Commit graph

8313 commits

Author SHA1 Message Date
Sebastian Dröge
983f57dc7d rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
and 4:4:4 formats.
2015-08-08 17:43:03 +02:00
George Kiagiadakis
2e590a32eb rtpklv(de)pay: add "RTP" in the klass string
GstRTSPMedia uses this classification to detect the real payloader
inside a dynpay bin and asserts if it doesn't find it, therefore
it is required

https://bugzilla.gnome.org/show_bug.cgi?id=753325
2015-08-07 10:15:05 +02:00
Hyunjun Ko
b0d6020862 rtprtxsend: print valid type where guint32 is expected
https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-08-06 01:39:43 -03:00
Hyunjun Ko
5a17572119 rtppayload: set standard payload type as default
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.

Also use the PT constants from the rtp lib when possible

https://bugzilla.gnome.org/show_bug.cgi?id=747965
2015-08-06 01:38:43 -03:00
Thiago Santos
e0878d6325 qtdemux: store the moof-offset also for push mode
It will be used in some cases for getting the correct offsets
from trun atoms.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2015-08-05 18:12:45 -03:00
Thiago Santos
bb336840c0 qtdemux: handle default-base-is-moof flag
Handle the flag from the tfhd that signals the base offset to
start from the moof atom

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2015-08-05 18:12:45 -03:00
Glen Diener
cd57697a2c matroskademux: Preserve forward referenced track tags
https://bugzilla.gnome.org/show_bug.cgi?id=752850
2015-08-05 16:46:33 -04:00
Sebastian Dröge
c9ea95481c rtpstreamdepay: Only allow activation in push mode
We need a proper caps event from upstream with the full RTP caps as we can't
create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
a filesrc or any other element that supports pull mode.

https://bugzilla.gnome.org/show_bug.cgi?id=753066
2015-08-04 21:00:31 +03:00
Sebastian Dröge
9ae316974d rtph264depay: Put the profile and level into the caps 2015-08-04 12:45:06 +03:00
Sebastian Dröge
8dda570e47 rtph264depay: Only update the srcpad caps if something else than the codec_data changed
h264parse does the same, let's keep the behaviour consistent. As we now
include the codec_data inside the stream too here, this causes less caps
renegotiation.
2015-08-04 12:45:06 +03:00
Sebastian Dröge
e0c124f76d rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
The spec says:

When a picture parameter set NAL unit with a particular value of
pic_parameter_set_id is received, its content replaces the content of the
previous picture parameter set NAL unit, in decoding order, with the same
value of pic_parameter_set_id (when a previous picture parameter set NAL unit
with the same value of pic_parameter_set_id was present in the bitstream).
2015-08-04 12:45:06 +03:00
Thiago Santos
72212198c7 splitmuxsink: remove extra \n at debug message 2015-08-03 13:46:16 -03:00
Thiago Santos
a930a1364a splitmuxsink: prevent deadlock when states change too fast
If the GOP is completed, pads have to start gathering for the
next one but it is possible that the the state might go to
COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
thread has a chance to wake up and proceed, leaving it trapped in
the check_completed_gop loop and deadlocking the other threads
waiting for it to advance.

To solve it, this patch also checks that tha input running time
hasn't changed to prevent this scenario.
2015-08-03 13:46:16 -03:00
Sebastian Dröge
ef7863355c rtph264depay: Insert SPS/PPS NALs into the stream
h264parse does the same and this fixes decoding of some streams with 32 SPS
(or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.

This looks like a mistake in the part of the spec about the codec_data.
2015-08-03 18:24:18 +03:00
Vineeth TM
cf19525d5c rtspsrc: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-30 15:51:25 +03:00
Vineeth TM
969bcf25a1 rtpmp4vdepay: rtpbuffer is being unref'ed twice
process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
the refernce should not be removed here

https://bugzilla.gnome.org/show_bug.cgi?id=753042
2015-07-30 12:20:19 +01:00
Sebastian Dröge
39a90710b7 rtspsrc: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 14:31:49 +01:00
Jan Schmidt
a0182dd943 splitmuxsink: Support mpegtsmux as a muxer.
As a fallback, look for a pad template sink_%d on
the muxer when requesting pads, to support mpegtsmux

https://bugzilla.gnome.org/show_bug.cgi?id=752999
2015-07-29 23:03:30 +10:00
Jan Schmidt
e7ec32801a splitmuxsrc: Use a separate lock to delay typefind.
Don't hold the main splitmux part lock over
the parent state change function, as it prevents
posting error messages that happen. Since the purpose
is to prevent typefinding from proceeding, use a
separate mutex just for that.
2015-07-29 23:03:18 +10:00
Vineeth TM
72b86ae868 matroska: fix memory leak
After adding to tag list, key_val is not being free'd
resulting in memory leak

https://bugzilla.gnome.org/show_bug.cgi?id=752992
2015-07-29 09:14:31 +01:00
Manasa Athreya
e6381ef285 qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
be more bits as well, in which case it's just like 'twos'.

https://bugzilla.gnome.org/show_bug.cgi?id=752613
2015-07-27 19:06:43 +01:00
Olivier Crête
7917bea855 avidemux: Stop without posting error on flushing
This could just be a normal pipeline shutdown.
2015-07-25 03:25:28 -04:00
Dimitrios Christidis
744167056c matroskademux: fix for subtitle buffers with NUL terminators
Commit 45892ec8 created a regression where g_utf8_validate() would fail
if the subtitle buffer had a NUL terminator as part of the data.

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-21 14:25:12 +01:00
Stian Selnes
45e05706e2 rtpvp8depay: Check available bytes before copy
Need to check that the number of bytes we want to copy from the adapter
actually is available and handle the error case gracefully. This error
may happen if malformed packets are received and we don't have a
complete frame.

https://bugzilla.gnome.org/show_bug.cgi?id=752663
2015-07-21 13:14:01 +01:00
Paul Hyunil
3740e69957 qtdemux: Support subtitle when track subtype is fourcc_subt
https://bugzilla.gnome.org/show_bug.cgi?id=752655
2015-07-21 12:24:15 +01:00
Havard Graff
764bbf99a8 rtpmux: handle different ssrc's on sinkpads
Do this by not putting the ssrc from the src pads in the caps used to
probe other sinkpads, and then  intersecting with it later.

https://bugzilla.gnome.org/show_bug.cgi?id=752491
2015-07-16 16:46:11 -04:00
Tim-Philipp Müller
2e3a5ba227 Update mailing list address from sourceforge to freedesktop 2015-07-16 17:19:03 +01:00
Dimitrios Christidis
45892ec8be matroskademux: fix trailing '*' displayed with some text subtitles
The subtitle buffer we push out should not include a NUL terminator
as part of the data, we just add such a terminator for safety, but
it should not be included in the buffer size.

A NUL terminator is not valid UTF-8, so checks will fail if it's
included in the size, and the NUL will be replaced by the fallback
character specified when converting, i.e. '*'.

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-16 13:18:06 +01:00
Ravi Kiran K N
1c00801585 audiofx: Fix typo in example pipelines
Fix typo in example pipelines of audiowsincband and audioinvert.

https://bugzilla.gnome.org/show_bug.cgi?id=752416
2015-07-15 13:51:13 +01:00
George Kiagiadakis
bbfa46363c splitmuxsink: add a "format-location" signal that allows better control over filenames
In certain applications, splitting into files named after a base
location template and an incremental sequence number is not enough.

This signal gives more fine-grained control to the application to
decide how to name the files.

https://bugzilla.gnome.org/show_bug.cgi?id=750106
2015-07-14 18:45:49 +02:00
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