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Author SHA1 Message Date
Sebastian Dröge
56062768af capssetter: Prevent unneeded caps copying and allocation 2013-03-25 10:12:03 +01:00
Dirk Van Haerenborgh
766c5b22ed capssetter: Pass any or filter caps upstream 
capsetter accepts anything and just forwards different caps,
as such it should return ANY caps on the sinkpad.

https://bugzilla.gnome.org/show_bug.cgi?id=693005
 2013-03-25 10:11:32 +01:00
Tim-Philipp Müller
35769f7c5d wavparse: expose CUE sheet items as tracks not chapter entries in TOC 
https://bugzilla.gnome.org/show_bug.cgi?id=677306
 2013-03-24 17:55:55 +00:00
Tim-Philipp Müller
163a7afa1a wavenc: add some example pipelines 2013-03-23 12:59:26 +00:00
Anton Belka
e808173483 wavenc: add TOC support 
https://bugzilla.gnome.org/show_bug.cgi?id=680998
 2013-03-23 12:55:08 +00:00
Matej Knopp
f29e62c131 qtdemux: make empty subtitle buffer recognition more robust 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-23 11:24:23 +00:00
David Schleef
c0443a17c4 qtmux: Fix test regression with one buffer streams 2013-03-22 15:14:15 -07:00
David Schleef
5bd2864101 qtdemux: split large raw audio samples 
In order to deal with a file that has samples that are 24 seconds
long.  Seeking still doesn't work with such files.
 2013-03-22 14:14:05 -07:00
David Schleef
364433c105 qtmux: Remove documentation for dts-method 2013-03-22 14:14:04 -07:00
David Schleef
6571e388be qtmux: deprecate dts-method property 2013-03-22 14:14:04 -07:00
David Schleef
ee56a7cb99 qtmux: Fix problems causing bad durations in file 
- Fix up out-of-order incoming DTS values.
- Fix duration of initial sample.
 2013-03-22 14:14:04 -07:00
David Schleef
816e186029 qtmux: fix all timestamps once first_ts is determined 2013-03-22 14:14:04 -07:00
David Schleef
258c40c6dd qtmux: Use PTS/DTS from incoming buffers 
Remove old DTS guessing code.
 2013-03-22 14:14:04 -07:00
Nicola Murino
709f05234f qtmux: expose mulaw caps 
https://bugzilla.gnome.org/show_bug.cgi?id=696052
 2013-03-22 20:08:06 +00:00
Rodolfo Schulz de Lima
874808fd2c qtdemux: fix sample leak when processing private qt tags 
https://bugzilla.gnome.org/show_bug.cgi?id=696355
 2013-03-22 08:47:17 +00:00
Matej Knopp
d8ac666137 qtmux: set stream language code from tag 
https://bugzilla.gnome.org/show_bug.cgi?id=696358
 2013-03-22 08:40:26 +00:00
Matej Knopp
49d9050e9a qtdemux: send GAP events for subtitle streams 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:37 +00:00
Matej Knopp
516a0b8acb qtdemux: ignore empty subtitle buffers 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:34 +00:00
Matej Knopp
f494635126 qtdemux: recognize SBTL subtype for subtitles 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:14 +00:00
Anton Belka
0f97b6f978 flacparse: add support for the toc-select event 
Select tracks from the CUE sheet by sending a toc-select
event based on the uid in the TOC.

https://bugzilla.gnome.org/show_bug.cgi?id=540891
 2013-03-21 00:38:48 +00:00
Michael Smith
b85c5f236b mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end. 2013-03-19 18:09:31 -07:00
Tim-Philipp Müller
5240b7453c sbcparse: pack multiple frames into one output buffer 
Don't output a single buffer for every tiny SBC frame
 2013-03-20 00:35:17 +00:00
Kishore Arepalli
288e05c99d deinterlace: fix infinite loop on EOS with non-default methods or fields 
Fixes problem of infinite loop in gst_deinterlace_reset_history.
Last field in the history was never deinterlaced because idx becomes negative.

Happens e.g. with method=scalerbob fields=bottom or
method=greedyl fields=top

https://bugzilla.gnome.org/show_bug.cgi?id=695644
https://bugzilla.gnome.org/show_bug.cgi?id=693173
 2013-03-17 14:47:26 +00:00
Tim-Philipp Müller
dfde4179e8 avimux: change raw video caps order so that GRAY8 is last 
People like colours.

https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-12 00:16:18 +00:00
Ognyan Tonchev
3f8ad30cee rtph264pay: Don't use upstream caps with peer_query_caps () 
Calling gst_pad_peer_query_caps () on the src pad with the caps
upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
is wrong and makes caps negotiation fail if upstream caps are not
NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=695629
 2013-03-11 16:55:13 -04:00
Dirk Van Haerenborgh
065bdf5925 avimux: support raw BGR 
https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-11 14:51:00 +01:00
Dirk Van Haerenborgh
d7743cf7c6 avidemux: support raw video with negative height 
https://bugzilla.gnome.org/show_bug.cgi?id=695541
 2013-03-11 14:23:46 +01:00
Tim-Philipp Müller
694dbcc5a0 dtmf: move dtmf plugin from -bad to -good 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 01:18:30 +00:00
Tim-Philipp Müller
a4c5aa38ec Merge branch 'dtmf-moved-from-bad' 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 00:30:38 +00:00
Sebastian Dröge
539126c097 matroska: Include config.h, it's needed for _stdint.h 2013-03-03 11:59:31 +01:00
Sebastian Dröge
1810786083 flacparse: Fix (wrong) use of uninitialized variable compiler warning 2013-03-03 11:53:04 +01:00
Tim-Philipp Müller
677bfecc6f qtdemux: add variant field to H.263 caps 
avdec_h263 won't get plugged otherwise.
 2013-03-02 13:59:52 +00:00
Arnaud Vrac
1cff6427f1 qtdemux: skip disabled tracks 
ISO/IEC 14496-12 specifies disabled tracks should be completely
ignored, so just do it.

Avoids deadlock during prerolling for some files.

Also prevents 'chapter' subtitle tracks from showing up.

https://bugzilla.gnome.org/show_bug.cgi?id=693993
https://bugzilla.gnome.org/show_bug.cgi?id=628790
 2013-03-02 13:54:23 +00:00
Stefan Sauer
15a81baea5 spectrum: remove the since doc-comment from 0.10 2013-02-28 09:43:12 +01:00
Stefan Sauer
b62cb3edcd level: add a "post-messages" property and deprecate "message" 
In spectrum this was changed from 0.10 to 1.0, lets do this here too.
 2013-02-28 09:43:12 +01:00
Olivier Crête
df5ca83baf rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional 
Specific case here is Wowza 3.5.0
 2013-02-26 14:19:10 -05:00
Thomas Vander Stichele
df8f5f2f83 level: put back deprecation warnings 2013-02-25 00:35:58 +01:00
Thomas Vander Stichele
52b7aab711 level: send last message on EOS 2013-02-25 00:19:22 +01:00
Mark Nauwelaerts
56e2767c20 avidemux: push mode: handle some more 0-size buffer cases 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944
 2013-02-24 19:28:07 +01:00
Tim-Philipp Müller
8004ae0369 matroskamux: fix up example pipeline in docs 2013-02-23 18:50:52 +00:00
Paul HENRYS
10802cae73 rtpsession: Fix wrong code organisation in case of collision 
change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
 2013-02-22 09:28:07 +02:00
Jean-François Fortin Tam
f5cb19e287 alpha: improve descriptions of chroma keying-related properties and enums 
https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:09:56 +00:00
Youness Alaoui
a65fd146f8 alpha: Do not override the method with custom r/g/b values 
Depending on the order g_object_set() calls aare made, the
target r/g/b settings will override the method if set to
green/blue. Change that so we do not use the target-r/g/b values
unless the method is set to custom.

https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:04:51 +00:00
Ognyan Tonchev
42d8b96f2d auparse: do not leak src_caps 
https://bugzilla.gnome.org/show_bug.cgi?id=694275
 2013-02-21 19:31:59 +00:00
Wim Taymans
a61055809f rtpsession: only delay RTCP when we are a sender 
Only delay the RTCP thread when we are a sender, which we can know because we
have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
are only a receiver and then there is no code path that wakes up the
RTCP thread and we end up without RTCP packets.
 2013-02-20 21:07:41 +02:00
Tim-Philipp Müller
5b19be933b qtdemux: fix up dodgy code that tries to fix up a broken moov atom 
After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely
append to the already-existing memory instead of filling it.
 2013-02-18 20:04:05 +00:00
Tim-Philipp Müller
34b81f7c93 qtdemux: fix potential crash on short MOOV atom 
Don't unmap short MOOV atom buffer twice, which happened
in the case where we don't fix up the MOOV atom.

