Commit graph

10270 commits

Author SHA1 Message Date
Kristofer Björkström
54b6ee0c55 buffermemory: keep track of buffer size and current offset
Added the possibility to get current offset and the total size of the
buffer.
2020-04-03 17:01:24 +02:00
Havard Graff
d9aaa15a30 rtpopuspay: make depay ! pay work
There is a use-case for a server to re-payload opus going through it.

Problem was that the payloader requires channels in the caps, but
this is not something the depayloader can parse out of the stream, meaning
caps-negotiation would fail.

Removing the requirement of channels in the template-caps fixes this.
2020-04-03 09:04:32 +00:00
Seungha Yang
599066726f splitmuxsink: Don't send too many force key unit event
splitmuxsink should requst keyframe depending on configured
threshold and previously requested time in order to avoid too many
keyframe request.
2020-04-03 15:00:37 +09:00
Jan Schmidt
78eaa7c6ed matroska: Check the return value of gst_segment_do_seek()
gst_segment_do_seek() can fail.
2020-04-02 05:23:17 +00:00
Sebastian Dröge
f757fbe0f7 qtdemux: Send instant-rate-change event if requested in the SEEK event
Handle an instant rate change seek immediately by reflecting
it downstream as an instant-rate-change event, and do no
further seek handling.
2020-04-02 05:23:17 +00:00
Sebastian Dröge
5d0657d4ae matroska-demux: Send instant-rate-change event if requested in the SEEK event
Short-circuit instant rate change events by generating
a downstream instant-rate-change event and doing no further
seek processing.
2020-04-02 05:23:17 +00:00
Seungha Yang
cb8c83e799 matroska: Update for video-hdr struct change
See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594
2020-04-01 05:19:24 +00:00
Aaron Boxer
3b65663846 rtpbin: make warning messages more meaningful 2020-03-31 15:51:27 -04:00
Nicolas Pernas Maradei
ce0fb9bd29 rtpsession: rename RTCP thread
RTP session starts a new thread for RTCP and names it
"rtpsession-rtcp-thread" which happens to be longer than the maximum 16B
allowed by pthread_setname_np and causes the naming to fail.
See docs for more details.

This commit simply shortens the thread's name so it can actually be set.
2020-03-31 13:34:07 +02:00
Havard Graff
3368ed44a3 rtpjitterbuffer: create specific API for appending buffers, events etc
To avoid specifying a bunch of mystic variables.
2020-03-31 10:02:57 +00:00
Havard Graff
818b38ebdd rtpjitterbuffer: fix waiting timer/queue code
Changing the types from boolean to guint due to the ++ operand used on
them, and only call JBUF_SIGNAL_QUEUE after settling down,
or else you end up signaling the waiting code in chain() for every buffer
pushed out.
2020-03-30 22:32:21 +02:00
Sebastian Dröge
d427b9bddf qtmux: Error out instead of crashing if reserved-max-duration is 0 or no samples could be created in prefill mode 2020-03-27 10:35:04 +00:00
Jan Schmidt
00a08c69ac splitmuxsrc: Fix some deadlock conditions and a crash
When switching the splitmuxsrc state back to NULL quickly, it
can encounter deadlocks shutting down the part readers that
are still starting up, or encounter a crash if the splitmuxsrc
cleaned up the parts before the async callback could run.

Taking the state lock to post async-start / async-done messages can
deadlock if the state change function is trying to shut down the
element, so use some finer grained locks for that.
2020-03-26 14:44:54 -04:00
Seungha Yang
a40eacabb4 splitmuxsink: Split fragment only if queued time is larger than threshold
The queued time includes the duration of the last queued frame
(i.e., new keyframe) so the condition check should not be inclusive.
Note that the new fragment will be cut excluding the last frame
and therefore if the condition is inclusive way,
the fragment might have one frame shorter duration for all keyframe
stream such as jpeg or all-inter video streams.
2020-03-25 13:22:31 +00:00
Seungha Yang
6256fc67e4 splitmuxsink: Don't need to trace next timecode for split decision
Since the commit 94bb76b6b9, splitmuxsink
will split fragments based on queued time and the threshold of that.
So don't need to store the next timecode for split decision.
2020-03-25 13:22:31 +00:00
Seungha Yang
0acd5d9f8b splitmuxsink: Mark some split decision related properties as MUTABLE_READY
The change of various criteria for split decision while muxing is on progress
wouldn't work well as expected.
2020-03-24 22:09:48 +09:00
Seungha Yang
94bb76b6b9 splitmuxsink: Take account queued time and max-size-timecode for split decision
Not only the requested keyframe time, the queued size should be
a criterion for the split decision of timecode based mode
(same as max-size-time based split case).
2020-03-24 22:04:21 +09:00
Xavier Claessens
6e1758d509 Fix usage of C99
It's 2020, way too early for that, let's stick to C89 for now.
2020-03-23 21:32:04 -04:00
Havard Graff
a710bda1ab rtptimerqueue: remove ->num from the timer
This concept was only used by the "multi"-lost timer, and since that
one is not around any longer, the "num" concept is superfluous.
2020-03-20 13:17:20 +00:00
Havard Graff
f1ff80ced0 rtpjitterbuffer: remove the concept of "already-lost"
This is a concept that only applies when a buffer arrives in the chain
function, and it has already been scheduled as part of a "multi"-lost
timer.

However, "multi"-lost timers are now a thing of the past, making this
whole concept superflous, and this buffer is now simply counted as "late",
having already been pushed out (albeit as a lost-event).
2020-03-20 13:17:20 +00:00
Havard Graff
5dacf366c0 rtpjitterbuffer: immediately insert a lost-event on multiple lost packets
There is a problem with the code today, where a single timer will
be scheduled for a series of lost packets, and then if the first packet
in that series arrives, it will cause a rescheduling of that timer, going
from a "multi"-timer to a single-timer, causing a lot of the packets
in that timer to be unaccounted for, and creating a situation in where
the jitterbuffer will never again push out another packet.

This patch solves the problem by instead of scheduling those lost packets
as another timer, it instead asks to have that lost-event pushed straight
out.

This very much goes with the intent of the code here: These packets are
so desperately late that no cure exists, and we might as well get the
lost-event out of the way and get on with it.

This change has some interesting knock-on effect being presented in
later commits. It completely removes the concept of "already-lost", so
that is why that test has been disabled in this commit, to be
removed later.
2020-03-20 13:17:20 +00:00
Havard Graff
2fa7e6a6d4 rtpjitterbuffer: refactor lost_timeout code
Split it up in code related to the timer, (do_lost_timeout) and code
to insert a lost-item/event and update private jitterbuffer-variables.
2020-03-20 13:17:20 +00:00
Seungha Yang
4f443c81cf qtmux: Fix build warning
gstqtmux.c(644): warning C4133: '=':
  incompatible types - from 'gboolean (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'
  to 'GstFlowReturn (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'
2020-03-19 19:20:05 +00:00
Jan Schmidt
c5181c23a4 splitmuxsink: Reset cleanly for reuse
Reset the splitmuxsink completely when changing states so that
it can be reused.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1241
2020-03-19 15:37:14 +00:00
Zebediah Figura
71bb53a648 mpegaudioparse: Use a constant bit rate to convert between time and bytes if possible.
This should result in no worse accuracy than the base parse element, and may
result in better accuracy. In particular, the number of bytes processed at any
given point, as accumulated by baseparse, can be only accurate to
(1 / # of frames) bytes per second, and if we try to seek immediately after
pausing the pipeline to a large offset, this small inaccuracy can propagate to
something noticeable.

The use case that prompted this patch is a 45-minute MPEG-1 layer 3 file, which
has a constant bit rate but no seek tables. Trying to seek the pipeline
immediately after pauisng it, without the ACCURATE flag, to a location 41
minutes in, yields a location that is, even with <https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/374>,
still audibly incorrect. This patch yields a much closer position, no longer
audibly incorrect, and likely within a frame of the most correct position.
2020-03-19 14:02:44 +00:00
Mathieu Duponchelle
56e5243f03 qtmux: fix renegotiation check
By the time sink_event is called, the pad's current caps have
already been updated. To address this, implement sink_event_pre_queue,
and check if the pad can be renegotiated there.

