mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-28 12:41:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
8117 commits 44 branches 1509 tags 436 MiB
Commit graph

8117 commits

This branch
This branch
All branches
Author SHA1 Message Date
Mark Nauwelaerts
f79de81bb4 flvdemux: also collect index info on-the-fly in pull mode 2010-03-10 11:48:07 +01:00
Mark Nauwelaerts
86a1aec2c0 flvdemux: incrementally build index in pull mode 
Scan for needed part upon a seek as opposed to doing a complete scan
at startup, which may take some time depending on file and/or platform.
Also accept index metadata in pull mode and peek for some metadata
at the end of the file when deemed appropriate.
 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
66fabd8bfd flvdemux: some more variable cleanup 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
122daaf6af flvdemux: refactor adding index entry 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
6aa4d5df2d flvdemux: fix setting DELTA_UNIT flag on outgoing buffers 
... which should not depend on having index available or not.
Also refactor resulting collapsed code.
 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
561a506822 qtdemux: avoid erroneous codec-data overriding of stsd information 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
4abf31067f speexdec: adapt to new oggdemux 
Remove all granulepos hacks and simply use upstream timestamps.
 2010-03-10 11:48:06 +01:00
Mark Nauwelaerts
972ce3006b speexdec: refactor granulepos hacks 2010-03-10 11:48:06 +01:00
Wim Taymans
821096c4f1 rtspsrc: parse connection information 
Parse the connection information from the SDP and use it to figure out if we are
dealing with ipv4 or ipv6 connections.
 2010-03-10 11:28:22 +01:00
Wim Taymans
8eb5c2c794 rtspsrc: require a destination for multicast 
When setting up the multicast sockets, we need a destination address to listen
on or else we error.
 2010-03-10 11:21:20 +01:00
Wim Taymans
574447b092 rtspsrc: handle ipv6 listening ports when needed 
Add some code to make udpsrc listen on an ipv6 address when needed. The
detection of IPV6 is not yet implemented.
 2010-03-10 11:21:20 +01:00
Wim Taymans
455f53c896 udp: use uri parsing code 
Use the uri parsing helper functions to manage the host and port pairs. This
adds support for IPV6.
 2010-03-10 11:21:19 +01:00
Wim Taymans
14ae2080d2 udpnetutils: add helper functions for udp uri handling 
Add some helpers to parse udp uris. Make sure IPV6 is supported too.
 2010-03-10 11:21:19 +01:00
Olivier Crête
a6dfe96169 rtpsession: Make it possible to favor new sources in case of SSRC conflict 
Add a "favor-new" property that tells the session to favor new sources when
there is a SSRC conflict. This is useful for SIP calls and other such cases
where a remote loop is extremely unlikely.

Fixes #607615
 2010-03-10 11:21:19 +01:00
Olivier Crête
f336ea283f rtpsession: Move SSRC conflicts lists into RTPSource 
We will also need to track SSRC conflicts in remote sources.

See #607615
 2010-03-10 11:21:18 +01:00
Wim Taymans
38f2b4735d rtspsrc: send keep alive when paused 
When we are paused, send keep alive messages to the server so that our session
doesn't time out when we go back to playing later.
 2010-03-10 11:21:18 +01:00
Tim-Philipp Müller
8599aff951 Automatic update of common submodule 
From 7aa65b5 to 7cc5eb4
 2010-03-10 01:10:07 +00:00
David Schleef
acb6ebbc9a multifilesink: Add key-frame option to next-file 
This allows segmenting of MPEG-TS files at key frames, which is
exactly what is needed for Apple's HTTP streaming.
 2010-03-09 13:52:30 -08:00
Sebastian Dröge
e31922913f Automatic update of common submodule 
From 44ecce7 to 7aa65b5
 2010-03-09 21:32:47 +00:00
Sebastian Dröge
42ff597f41 videobox: Fix autocropping for odd width/height differences 2010-03-09 21:03:18 +00:00
Sebastian Dröge
1819fcb4a5 videobox: Use libgstvideo for format specific stuff 2010-03-09 21:03:18 +00:00
Sebastian Dröge
79e720052a audiofx: Sync properties to the stream time 2010-03-09 21:03:18 +00:00
Sebastian Dröge
4381b9296f videobox: Make properties controllable 2010-03-09 21:03:18 +00:00
Sebastian Dröge
7dcb294b54 videobox: Some cleanup 2010-03-09 21:03:18 +00:00
Sebastian Dröge
f8b7308c21 effectv: Use controller where possible, optimize a bit and make properties threadsafe 2010-03-09 21:03:18 +00:00
Sebastian Dröge
ca6566ea66 build: Make some more rules silent if requested 2010-03-09 21:03:18 +00:00
Sebastian Dröge
3e2e82afef configure: Use automake 1.11 silent rules instead of shave if available 
This makes sure that we use something that is still maintained and
also brings back libtool 1.5 support.
 2010-03-09 21:03:18 +00:00
Benjamin Otte
2a5f368bc7 png: fractions don't allow doubles 2010-03-09 19:15:07 +01:00
Benjamin Otte
b5a3b9cb1c flx: fix description 
It's video, not audio
 2010-03-09 19:15:07 +01:00
Tim-Philipp Müller
5d07a2c88f Back to development 2010-03-09 17:45:27 +00:00
Tim-Philipp Müller
e39295317b Release 0.10.21 2010-03-09 00:28:16 +00:00
Tim-Philipp Müller
e75942876b Update .po files 2010-03-09 00:24:45 +00:00
Tim-Philipp Müller
8127670c86 Revert "Add 4:2:2, 4:1:1, and 4:4:4 output support" 
This reverts commit 637c26f61a.
 2010-03-09 00:09:34 +00:00
Tim-Philipp Müller
01b6861527 Release 0.10.20 2010-03-08 23:42:51 +00:00
Tim-Philipp Müller
e83ecaa368 Update .po files 2010-03-08 23:42:06 +00:00
Tim-Philipp Müller
de56a874ae flacdec: don't send second newsegment event in framed mode, fixes long playback delay 
Don't send another newsegment event if the upstream muxer/parser has already
sent one (otherwise the sink will wait for $duration before starting playback).
Fixes long delay until playback starts with flac-in-ogg files.