Fixes crashes when thumbnailing partial mp4 file where
the MOOV atom is still incomplete.

https://bugzilla.gnome.org/show_bug.cgi?id=694010
 2013-02-18 16:35:08 +00:00
Stefan Sauer
99f84b8c4c audiopanorama: remove channel-mask from caps 
The channel-mask is only needed for channels>2 which we don't do.
 2013-02-15 21:30:15 +01:00
Tim-Philipp Müller
01c6512d5f udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions 
So we have to worry less about portability.

https://bugzilla.gnome.org/show_bug.cgi?id=692400
 2013-02-15 14:11:36 +00:00
Sebastian Dröge
a7ddbc03fe rtp-payloading: Fix unit test caps and AMR depayloader sink template caps 
Fields were missing from the actual caps, or too many fields
existed in the template caps.
 2013-02-13 12:02:46 +01:00
Michael Smith
e3430b0d07 qtdemux: extract codec_data for ProRes 2013-02-12 13:19:53 -08:00
Tim 'mithro' Ansell
c499a81848 avimux: Fixing buffer leak in gst_avi_mux_do_buffer 
gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.
 2013-02-12 10:09:05 +01:00
Mark Nauwelaerts
bf81dce432 avidemux: correct duration for audio VBR buffers in pull mode 2013-02-10 15:10:32 +01:00
Mark Nauwelaerts
f0645b79c5 avidemux: proper position reporting and push mode timestamping 
... and align current_total semantics in push and pull mode,
which tracks bytes for CBR and blocks for VBR.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481
 2013-02-08 21:41:55 +01:00
Wim Taymans
2d5319c1fa rtpsession: delay RTCP until first RTP packet 
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
 2013-02-08 17:05:27 +01:00
Wim Taymans
2971ed44ee rtpsession: remove dead code 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
 2013-02-07 15:06:40 +01:00
Paul HENRYS
0e91c949d8 rtpptdemux: forward sticky events and then set caps 
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
 2013-02-07 14:38:20 +01:00
Markovtsev Vadim
7cebe2fc41 rtpjitterbuffer: improve debug output 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935
 2013-02-07 14:32:26 +01:00
Wim Taymans
978cc9f538 rtpbin: rework cleanup of streams 
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.

Based on patch by Sujay <sdatar@cisco.com>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
 2013-02-07 13:02:34 +01:00
Tim 'mithro' Ansell
3a5d17e852 videomixer2: avoid caps leak 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307
 2013-02-07 11:40:35 +01:00
Wim Taymans
c3077012c0 jitterbuffer: do skew estimation only for new timestamps 
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
 2013-02-06 17:15:11 +01:00
Wim Taymans
640de61740 rtspsrc: only EOS when our source sends BYE 
Only EOS when we receive a BYE event from the SSRC of our stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
 2013-02-06 14:01:16 +01:00
Wim Taymans
0540492ab2 rtspsrc: save the stream SSRC 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2013-02-06 14:00:56 +01:00
Wim Taymans
c8fb1c720c rtspsrc: flush connection when stopping 
When we stop, we can flush all pending commands so that we can stop and
join the task.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
 2013-02-06 13:18:18 +01:00
Stefan Sauer
96f8775a0d spectrum: remove outdates readme 
Lets remove the readme from pre-0.1.0 that is completely irrelevant now.
 2013-02-05 22:02:13 +01:00
Stefan Sauer
86ae581928 audiopanorama: add more debug logging 2013-02-05 18:51:27 +01:00
Rico Tzschichholz
682e49a752 audioparsers: fix typo in noinst_headers 2013-02-04 18:38:41 +00:00
Stefan Sauer
1f1fe47cb6 audiopanorama: further port to 1.0 
Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.
 2013-02-04 11:08:23 +01:00
Stefan Sauer
d187b96ee2 audiopanorama: fix caps 
We don't turn float into 32bit pcm. Looks like a typo from updating the caps.
 2013-02-03 22:45:52 +01:00
Olivier Crête
fe3e535853 level: Add missing coma between formats 2013-02-03 13:14:50 +01:00
Matthew Waters
b9151a9c28 videomixer: fix eos timestamp check 
fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935
 2013-01-31 16:45:38 +01:00
Dirk Van Haerenborgh
18ff57d6b3 avimux: add support for raw monochrome 8-bit video 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932
 2013-01-31 13:00:17 +01:00
Wim Taymans
747447d298 rtpsession: avoid '...is used uninitialized' 2013-01-29 10:32:51 +01:00
Youness Alaoui
f6a00ad6e9 qtdemux: set interleaved layout correctly for LPCM audio 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:44:01 +00:00
Youness Alaoui
a76524ea08 qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7) 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:57 +00:00
Youness Alaoui
69b814546a qtdemux: print all debug for sound sample description v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:49 +00:00
Youness Alaoui
92ff8a9b09 qtdemux: sound sample description v2 doesn't override samples_per_packet 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:42 +00:00
Youness Alaoui
ee3d9cbd98 qtdemux: pass stsd data to qtdemux_audio_caps() 
We will need that later for LPCM format support. Disable
QDM2 parsing of stsd data which dead code before as well
because data was always NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:38 +00:00
Youness Alaoui
6d3ff78575 qtdemux: add len check for sound sample descriptions v1 and v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:28 +00:00
Tim-Philipp Müller
629772f735 rtpmanager: use C89-style comments 2013-01-28 23:07:34 +00:00
Olivier Crête
451217c437 gstrtpsession: Fix double-declared variable 2013-01-28 18:06:15 -05:00
Olivier Crête
7300d489fe rtp: Fix compilation errors in previous patches 2013-01-28 17:58:20 -05:00
Haakon Sporsheim
86c13ceae6 rtpsession: Ensure MT safe event handling and plug event leak. 
https://bugzilla.gnome.org/show_bug.cgi?id=667826
 2013-01-28 17:44:31 -05:00
Idar Tollefsen
268c998a32 rtpsession: mt-safe event-push 
By taking a ref of the sink-pad under lock, it won't dissappear
while the push is taking place

https://bugzilla.gnome.org/show_bug.cgi?id=667816
 2013-01-28 17:34:50 -05:00
Pascal Buhler
f459fe2673 rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE 
https://bugzilla.gnome.org/show_bug.cgi?id=667815
 2013-01-28 17:01:27 -05:00
Tim-Philipp Müller
721dd1ab26 sbcparse: init some variables to avoid bogus compiler warnings 2013-01-28 11:58:50 +00:00
Wim Taymans
4397c8ffbf rtpdepay: remove payload type restrictions 
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:41:04 +01:00
Marc Leeman
bab2f3c92b rtp: remove payload requirements from selected depayloaders 
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.