Fixes #707
2020-03-19 23:34:52 +11:00
Seungha Yang
18e09de0a2 splitmuxsink: Decouple keyframe request and the decision for fragmentation
Split the decision for keyframe request and fragmentation in order to
ensure periodic keyframe request.
2020-03-19 10:17:21 +00:00
Stian Selnes
81a87c26f9 rtpvp8pay, rtpvp9pay: fix caps leak in set_caps() 2020-03-12 16:49:58 +00:00
Edward Hervey
5a893f2a95 videomixer: Don't leak peer caps 2020-03-12 11:22:56 +01:00
Thibault Saunier
21bc0d527b imagesequencesrc: Cleanup and add some features
* Implement the GstURIHandlerInterface
* Rework the locking
* Implement backward seeking handling
* Generate documentation
2020-03-11 15:11:54 +00:00
Fabian Orccon
7511999083 Add an imagesequencesrc element to stream sequence of images
See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/121
2020-03-11 15:11:54 +00:00
yychao
7f89085251 qtdemux: Add support for AC4
The caps received from qtdemux for AC-4 content are audio/x-gst-fourcc-ac_4

Based on patch by: Savinderjit Kaur

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/413
2020-03-10 15:28:01 +00:00
Matthew Waters
dacdc74043 imagefreeze: handle reconfigure events on the srcpad 2020-03-10 21:22:20 +11:00
Matthew Waters
07a8a1c484 imagefreeze: properly ignore setting caps failures
Ignore the return value of gst_pad_set_caps() so that setcaps will set a
framerate that is usable.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/705
2020-03-10 21:22:03 +11:00
Matthew Waters
28f49e1fd5 imagefreeze: don't fail sending sticky events downstream
They will be repropagated anyway.
2020-03-10 21:08:45 +11:00
Markus Ebner
5dcbb6b0d8 videocrop: Add support for Y41B and Y42B 2020-03-10 08:24:56 +00:00
Markus Ebner
b562235283 videocrop: Add support for Y444
- Refactored the planar transform method to support all video formats
  that are stored planar, independent of the used subsampling
- Added support for Y444
2020-03-10 08:24:56 +00:00
Markus Ebner
4a9e5bbf8b videocrop: Use G_VALUE_INIT to initialize GValues 2020-03-10 08:24:56 +00:00
Ognyan Tonchev
a78a74bff0 rtph26x: Use gst_memory_map() instead of gst_buffer_map() in avc mode
gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory and when AVC (length-prefixed) alignment is used.
This has quite an impact on performance on systems with limited amount of
resources. With this patch the whole GstBuffer will not be mapped at once,
instead each individual GstMemory will be iterated and mapped separately.
2020-03-06 10:44:16 +00:00
Havard Graff
4046970b01 rtptwcc: make RTPTWCCManager a GObject 2020-03-04 16:48:04 +01:00
Havard Graff
026223cde2 rtpjitterbuffer: fix stalling when resetting timers
When calling gst_rtp_jitter_buffer_reset you pass in a seqnum.

This is considered the starting-point for a new stream.

However, the old behavior would unref this buffer, basically lying to
the thread that is pushing out buffers saying that it can expect
this buffer, when it would never arrive. The resulting effect being no
more buffer pushed out of the jitterbuffer, and it would buffer
incoming data indefinitely.

By instead inserting the buffer in the gap_packets queue, the _reset()
function will take responsibility for using that as the first buffer
of the new stream.

Fixes #703
2020-03-04 12:55:52 +01:00
Jan Schmidt
f490c38416 splitmux: Avoid negative DTS
In order to concatenate fragments, splitmuxsrc offsets
the start of each fragment PTS to 0 to align it with the
previous file. This means that DTS can go negative for
the first fragment, with really bad results.

Add a fixed offset to outgoing timestamp ranges to
avoid that.
2020-03-04 05:42:21 +00:00
Jan Schmidt
54f68ff36b qtmux: Remove warning in the log for mono video
Vanilla mono video was generating a spurious warning into the debug log
that's just misleading. Handle mono caps explicitly to avoid the warning.
2020-03-04 04:14:40 +00:00
Guillaume Desmottes
d43ad6e029 deinterlace: add alternate support
In this mode each field is carried using its own buffer.
Allow deinterlace to negotiate caps with the Interlaced feature and
adjust the algorithm fetching lines.

Fix #620
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
b3d96e06c6 deinterlace: add wrapper to get field lines from history
No semantic change so far, will be used to implement alternate support.
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
f0eb1419f6 deinterlace: stop checking line index boundaries
The LINE2() macro already prevents out of bound indexes using CLAMP_HI()
and CLAMP_LOW().
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
cca8008779 deinterlace: fix video info on output frames
Output frames used to have their interlace mode set to the same one as
the input. This breaks their field and comp heights when deinterlacing
an alternate stream.
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
6dde6038cc deinterlace: use output caps to compute buffer size
In interlace-mode=alternate the input buffers have half the size of the
output ones as each field has its own buffer.
2020-03-03 17:15:00 +00:00
Jennifer Berringer
3287f1cb3f flacparse: fix broken reordering of flac metadata
Each FLAC metadata block starts with a flag denoting whether it is the
last metadata block. The existing flacparse code moves any existing
VORBISCOMMENT block to immediately follow the STREAMINFO block without
changing any block's last-metadata-block flag. If no VORBISCOMMENT block
exists, it created one with the last-metadata-block flag set to true.
This results in gstflacdec sometimes giving bad headers to libflac when
trying to play perfectly valid FLAC files depending on the file's
metadata ordering. Depending on the contents of the other metadata
blocks, current versions of libflac may or may not return
FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER when given this broken
metadata. This is most noticeable with files that have a large cover art
image attached where VORBISCOMMENT is the very last metadata block with
no PADDING afterwards.

This patch changes that behavior so that:

1. For FLAC files that already have a VORBISCOMMENT block, the metadata
   order is preserved.
2. For FLAC files that do not have a VORBISCOMMENT block, the generated
   dummy VORBISCOMMENT is placed immediately after STREAMINFO and
   inherits the last-metadata-block flag from STREAMINFO.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/484
2020-03-03 08:03:32 +00:00
Sebastian Dröge
885d330ee6 qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values 2020-02-28 13:52:40 +00:00
Sebastian Dröge
9e9af6711d qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats
Previously it was only used for in24.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
67be373221 qtdemux: Fix up header information for various fixed-format raw audio formats
Sometimes the headers contain useless, wrong or zero values for e.g. the
sample size with these formats. There's only a single valid value for
them so let's set these instead.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
2c5f6e508c qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs 2020-02-28 13:52:40 +00:00
Sebastian Dröge
65b30ecce6 qtdemux: Add some more raw audio fourccs to the header instead of duplicating them 2020-02-28 13:52:40 +00:00
Nirbheek Chauhan
42e7864e90 rtpjitterbuffer: Don't use glib format modifiers with sscanf
We do not have a way to know the format modifiers to use with string
functions provided by the system. G_GUINT64_FORMAT and other string
modifiers only work for glib string formatting functions. We cannot
use them for string functions provided by the stdlib. See:
https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description

```
../gst/rtpmanager/gstrtpjitterbuffer.c: In function 'gst_jitter_buffer_sink_parse_caps':
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: unknown conversion type character 'l' in format [-Werror=format=]
           || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
                                ^~~~~~~~~~
In file included from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib.h:30,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/rtp/gstrtpbuffer.h:27,
                 from ../gst/rtpmanager/gstrtpjitterbuffer.c:108:
/home/nirbheek/cerbero/build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
 #define G_GUINT64_FORMAT "llu"
                            ^
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: too many arguments for format [-Werror=format-extra-args]
           || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
                                ^~~~~~~~~~
```

See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/379
2020-02-26 19:05:24 +05:30
Sebastian Dröge
35a1cedb97 qtmux: Add support for 8k resolutions in prefill mode with ProRes 2020-02-25 15:46:44 +02:00
Sebastian Dröge
3998b7cb4c rtpjitterbuffer: Include string.h for memcpy() / memset()
Usually something else is pulling it in somehow already, but not on
Windows.
2020-02-25 09:07:47 +00:00
Håvard Graff
fdf002d069 rtpsession: fix crash when no extension-header present for twcc 2020-02-24 13:06:27 +00:00
Johan Bjäreholt
ce802f033c matroska-mux: Fix incorrect rounding of timestamps
Previously we saved the buffer_timestamp straight into
mux->cluster_time. Since the cluster time saved into the file does not
have as high precision as GstClockTime depending on the timecodescale
the rounding of relative_timestamp was invalid as mux->cluster_time
which it was calculated relative to was not equal to the cluster time
written to the matroska file.

Example of "mkvinfo -v" of how it looks before and after this change in
an scenario where previously timestamps got out of order because of this
issue.

Notice the timestamp of the SimpleBlock right before and right after the
Cluster now being in order. The consequence of this however is that the
cluster timestamp is not necessarily the same as the timestamp of the
first buffer in the cluster however (in case it's rounded up).

Before

| + SimpleBlock (track number 1, 1 frame(s), timecode 126.922s = 00:02:06.922)
|  + Frame with size 432
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.933s = 00:02:06.933)
|  + Frame with size 329
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.955s = 00:02:06.955)
|  + Frame with size 333
|+ Cluster
| + Cluster timecode: 126.954s
| + Cluster previous size: 97344
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 126.954s = 00:02:06.954)
|  + Frame with size 61239
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.975s = 00:02:06.975)
|  + Frame with size 338

After

| + SimpleBlock (track number 1, 1 frame(s), timecode 135.456s = 00:02:15.456)
|  + Frame with size 2260
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.468s = 00:02:15.468)
|  + Frame with size 332
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.490s = 00:02:15.490)
|  + Frame with size 335
|+ Cluster
| + Cluster timecode: 135.489s
| + Cluster previous size: 158758
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 135.490s = 00:02:15.490)
|  + Frame with size 88070
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.511s = 00:02:15.511)
|  + Frame with size 336
2020-02-21 12:49:28 +00:00
Stefano Buora
2d3dccdba7 rtspsrc: remove useless function calls
Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls.

The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed.

