Commit graph

8347 commits

Author SHA1 Message Date
Luis de Bethencourt
e87113781a rtph263pay: remove uneeded variable
We just need to save the ebit information in case there is an error decoding.
2015-03-09 16:17:45 +00:00
Luis de Bethencourt
db3ade5bfb matroska: error mode if can't push buffer
If gst_pad_push() fails, inform and return flow error.
2015-03-09 12:51:21 +00:00
Luis de Bethencourt
f494da89b4 matroska: unused value
Value set in ret will be overwritten just before exiting the function.

CID #1226469
2015-03-09 12:13:40 +00:00
Sebastian Dröge
9e934d076b rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
2015-03-09 11:10:35 +01:00
Linus Svensson
398296d978 rtspsrc: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2015-03-09 10:18:35 +01:00
Sebastian Dröge
38bf3d3808 rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams 2015-03-09 10:05:14 +01:00
Mark Nauwelaerts
d0587467fc avidemux: resurrect some flow return handling 2015-03-07 20:22:33 +01:00
Nicolas Huet
5ead23a14a aacparse: fix LOAS parsing issue
Fix missing index in syncword searching

https://bugzilla.gnome.org/show_bug.cgi?id=745585
2015-03-06 14:34:08 -03:00
Jan Schmidt
b0ce43cde3 splitmuxsink: Protect property variables with the object lock.
Use the object lock instead of the splitmux lock to protect
internal property variables, so they're not locked when
switching to a new file.

https://bugzilla.gnome.org/show_bug.cgi?id=744420
2015-03-07 00:55:47 +11:00
Sebastian Dröge
c34a7cb90d rtspsrc: Fix handling of interleaved (TCP) streams
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.

https://bugzilla.gnome.org/show_bug.cgi?id=745599
2015-03-05 12:15:04 +01:00
Sebastian Dröge
b4aaa11f97 rtspsrc: Don't unref caps we don't own 2015-03-05 09:56:37 +01:00
Sebastian Dröge
297d808acc rtspsrc: Push RTCP caps on the RTCP pads
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
2015-03-05 09:47:29 +01:00
Sebastian Dröge
788074733c rtspsrc: Make sure to send SEGMENT events on all pads 2015-03-05 09:47:29 +01:00
Santiago Carot-Nemesio
e05378ec16 rtp: Add Full Intra Request (FIR) packets to statistics
https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:40 +01:00
Santiago Carot-Nemesio
22791413f9 rtp: Add Packet Loss Indication (PLI) to statistics
This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.

https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:07 +01:00
Nicola Murino
c4e542de69 matroskamux: Remove duration accumulation logic
Duration accumulation can cause rounding errors and generate wrong
duration with different buffers that share the same timestamp.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:37:48 +01:00
Nicola Murino
f727762c1f matroska: Add an helper method to get buffer timestamps
... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
that return PTS or DTS based on stream type.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:36:24 +01:00
Sebastian Dröge
8984e18ef7 rtpsession: Add explanation why we have space for 32 hash tables
And also create only one, there's no need yet to create all 32 until
we implement RFC2762.
2015-03-04 11:30:43 +01:00
Sebastian Dröge
af2bdd6e15 Revert "rtpsession: Do not use an array of maps if they are not being used"
This reverts commit 1591adf4cd.

https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
2015-03-04 11:26:57 +01:00
Santiago Carot-Nemesio
1591adf4cd rtpsession: Do not use an array of maps if they are not being used
rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession

https://bugzilla.gnome.org/show_bug.cgi?id=745586
2015-03-04 11:25:30 +01:00
Jimmy Ohn
42599eab76 avidemux: remove not needed code
In gst_avi_demux_handle_src_query, there is not needed code.
We already check about stream is vbr or not at the upper line.
o, we don't need to check this condition becase stream is not
vbr 100% in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=745276
2015-03-04 10:08:21 +01:00
Matej Knopp
f75e443a7a qtdemux: fix key unit seek
Unlike many other seek flags, the KEY_UNIT seek
flag is not copied over into the GstSegment,
since it's only relevant for the seek itself,
so we need to pass it explicitly to the seek
handler here.

https://bugzilla.gnome.org/show_bug.cgi?id=745339
2015-03-01 13:06:55 +00:00
Nicola Murino
e676b8ba9c matroskamux/demux: initialize dts_only
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Nicola Murino
09b8f0efc3 matroskamux: store DTS for V_MS/VFW/FOURCC streams
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Tim-Philipp Müller
f5b511b42b multifile: attempt to fix docs build issue on build bot 2015-02-26 19:48:33 +00:00
Arun Raghavan
0c06553fb2 interleave: Drop custom latency query handling
This is implemented by the default query handler now.
2015-02-27 00:59:43 +05:30
Arun Raghavan
dbc142afec videomixer: Drop custom latency querying logic
This is now implemented in the default latency query handler.
2015-02-27 00:59:43 +05:30
Sebastian Rasmussen
d331d931db rtpvorbispay: fix payloader description and author e-mail
https://bugzilla.gnome.org/show_bug.cgi?id=745226
2015-02-26 15:57:08 +00:00
Matej Knopp
fa283f407f matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
When such stream is present demuxer should set DTS on buffers instead
of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
streams.

Sample file
https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-26 11:12:34 +02:00
Krzysztof Kotlenga
e3ca4d1c86 rtspsrc: improve error message when unauthorized
Make use of NOT_AUTHORIZED error code instead of falling back to generic
READ error.

https://bugzilla.gnome.org/show_bug.cgi?id=601733
2015-02-24 11:08:27 +02:00
Thibault Saunier
fa0870658d qtdemux: All segment resulting from a seek should have the same seqnum
https://bugzilla.gnome.org/show_bug.cgi?id=744983
2015-02-23 20:05:20 +01:00
Vincent Penquerc'h
dc73d153cb rtpvp8pay: default encoding name to VP8
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:29:02 +00:00
Vincent Penquerc'h
b88ea286d2 rtpvp8pay: make caps writable before truncating them
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:06:51 +00:00
Vincent Penquerc'h
b866c989f5 rtpvp8pay: negotiate encoding name
Chrome uses a different one than gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 13:52:29 +00:00
Sebastian Dröge
939a95d44b rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
Otherwise we will just send buffers on the pad without any events beforehand
and will get g_warnings() about that.
2015-02-19 13:34:47 +02:00
Thiago Santos
84b7cf6795 qtmux: remove not needed condition
gst_buffer_replace can handle NULL inputs by itself
2015-02-18 10:36:06 -03:00
Thiago Santos
a12e41c106 qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
The tfdt should be more accurate as the buffer timestamp is provided
by the fragmented format manifest and it might just be an approximation.
2015-02-18 09:57:48 -03:00
Sebastian Dröge
735c6c40f8 rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
confuse downstream with buffers that come before such events.
2015-02-17 16:57:55 +02:00
Edward Hervey
6798dc7912 isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
2015-02-17 12:31:06 +01:00
Luis de Bethencourt
ea1d67abe3 goom2k1: use fractional part of float division 2015-02-16 14:31:05 +00:00
Luis de Bethencourt
4af5a2b760 splitmuxsin: remove dead code
Every instance of goto beach has buf_info equal NULL. Don't check
for a condition that never happens.

CID #1268399
2015-02-16 13:59:17 +00:00
Nicolas Dufresne
b8142bde07 spectrum: Fix min and max for bands property
The number of FFTs is calculated with the following formula:

  guint nfft = 2 * bands - 2;

nfft is passed to gst_fft_f32_new() as the len argument and is of type
unsigned integer. This method required that len is at leas 1, then
maximum G_MAXINT, as other values would be negative. If we extrapolate
from the formula above it means we need "bands" to be between 2 and
((guint)G_MAXINT + 2) / 2).

https://bugzilla.gnome.org/show_bug.cgi?id=744213
2015-02-15 21:34:28 -05:00
Thiago Santos
afa5481c50 qtdemux: do not use sparse streams in push-based seeking
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
 if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.

https://bugzilla.gnome.org/show_bug.cgi?id=742661
2015-02-14 11:36:11 -03:00
Tim-Philipp Müller
3f5b690e78 splitmuxsink: flag as sink from the start 2015-02-13 20:40:48 +00:00
Philippe Normand
3a9b0188cd qtdemux: Initial 'sidx' atom parsing support
Parse the 'sidx' atom and update the total duration according to the
parser result. The isoff parser code is imported from
gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
function was factored out of the gst_isoff_sidx_parser_add_buffer()
function.

https://bugzilla.gnome.org/show_bug.cgi?id=743578
2015-02-12 14:23:21 -03:00
Jan Schmidt
2e00311fe1 flvdemux: Use gst_video_guess_framerate()
Use gst_video_guess_framerate() from libgstvideo to guess
sensible common framerates where possible from the
floating point fps in the stream.
2015-02-12 23:38:47 +11:00
Sebastian Dröge
f4b5107796 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 13:53:02 +01:00
Sebastian Dröge
b79eff7f9b rtpsession: Handle first RTCP packet and early feedback correctly
According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
an early RTCP packet for the very first one. It must be a regular one.

Also make sure to not use last_rtcp_send_time in any calculations until
we actually sent an RTCP packet already. In specific this means that we
must not use it for forward reconsideration of the current RTCP send time.
Instead we don't do any forward reconsideration for the first RTCP packet.
2015-02-11 10:32:46 +01:00
Wim Taymans
009a62fddb rtph263depay: fix compilation with gcc 5.0 2015-02-10 18:54:24 +01:00
Tim-Philipp Müller
90badeebad splitmuxsink: fix example pipeline properly
x264enc might not have a max-key-int property, but it
has a key-int-max property...
2015-02-10 16:00:07 +00:00
Luis de Bethencourt
102ae8511a splitmux: fix typo 2015-02-10 14:57:55 +00:00
Luis de Bethencourt
12aa2428e0 splitmux: update example pipeline
Element x264enc doesn't have a max-key-int property
2015-02-10 14:56:23 +00:00
Luis de Bethencourt
0373fd8f65 splitmux: fix memory leak
If execution goes to the beach in line 981, buf_info goes out of scope without
the memory being free'd. Handle this case.

CID #1268403
2015-02-10 13:33:09 +00:00
Tim-Philipp Müller
603c1d71a1 rtspsrc: fix awkward if clause 2015-02-08 12:03:10 +00:00
Jan Schmidt
8ceb58122e splitmux: Add unit test for file splitting
Add a unit test for file splitting, and fix the leaks in the
splitmuxsink it found
2015-02-07 03:58:30 +11:00
Luis de Bethencourt
eb975ce880 wavparse: fix which stop variable is used in assignment
Assignment is done to variable segment.stop when the intention was to assign to
local variable stop. Instead of overwriting it, the value is now clamped and
segment.stop is set to it soon after.