In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:23:41 +01:00
Mark Nauwelaerts
a1a579afeb qtdemux: push mode: only parse moov 1 once 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570
 2013-01-27 12:54:20 +01:00
Tim-Philipp Müller
47fccbe635 rtpdtmfsrc: fix compiler warning 
gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function
 2013-01-26 22:58:29 +00:00
Olivier Crête
db5c3f4048 rtpdtmfdepay: Fix missing work in doc 2013-01-25 21:06:05 -05:00
Olivier Crête
92f9a9d9ff rtpdtmfsrc: Post the messages after the clock wait 
This way, the messages will be closer in time to when the packets are sent out
 2013-01-25 20:45:43 -05:00
Olivier Crête
0d316b4f43 rtpdtmfsrc: Only set the duration when starting to send 
The duration depends on the clock rate, which could change due to renegotiation
 2013-01-25 20:45:43 -05:00
Olivier Crête
90497aa3cd rtpdtmfsrc: remove "ssrc" from caps 
ssrc is uint and we don't have a uint range type
 2013-01-25 20:45:43 -05:00
Tim-Philipp Müller
d62019fff2 qtmux: set language to 'undefined' instead of English by default 2013-01-24 21:08:51 +00:00
Mark Nauwelaerts
0777a600e3 audioparsers: sbc: fix bogus compiler warning 
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
 2013-01-22 19:26:09 +01:00
Thijs Vermeir
16128f0234 autoparsers: use appropriate printf format for gsize 2013-01-16 14:32:56 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name 
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
 2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
9f7a949773 audioparsers: add SBC audio parser 
From-scratch rewrite, the bluez one was useless and broken.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0 
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-10 12:43:50 +00:00
Olivier Crête
c6dea5d09c dtmf/spandsp: Move dtmfdetect to use libspandsp 
Remove our copy of the tone_detect.c file and use the original
from libspandsp. Also move the element to the spandsp plugin.
 2013-01-09 20:05:16 -05:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574 jitterbuffer: refactor handle sync code 
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
 2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf rtp: include downstream latency in SR calculations 
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
 2013-01-07 15:45:10 +01:00
Wim Taymans
c631ed3300 rtpsession: don't cast event functions 
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
 2013-01-07 14:25:14 +01:00
Wim Taymans
8dcde8b3ea rtp: more debug 2013-01-07 14:23:34 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
cf1f6aff0d udpsrc: sanity check size of available packet data for reading to avoid memory waste 
On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.

https://bugzilla.gnome.org/show_bug.cgi?id=610364
 2013-01-04 14:00:55 +00:00
Tim-Philipp Müller
95a37196b3 rtspsrc: add "proxy-id" and "proxy-pw" properties 
to match souphttpsrc. user/password passed via the URI
will still take precedence though.

https://bugzilla.gnome.org/show_bug.cgi?id=395427
 2012-12-31 00:22:27 +00:00
Wim Taymans
8cfec6a88d rtspsrc: fix cmd comparison 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476
 2012-12-20 17:12:30 +01:00
Wim Taymans
75616fac9a rtspsrc: add some more debug 2012-12-20 17:12:20 +01:00
Jonas Holmberg
e12457f138 rtpjpegpay: handle width and height > 2040 
If width or height is greater than 2040 set width and height to zero in
the rtp header and add x-dimensions to outcaps.

Solves #684955
 2012-12-20 15:40:49 +01:00
Wim Taymans
dcb0e0af93 avidemux: cleanup in flag define 2012-12-20 13:04:52 +01:00
Wim Taymans
0e522bc69a avidemux: improve debug 2012-12-20 13:04:52 +01:00
Thijs Vermeir
de41376231 rtp: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Thijs Vermeir
df88341ffb deinterlace: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Philippe Normand
2bd77e1c8a interleave: set src pad caps upon last sink pad CAPS event 
Gather caps on all sink pads before setting the src pad caps. This is
specially needed when the audio channel mapping is set on the sink
pads and the element needs to preserve it on its src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=690267
 2012-12-18 12:58:43 +01:00
Tim-Philipp Müller
f4cb0c4315 matroskademux: skip empty tags 
instead of trying to add tags with empty strings, which
causes criticals at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=690358
 2012-12-17 22:55:12 +00:00
Sebastian Dröge
c49dede772 audioparsers: Make sure the caps are actually writable before changing them 2012-12-17 15:17:12 +01:00
Sebastian Dröge
26040ee38c audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps 
Otherwise we will intersect with the srcpad template caps and add all the caps fields
that the parser will ever set, no matter if downstream restricts this field or not.
This requires upstream to set this field on the caps to successfully negotiate.

https://bugzilla.gnome.org/show_bug.cgi?id=690184
 2012-12-17 15:01:02 +01:00
Alexey Fisher
7e47e3b92d matroskamux: set appropriate block header flag for VP8 invisible frames 
Useful for debugging mostly.

https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-12-16 23:30:13 +00:00
Tim-Philipp Müller
8a3b116d1f docs: add rtpmux and rtpdtmfmux to plugin docs 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
3295b5d791 rtpmanager: move rtpmux and rtpdtmfmux elements from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
de204ba754 rtpmux: Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
2778a1757f rtpmux: Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-16 16:36:39 +00:00
Olivier Crête
15dfdc58d4 rtpmux: Misc fix for 0.11 
Convert the incoming caps before proxying them
Clear the last_pad when going to ready

tests: Implement accept_caps, don't leak event
 2012-12-16 16:36:38 +00:00
Wim Taymans
83262be703 rtpmux: update for RTP buffer api changes 2012-12-16 16:36:38 +00:00
Sebastian Dröge
f17064a8ea rtpmux: Update for GST_PLUGIN_DEFINE() API changes 2012-12-16 16:36:34 +00:00
Wim Taymans
c86156ad8f rtpmux: fix compilation 2012-12-16 16:35:36 +00:00
Wim Taymans
6826bbb6da rtpmux: fix for caps api changes 2012-12-16 16:35:33 +00:00
Matej Knopp
bb345a584d rtpmux: Fix compiler warnings 2012-12-16 16:35:29 +00:00
Olivier Crête
af4e999c59 rtpmux: Unref non-forwarded events 
Also, don't unref forwarded ones
 2012-12-16 16:35:29 +00:00
Olivier Crête
a8789d1df1 rtpmux: resync iterator on resync 2012-12-16 16:35:29 +00:00
Olivier Crête
0c54079af5 rtpmux: Re-push sticky events on input pad change 2012-12-16 16:35:29 +00:00
Olivier Crête
21831b430f rtpmux: Don't leak gvalue from iterator 2012-12-16 16:35:29 +00:00
Wim Taymans
ccc4b960fc rtpmux: more porting 2012-12-16 16:35:26 +00:00
Olivier Crête
f20a6b1d16 rtpmux: port to 0.11 2012-12-16 16:35:26 +00:00
Wim Taymans
35b6668fb6 rtpmux: make request pads take _%u 2012-12-16 16:35:22 +00:00
Olivier Crête
aa3607ef5c rtpdtmfmux: Add last-stop to dtmf-event upstream events 
Add the running time of the last outputted buffer to the
upstream "dtmf-event" events so that the dtmf source does not
leave a gap.
 2012-12-16 16:35:22 +00:00
Edward Hervey
d137482fe5 rtpmux: Remove dead assignments 2012-12-16 16:35:22 +00:00
Stefan Kost
55aae6bfab rtpmux: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-16 16:35:15 +00:00
Olivier Crête
9674d5cc23 rtpmux: Improve documentation 
Add an example pipeline, and try to explain a bit more what it does.
 2012-12-16 16:35:15 +00:00
Stefan Kost
ca27a279ba rtpdtmfmux: remove unused variable 2012-12-16 16:35:15 +00:00
Stefan Kost
c85dceeacb rtpdtmfmux: remove unused signal boilerplate 2012-12-16 16:35:15 +00:00
Stefan Kost
2353f8d852 rtpmux: no need to ref pad in _chain() 2012-12-16 16:35:15 +00:00
Youness Alaoui
e42d2eebcb rtpmux: Unlock the right mutex 
The mutex locked is for the 'mux' object, but we unlock the
pad, which means that if the rtpmux gets a flush, then the
object lock will stay locked forever, causing it to freeze
the next time it tries to take it.

