Commit graph

8551 commits

Author SHA1 Message Date
Vineeth TM
77c9e2cd4d matroska: read: fix tag list memory leak
gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
be unref'ed as soon as the tags are merged to the tocentry

https://bugzilla.gnome.org/show_bug.cgi?id=753904
2015-08-21 10:22:54 +03:00
Tim-Philipp Müller
29afa75858 multifilesrc: fix regression with starting from index set via index property
When we haven't started yet, set the start_index when we set the index property,
so that we start at the right index position after the initial seek. The index
property was never really meant to be for writing, but it used to work, so let's
support it for backwards compatibility.

https://bugzilla.gnome.org/show_bug.cgi?id=739472
2015-08-18 13:17:34 +01:00
Alex Ashley
5d99d0dfa0 qtdemux: fix offset calculation when parsing CENC aux info
Commit 7d7e54ce68 added support for
DASH common encryption, however commit
bb336840c0 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.

The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
2015-08-18 11:48:03 +01:00
Hyunjun Ko
38d269f80d rtp: copy metadata in the (de)payloaders which is missed before
https://bugzilla.gnome.org/show_bug.cgi?id=753706
2015-08-17 14:12:50 +02:00
Thiago Santos
5838940681 y4mencode: fix gst-launch version in documentation 2015-08-16 14:30:57 -03:00
Thiago Santos
a1aa942acf audioencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough

Elements: speexenc, wavpackenc, mulawenc, alawenc
2015-08-16 14:30:57 -03:00
Thiago Santos
1b27badcfd videoencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough

Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
2015-08-16 14:30:57 -03:00
Tim-Philipp Müller
a39bebb5fe mpegaudioparse: use new baseparse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 17:21:24 +01:00
Olivier Crête
b1dfe209c2 audioparsers: use new base parse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 17:02:19 +01:00
Tim-Philipp Müller
a042a98159 flacparse: use new baseparse API and fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 16:33:55 +01:00
Sebastian Dröge
e9aa4c7467 qtdemux: Use signed integer type to be able to check for negative subtraction results
CID 1315829
2015-08-16 13:04:02 +02:00
Luis de Bethencourt
1aee15050c rtpvorbisdepay: remove dead code
payload_buffer must be NULL in ignore_reserved. Check will always be false.

Introduced by b1089fb520

CID #1316476
2015-08-16 11:52:44 +01:00
Thiago Santos
1328289474 alawenc: port to AudioEncoder base class 2015-08-15 22:46:46 -03:00
Thiago Santos
65676c22ee audiodecoders: use default pad accept-caps handling
Avoids useless check of downstream caps when handling an
accept-caps query

Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
2015-08-15 11:46:34 -03:00
Thiago Santos
65d2af6462 alawdec: make error handling a bit nicer
Print the element along with the debug to make it easier to trace
the failures
2015-08-15 11:31:04 -03:00
Thiago Santos
7ab3178cc4 alawdec: port to audiodecoder base class
mulawdec was already ported, alawdec was left behind.
2015-08-15 11:06:02 -03:00
Thiago Santos
41a4b68390 qtdemux: only look for more samples in moofs in pull-mode
For playback of some fragmented formats with qtdemux it will
try to look for the next moof after finishing one but it is only
possible for pull-mode. For playback of streaming fragmented formats
such as DASH it should just not try to look for another moof but
instead wait for more data.

https://bugzilla.gnome.org/show_bug.cgi?id=752602

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2015-08-15 11:06:02 -03:00
Sebastian Dröge
64b06d1829 dcaparse: Don't look for a second syncword
There are streams out there that consistently contain garbage between
every frame so we never ever find a second consecutive syncword.

See https://bugzilla.gnome.org/show_bug.cgi?id=738237
2015-08-15 13:00:06 +02:00
Thiago Santos
9523fb23ed audioparsers: enable accept-template flag
Do a quick check with the pad template caps as it is enough. Users
should have figured the appropriate full caps on a previous caps query

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:42:27 -03:00
George Kiagiadakis
e2f2f087ec rtspsrc: send the User-Agent header
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.

This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"

https://bugzilla.gnome.org/show_bug.cgi?id=750101
2015-08-14 15:59:06 +02:00
George Kiagiadakis
af03341e26 rtspsrc: wrap gst_rtsp_message_init_request in a local function
This will allow adding common request initialization, like the
user agent string, in just one place.
2015-08-14 15:59:06 +02:00
Prashant Gotarne
0671ea85af audioecho: make sure buffer gets reallocated if max_delay changes
https://bugzilla.gnome.org/show_bug.cgi?id=753490
2015-08-14 11:50:22 +01:00
Ramiro Polla
23b5a34675 rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
Use constantDuration to calculate the timestamp of non-first AU in the
RTP packet.

