The matroska codec specs is unfortunately vague on the subject,
stating for H264:
AVC/H.264 stored as described in [@!ISO.14496-15]
and for H265:
HEVC/H.265 stored as described in [@!ISO.14496-15]
This spec however specifies multiple stream formats, our
implementation has opted for interpreting this as avc1 / hvc1,
both of which disallow in-band SPS.
Most decoders however will support in-band SPS / PPS, and
this commit gives the option to explicitly mux in avc3 / hev1,
which allows changing stream parameters on the fly, that is
useful for smart encoding for example.
When either of these stream formats are picked as the input,
changes in codec_data / tier / level / profile do not cause
renegotiation failure, a warning is logged however as it isn't
clear how compliant such a stream is.
The stream-format field is correctly ordered in the template
caps to avoid selecting potentially non-compliant options on
automatic negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
Adds a user-controllable timestamp offset to clusters and blocks. This
should be useful if we want to have timestamps that have significance
outside of the current file (for example, we might set the offset to the
wallclock when the file is being created, or some other common base, if
we want to correlate streams across multiple files).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1051>
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.
1. A server use Apache combined with a separate RTSP process to handle
Https request on port 443. In this case Apache handle TLS and
connects to the local RTSP server, which results in a local
address 127.0.0.1 or ::1 in the x-server reply. This address is
returned to the actual RTSP client in the x-server header.
The client will receive this address and try to connect to it
and fail.
2. The client use a ipv6 link local address with a specified scope id
fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
The RTSP server receives the connection and returns the address
in the x-server header. The client will receive this address and
try to connect to it "as is" without the scope id and fail.
In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1007>
By this new property, user can select physical port to connect,
and element will pick requested port instead of random ones.
User should provide full port name including "client_name:" prefix.
An example is
jackaudiosrc port-names="system:capture_1,system:capture_3" ! ...
jackaudiosink port-names="system:playback_2"
In addition to "port-names" property, a new connect type "explicit"
is added so that element can post error message if requested
"port-names" contains invalid port(s).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1037>
Currently, videocrop, only negotiates raw caps (system memory) because
it's the type of memory it can modify. Nonetheless, it's also possible
for the element to handle non-raw caps when only adding the crop meta
is possible, in other words, when downstream buffer pools expose the
crop API.
This patch enable non-raw caps negotiation. If downstream doesn't
expose crop API and negotiated caps are featured, the negotiation
fails.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/915>
This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file.
Wavpack float mode is always in 32-bit IEEE format.
The following pipeline plays distorted audio if source file is encoded in float mode:
gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/894>
A classic case of not enough locking.
One interesting thing with this is the interaction between the
rotation value and caps negotiation. i.e. the width/height of the caps
can be swapped depending on the video-direction property. We can't lock
the entirety of the caps negotiation for obvious reasons so we need to
do something else. This takes the approach of trying to use a single
rotation value throughout the entirety of the negotiation and then
subsequent output frame in a kind of latching sequence.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
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>
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>