Fixes bug #627991
 2012-12-16 16:35:15 +00:00
Olivier Crête
78d1ebac9e rtpmux: Add support for GstBufferList 
Factor out most of the buffer handling and implement a chain_list
function. Also, the DTMF muxer has been modified to just have a
function to accept or reject a buffer instead of having to subclass
both chain and chain_list.
 2012-12-16 16:35:15 +00:00
Olivier Crête
c00f14419b rtpmux: Don't leak invalid buffers 2012-12-16 16:35:15 +00:00
Tim-Philipp Müller
a45429d81d rtpmux: fix missing debug log message argument 2012-12-16 16:35:15 +00:00
Olivier Crête
4a8d0243b5 rtpdtmfmux: Add some debug messages 2012-12-16 16:35:14 +00:00
Olivier Crête
423ce98666 rtpdtmfmux: Remove stream-lock event handling 2012-12-16 16:35:14 +00:00
Olivier Crête
a4500c0e74 rtpdtmfmux: Update doc for simplification 2012-12-16 16:35:14 +00:00
Olivier Crête
70097866de rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink 2012-12-16 16:35:14 +00:00
Olivier Crête
f6548fe9b6 rtpdtmfmux: Add priority sink pads 2012-12-16 16:35:14 +00:00
Olivier Crête
2bcea1537b rtpdtmfmux: Cleanup event function 2012-12-16 16:35:14 +00:00
Olivier Crête
8e58646f5c rtpmux: Aggregate incoming segments 2012-12-16 16:35:14 +00:00
Olivier Crête
7be57cac3a rtpdtmfmux: Update documentation 2012-12-16 16:35:14 +00:00
Olivier Crête
e590fc1f32 rtpmux: Simplify request pad creation 2012-12-16 16:35:14 +00:00
Benjamin Otte
2867e00225 rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-16 16:35:10 +00:00
unknown
fb7266884d rtpmux: update the current_ssrc from the caps 
Fixes #604101
 2012-12-16 16:33:47 +00:00
Håvard Graff
eab65e84ca rtpmux: release pads when disposing 
Because of an allocated priv (GstRTPMuxPadPrivate), the element will
leak memory if not gst_rtp_mux_release_pad() is called. This would
previously only happen if release_request_pad() was called explicitly,
somthing that should not be neccesary.