If constantDuration is not present in the MIME parameters, its value
must be calculated based on the timing information from two consecutive
RTP packets with AU-Index equal to 0.

https://bugzilla.gnome.org/show_bug.cgi?id=747881
2015-08-14 12:38:32 +02:00
Sebastian Dröge
3ede3105d6 goom: Rename get_type() function of base class to prevent symbol conflicts
This is a problem when statically linking.
2015-08-14 09:21:25 +02:00
Sebastian Dröge
68a9209408 rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
Otherwise we will just output buffers without timestamps after a reset if no
timestamps are provided by upstream, e.g. when using RTSP over TCP.

https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-08-13 16:45:16 +02:00
Ravi Kiran K N
6eee26b24b matroska: Remove unused variable
https://bugzilla.gnome.org/show_bug.cgi?id=753556
2015-08-13 14:11:12 +01:00
Sebastian Dröge
b1089fb520 rtp: Copy metadata in the (de)payloader, but only the relevant ones
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
2015-08-11 12:47:23 +02:00
Thiago Santos
288b0bbb38 qtdemux: fix small typo in comment 2015-08-10 19:11:17 -03:00
Nicolas Dufresne
109995707e goom2k1/doc: Fixup previous commit 2015-08-10 16:19:18 -04:00
Nicolas Dufresne
18c43aa845 goom2k1/doc: Use GstGoom2k1 namespace
The doc generator isn't happy when we have class name clash. Simply
use it's own namespace.
2015-08-10 15:55:19 -04:00
Prashant Gotarne
cde1f12ad3 audioecho: removed unused variable in set_property
unused local variable 'delay' is removed.

https://bugzilla.gnome.org/show_bug.cgi?id=753450
2015-08-10 13:34:11 +01:00
Tim-Philipp Müller
604cc2a548 qtdemux: fix suboptimal queue iteration code 2015-08-10 12:45:50 +01:00
Tim-Philipp Müller
0fbf5f3d9e qtdemux: don't use glib 2.44-only API 2015-08-10 12:32:23 +01:00
Alex Ashley
7d7e54ce68 qtdemux: add support for ISOBMFF Common Encryption
This commit adds support for ISOBMFF Common Encryption (cenc), as
defined in ISO/IEC 23001-7. It uses a GstProtection event to
pass the contents of PSSH boxes to downstream decryptor elements
and attached GstProtectionMeta to each sample.

https://bugzilla.gnome.org/show_bug.cgi?id=705991
2015-08-10 12:32:17 +01:00
Hyunjun Ko
9c5c16eb57 rtph264depay: checking if depay has sps/pps nals before insertion
https://bugzilla.gnome.org/show_bug.cgi?id=753430
2015-08-10 10:49:02 +02:00
Tim-Philipp Müller
a95c761fde matroskamux: fix outdated comment
The default behaviour was changed in the 0.10 -> 1.x
transition, but the comment was not updated.
2015-08-08 16:44:49 +01:00
Sebastian Dröge
c8551b6285 rtptheorapay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:43:03 +02:00
Sebastian Dröge
4957cc7459 rtpvorbispay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:43:03 +02:00
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
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
Sebastian Dröge
ef5e14989b rtpgstpay: Use the return value of gst_buffer_append() 2015-07-01 21:39:25 +02:00
Sebastian Dröge
137672ff18 rtpgsmpay: Attach payload to the output buffer instead of copying it 2015-07-01 21:39:25 +02:00
Sebastian Dröge
cb0232ba4e rtpg729pay: Attach payload directly to output buffers instead of copying 2015-07-01 21:39:25 +02:00
Sebastian Dröge
0a71dbc80c rtpg723pay: Attach payload buffer to the output instead of copying 2015-07-01 21:39:25 +02:00
Sebastian Dröge
8aca30799a rtpdvdepay: Map the output buffer once instead of once every 80 bytes 2015-07-01 21:39:25 +02:00
Jimmy Ohn
4f4605f481 avidemux: fix return type of index_entry_offset_search()
It's a compare function and may return a negative value,
so should for correctness and consistency return a signed
integer.

https://bugzilla.gnome.org/show_bug.cgi?id=751780
2015-07-01 19:18:11 +01:00
Miguel París Díaz
2176f31174 rtpjitterbuffer: refactor handle_next_buffer
The goal of this patch is making handle_next_buffer function
more readable avoiding unnecesary gotos and adding other
cosmetic changes.
2015-07-01 16:06:40 +02:00
Sebastian Dröge
3af36ed8fe rtpac3pay: Attach the payload to the output buffer instead of copying it
Might also want to produce buffer lists here if needed.
2015-07-01 15:46:07 +02:00
Sebastian Dröge
adf2d8459f rtp: Fix indention 2015-07-01 15:46:06 +02:00
Sebastian Dröge
978903cd87 rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() 2015-07-01 11:58:26 +02:00
Sebastian Dröge
ceaf90f027 vp8depay: Don't lock/map every non-keyframe buffer twice
Just copy the complete header instead of first looking at the first byte
and then at the remaining 10 bytes.
2015-06-30 14:07:28 +02:00
Sebastian Dröge
de5cd0995b Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
This reverts commit 0c21cd7177.

If we have multiple immediate timers, we want to first handle the one with the
lowest sequence number... which would be broken now.