CID #1265773
2015-02-06 14:46:14 +00:00
Jan Schmidt
aa4c29c5d6 splitmux: Fix memory leaks until the test valgrinds clean 2015-02-07 00:19:36 +11:00
Jan Schmidt
ace6be8abb splitmux: Handle early EOS during part preparation
Handle the case where a short file reaches EOS while we're still
waiting for no-more-pads, and make sure we continue to the internal
READY state for real playback to work properly later.
2015-02-06 06:42:17 +11:00
Jan Schmidt
5e2214d309 splitmux: Implement new elements for splitting files at mux level.
Implement 2 new elements - splitmuxsink and splitmuxsrc.

splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.

splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
2015-02-06 04:26:59 +11:00
Thiago Santos
a6d73797d0 rtph264depay: prevent trying to get 0 bytes from adapter
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to
a segfault

https://bugzilla.gnome.org/show_bug.cgi?id=737199
2015-02-04 21:37:50 -03:00
Jan Schmidt
a3059bec1f qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS
When the trickmode key-units flag is set on the segment, simply skip
any sample on a video stream that isn't a keyframe
2015-02-04 21:58:31 +11:00
Wim Taymans
852c040c89 rtspsrc: fix container handling
We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.

See https://bugzilla.gnome.org/show_bug.cgi?id=739391
2015-02-03 17:39:10 +01:00
Thiago Santos
7772a25fdc matroskamux: store and write stream tags
Separate global from stream tags storage and write them to the
appropriate tags entry in the output
2015-02-02 20:07:13 -03:00
Thiago Santos
75dee31b0d qtdemux: parse stream tags
Keep global and stream tags separately and parse the udta node
that can be found under the trak atom. The udta will contain
stream specific tags and will be pushed as such

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-02-02 14:05:51 -03:00
Thiago Santos
e52b2cb2cf qtmux: store stream and container tags separately
Tags received via events, when marked as stream tags, will
be stored on that stream's trak atom instead of being stored
in the main tags atom. This allows the resulting file to have
global and stream tags stored.

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:23:01 -03:00
Thiago Santos
6321cdedb3 qtmux: refactor tags functions to accomodata UDTA at trak level
Refactor the functions that were bound to the 'moov' atom to
directly pass the desired 'udta' that should receive the tags.
This allows the tags to be written to 'udta' at the 'moov' or
the 'trak' level, creating tags that are for the container or
for a stream only.

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:22:57 -03:00
Thiago Santos
f0fde8be88 qtmux: map application name to _swr tag
It refers to the application name and version used to create the
file

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:22:44 -03:00
Jan Schmidt
4a77c8a84f matroska: Fix seeking past the end of the file in reverse mode.
Snap to the end of the file when seeking past the end in reverse mode,
and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
for the stop position by always seeking on a segment in stream time
2015-01-31 06:15:44 +11:00
Sebastian Dröge
075eb10e65 rtpsession: Fix signal name
This wasn't meant to be pushed at all yet, but now that it's there
already it won't hurt to make it correct at least.
2015-01-30 18:22:31 +01:00
Sebastian Dröge
ec99bbb5e1 rtpstats: Fix typo in documentation 2015-01-30 16:56:35 +01:00
Sebastian Dröge
77511b156e rtpsession: Add new on-receiving-rtcp signal
This will be emitted whenever an RTCP packet is received. Different to
on-feedback-rtcp, this signal gets every complete RTCP packet and not
just the individual feedback packets.
2015-01-30 16:50:36 +01:00
Thiago Santos
9a9d4eccea qtdemux: simplify segment.base math
Remove a fix for heavily edited files added for fixing
https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work
with seeks and proper gaps playback. The fix was replaced
for a more general solution that bases on using previous
segment's duration, just like it works for media segments
playback.

https://bugzilla.gnome.org/show_bug.cgi?id=743518
2015-01-28 15:20:58 -03:00
Luis de Bethencourt
5ff1229754 videomixer: update orc files 2015-01-27 14:00:35 +00:00
Thiago Santos
2586a219f6 qtdemux: Fix data dropping for fragmented streams
For fragmented streams with extra data at the end of the mdat
qtdemux was not dropping those bytes and would try to use
that extra data as the beginning of a new atom, causing the
stream to fail.

https://bugzilla.gnome.org/show_bug.cgi?id=743407
2015-01-27 08:54:19 -03:00
Sebastian Dröge
e4ed852041 rtpsession: Deprecate rtcp-immediate-feedback-threshold property
It had no effect since quite some time and also is not needed in general,
especially not to switch between immediate feedback mode and early feedback
mode. The latest understanding of the RFC is that from the endpoint point of
view, both modes are exactly the same. RTCP is only allowed to use the
bandwidth as given by the RFC constraints, as such it is only ever possible
to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
packets.

The difference between immediate feedback mode and early feedback mode is that
the former guarantees that an RTCP packet can be sent for every event
"immediately", which means that the bandwidth calculations from the RFC have
resulted in an RTCP scheduling interval that is small enough. Early feedback
mode on the other hand means that we can schedule some packets early to make
that happen, but it's not guaranteed at all that it's possible to schedule
an RTCP packet per event (i.e. they need to be accumulated or dropped).
2015-01-26 18:49:31 +01:00
Sebastian Dröge
b07b7736b3 rtpsession: Delay the next regular RTCP packet after early RTCP
This is required to not exceed the short term average RTCP bitrate when
using early feedback as compared to without early feedback.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
bc9111a03d rtpsession: Add new send-rtcp-full signal
This indicates with a boolean return value if scheduling a new RTCP packet
within the requested delay was possible. Otherwise it behaves exactly like
send-rtcp. The only reason for adding a new signal is ABI compatibility.
2015-01-26 18:49:31 +01:00
Luis de Bethencourt
1e15808563 matroskademux: remove unnecessary check
No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
flow is OK or not, the check there will be a break from the switch. Removing the
check since the outcome is the same.

CID #1265762
2015-01-23 17:35:51 +00:00
Edward Hervey
932b32bb6e matroskamux: Avoid using freed variable
the name variable might have been attributed to pad_name, make sure we
free it only *after* pad_name has been used.

Coverity CID : 1265774
2015-01-23 15:16:25 +01:00
Edward Hervey
8abfd9d720 avimux: Avoid using freed variable
the name variable might have been attributed to pad_name, make sure we
free it only *after* pad_name has been used.

Coverity CID : 1265775
2015-01-23 15:15:07 +01:00
Sebastian Dröge
60e2d0c84f rtpsession: Fix indention 2015-01-22 11:03:25 +01:00
Edward Hervey
7203c4751c qtdemux_dump: Bypass even more code if debugging is disabled
And avoid using variables that won't exist when debugging is disabled
2015-01-21 17:36:26 +01:00
Edward Hervey
906f4c4360 qtdemux: Only traverse/dump nodes if guaranteed to be used
__gst_debug_min is the "global" lowest debug level set. There's no
guarantee the qtdemux debug category is actually set at that level.
2015-01-21 15:32:01 +01:00
Edward Hervey
9fa85f72e1 matroska: Avoid debugging below category threshold
This part alone was what made the matroska thread take a full core
on an android phone ...
2015-01-21 15:26:41 +01:00
Sebastian Dröge
d5aab81a77 Constify some static arrays everywhere 2015-01-21 09:55:53 +01:00
Vincent Penquerc'h
d854cfff9d qtdemux: fix deadlock seeking in files without seek entries
A mutex unlock was missing.

https://bugzilla.gnome.org/show_bug.cgi?id=739975
2015-01-19 17:49:54 +00:00
Vincent Penquerc'h
84c44fceac videomixer: fix illegal memory access in blend function with negative ypos
https://bugzilla.gnome.org/show_bug.cgi?id=741115
2015-01-19 12:34:25 +00:00
Sebastian Dröge
dc2251a664 qtmux: Add support for v210 2015-01-13 19:05:40 +01:00
Sebastian Dröge
b7134435ee qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420
Also add a few other raw video formats we support: v308, v216
and add comments for a few others we don't support yet.

https://developer.apple.com/library/mac/technotes/tn2162/
2015-01-13 19:05:40 +01:00
Thiago Santos
3e0be85840 qtdemux: fix stream time conversion
Use the right macro to convert to the correct scale or the
segment information will be wrong

https://bugzilla.gnome.org/show_bug.cgi?id=742572
2015-01-09 11:40:40 -03:00
Matej Knopp
ff5b235c32 ac3parse: request at least 8 bytes to properly parse header
https://bugzilla.gnome.org/show_bug.cgi?id=742325
2015-01-08 14:45:23 +01:00
Michael Smith
e8f3d596bc wavparse: skip an additional uninteresting chunk type before the fmt chunk. 2015-01-07 16:20:03 -08:00
Luis de Bethencourt
42535107ca audiodynamic: assert func_index is inside bounds
Bringing back the check removed in the previous commit but have that check be a
g_assert. Changing the function to static void since return can never be False,
because audio format will never be unkown.
2015-01-07 18:16:12 +00:00
Luis de Bethencourt
1db92a91de audiodynamic: remove always-true conditional
func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
The conditional checking if func_index is >= 0 and < 8 will always be true.
Removing it.

CID 1226442
2015-01-07 17:31:39 +00:00
Sebastian Dröge
87c8c163a8 rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
2015-01-07 18:05:18 +01:00
Aleix Conchillo Flaqué
07c5d1820a rtspsrc: set PLAYING state after configuring caps
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.

https://bugzilla.gnome.org/show_bug.cgi?id=740505
2014-12-31 12:49:11 +00:00
Sebastian Dröge
67d4b85d6a matroskademux: Improve detection of being stuck at the same offset
Only error out if we read from the same position again and got the
same length. Just the same position is not necessarily enough.
2014-12-29 15:35:19 +01:00
Sebastian Dröge
e596a3b6a7 matroskademux: Don't get stuck at the same offset when searching for clusters
This could happen if there is an invalid cluster with size 0, and in that
case just error out instead of looping forever.
2014-12-29 15:02:52 +01:00
Tim-Philipp Müller
aa94fc6beb qtmux: fix ALAC muxing
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.

Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink

https://bugzilla.gnome.org/show_bug.cgi?id=741783
2014-12-25 21:37:49 +00:00
Tim-Philipp Müller
c62209d050 rtpptdemux: just drop invalid rtp packets instead of erroring out
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).

https://bugzilla.gnome.org/show_bug.cgi?id=741398
2014-12-25 15:48:04 +00:00
Tim-Philipp Müller
bcad30510b rtpptdemux: fix 0.10-ism in docs 2014-12-25 15:44:15 +00:00
Edward Hervey
cbe56d2331 matroska-demux: Cache upstream length
Instead of constantly querying upstream, just cache the last duration,
and in the unlikelyness we might have gone over query again before
deciding we are EOS.