Fixes #604099
 2012-12-16 16:33:46 +00:00
Wim Taymans
0d54122804 dtmfmux: method name cleanups 2012-12-16 16:33:46 +00:00
Olivier Crête
3841cc74cf rtpmux: Don't ignore requested pad name 2012-12-16 16:33:46 +00:00
Olivier Crête
d93295ff9d rtpmux: Remove empty finalize 2012-12-16 16:33:46 +00:00
Olivier Crête
5e90a4e86b rtpmux: Free the pad private data on pad release 
Free the pad private data on pad release instead of using a weak ref,
which is not thread safe. Also, lock the content of the pad private using the element's
object lock.
 2012-12-16 16:33:46 +00:00
Olivier Crête
4be63c9add rtpmux: Reject wrong caps 2012-12-16 16:33:46 +00:00
Olivier Crête
0111bafb3a rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr> 2012-12-16 16:33:46 +00:00
Olivier Crête
fcc1522d2e rtpmux: Fix leak 
Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>
 2012-12-16 16:33:46 +00:00
Olivier Crête
ff6686f1c7 rtpmux: Fix warning 2012-12-16 16:33:46 +00:00
Olivier Crête
00791f930b rtpmux: Set different caps depending on the input 2012-12-16 16:33:46 +00:00
Olivier Crête
ed0b407038 rtpmux: Only free pad private when pad is disposed 2012-12-16 16:33:45 +00:00
Olivier Crête
92bb5199ac rtpmux: Remove useless caps mangling 2012-12-16 16:33:45 +00:00
Olivier Crête
3ccf3217fe rtpmux: Rename variable for more clarity 2012-12-16 16:33:45 +00:00
Olivier Crête
4b958f6d8d rtpmux: Use GST_BOILERPLATE 2012-12-16 16:33:45 +00:00
Olivier Crête
abe57be248 rtpmux: Do the includes locally 2012-12-16 16:33:45 +00:00
Olivier Crête
05844c89e9 rtpmux: Add GST_DEBUG_FUNCPTRs 2012-12-16 16:33:45 +00:00
Olivier Crête
fd102b95ab rtpdtmfmux: Release locked pad on release_pad 
Release the special pad if the pad is removed from the muxer.
 2012-12-16 16:33:45 +00:00
Laurent Glayal
00f8bab712 rtpdtmfmux: Release special on pad dispose 
Fixes #577690
 2012-12-16 16:33:45 +00:00
Stefan Kost
a4a22454dc docs: various doc fixes 
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
 2012-12-16 16:33:41 +00:00
Olivier Crête
7d4395a910 rtpmux: Move rtpmux from gst-plugins-farsight to -bad 2012-12-16 16:33:27 +00:00
Olivier Crête
68215752f4 rtpmux: Re-indent to Gst style 2012-12-16 16:33:24 +00:00
Olivier Crête
c7d0809434 rtpmux: Document rtp muxer a bit 2012-12-16 16:33:20 +00:00
Laurent Glayal
47c7a93df2 rtpmux: Add signals before stream lock and after unlocking 2012-12-16 16:33:17 +00:00
Olivier Crête
f1656ed8b0 rtpmux: Let ssrc through getcaps 2012-12-16 16:33:14 +00:00
Olivier Crête
1529dffaf9 rtpmux: Rename have_base to have_ts_base 2012-12-16 16:33:11 +00:00
Olivier Crête
57563517bd rtpmux: Protect the seqnum with object lock in rtpmux 2012-12-16 16:33:08 +00:00
Olivier Crête
d3237eaf95 rtpmux: Remove unused sink_ts_base 2012-12-16 16:33:04 +00:00
Olivier Crête
cc23958183 rtpmux: Have getcaps to force the same clockrate on all pads 2012-12-16 16:33:01 +00:00
Olivier Crête
dc36590d0c rtpmux: Validate RTP data in RTP Mux 2012-12-16 16:32:57 +00:00
Olivier Crête
360c8d4f1d rtpmux: Remove unused clock-rate property 2012-12-16 16:32:54 +00:00
Olivier Crête
b86232d0dc rtpmux: Clarify locking in rtpdtmfmux 2012-12-16 16:32:50 +00:00
Laurent Glayal
4b607cdda5 rtpmux: Missing format parameter 2012-12-16 16:32:47 +00:00
Håvard Graff
b313c80367 rtpmux: Update seqnum base in rtp muxer 
With help from Wim
 2012-12-16 16:32:43 +00:00
Håvard Graff
c479f90274 rtpmux: Fix some more leaks 2012-12-16 16:32:40 +00:00
Håvard Graff
1b5e769e0b rtpmux: Fix leak 2012-12-16 16:32:37 +00:00
Olivier Crête
5cbb0de823 rtpmux: Don't unref caps we don't know (thanks Wim) 2012-12-16 16:32:32 +00:00
Olivier Crête
cebf506949 rtpmux: Put per-buffer debug at level LOG 2012-12-16 16:32:29 +00:00
Olivier Crête
3c12a423b7 rtpmux: Make debug print accurate 2012-12-16 16:32:25 +00:00
Olivier Crête
c49f4c87c6 rtpmux: Set our caps on the buffers 2012-12-16 16:32:22 +00:00
Olivier Crête
ec63da9366 rtpmux: Take the clock-base stored from the last setcaps 2012-12-16 16:32:18 +00:00
Olivier Crête
674c074114 rtpmux: Store the clock-base on setcaps 2012-12-16 16:32:15 +00:00
Olivier Crête
90264b9686 rtpmux: Add padprivate to the request pads 2012-12-16 16:32:11 +00:00
Olivier Crête
15d661ba3e rtpmux: Make indentation more correct 2012-12-16 16:31:56 +00:00
Olivier Crête
3a7d09a749 rtpmux: Fix typo 2012-12-16 16:31:53 +00:00
Olivier Crête
91aef3ec5e rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer 2012-12-16 16:31:50 +00:00
Zeeshan Ali
6ea5ca354d rtpmux: more debug 
20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz
 2012-12-16 16:31:46 +00:00
Youness Alaoui
f0e209b638 rtpmux: missing comment 
20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz
 2012-12-16 16:30:33 +00:00
Olivier Crete
3ed5590da6 rtpmux: Make buffer writable before writing into it 
20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz
 2012-12-16 16:30:31 +00:00
Olivier Crete
dd13f7c8ef rtpmux: Set pads active when adding them to a potentially running element 
20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz
 2012-12-16 16:30:27 +00:00
Olivier Crete
1c5075f927 rtpmux: Fix multiple ref leaks (patches by SP GLE) 
20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz
 2012-12-16 16:30:23 +00:00
Zeeshan Ali
42f455e902 rtpmux: send event to all src pads 
20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz
 2012-12-16 16:30:18 +00:00
Zeeshan Ali
dba101bb0f rtpmux: print a warning if receive an error iterating sinkpads 
20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz
 2012-12-16 16:30:15 +00:00
Zeeshan Ali
baa48dc6bc rtpmux: deal with all the gst_iterator_next() return values 
20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz
 2012-12-16 16:30:12 +00:00
Zeeshan Ali
de40874670 rtpmux: Return correct value from the event handler 
20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz
 2012-12-16 16:30:08 +00:00
Zeeshan Ali
ed76f67e96 rtpmux: Ville's original patch to fix the traversal of dtmf event 
20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz
 2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com
94ebe07862 rtpmux: Set the correct ts-offset on the get_prop value 
20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz
 2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com
1ee542c378 rtpmux: Refactorize state_change 
20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz
 2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com
2498ba671a rtpmux: set SSRC on the packets 
20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz
 2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com
ee69c2690d rtpmux: Code clean-up and more debug output 
20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz
 2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com
1c799ce964 rtpmux: Use own clock-base 
20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz
 2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com
b04630d7a2 rtpmux: Only accept RTP streams that have the same clock-rate 
20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz
 2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com
6fe1e02efd rtpmux: Some more code-cleanups 
20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz
 2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com
1603223ee5 rtpmux: return newpad instead of NULL and warn if failed to create a pad 
20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz
 2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com
23d3ed5c5f rtpmux: Refactorize the RTPMux code 
20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz
 2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com
21e6e951f6 rtpmux: Some more doc fixing 
20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz
 2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com
0de7fb6f37 rtpmux: More Refactoring 
20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz
 2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com
0f755657ce rtpmux: More documentation 
20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz
 2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com
5483c78ac0 rtpmux: Refactor the event handler function 
20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz
 2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com
db1523ae60 rtpmux: Add RTPDTMFMux element 
20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz
 2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com
97ff54dce7 rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 
20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz
 2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com
1a227ac7e5 rtpmux: Put more helpful description 
20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz
 2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com
d876c0d8cc rtpmux: remove the (commented-out) code for blocking the pads 
20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz
 2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com
209228c44d rtpmux: Drop buffers instead of blocking the sinkpads 
20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz
 2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com
795822ffa5 rtpmux: Implement stream locking, needed for DTMF 
20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz
 2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com
fd209faa56 rtpmux: use GST_*_OBJECT instead of g_* 
20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz
 2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com
b0208cb0a6 rtpmux: No need to manage pads, parent does that for us 
20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz
 2012-12-16 16:28:55 +00:00
zeenix@gmail.com
74e9071dad rtpmux: Fix copyright header 
20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz
 2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com
3c4cdf1541 rtpmux: The first implementation of RTP muxer 
20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz
 2012-12-16 16:28:41 +00:00
Tim-Philipp Müller
b19122bac8 scaletempo: no need for a private struct 2012-12-15 21:27:01 +00:00
Tim-Philipp Müller
61913ab7b4 audiofx: move scaletempo element from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=687262
 2012-12-14 13:16:17 +00:00
Sebastian Dröge
314765c294 scaletempo: Fix event leak 2012-12-14 13:16:17 +00:00
Sebastian Dröge
490e408991 scaletempo: Fix timestamp tracking 2012-12-14 13:16:17 +00:00
Sebastian Dröge
502eb8d1b7 scaletempo: Implement LATENCY query 2012-12-14 13:16:17 +00:00
Sebastian Dröge
c7589817cb scaletempo: Store instance private data in the instance struct 
Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
is really slow.
 2012-12-14 13:16:17 +00:00
Tim-Philipp Müller
e552bd484f scaletempo: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
d2dd91ac47 scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-12-14 13:16:17 +00:00
Wim Taymans
cb1743d578 scaletempo: ffmpegcolorspace is no more 2012-12-14 13:16:17 +00:00
Sebastian Dröge
93e1091d7f scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
3286cdd542 scaletempo: port to 0.11 2012-12-14 13:16:16 +00:00
Stefan Kost
62d780cd51 scaletempo: improve the docs 
Fix the syntax, add more explanation and xref the properties.
 2012-12-14 13:16:16 +00:00
Chris E Jones
caf2b6cb5c scaletempo: Correctly handle newsegment events with stop==-1 
Fixes bug #645420.
 2012-12-14 13:16:16 +00:00
Stefan Kost
6d54058982 scaletempo: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-14 13:16:16 +00:00
Benjamin Otte
38bc2dfb4a scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-14 13:16:16 +00:00
Thiago Santos
2d72ec153a scaletempo: properly update new segments 
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.

Fixes #599903

Based on patch by: Bastian Hecht <hechtb@gmail.com>
 2012-12-14 13:16:16 +00:00
Maximilian Högner
2fe7a97f1c scaletempo: Explicitely cast to signed integers to fix a segfault 
Fixes bug #585660.
 2012-12-14 13:16:16 +00:00
Michael Smith
1b1f6f56d6 scaletempo: Do not use void pointer arithmetic. 2012-12-14 13:16:16 +00:00
Stefan Kost
9284c85b33 scaletempo: Return the result of parent_class->event() 
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
 2012-12-14 13:16:16 +00:00
Rov Juvano
43e79f7769 Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... 
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/audiofx/Makefile.am:
* gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/audiofx/gstscaletempo.h:
* gst/audiofx/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
 2012-12-14 13:16:15 +00:00
Havard Graff
9c94f1187c jitterbuffer: bundle together late lost-events 
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.

Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.

So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...

The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.

See https://bugzilla.gnome.org/show_bug.cgi?id=667838
 2012-12-13 12:00:43 +01:00
Wim Taymans
a858bf46db rtspsrc: fix TCP reconnect 
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
 2012-12-13 09:30:59 +01:00
Philippe Normand
a8fa9f2b47 deinterleave: properly set srcpad channel position 
The src pad caps always describe a single audio channel so only the
first position matters if deinterleave is configured to keep channel
positions in its src pads.
 2012-12-12 11:20:56 +00:00
Wim Taymans
b1dc816772 rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:09:42 +01:00
Wim Taymans
32bd981303 udpsrc: improve timeouts 
Make it possible to set the timeout after we went to the READY state by using
the timeout when checking the condition. This also makes it possible to set the
timeout with a higher granularity than seconds.
 2012-12-12 11:08:13 +01:00
Wim Taymans
abd7e33db6 deinterlace: add support for strides 
Implement stride support correctly by taking it from the GstVideoFrame.
Propose a bufferpool upstream when not operating in passthrough.
 2012-12-11 13:00:46 +01:00
Aleix Conchillo Flaque
3503aef946 rtspsrc: do not change state to PLAYING if currently chaning state 
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
  happening in the application thread, so we don't change the state to
  PLAYING in the gstrtspsrc thread unless it is safe.

