Commit graph

3878 commits

Author SHA1 Message Date
Tim-Philipp Müller
b163f111c8 rtspdefs: remove outdated comments 2014-05-02 19:36:34 +01:00
Vincent Penquerc'h
db8460cbd8 rtpbuffer: avoid underflow in size calculation 2014-05-02 15:10:07 +01:00
Thiago Santos
5ac9476b40 videodecoder: do not parse caps for not using it
Saving some cpu
2014-05-01 19:31:25 -03:00
John Bassett
0fd60ac858 rtpbasepayload: restrict initial random sequence number to be <= 32767
In order to prevent SRTP roll over counter issues the initial sequence
number is restricted to <= 32767. This is recommended by RFC 4568 section 6.4.
2014-05-01 17:00:47 -04:00
Sebastian Dröge
cceb3303ab sdp: Add some more gobject-introspection annotations for bindings
https://bugzilla.gnome.org/show_bug.cgi?id=729123
2014-05-01 15:11:04 +02:00
Vincent Penquerc'h
ffdf87b121 rtcpbuffer: check claimed data size against available size
Coverity 1208773
2014-04-30 18:13:15 +01:00
Göran Jönsson
9685e7a583 rtspconnection: Empty queue when flush.
Empty the watchs queue when calling
gst_rtsp_watch_set_flushing with flushing variabel is TRUE.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728772
2014-04-30 16:37:17 +02:00
Haakon Sporsheim
7c97a1c6cf audiodecoder: Make caps writable before fixating
https://bugzilla.gnome.org/show_bug.cgi?id=729114
2014-04-29 09:58:21 +02:00
Sebastian Dröge
05ed7d5ba7 sdpmessage: Add array length annotation to gst_sdp_message_parse_buffer
https://bugzilla.gnome.org/show_bug.cgi?id=729123
2014-04-29 09:54:38 +02:00
Stian Selnes
0011d8cbb5 rtpbuffer: fix memory leak when gst_rtp_buffer_map fails
Make sure rtp->data[3] is set before jumping to error path.

https://bugzilla.gnome.org/show_bug.cgi?id=729117
2014-04-29 08:46:29 +02:00
Tim-Philipp Müller
bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
2014-04-26 23:28:57 +01:00
Edward Hervey
74eb5fa995 audiodecoder: Plug caps leaks
We were returning in various places without unreffing the caps, and
we were also leaking (overwriting) the caps we got from _get_current_caps()

Spotted by Haakon Sporsheim in #gstreamer
2014-04-25 11:30:37 +02:00
Thiago Santos
03ba34f3a8 tag: exif: avoid adding empty strings
Fixes assertion with some jpeg files
2014-04-22 00:21:45 -03:00
Sebastian Dröge
9398207101 videoframe: Initialise GstVideoFrame to zeroes if mapping fails
This should allow for more meaningful errors. Dereferencing NULL
is more useful information than dereferencing a random address
happened to be on the stack.
2014-04-16 19:53:46 +02:00
Vincent Penquerc'h
22e12979ee exiftag: catch buffer mapping failure
Might be what caused:

Coverity 1139734
2014-04-16 11:43:40 +01:00
Sebastian Dröge
dce51830f4 encoding-profile: Free preset name in finalize 2014-04-15 13:48:17 +02:00
Wim Taymans
f2ee068729 rtpbasedepay: add stats property
Add a stats property that holds a structure with all the current
values of the depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=646577
2014-04-12 07:10:36 +02:00
Wim Taymans
314eee6dd1 rtpbasepayload: update docs 2014-04-12 06:43:24 +02:00
Wim Taymans
f0348d7005 rtpbasepayload: add current timestamp and seqnum offset to stats
Expose the current timestamp and seqnum offset in the stats

See https://bugzilla.gnome.org/show_bug.cgi?id=646577
2014-04-12 06:27:36 +02:00
Vincent Penquerc'h
3d1fdf8dd8 video-overlay-composition: guard against NULL pointer dereference on error
If gst_video_overlay_rectangle_apply_global_alpha is called with
a rectangle with unsuitable alpha, expanding the alpha plane will
fail, and thus lead to dereferencing a NULL src pointer. It's not
certain this will happen in practice, as the function is static
and callers might ensure suitable alpha before calling, but there
is no apparent explicit such check.
Add prologue asserts for proper alpha to explicitely prevent this.

Coverity 1139707
2014-04-10 12:35:03 +01:00
Vincent Penquerc'h
4a401adf38 videometa: fix texture_type memcpy size
Coverity 1139589, 1139588
2014-04-10 12:35:03 +01:00
Vincent Penquerc'h
505c803765 sdpmessage: fix multi statement macros
Wasn't playing nice with an if statement below.

Coverity 1139767
2014-04-10 12:35:03 +01:00
Vincent Penquerc'h
dda777803c audiocdsrc: guard aginst overflow
An audio CD may contain about a tenth of the samples 32 bit can
represent, so it doesn't seem likely this will be hit in practice.

Coverity 1139805
2014-04-10 12:35:03 +01:00
Tim-Philipp Müller
374e756eee pbutils: descriptions: default to systemstream=false for partial video/mpeg caps
Assume systemstream=false for video/mpeg caps where that field
is missing.
2014-04-10 12:30:50 +01:00
Vincent Penquerc'h
7618699ffd audiobasesink: avoid possible sample count overflow
At 48 kHz, 2<<31 samples is reached before 13 hours so it
sounds plausible this would be hit.