Instead of this we should just use a GSequence for the timers, and have them
sorted first by timestamp, and for equal timestamps by sequence number. Then
we would always only have to take the very first timer from the list and never
have to look at any others.
2015-06-29 10:36:58 +02:00
Sebastian Dröge
0c21cd7177 rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
If we have lots of such immediate timeouts, we would otherwise have quadratic
runtime in the number of timeouts.
2015-06-29 10:14:05 +02:00
Thiago Santos
121fcbf7da splitmuxsrc: sticky events are sent automatically from the pad
No need to send them explicitly from the element

https://bugzilla.gnome.org/show_bug.cgi?id=751240
2015-06-25 17:13:43 -03:00
Thiago Santos
af6a09ae4c splitmuxsrc: make sure to push sticky events before adding pad
It allows the caps to be set on the pad before being added for
dynamic autoplugging to work.

https://bugzilla.gnome.org/show_bug.cgi?id=751240
2015-06-25 17:13:43 -03:00
Hyunjun Ko
f560a3d223 rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
Enable to use new ntp-time-source property of rtpbin

https://bugzilla.gnome.org/show_bug.cgi?id=751496
2015-06-25 17:16:49 +02:00
Hyunjun Ko
a1bff413a1 rtpbin/session: fix description
https://bugzilla.gnome.org/show_bug.cgi?id=751496
2015-06-25 16:31:51 +02:00
Luis de Bethencourt
063f553275 docs: decodebin2 -> decodebin 2015-06-25 10:57:29 +01:00
Luis de Bethencourt
34caf9d7c5 deinterlace: update example pipeline
Update reference to decodebin2 to decodebin
2015-06-25 10:47:40 +01:00
Luis de Bethencourt
72f63c58ad deinterlace: remove dead assignments
Values in fields_required and same_buffer are overwritten before used. Removing
assignment
2015-06-25 10:46:43 +01:00
Gilbok Lee
0dcd76447a qtdemux: does not detect orientation
Most files don't contain the values for transposing the coordinates
back to the positive quadrant so qtdemux was ignoring the rotation
tag. To be able to properly handle those files qtdemux will also ignore
the transposing values to only detect the rotation using the values
abde from the transformation matrix:

[a b c]
[d e f]
[g h i]

https://bugzilla.gnome.org/show_bug.cgi?id=738681
2015-06-25 00:24:21 -03:00
Nicolas Dufresne
2359ee29e8 qtmux: Correctly calculate the elst media start
The media start has nothing to do with the shift we have applied
but with the value of the first PTS. This is defined as:

  Dt(0) = 0
  Ct(0) = Dt(0) + CTTS(0)

So the media start is always the first CTTS.

https://bugzilla.gnome.org/show_bug.cgi?id=751361
2015-06-23 22:34:36 -04:00
Thiago Santos
eceb2ccc73 qtdemux: accumulate previous edts entries into segment.base
Allows playing edts editted files with proper synchronization of
streams. This patch fixes the regression introduced by
bf95f93c01 that was added to fix
segment seeks handling.

Having the accumulated_base separated from the main segment.base
allows handling both segment seeks and edts editted files.

https://bugzilla.gnome.org/show_bug.cgi?id=751361
2015-06-23 22:34:36 -04:00
Thiago Santos
aef61c2251 qtdemux: improve some debug messages
Those messages are about the stream, use the pad as the
debug object to make it clear from the logs

https://bugzilla.gnome.org/show_bug.cgi?id=751361
2015-06-23 22:34:35 -04:00
Thiago Santos
1ec9a86e72 qtmux: store last_dts of the first buffer
Buffers need not to start at running-time 0 so the last_dts needs
to be the value of the first buffer's dts as it is used to compute
the duration of the buffers. If it was left at 0 the first buffer
would have a larger duration when it shouldn't

https://bugzilla.gnome.org/show_bug.cgi?id=751361
2015-06-23 22:34:35 -04:00
Vineeth TM
e44ce40455 flacparse: fix possible memory leak
when buffer is stored to seektable, and stop gets called due to
corrupt flac file, then the seektable is not being released

https://bugzilla.gnome.org/show_bug.cgi?id=751364
2015-06-23 10:17:53 +02:00
Jan Schmidt
b26bbae695 Revert "splitmuxsink: Mask async-start/done while switching files."
This reverts commit d61e5393f1.

Causes failures muxing larger GOP sizes for some reason. Reverting
while I figure it out
2015-06-23 17:33:03 +10:00
Jan Schmidt
600bab0056 splitmuxsrc: Fix startup and shutdown races.
Fix 2 startup races when things happen too quickly, and 1
at shutdown by holding a ref to the pads in use until the
loop functions exit.

Handle errors activating file parts and publish them on
the bus.

https://bugzilla.gnome.org/show_bug.cgi?id=750747
2015-06-23 12:03:14 +10:00
Jan Schmidt
d61e5393f1 splitmuxsink: Mask async-start/done while switching files.
Sometimes, extra async-start/done from the internal sink
while the element is still starting up can cause splitmuxsink
to stall in PAUSED state when it has been set to PLAYING
by the app. Drop the child's async-start/done messages while
switching, so they don't cause state changes at the
splitmuxsink level.

https://bugzilla.gnome.org/show_bug.cgi?id=750747
2015-06-23 12:03:14 +10:00
Jan Schmidt
e5db2673bd matroska-demux: Use gst_video_multiview_guess_half_aspect()
Use the gst_video_multiview_guess_half_aspect() utility function
to set the half-aspect flag (or not) on stereoscopic frame-packed
videos.
2015-06-23 11:58:41 +10:00
Jan Schmidt
b7cbfe1fa1 qtdemux: Move multiview caps calculations, add half-aspect heuristics
Move the multiview caps calculations to the configure_stream()
function, so the rest of the video info is available, and
use the gst_video_multiview_guess_half_aspect() function to
determine if the half-aspect flag should be set on frame-packed
video.
2015-06-23 11:58:41 +10:00
Nicolas Dufresne
dd72267a8d qtdemux: Add cslg support
The cslg atom provide information about the DTS shift. This is
needed in recent version of ctts atom where the offset can be
negative. When cslg is missing, we parse the CTTS table as proposed
in the spec to calculate these values.