  A specific case is when chaning the state to NULL from the application
  thread. This will synchronously try to stop the task (with the element
  state lock acquired), but we will try a gst_element_set_state from
  gstrtspsrc thread which will block on the element state lock causing a
  deadlock.

  https://bugzilla.gnome.org/show_bug.cgi?id=684312
 2012-12-10 15:13:22 +01:00
Tim-Philipp Müller
672ab8fb5b webmux: fix linking with shout2send element 
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.

Also add unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=689336
 2012-11-30 17:22:34 +00:00
Wim Taymans
64cdbb77a9 rtspsrc: use new option parser function 2012-11-27 11:13:37 +01:00
Tim-Philipp Müller
5dee61a8d5 law: fix accidental file permissions change 
https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-26 15:17:13 +00:00
Tim-Philipp Müller
314efb684b qtdemux: avoid criticals if unknown fourcc has space at beginning or end 
https://bugzilla.gnome.org/show_bug.cgi?id=682936
 2012-11-25 14:16:09 +00:00
Tim-Philipp Müller
efaa80fbc6 videobox: fix border filling for planar YUV formats 
We would get a green border instead of a black one, for
example.

https://bugzilla.gnome.org/show_bug.cgi?id=684991
 2012-11-24 19:32:51 +00:00
Tim-Philipp Müller
ef6c16a32e mulaw: const-ify some arrays 2012-11-24 14:27:33 +00:00
Roland Krikava
3be45f7022 mulawdec: fix integer overrun 
There might be more than 65535 samples in a chunk of data.

https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-24 14:24:41 +00:00
Wim Taymans
5d0507c09e rtspsrc: pause the task instead of spinning 
Actually pause the loop task instead of spinning forever.
 2012-11-22 11:34:31 +01:00
Joshua M. Doe
fe9fb8d8a7 videoflip: Add gray 8/16 support 2012-11-20 12:49:49 +01:00
Wim Taymans
c28bfa8902 rtspsrc: handle segment event 
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
 2012-11-16 15:38:29 +01:00
Wim Taymans
bd91bd3193 rtspsrc: fix check for active streams 
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
 2012-11-16 15:22:46 +01:00
Wim Taymans
11cf4d4fd3 rtspsrc: create and add pads outside of lock 
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
 2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque
6c855edf03 rtspsrc: allow client to disable reconnection 
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
  rtspsrc always tried to reconnect to the server when the RTSP
  connection was closed by the server. This property lets the user
  decide whether it wants rtspsrc to reconnect or not.

  https://bugzilla.gnome.org/show_bug.cgi?id=683912
 2012-11-16 12:55:10 +01:00
Wim Taymans
e2a4d28c1f rtspsrc: clear variables before retrying 
Else we might unref an old udpsrc twice in cleanup.
 2012-11-16 12:17:37 +01:00
Wim Taymans
cc9cb26be1 rtspsrc: propose ports in multicast 
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-16 12:17:37 +01:00
Wim Taymans
5025b3f1b3 rtspsrc: add more debug 2012-11-16 12:17:37 +01:00
Tim-Philipp Müller
6f1aa3e4d5 multifilesink: post messages in max-size mode as well 
No reason not to really.
 2012-11-16 09:13:22 +00:00
Wim Taymans
c33507f186 udpsrc: post error before stopping 2012-11-15 14:48:59 +01:00
Tim-Philipp Müller
bdf3c77828 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:13:36 +00:00
Nicolas Dufresne
673d2d24b8 videoflip: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=688225
 2012-11-13 14:25:04 +01:00
Wim Taymans
c755af0cb0 rtpsource: protect against invalid RTP packets 2012-11-12 11:18:30 +01:00
Tim-Philipp Müller
35fafae241 videocrop: add support for YV12 
We can do I420, so we can do YV12 as well.
 2012-11-10 18:21:28 +00:00
Alessandro Decina
b916d2b398 multifilesink: don't write stream headers with key-unit-event 
Don't write stream headers, let upstream elements insert them in the stream if
all_headers=true is set in key unit events.
 2012-11-10 12:41:33 +01:00
Nicolas Dufresne
e111068f7b videocrop: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=687964
 2012-11-10 01:52:44 +01:00
Sebastian Dröge
c70ba7765a udpsrc: Also clear GError 2012-11-09 11:22:30 +01:00
Sebastian Dröge
b86d20e45b udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets 
See bug #529454 and #687782 and commit
751f2bb364
 2012-11-09 11:20:27 +01:00
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
Nicolas Dufresne
1ad8ebac44 videocrop: Add support for automatic cropping 
This change enable automatic cropping using -1 set to left, top, right or
bottom property. In the case both side are set to automatic cropping, the
croping will be done equally on both side (in the odd case, right and
bottom cropping will be 1 pixel more).

https://bugzilla.gnome.org/show_bug.cgi?id=687761
 2012-11-07 11:20:24 +01:00
Marc Leeman
7cbca3dcd1 rtsp: the RTCP port number is inclusive 
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.

See https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-06 13:22:58 +01:00
Tim-Philipp Müller
beb3c9c9be Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:09:59 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans
9857e6af4d vrawdepay: don't access rtp buffer after unmap 
Read the marker bit before we unmap the rtp packet.
 2012-11-02 18:48:17 +00:00
Douglas Bagnall
0b898ab911 videoconvert: Compare y offset with height, not width, when testing for overlap 
This could have prevented images showing that should have when the
source height is greater than its width.

When width exceeds height, as is common, it probably only caused a
miniscule amount of unnecessary work.  I haven't tested.
 2012-11-02 09:29:30 +01:00
Tim-Philipp Müller
5ac789408b rtpvp8: include config.h and minor style fixes 2012-11-01 21:10:21 +00:00
Tim-Philipp Müller
4a849d6690 rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
321acd14dc rtp: move VP8 payloader and depayloader from -bad 
Spec is still in draft state, but should hopefully not
change much now. Besides, we announce things as VP8-DRAFT-IETF-01
in our caps, so even if things change in incompatible ways it
should not break anything.

https://bugzilla.gnome.org/show_bug.cgi?id=687263
 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
44efab8e3d rtpvp8: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
bc7dbbbd4f rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-11-01 20:53:48 +00:00
Sebastian Dröge
4853001547 rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-11-01 20:53:48 +00:00
Wim Taymans
fccfca38d4 rtpvp8: update for buffer changes 2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula
3edffb13e3 rtpvp8; fix compatibility with the third draft 
https://bugzilla.gnome.org/show_bug.cgi?id=671073
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
d9581832a0 rtpvp8: port some more to new memory API 2012-11-01 20:53:47 +00:00
Olivier Crête
c6761daa27 rtpvp8: port to 0.11 2012-11-01 20:53:47 +00:00
Sebastian Dröge
2c5ea76bdc rtpvp8pay: Fix typo 2012-11-01 20:53:47 +00:00
Youness Alaoui
1cf155d70d rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2012-11-01 20:53:47 +00:00
Vincent Penquerc'h
88aade4150 rtpvp8: fix bitstream parsing using the wrong kind of bitreader 
VP8 uses a probabilistic bool coder, not a straight bit coder.
This fixes parsing when error-resilient is set.