Cut 15% cpu off matroskademux streaming thread (srsly...)
2014-12-19 10:59:18 +01:00
Vincent Penquerc'h
b7413279d9 matroska: mux/demux the OpusHead header
This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
it is marked as a draft, this part was confirmed to be correct on
IRC), and allows one to determine whether a demuxed stream is
multistream or not, and thus set the multistream caps field
accordingly. In turn, this means downstream does not have to guess.

https://bugzilla.gnome.org/show_bug.cgi?id=740744
2014-12-18 11:38:49 +00:00
Sebastian Dröge
d18b893d28 rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
CID 1258717
2014-12-18 11:51:12 +01:00
Tim-Philipp Müller
4dd7d79b52 udpsink: allocate scratch space for render functions on the heap
and not the stack. Our allocations could get a bit too large
to be sure it's not going to cause trouble using the stack.
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
97a2eb7afb multiudpsink: re-use send_buffers() code path for render() function
It's like rendering a buffer list, just with one buffer.
Has the added advantage that if there are multiple clients
we can send the buffer to all the clients in one go.
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
54a9a436ba multiudpsink: keep client list consistent during removals
We unlock and re-lock the client lock while emitting the
removed signal, which causes inconsistencies in the client
list vs. the client counts. Instead, remove the client from
the list already before emitting the signal and put it into
a temporary list of clients to be removed. That way things
look consistent to the streaming thread, but signal callbacks
can still do things like get stats from removed clients.
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
fa3ef2e54c multiudpsink: fix client count after removal 2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
7bdf7500a1 multiudpsink: keep client list sorted by socket family
We make use of in the send_buffers() function if we
need to use different sockets to send to IPv4 and
IPv6 destinations.
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
e1a7deb27f multiudpsink: add sendmmsg-ready render_list function prototype
Add prototype for a render_list() function that can use a
sendmmsg-style g_socket_send_messages() function once it lands
in GLib. We can use this infrastructure to send multiple buffers
made up by multiple memories to multiple clients in one go, which
drastically reduces the number of syscalls made when sending
high-bitrate video streams.

https://bugzilla.gnome.org/show_bug.cgi?id=732152
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
dead5c2476 multiudpsink: make udp client structure refcounted
Use the refcount for memory management and keep track
of the number of duplicate clients in a separate
variable. This will be useful later, and means we
don't have to hold the OBJECT_LOCK all the time.

https://bugzilla.gnome.org/show_bug.cgi?id=732866
2014-12-16 20:26:36 +00:00
Tim-Philipp Müller
675384a8cb multiudpsink: keep count of number of unique and non-unique IPv4 and IPv6 clients
This will come in handy later.
2014-12-16 20:26:36 +00:00
Sebastian Dröge
6b2fc2de8d rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
... because the application already has a signal handler set up here.
2014-12-16 16:40:08 +01:00
Matthew Waters
bf0a19bf02 rtspsrc: add retransmission support according to RFC4588
Based on the client-rtpaux example
2014-12-16 16:40:08 +01:00
Nicolas Dufresne
9c468ef2da videocrop: Remove todo about caps filter
The filter is already interected.
2014-12-15 18:30:01 -05:00
Nicolas Dufresne
36f1a9bce1 videocrop: Make sure new crop is applied
Since "basetransform: Fix caps equality check" commit a7f357,
set_info() will not be called anymore if crop didn't change
the caps. This is fixed by setting "need_update" boolean when
cropping properties has been changed, and then applying these
if they where not applied before rendering the next frame. This
patch also fixed the locking, dropping un-needed custom lock,
and no holding needless lock while doing the operation as we
already hold the streaming lock.

https://bugzilla.gnome.org/show_bug.cgi?id=740787
2014-12-15 18:27:09 -05:00
Thibault Saunier
76944350c0 Deinterlace: in query_caps return only supported formats if filter is interlaced
In some cases the currently set GstVideoInfo is not interlaced, but
upstream caps are interlaced and the info is passed in the filter,
we should take that info into account and make sure that we do not
consider that case as a "pass through" case.

https://bugzilla.gnome.org/show_bug.cgi?id=741407
2014-12-14 12:41:16 +01:00
Edward Hervey
6b69ef24a1 qtdemux: Fix debug statement
It was using the non-increasing offset variable, which made that statement
not so useful :)
2014-12-12 11:06:17 +01:00
Edward Hervey
d1ae39d6d6 qtdemux: Add macros for the various timescale conversions
This helps make the code more readable and avoid future bad usage of
scaling function argument order.
2014-12-12 11:03:15 +01:00
Patrick Radizi
0a359cdbdc rtph264pay: fix potential crash when shutting down
A race condition in the state change function may cause buffers
to be unreffed while they are still used by the streaming thread
in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
up to the parent class first in the state change function to
make sure streaming has stopped and only then free those buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=741381
2014-12-11 14:00:19 +00:00
Jan Schmidt
de8d00348e qtdemux: Copy flags of the overall segment to output segments
Preserve the segment flags of the overall demux segment on the output
segments for each pad.
2014-12-12 00:56:49 +11:00
Matej Knopp
2505e343b1 qtmux: use 64bit chunk_offset
https://bugzilla.gnome.org/show_bug.cgi?id=741279
2014-12-10 18:42:30 -03:00
Edward Hervey
9a903c994f qtdemux: Fix rounding errors in duration update
Make sure we store updated segment stop/duration with the same
granularity as the duration timescale.

And add more debug
2014-12-10 17:39:17 +01:00
Edward Hervey
b40cfcfffb qtdemux: Update duration when we get more information
When dealing with fragmented files, we will get more accurate duration
information via the mfra and moof atoms.

In order for playback to not stop at the initial duration (from the
moov atom), we need to check and update the various duration variables
when we find more information.

Fixes playback of fragmented files in pull mode
2014-12-10 16:55:44 +01:00
Edward Hervey
799609583e qtdemux: Remove variable assignments never read
As detected by clang/scan-build
2014-12-10 15:09:25 +01:00
Edward Hervey
7828f73516 qtdemux: Use GstClockTime for nanosecond-based time variables/fields
Avoids confusion with timescaled-based variables and bytes (offset)
variables.
And use GST_CLOCK_TIME_NONE where applicable
2014-12-10 15:09:25 +01:00
Edward Hervey
0a381b9edd pushfilesrc: Add TIME SEGMENT capability
Adds a new set of properties to make pushfilesrc output a TIME SEGMENT
(instead of the filesrc BYTE SEGMENT).

When time-segment is set to True the following will happen:
* Seeks are refused (data starts from the beginning of the file)
* The BYTE segment will be replaced by a TIME segment with the values
  specified in the various properties
* The first outgoing buffer will have a timestamp set on it (by default
  it has a value of GST_CLOCK_TIME_NONE)
2014-12-10 15:09:25 +01:00
Sebastian Dröge
f5d26af3c9 aacparse: Also only unref caps if they're not NULL 2014-12-10 11:35:29 +01:00
Sebastian Dröge
6d6c6aac13 aacparse: gst_pad_get_allowed_caps() will return NULL if there is no peer 2014-12-10 11:35:02 +01:00
Thibault Saunier
52a1773b40 rtpsession: Use an empty iterator in iterate_internal_link when no links
And not a NULL Iterator, so it is consistent with the way it usually
works and avoid user to need a different code paths to handle that.
2014-12-09 20:38:22 +01:00
Patrick Radizi
fef1a8d88a rtph264pay: Fixes buffer leak when using SPS/PPS
Fixes a buffer leak that would occurr if the pipeline was shutdown
while a SPS/PPS header was being created.

https://bugzilla.gnome.org/show_bug.cgi?id=741271
2014-12-09 09:47:23 +01:00
Mathieu Duponchelle
a5694b213a agingtv: fix memcpy when no color aging requested.
video_size is the size in pixels, actual size of the memcpy
has to be stride * height.
2014-12-09 04:44:40 +01:00
Nicola Murino
c466ff4748 matroskademux: set framerate 0/1 when duration is not known
https://bugzilla.gnome.org/show_bug.cgi?id=740130
2014-12-04 18:20:37 +01:00
Jan Schmidt
f4ca3c255a qtdemux: More fixes for reverse playback
When seeking or finding the previous keyframe, do
comparisons against targets and segments using composition time
to correctly decide which sample times match.
2014-12-04 22:53:07 +11:00
Thibault Saunier
aa89278ade rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links
We used to setup an iterator with 1 GValue set with a NULL object
pointer which is not the normal way to do that. Instead we should make
sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
2014-12-03 11:17:11 +01:00
Jan Schmidt
b3d1ab5267 qtdemux: Handle seeks past EOS as a seek to the end
Fix reverse playback of every frame by making seeks past/to EOS
find the last segment and start there.
2014-12-03 13:23:35 +11:00
Olivier Crête
e3b0fb2a5d rtpmpadepay: Relax caps to allow any clock-rate
Some Wowza setups seem to send an invalid non-90000 clock-rate.
2014-12-02 15:33:25 -05:00
Thiago Santos
148da6210a qtdemux: don't use GST_CLOCK_TIME_NONE in non GstClockTime variables
Use -1 instead as those are gint64/guint64 variables and not GstClockTime
2014-12-02 00:46:35 -03:00
Tim-Philipp Müller
d65c3bbe7e qtdemux: implement seeking in fragmented mp4 files in pull mode based on the mfra table 2014-11-30 15:33:13 +00:00
Tim-Philipp Müller
77f37a6b22 qtdemux: use track fragment decoding time (tfdt) in parse_trun() for interpolation
As fallback if we don't have any existing samples
as reference point yet.

Based on patch by David Corvoysier <david.corvoysier@orange.com>
2014-11-30 15:33:13 +00:00
Tim-Philipp Müller
e24f903b13 qtdemux: parse mfra random access box for fragmented mp4 files
If it's present, and we operate in pull mode.
2014-11-30 15:33:13 +00:00
Tim-Philipp Müller
8a0f4e74e4 qtdemux: stop parsing headers for fragmented mp4s at the first moof
Currently during header parsing, we scan through the entire file
and skip every moof+mdat chunk for fragmented mp4s, which makes
start-up incredibly slow. Instead, just stop at the first moof
chunk when have a moov, and start exposing the streams, so we
can go and start handling the moofs for real.
2014-11-30 15:30:04 +00:00
Olivier Crête
ccac1f8c0b rtprtxreceive: Use offset when copying header
The header is not always at the start of the packet, so we need to compute
the offset first.
2014-11-29 18:38:12 -05:00
Andrei Sarakeev
6348de195d aspectratiocrop: Handle resolution changes properly
When an caps-event is received, we must immediately change the crop
to videocrop correctly changed caps-event dimension, otherwise the
videocrop will first use the previous value of the crop that when
resizing video to a smaller resolution may cause an error.

https://bugzilla.gnome.org/show_bug.cgi?id=740671
2014-11-28 11:19:23 +01:00
Edward Hervey
5b5e9f320f isomp4: Check presence of mfhd in moof
The 'mfhd' atom is mandatory in 'moof'. We can later on check whether
the fragment number properly increases
2014-11-26 16:36:39 +01:00
Edward Hervey
5e3e97353d isomp4: Fix mfro and tfra atom dumping
mfro was skipping the version/flags
tfra had wrong byte_reader return value checks
2014-11-26 16:36:39 +01:00
Edward Hervey
c45533bcd7 isomp4: Add mfhd atom dumping 2014-11-26 16:36:39 +01:00
Jan Schmidt
61bbd2d226 qtdemux: Handle empty segments when seeking in reverse play.
Empty segments in an edit list have a media_start time of -1,
as they don't actually play any media. Allow for that when
aligning to the reference stream in reverse play.
2014-11-27 00:17:03 +11:00
Tim-Philipp Müller
69ec922c16 icydemux: does not need to link against zlib 2014-11-23 16:24:06 +00:00
Miguel París Díaz
6daa57868f rtpjitterbuffer: ensure rtx_retry_period >= 0
https://bugzilla.gnome.org/show_bug.cgi?id=739344
2014-11-22 14:48:57 +00:00
Arun Raghavan
45e716e75d rtpbin: Fix up new_jitterbuffer signal prototype 2014-11-20 22:42:59 +05:30
Arun Raghavan
56436ccced rtpbin: Document how to control per-SSRC retransmission 2014-11-20 20:24:42 +05:30
Wim Taymans
3d7b0f30d7 rtpgstpay: put 0-byte at the end of events
Put a 0-byte at the end of the event string. Does not break ABI because
old depayloaders will skip the 0 byte (which is included in the length).
Expect a 0-byte at the end of the event string or a ; for old
payloaders.

