This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
associated with the session. This allow rtpbin users to clear pads and
elements for a specific ssrc that is known to no longer be in use. This
happens when a pad is reused in rtpsrc or ristsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
Add property to set the initial value for picture-id. RFC7741 says
that picture-id MAY be initialized to a random value, thus it's also
valid to simply set it to a fixed initial value. A fixed value is very
useful for testing.
Default behavior is not changed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
- introduce two new properties:
* temporal-scalability-layer-flags:
Provide fine-grained control of layer encoding to the
outside world. The flags sequence should be a multiple of
the periodicity and is indexed by a running count of encoded
frames modulo the sequence length.
* temporal-scalability-layer-sync-flags:
Specify the pattern of inter-layer synchronisation (i.e.
which of the frames generated by the layer encoding
specification represent an inter-layer synchronisation).
There must be one entry per entry in
temporal-scalability-layer-flags.
- apply temporal scalability settings and expose as buffer
metadata.
This allows the codec to allocate a given frame to the correct
internal bitrate allocator. Additionally, all the
non-bitstream metadata needed to payload a temporally scaled
stream is now attached to each output buffer as a
GstVideoVP8Meta.
- add unit test for temporally scaled encoding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.
Remove compat code as glib requirement
is now > 2.56
Version used by Ubuntu 18.04 LTS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
Scenario:
- gap event causes h264parse to push made up caps that may fail checks
inside qtmux (e.g missing codec_data).
- the caps event has already been marked as received and is sticky on
the sink pad
- gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
using gst_pad_get_current_caps() and reject the correct updated caps
with codec_data.
- Failure!
Keep track of the configured caps ourselves instead of relying on the
sticky event on the pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Since !348, the jitterbuffer was only removed with the session. This restores
the original behaviour and removes the jitterbuffer when the stream is
removed. This avoid accumulating jitterbuffer objects into the bin when a
session is reused.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
The rtpjitterbuffer is now part of the session elements, we no longer need
to do the ref_sink dance when signalling it. It is already owned by the bin
when signalled. Also, the code that handles generic session elements already
handle the ref_sink() calls since:
03dc22951b
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
If we have not received a FU with a start bit set, any subsequent FU
data is not useful at all and would result in an invalid stream.
This case is constructed from multiple requirements in
RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3. Following are excerpts
from RFC 3984 but RFC 7798 contains similar language.
The FU in a single FU case is forbidden:
A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
Start bit and End bit MUST NOT both be set to one in the same FU
header.
and dropping is possible:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
The jump in seqnum case is supported by this from the specification
instead of implementing the forbidden_zero_bit mangling:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
A receiver in an endpoint or in a MANE MAY aggregate the first n-1
fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
n of that NAL unit is not received. In this case, the
forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
syntax violation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
Used by some proprietary software for their fragmented files.
Adds some support for multi-stream fragmented files
Flow is as follows.
1. The first 'fragment' is written as a self-contained fragmented
mdat+moov complete with an edit list and durations, tags, etc.
2. Subsequent fragments are written with a mdat+moof and each stream is
interleaved as data arrives (currently ignoring the interleave-*
properties). data-offsets in both the traf and the trun ensure
data is read from the correct place on demuxing. Data/chunk offsets
are also kept for writing out the final moov.
3. On finalisation, the initial moov is invalidated to a hoov and the
size of the first mdat is extended to cover the entire file contents.
Then a moov is written as regularly would in moov-at-end mode (the
default).
This results in a file that is playable throughout while leaving a
finalised file on completion for players that do not understand
fragmented mp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
When we are fragmented, the edit list may only refer to the portion of
the media that is in the moov. Extend the edit list stop time when we
if there is only one qt segment and we are reading a fragmented file.
Fixes playback of some fragmented mp4 files generated by proprietary
programs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
This is modelled after the DASH Common Encryption scheme, but is somewhat
simpler as more parts are fixed, i.e. just one encryption scheme.
The output caps are fixed to 'application/x-aavd'. All information
required for decryption are part of the 'adrm' atom, which is passed
on as a property. The property is attached to the buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
box follows the layout of an 'mp4a' entry, i.e. it contains a single
standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
extension boxes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
With recent libvpx versions, multithreading can be enabled on
a per-tile basis, instead of on a per tile-column basis.
In combination with the new tile-rows property, this allows the
encoder to make much better use of the available CPU power.
Bump minimum libvpx version to 1.7.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
- Release the split mux lock while removing the probes
- Flush the sinkpad to unblock other pads
- Turn check_completed_gop into a do while statement, when
waking up we want to recheck whether the current GOP is
ready for sending
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
- make test_encode_simple cope with libvpx built with
CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
runtime beyond trying to set lag-in-frames to >0, pushing a
buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
- fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
- port test_encode_simple to GstHarness and introduce a separate
test for the lag-in-frames property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
