Directly setting audio/x-raw caps leads to problems when the delivered
data blocks do not align properly at sample boundaries (for example, a
data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
let a parser be autoplugged.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
segment_duration and media_time should be parsed based on version
of elst box. Specification defines that an elst box with version 1
has uint64 and int64 values for segment_duration and media_time,
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=766301
Non-blocking read will return the amount of data available without
blocking to wait for the full requested size.
The downside is that now it souphttpsrc needs to have a waiting
mechanism in case there is no data available yet to avoid busy
looping arond the inputstream.
Set the async-handling property on GstBin to let it manage
async-handling instead of the local handling from the previous
commit. Works because of #174a5e in core
Avoid using soup_server_run_async and old get_port() APIs,
replace with me soup_server_listen and get the port through the
URIs list returned from the server.
When switching fragments, hide the async-start/async-done
messages from the parent bin, as otherwise we sometimes (very rarely)
hang in PAUSED instead of returning / continuing to PLAYING
state.
1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles.
However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet.
2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id.
https://bugzilla.gnome.org/show_bug.cgi?id=745187
Properly handle edts segments for push-based operation seeking.
We only support edts that a single segment that has media at the end,
being preceeded by any number of gap segments.
This also allows the qt segment rate to be respected after seeks
https://bugzilla.gnome.org/show_bug.cgi?id=765669
When a packet arrives that has already been considered lost as part of a
large gap the "lost timer" for this will be cancelled. If the remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.
The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.
In practice the issue is rare since large gaps are scheduled immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.
https://bugzilla.gnome.org/show_bug.cgi?id=765933
The access to the session hash table must happen while the session lock is
taken, otherwise another thread might modify the hash table while we're
creating the stats.
https://bugzilla.gnome.org/show_bug.cgi?id=766025
The previous ones resulted in odd display aspect ratios and were different
from the ones used by e.g. ffmpeg. The new ones now result in display aspect
ratios of 4:3 and 16:9.
https://bugzilla.gnome.org/show_bug.cgi?id=765946
This signal allows a user to directly return a sorted list of
files to be joined, so that they don't have to follow the
filename pattern that the "location" property expects.
https://bugzilla.gnome.org/show_bug.cgi?id=753625
The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'.
There is no requirement for 'fmt' to be first. We already had a list of chunks
to skip, but it is easier to just skip any chunk while seeking for 'fmt'.
This fixes reading files generated by ProTools.
Via the MPEG-4 Part 3 spec we can support the other layers too.
Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
MPEG-2.5.
https://bugzilla.gnome.org/show_bug.cgi?id=765725
We only changed them for UDP so far, which caused the wrong seqnum-base and
other information to be passed to rtpjitterbuffer/etc when seeking. This
usually wasn't that much of a problem as the code there is robust enough, but
every now and then it causes us to drop up to 32756 packets before we
continue doing anything meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=765689
set_fields() should only be called in the beginning, otherwise we will never
remember the maximum audio chunk size and write a wrong block align... which
then causes wrong timestamps and other problems.
3ea338ce27 changed avimux to do that, but it
never actually kept track of the max audio chunk for MP3 and MP2. These are
knowing the hdr.scale only after parsing the frames instead of at setcaps
time.
When a frame's duration is too low, calling gst_util_uint64_scale()
to scale its value can result into it being truncated to zero, which
will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
when trying to encode.
To prevent this from happening, we simply ignore the duration when
encoding if it becomes zero after scaling, logging a warning message.
https://bugzilla.gnome.org/show_bug.cgi?id=765391
timescale/1 is unreliable value for framerate. Due to downstream
element usually use framerate generated by qtdemux, let it be omitted
until the framerate can be reliably calculated.
https://bugzilla.gnome.org/show_bug.cgi?id=764733
