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If we also replace all headers when receiving any possibly following
comments header, we would throw away the config header before being able
to make use of it.

A sparse stream's ending timestamp can be considerably smaller
than the ending timestamps of the other streams, which can lead
to skipping considerable time from the next part.

https://bugzilla.gnome.org/show_bug.cgi?id=761086

https://bugzilla.gnome.org/show_bug.cgi?id=779230

In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
timestamps is more than (3 * jbuf->clock_rate) we call
rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
the pipeline (playes, mixers) just skip frames/samples until pts becomes
equal to pts before gap.

In version 1.10.2 and before this checking was bypassed for packets with
"estimated dts", and gaps were handled correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=778341

The payloader needs to reset and update the vorbis config data which is
pushed on the network if it receives new headers, or at least, it may
have to do so.

Without this, the stream configuration could change without the
payloader sending the new configuration to the other side.

https://bugzilla.gnome.org/show_bug.cgi?id=761761

https://bugzilla.gnome.org/show_bug.cgi?id=761761

https://bugzilla.gnome.org/show_bug.cgi?id=761761

https://bugzilla.gnome.org/show_bug.cgi?id=761761

This reverts commit 107902ec.

This commit intended to ensure that keyframe seeks land at the
start timestamp of a keyframe, rather than in the middle of one,
but they cause trouble on files with sparse streams, or with
JPEG 'cover art' tracks that have only one or a few JPEG samples
with very long durations.

That's still desirable for doing seamless cutting of videos,
but needs a rethink for implementation.

https://bugzilla.gnome.org/show_bug.cgi?id=778690

Add a new boolean surround-delay property that makes
audioecho just apply a delay to certain channels to create
a surround effect, rather than an echo on all
channels. This is useful when upmixing from stereo - for example.

Add a surround-mask property to control which channels
are considered surround sound channels when adding a
delay with surround-delay = true

Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>

This goes around the inefficient control message based filtering and
does all the filtering kernel-side. Unfortunately this is Linux-only and
there is no IPv6 variant of it (yet).

Some radio streams uses StreamTitle='' to reset the title after a
track stopped playing, e.g. while the host talks between tracks or
during news segments.
This change forces an empty tag object to be distributed if
StreamTitle or StreamUrl is received with empty value, thus allowing
downstream elements to get notified about this.

https://bugzilla.gnome.org/show_bug.cgi?id=778437

If the srtp elements are not present, post a message on the bus
informing about the missing plugins.

Upstream elements like videoflip can transform caps, such as changing width and height.
When an imagefreeze downstream receives an ACCEPT_CAPS query it will NOW return
all caps that it can accept.

https://bugzilla.gnome.org/show_bug.cgi?id=778389

Don't spam debug log with uninteresting stuff.

We recalculate them, so any old information has to be forgotten.
Otherwise we write invalid edit lists when writing headers multiple
times.

https://bugzilla.gnome.org/show_bug.cgi?id=778330

Used signed calculations when measuring the max_ts of an input
fragment, so as to calculate the correct duration and offset
when buffers have timestamps preceding their segment

This is needed in order to:
 - Avoid ignoring requests for different media sources.
 - Add SSRC field in the GstForceKeyUnit event.

https://bugzilla.gnome.org/show_bug.cgi?id=778013

Fixes crash with fuzzed file.

https://bugzilla.gnome.org/show_bug.cgi?id=777940

Don't need to querying byte-format seeking for time-format
upstream case

https://bugzilla.gnome.org/show_bug.cgi?id=776715

When multiple demuxer's are used, upstream might want to indicate
default streams using GST_STREAM_FLAG_{SELECT, UNSELECT}

https://bugzilla.gnome.org/show_bug.cgi?id=775440

The n_frames field (frames per second) should follow the nominal frame
rate for drop-frame timecodes.

Also, the trak's timescale (and duration, accordingly) should follow the
STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
not the other way around.

https://bugzilla.gnome.org/show_bug.cgi?id=777832

https://bugzilla.gnome.org/show_bug.cgi?id=777101

https://bugzilla.gnome.org/show_bug.cgi?id=777362

In case wavparse receives a manually injected FLUSH_STOP event
while operating in pull mode we get criticals because we'd try
to clear a NULL adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=777123

For MP4 this is not defined, and it actually breaks things for MSE in
Chrome if we do this. For MOV this is required by some broken software
but the official specification says it's optional:
https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html

https://bugzilla.gnome.org/show_bug.cgi?id=777540

Currently, the nack packets sent or received are kept at session level,
which makes it impossible to distinguish how many of these packages were
sent/received per ssrc when several sources are in the same session. This
patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats

https://bugzilla.gnome.org/show_bug.cgi?id=776714

Insert VPS/SPS/PPS before the first NAL unit containing an I-frame in an
access unit only. If an access unit consists of several such NAL units
(tiles) VPS/SPS/PPS should only be inserted before the first of them so
that parameters are only updated between frames.

Do not insert VPS/SPS/PPS before P-frames when config-interval is -1.

https://bugzilla.gnome.org/show_bug.cgi?id=775817

https://bugzilla.gnome.org/show_bug.cgi?id=777532

https://bugzilla.gnome.org/show_bug.cgi?id=777500

Otherwise we could read more chunks than there are available, doing an
out of bounds read and potentially crash.

https://bugzilla.gnome.org/show_bug.cgi?id=777469

This reverts commit 99d5d757. It broke
playback of various valid files.

Otherwise we could read more chunks than there are available, doing an
out of bounds read and potentially crash.

https://bugzilla.gnome.org/show_bug.cgi?id=777469

qtdemux_handle_xmp_taglist() requires a writable taglist,
but qtdemux->tag_list can become non-writable, specifically
after sending global tags (qtdemux.c:958), which adds a
second reference.  Ensure the list is made writable before
calling (make_writable will copy the list if necessary).

https://bugzilla.gnome.org/show_bug.cgi?id=766177

Keep taglist around during element existance to avoid having to
create it at different places before usage. Makes code simpler to handle.

https://bugzilla.gnome.org/show_bug.cgi?id=766177

They are handled since commit 7b565475
(qt: Add support for ProRes 4444 XQ).

https://bugzilla.gnome.org/show_bug.cgi?id=777377