Commit graph

8283 commits

Author SHA1 Message Date
Tim-Philipp Müller
6717c86061 rtp: depayloaders: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the mapping the base class has already done.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:28:29 +01:00
Tim-Philipp Müller
fe787425bc rtpvrawdepay: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the map the base class has already done.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:28:25 +01:00
Sebastian Dröge
582ade2c42 rtpjitterbuffer: Fix indention 2015-07-10 00:13:32 +03:00
Sebastian Dröge
ae8acc0973 rtpjitterbuffer: Always estimate DTS from the current clock time
Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
we would produce wrong DTS. As now the estimated DTS is based on the clock,
don't store it in the jitterbuffer items as it would otherwise be used in the
skew calculations and would influence the results. We only really need the DTS
for timer calculations.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-10 00:13:22 +03:00
Thiago Santos
30b3aa3030 qtdemux: rework segment event handling for adaptive streaming
When a new time segment is received upstream is going to restart
with a new atom. Make the neededbytes and todrop variables
reflect that to avoid waiting too much or dropping the
initial bytes that contain the header.
2015-07-08 23:23:53 -03:00
Thiago Santos
38520a1e12 qtdemux: push data from adapter before starting new segment
The adapter might have data remaining from the previous segment,
push it all before clearing the adapter and starting a new segment.

It can accumulate data if it had pushed and got not-linked, returning
immediately without processing all the data. Before starting a new
segment this data should be handled.
2015-07-08 23:23:53 -03:00
Sebastian Dröge
6e7c724afa rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 23:19:52 +03:00
Havard Graff
ddd032f56b rtpjitterbuffer: fix gap-time calculation and remove "late"
The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.

The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)

Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.

https://bugzilla.gnome.org/show_bug.cgi?id=738363
2015-07-08 23:18:48 +03:00
Stian Selnes
40524e5a49 Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
This reverts commit 05bd708fc5.

The reverted patch is wrong and introduces a regression because there
may still be time to receive some of the packets included in the gap
if they are reordered.
2015-07-08 23:18:48 +03:00
Thiago Santos
ee7ddf6c67 qtdemux: flush samples before adding more from moof
Avoids accumulating all samples from a fragmented stream that could
lead to a 'index-too-big' error once it goes over 50MB of data. It
could reach that before 2h of playback so it doesn't take that long.

As upstream elements are providing data in time format they should
be the ones that have more information about the full media index
and should be able to seek if possible.
2015-07-08 11:53:44 -03:00
Thiago Santos
6ee4b31c0e qtdemux: rename upstream_newsegment to upstream_format_is_time
upstream_newsegment isn't really clear on what it means, it is set
to TRUE when the upstream element sends a segment in TIME format, so
rename it to be more clear about it.

It is important to know this because it means that upstream has
a notion of time and qtdemux is likely being driven by an upstream
element that is reading from a higher level abstraction than a file,
such as a DASH, MSS or DLNA element.
2015-07-08 11:53:44 -03:00
Thiago Santos
5994b30257 qtdemux: fix leak by flushing previous sample info from trak
In fragmented streaming, multiple moov/moof will be parsed and their
previously stored samples array might leak when new values are parsed.
The parse_trak and callees won't free the previously stored values
before parsing the new ones.

In step-by-step, this is what happens:

1) initial moov is parsed, traks as well, streams are created. The
   trak doesn't contain samples because they are in the moof's trun
   boxes. n_samples is set to 0 while parsing the trak and the samples
   array is still NULL.
2) moofs are parsed, and their trun boxes will increase n_samples and
   create/extend the samples array
3) At some point a new moov might be sent (bitrate switching, for example)
   and parsing the trak will overwrite n_samples with the values from
   this trak. If the n_samples is set to 0 qtdemux will assume that
   the samples array is NULL and will leak it when a new one is
   created for the subsequent moofs.