Fixes #610959.
 2010-03-08 17:10:31 +00:00
Wim Taymans
66709a7a68 rtspsrc: configure multicast correctly 
Take the transport destination for multicast.
Disable loop and autojoin for multicast on the udpsinks.
 2010-03-08 17:48:46 +01:00
Wim Taymans
83a0c73dc0 multicast: always configure loop and ttl 
Also configure TTL and loop parameters when we add a client after initializing
the sender.
 2010-03-08 17:48:37 +01:00
Wim Taymans
cabe01ef95 Revert "rtph263depay: baseclass handles timestamps for us" 
This reverts commit 564581e1b8.

If we don't call push_ts, there will be no timestamp at all on the outgoing
buffer.

Fixes #612154
 2010-03-08 17:48:27 +01:00
Benjamin M. Schwartz
637c26f61a Add 4:2:2, 4:1:1, and 4:4:4 output support 2010-03-08 17:48:11 +01:00
Wim Taymans
529f443a61 rtpsource: use payload size to estimate bitrate 
Use the length of the payload for estimating the receiver bitrate so that it
matches the calculations done on the sender side. Together with the number of
packets one can scale the bitrate with the header overhead of the lower
transport.
 2010-03-08 17:48:04 +01:00
Wim Taymans
c971d1a9ab rtpsource: refactor bitrate estimation 
Don't reuse the same variable we need for stats for the bitrate estimation
because we're updating it.
Refactor the bitrate estimation code so that both sender and receivers use the
same code path.
 2010-03-08 17:48:00 +01:00
Tristan Matthews
a0a6d4ff3b added bitrate estimation to receiver-side stats, fixes #611213 2010-03-08 17:47:55 +01:00
Wim Taymans
968c981e74 h263pay: fix typo in debug 2010-03-08 17:47:14 +01:00
Tim-Philipp Müller
4379378239 Release 0.10.19 2010-03-06 00:43:03 +00:00
Tim-Philipp Müller
bcad939757 Update .po files 2010-03-06 00:42:09 +00:00
Tim-Philipp Müller
552d6b703d 0.18.4 pre-release 2010-03-03 20:29:30 +00:00
Edward Hervey
869ff4263f matroskademux: Make sure we don't send invalid newsegments 
Fixes #611501
 2010-03-02 21:20:45 +01:00
Edward Hervey
be186bd089 matroskademux: Mark streams as being EOS at the right time. 
This allows us to stop streaming only when all streams have gone past the
segment.stop and not before.

Fixes #611501
 2010-03-02 21:20:31 +01:00
Sebastian Dröge
ad71d43f52 matroskademux: Advance sparse streams only as much as required to keep the gap smaller than 500ms 
Changing it to the newest timestamp that was ever pushed will
increase the segment start in 500ms jumps, which could be just
after the next sparse stream buffer. E.g.

Video at 1.0s, sparse stream at 0.5s would jump the
sparse stream to 1.0s. Now a new sparse stream buffer could
appear that has a timestamp of 0.9s and this would be
dropped for no good reason because of bad luck.
 2010-02-27 12:20:06 +01:00