This commit includes a copy of libvpx's bool coder, BSD licensed.

https://bugzilla.gnome.org/show_bug.cgi?id=652694
 2012-11-01 20:53:47 +00:00
Olivier Crête
97c3f3617c rtpvp8: Reject unknown bitstream versions 2012-11-01 20:53:47 +00:00
Edward Hervey
74a1a704bf rtpvp8: Fix unitialized variable 
Makes macosx compiler happy.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
6ed6318076 rtpvp8depay: Accept packets with only one byte of data 
When fragmenting partions it can happen that an RTP packet only caries 1
byte of RTP data.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
a45e7a3fc0 rtpvp8pay: Treat the frame header just like any other partition 
When setting up the initial mapping just act as if the global frame
information is another partition. This saves special-casing it later in
the actual packetizing code.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
e9f4e9342f rtpvp8: Add simple payloaders and depayloaders for VP8 
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
 2012-11-01 20:53:47 +00:00
Wim Taymans
d6fd0ebd04 gstpay: fix for 1.0 events 
Caps events are sometimes not followed by a buffer but by an event. Flush any
pending caps before we make a packet with the event.
Chain up to the parent event handler before we attempt to push RTP packets, it
might be a segment event.
 2012-11-01 18:42:39 +00:00
Wim Taymans
05232c55a5 gstdepay: fix small leak 2012-11-01 18:42:24 +00:00
Wim Taymans
08e5a197b4 gstdepay: add support for events 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 18:18:19 +00:00
Wim Taymans
54b783b5a3 rtpgstpay: add support for sending events 
We currently only send tags and custom events. The other events
might interfere with the receiver timings or are otherwise handled
by RTP.

Conflicts:
	gst/rtp/gstrtpgstpay.c
 2012-11-01 18:06:11 +00:00
Wim Taymans
6502d08e43 gstpay: rewrite payloader 
Use adapter to assemble the payload and make a flush function to
turn this payload into (fragmented) packets.

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
 2012-11-01 17:57:52 +00:00
Douglas Bagnall
e3c77ba709 videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH 
https://bugzilla.gnome.org/show_bug.cgi?id=687330
 2012-11-01 13:03:44 +00:00
Douglas Bagnall
79403bcb0c videbox: fix border filling for gray formats 
Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH.

https://bugzilla.gnome.org/show_bug.cgi?id=687330
 2012-11-01 13:02:16 +00:00
Wim Taymans
c0713e4b80 gstdepay: check for correct fragment offset 
Make sure we only insert the rtp packet in the adapter when the
frag_offset matches. When the first packet of a fragment is dropped,
it avoids putting the remaining packets in the adapter and processing
the partial fragment.

Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 12:09:47 +00:00
Wim Taymans
8a402e0c06 gstpay: set C flag on all buffers of the fragment 
Set the C flags on all the fragments instead of only those with
caps in them. This makes it easier in the receiver to check if there
is a caps in the assembled fragments just by looking at the last RTP
packet flags.
 2012-11-01 12:06:08 +00:00
Wim Taymans
d78ff07f7d gstdepay: use the capsversion 
Take the caps from the input caps and store it in the slot given
by capsversion.
 2012-11-01 11:37:44 +00:00
Wim Taymans
936c3819b5 gstpay: send caps inline 
Place the capsversion on the outgoing caps so that they end up in
an SDP as well. Receivers need to know what capsversion a particular
caps is for to be able to match the caps to the CV in the RTP packets.
Place the caps inside the RTP packet whenever the caps change.

Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
 2012-11-01 11:34:33 +00:00
Andrzej Bieniek
3b1931a039 gstpay: add debug 
Conflicts:
	gst/rtp/gstrtpgstpay.c
 2012-11-01 11:28:50 +00:00
Andrzej Bieniek
ee5ecc7773 depay: correctly skip caps header size 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 11:27:13 +00:00
Tim-Philipp Müller
ef0805ea14 matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work 
https://bugzilla.gnome.org/show_bug.cgi?id=640589
 2012-10-30 23:29:46 +00:00
Tim-Philipp Müller
752cf98745 gst: fix variable order in some Makefile.am 
https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:27:38 +01:00
Antoine Tremblay
a1c86de09a gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am 
Those plugins depend on either libgstaudio or libgstvideo,
which are in gst-plugins-base.

https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:26:41 +01:00
Alexey Fisher
29cd24bc41 matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag 
https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-10-27 14:46:02 +01:00
Stas Sergeev
238a5ec826 multifilesrc: fix stop index handling 
Make sure the stop index is always honoured. Avoids
endless loop if one wants to read and output the same
file N times, for example.

https://bugzilla.gnome.org/show_bug.cgi?id=654853
 2012-10-26 11:04:01 +01:00
Руслан Ижбулатов
78193dfe71 matroskademux: Support recursive SimpleTags 
Fixes #682644
Depends on #682615
 2012-10-26 10:16:42 +02:00
Руслан Ижбулатов
cd719bb808 matroskademux: Expand the tag mapping. 
* Also expose unknown tags as key=value pairs.
* Arrange tag map in the same order tags are listed in Matroska spec, leaving
unmapped tags as comments.
* More specific TODOs.
* Remove duplicate DATE define.

Fixes #682615
Depends on #682524
 2012-10-26 10:12:52 +02:00
Sebastian Dröge
6c635ce64f matroskademux: Fix uninitialized variable compiler warning 2012-10-26 10:09:39 +02:00
Руслан Ижбулатов
71fd688ef0 matroskademux: Matroska tag TargetType support 
* Reads TargetType and TargetTypeValue from a Tag.
* After Tag is completely read, processes taglist, substituting some of the
tags depending on target type value and the presence of video/subtitle streams.
* Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS

Depends on #682448
Fixes #682524
 2012-10-26 10:08:18 +02:00
Руслан Ижбулатов
b75628f041 matroskademux: Per-track tags for Matroska 
Requires Matroska file to have sane layout (track info before tag info).
Uses replace-merge.
Makes track UIDs 64-bit.

Fixes #682448
 2012-10-26 10:03:55 +02:00
Tim-Philipp Müller
fe7236230c multifilesrc: fix typo in property description 2012-10-25 20:19:44 +01:00
Michael Smith
a88caf84b4 qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. 
Fixes aspect ratio of prores files.
 2012-10-25 12:18:50 -07:00
Thiago Santos
02d91dcd24 imagefreeze: the new get_caps already does the filter intersection 
It should be faster to pass the caps to intersect as the filter caps,
rather than using NULL and intersecting 'manually' later.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 10:32:17 -03:00
Thiago Santos
115581eb2e imagefreeze: avoid assertion when using accept caps query 
This query must receive a fixed caps, so imagefreeze should
fixate its framerate before sending the query downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 09:39:36 -03:00
Arnaud Vrac
bc79fe565c qtdemux: use correct type for channel-mask bitmask 
Fixes crash on 32-bit systems.
 2012-10-24 12:54:08 +01:00
Tim-Philipp Müller
7275860bdd flacparse: fix coverart extraction if vorbis comments come after picture header 
See sample file for bug #684701.
 2012-10-23 16:02:05 +01:00
Tim-Philipp Müller
7c41f42eec flacparse: ignore bad headers if we have a valid STREAMINFO header 
If we run into any header parsing issues and we have a valid
STREAMINFO header already, don't error out, but just stop
header parsing and try to find some audio frames.

https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Tim-Philipp Müller
49cc719809 flacparse: post proper error message and fix buffer leak on header parsing error 
https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Michael Smith
150bd97e96 qtdemux: with raw audio, set a default channel-mask for multichannel audio. 
This doesn't actually parse 'chan' because it's absurdly complex.
 2012-10-22 22:34:43 -07:00
Sebastian Rasmussen
9fc62a58e3 updsrc: fix typo causing compilation error 
gstudpsrc.c: In function 'gst_udpsrc_create':
gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function

https://bugzilla.gnome.org/show_bug.cgi?id=686642
 2012-10-22 23:19:28 +01:00
Wim Taymans
a2eead3d60 avi_ fix invert function 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:55:59 +02:00
Wim Taymans
0e3ef30c31 avi: fix debug 2012-10-22 11:55:22 +02:00
Wim Taymans
199aaa4021 qtdemux: add support for 'generic' samples 
Add support for stuffing a complete stream into 1 sample.

