=793333
The behaviour of split-now is to output the current GOP after
starting a new file.
The newly-added split-after signal will output the current GOP
to the old file if possible once a new GOP is opened.
https://bugzilla.gnome.org/show_bug.cgi?id=796982
For 59.94 FPS, it's common to set 60000 as timescale. For that
timescale, if the audio is late by as little as 0:00:00.000016666
(definitely less than one audio sample), lateness gets rounded to 1.
Added a safeguard that allows lateness up to 1 sample with the specific
trak's timescale, to make sure that values less than e.g. one audio
sample won't break the prefill mode. What will happen in this case is
that the audio will get squeezed back to the video's timestamp, which in
practice means that the audio will be 0.000016666 seconds early (with
the patch).
https://bugzilla.gnome.org/show_bug.cgi?id=797133
Accept wavpack correction streams (.wvc) on sink pad, so
that wavpackparse can also be used to packetise correction
streams.
Fix parsing of subblock ID tags - the higher bits are
flags and are not part of the ID. This resulted in
correction blocks not being recognised properly and
the output not having the right (correction) caps.
Currently, whenever we generate a 128-bit UID, we store it in a list and
return 0 if we ever encounter a collision. This is so mathematically
improbable that it's not worth checking for, so we can save memory and
time by not tracking the UID. Even if a collision happened, a list of
only 10 UIDs would be unlikely to detect it.
This article has a good description of how improbable a collision is:
https://en.wikipedia.org/wiki/Universally_unique_identifier#Collisionshttps://bugzilla.gnome.org/show_bug.cgi?id=797086
This patch clears the sample table whenever the demuxing of a new
fragment begins. This avoids increasing memory usage for long videos.
This behavior was already present when upstream_format_is_time; this
patch extends it to all push mode operation (e.g. Media Source
Extensions).
https://bugzilla.gnome.org/show_bug.cgi?id=796899
Both rtpmp4vpay and rtpmp4gpay support MPEG4 elementary streams. But
the most supported variant is the video-specific one (rtpmp4vpay),
therefore increase the rank of that one so that auto-plugging of
payloaders for MPEG4 elementary streams ends up picking that one
and not the generic one.
If we have cluster prev size (GStreamer muxer will write it by default),
we can go back to the previous cluster efficiently, but if we don't then
just search backwards until we find a cluster ebml identifier, like we
do when searching for clusters in the bisection loop.
Add property instead of hardcoding it in the code.
In some scenarios such as CCTV variable fps and extra long GOPs are
used to minimise storage space, for example. In those cases there might
not be any keyframes for many minutes, so provide a property to override
the max allowed distance.
https://bugzilla.gnome.org/show_bug.cgi?id=790696
When seeking in pull mode without an index (because there is no index
or the file is still being written to) we bisect to find the right
cluster to jump to. However, it's possible the cluster we found doesn't
start with a keyframe, which leads to decoding errors, so if we know
that the found cluster starts with a delta frame try to scan back to
previous clusters until we find one that starts with a keyframe or
we are back at the beginning. Theoretically it's possible that all
clusters but the first one do not start with a keyframe and the
keyframes are in the middle of clusters, but this is extremely
unusual, so we will cover this case with a basic sanity check.
This problem is especially problematic with content recorded with
dynamic GOP and FPS, where long GOP lengths and low FPS may cause a
large set of clusters to lack key frames. Playback would then be
started on a non-keyframe cluster, and the large number of such frames
would make the content impossible to decode fo a long stretch of time.
Based on patch by: Mats Lindestam <matslm@axis.com>
https://bugzilla.gnome.org/show_bug.cgi?id=790696
This is useful for reverse playback/trickmodes
without an index, and will also be useful in the
seek handler if we need to scan back to find a cluster
that starts with a keyframe.
https://bugzilla.gnome.org/show_bug.cgi?id=790696
This is an enum not a boolean, and a value of 2 signals
that the video is progressive, but we would mistakenly set
interlace-mode=mixed on the output caps.
https://bugzilla.gnome.org/show_bug.cgi?id=787206
On Linux, the kernel returns twice the size as it will allocate extra
space for accouting. We devides this value by two in order to ensure
that get/set value now match. This fixes the set buffer size validation
and allow having a nice warning when the size if surpassed and the
process does not have CAP_NET_ADMIN capabilities.
https://bugzilla.gnome.org/show_bug.cgi?id=727067
The udp buffer size is limited to a maximum of around 100K.
Some apps need to set the force bufsize for their own operation.
Use the SO_RCVBUFFORCE option in order to override the rmem_max limit
of linux kernel. Require user to have the CAP_NET_ADMIN privilege to
work.
Original patch from Kyungnam Bae <kyungnam.bae@lge.com>
https://bugzilla.gnome.org/show_bug.cgi?id=727067
rtph264pay and rtph265pay skip updating the parameter set timestamp if
the units they see contain no new configuration. This can result in
them injecting duplicate parameters.
https://bugzilla.gnome.org/show_bug.cgi?id=796748
The stream context was holding a reference to the
internal queue and pads, with pad probes that were
in turn holding references to the stream context.
This lead to a leak if the request pads weren't explicitly
released.
https://bugzilla.gnome.org/show_bug.cgi?id=796893
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