In this implementation, we only need to know the shift. As GStreamer
cannot transport negative timestamps, we shift the timestamps forward
using that value and adapt the segment to compensate. This patch also
removes bogus offset of ctts_soffset, this offset shall be included
in the edit list.

https://bugzilla.gnome.org/show_bug.cgi?id=751103
2015-06-22 17:51:49 -04:00
Nicolas Dufresne
7b8615d4fc qtmux: Use PTS to figure-out presence of gaps
We need to look at the presentation timestamp in order to conclude if
there is a gap at the start of a stream.

https://bugzilla.gnome.org/show_bug.cgi?id=751242
2015-06-22 17:45:30 -04:00
Nicolas Dufresne
feda525591 qtmux: Set edit list to compensate DTS shift
We shift DTS forward to avoid negative timestamps which cannot be
represented with version 0 of the CTTS table. To stick with that
version (backward compatibility), the spec recommend using an
edit list entry to move back the presentation time to where it
should be.

https://bugzilla.gnome.org/show_bug.cgi?id=751242
2015-06-22 17:45:30 -04:00
Nicolas Dufresne
bbea34bb6e flvmux: Insert AVC end of sequence
This FLV specific mark is needed to prevent Flow Player (most likely
all Flash base player) from going into buffering state when near EOS.

https://bugzilla.gnome.org/show_bug.cgi?id=751320
2015-06-22 17:24:16 -04:00
Vineeth TM
9a1ed36b7a matroska: remove useless check
No need to check for context availability while freeing. We are inside
inside a code block with a condition that dereferences context.
if (context->type == 0 ...

https://bugzilla.gnome.org/show_bug.cgi?id=751306
2015-06-22 12:26:23 +01:00
Vineeth T M
e97df1e097 lzo: fix memory leak
the opened file is not being closed during test, which will result
in memory leak.

https://bugzilla.gnome.org/show_bug.cgi?id=751306
2015-06-22 12:22:06 +01:00
Sangkyu Park
2663388000 rtpjitterbuffer: Minor clean-up
1. Fix the code which is wrong coding style.
2. Fix a typing error of comment.

https://bugzilla.gnome.org/show_bug.cgi?id=751316
2015-06-22 13:08:12 +02:00
Jose Antonio Santos Cadenas
11f298a338 rtpsource: Do not try to push NULL buffers
If update_receiver_stats() fails, we can't really do anything with this buffer
anymore and have to drop it. This happens if there's a big seqnum
discontinuity for example.

https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-06-22 12:26:59 +02:00
Vineeth TM
636d47d2d6 flvdemux: trivial cleanup
trivial patch to add proper ( while checking for if(G_UNLIKELY())

https://bugzilla.gnome.org/show_bug.cgi?id=751306
2015-06-22 11:13:32 +01:00
Vineeth TM
78fcd03ca3 dcaparse: initialize size variable
size can be used in cleanup without being initialized. Hence
setting it to 0 when declaring

https://bugzilla.gnome.org/show_bug.cgi?id=751306
2015-06-22 10:58:35 +01:00
Vineeth TM
331fca4dfb mpegaudioparse: initialze bpf variable
bpf variable might be used in cleanup without being intialized.

https://bugzilla.gnome.org/show_bug.cgi?id=751306
2015-06-22 10:57:35 +01:00
Miguel París Díaz
40957a9212 rtprtxqueue: reverse pending list before pushing buffers
With this we send the RTX buffers in the same order
that they were requested.

https://bugzilla.gnome.org/show_bug.cgi?id=751297
2015-06-22 11:36:22 +02:00
Nicolas Dufresne
212f39ee1d flvmux: Fix DTS validity check
This check was up-side-down, causing a bad timestamp at start
and then all timestamp being delayed.

https://bugzilla.gnome.org/show_bug.cgi?id=751298
2015-06-21 19:23:22 -04:00
Nicolas Dufresne
d6e1e744a7 cslg: Add Composition Shift Least Greatest Atom
This simply add fourcc and dump function for the cslg Atom.

https://bugzilla.gnome.org/show_bug.cgi?id=751103
2015-06-17 15:21:16 -04:00
Nicolas Dufresne
8a406c9c38 ctts_dump: Fix signess issues
It didn't bug, but use correct signess in traces. The number of
entries is unsigned while the offset can be signed according to
recent spec.