This patch makes qtdemux properly free previous sample data before
creating new ones and adds an assert to catch future occurrences of
this issue when the code changes.
2015-07-08 11:53:44 -03:00
Thiago Santos
63f35eeb12 qtdemux: fix index size check and debug message
It is allocating samples_count + n_samples, not only n_samples
2015-07-08 11:53:44 -03:00
Sebastian Dröge
4e23481d9f rtpjitterbuffer: Calculate receive time if we don't have any
This is required to properly schedule packet loss timers and make
sure all our calculations work properly.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 17:02:05 +03:00
Sebastian Dröge
243730ced4 rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
That is, handle DTS==GST_CLOCK_TIME_NONE correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-07-08 15:15:00 +03:00
Vineeth T M
5439fc9a0c avidemux: fix event leak
when seek fails in avidemux, event is not being freed.

https://bugzilla.gnome.org/show_bug.cgi?id=752117
2015-07-08 12:57:43 +01:00
Stian Selnes
8a0dbff3f4 rtph263depay: Make sure payload is large enough
Plus new unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=752112
2015-07-08 11:36:55 +01:00
Vineeth TM
ffe9cbc1f6 rtpklvdepay: fix printf format compiler warning
v_len is of type guint64, but while print the value(16 + len_size + v_len)
G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT

https://bugzilla.gnome.org/show_bug.cgi?id=752100
2015-07-08 08:49:37 +01:00
Tim-Philipp Müller
b105e1e3d1 rtpklvdepay: improve start detection and handle fragmented KLV units 2015-07-07 20:11:27 +01:00
Tim-Philipp Müller
740f10bae9 rtp: add SMPTE 336M KLV metadata depayloader
http://tools.ietf.org/html/rfc6597
2015-07-07 20:11:27 +01:00
Tim-Philipp Müller
7db7da1acb rtp: add SMPTE 336M KLV metadata payloader
http://tools.ietf.org/html/rfc6597
2015-07-07 20:11:23 +01:00
Stefan Sauer
12930c2f8c docs: fix "Symbol name not found at the start of the comment block"
Add symbols or change comment into a regular comment.
2015-07-07 17:12:02 +02:00
Stefan Sauer
093e8f8a75 docs: remove outdated doc strings 2015-07-07 17:12:02 +02:00
Luis de Bethencourt
55175561f6 Revert "imagefreeze: Remove impossible error condition"
This reverts commit d46631c5c7.

pad only handle EOS events but not EOS flow, and will push the buffer again
resulting in an assertion error. So we should not handle the buffer
and return EOS flow.
2015-07-07 15:57:19 +01:00
Tim-Philipp Müller
f0c6b728f8 rtpg729depay: unmap rtp buffer in error path 2015-07-07 15:50:50 +01:00
Tim-Philipp Müller
f07d61a9ef rtpg729pay: fix buffer leak
The handle_buffer vfunc takes ownership of the input buffer.
Fixes elements/rtp-payloading under valgrind.
2015-07-07 15:50:37 +01:00
Tobias Mueller
6faeb75170 goom: Initialised variables to remove compiler warnings
goom_core.c: In function 'goom_update':
goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
     goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
     ^
goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
     goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
     ^

https://bugzilla.gnome.org/show_bug.cgi?id=752053
2015-07-07 13:18:49 +03:00
Tim-Philipp Müller
19e7f188fa rtph261pay: fix indentation 2015-07-07 09:18:39 +01:00
Jimmy Ohn
2f016f3f9d rtph261pay: Fix uninitialized variable compiler error
endpos variable does not correctly understand in the
4.6.3 GCC version. So compile error appears when we do
compile rtph261pay using jhbuild.
This patch is fixed the compile error in 4.6.3 GCC version.

https://bugzilla.gnome.org/show_bug.cgi?id=751985
2015-07-07 09:18:06 +01:00
Jan Alexander Steffens (heftig)
439f98ed9a flvdemux: Handle seek flags properly
Allows for non-keyframe seeks.

https://bugzilla.gnome.org/show_bug.cgi?id=738570
2015-07-06 10:30:42 -04:00
Thiago Santos
f40c1f8b09 qtdemux: avoid looping reading the 'moof' atom forever
It gets stuck if it only finds a moof and no mfra/mfro or moov
atoms. Skip the moof to continue the parsing to have it either
play or error out.