See https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 13:14:14 +01:00
Wim Taymans
9d2902d978 rtpgstdepay: avoid buffer overread.
Check that a caps event string is 0 terminated and the event string is
terminated with a ; to avoid buffer overreads.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 12:44:26 +01:00
Tim-Philipp Müller
488d0b93cd qtmux: don't limit max video resolution to 4096x4096
MAX isn't entirely correct as upper limit either,
it should really be MAXUINT32, but it's unlikely
to be a problem in the near future.

https://bugzilla.gnome.org/show_bug.cgi?id=740407
2014-11-20 10:45:53 +00:00
Aleix Conchillo Flaqué
00ca83629b rtspsrc: fix leak for mikey base64 decoded key-mgmt
https://bugzilla.gnome.org/show_bug.cgi?id=740392
2014-11-20 09:15:56 +01:00
Wim Taymans
e95da8410f videobalance: fix unhandled format in passthrough
In passthrough we can handle all formats.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387
2014-11-20 09:02:36 +01:00
Jan Alexander Steffens (heftig)
bf73d834b2 flvdemux: Restrict resyncing to TS regressions
The behavior of resyncing video and audio indepen-
dently can cause A/V desyncs. Lets restrict resyncs
to jumps backward for now.

https://bugzilla.gnome.org/show_bug.cgi?id=736397
2014-11-19 11:58:19 -05:00
Matthew Waters
0053ad0847 videomixer: fix up QoS handling for live sources
Only attempt adaptive drop when we are not live

https://bugzilla.gnome.org/show_bug.cgi?id=739996
2014-11-17 23:16:03 +11:00
Arun Raghavan
1c3b233fef rtpmanager: Trivial typo fix 2014-11-10 13:16:50 +05:30
Sebastian Dröge
7a909917b5 matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it 2014-11-09 11:04:33 +01:00
Göran Jönsson
ec05d3b6d8 matroskamux: make GstMatroskamuxPad get_type() function thread-safe
https://bugzilla.gnome.org/show_bug.cgi?id=739722
2014-11-07 21:20:31 +00:00
Josep Torra
038cc7b004 rtsp: fix build in gst-uninstalled setup 2014-11-06 21:38:43 +01:00
Thibault Saunier
99bbc2bbe4 imagefreeze: Handle seqnums
https://bugzilla.gnome.org/show_bug.cgi?id=739366
2014-11-06 12:20:25 +01:00
Wim Taymans
26d682d23f videomixer2: reverse order of params for converter 2014-11-03 15:26:06 +01:00
Tim-Philipp Müller
c756fd6a55 goom2k1: post QoS messages when dropping frames due to QoS 2014-11-02 19:42:03 +00:00
Tim-Philipp Müller
b03056eede goom: post QoS messages when dropping frames due to QoS 2014-11-02 19:31:01 +00:00
Tim-Philipp Müller
85c3c36712 matroskamux: tweak writing app tag string a little 2014-11-02 19:02:35 +00:00
Tim-Philipp Müller
3956f5addc Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED 2014-11-02 16:58:30 +00:00
Tim-Philipp Müller
d940c21b78 rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
Properties are so much more useful if you can actually set
and get their values.
2014-11-02 13:06:33 +00:00
Nicolas Dufresne
0f4f948f5f rtpvp8: Use VP8 encoding name
Both Firefox and Chrome uses VP8 as the encoding in their SDP.
Adding this now defacto standard name removes the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2014-11-01 11:26:26 -04:00
Tim-Philipp Müller
92c1d289b8 rtpmp2tpay: fix up template caps so we can output the default pt 33
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.

There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
2014-11-01 12:40:07 +00:00
Aleix Conchillo Flaqué
d15ebcbf62 rtspsrc: mikey related memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739430
2014-10-31 10:03:47 +00:00
Sebastian Dröge
4aac09e708 aacparse: Always set profile/level on the caps
We have the information already, so why not use it?
2014-10-26 11:47:25 +01:00
Tim-Philipp Müller
b02d73a0ed rtpjitterbuffer: fix crash on some 32-bit systems
Make sure to pass right number of bits to gst_structure_new()
which is a vararg function.

Fixes elements/rtpaux unit test on ppc32.
2014-10-25 12:45:31 +01:00
Tim-Philipp Müller
401782c19d interleave: intersect result with filter caps in caps query
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
2014-10-25 11:08:48 +01:00
Wim Taymans
bd09dc96e9 rtpjitterbuffer: limit the retry frequency
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
2014-10-22 15:04:24 +02:00
Miguel París Díaz
4b5243c43d gstrtpjitterbuffer: add "rtx-min-delay" property
This property is useful to set a min time to wait before sending a
retransmission event.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 15:00:27 +02:00
Wim Taymans
0b81b316b5 jitterbuffer: Refactor code
Refactor some code dealing with calculating various timeouts.

See https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 14:59:57 +02:00
Miguel París Díaz
e6504e3a65 rtpsession: fix Early Feedback Transmission
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.

After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.

Remove the condition for the immediate feedback for now.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
2014-10-22 13:13:47 +02:00
Wim Taymans
09f179139d rtpjitterbuffer: make debug line less confusing 2014-10-21 13:10:53 +02:00
Wim Taymans
2e7f5c08cf jitterbuffer: rework resync handling
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
2014-10-21 11:57:34 +02:00
Aleix Conchillo Flaqué
bd392d72ee rtspsrc: set full stream caps on internal src TCP pads
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.

https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-21 11:33:01 +02:00
Sjoerd Simons
0ee384b251 rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.

https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-21 10:52:00 +02:00
Vineeth T M
1131db8c1f videobox: critical error when element properties set as max/min
left, right, top, bottom can be set from range of -2147483648 to 2147483647
when i launch the videobox element with that values, it gives a critical error

(gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.

https://bugzilla.gnome.org/show_bug.cgi?id=738838
2014-10-20 12:53:51 +02:00
Tim-Philipp Müller
f3fec86bc9 Revert "rtp: add h265 RTP payloader + depayloader"
This reverts commit d06ba9051f.

This breaks the build, as it depends on parser API in -bad.
2014-10-15 17:48:46 +01:00
Jurgen Slowack
d06ba9051f rtp: add h265 RTP payloader + depayloader 2014-10-15 17:34:50 +02:00
Peter G. Baum
b5e46c05d7 wavenc: Support RF64 format
https://bugzilla.gnome.org/show_bug.cgi?id=725145
2014-10-14 10:24:50 +02:00
David Sansome
8154c90c9b equalizer: Don't call iirequalizer's transform_ip in passthrough mode
It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.

https://bugzilla.gnome.org/show_bug.cgi?id=737886
2014-10-13 08:30:03 +02:00
Olivier Crête
51a8bedced rtpsource: Rename seqnum-base to seqnum-offset in caps
This was modified back in 1.0 in GstRtpBasePayload
2014-10-10 18:33:34 -04:00
Olivier Crête
155ed569c3 rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:32 -04:00
Olivier Crête
b3069634bd rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:23 -04:00
Luis de Bethencourt
cff880401d goom2k1: removing block of code that does nothing
The loop in zoomFilterSetResolution is meant to change the values in the
zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
but no conditions that change the value of decc are ever met and the array is
filled with zero for each element. Which is the initial state of the
array before the loop begins.

The loop does nothing.

https://bugzilla.gnome.org/show_bug.cgi?id=728353
2014-10-08 14:07:56 +01:00
Stefan Sauer
98222a67ff rtpjitterbuffer: don't log all clock_rate changes as warnings.
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
2014-10-04 17:17:13 +02:00
Matej Knopp
e1d275cfec aacparse: fix memory leak when prepending ADTS headers
https://bugzilla.gnome.org/show_bug.cgi?id=737761
2014-10-02 10:41:28 +03:00
Antonio Ospite
7ae7f657fa interleave: interleave samples following the Default Channel Ordering
In order to have a full mapping between channel positions in the audio
stream and loudspeaker positions, the channel-mask alone is not enough:
the channels must be interleaved following some Default Channel Ordering
as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.

As a Default Channel Ordering use the one implied by
GstAudioChannelPosition which follows the ordering defined in SMPTE
2036-2-2008[2].

NOTE that the relative order in the Top Layer is not exactly the same as
the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
using so may channels are already aware of such discrepancies.

[1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
[2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
2014-10-02 10:21:26 +03:00
Sebastian Dröge
7729f4ce81 wavenc: Send CAPS event after the pad was activated
Otherwise the CAPS event will be dropped and we never configure any caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.

https://bugzilla.gnome.org/show_bug.cgi?id=737735
2014-10-02 10:10:11 +03:00
Sebastian Dröge
1a2adf5123 videomixer: Actually use the correct GstVideoInfo for conversion 2014-10-01 17:29:29 +03:00
Sebastian Dröge
c1a96113db videomixer: Revert the last commit and handle resolutions differences properly
This is about converting the format, not about converting any widths and
heights. Subclasses are expected to handler different resolutions themselves,
like the videomixers already do properly.
2014-10-01 17:24:59 +03:00
Sebastian Dröge
af7916ca4a videomixer: GstVideoConverter currently can't rescale and will assert
Leads to ugly assertions instead of properly erroring out:
CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed
2014-10-01 17:12:59 +03:00
Antonio Ospite
eca3e2474d wavenc: print channel masks in hexadecimal 2014-09-29 17:45:59 +03:00
Sebastian Dröge
d1c7f2e4d1 rtspsrc: Fix compiler warnings
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_new (&sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_parse_uri (uri, sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2014-09-26 13:46:16 +03:00
Jonas Holmberg
1371fa0c61 matroskademux: make demuxer reusable
Remove pads from flow combiner and reset last
flow return to FLOW_OK by resetting the flow combiner.
This prevents FLOW_FLUSHING when trying to re-use the
demuxer after setting it back to NULL/READY state.