https://bugzilla.gnome.org/show_bug.cgi?id=751103
2015-06-17 15:21:16 -04:00
Sebastian Dröge
e9902430da rtpjitterbuffer: gst_rtp_buffer_ext_timestamp() modifies its first argument, keep a copy around 2015-06-16 11:43:39 +02:00
Sebastian Dröge
62a7bcb395 rtpjitterbuffer: Compare ext RTP times, not plain RTP time and ext RTP time when calculating elapsed time
Otherwise all RTP times after a wraparound would be considered as going
backwards, they will always be smaller than the ext RTP time.
2015-06-16 10:31:47 +02:00
Sebastian Dröge
f4e01b13ee rtpbin: The default rtp-profile should be AVP, not AVPF 2015-06-15 19:25:12 +02:00
Sangkyu Park
6696bd62ef rtpjitterbuffer: Minor cleanup
1. Add Null check in 'free_item' function.
2. Fix a typing error of comment.

https://bugzilla.gnome.org/show_bug.cgi?id=750965
2015-06-15 11:55:57 +02:00
Nicolas Dufresne
717265ebfb flmux: Make sure best_time is initialized 2015-06-12 17:45:23 -04:00
Sebastian Dröge
dc513eb949 rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property
The new property allows to select the time source that should be used for the
NTP time in RTCP packets. By default it will continue to calculate the NTP
timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
clock time. The latter is especially useful for synchronizing multiple
receivers if all of them share the same clock.

If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
setting and continue to use the running time plus 70 years. This is only kept
for backwards compatibility.
2015-06-12 23:35:42 +02:00
Nicolas Dufresne
135e516730 qtdemux: Adjust segment according to ctts offset
In presence of a CTTS, the segment start/stop must be offset so
the segment start/stop include the PTS. This is needed since the
PTS cannot be negative in this format. This fixes issues where the
running time of the first buffer isn't at the start.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Nicolas Dufresne
12181efddc qtmux: Handle DTS with negative running time
As QT works with duration, simply bring back first DTS to 0 and shift
forward the PTS of the same amount.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Nicolas Dufresne
2274ca7d07 flvmux: Add negative runtime DTS support
This is done by using new feature of the CollectPad clip function
which sets the DTS as a gint64 in the collected data. It also simplify
the code a bit.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Sebastian Dröge
37e3ca1447 rtpbin: Rename some variables and debug output to make more sense
Local and remote were mixed up in a few places, and the time we store here is
not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds.
2015-06-12 23:07:27 +02:00
Jan Schmidt
0c46c5c3e2 matroska-demux: Actually set detected 3D info into output caps.
Use the information read from the StereoMode info
to configure multiview-mode and multiview-flags in the
video caps.
2015-06-12 01:57:36 +10:00
Jan Schmidt
3f39d06338 splitmuxsink: Take released-but-not-yet-output bytes into account
When deciding whether it's time to switch to a new file, take into
account data that's been released for pushing, but hasn't yet
been pushed - because downstream is slow or the threads haven't been
scheduled.

Fixes a race in the unit test and probably in practice - sometimes
failing to switch when it should for an extra GOP or two.

Also fix a problem in splitmuxsrc where playback sometimes
stalls at startup if types are found too quickly.

https://bugzilla.gnome.org/show_bug.cgi?id=750747
2015-06-12 01:57:36 +10:00
Thiago Santos
03f1a2ea67 atoms: remove custom gst_buffer_new function in favor of core version
Remove a custom specialized version of gst_buffer_new_wrapped by
using gst_buffer_new_wrapped_full inside a macro to simplify
parameters and give it a more meaningful name.
It is only used to create temporary buffers to have its data copied.
2015-06-11 01:11:31 -03:00
Thiago Santos
1596972674 atoms: simplify free form data atoms creation
Avoid creating an intermediary buffer or memory area just
to copy into an atom's data area.
2015-06-11 01:11:31 -03:00
Thiago Santos
ab18f5035c qtmux: add AC-3 muxing support
Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.

One extra feature that was added was the ability to add extension
atoms after set_caps as the AC-3 extension atom needs some data
that has to be extracted from the stream itself and is not
present on caps.
2015-06-11 01:11:31 -03:00
Thiago Santos
674e0cc2df qtmux: remove unused type MP4S 2015-06-11 01:11:31 -03:00
Thiago Santos
f83fd7a88f qtmux: remove duplicate attribute value set
It is also set a few lines below
2015-06-11 01:11:18 -03:00
Jan Schmidt
ec5bc9dccb matroska: Implement basic stereoscopic video support
Implement support for the packed video formats WebM
uses, not all the values that Matroska might use.

In practice, it's really hard to find any samples in the
wild of any.

Supported in both the muxer and demuxer.
2015-06-11 12:11:42 +10:00
Jan Schmidt
fff76157d8 qtdemux: Add basic support for MPEG-A stereoscopic video
The MPEG-A format provides an extension to the ISO base media
file format to store stereoscopic content encoded with different
codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
atom declares the presence and storage method for the video.

Stereo video information for MPEG-A can also be supplied through
the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
is not implemented in this patch.

Also missing is support for stereo video encoded as separate video tracks
for now.

Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>

https://bugzilla.gnome.org/show_bug.cgi?id=611157
2015-06-11 12:11:42 +10:00
Sebastian Dröge
dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00
Ilya Konstantinov
c7e168ec70 rtpmanager: clarify negative lost packets in stats
Also:
- Move notes on units before field documentation.
- Unify documentation style.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-10 14:10:52 +02:00
Vineeth TM
720ff75c72 qtdemux: fix reverse playback
When performing seek, segment->start is being updated with desired_offset,
but in case of reverse playback segment->start should be 0 and
segment->stop should be updated with desired offset.

https://bugzilla.gnome.org/show_bug.cgi?id=750675
2015-06-10 10:41:13 +02:00
Xavier Claessens
b0b3e8e2cc rtspsrc: Add a GTlsInteraction property
It can be used for TLS client authentication.

https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 20:03:18 -04:00
Ilya Konstantinov
0a578c235a rtpmanager: document units of stats and arguments
Also, minor spelling and style corrections.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-09 18:21:59 +02:00
Luis de Bethencourt
e56ef6bcf0 goom: possible uninitialized variables warning
Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might
possibly be used uninitialized. They are set depending on the cases of a switch
statement and the compiler sees this as not a complete guarantee.
Set them to 0 if the switch statement falls down to the default case.

https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6
2015-06-08 23:06:39 +01:00
Chris Clayton
e29f231e5d rtpvp8depay: potential access beyond end of array
Compiling (with gcc-4.9-20150603) produces an error because of an access beyond
the end of an array. This patch fixes the error by initializing the loop
control/array index variable (i) to 1 and returning i - 1 when a match is found.
Also, because the values stored in the array increase in value as the index
increases, the >= test unnecessary, so it is removed.
2015-06-08 20:16:20 +01:00
Jan Schmidt
d78502deb1 splitmuxsink: Don't accumulate more than 2 GOPs
Don't allow large amounts of data to queue up - we only need
the GOP we're writing, and the GOP we're accumulating.
2015-06-08 18:58:43 +10:00
Jan Schmidt
23d610140d isomp4: fsync after sending updates in robust mode
Use the new GstBuffer SYNC_AFTER flag to trigger an fsync
after updating the moov or mdat atom, and after updating the free
atom to make it visible.
2015-06-08 14:49:11 +10:00
Jan Schmidt
3e17cd8acb isomp4: Only set moov header into streamheader at EOS
Only update the moov header into the caps if it's the finalised
moov at EOS time. Avoids posting a bogus moov at startup and
repeated updates in robust-recording mode
2015-06-08 14:49:11 +10:00
Jan Schmidt
1d058c7d8a isomp4: Implement robust muxing using ping-pong strategy
Implement a robust recording mode, where the output
file is always in a playable state, seeking and rewriting
the moov header at a configurable interval. Rewriting
moov is done using reserved space at the start of
the file, and a ping-pong strategy where the moov
is replaced atomically so it's never invalid.

Track when tags have actually changed, and don't write them into
the moov unless they've changed. Clear any existing tags when
re-writing them, so we can do progressive moov updating in robust
recording mode.

Write placeholder mdat as a free atom plus a 32-bit mdat
with '0' size, which means "rest of the file" in the spec.

Re-write it later to a full 64-bit extended size atom if needed.
2015-06-08 14:49:11 +10:00
Jan Schmidt
3d7b343525 isomp4: Update edit list when re-writing moov
Correctly update any edit lists each time the moov is recalculated,
updating existing table entries if they already exist instead of just
adding new ones.
2015-06-08 14:16:36 +10:00
Jan Schmidt
0c1bcc629d isomp4: Remove an extra bracket in a comment. 2015-06-08 14:16:36 +10:00
Jan Schmidt
94e113c6c6 splitmuxsrc: Protect total_duration state variable with the object lock.
Prevent deadlocks from downstream querying duration from the streaming thread.
2015-06-08 14:16:36 +10:00
Luis de Bethencourt
0b8c7ab797 goom: clean dereferences of private structure
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-07 19:24:20 +01:00
Luis de Bethencourt
fce8e5fb26 goom2k1: clean dereferences of private structure
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-07 19:20:49 +01:00
Sebastian Dröge
a7faa3e0a2 Release 1.5.1 2015-06-07 10:46:34 +02:00
Sebastian Dröge
b549ebd066 rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property 2015-06-07 10:33:27 +02:00
Sebastian Dröge
d650a310da rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
https://bugzilla.gnome.org/show_bug.cgi?id=749581
2015-06-05 16:45:54 +02:00
Vineeth TM
0e5631c5c0 interleave: error when channel-positions-from-input=False
self->channels is being incremented only when
channel-positions-from-input is set as TRUE. So in case of FALSE
self->func is not set and hence creating assertion error.
Hence removing the condition to increment self->channels.

https://bugzilla.gnome.org/show_bug.cgi?id=744211
2015-06-05 08:48:25 -03:00
Sebastian Dröge
8f5bdf9690 rtpjitterbuffer: Add support for receiving reduced size RTCP
It worked before but gave warnings, now we just ignore RTCP
packets that don't start with a SR. As all we're interested
in here are SRs.
2015-06-05 10:33:11 +02:00
Jose Antonio Santos Cadenas
f563176349 rtpssrcdemux: Add support for reduce size rtcp
According to RFC 5506, reduce size packages can be sent, this
packages may not be compound, so we need to add support for
getting ssrc from other types of packages.