The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable.
2020-02-20 08:27:35 +00:00
Matthew Waters
1326fcdbcc rtpbin: fix typo setting max-dropout/misorder-time
we were setting the max-dropout-time to the value of the
max-misorder-time which by default has a factor of 30 difference in
value.
2020-02-20 13:46:06 +11:00
Seungha Yang
f286f30640 qtdemux: Parse VP Codec Configuration Box
The VP Codec Configuration Box (vpcC) contains vp9 profile and
colorimetry information. Especially the profile information might
be useful for downstream to select capable decoder element.
2020-02-19 23:18:51 +09:00
Yeongjin Jeong
e836640bd5 flvmux: Support rollover in timestamp
For live streams, if we keep the stream for a long time, the timestamp
will be larger than max_uint32. In that case, timestamp should be handled
as a rollover timestamp rather than a backward timestamp.
2020-02-18 18:39:31 +09:00
Havard Graff
63ae338c24 rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK
It could have been freed (rtp_timer_free) in the meantime.
2020-02-17 15:04:45 +01:00
Havard Graff
1df706448c rtpmanager: Google Transport-Wide Congestion Control RTP Extension
Generating and parsing the RTCP-messages described in:
https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
2020-02-14 10:09:02 +00:00
Håvard Graff
9ba9837058 rtpfunnel: various cleanups
* Organize GstRtpFunnelPad and GstRtpFunnel separately
* Use G_GNUC_UNUSED instead of (void) casts
* Don't call an event "caps"
* Use semicolons after GST_END_TEST (helps gst-indent)
2020-02-14 10:08:05 +00:00
Sebastian Dröge
9593a3679e qtdemux: Merge sample tables for raw audio streams with one container sample per audio sample
Instead of having chunks with one sample per raw audio sample, have
chunks with a single sample that contains lots of raw audio samples. If
necessary these are still split again later when reading the stream.

With this we are allocating a lot less memory for the parsed sample
tables and can play files that previously triggered our limit of 200MB
for the sample table. For example, one file here would previously
allocate 3.5GB for the sample table and now only allocates 70KB.
2020-02-14 08:48:01 +00:00
Sebastian Dröge
be1c97d3c9 qtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame
Outputting 48000 buffers per second is not a good idea performance-wise.
If a container sample is less than 1024 raw audio frames, combine
multiple samples to get at least 1024 raw audio samples as long as
they're stored contiguous in the file.

For the other direction, if a container sample contains more than 4096
samples there is already code for splitting them up.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692750
2020-02-14 08:48:01 +00:00
Mathieu Duponchelle
1471100f37 rtspsrc: fix requested range
When the server replies with a range "now-", it is presumed to
be a "live" stream and we should request a similar range.

This was the case prior to my refactoring to make use of
gst_rtsp_range_to_string in 5f1a732bc7,
this commit restores the behaviour for that case.
2020-02-12 05:47:54 +00:00
Mikhail Fludkov
57b0522cd7 rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
Refactoring to remove duplicate code and add test
2020-02-11 18:39:22 +00:00
Stian Selnes
629b71ac9c rtpptdemux: Fix debug to use GST_DEBUG_OBJECT 2020-02-11 18:39:22 +00:00
Mikhail Fludkov
851a2b7925 rtpbin: use max-streams on rtpssrcdemux
The proper way of capping on max-streams is to do it in rtpssrcdemux.
This patch uses the newly introduced property on rtpssrcdemux. Previous
behavior would not prevent rtpssrcdemux spawning new pads for every new
ssrc and potentialy causing performance trouble during teardown.
2020-02-11 15:12:07 +01:00
John Bassett
16d750bc01 rtpssrcdemux: Handle RTCP APP packets
Fix crash when processing RTCP APP packets.
2020-02-11 15:12:07 +01:00
John Bassett
5800950a2d rtpssrcdemux: Bad RTP/RTCP packet is not fatal
When used for processing bundled media streams within rtpbin the rtpssrcdemux element may
receive bad RTP and RTCP packets, these should not be treated as a fatal error.
2020-02-11 15:10:12 +01:00
Mikhail Fludkov
35596e7fac rtpssrcdemux: introduce max-streams property
The property is useful against atacks when the sender changes SSRC for
every RTP packet. The property with the same name introduced in rtpbin
was not enough, because we still can end up with thousands of pads
allocated in rtpssrcdemux.
2020-02-11 15:10:12 +01:00
Alexander Lapajne
54c4ba82f8 rtspsrc: Fix for segmentation fault when handling set/get_parameter requests
gstrtspsrc uses a queue, set_get_param_q, to store set param and get
param requests. The requests are put on the queue by calling
get_parameters() and set_parameter(). A thread which executs in
gst_rtspsrc_thread() then pops requests from the queue and processes
them. The crash occured because the queue became empty and a NULL
request object was then used. The reason that the queue became empty
is that it was popped even when the thread was NOT processing a get
parameter or set parameter command. The fix is to make sure that the
queue is ONLY popped when the command being processed is a set
parameter or get parameter command.
2020-02-10 09:43:17 +01:00
Olivier Crête
c00796eaa5 rtpsession: Add test for packet rate maths 2020-02-06 14:01:38 -05:00
olivier.crete@collabora.com
774ddb15b8 rtpstats: Base the packet rate average on the packet rate itself
Do this so that the average update speed is in time instead of varying
based on the actual packet arrival rate.
2020-02-06 14:00:48 -05:00
olivier.crete@collabora.com
a637ec3da8 rtpstats: Don't save the ts & seqnum if the avg is not updated
This makes it update correctly when you have more than one packet per
frame.
2020-02-06 14:00:48 -05:00
Sebastian Dröge
f6e383b749 splitmuxsink: Include actual sink element in the fragment-opened/closed messages
If not configuring the sinks via the "location" property this can be
useful to know for which sink the fragment was actually opened/closed,
especially if finalization of the fragments is happening asynchronously.
2020-01-29 13:30:00 +00:00
Juergen Werner
755dba4561 rtpjitterbuffer: fix scaling from RTP-time to NTP-time
The scaling was inverse.
2020-01-29 12:05:07 +01:00
Mathieu Duponchelle
a245e85fb1 rtprtxsend: allow generic input caps
When connected to an upstream rtpfunnel element, payload-type,
ssrc and clock-rate will not be present in the received caps.

rtprtxsend can already deal with only the clock rate being
present there, a new property is exposed to allow users to
provide a payload-type -> clock-rate map, this enables the
use of the max-size-time property for bundled streams.
2020-01-28 15:44:13 +00:00
Sebastian Dröge
eb0b676fae splitmuxsink: Check the correct sink class for the existence of the "location" property 2020-01-27 15:53:40 +02:00
Sebastian Dröge
5877d945a4 qtdemux: Always prefer information from v1/v2 sound sample description over sample description entry
ffmpeg is doing the same and various files in the wild have bogus
information in the sample description if the same information is also
duplicated afterwards in the v1/v2 sound sample desription.

Previously we only did this for non-raw audio due to
  https://bugzilla.gnome.org/show_bug.cgi?id=374914
but this specific file is already worked around differently. It still
works after this change.

Also remove ad-hoc GST_READ_DOUBLE_BE re-implementation and move the
switch for legacy audio formats after reading all the sample
descriptions as we want to override the values from there.
2020-01-27 14:14:50 +02:00
Sebastian Dröge
c4f6ce789d avimux: Add support for >2 raw audio channels
For this case write a WAVEFORMATEXTENSIBLE header and also reorder the
raw audio channels to the AVI channel order if needed.
2020-01-19 12:09:38 +00:00
Sebastian Dröge
451fc5c112 wavenc: Fix writing of the channel mask with >2 channels
The channel position is an enum but the conversion code assumed it's a
mask. Convert accordingly.
2020-01-13 19:50:06 +00:00
Kristofer Björkström
9c86414279 rtph265pay: TID for NALU type 48 was always set to 7
A typo bug: | instead of & resulted in TID alwasy being set to 7
for the aggregated NALU of type 48
2020-01-13 15:41:30 +01:00
Sebastian Dröge
c17d5e36ad imagefreeze: Add support for replacing the output buffer
By default imagefreeze will still reject new buffers after the first one
and immediately return GST_FLOW_EOS but the new allow-replace property
allows to change this.

Whenever updating the buffer we now also keep track of the configured
caps of the buffer and from the source pad task negotiate correctly
based on the potentially updated caps.

Only the very first time negotiation of a framerate with downstream is
performed, afterwards only the caps themselves apart from the framerate
are updated.
2020-01-11 08:04:43 +00:00
Alicia Boya García
8dd42666e3 qtdemux: Fix race on pad reconnection
Elements emitting frames through several srcpads should use a
flow combiner to aggregate the chain returns and therefore only return
GST_FLOW_NOT_LINKED to upstream when all the downstream pads have
received GST_FLOW_NOT_LINKED.

In addition to that, in order to handle pads being relinked downstream,
the flow combiner should be reset in response to RECONFIGURE events.
This ensures that a both srcpads process a chain operation before a
GST_FLOW_NOT_LINKED can be propagated upstream (which would usually stop
the pipeline).

Otherwise, in a configuration with two srcpads, only one linked at a
time, after the relink the element could chain data through the now
unlinked pad and the flow combiner would resolve as GST_FLOW_NOT_LINKED
(stopping the pipeline) just because the now linked pad has not been
chained yet to update the flow combiner.

This patch adds handling of RECONFIGURE events to qtdemux. Also, since
this event handling causes the flow combiner to be used from a thread
other than the qtdemux streaming thread, usages of the flow combiner
has been guarded by the object lock.
2020-01-09 18:43:02 +00:00
Seungha Yang
8445685a21 splitmuxsink: Fix assertion failure on set_property()
GValue might have null object.