https://bugzilla.gnome.org/show_bug.cgi?id=737359
2014-09-25 16:14:18 +01:00
Wim Taymans
84ec78bd86 videomixer: use video library code instead of copy 2014-09-24 16:46:36 +02:00
Sanjay NM
323683db96 audioparsers: Added index check before using the index
https://bugzilla.gnome.org/show_bug.cgi?id=736878
2014-09-24 10:21:35 +03:00
Matej Knopp
9f85dfd733 qtmux: Do not infer DTS on buffers from sparse streams.
DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 22:25:47 -03:00
Sanjay NM
36140ccf69 goom: Clarified precedence between % and ?
https://bugzilla.gnome.org/show_bug.cgi?id=736887
2014-09-24 00:48:09 +01:00
Sanjay NM
f62076e49c rtsp: clarify expression so operator precedence is clear
https://bugzilla.gnome.org/show_bug.cgi?id=736903
2014-09-24 00:48:09 +01:00
Sanjay NM
26a1344f37 Miscellaneous minor cleanups
Fix redundant variables and assignments,
and unreachable breaks.

https://bugzilla.gnome.org/show_bug.cgi?id=736875
https://bugzilla.gnome.org/show_bug.cgi?id=736876
https://bugzilla.gnome.org/show_bug.cgi?id=736879
https://bugzilla.gnome.org/show_bug.cgi?id=736880
https://bugzilla.gnome.org/show_bug.cgi?id=736881
https://bugzilla.gnome.org/show_bug.cgi?id=736888
https://bugzilla.gnome.org/show_bug.cgi?id=736890
https://bugzilla.gnome.org/show_bug.cgi?id=736892
https://bugzilla.gnome.org/show_bug.cgi?id=736893
https://bugzilla.gnome.org/show_bug.cgi?id=736894
2014-09-24 00:45:31 +01:00
Tim-Philipp Müller
208e12dca2 videobox: remove duplicate assignments
https://bugzilla.gnome.org/show_bug.cgi?id=736897
2014-09-24 00:12:14 +01:00
Sebastian Dröge
91a3d044f0 flacparse: Only calculate with durations != -1 2014-09-23 22:56:21 +03:00
Matej Knopp
fd3e8c5672 qtmux: collect pad for sparse stream should be created with lock set to false
Avoids waiting for buffers from sparse streams

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:25:45 -03:00
Matej Knopp
6695341583 qtmux: fix subtitle buffer duration and strip null termination
Strip the \0 off the subtitle as we already know the size and also remember
to set the duration as buffer copying doesn't do it.

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:25:28 -03:00
Matej Knopp
f57e9c4516 qtmux: move subtitle layer above video and set alternate group
layer -1 is above video, that is 0
And having all subtitles in alternate group 2 means that only one
should be selected at a time.

https://bugzilla.gnome.org/show_bug.cgi?id=737095
2014-09-23 15:20:37 -03:00
Matej Knopp
8a4931726d qtdemux: Handle mp4a without ESDS atom
https://bugzilla.gnome.org/show_bug.cgi?id=736986
2014-09-22 13:04:52 -03:00
Sanjay NM
89eb378598 dtmf: Removed unused structure members
https://bugzilla.gnome.org/show_bug.cgi?id=736883
2014-09-19 15:42:04 -04:00
Reynaldo H. Verdejo Pinochet
e655d47dfc isomp4: fix wrong DAR calculation for PAR <= 1
CID #1226452

https://bugzilla.gnome.org/show_bug.cgi?id=736396
2014-09-18 18:53:38 -03:00
Sanjay NM
ba4b9b22d0 flv: Removed unreachable break statements
https://bugzilla.gnome.org/show_bug.cgi?id=736884
2014-09-18 09:42:43 -04:00
Ognyan Tonchev
f7ae4288a2 rtpbin: do not leak encsink pad in error case
https://bugzilla.gnome.org/show_bug.cgi?id=736807
2014-09-18 12:49:53 +03:00
Ognyan Tonchev
3bf81ad12c multipartdemux: do not leak new stream event
https://bugzilla.gnome.org/show_bug.cgi?id=736805
2014-09-18 12:49:53 +03:00
Ravi Kiran K N
5480f6d2dd y4menc: port y4menc to use GstVideoEncoder base class
https://bugzilla.gnome.org/show_bug.cgi?id=735085
2014-09-17 18:28:00 -03:00
Ognyan Tonchev
7cd335e9b9 flacparse: do not leak uid after parsing TOC event
https://bugzilla.gnome.org/show_bug.cgi?id=736739
2014-09-17 09:51:15 +03:00
Sebastian Dröge
4bc10e755a rtpvrawdepay: Declare some more required caps fields in the sink template caps
Now only missing are width and height, which are expressed as strings
for RTP... so we can't put them into the template caps.
2014-09-16 22:47:13 +03:00
Wim Taymans
711e1407a1 capssetter: update to 1.0 transform_caps sematics
In 1.0, we pass the complete caps to transform_caps to allow for better
optimizations. Make this function actually work on non-simple caps
instead of just ignoring the configured filter caps.
2014-09-15 18:14:06 +02:00
Peter G. Baum
f8f61237f8 wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels
https://bugzilla.gnome.org/show_bug.cgi?id=733444
2014-09-15 11:19:23 +03:00
Sebastian Dröge
a9d7c1d95e wavparse: Fix parsing of adtl chunks
We have to skip 12 bytes of data for the chunk, and the data size
passed to the sub-chunk parsing functions should have 4 bytes less
than the data size.

Also when parsing the sub-chunks, check if we actually have enough
data to read instead of just crashing.

https://bugzilla.gnome.org/show_bug.cgi?id=736266
2014-09-12 15:08:23 +03:00
Sanjay NM
66810a32f6 udp: include string.h for memcmp and memset
https://bugzilla.gnome.org//show_bug.cgi?id=736528
2014-09-12 10:45:39 +01:00
Anuj Jaiswal
4242495ea7 matroskamux: don't bitwise OR the same flag twice
https://bugzilla.gnome.org//show_bug.cgi?id=736543
2014-09-12 10:37:31 +01:00
Tim-Philipp Müller
4c08f2694d matroskademux: handle real audio 28_8
Fixes duplicate check for 14_4.

https://bugzilla.gnome.org//show_bug.cgi?id=736543
2014-09-12 10:35:36 +01:00
Anuj Jaiswal
86579c59bf multifilesink: don't OR the same flag twice
https://bugzilla.gnome.org/show_bug.cgi?id=736462
2014-09-11 11:05:35 +01:00
Tim-Philipp Müller
e6f77948ac udpsrc: more efficient memory handling
Drop use of g_socket_get_available_bytes() which is
not useful on all systems (where it returns the size
of the entire buffer not that of the next pending
packet), and is yet another syscall and apparently
very inefficient on Windows in the UDP case.

Instead, when reading UDP packets, use the more featureful
g_socket_receive_message() call that allows to read into
scattered memory, and allocate one memory chunk which is
likely to be large enough for a packet, while also providing
a larger allocated memory chunk just in case the packet
is larger than expected. If the received data fits into the
first chunk, we'll just add that to the buffer we return
and re-use the fallback buffer for next time, otherwise we
add both chunks to the buffer.

This reduces memory waste more reliably on systems where
get_available_bytes() doesn't work properly.

In a multimedia streaming scenario, incoming UDP packets
are almost never fragmented and thus almost always smaller
than the MTU size, which is also why we don't try to do
something smarter with more fallback memory chunks of
different sizes. The fallback scenario is just for when
someone built a broken sender pipeline (not using a
payloader or somesuch)

https://bugzilla.gnome.org/show_bug.cgi?id=610364
2014-09-09 17:38:52 +01:00
Tim-Philipp Müller
39505584e1 udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into
First chunk is the likely/expected buffer size, second is as
fallback in case the packet is larger in the end.

Next step: actually use these.
2014-09-09 17:35:38 +01:00
Tim-Philipp Müller
305e4c2f46 udpsrc: track max packet size and save allocator negotiated by GstBaseSrc 2014-09-09 17:35:14 +01:00
Tim-Philipp Müller
8e28994207 audioecho: fix example command line 2014-09-08 16:15:32 +01:00
Tim-Philipp Müller
7271ff253b avidemux: fix crash with certain videos
This is a regression from 1.2 caused by the port
to the pad flow combiner.

https://bugzilla.gnome.org/show_bug.cgi?id=736192
2014-09-07 12:48:16 +01:00
Sebastian Dröge
a3a5530518 matroska-demux: Don't handle parse errors at the end of file as an error
But only if they happen after the Matroska segment.

https://bugzilla.gnome.org/show_bug.cgi?id=735833
2014-09-05 11:36:30 +03:00
Andrei Sarakeev
558f9a2a6f videomixer: Fix synchronization if dynamically changing the FPS
https://bugzilla.gnome.org/show_bug.cgi?id=735859
2014-09-04 11:34:26 +03:00
Ravi Kiran K N
ea43ef214a smpte: Check if input caps are the same and create output caps from video info
This makes sure that also properties like the pixel-aspect-ratio are the same
between both streams and that the output caps contain all fields necessary for
complete video caps.

https://bugzilla.gnome.org/show_bug.cgi?id=735804
2014-09-04 10:47:34 +03:00
Vineeth T M
6ff397eccc imagefreeze: replace with gst_buffer_copy
gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.

replacing the same with gst_buffer_copy as the functionality is same.

https://bugzilla.gnome.org/show_bug.cgi?id=735880
2014-09-03 21:33:09 -03:00
Tim-Philipp Müller
884f81ba28 qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too
https://bugzilla.gnome.org/show_bug.cgi?id=735971
2014-09-03 23:08:16 +01:00
Jan Schmidt
9375e90203 qtdemux: Silence some warnings for normal file contents 2014-09-03 23:47:49 +10:00
Nicolas Huet
15894c1853 aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame
https://bugzilla.gnome.org/show_bug.cgi?id=735520
2014-09-02 09:43:14 +03:00
Vineeth T M
3a1e010221 imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced
Adding an extra condition while calling gst_caps_unref (templ)
and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
gst_caps_copy (caps) in line 177, since the functionality is same.

https://bugzilla.gnome.org/show_bug.cgi?id=735795
2014-09-01 14:34:43 +03:00
Sebastian Dröge
f5df8af59e wavparse: Store size of data tag in a 64 bit integer locally too
Otherwise we will clip the DS64 value of RF64 files to 32 bits again.
2014-08-29 11:55:26 +03:00
Sebastian Dröge
d924f8a955 wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer 2014-08-29 11:53:23 +03:00
Peter G. Baum
5c838af300 wavparse: support rf64 format
https://bugzilla.gnome.org/show_bug.cgi?id=735627
2014-08-29 11:49:42 +03:00
Jason Litzinger
bcbdcbf638 multipartdemux: Ensure caps before pad added.
This stores the stream-start, sets caps, and then adds the pad,
which ensures that the caps are set for the "pad-added" callback.