https://bugzilla.gnome.org/show_bug.cgi?id=750327
2015-06-05 10:30:15 +02:00
Jose Antonio Santos Cadenas
f8f23bbf5d rtpsession: Add support for receiving reduced size rtcp
See RFC 5506

https://bugzilla.gnome.org/show_bug.cgi?id=750332
2015-06-05 10:24:17 +02:00
Sebastian Dröge
ec82eba96b aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels
ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14.
2015-06-04 16:09:41 +02:00
Nicolas Dufresne
3ab70e4677 asteriskh263: Un-rank clashing depayloader
This depayloader clash with the standard one for H263p. It produces an
H263p stream with a modified header. It uses encoding-name that is the
same as H263p (H263-1998) though the resulting ES is not decodable or
parsable in GStreamer, making it unsuable in dynamic pipeline. This
patch unrank this specialized depayloader since it can only be used in
custom pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=739935
2015-06-03 08:57:57 -04:00
Luis de Bethencourt
ffe7507512 goom2k1: remove variables not needed anymore
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-02 18:09:48 +01:00
Luis de Bethencourt
8756b6a9d4 goom2k1: rebase to use the audiovisualizer class
Rebase to have goom2k1 using the common GstAudioVisualizer class

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-02 18:02:08 +01:00
Luis de Bethencourt
89903bf66a goom: rebase to use the audiovisualizer class 2015-06-02 17:47:57 +01:00
Sebastian Dröge
647eefea67 rtpsession: Only schedule a timer when we actually have to send RTCP
Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
feedback is actually pending and no regular RTCP has to be sent).

This improves CPU usage and battery life quite a lot.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
8ada98964d rtpsession: Remove useless goto
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0a7823b30f rtspsrc: Set RTP profile on the rtpsession objects
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
506a8a8857 rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0f7e80ed59 rtpsession: Only put RRs and full SDES into regular RTCP packets
If we may suppress the packet due to the rules of RFC4585 (i.e. when
below the t-rr-int), we can send a smaller RTCP packet without RRs
and full SDES. In theory we could even send a minimal RTCP packet
according to RFC5506, but we don't support that yet.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
6f830e5bd5 rtpsession: Keep track of tp/tn and t_rr_last separately
Otherwise we can't properly schedule RTCP in feedback profiles as we need to
distinguish the time when we last checked for sending RTCP (tp) but might have
suppressed it, and the time when we last actually sent a non-early RTCP
packet.

This together with the other changes should now properly implement RTCP
scheduling according to RFC4585, and especially allow us to send feedback
packets a lot if needed but only send regular RTCP packets every once in a
while.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
3122ef4ae3 rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
And modify our RTCP scheduling algorithm accordingly. We now can send more
RTCP packets if needed for feedback, but will throttle full RTCP packets by
rtcp-min-interval (t-rr-int from RFC4585).

In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
statically set to 1s or 0s by RFC4585. Tmin defines how often we should
send RTCP packets at most.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Olivier Crête
8fd3e0e125 mulawdec: Let baseclass estimate bitrate
This makes playback directly from a file work with the right caps.
2015-05-30 17:41:44 -04:00
Tim-Philipp Müller
2e5df10ed9 dynudpsink: keep GCancellable fd around instead of re-creating it constantly
And create it only when starting the element.
2015-05-27 17:08:47 +01:00
Tim-Philipp Müller
b33d30621c udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every time we call g_socket_condition_timed_wait() or
g_socket_send_message(s)(), i.e. a lot. Which is not
particularly good for performance.

Can't create GCancellable in ::start() here because it's used
in client_new() which may be called via the add-client action
signal which may be called before the element is up and running.
2015-05-27 17:08:47 +01:00
Tim-Philipp Müller
11bb21f3c2 udpsrc: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every single time we call g_socket_condition_timed_wait() or
g_socket_receive_message(), i.e. twice per packet received!
This was not particularly good for performance.

Also only create GCancellable on start-up.
2015-05-27 17:08:47 +01:00
Luis de Bethencourt
6d06a74f7f matroska: overwritten value assignment
curpos is set and immediately after, set again. Remove the redundant
assignment.

https://bugzilla.gnome.org/show_bug.cgi?id=749909
2015-05-27 16:56:15 +01:00
Tim-Philipp Müller
80998dadba rtpvrawdepay: don't shadow existing outbuf variable
And fix unref of the wrong one which will contain NULL
in an error code path.
2015-05-25 16:16:47 +01:00
Tim-Philipp Müller
2aafb3951d rtpvrawdepay: map/unmap output frame only once, not for every input packet
Map output buffer after creating it and keep it mapped
until we're done with it instead of mapping/unmapping
it for every single input buffer.
2015-05-25 16:16:42 +01:00
Thiago Santos
d03b9513f1 qtdemux: remove fixme from 2006
It has been verified by use over time.
2015-05-25 08:47:47 -03:00
Thiago Santos
fc0a184592 qtdemux: fix reverse playback of fragmented media
qtdemux creates a samples array and gets the timestamps for buffers by
accumulating their durations. When doing reverse playback of fragments,
accumulating samples will lead to wrong timestamps as the timestamps
should go decreasing from fragment to fragment and the accumulation
will produce wrong results.