(gst-inspect-1.0:10304): GStreamer-CRITICAL ...
    gst_object_ref_sink: assertion 'object != NULL' failed
2020-01-07 01:24:01 +09:00
Daniel Molkentin
bb1ce82e39 videocrop: allow properties to be animated by GstController 2020-01-03 15:16:02 +01:00
Aaron Boxer
09d4514814 rtspsrc: improved handling of control concatenation with base
Also, `control_url` variable has been renamed to `control_path`,
as it is actually a path.
2019-12-30 16:52:45 +00:00
Aaron Boxer
ed6b5a3a63 rtspsrc: append aggregate control string to base URL before query string
Appending control string to end of query changes meaning of query string
Fixes #650
2019-12-30 16:52:45 +00:00
Niels De Graef
acab06b2e8 alpha: Cleanup using G_DECLARE_FINAL_TYPE
We started depending on GLib 2.44, so we can clean up all the GObject
boilerplate macros.
2019-12-28 04:05:13 +00:00
Stéphane Cerveau
b928517f1e good: use of g_value_dup_string
Use helper method to get string from GValue.
2019-12-20 09:30:26 +00:00
Havard Graff
8b96d8ee8d rtpbin: fix shutdown crash in rtpbin
The key is to make sure the jitterbuffer is set to NULL *before* the
ptdemux.

The race that existed would basically happen when ptdemux had reached
READY, and the jitterbuffer would then push a buffer, triggering a new
pad with a new payloadtype being added and ghosted to the rtpbin itself.

However, the srcpad of the ptdemux would now be inactive, and all the
sticky-event pushed on it would be swallowed, not allowing any to reach
the ghost-pad. Then the buffer in-flight would come to the ghostpad,
and we would assert that a buffer arrived before the necessary
events.

By simply re-ordering the state-changes, we ensure that there will be
no buffer racing into the ptdemux while its state is being changed,
and the problem disappears completely.

Notice also that there is not point in disconnecting the signals on the
ptdemux before this point, since we need the push-thread to settle
down before we can do this in a non-racy way.
2019-12-20 08:27:07 +00:00
Aaron Boxer
4155c59cc4 rtspsrc: avoid seek DISCONT when only rate changes in same direction
Not setting DISCONT avoids a noticable delay when seeking
with only rate changing, in the same direction as current
rate.
2019-12-19 05:54:38 +00:00
Olivier Crête
9db1d740e8 rtspsrc: Remove deprecated GTimeVal
GTimeVal won't work past 2038
2019-12-18 19:48:34 +00:00
Sebastian Dröge
04806a75bd avimux: Add support for S24LE and S32LE raw audio
avidemux already handles this correctly.
2019-12-18 11:16:30 +00:00
Sebastian Dröge
4dbaff424f avimux: Allow muxing v210 video into AVI
avidemux already handles this.
2019-12-18 10:20:25 +00:00
Vivia Nikolaidou
7cbc351e05 flvdemux: Don't replace video codec data when we receive a PAR
Receiving a pixel-aspect-ratio should trigger a caps change, but not
replace the existing video codec tag
2019-12-16 21:51:38 +00:00
Mathieu Duponchelle
5766731bd4 qtmux: protect access to GstElement.sinkpads 2019-12-16 14:17:38 +00:00
Mathieu Duponchelle
e2462005fb qtmux: port to GstAggregator 2019-12-16 14:17:38 +00:00
Joakim Johansson
4d7d577496 gstrtspsrc: Add missing lock on free set_get_param_q
Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter.
2019-12-16 13:13:00 +01:00
Sebastian Dröge
9f6ed9ec72 splitmuxsink: Increment fragment_id even if no fragment location was provided
Applications might handle locations and generally configuration of the
sink by themselves instead of having splitmuxsink set the location on
the sink. Nonetheless it makes sense to increment the fragment_id that
is passed to the signal so that applications know which fragment is
requested.
2019-12-13 22:59:55 +00:00
Jan Alexander Steffens (heftig)
9e0eb77810
flvmux: Use the last DTS for the metadata timestamp
This avoids creating a timestamp regression during a stream.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/429
2019-12-12 11:09:31 +01:00
Mathieu Duponchelle
625eb00c06 qtdemux: send GAP events for lagging audio and video streams too
The logic is taken straight from matroskademux, see
77403d0afe
2019-12-11 19:59:13 +00:00
Seungha Yang
5009cad220 flvmux: Use thread-safe gmtime_r if available
gmtime on *nix is not thread-safe.
2019-12-10 23:48:35 +09:00
Stéphane Cerveau
b44d37a338 splitmuxsink: provides a start-index property
Allow to change the fragment-id start index.
2019-12-05 14:58:40 +00:00
Tim-Philipp Müller
1df530eaa7 rtpjpegdepay: outputs framed jpeg
Add parsed=true to output caps, as we always output
whole frames, timestamped and all. Means also that
the output can be decoded by avdec_mjpeg wihout
plugging an extra parser (which has no rank).
2019-12-04 13:02:54 +00:00
Jan Alexander Steffens (heftig)
06600b2cd9
flvmux: Correct metadata handling in file and stream mode
In file mode, only push one onMetaData at the start of the stream.

In stream mode, always push complete onMetaData. They get replaced, not
merged.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418
2019-12-03 14:01:19 +01:00
Jan Alexander Steffens (heftig)
6fdb6ece6e
flvmux: Don't calculate duration in streamable mode
There's no header to rewrite, so the duration is left unused.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418
2019-12-03 14:01:14 +01:00
Havard Graff
a7c887b197 rtpL16depay: don't crash if data is not modulo channels*width 2019-12-03 00:02:48 +00:00
Havard Graff
690c15bd78 rtpopuspay: use baseclass allocator for buffers
That way we get some of the meta -> rtp-extension goodies.
2019-12-02 13:05:12 +01:00
Havard Graff
f997859913 rtpsession: add locking for clear-pt-map
...or it will segfault from time to time...
2019-11-29 14:23:49 +01:00
Linus Svensson
08060dd97b matroskamux: Add property to set DateUTC
Add a property that makes it possible for an application to set the
DateUTC header field in matroska files. This is useful for live feeds,
where the DateUTC header can be set to a UTC timestamp, matching the
beginning of the file.

Needs gstreamer!323

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/481
2019-11-25 14:01:48 +01:00
Linus Svensson
0690bd1b21 matroskamux: Use nanosecond precision for DateUTC
DateUTC is specified with nanosecond precision in matroska, make use of
that.
2019-11-22 16:30:50 +01:00
Jan Alexander Steffens (heftig)
1e7d2e2bbd
matroskamux: Pass the right size to gst_collect_pads_add_pad
We were lucky that GstMatroskamuxPad is larger than GstMatroskaPad.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/393
2019-11-19 14:57:11 +01:00
aogun
a6e28ca268 aacparse: fix wrong offset of adts channel 2019-11-18 01:06:41 +00:00
Seungha Yang
a441779d39 splitmuxsink: Don't take lock during posting message
An application might try to access splitmuxsink from sync message handler
by g_object_{get,set} which takes lock also. In general, we don't
take lock around message handler.
2019-11-18 00:08:36 +00:00
Niels De Graef
7cf4ab6229 Don't pass default GLib marshallers for signals
By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-11-17 15:32:30 +00:00
Nicolas Dufresne
db187eec19 rtpjitterbuffer: Check the exit condition after executing timers
The do_expected_timeout() function may release the JBUF_LOCK, so we need
to check if nothing wanted the timer thread to exit after this call.
The side effect was that we may endup going back into waiting for a timer
which will cause arbitrary delay on tear down (or deadlock when test
clock is used).

Fixes #653
2019-11-14 17:52:16 -05:00
Nicolas Dufresne
fd6cd6f545 rtpjitterbuffer: Check exit condition immediately after JBUF_WAIT
JBUF_WAIT_QUEUE drops the JBUF_LOCK, which means the stop condition
for the chain function may have changed (change_state to NULL). Check
this immediately after the wait so that we don't delay shutting down.
2019-11-14 17:51:31 -05:00
Nicolas Dufresne
e66a4b64b3 videocrop: Also update the coordinate when in-place
This update is needed when the output caps is not changed (e.g. we are
moving a viewport around).

Fixes #669
2019-11-12 17:28:22 -05:00
Nicolas Dufresne
98a5726eba videocrop: Don't always re-run the allocation query
When in-place, running an allocation is not useful since videocrop
is not implicated in the allocation. So only force the allocation
query for the case it was in passthrough. This is needed since the
change in the crop region will likely pull us out of this mode. For the
case we where neither in passthrough or in-place, the allocation query
is already ran by the baseclass, so nothing special is needed.

This fixes performance issues when changing the crop region per frame.
This was reproduced using videocrop2-test.
2019-11-11 16:05:24 -05:00
Nicolas Dufresne
e09b4e9cde videocrop: Cleanup spurious assignment
These are just writing the same thing a second time.
2019-11-11 14:09:47 -05:00
Stéphane Cerveau
9dc1a32d5a splitmuxsink: add fakesink support
fakesink does not support "location" property and was generating
a warning.
2019-11-07 12:28:58 +01:00
Sergey Nazaryev
b4b79a211f multiudpsink: don't lose scope_id 2019-11-05 23:50:11 +00:00
Havard Graff
87457a862d rtpjitterbuffer: make sure not to drop packets based on skew
One of the jitterbuffers functions is to try and make sense of weird
network behavior.