https://bugzilla.gnome.org/show_bug.cgi?id=735626
2014-08-29 11:38:19 +03:00
Nicolas Dufresne
356defdfea flvmux: Fallback to PTS if DTS is missing
Fixing a regression introduce when fixing:
https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-28 15:05:56 -04:00
Vineeth T M
d46631c5c7 imagefreeze: Remove impossible error condition
We return EOS after the first buffer, and GstPad will make sure now that we
won't get any other buffer afterwards until a flush happens. No need to check
for it ourselves.

https://bugzilla.gnome.org/show_bug.cgi?id=735581
2014-08-28 14:55:00 +03:00
Nicolas Dufresne
a7a3cb343a flvmux: Correctly offset timestamp
The previous method would break AV sync in the case audio or video
didn't start at the same point in running time.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-27 21:09:57 -04:00
Nicolas Dufresne
aa5bd99127 flvmux: Save dts from buffer
We no longer set dts in muxed buffer. This would lead to encoding tags
with timestamp 0 instead of the timestamp of previous buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-27 21:08:21 -04:00
Nicolas Dufresne
c1e7bec616 flvmux: Ensure Timestamp starts at 0
FLV documentation stipulates that timestamp must start at zero.
In order to respect this rule, keep the first timestamp around
and offset the timestamp from this value. This allow for longer
recording time in presence of timestamp that does not start
at 0 already.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-26 16:46:03 -04:00
Nicolas Dufresne
ff2bce7b26 flv: Tag timestamp are DTS not PTS
The tags in FLV are DTS. In audio cases, and for many video format this makes
no difference, but for AVC with B-Frames, PTS need to be computed from
composition timestamp CTS, with PTS = DTS + CTS.

https://bugzilla.gnome.org/show_bug.cgi?id=731352
2014-08-26 16:45:59 -04:00
Youness Alaoui
a98341397d jitterbuffer: Allow rtp caps without clock-rate
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.

https://bugzilla.gnome.org/show_bug.cgi?id=734322
2014-08-21 18:32:58 -04:00
Thiago Santos
fa103ca5ad qtdemux: avoid crashing on dash streams
DASH/fragmented moov might have no samples as those are carried
in moof fragments. Avoid crashing or failing the stream because
of that.
2014-08-18 14:05:52 -03:00
Víctor Manuel Jáquez Leal
419332e287 udp: fix udpsrc documentation
udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
been removed. This patch replaces those references to socket and close-socket
respectively.

https://bugzilla.gnome.org/show_bug.cgi?id=734987
2014-08-18 11:01:31 +01:00
Jan Schmidt
6e7930a10c qtmux: Make the default timescale 1/1800 second
The old default timescale of 1 millisecond produces irrational
numbers for a lot of framerate/audio-packet-duration multiples.
1/1800 is a nicer number, as it tends to produce better fractions
and therefore slightly higher accuracy overall
2014-08-15 13:03:52 +10:00
Jan Schmidt
f1c3a40547 matroska: Use gst_video_guess_framerate() function
Remove local framerate guessing function in favour of
the new gst_video_guess_framerate() function.
2014-08-15 01:17:27 +10:00
Jan Schmidt
ca068865c3 qtdemux: Improve framerate calculation/guessing
Change the way the output framerate is calculated
to ignore the first sample (which is sometimes truncated
in my testing) and use the new gst_video_guess_framerate()
function to recognise common standard framerates better.

Remove the code that was sorting the first 20 sample
durations and then ignoring the result.
2014-08-15 01:12:20 +10:00
Sebastian Dröge
ce1d4d9f21 videomixer: Use the best width/height/etc if downstream can handle that
Before it was always using whatever downstream preferred, while
the code and documentation claimed something different.

https://bugzilla.gnome.org/show_bug.cgi?id=727180
2014-08-14 16:36:44 +03:00
Ravi Kiran K N
61fe02a018 videomixer: Avoid double free of VideoConvert
https://bugzilla.gnome.org/show_bug.cgi?id=734764
2014-08-14 15:31:48 +03:00
Tim-Philipp Müller
6ee2665b7c flvdemux: fix indentation 2014-08-13 11:59:39 +01:00
Tim-Philipp Müller
9afeb9652b flvdemux: un-break duration querying
Commit 2b9493b5 broke this in two ways: a) we should only
pass duration queries in TIME format upstream (or at least
not those in DEFAULT or BYTE format), and b) we mustn't
overwrite the default value of 'res' from TRUE to FALSE
and not set it again later. This led to bogus durations
being reported for FLV playback from file, because TIME
queries would fail (as 'res' had been set to FALSE) and
parsers then do a BYTE query as fallback and try to
guesstimate something in return, which of course goes
horribly wrong since the BYTE size returned is for the
muxed file.
2014-08-13 11:59:39 +01:00
Sebastian Dröge
0911307d7d videobalance: Allow any raw caps in passthrough mode, not just the ones we handle 2014-08-13 13:25:36 +03:00
Sebastian Dröge
a9eda81978 videobalance: Allow ANY capsfeatures, but only in passthrough mode
When changing the properties to not be in passthrough mode anymore,
we will only accept caps we can process ourselves, potentially causing
a not-negotiated error.

https://bugzilla.gnome.org/show_bug.cgi?id=720345
2014-08-13 13:24:38 +03:00
George Kiagiadakis
9dd48c503c qtdemux: forward DISCONT from upstream to the output streams
This makes sense in DASH reverse playback, where the upstream dashdemux
will download DASH segments in reverse order, but push their buffers
forward to qtdemux and mark each segment start as DISCONT. This needs
to be forwarded downstream to the parser/decoder, otherwise it won't work.

https://bugzilla.gnome.org/show_bug.cgi?id=734443
2014-08-11 10:28:14 +02:00
Sebastian Rasmussen
70a43758bb shapewipe: Unref caps and element after usage
https://bugzilla.gnome.org/show_bug.cgi?id=734478
2014-08-10 11:09:09 +01:00
Tim-Philipp Müller
e8321af983 qtdemux: improve debug logging of fourccs
If we can't show ASCII, at least show them
in big endian order.
2014-08-09 20:50:01 +01:00
Tim-Philipp Müller
f41d03cd4d qtdemux: add support for 'wma ' mapping as found in some ismv files
e.g. To_The_Limit_720_2962.ismv
2014-08-09 20:49:53 +01:00
Tim-Philipp Müller
6183f83190 qtdemux: add support for 'vc-1' mapping as found in some ismv files
e.g. To_The_Limit_720_2962.ismv
2014-08-09 20:49:49 +01:00
Sebastian Rasmussen
276269d956 rtph263ppay: Unref pad template caps after use
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435
2014-08-08 16:02:24 -03:00
Sebastian Rasmussen
1fa61632fe videomixer: Unref allowed caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474
2014-08-08 15:59:36 -03:00
Sebastian Rasmussen
c85ae43a6e imagefreeze: Unref pad template caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475
2014-08-08 15:54:39 -03:00
Sebastian Rasmussen
edf8728016 navseek: Unref peer pad after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476
2014-08-08 15:50:55 -03:00
Sebastian Rasmussen
1a35bf9647 rtpmux: Unref pad template caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473
2014-08-08 15:38:32 -03:00
Srimanta Panda
421b00cd17 rtph264pay: append packetization mode parameter to SDP
Append packetization-mode parameter to SDP description.
Packetization mode signals the properties of an RTP payload type.

https://bugzilla.gnome.org/show_bug.cgi?id=733556
2014-08-08 13:41:36 +01:00
Jan Schmidt
d9e1aa4959 isomp4/qtmux: Write correct file duration when gaps exist.
When writing out a trak with an edit list, make sure the
overall file duration is also updated to reflect the
lengthening of the stream.

Add some more debug to qtdemux to warn about streams that
are longer than the file and get truncated.
2014-08-08 04:01:19 +10:00
Sebastian Dröge
add40de469 rtspsrc: Push the correct segment in TCP mode when seeking 2014-08-05 16:28:04 +02:00
Mark Nauwelaerts
d5d28055c1 rtph264pay: unbreak au aligned byte-stream payloading 2014-08-03 14:42:45 +02:00
Srimanta Panda
dd9f716892 rtph264pay: append profile-level-id to SDP
Append profile-level-id to SDP if available.

https://bugzilla.gnome.org/show_bug.cgi?id=733539
2014-08-01 16:01:07 +01:00
Philippe Normand
b8b5704445 interleave: set output caps layout to interleaved
Set output caps layout independently from input caps layout which can
be either non-interleaved or interleaved.

https://bugzilla.gnome.org/show_bug.cgi?id=733866
2014-07-29 11:49:32 +02:00
Tim-Philipp Müller
5122410f11 qtdemux: fix language code parsing for 3-letter codes starting with 'a'
And handle special value for 'unspecified' explicitly.

https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html
2014-07-21 18:21:50 +01:00
Sebastian Dröge
b1f7681555 videobox: Don't overwrite the first component with the alpha value for BGRx
Instead leave the x component unset when filling the borders.

https://bugzilla.gnome.org/show_bug.cgi?id=733380
2014-07-19 11:31:45 +02:00
Sebastian Dröge
638a700463 aacparse: Properly report in the CAPS query that we can convert ADTS<->RAW
https://bugzilla.gnome.org/show_bug.cgi?id=733190
2014-07-16 17:27:57 +02:00
Sebastian Rasmussen
f45657f604 rgvolume: Avoid taking unnecessary refs
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
2014-07-16 16:45:43 +02:00
Sebastian Rasmussen
ca22ad8da9 rtpdtmfmux: Avoid taking an unnecessary ref
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
2014-07-16 16:45:31 +02:00
Tim-Philipp Müller
c2614e5253 rtspsrc: fix query leak
https://bugzilla.gnome.org/show_bug.cgi?id=733003
2014-07-10 17:19:42 +01:00
Sebastian Dröge
dd5144fd4e wavenc: Return not-negotiated if we got no caps or caps negotiation failed
And do it always, not inside a g_return_val_if_fail().

See https://bugzilla.gnome.org/show_bug.cgi?id=732939
2014-07-10 14:37:31 +02:00
Tim-Philipp Müller
deeef84d2c videomixer: fix double unlock in segment seek segment code path
We only want to unlock if we push an event downstream and
jump to done_unlock label afterwards. We would also unlock
in case of a segment seek and then unlock again later, and
nothing good can come of that.