Coverity 1139800, 1139801
2014-04-10 11:06:00 +01:00
Sebastian Dröge
8c7cbeed5b video-event: Update the running times in the force-keyunit events from the pad offsets 2014-04-10 09:18:05 +02:00
Wim Taymans
4a81605d02 sdp: guard against address parse errors. 2014-04-08 15:59:47 +02:00
Josep Torra
6ce7ade7c6 audioringbuffer: parse channels field from compressed audio caps
Also parse channels as an optional field in the caps for compressed
audio formats.
2014-04-08 12:54:04 +02:00
Thiago Santos
05e957106f videodecoder: do not deactivate the bufferpool, just unref
Videodecoder does late renegotiation, it will wait for the next
buffer before renegotiating its caps and bufferpool. It might happen
that downstream element switched from passthrough to non-passthrough
and sent a reconfigure upstream (that caused this renegotiation).
This downstream element will ask the video sink below for the bufferpool
with an allocation query and will get the same bufferpool that
videodecoder is holding, too.

When renegotiating, if videodecoder deactivates its bufferpool it
might be deactivating the bufferpool that some element downstream
is using and cause the pipeline to fail.

https://bugzilla.gnome.org/show_bug.cgi?id=727498
2014-04-04 13:50:03 -03:00
Vincent Penquerc'h
169166d0a2 audiobasesink: clip start samples to match clipped start time
Clock slaving can clip start time to zero, giving us a shorted
duration than we originally got. To keep in sync, we must then
discard the samples falling before that zero timestamp.

This possibly fixes random distortion caused by constant PA
underflows which are never resynced.
2014-04-04 17:04:06 +01:00
Wim Taymans
675d0400e1 mikey: Fix the KEMAC payload
The KEMAC payload actually needs to have subpayloads and the key should
go into the KEY_DATA subpayload. Add support for subpayloads and
implement the KEY_DATA payload.
Add some pointers to the conversion functions that allow us to add
encryption and decryption later.
2014-04-04 17:40:58 +02:00
Sebastian Dröge
6189847ed0 videodecoder: Always drain the decoder after a discont group in reverse playback mode 2014-03-30 18:26:59 +02:00
Sebastian Dröge
5a4fbb1638 videodecoder: Flush the decoder once per discont group, not once per keyframe 2014-03-30 18:00:53 +02:00
Sebastian Dröge
f1f8731ff5 videodecoder: Handle reverse playback with multiple GOPs per discont group properly
baseparse will reverse each GOP for us already, so the segment events can
be after our keyframe. Make sure to get it and all other relevant sticky
events before starting to decode.
2014-03-30 17:59:55 +02:00
Sebastian Dröge
50c2218d4d videodecoder: Log event types of events that are pushed downstream 2014-03-29 10:33:01 +01:00
Sebastian Dröge
1c26e5734c videodecoder: In reverse playback mode we need to finish the subclass after passing all frames to it 2014-03-29 10:33:01 +01:00
Wim Taymans
8d439edd7a rtspconnection: add flush method
Add a method to set/unset the flushing state that makes _wait_backlog()
unlock.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-28 09:34:33 +01:00
Wim Taymans
0348ee66f1 mikey: fix return values of g_return_* 2014-03-25 11:14:51 +01:00
Wim Taymans
183e441d88 rtsptransport: UDP is also default for SAVP and AVPF 2014-03-25 11:07:34 +01:00
Wim Taymans
51ca0bdf7b docs: add MIKEY docs 2014-03-24 17:12:52 +01:00
Wim Taymans
83888d6b13 mikey: add MIKEY parsing helpers
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
2014-03-24 17:12:52 +01:00
Ognyan Tonchev
d7857325c5 rtspconnection: Fix minor memory leaks in error handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
2014-03-24 12:45:14 +01:00
Ognyan Tonchev
e0af857445 rtspconnection: Fix connection_poll()
* Only check for conditions we are interested in.
* Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
  will always be reported if they are true.
* Do not create timed source if timeout is NULL.
* Correctly wait for sources to be dispatched, context_iteration() is
  not guaranteed to always block even if set to do so.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
2014-03-24 12:43:38 +01:00
Wim Taymans
bf4079277d rtpbasepayload: add pt and ssrc to stats 2014-03-20 09:19:46 +01:00
Руслан Ижбулатов
d6bd37460a rtspconnection: Silence a compiler warning
Cast the argument into (const char *) on W32, as winsock2 expects it.

https://bugzilla.gnome.org/show_bug.cgi?id=726433
2014-03-16 11:22:04 +01:00
Göran Jönsson
0b30fdbfbe rtspconnection: gst_rtsp_watch_wait_backlog
New method that wait until there is room in backlog queue.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-10 17:28:40 +01:00
David Svensson Fors
6cd0d10d30 rtspconnection: GstRTSPWatch func for tunnel GET response
Add a callback in GstRTSPWatch where the response to HTTP GET for
tunneled connections can be modified.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
2014-03-10 10:43:03 +01:00
Wim Taymans
4898c30537 rtspdefs: add RFC 4567 headers and status code
This new Header and status code is used for SRTP
2014-03-10 10:33:28 +01:00
Matthieu Bouron
a8951c16da video-overlay-composition: add GST_CAPS_FEATURE_META_GST_VIDEO_OVERLAY_COMPOSITION 2014-03-05 20:38:45 +01:00
Ognyan Tonchev
4220442441 rtspconnection: Call closed() when GET is closed in tunneled mode
This patch adds read source on the write socket in tunneled
mode and we get a callback when client disconnects the GET
channel.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313
2014-03-03 10:34:56 +01:00