In this case, when receiving a discont for fragmented reverse playback,
the previous samples information should be flushed before new data
is processed.
2015-05-25 08:46:18 -03:00
Jimmy Ohn
d3997773fc splitfilesrc: Implement binary search in find_part_for_offset
Implement binary search using gst_util_array_binary_search

https://bugzilla.gnome.org/show_bug.cgi?id=749690
2015-05-25 14:23:32 +10:00
Sebastian Dröge
565cd49643 rtpsession: Don't crash if we receive FIR/PLI from a source we don't know 2015-05-21 13:26:53 +03:00
Santiago Carot-Nemesio
2fb1fe2ee3 rtpsession: Fix collection of statistics
Stats should be collected on the media rtp source not in the
sender one.

https://bugzilla.gnome.org/show_bug.cgi?id=749669
2015-05-21 12:56:12 +03:00
Edward Hervey
27c91bc881 multifilesink: Add a new max-duration file switching mode
This new mode ensures that files will never exceed a certain duration
based on incoming buffer PTS (and duration if present)

Note:
* You need timestamped buffers (duh). If some of the incoming buffers don't
  have PTS, then it will just accept them in the current file
2015-05-20 15:50:07 +02:00
Edward Hervey
f1ceaab02f multifilesink: streamline the file-switch code a bit
Use the same functions regardless of the mode we are using
2015-05-20 15:50:07 +02:00
Edward Hervey
db0abbd531 multifilesink: add "aggregate-gops" property to process GOPs as a whole
This property can be used in combination with next-file=max-size
(and perhaps a future next-file=max-duration) to make sure that
each file part starts cleanly with a key frame and the appropriate headers.

In order for this property to work correctly, upstream elements should make
sure than any headers that need to be written in a standalone file are:
1) in the streamheader caps field
2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
   that are just before the keyframe buffer

This is useful for MPEG-TS/MPEG-PS file segmenting in
combination with mpegtsmux or mpegpsmux.

Original patch by: Tim-Philipp Müller <tim@centricular.com>
2015-05-20 15:49:57 +02:00
Sebastian Dröge
9b14170355 rtspsrc: Use single-include header for the RTSP library 2015-05-20 16:37:55 +03:00
Tim-Philipp Müller
f54110fd3e udp: don't use soon-to-be-deprecated g_cancellable_reset()
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."

https://bugzilla.gnome.org/show_bug.cgi?id=739132
2015-05-19 19:00:20 +01:00
Stefan Sauer
168881a186 Revert "doc: Workaround gtkdoc issue"
This reverts commit 1797c8f8b1.

This is fixed by the gtk-doc 1.23 release.
<para> cannot contain <refsect2>:
http://www.docbook.org/tdg/en/html/para.html
http://www.docbook.org/tdg/en/html/refsect2.html
2015-05-18 20:13:01 +02:00
Nicola Murino
5e226d63f9 rtpg726pay: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=749544
2015-05-18 17:40:55 +01:00
Nicola Murino
335afc982b rtpg726depay: don't leak input buffer
https://bugzilla.gnome.org/show_bug.cgi?id=749543
2015-05-18 17:40:39 +01:00
Sebastian Dröge
c60038f188 rtpsource: Queue bad packets instead of dropping them
So we can send them out once we found the next, consecutive sequence number in
case one is following.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
9f18a271f3 rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 18:43:16 +03:00
Sebastian Dröge
54e924332e rtpsource: Refactor seqnum comparison code a bit 2015-05-18 18:43:16 +03:00
Sebastian Dröge
1974b24ef4 rtpsource: Allow sequence number wraparound during probation 2015-05-18 18:43:16 +03:00
Sebastian Dröge
3386de7a8a rtpsource: Make sequence number comparison code more readable
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
ca110fb0b8 rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.

https://bugzilla.gnome.org/show_bug.cgi?id=747922
2015-05-18 18:43:15 +03:00
Nicolas Dufresne
1797c8f8b1 doc: Workaround gtkdoc issue
With gtkdoc 1.22, the XML generator fails when a itemizedlist is
followed by a refsect2. Workaround the issue by wrapping the
refsect2 into para.
2015-05-16 23:37:06 -04:00
Stefan Sauer
426eb3e300 qtdemux: avoid wrong warnings on unknown node types
Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
trigger a warning.
2015-05-15 14:56:07 +02:00
Nicola Murino
fefeda5e6c rtpg726depay: add block_align to output caps
It is needed to correctly negotiate caps with matroskamux
and most other muxers.

https://bugzilla.gnome.org/show_bug.cgi?id=749129
2015-05-13 12:39:07 +01:00
Sebastian Dröge
e11a537b65 audiofxbasefirfilter: Fix time-domain convolution with >1 channels
input_samples is the number of frames, but we used it as the number of
samples.

https://bugzilla.gnome.org/show_bug.cgi?id=747204
2015-05-12 13:41:58 +03:00
Tim-Philipp Müller
2e412a447a docs: update example pipelines in element docs
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
2015-05-10 11:05:00 +01:00
Tim-Philipp Müller
3755409409 splitmuxsrc: minor error message clean-up
Don't put filename in error message shown to user.
2015-05-10 10:53:13 +01:00