It is quite unhelpful for the jitterbuffer to start dropping packets
itself when what you are trying to achieve is better network resilience.

In the case of a skew, this could often mean the sender has restarted
in some fashion, and then dropping the very first buffer of this "new"
stream could often mean missing valuable information, like in the case
of video and I-frames.

This patch simply reverts back to the old behavior, prior to 8d955fc32b
and includes the simplest test I could write to demonstrate the behavior,
where a single packet arrives "perfectly", then a 50ms gap happens,
and then two more packets arrive in perfect order after that.

# Conflicts:
#	tests/check/elements/rtpjitterbuffer.c
2019-11-02 23:05:32 +00:00
Patricia Muscalu
203ad39d53 qtmux: Fix memory leak while pushing fragmented data
The memory leak occurs in the case when the buffer has been
added to the fragment_buffers array of the current pad and
never been sent because of the push failure of the previous
buffers: moof or mdat header or fragmented buffer(s).
2019-10-24 10:21:11 +00:00
Edward Hervey
8e1c224fbc good: Avoid usage of deprecated API
GTimeval and related functions are now deprecated in glib.
Replacement APIs have been present since 2.26
2019-10-16 07:46:58 +00:00
Tim-Philipp Müller
c9a47c0c8d Remove autotools build system 2019-10-14 11:04:18 +01:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +00:00
Simon Arnling Bååth
8173596ed2 gstrtpjitterbuffer: Custom messages when dropping packets
This commit adds custom element messages for when gstrtpjitterbuffer
drops an incoming rtp packets due to for example arriving too late.
Applications can listen to these messages on the bus which enables
actions to be taken when packets are dropped due to for example high
network jitter.

Two properties has been added, one to enable posting drop messages and
one to set a minimum time between each message to enable throttling the
posting of messages as high drop rates.
2019-10-04 20:31:56 +00:00
Thibault Saunier
a55576d1fd qtdemux: Specify REDIRECT information in error message
There are in the wild (mp4) streams that basically contain no tracks
but do have a redirect info[0], in which case, we won't be able
to expose any pad (there are no tracks) so we can't post anything but
an error on the bus, as:

- it can't send EOS downstream, it has no pad,
- posting an EOS message will be useless as PAUSED state can't be
  reached and there is no sink in the pipeline meaning GstBin will
  simply ignore it

The approach here is to to add details to the ERROR message with a
`redirect-location` field which elements like playbin handle and use right
away.

[0]: http://movietrailers.apple.com/movies/paramount/terminator-dark-fate/terminator-dark-fate-trailer-2_480p.mov
2019-09-30 12:15:43 -03:00
Olivier Crête
a24596423a rtpjitterbuffer: Cancel timers instead of just unlocking loop thread
When the queue is full (and adding more packets would risk a seqnum
roll-over), the best approach is to just start pushing out packets
from the other side.  Just pushing out the packets results in the
timers being left hanging with old seqnums, so it's safer to just
execute them immediately in this case. It does limit the timer space
to the time it takes to receiver about 32k packets, but without
extended sequence number, this is the best RTP can do.

This also results in the test no longer needed to have timeouts or
timers as pushing packets in drives everything.

Fixes #619
2019-09-28 07:47:54 -04:00
Nicolas Dufresne
4a9f42430a rtpjitterbuffer: Optimize offset update
As we are applying the same offset over all timers, there timer
ordering won't change, so we can safely skip time-reordering.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
af1c586c7b rtptimerqueue: Optimize reschedule optations
This basically add ability to choose between inserting from head, tail
or in-place in order to try and minimize the distance to walk through in
the timer queue. This removes an overhead we had seen on high drop rate.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
1897c1fbe6 rtpjitterbuffer: Fix a typo in comment 2019-09-27 17:34:04 -04:00
Nicolas Dufresne
9ebcadb349 rtpjitterbuffer: Don't use stats timer on the timers queue
The timer passed to update_timers may be from the stats timer. At the
moment, we could endup rescheduling (reusing) that timer onto the normal
timer queue, unschedul it as if it was from the normal timer queue or
duplicate it into the stats timer queue again. This was protected before
as the with the fact the stats timer didn't have a valid idx.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
81bffb5e5c rtpjitterbuffer: Update timers on ts-offset changes
As the offset is already applied now, we need to update and reschedule
all timers each time the offset is changed. I'm not sure who expect this
to be retro-actively applied, but there was a unit test for it.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
d4c6c335c5 rtpjitterbuffer: No need to wake the timer thread on head changes
If the jitterbuffer head change, there is no need to systematically
wakeup the timer thread. The timer thread will be waken up on if
an earlier timeout has been pushed. This prevent some more spurious
wakeup when the system is loaded. As a side effect, cranking the clock
may set the clock at an earlier position.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
36771b75e9 rtpjittterbuffer: Port timers array to RtpTimerQueue
In this patch we now make use of the new RtpTimerQueue instead of the
old GArray. This required a lot of changes all over the place, some of
the important changes are that `timer->timeout` is no longer a PTS but
the actual timeout. This was required to get the RtpTimerQueue sorting
right. The applied offset is saved as `timer->offset`, this allow
retreiving back the PTS when needed.

The clockid updates only happens once per incoming packet. If the
currently schedule timer is before the earliest timer in the queue, we
no longer wakeup the thread. This way, if other timers get setup in the
meantime, this will reduce the number of wakup.

The timer loop code has been mostly rewritten, though the behaviour of
running the lost timers first has been kept (even though there is no
test to show what would be the side effect of doing this differently).

Fixes #608
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
d4b2231de2 rtpjittterbuffer: Port from TimerQueue to RtpTimerQueue 2019-09-27 17:34:04 -04:00
Nicolas Dufresne
f5e3280dbe rtpjitterbuffer: Port use the new RtpTimer structure
First iteration toward porting to the new timer queue.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
37742cd36d rtptimerqueue: Consolidate a data structure for timers
Implement a single timer queue for all timers. The goal is to always use
ordered queues for storing timers. This way, extracting timers for
execution becomes O(1). This also allow separating the clock wait
scheduling from the timer itself and ensure that we only wake up the
timer thread when strictly needed.

The knew data structure is still O(n) on insertions and reschedule,
but we now use proximity optimization so that normal cases should be
really fast. The GList structure is also embeded intot he RtpTimer
structure to reduce the number of allocations.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
c917f11ae8 rtpjitterbuffer: Move item structure outside of the element
This moves the RtpJitterBufferStructure type, alloc, free into
rtpjitterbuffer.c/h implementation. jitterbuffer.c strictly rely on
the fact this structure is compatible with GList, and so it make more
sense to keep encapsulate it. Also, anything that could possibly
reduce the amount of code in the element is a win.

In order to support that move, a function pointer to free the data
was added. This also allow making the free function option when
flushing the jitterbuffer.
2019-09-27 13:02:16 -04:00
Nicolas Dufresne
9b706b6220 rtpjitterbuffer: Constify timer pointers where possible
This helps understanding which function modify the Timerdata
and which one does not. This is not always obvious from thelper
name considering recalculate_timer() does not.
2019-09-27 13:02:16 -04:00
Mathieu Duponchelle
b5e414cdc2 rtpbin: add request-jitterbuffer signal
This can be used to pass the threadsharing jitterbuffer from
gst-plugins-rs for example.
2019-09-24 15:33:21 +00:00
Matthew Waters
5ffd733317 build: fix werror build with newer gcc
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:55,
                 from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/tag/tag.h:25,
                 from ../gst/isomp4/qtdemux.c:56:
In function ‘qtdemux_inspect_transformation_matrix’,
    inlined from ‘qtdemux_parse_trak’ at ../gst/isomp4/qtdemux.c:10676:5,
    inlined from ‘qtdemux_parse_tree’ at ../gst/isomp4/qtdemux.c:14210:5:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:645:5: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
  645 |     gst_debug_log ((cat), (level), __FILE__, GST_FUNCTION, __LINE__, \
      |     ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  646 |         (GObject *) (object), __VA_ARGS__);    \
      |         ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:1062:35: note: in expansion of macro ‘GST_CAT_LEVEL_LOG’
 1062 | #define GST_DEBUG_OBJECT(obj,...) GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_DEBUG,   obj,  __VA_ARGS__)
      |                                   ^~~~~~~~~~~~~~~~~
../gst/isomp4/qtdemux.c:10294:5: note: in expansion of macro ‘GST_DEBUG_OBJECT’
10294 |     GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s",
      |     ^~~~~~~~~~~~~~~~
../gst/isomp4/qtdemux.c: In function ‘qtdemux_parse_tree’:
../gst/isomp4/qtdemux.c:10294:64: note: format string is defined here
10294 |     GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s",
      |                                                                ^~
2019-09-23 18:46:16 +10:00
Sebastian Dröge
d7738da285 qtmux: Use the new helper functions for mapping the colr atom values to colorimetry 2019-09-18 18:32:02 +03:00
Sebastian Dröge
5d4a46aa63 qtdemux: Use the new helper functions for mapping the colr atom values to colorimetry 2019-09-18 18:29:27 +03:00
Mathieu Duponchelle
eeccb330d0 smpte: don't register transition types twice 2019-09-10 20:52:17 +00:00
Doug Nazar
42dea672fa alpha: Fix one_over_kc calculation
On arm/aarch64, converting from float directly to unsigned int uses
a different opcode and negative numbers result in 0. Cast to
signed int first.
2019-09-09 00:51:53 -04:00
Jan Schmidt
31be44c47f splitmux: Add muxer-pad-map property
Add a property which explicitly maps splitmuxsink pads to the
muxer pads they should connect to, overriding the implicit logic
that tries to match pads but yields arbitrary names.
2019-09-06 12:38:56 +00:00
Jan Schmidt
8ec695e55d splitmuxsink: In async mode, retain previous muxer pad names.
When running in async-finalize mode, request new pads from the muxer
using the same names as old pads, instead of letting the muxer assign
new ones based on the pad template name.
2019-09-06 12:38:56 +00:00
Jan Schmidt
83ef7a6d1c splitmuxsink: Mark split-* signals as action signals. Doc fixes.
Add the G_SIGNAL_ACTION flag to the split-* signals on splitmuxsink,
and make some improvements to their docstrings
2019-09-06 12:38:56 +00:00
Seungha Yang
2ef74f2c81 qtmux: Fix incompatible type warning with MSVC
gstqtmux.c(5582): warning C4133: 'function':
  incompatible types - from 'GstVideoMultiviewFlags *' to 'guint *'
2019-09-02 15:07:17 +00:00
Mathieu Duponchelle
c5e8a8f320 rtspsrc: fix git diff indentation 2019-09-02 16:33:05 +02:00
Mathieu Duponchelle
3bc5d3d3b5 rtspsrc: normalize variable to boolean 2019-08-30 22:42:58 +02:00
Mathieu Duponchelle
37eca8a12c rtspsrc: clip output segment on accurate seeks
The output segment is only used in ONVIF mode.