https://bugzilla.gnome.org/show_bug.cgi?id=745089
2015-07-06 11:00:20 -03:00
Stian Selnes
a675e18935 rtph263pdepay: init debug category
https://bugzilla.gnome.org/show_bug.cgi?id=752012
2015-07-06 13:35:04 +03:00
Stian Selnes
d91ef9dcbf rtpv8depay: ignore reserved bit in payload descriptor
Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:

R: Bit reserved for future use.  MUST be set to zero and MUST be
   ignored by the receiver.

https://bugzilla.gnome.org/show_bug.cgi?id=751929
2015-07-06 12:03:51 +03:00
Stian Selnes
f682772898 rtph261pay: rtph261depay: Add documentation
https://bugzilla.gnome.org/show_bug.cgi?id=751982
2015-07-05 16:09:02 +01:00
Sebastian Dröge
ab77906a37 rtph261pay: Fix compiler warning
gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
   GObjectClass *gobject_class;
2015-07-03 14:29:16 +02:00
Sebastian Dröge
e0204938a8 rtph261depay: Let the base class push the buffer so it can deal with the flow return 2015-07-03 14:15:31 +02:00
Sebastian Dröge
b653fae8c9 rtph261pay: Remove unused adapter 2015-07-03 14:15:29 +02:00
Sebastian Dröge
90d47bff9e speexpay: Directly attach payload to the output buffer instead of copying it 2015-07-03 14:00:04 +02:00
Sebastian Dröge
6675e33109 sbcpay: Attach payload directly to the output instead of copying 2015-07-03 14:00:04 +02:00
Stian Selnes
ef8d630a59 rtp: add H.261 RTP payloader and depayloader
Implementation according to RFC 4587.

Payloader create fragments on MB boundaries in order to match MTU size
the best it can. Some decoders/depayloaders in the wild are very strict
about receiving a continuous bit-stream (e.g. no no-op bits between
frames), so the payloader will shift the compressed bit-stream of a
frame to align with the last significant bit of the previous frame.

Depayloader does not try to be fancy in case of packet loss. It simply
drops all packets for a frame if there is a loss, keeping it simple.

https://bugzilla.gnome.org/show_bug.cgi?id=751886
2015-07-03 11:48:41 +01:00
Sebastian Dröge
9dfae82566 rtpmpvdepay: Don't forget to unmap the input buffer 2015-07-03 12:19:05 +02:00
Sebastian Dröge
7e1d28d27f rtpmpvpay: Create buffer lists instead of pushing each buffer individually 2015-07-03 12:15:10 +02:00
Sebastian Dröge
f67bafb90d rtpmpapay: Use buffer lists instead of pushing each fragment individually 2015-07-03 12:04:18 +02:00
Sebastian Dröge
002bba37f7 rtpmp4apay: Create buffer lists and don't copy payload memory 2015-07-03 12:00:26 +02:00
Miguel París Díaz
5ae672fd22 rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
When there are a lot of small gaps, we can consider that there is
a big gap (too losses) to reset the buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=751636
2015-07-02 18:38:46 +02:00
Sebastian Dröge
3df0cce65d rtpjitterbuffer: If possible, always update the current time before looping over all timers
If we have a clock, update "now" now with the very latest running time we have.
If timers are unscheduled below we otherwise wouldn't update now (it's only updated
when timers expire), and also for the very first loop iteration now would otherwise
always be 0.

Also the time is used for the timeout functions, e.g. to calculate any times
for the next timeouts and we would otherwise pass too old times there.

https://bugzilla.gnome.org/show_bug.cgi?id=751636
2015-07-02 16:45:59 +02:00
Sebastian Dröge
6a59cc4b76 rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it 2015-07-02 12:26:03 +02:00
Sebastian Dröge
9ceb15bcf8 rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us 2015-07-02 09:49:44 +02:00
Sebastian Dröge
8b0d11a0ee rtph263pay: Stop using an adapter and directly use the buffer
We always pushed one buffer into the adapter, then handled exactly that one
buffer and flushed it from the adapter. Now also don't memcpy() the actual
payload but just attach the input buffer's data to the output buffer.

This code still needs some serious refactoring/rewriting.
2015-07-02 09:26:27 +02:00
Sebastian Dröge
51cd22c912 rtpgsmpay: Remove non-existing includes for now
git add -p mistake.
2015-07-01 21:57:28 +02:00