Commit graph

3855 commits

Author SHA1 Message Date
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
Sebastian Rasmussen 900c204eb9 videoformat: Remove duplicate/incorrect section
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:41:51 +00:00
Sebastian Rasmussen 35bb1b3328 docs: Add annotations for return values
Rephrase and clarify some return value descriptions

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:41:18 +00:00
Sebastian Rasmussen 5b4f2ba20b docs: Fix argument and annotation typos
* colorbalance: Fix misspelled annotation
 * rtsp: Replace incorrectly documented function argument
 * sdp: Escape @ character to avoid gtk-doc warning
 * video-*: Add missing annotation colon
 * videodecoder/video-color: Fix function argument typos
 * videoutils: Remove unknown annotation field

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:22:51 +00:00
Tim-Philipp Müller 14b82bbc9a rtsp: fix build with older GLib versions
The gio/gnetworking.h header is only available since glib 2.36

https://bugzilla.gnome.org/show_bug.cgi?id=725206
2014-02-26 11:44:18 +00:00
Ognyan Tonchev 5445682c6a rtspconnection: Add missing include
https://bugzilla.gnome.org/show_bug.cgi?id=725206
2014-02-26 11:25:13 +00:00
Sebastian Rasmussen d6dc1b6c46 rtpbasepayload: Let caps event also configure seqnum-offset
Previously the sequence number kept track of by GstRTPBasePayload would
only be set when going from READY to PAUSED state. This meant that a
downstream element that attempted to configure a basepayloader by
setting seqnum-offset e.g. in its sinkpad's caps template would have
trouble configuring the basepayloader. The reason was that the caps
event which arrives with the desired value for seqnum-offset did not
arrive at the basepayloader until caps negotiation took place,
significantly later than the transition from READY to PAUSED.

The result after this patch is that the default value for the
seqnum-offset property, or later set values for this property, will take
effect when going from READY to PAUSED like before. In addition the an
arriving caps event will also affect the basepayloaders configured
sequence number as the event arrives.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen 638d069c91 rtpbasepayload: Fix payload type property boundary value
The payload type field in an RTP packet header is 7 bits wide, hence the
boundary values ought to be 0x00 and 0x7f, not the previously stated
values 0x00 and 0x80.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen 3cc67ff494 rtpbasedepayload: Fix typos in comments 2014-02-24 12:10:26 +01:00
Tim-Philipp Müller 6442e76e9f docs: add GstVideoPool to docs 2014-02-23 14:42:12 +00:00
Ognyan Tonchev ebe3530f51 rtspconnection: Remove read child source when POST is disconnected
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720
2014-02-21 16:21:45 +01:00
Aleix Conchillo Flaqué 0a115bd31f rtspconnection: allow specifying a certificate database
Two new functions have been added,
gst_rtsp_connection_set_tls_database() and
gst_rtsp_connection_get_tls_database(). The certificate database will be
used when a certificate can't be verified with the default database.

https://bugzilla.gnome.org/show_bug.cgi?id=724393
2014-02-19 21:48:13 +01:00
Aleix Conchillo Flaqué 9121b16aa0 rtspconnection: get rid of superfluous whitespaces 2014-02-19 21:22:30 +01:00
Nicolas Dufresne 6b77971097 video: Fix NV12_64Z32 default offset and size
This was a regression introduced by f52fd7a68, where we started using
the stride to encode the dimensions in tiles. This patch simply updates
offset and size calculation as described in the documentation,
part-mediatype-video-raw.txt.
2014-02-18 13:09:21 -05:00
Rafał Mużyło 5496d09eb4 audio: map channels=1,channel-mask=0 to MONO instead of NONE
Fixes problem in audioconvert, which would end up using
a mixmatrix when converting between different mono format
because it thinks MONO positioning is different from
unpositioned channels, which is not the case in this
special case. The mixmatrix would end up being 0.0 so
audioconvert would convert to silence samples.

https://bugzilla.gnome.org/show_bug.cgi?id=724509
2014-02-18 10:41:47 +00:00
Sebastian Dröge bc92cd8f67 audiosrc: Fix typo in docs
We read *from* the audio device, not to it.
2014-02-09 11:28:48 +01:00
Eric Trousset 2ca256acdb tagdemux: Forward TIME seeks upstream too, maybe upstream can handle that
https://bugzilla.gnome.org/show_bug.cgi?id=723597
2014-02-04 13:56:29 +01:00
Stefan Sauer 76ec6d3760 docs: doc fixes for audio library
Add sections docs for audiometa. Fix sections docs for audiochannels. Remove old
mixerutil section.
2014-02-03 09:36:43 +01:00
Thiago Santos e00dc5b879 audioencoder: push pending events and tags before EOS
if there are tags or events pending and an EOS is received, push those
events and tags before the EOS.
2014-01-29 12:33:59 -03:00
Thiago Santos da54836a33 videoencoder: push tags and events before eos
if any tags or events are pending, push them before pushing eos
2014-01-29 12:33:59 -03:00
Sebastian Rasmussen 125b9c19c0 rtpbasepayload: Do cosmetic changes to rtptime calculations
* Change running time type to guint64
 * Use GST_CLOCK_TIME_NONE() to check for invalid timestamps
 * Name variables so ns-based and hz-based timestamps are evident

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen 0142cd5e35 rtpbasepayload: Expose running-time of payloaded stream
https://bugzilla.gnome.org/show_bug.cgi?id=719415
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen 865a5d1c8f rtpbasepayload: Improve documentation for perfect-rtptime
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00
Sebastian Rasmussen 713dfe0d70 rtpbasepayload: Fix typos in documentation for properties
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
2014-01-28 14:41:41 +01:00