The previous behaviour was to output a segment computed from
the Range response sent by the server.

In ONVIF mode, servers will start serving from the appropriate
synchronization point (keyframe), and the Range in response will
start at that position.

This means rtspsrc can now perform truly accurate seeks in that
mode, by clipping the output segment to the values requested in
the seek. The decoder will then discard out of segment buffers
and playback will start without artefacts at the exact requested
position, similar to the behaviour of a demuxer when an accurate
seek is requested.
2019-08-30 14:50:21 +00:00
Mathieu Duponchelle
3429ddde38 docstrings: port ulinks to markdown links 2019-08-23 18:56:01 +02:00
Tim-Philipp Müller
0dc9e5bff8 replaygain: fix up doc links to defunct replaygain.org website
Fixes #624
2019-08-23 13:12:39 +03:00
Amr Mahdi
cbe61c4ff5 wavparse: Fix push mode ignoring audio with a size smaller than segment buffer
In push mode (streaming), if the audio size is smaller than segment buffer size, it would be ignored.
This happens because when the plugin receives an EOS signal while a single audio chunk that is less than the segment buffer size is buffered, it does not
flush this chunk. The fix is to flush the data chunk when it receives an EOS signal and has a single (first) chunk buffered.

How to reproduce:
1. Run gst-launch with tcp source
```
gst-launch-1.0  tcpserversrc port=3000 !  wavparse ignore-length=0 ! audioconvert ! filesink location=bug.wav
```
2. Send a wav file with unspecified data chunk length (0). Attached a test file
```
cat test.wav | nc localhost 3000
```
3. Compare the length of the source file and output file
```
ls -l test.wav bug.wav
-rw-rw-r-- 1 amr amr    0 Aug 15 11:07 bug.wav
-rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
```

The expected length of the result of the gst-lauch pipeline should be the same as the test file minus the headers (44), which is ```3564 - 44 = 3520``` but the actual output length is ```0```

After the fix:
```
ls -l test.wav fix.wav
-rw-rw-r-- 1 amr amr 3520 Aug 15 11:09 fix.wav
-rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav
```
2019-08-19 07:30:17 +00:00
Sebastian Dröge
2a4d0a9b09 rtpvp8depay: Add property for waiting until the next keyframe after packet loss
If VP8 is not encoded with error resilience enabled then any packet loss
causes very bad artefacts when decoding and waiting for the next
keyframe instead improves user experience considerably.
2019-08-12 17:10:20 +00:00
Mart Raudsepp
67958ccce8 matroska: Provide audio lead-in for some lossy formats
Various audio formats require an audio lead-in to decode it properly.
Most parsers would take care of it, but when a container like matroska is
involved, the demuxer handles the seeking and without its own lead-in
handling would never even pass the lead-in data to the parser.
This commit provides an initial implementation of that for audio/mpeg,
audio/x-ac3 and audio/x-eac3 by calculating the worst case lead-in time
needed from known samplerate, potential lead-in frames need and the
maximum blocksize possible for the format (as we don't parse that out
exactly in matroskademux) and seeking that much earlier in case of
accurate seeks. This is especially important for NLE use-cases with GES.

If accurate seeking to a position that happens to have a video keyframe,
it'll go back to the previous keyframe than needed, but with typical
video files that's the best we can do anyway without falling back to
scanning the clusters, as typically only keyframes are indexed in
Cueing Data.
If the media doesn't have a CUE, then we bisect for the cluster to seek
to with the same modified time as well in case of accurate seeking,
ensuring sufficient lead-in. This code path is typically hit only with
(suboptimal) audio-only matroska files, e.g. when created with ffmpeg,
which doesn't add a CUE for audio-only mkv muxing.
2019-08-07 18:51:57 -04:00
Antonio Ospite
8dd03042cc rtpsession: add support for buffer lists on the recv path
The send path in rtpsession processes the buffer list along the way,
sharing info and stats between packets in the same list, because it
assumes that all packets in a buffer list are from the same frame.

However, in the receiving path packets can arrive in all sorts of
arrangements:

  - different sources,
  - different frames (different timestamps),
  - different types (multiplexed RTP and RTCP, invalid RTP packets).

so a more general approach should be used to correctly support buffer
lists in the receive path.

It turns out that it's simpler and more robust to process buffers
individually inside the rtpsession element even if they come in a buffer
list, and then reassemble a new buffer list when pushing the buffers
downstream.

This avoids complicating the existing code to make all functions
buffer-list-aware with the risk of introducing regressions,

To support buffer lists in the receive path and reduce the "push
overhead" in the pipeline, a new private field named processed_list is
added to GstRtpSessionPrivate, it is set in the chain_list handler and
used in the process_rtp callback; this is to achieve the following:

  - iterate over the incoming buffer list;
  - process the packets one by one;
  - add the valid ones to a new buffer list;
  - push the new buffer list downstream.

The processed_list field is reset before pushing a buffer list to be on
the safe side in case a single buffer was to be pushed by upstream
at some later point.

NOTE:

The proposed modifications do not change the behavior of the send path.

The process_rtp callback is called in rtpsource.c by the push_rtp
callback (via source_push_rtp) only when the source is not internal.

So even though push_rtp is also called in the send path, it won't end up
using process_rtp in this case because the source would be internal in
the send path.

The reasoning from above may suggest a future refactoring: push_rtp
might be split to better differentiate the send and receive path.
2019-08-07 15:32:30 -04:00
Doug Nazar
b0534c65d1 matroska: Handle interlaced field order 2019-08-07 14:12:32 +00:00
Amr Mahdi
5f01b9da05 wavparse: Fix ignoring of last chunk in push mode
In push mode (streaming), if the last audio payload chunk is less than the segment rate buffer size, it would be ignored since the plugin waits until it has at least segment rate bufer size of audio.

The fix is to introduce a flushing flag that indicates that no more audio will be available so that the plugin can recognize this condition and flush the data is has even if it is less
than the desired segment rate buffer size.
2019-08-07 12:09:46 +00:00
luke.lin
d6ae59c32d qtdemux: enlarge the maximal atom size
For 8K content, frame size is over 25MB, and cause the negotiation failure.
Enlarge the limitation of QTDEMUX_MAX_ATOM_SIZE to 32MB.
2019-08-07 02:46:20 +00:00
Mathieu Duponchelle
5c7423d73c rtspsrc: expose and implement is-live property
This is useful to support the ONVIF case: when is-live is set to
FALSE and onvif-rate-control is no, the client can control the
rate of delivery and arrange for the server to block and still
keep sending when unblocked, without requiring back and forth
PAUSE / PLAY requests. This enables, amongst other things, fast
frame stepping on the client side.

When is-live is FALSE, we don't use a manager at all. This case
was actually already pretty well handled by the current code. The
standard manager, rtpbin, is simply no longer needed in this case.

Applications can instantiate a downloadbuffer after rtspsrc if
needed.
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
75f53631e5 rtspsrc: reset_time when flush stopping 2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
5f1a732bc7 rtspsrc: expose and implement onvif-mode property
Refactor the code for parsing and generating the Range, taking
advantage of existing API in GstRtspTimeRange.

Only use the TCP protocol in that mode, as per the specification.

Generate an accurate segment when in that mode, and signal to the
depayloader that it should not generate its own segment, through
the "onvif-mode" field in the caps, see
<https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/328>
for more information.