(This code looks a bit dodgy anyway though, shouldn't it
also bail out with FLOW_EOS here in case of a segment seek
scenario, just without the event?)
2014-07-04 20:26:46 +01:00
Sebastian Rasmussen
d33d8cf026 avidemux, wavparse: Print invalid fourcc in hex
Previously this was printed as characters which caused later processing
of the error message to sometimes warn about non-UTF-8 characters.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732714
2014-07-04 09:21:07 +01:00
Wim Taymans
db1d9444d6 rtspsrc: fix for mikey api change 2014-07-02 16:01:47 +02:00
Vincent Penquerc'h
bbb1a8de1f videomixer: reset QoS on segment event
https://bugzilla.gnome.org/show_bug.cgi?id=732540
2014-07-01 16:35:05 +01:00
Vincent Penquerc'h
5653b1a25a matroskademux: send gap events instead of segment tricks
This fixes missing frames from being time skipped.

https://bugzilla.gnome.org/show_bug.cgi?id=732372
2014-07-01 15:14:34 +01:00
Sebastian Dröge
2f47105129 rtpbin: Don't leak caps 2014-06-29 23:55:19 +02:00
Sebastian Dröge
bbca040336 rtpssrcdemux: Fix compiler warning when compiling with G_DISABLE_ASSERT 2014-06-29 19:59:53 +02:00
Sebastian Dröge
5500dd4a20 matroskamux: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 19:57:57 +02:00
Sebastian Dröge
b03a4d9155 deinterlace: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 19:54:44 +02:00
Tim-Philipp Müller
155a3fec93 matroskaparse: don't error out if there's not enough data in the adapter
gst_matroska_parse_take() would return FLOW_ERROR instead of
FLOW_EOS in case there's less data in the adapter than requested,
because buffer is NULL in that case which triggers the error
code path. This made the unit test fail (occasionally at least,
because of a bug in the unit test there's a race and it would
happen only sporadically).
2014-06-28 17:39:36 +01:00
Sebastian Dröge
c0f5644b80 videomixer: Update dist generated ORC files 2014-06-28 16:56:18 +02:00
Sebastian Dröge
db43a39bbf videomixer: Update videoconvert code from -base
And also rename the remaining symbols to prevent conflicts
during static linking.

https://bugzilla.gnome.org/show_bug.cgi?id=728443
2014-06-28 16:56:18 +02:00
Tim-Philipp Müller
8b7f0ae3fe autovideosrc: use videotestsrc as fallback element instead of fakesrc
fakesrc doesn't announce video caps, so most video pipelines will
just error out with not-negotiated if a fallback element is created.
2014-06-28 14:25:25 +01:00
Tim-Philipp Müller
7dcc3ffe5a autoaudiosrc: use audiotestsrc as fallback element instead of fakesrc
fakesrc doesn't announce audio caps, so most audio pipelines will
just error out with not-negotiated if a fallback element is created.
2014-06-28 14:25:25 +01:00
Thibault Saunier
45b9ef1825 videomixer: Declare as Compositor in 'klass' 2014-06-26 17:49:23 +02:00
Tim-Philipp Müller
e9f2d63011 flvdemux: fix speex caps
Decoder complains about "notification: Invalid mode encountered.
The stream is corrupted" though, even if it works, so there's
probably something wrong with the generated codec headers.
2014-06-26 13:50:19 +01:00
Tim-Philipp Müller
d98b996523 flvmux: fix speex in FLV
Speex in FLV is always mono @ 16kHz, see
http://download.macromedia.com/f4v/video_file_format_spec_v10_1.pdf
section E.4.2.1: "If the SoundFormat indicates Speex, the audio is
compressed mono sampled at 16 kHz, the SoundRate shall be 0, the
SoundSize shall be 1, and the SoundType shall be 0"

Also see https://bugzilla.gnome.org/show_bug.cgi?id=683622
2014-06-26 13:43:33 +01:00
Jan Schmidt
8da6ee0312 isomp4: Add object type id and fourcc for DTS/DTS-HD
Enables playback for files with DTS audio tracks.
Also add an extra AC-3 variant fourcc from Nero
2014-06-26 19:57:41 +10:00
David Fernandez
4ed74d3ab0 videomixer2: Solve segmentation fault when src caps are configured
Change function pointers to NULL while holding the lock to avoid
race conditions

https://bugzilla.gnome.org/show_bug.cgi?id=701110
2014-06-25 16:44:38 +02:00
Wim Taymans
ca9cfd40dd jitterbuffer: improve SR packet handling
Implement 3 different cases for handling the SR:

 1) we don't have enough timing information to handle the SR packet and
    we need to wait a little for more RTP packets. In that case we keep
    the SR packet around and retry when we get an RTP packet in the
    chain function.

 2) the SR packet has a too old timestamp and should be discarded. It is
    labeled invalid and the last_sr is cleared.

 3) the SR packet is ok and there is enough timing information, proceed
    with processing the SR packet.

Before this patch, case 2) and 1) were handled in the same way,
resulting that SR packets with too old timestamps were checked over and
over again for each RTP packet.
2014-06-25 16:14:46 +02:00
Olivier Crête
64f28e2552 avimux: Add UYVY format 2014-06-23 19:55:29 -04:00
Miguel París Díaz
b22aed9bbc gstrtpssrcdemux: manage ssrc of RTCP RR packets
https://bugzilla.gnome.org/show_bug.cgi?id=731324
2014-06-23 16:23:00 -04:00
Sebastian Dröge
efaf996b1a wavparse: Update offset after parsing adtl chunk
Otherwise we will parse it over and over again without ever
getting past it.

https://bugzilla.gnome.org/show_bug.cgi?id=731533
2014-06-23 20:53:50 +02:00
Sebastian Dröge
daf25482ed matroskademux: Don't call GST_DEBUG_OBJECT() and other macros with non-GObject objects
It will crash with latest GLib GIT and was never supposed to work before
either.
2014-06-22 19:26:03 +02:00
Tim-Philipp Müller
41c895de4d multiudpsink: optimisation: avoid unnecessary memory ref/unrefs
We know the buffer will stay valid and we will also not
modify the buffer, we just want to send out the data.
2014-06-20 12:21:05 +01:00
Tim-Philipp Müller
3512ad3be0 multiudpsink: avoid some unnecessary run-time type checks 2014-06-20 12:06:57 +01:00
Wim Taymans
98a4ee0f92 rtspsrc: pass the stream id when asking for crypto params
This way the app can choose different parameters for each stream.
2014-06-19 16:17:23 +02:00
Aleix Conchillo Flaqué
7ce0ea3946 rtspsrc: add support for key length parameters
This patch adds supports for the incoming key management parameters for
encryption and authentication key lengths.

It also adds a new signal request-rtcp-key that allows the user to
provide the crypto parameters and key for the RTCP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=730473
2014-06-19 16:11:19 +02:00
Wim Taymans
8a78fa1ff5 vp8depay: fix header size checking
Use a different variable name to make it clear that we are calculating
the header size.
Correctly check that we have enough bytes to read the header bits. We
were checking if there were 5 bytes available in the header while we
only needed 3, causing the packet to be discarded as too small.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595
2014-06-19 15:29:46 +02:00
Guillaume Desmottes
f00c2b7155 rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:49 +02:00
Guillaume Desmottes
4be99ec7d5 rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).

We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:38 +02:00
Guillaume Desmottes
42ff642372 gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Guillaume Desmottes
9a7479fb0d rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Tim-Philipp Müller
460ab3dd76 avidemux: don't leak flow combiner 2014-06-18 15:03:25 +01:00
Tim-Philipp Müller
6347ec522d rtpjp2kpay: pre-allocate buffer-list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
ccb7380689 rtpjpegpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
70bfc35756 rtpmp4vpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
4b1f771e4d rtpvp8pay: allocate bitreader on the stack 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
725b8f272b rtpvp8pay: post error message on bus on error and don't use g_message() 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
f4db7443ae rtpvp8pay: couple of minor optimisations
Pre-allocate buffer list of the right size to avoid re-allocs.
Avoid plenty of double runtime cast checks and re-doing the
same calculation over and over again in rtp_vp8_calc_payload_len().
Only call gst_buffer_get_size() once.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
6c9e2194d2 rtpgstpay: pre-allocate buffer list of the right size
To avoid re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
01ee993d8d rtph264pay: pre-allocate bufferlist of the right size
To avoid unnecessary re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
c7c72c00b1 rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
0f5da64de3 rtpvrawpay: make chunks per frame configurable
Bit of a misnomer because it's really chunks per field
and not per frame, but we're going to ignore that for
the time being.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
2cf13b603f rtpvrawpay: remove unused variables 2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
a09e237b85 rtpvrawpay: pre-allocate buffer lists of sufficient size
Avoids unnecessary reallocs when appending buffers
to the bufferlist.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
15a33ccc65 rtpvrawpay: micro-optimise variable access in inner loop
Store some values that don't change during the execution
of the inner loops locally, so the compiler knows that too.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
fdf95fecbd rtpvrawpay: use buffer lists
Collect buffers to send out in buffer lists instead of
pushing out single buffers one at a time. For HD video
each frame might easily add up to a couple of thousand
packets, multiply that by the frame rate and that's a
lot of push() and sendmsg() calls per second.

A good reason to push out buffers as early as possible is
latency, so we don't accumulate the whole frame in a single
buffer list, but instead push it out in a few chunks, which
is hopefully a reasonable compromise.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
884d1af074 udp: improve element descriptions for dynudpsink and multiudpsink 2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
6c1231eed3 udp: remove suppression of compiler warnings for deprecated GLib API
Not needed any more.
2014-06-18 14:54:58 +01:00
Ravi Kiran K N
3c4c130c5e videobox: Fix caps negotiation issue
Make sure that if AYUV is received it will detect that it can produce
both RGB and YUV formats

Signed-off-by: Ravi Kiran K N <ravi.kiran@samsung.com>

https://bugzilla.gnome.org/show_bug.cgi?id=725248
2014-06-17 09:27:45 -03:00
Tim-Philipp Müller
054f774455 rtptheoradepay: fix double frees
Fix double-frees introduced to fix another coverity report.

CID 1223053
2014-06-16 12:03:38 +01:00
Tim-Philipp Müller
bb51ec5842 dynudpsink: return FLUSHING when sendto got canceled, not an error 2014-06-13 10:12:07 +01:00
Vincent Penquerc'h
25c26a4c4c rtptheordepay: fix leaks
Coverity 1212163
2014-06-12 11:24:15 +01:00
Vincent Penquerc'h
8e80478cf7 rtpg729pay: leak fixes
Coverity 1212159
2014-06-12 11:16:08 +01:00
Vincent Penquerc'h
fe4c5b92b1 rtph263pay: fix leak
Coverity 1212157
2014-06-12 11:11:38 +01:00
Vincent Penquerc'h
6ef26e4a8a rtph263pay: fix leaks
Coverity 1212149
2014-06-12 10:43:53 +01:00
Vincent Penquerc'h
c58a2d9bbb rtpdvpay: catch failures to map buffer
Coverity 1139741
2014-06-12 10:31:47 +01:00
Vincent Penquerc'h
7e278e6b22 multipartdemux: guard against having no MIME type
The code would previously crash trying to insert a NULL string
into a hash table.
It does seem a little broken that indexing is done by MIME type
and not by index though, unless the spec says there cannot be
two parts with the same MIME type.

https://bugzilla.gnome.org/show_bug.cgi?id=659573
2014-06-11 17:44:56 +01:00
Nicolas Dufresne
9966fdfa75 multipartdemux: Send stream-start event
This event was not sent. Send it before caps, this requires the pad to
be parented. This removes warning like: "Got data flow before
stream-start event".

