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.
When the codec is AAC-LC, some server implementation (e.g. Microsoft IIS) doesn't add the CodecPrivateData attribute. The element needs to re-create the codec data from the Quality Level attributes (channels and sampling rate).
Live streams force the demuxer to keep reloading the Manifest from
time to time, as the new fragments are being added as they are recorded.
The demuxer should also try to keep up and detect when it had to skip
fragments, marking the discont flag when that happens.
Curiously, the spec doesn't seem to mention when/how a live stream is supposed
to end, so keep trying downloads until the demuxer errors out.
This speed limits the maximum bitrate of streams. Currently it
is only read when starting the pipeline, but it should be used
for switching bitrates during playback to adapt to network
changes.
mssdemux should set the streams it has exposed as active so that
the manifest won't use the non-active streams to compute total bitrates
or providing fragments