Translate trickmode seek flags to their ONVIF representation

Expose an onvif-rate-control property
2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
544f8fecf4 rtspsrc: improve handling of rate in seeks 2019-08-06 22:45:37 +00:00
Mathieu Duponchelle
e18d5d6ec6 rtpfunnel: forward correct segment when switching pad
Forwarding a single segment event from the pad that first gets
chained is incorrect: when that first event was sent by an element
such as x264enc, with its offset start, we end pushing out of segment
buffers for the other pad(s).

Instead, everytime the active pad changes, forward the appropriate
segment event.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028
2019-08-06 14:02:50 +00:00
Sebastian Dröge
86ec5c1031 rtspsrc: Use new GstRTSPMessage API to set message body from a buffer directly 2019-08-05 19:35:36 +03:00
Antonio Ospite
ae48646d8e rtpsource: fix receiver source stats to consider previously queued packets
When it is not clear yet if a packet relative to a source should be
pushed, the packet is put into a queue, this happens in two cases:

  - the source is still in probation;
  - there is a large jump in seqnum, and it is not clear what
    the cause is, future packets will help making a guess.

In either case stats about received packets are not updated at all; and
even if they were, when init_seq() is called it resets all receiver
stats, effectively loosing any possible stat about previously received
packets.

Fix this by taking into account the queued packets and update the stats
when calling init_seq().
2019-08-02 17:22:51 +02:00
Antonio Ospite
cf0ffd8693 rtpsource: clarify meaning of the octets-sent and octets-received stats
The octets-send and octets-received stats count the payload bytes
excluding RTP and lower level headers, clarify that in the
documentation.
2019-08-02 17:22:51 +02:00
Antonio Ospite
821994240e rtpsource: expose field bytes_received in RTPSourceStats
Since commit c971d1a9a (rtpsource: refactor bitrate estimation,
2010-03-02) bytes_received filed in RTPSourceStats is set but then never
used again, expose it so that it can be used  by user code to verify how
many bytes have been received.
2019-08-02 17:22:51 +02:00
Antonio Ospite
9d800cad43 rtpmanager: consider UDP and IP headers in bandwidth calculation
According to RFC3550 lower-level headers should be considered for
bandwidth calculation.

See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4:

  Bandwidth calculations for control and data traffic include
  lower-layer transport and network protocols (e.g., UDP and IP) since
  that is what the resource reservation system would need to know.

Fix the source data to accommodate that.

Assume UDPv4 over IP for now, this is a simplification but it's good
enough for now.

While at it define a constant and use that instead of a magic number.

NOTE: this change basically reverts the logic of commit 529f443a6
(rtpsource: use payload size to estimate bitrate, 2010-03-02)
2019-08-02 17:22:51 +02:00
Seungha Yang
4146dc905d qtdemux: Use empty-array safe way to cleanup GPtrArray
Fix assertion fail
GLib-CRITICAL **: g_ptr_array_remove_range: assertion 'index_ < rarray->len' failed
2019-08-02 12:32:59 +09:00
Marc Leeman
d365c4fdf9 rtpmp4vpay: config-interval -1 send at idr
adjust/port from rtph264pay and allow sending the configuration data at
every IDR

The payloader was stripping the configuration data when the
config-interval was set to 0. The code was written in such a way !(a >
0) that it stripped the config when it was set at -1 (send config_data
as soon as possible).

This resulted in some MPEG4 streams where no GOP/VOP-I was detected to
be sent out without configuration.
2019-08-01 14:28:04 +00:00
Doug Nazar
5451e4e900 matroskademux: Ignore crc32 element while peeking at cluster. 2019-07-27 14:21:34 -04:00
Mathieu Duponchelle
4830bbe6ca qtdemux: fix reverse playback EOS conditions
In reverse playback, we don't want to rely on the position of the current
keyframe to decide a stream is EOS: the last GOP we push will start with
a keyframe, which position is likely to be outside of the segment.

Instead, let the normal seek_to_previous_keyframe mechanism do its job,
it works just fine.
2019-07-26 02:42:11 +00:00
Mathieu Duponchelle
104f459258 qtdemux: fix key unit seek corner case
If a key unit seek is performed with a time position that matches
the offset of a keyframe, but not its actual PTS, we need to
adjust the segment nevertheless.

For example consider the following case:

* stream starts with a keyframe at 0 nanosecond, lasting 40 milliseconds
* user does a key unit seek at 20 milliseconds
* we don't adjust the segment as the time position is "over" a keyframe
* we push a segment that starts at 20 milliseconds
* we push a buffer with PTS == 0
* an element downstream (eg rtponviftimestamp) tries to calculate the
  stream time of the buffer, fails to do so and drops it
2019-07-26 01:50:47 +00:00
Knut Andre Tidemann
dbd7234191 rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps.
The src caps were never dereferenced, causing a memory leak.
2019-07-22 10:33:41 +02:00
Mathieu Duponchelle
5fde140e6e qtdemux: implement support for trickmode interval
When the seek event contains a (newly-added) trickmode interval,
and TRICKMODE_KEY_UNITS was requested, only let through keyframes
separated with the required interval
2019-07-18 17:54:43 +02:00
Seungha Yang
aa0544ab8f matroska: Port to color_{primaries,transfer,matrix}_to_iso
... and remove duplicated code.
2019-07-15 23:25:53 +09:00
Jan Schmidt
436d33b288 splitmuxsink: add the ability to mux auxilliary video streams
The primary video stream is used to select fragment cut points
at keyframe boundaries. Auxilliary video streams may be
broken up at any packet - so fragments may not start with a keyframe
for those streams.
2019-07-15 11:46:36 +00:00
Jan Schmidt
b5d8484b0b splitmuxsrc: Add video_%d pad template.
splitmuxsrc actually supports multiple video pads. Make that clear,
especially since it was already creating pads named "video_0" etc.
2019-07-15 11:46:36 +00:00
Mathieu Duponchelle
9deb3c27ac qtdemux: fix conditions for end of segment in reverse playback
The time_position field of the stream is offset by the media_start
of its QtDemuxSegment compared to the start of the GstSegment of
the demuxer, take it into account when making comparisons.
2019-07-09 21:21:20 +00:00
Seungha Yang
67b8ce3167 matroskademux: Fix mismatched transfer characteristic
TransferCharacteristics(18) should be ARIB STD-B67 (HLG)
See https://www.webmproject.org/docs/container/#TransferCharacteristics

Also map more color primaries indexes which have been handled by matroska-mux.
2019-07-09 23:11:45 +09:00
Olivier Crête
9d9d543d5c rtpsession: Also send conflict event when sending packet
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.

Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
2019-07-06 14:23:20 +00:00
Olivier Crête
061afa33ee rtph265pay: Also immediately send packet if it is a suffix NAL
Immediately send packet if it contains any suffix NAL, this is required
in case they come after the VCL nal to not have to wait until the next frame.
2019-07-03 19:05:29 +00:00
Olivier Crête
43e83695fd rtph265pay: Don't drop second byte of NAL header
At least keep 2 bytes per NAL even if the second one is 0, the
second byte of the NAL header could very well be 0.
2019-07-03 19:05:29 +00:00
Olivier Crête
6fed30c48e rtph26xpay: Avoid print when there is no bundle at end of packet 2019-07-03 19:05:29 +00:00
Olivier Crête
97f2fb4cc8 rtph26xpay: Wait until there is a VCL or suffix NAL to send
With unit tests.
2019-07-03 19:05:29 +00:00
Olivier Crête
1b32cb1eae rtph265pay: Implement Aggregation packets
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
5a9b602c9e rtph264pay: Report latency when in maximal aggregation mode 2019-07-03 19:05:29 +00:00
Olivier Crête
cede4f993d rtph264pay: Default to not adding latency when aggregating
Send the bundle as soon as there is one VCL unit in the packet at
the end of an incoming buffer.

The DELTA_UNIT flag is not reliable, so ignore it.
2019-07-03 19:05:29 +00:00
Olivier Crête
13d25583db rtph265pay: Replace fragmentation while-loop with for-loop
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
9be70dc360 rtph265pay: Rename payload_len to max_fragment_size
Align to rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
34c23bdc5d rtph265pay: Clean up _payload_nal
Move determining whether we need to fragment at all into the
fragmenter.

Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
f5765ccf05 rtph265pay: Extract sending fragments into _payload_nal_fragment
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
378c422e0c rtph265pay: Extract sending a single packet into _payload_nal_single
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
b841fd4c8a rtph265pay: Define and use FU_A_TYPE_ID
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
a6d50889af rtph265pay: Use snake_case variables
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
d4268ab2bf rtph265pay: Clean up whitespace and syntax
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
b46dab13d2 rtph264pay: Support STAP-A bundling
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.

