For files containing video only video keyframes are valid points to
which a player can seek. For audio-only files any tag start is a valid
seek point.
See #601236.
* The nb_elements for arrays is just an indication, we can therefore ignore
it and carry on parsing metadata items until we reach the end marker.
* If type == 3, then the script tag contains a list of object followed
by the end marker.
Refactor code slightly to handle both cases
https://bugzilla.gnome.org/show_bug.cgi?id=610447
This just replaces every "$ERROR_CFLAGS" usage with a usage of
"$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
previously.
Actually using that separation will happen later.
Push mode seeking uses same index data as pull mode, and stores
offset to data in chunk, whereas push mode operates in chunks,
and as such needs offset consistently corresponding to chunk headers.
Also fix determining best matching stream for incoming newsegment event,
as well as setting some stream state accordingly.
Scan for needed part upon a seek as opposed to doing a complete scan
at startup, which may take some time depending on file and/or platform.
Also accept index metadata in pull mode and peek for some metadata
at the end of the file when deemed appropriate.
Add a "favor-new" property that tells the session to favor new sources when
there is a SSRC conflict. This is useful for SIP calls and other such cases
where a remote loop is extremely unlikely.
Fixes#607615
Use the length of the payload for estimating the receiver bitrate so that it
matches the calculations done on the sender side. Together with the number of
packets one can scale the bitrate with the header overhead of the lower
transport.
Don't reuse the same variable we need for stats for the bitrate estimation
because we're updating it.
Refactor the bitrate estimation code so that both sender and receivers use the
same code path.
Changing it to the newest timestamp that was ever pushed will
increase the segment start in 500ms jumps, which could be just
after the next sparse stream buffer. E.g.
Video at 1.0s, sparse stream at 0.5s would jump the
sparse stream to 1.0s. Now a new sparse stream buffer could
appear that has a timestamp of 0.9s and this would be
dropped for no good reason because of bad luck.
Multiple flvparse/flvdemux instances should be able to operate without
trampling over each other by accidentally re-using the same (static)
variables. (Spotted by Mark Nauwelaerts)
For calculating the durations of each sample, we are supposed to add each
duration modulo 1<<32 so make the elapsed time counter a uint32.
Fixes#610280
Make the handing of the mime type within the "boundary" a bit less naive.
The standard for MIME allows parameters to follow the "type" / "subtype"
clause separated from the mime type by ';'.
Modifies the multipartdemuxer's header parsing so it doesnt assume
the whole line after "content-type:" is the mime type and thus makes it a bit
more resilient to finding absurd mime types in the case where parameters are
added.
Fixes#604711
ALAC codec-data apparently comes in (at least) two flavours (mov, mp4),
so use atom based parsing to retrieve required data, rather than
aiming for a specific offset.
See also #580731.
Remove some code where we pass ntpnstime around, we can do most things with the
running_time just fine.
Rename a variable in the ArrivalStats struct so that it's clear that this is the
current system time.
Don't calculate the NTP time based on the running_time of the pipeline but from
the systemclock. This allows us to generate more accurate NTP timestamps in case
the systemclock is synchronized with NTP or similar.
Used the _add_associationv variant of GstIndex since we know how many
associations we're adding. Trims up to 50% from index generation time.
Note : It would be great if the index could be generated on the fly or
on request as opposed to being fully created at startup.
If we detect backward timestamps on the server, don't try to resync when we
don't have an input timestamp (such as when using RTSP over TCP) instead, do
nothing but assume the timestamp was ok, it will correct itself when time goes
forwards.
There is no need to set the latency in the jittebuffer in _init, we will set
that later when going to PAUSED.
Set the jitterbuffer active and not buffering when starting.
When deactivating jitterbuffers when the buffering starts, keep the current
percent of the jitterbuffer and also set the jitterbuffer in the buffering state
so that we know when it's filled again.
Add property to get the buffering percentage of the jitterbuffer.
When we are in buffer mode, adjust the buffering low/high thresholds based on
the total configured latency. If we don't and there is a huge queue or element
with a big latency downstream we might drain the complete queue immediately and
start buffering again.
