It is more appropriate to start closer to the live edge in
live streams. Some live streams maintain a large dvr window
(over few hours in some cases), so starting from the first
fragment will be too far away from the live edge.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1346>
The SPS parsing functions take a parse_vui_param flag
to skip VUI parsing, but there's no indication in the output
SPS struct that the VUI was skipped.
The only caller that ever passed FALSE seems to be the
important gst_h264_parser_parse_nal() function, meaning - so the
cached SPS were always silently invalid. That needs changing
anyway, meaning noone ever passes FALSE.
I don't see any use for saving a few microseconds in
order to silently produce garbage, and since this is still
unstable API, let's remove the parse_vui_param.
Without this, for streams where the content is stored indefinitely and
can be seeked on, the duration would never increase when in paused or,
until we reached near the end of the currently advertised stream (where
the internal fragment parser would see descriptions of new fragments).
A live manifest may have a set (> LookAheadFragmentCount) of fragments
that have already been served and are stored on the server, maybe
indefinitely. Adding the parsed live fragments after the manifest
fragments breaks duration reporting and the seekable range.
Fix by only adding parsed fragments outside the list of fragments which
assumes that the fragment list in the manifest is accurate enough to not
stray too far off what's in the retrieved data.
https://bugzilla.gnome.org/show_bug.cgi?id=779447
For duration queries on live streams, adaptivedemux ignores the query.
The problem then is that the query is answered by the downstream
qtdemux element, with the duration of the currently passing fragment.
This commit changes the behaviour of adaptivedemux to answer the duration
queries for live streams, returning GST_CLOCK_TIME_NONE.
https://bugzilla.gnome.org/show_bug.cgi?id=753879
When a MSS server hosts a live stream the fragments listed in the
manifest usually don't have accurate timestamps and duration, except
for the first fragment, which additionally stores timing information
for the few upcoming fragments. In this scenario it is useless to
periodically fetch and update the manifest and the fragments list can
be incrementally built by parsing the first/current fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=755036
Allows seeking through the available fragments that are still available
on the server as specified by the DVRWindowLength attribute in the
manifest.
https://bugzilla.gnome.org/show_bug.cgi?id=774178
Use new gst_h264_video_calculate_framerate() API instead of fps_n/fps_d
fields in SPS struct which are to be removed.
Apparently H264 content in MSS is always non-interlaced/progressive,
so we can just pass 0 for field_pic_flag and don't need to parse any
slice headers first if there's no external signalling. But even if
that's not the case the new code is not worse than the existing code.
https://msdn.microsoft.com/en-us/library/cc189080%28VS.95%29.aspxhttps://bugzilla.gnome.org/show_bug.cgi?id=723352
stream->current_fragment has the value of g_list_previous (iter) which has
just been checked. No need to check it again.
Just to be safe, use a g_assert() to check fragment before dereferencing.
CID #1352041
Adaptive demuxers need to start downloading from specific positions
(fragments) for every stream, this means that all streams can snap-seek
to a different position when requested. Snap seeking in this case will
be done in 2 steps:
1) do the snap seeking on the pad that received the seek event and
get the final position
2) use this position to do a regular seek on the other streams to
make sure they all start from the same position
More arguments were added to the stream_seek function, allowing better control
of how seeking is done. Knowing the flags and the playback direction allows
subclasses to handle snap-seeking.
And also adds a new return parameter to inform of the final
selected seeking position that is used to align the other streams.
https://bugzilla.gnome.org/show_bug.cgi?id=759158
Not doing this can lead the demuxer to attempt downloading fragments
for an invalid start time. The server would then send a HTTP
Precondition failed error, the demuxer would try some more times to
download the invalid fragment and eventually error out.
https://bugzilla.gnome.org/show_bug.cgi?id=754523
q->bitrate is a guint64, but G_TYPE_INT may read fewer bits
off the stack, and if we pass more then the NULL sentinel
may not be found at the right place, which in turn might
lead to crashes.
https://bugzilla.gnome.org/show_bug.cgi?id=741751
Rework reverse fragment traversing with repetition fields to prevent
NULL pointer deref and avoid never advancing a fragment as the variable
is unsigned and would always be non-negative.
CID #1257627
CID #1257628
Read the "r" attribute from fragments to support fragments nodes
that use repetition to have a shorter Manifest xml.
Instead of doing:
<c d="100" />
<c d="100" />
You can use:
<c d="100" r="2" />
WmaPro is actually wmaversion 3, and can also be found by the
WMAP fourcc.
Some manifests also contain the block_align field as "PacketSize"
in the audio track description, the libav decoders require it
to be present in caps.
Fixes#699921
wma v2 expects block_align, channels and rate fields set to its caps.
This isn't present direclty on the manifests, so mssdemux should parse
it from the waveformatex structure
https://bugzilla.gnome.org/show_bug.cgi?id=699924
bitrate info is always present on the QualityLevel xml node as part
of the adaptive selection processing, put it into caps as some
decoders require it (avdec_wmav2 for example)
https://bugzilla.gnome.org/show_bug.cgi?id=699924
g_ascii_strtoull() returns a long long integer, but we need to
pass a normal int to gst_structure_set() for fields of G_TYPE_INT,
so cast appropriately.
connection setup times seem to matter when measuring the download
rate of different streams. Streams with longer fragments have a
*relatively* lower connection setup time and achieve higher bitrates.
Using the average seems unfair here, so use each stream's measured bitrate
to select its best quality option.