*: The property-name is kept generic since it might apply more widely,
   e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
66a3db2083 rtph264pay: Fix delta-unit/discont handling when injecting SPS/PPS
Apply the wanted delta-unit and discont to the first packet; following
packets for this frame are always delta units and not discont.
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
2a16160b57 rtph264pay: Replace fragmentation while-loop with for-loop 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
00936a8362 rtph264pay: Calculate the right max_fragments 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
fe99982dec rtph264pay: Rename payload_len to max_fragment_size 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
5051569713 rtph264pay: Clean up _payload_nal_fragment 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
d97c3f045c rtph264pay: Clean up _payload_nal
Move determining whether we need to fragment at all into the fragmenter.
2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
166c49b800 rtph264pay: Clean up _payload_nal_single 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
b3291620ca rtph264pay: Extract sending fragments into _payload_nal_fragment 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
e493f0ba09 rtph264pay: Extract sending a single packet into _payload_nal_single 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
40c23c06b1 rtph264pay: Define and use FU_A_TYPE_ID 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
bc0018370b rtph264pay: Use snake_case variables 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
28d6dfa51f rtph264pay: Clean up whitespace and syntax 2019-07-03 19:05:29 +00:00
Olivier Crête
37d22186ff rtpjitterbuffer: Unlock output if the queue is full 2019-07-03 18:03:42 +00:00
Thomas Bluemel
080eba64de rtpjitterbuffer: Ignore unsolicited rtx packets.
If an rtx packet arrives that hasn't been requested (it might
have been requested from prior to a reset), ignore it so that
it doesn't inadvertently trigger a clock skew.
2019-07-03 06:23:07 -06:00
Thomas Bluemel
8d955fc32b rtpjitterbuffer: Only calculate skew or reset if no gap.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.

Fixes #612
2019-07-03 06:23:07 -06:00
Mart Raudsepp
ade531183f qtdemux: Provide a 30 frames lead-in for MP3
mpegaudioparse suggests MP3 needs 10 or 30 frames of lead-in (depending on
mpegaudioversion, which we don't know here), thus provide at least 30 frames
lead-in for such cases as a followup to commit cbfa4531ee.
2019-07-02 20:50:21 +00:00
Olivier Crête
af618cb081 rtpjitterbuffer: max-dropout-time gets cast to int32
So any value over MAXINT32 gets considered as negative and is silently ignored.
2019-07-02 19:59:49 +00:00
Mathieu Duponchelle
f4f11530c2 qtdemux: do_seek can never be called with a NULL event 2019-07-02 13:39:55 +02:00
Mathieu Duponchelle
83704e32e6 qtdemux: only adjust segment time when adjusting segment start
We ended up setting segment.time to segment.position when doing
reverse playback, which is obviously wrong.
2019-07-02 13:39:55 +02:00
Mathieu Duponchelle
33277da781 rtspsrc: unref the event in element seek handler 2019-07-01 13:54:13 +02:00
Mathieu Duponchelle
bcd367b81d rtspsrc: handle seek event on the element
Without this, the user has to wait for rtspsrc to have sent a PLAY
request and exposed its pads before seeking it.
2019-06-29 00:25:26 +02:00
Nicolas Dufresne
2c3c1072f7 multiudpsink: Add missing socket.h include
Without this include, macro like SO_BINDTODEVICE is not visible and
associated feature gets out-compiled. This also affects the support for
SO_SNDBUF.
2019-06-26 18:03:29 -04:00
Jan Alexander Steffens (heftig)
152b002658
flvmux: Clear new_tags if sending metadata in header
This avoids sending an additional metadata object right after the
headers.
2019-06-24 17:37:51 +02:00
Mart Raudsepp
ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it
The pre_push_frame default clipping behaviour was introduced in 2010
with commit 30be03004e and modified with commit 4163969a24 in 2011,
when most parsers didn't implement a pre_push_frame yet. Not having it
meant that clipping was done by default. Those that did implement a
pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag
adjusted as part of the 2011 refactor work.

All other parsers got a pre_push_frame vfunc implementation only in
2013, but seem to have forgot to keep the clipping behaviour, as
was done automatically when a pre_push_frame implementation doesn't
exist for the parser. aacparse lost it with commit 91d4abcea in
July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting
in commits 6f89b430e, d2ab5199b, 29f2cae12, 753d3c23a and 292780574.
2019-06-24 14:40:58 +03:00
Jan Alexander Steffens (heftig)
9528bfd78f
flvmux: Simplify an if-else chain
Merge the identical branches and turn the condition around to make it
easier to read.
2019-06-19 14:36:21 +02:00
Jan Alexander Steffens (heftig)
9a70ce87db
flvmux: Avoid crash when changing caps without both streams
mux->video_pad and mux->audio_pad can be NULL if the corresponding pad
has not been requested.
2019-06-19 14:36:21 +02:00
Sebastian Dröge
b18ad8b54c rtpgstpay: Send caps anyway if caps are pending in the adapter but are different from the new ones
Otherwise it can happen that we receive a caps event, then another caps
event and only then buffers. We would then send out the first caps event
in the stream but mark buffers with the caps version of the second caps
event.
2019-06-18 08:35:12 +00:00
Sebastian Dröge
44a697deba rtpgstdepay: Only store the current caps and drop old caps immediately
Otherwise it can happen that we already collected 7 caps, miss the 8th
caps packet (packet loss) and then re-use the 1st caps for the following
buffers instead of the 8th caps which will likely cause errors further
downstream unless both caps are accidentally the same.

Keeping old caps around does not seem to have any value other than
potentially causing errors. We would always receive new caps whenever
they change (even if they were previous ones) and it's very unlikely
that they happen to be exactly the same as the previous ones.

Also after having received new caps or a buffer with a next caps
version, no buffers with old caps version will arrive anymore.
2019-06-18 08:35:12 +00:00
Jan Schmidt
53b3f2ddbb rtpjitterbuffer: Clear clock master before unreffing
Make sure to clear any master clock on the media_clock
before unreffing it to release the timer callback that's
updating the clock and keeping it reffed.
2019-06-16 20:36:55 +10:00
Jan Schmidt
2479ccac7d matroska: Initialise a video_context field to satisfy valgrind
Clear the mastering_display_info_present field explicitly
after reallocating the track context into a video context
to avoid uninitialised warnings in valgrind
2019-06-16 11:10:41 +10:00
Thibault Saunier
ac55681bbf docs: Fix link to strings
We can't link to #gchar* this way.
2019-06-14 17:34:43 -04:00
Mathieu Duponchelle
ebe2756434 jitterbuffer: unset DTS on output buffers 2019-06-14 16:02:59 +02:00
Mathieu Duponchelle
ddbbe5d277 splitmuxsink: set the same seqnum on flush_start / flush_stop
It's currently not made mandatory by aggregator, but it might
eventually be, and is more consistent behaviour

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/977
2019-06-13 16:44:47 +02:00
Mikhail Fludkov
ec5fa49631 rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic
effect on the jitterbuffer clock-skew logic, having it being reset
and losing track of the current dts-to-pts calculations, directly affecting
the packets that arrive later.

This is demonstrated in the test, where a RTX packet is pushed in really
late, and without this patch the last packet will have its PTS affected
by this, where as a late RTX packet should be redundant information, and
not affect anything.
2019-06-13 11:55:10 +02:00
Mikhail Fludkov
b9c3e354ee rtpjitterbuffer: fix rtx delay calulation when large packet spacing 2019-06-12 11:39:32 +02:00
Stian Selnes
6269ed49ab rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
This patch corrects the delay set on EXPECTED timers that are added when
processing gaps. Previously the delay could be too small so that
'timout + delay' was much less than 'now', causing the following retries
to be scheduled too early. (They were sent earlier than
rtx-retry-timeout after the previous timeout.)
2019-06-12 11:39:32 +02:00
Havard Graff
8ed7ab178b rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping
Turns out that the "big-gap"-logic of the jitterbuffer has been horribly
broken.

For people using lost-events, an RTP-stream with a gap in sequencenumbers,
would produce exactly that many lost-events immediately.
So if your sequence-numbers jumped 20000, you would get 20000 lost-events
in your pipeline...

The test that looks after this logic "test_push_big_gap", basically
incremented the DTS of the buffer equal to the gap that was introduced,
so that in fact this would be more of a "large pause" test, than an
actual gap/discontinuity in the sequencenumbers.

Once the test was modified to not increment DTS (buffer arrival time) with
a similar gap, all sorts of crazy started happening, including adding
thousands of timers, and the logic that should have kicked in, the
"handle_big_gap_buffer"-logic, was not called at all, why?

Because the number max_dropout is calculated using the packet-rate, and
the packet-rate logic would, in this particular test, report that
the new packet rate was over 400000 packets per second!!!

I believe the right fix is to don't try and update the packet-rate if
there is any jumps in the sequence-numbers, and only do these calculations
for nice, sequential streams.
2019-06-12 11:39:31 +02:00
Jan Schmidt
f6b91fe303 splitmuxsrc: Protect initial pad configuration with the object lock
gst_splitmux_src_activate_part() configures the pad information
before starting the pad task, but occasionally the changes it makes
to the pad are not seen in the pad task because they're not
protected by the right locking. Use the pad's object lock to
protect those variables.
2019-06-12 02:46:48 +10:00
Jan Schmidt
715c6896a2 splitmuxsrc: Restart pad task on a reconfigure
On a reconfigure event, restart streaming on the pad so
that switching tracks in playbin works cleanly
2019-06-12 02:46:48 +10:00
Jan Schmidt
86c131b668 splitmuxsrc: Use an RW lock instead of a mutex to protect the pad list
Fix a deadlock around the pads list by using an RW lock to
allow simultaneous readers. The pad list doesn't really changes
except at startup and shutdown.
2019-06-12 02:46:48 +10:00
Jan Schmidt
26d6532702 splitmuxsrc: Ignore duplicate seeks
Use the seqnum to ignore duplicated seek events.
2019-06-12 02:46:41 +10:00