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475
2014-06-10 15:43:21 -04:00
Thiago Santos
9fda7b107f qtdemux: avoid looping indefinitely in broken svq3 files
Abort if an atom with size 0 is read from within the svq3 stsd
atoms

https://bugzilla.gnome.org/show_bug.cgi?id=726512
2014-06-10 15:33:33 -03:00
Edward Hervey
f7fc8d74c9 flvdemux: Attempt upstream seek first
If we have an upstream element that can handle the seek (such as
rtmpsrc), try to do that first before attempting it ourself.
2014-06-09 10:04:38 +02:00
Vincent Penquerc'h
40ae581ef2 wavparse: do not include codec_data on raw audio caps
If the wav header contains an extended chunk, we want to keep
the codec_data field, but not for raw audio.

This fixes some elements (such as adder) from failing to intersect
raw audio caps which would otherwise be intersectable.
2014-06-05 10:34:49 +01:00
Edward Hervey
2b9493b5f0 flvdemux: Query duration upstream first
Upstream elements (like rtmpsrc) might be able to provide the duration
more accurately than flvdemux. Especially with index-less vod files
2014-06-05 09:38:29 +02:00
Jan Alexander Steffens (heftig)
303883752e flvdemux: set RESYNC buffer flag when bridging large PTS gaps
So downstream gets notified when this happens.

https://bugzilla.gnome.org/show_bug.cgi?id=725903
2014-06-04 10:28:47 -04:00
Tim-Philipp Müller
341b691b18 matroskademux: don't leak doctype string in error code path
CID 1212145.
2014-06-02 09:57:42 +02:00
Thiago Santos
c25d94b7ef qtdemux: upstream handles seek if fragmented and on time segment
Otherwise we can reject seeks on local files that contain fragmented-like
atoms like 'mvex'. Also improve a message log

https://bugzilla.gnome.org/show_bug.cgi?id=730722
2014-05-30 15:01:50 -03:00
Wim Taymans
a5a7649831 h264depay: make sure we call handle_nal for each NAL
Call handle_nal for each NAL in the STAP-A RTP packet. This makes
sure we correctly extract the SPS and PPS.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999
2014-05-30 16:51:37 +02:00
Thiago Santos
fd6b348898 avidemux: remove stream last flow return
GstPad already stores that information

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:13 -03:00
Thiago Santos
2b454bf87f qtdemux: remove last flow return from stream struct
It is already stored on GstPad on core

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:12 -03:00
Thiago Santos
3b887887be flvdemux: Use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:07 -03:00
Thiago Santos
c7c25071e3 matroskademux: use GstFlowCombiner
Use the flow combiner to have the standard combination results and avoid
repeating the same code

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:51:02 -03:00
Thiago Santos
da3c031627 avidemux: use GstFlowCombiner
Removes flow return combination code to use the newly added GstFlowCombiner
2014-05-26 15:30:12 -03:00
Thiago Santos
4b0ce7dc30 qtdemux: use GstFlowCombiner
Removes the common code to combining flow returns to let it be
handled by core gstutils' GstFlowCombiner

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 15:30:12 -03:00
Thiago Santos
d423b9f63e qtdemux: parse tkhd transformation matrix and add tags if appropriate
Handle the transformation matrix cases where there are only simple rotations
(90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
when recording with mobile devices

https://bugzilla.gnome.org/show_bug.cgi?id=679522
2014-05-24 15:38:54 -04:00
Thiago Santos
f0b99d96a9 qtdemux: add tag mappings for _swr, _mak and _mod tags
swr -> Application name
mak -> device manufacturer
mod -> device model
2014-05-23 03:15:42 -03:00
Sebastian Dröge
1cdd3765d6 goom: Use fabs() instead of abs() to calculate the floating point absolute value
tentacle3d.c:268:7: error: using integer absolute value function 'abs' when
      argument is of floating point type [-Werror,-Wabsolute-value]
  if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) {
      ^
2014-05-19 11:24:06 +02:00
Sebastian Dröge
97fb3655df debugutils: Properly calculate the difference with unsigned types
tests.c:161:16: error: taking the absolute value of unsigned type
      'unsigned long' has no effect [-Werror,-Wabsolute-value]
    t->diff += labs (GST_BUFFER_TIMESTAMP (buffer) - t->expected);
2014-05-19 11:21:36 +02:00
Aleix Conchillo Flaqué
782d65cab1 rtspsrc: always use a random ssrc for the internal session
Use a random SSRC different than 0 for the internal session SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=730212
2014-05-16 16:58:44 +02:00
Wim Taymans
d004eda79d rtpsession: update last_activity when sending RTP
Also update last_activity when doing something with the internal
source to make sure don't timeout early.

See https://bugzilla.gnome.org/show_bug.cgi?id=730217
2014-05-16 16:55:17 +02:00
Aleix Conchillo Flaqué
a62b280873 rtpbin: update rtp encoder/decoder docs
Use %u in RTP encoder/decoder pads to match other rtpbin pads.

https://bugzilla.gnome.org/show_bug.cgi?id=730146
2014-05-15 15:48:21 +02:00
George Kiagiadakis
7e2138794f rtpsession: remove unused if branch
1) sources that have sent BYE in the past cannot be senders, since
they would have timed out to being receivers in the meantime...
2) sources that have sent BYE are now being removed earlier inside
this function
2014-05-14 16:01:50 +02:00
George Kiagiadakis
85d4c031d4 rtpsession: cleanup sources that have sent BYE 2014-05-14 16:01:50 +02:00
George Kiagiadakis
7d7840cc4a rtpsession: unify nested if clauses 2014-05-14 16:01:50 +02:00
George Kiagiadakis
0e6a31411b rtpsession: timeout internal sources that are inactive for a long time and send BYE 2014-05-14 16:01:50 +02:00
Aleix Conchillo Flaqué
bcd469ff31 rtpjitterbuffer: don't stop looping if event found in the queue
If we are inserting a packet into the jitter queue we need to keep
looping through the items until the right position is found. Currently,
the code stops as soon as an event is found in the queue.

Regarding events, we should only move packets before an event if there
is another packet before the event that has a larger seqnum.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078
2014-05-14 10:23:28 +02:00
Adrien SCH
8ac30d4c26 matroskamux: fix the memory leak of language attribute
https://bugzilla.gnome.org/show_bug.cgi?id=728418
2014-05-13 19:55:21 -03:00
Edward Hervey
420661bd95 qtdemux: Fix leak of palette_data in error cases
CID #1212151
2014-05-12 16:56:35 +02:00
Edward Hervey
112d948b7e qtmux: Free node_header in error cases
CID #1212134
2014-05-12 16:53:32 +02:00
Edward Hervey
6c4882996f flvdemux: Don't use WARNING for not-linked flow return
Pollutes debug logs for no reason. It's only an error if all pads
return not-linked
2014-05-12 13:46:01 +02:00
Edward Hervey
c09b14c931 flvdemux: Skip unknown tags in push-mode
We add a new mode (SKIP) in push-mode to skip tags that we don't known about

Partially fixes https://bugzilla.gnome.org/show_bug.cgi?id=670712
2014-05-12 13:45:06 +02:00
Wim Taymans
b2e1598e4a rtpjitterbuffer: increment accepted packets after loss
When we detect a lost packet, expect packets with higher
seqnum on the input.

Also update the unit test.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729524
2014-05-09 18:10:32 +02:00
Jason Litzinger
9068e1bb8e Add new test case. 2014-05-09 18:10:32 +02:00
Eric Trousset
bd51aa7aa8 qtdemux: don't respond to a position query in BYTE format with a TIME position
https://bugzilla.gnome.org/show_bug.cgi?id=729553
2014-05-09 16:12:45 +01:00
Tim-Philipp Müller
9872c19491 matroskademux: don't leak doctype string in error code path
CID 1212145.
2014-05-09 14:22:42 +01:00
Tim-Philipp Müller
615f6e55c1 flacparse: skip PICTURE headers without any image data
Fixes warning if the image length is 0.
2014-05-07 00:58:15 +01:00
Guillaume Desmottes
d089f99a39 rtp/README: update pipelines to work with 1.0
- Use gst-libav encoders/decoders instead of gst-ffmpeg
- gstrtpjitterbuffer -> rtpjitterbuffer
- gst-launch-0.10 -> gst-launch-1.0
- Add 'videoconvert' element
- xvimagesink -> autovideosink

https://bugzilla.gnome.org/show_bug.cgi?id=729247
2014-05-05 20:23:56 -04:00
Vincent Penquerc'h
ec38c62563 matroska: rejig test to avoid undefined shift behavior
Coverity 1195121, 1195120
2014-05-05 14:44:57 +01:00
Vincent Penquerc'h
9589c43516 matroskamux: ensure we don't dereference a NULL pointer
while working out the codec ID.

Coverity 1195148
2014-05-05 14:32:06 +01:00
Olivier Crête
b2a52035bf rtprtxreceive: Wait until timeout to clear association requests
If two streams request a retranmission for the same SSRC, ignore the second
one if the first oen is less than one second old, otherwise time out the first
one and ignore the second.
2014-05-04 22:36:59 -04:00
Olivier Crête
0742a5a257 rtpmux: Always let upstream chose the ssrc if it wishes 2014-05-04 19:11:03 -04:00
Mark Nauwelaerts
6c584bc833 rtpjitterbuffer: avoid stall by corrupted seqnum accounting 2014-05-04 13:38:26 +02:00
Olivier Crête
2e54d38dd0 rtpsession: Keep local conflicting addresses in the session
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.

Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
2014-05-03 18:30:20 -04:00
Sebastian Dröge
1d4404d883 Release 1.3.1 2014-05-03 18:02:23 +02:00
Sebastian Dröge
859c751a5d imagefreeze: Set segment position to the stop position of the buffer 2014-05-02 17:14:29 +02:00
Sebastian Dröge
4282d75597 imagefreeze: Properly report errors before stopping the srcpad task 2014-05-02 17:14:29 +02:00
Sebastian Dröge
4933394d35 imagefreeze: Error out if we have no caps yet 2014-05-02 17:14:29 +02:00
Vincent Penquerc'h
218294b9f3 wavparse: avoid dividing by a 0 blockalign
This can be 0. In that case, do not try to cut off the last few
bytes from the last buffer.

Coverity 1146971
2014-05-02 14:49:27 +01:00
Vincent Penquerc'h
590e20cbc9 matroskamux: do not use uinitialized clut on error
If we're missing part of the clut, do not try to use it. It seems
very likely the break was meant to break out of the switch rather
than from the loop.

Coverity 1139878
2014-05-02 14:25:01 +01:00
Vincent Penquerc'h
d917c94037 flxdec: fix integer overflow
Coverity 1139859
2014-05-02 14:18:08 +01:00
Vincent Penquerc'h
60ba2d7aee rtpqdmdepay: remove pointless check
Besides, the pointer was dereferenced earlier anyway.

Coverity 1139853
2014-05-02 14:09:02 +01:00
Vincent Penquerc'h
a846e84349 rtspsrc: remove duplicate test
item was dereference previously.

While there, reorder some test for faster early out.

Coverity 1139844
2014-05-02 14:06:25 +01:00