See https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:39:37 +02:00
Tim-Philipp Müller
aa3ba65eb5 qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio 
https://bugzilla.gnome.org/show_bug.cgi?id=681192
 2012-10-19 19:26:45 +01:00
Mark Nauwelaerts
35cd53867c matroskamux: unsigned subtitle template 2012-10-19 16:14:01 +02:00
Youness Alaoui
13328bc129 videomixer2: Fix race condition where a src setcaps is ignored 
If both pads receive data at the same time, they will both get their
sink_setcaps called which will call the src_setcaps, but there is
a race condition where the second one might not be called.
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842
 2012-10-19 12:10:31 +02:00
Mark Nauwelaerts
5742352e10 matroskamux: do not use unoffical V_MJPEG codec id 
Since it's not spec'ed, consider it a VfW compatibility
case. Many applications (e.g. avidemux) don't understand
the unofficial V_MJPEG id.

Fixes #659837.

Conflicts:
	gst/matroska/matroska-mux.c
 2012-10-18 18:29:40 +01:00
Tim-Philipp Müller
488549bb54 Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-10-17 17:46:34 +01:00
Wim Taymans
e9040e90a5 jpegdepay: store quant tables in zigzag order 2012-10-17 14:23:01 +02:00
Wim Taymans
d5fd524a0c rtsession: fix compiler warning 2012-10-17 13:55:45 +02:00
Wim Taymans
26a21e85e2 rtpbin: clarify the ntp-sync option 2012-10-17 13:35:07 +02:00
Wim Taymans
f17db5c4ed rtpsession: update caps in the source 
Inform the source when caps changed. This was removed in the port to 1.0
leaving the source unaware of the clock-rate and unable to interpollate
rtp timestamps for SR packets.
 2012-10-17 13:22:40 +02:00
Wim Taymans
f4eef3f48d rtpbin: set PTS and DTS in jitterbufffer 2012-10-17 12:46:32 +02:00
Wim Taymans
796c1d8029 rtpbin: disable check for ntp-sync 
Disable the check for the ntp-sync method. It is expected that
a rather larger offset needs to be applied with this method.
 2012-10-17 12:27:03 +02:00
Wim Taymans
1cebcfa8c2 rtpbin: use running-time for NTP time 
When use-pipeline-clock is set, use the running-time of the
pipeline to calculate the NTP timestamps. This method would previously
only work when the base-time is set to 0 but with this change it can
also work with different offsets and we can also implement pause/resume
of the sender and receiver now.
 2012-10-17 12:26:05 +02:00
Wim Taymans
5ec642d0c3 videocrop: port to videofilter 2012-10-17 10:20:12 +02:00
Wim Taymans
3ef7c8ab93 videobox: use out_info for out properties 2012-10-17 09:36:50 +02:00
Wim Taymans
f701d980e6 median: small cleanups 2012-10-16 14:40:19 +02:00
Wim Taymans
0e21e80e9b median: remove now that it is in videofilter 2012-10-16 13:56:19 +02:00
Wim Taymans
9e67891f72 videomedian: copy media to videomedian 
Copy the median video filter to videofilters and rename to
videomedian.
 2012-10-16 13:47:24 +02:00
Wim Taymans
b893197317 media: port to 1.0 2012-10-16 13:16:29 +02:00
Tim-Philipp Müller
f94572fb36 avidemux: append palette data to paletted 8-bit RGB frames 
Fixes playback of 8-bit indexed RGB videos, with fixes in -base.

https://bugzilla.gnome.org/show_bug.cgi?id=686046
 2012-10-16 01:09:05 +01:00
Tim-Philipp Müller
e9682b938a qtdemux: don't assert if upstream size is not available when guessing bitrates 
Fixes abort in push mode where the source is not seekable and the
size of the file is not available, as with

  cat foo.mp4 | gst-launch-1.0 playbin uri=fd://0

Less noticable with releases, since we disable all
g_assert() there.

https://bugzilla.gnome.org/show_bug.cgi?id=686008
 2012-10-13 00:08:01 +01:00
Michael Smith
3a3a7c38aa qtdemux: allow more streams. Bump this constant to 32, which should be 
enough for real-world files.
 2012-10-12 14:38:33 -07:00
Michael Smith
d60c9ce2a4 qtdemux: support more different fourcc values for other ProRes variants. 2012-10-12 14:35:49 -07:00
Wim Taymans
adb70e89f9 rtspsrc: remove unused include 2012-10-10 12:05:34 +02:00
Rasmus Rohde
11ed7c0373 multiudpsink: add multicast-iface property 
udpsrc already has support for setting the multicast interface, which
is useful for multi-homed machines. This patch adds the same code to
the multiudpsink.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685864
 2012-10-10 11:48:25 +02:00
Wim Taymans
54f049c355 multiudpsink: don't error on send errors but only warn 
Don't error on send errors but simply post a warning, it's possible
that the next packet will be fine.
 2012-10-10 11:32:17 +02:00
Rasmus Rohde
6c169312d1 multiudpsink: add force-ipv4 option 
Add an option to the multiudpsink that makes it possible to force
the use of an IPv4 socket.

This can e.g. be used to handle the issue described in
https://bugzilla.gnome.org/show_bug.cgi?id=682481
 2012-10-10 10:28:24 +02:00
Wim Taymans
2955f0e10c multiudpsink: remove unused field 2012-10-10 10:18:52 +02:00
Wim Taymans
f4e1bb02b7 udpsrc: use negotiated allocator or pool 
Use the base class to allocate a buffer for us because it knows how
to use the negotiated allocator or bufferpool.
 2012-10-10 10:10:26 +02:00
Wim Taymans
e8d951ed68 multiudpsink: post error when something goes wrong 2012-10-10 10:09:37 +02:00
Wim Taymans
15c2b997e9 spectrum: elements post element messages 2012-10-10 10:09:10 +02:00
Michael Smith
7aed5a4b4b deinterleave: output channels should be marked as MONO, not FRONT_LEFT, if 
we're not preserving input channel positions.
 2012-10-05 15:12:27 -07:00
Michael Smith
7522cd1595 interleave: use gst_audio_channel_positions_to_mask instead of a local copy 
of half of it. Handles some values more correctly.
 2012-10-04 15:13:20 -07:00
Rasmus Rohde
47a8eb7ca8 gstrtpdepay: don't leak input buffer 
The rtp buffer is never unmapped in the normal code exit path
of gst_rtp_gst_depay_process(..) resulting in a memory leak.

https://bugzilla.gnome.org/show_bug.cgi?id=685512
 2012-10-04 19:44:28 +01:00
Sebastian Dröge
1ac6a782c3 videobalance: Add support for NV12 and NV21 2012-10-04 18:37:48 +02:00
Patricia Muscalu
7a863e4d8d rtph264pay: do not push unmapped data 
Also do not use a GstBuffer after it has been pushed into the adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=685213
 2012-10-04 09:22:50 +01:00
Michael Smith
b04b1b5089 meta info: threadsafe registration using g_once 2012-10-03 10:51:45 -07:00
Mark Nauwelaerts
b10829d6c8 avidemux: push mode; handle some initial junk before hdrl list 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685059
 2012-10-01 15:50:53 +02:00
Tim-Philipp Müller
e6d37eb30a Purge references to liboil 
https://bugzilla.gnome.org/show_bug.cgi?id=673285
 2012-09-29 12:41:37 +01:00
Mark Nauwelaerts
cb0e4b2059 avidemux: recognize all xsub frames as keyframes 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977
 2012-09-28 17:04:42 +02:00
Mark Nauwelaerts
511dfa5ee5 avidemux: push mode: find the correct chunk for segment following seek 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977
 2012-09-28 17:04:42 +02:00
Arnaud Vrac
f0db4a8213 qtdemux: fix parsing in push mode when moov atom is at the end 
When playing an mp4 file with the MOOV atom at the end of the file, playback
fails with the error message "no 'moov' atom within the first 10 MB". This is
due to a mistake in the upstream_size typing, making the seek to the end of
file never happening.

https://bugzilla.gnome.org/show_bug.cgi?id=684972
 2012-09-27 22:20:19 +01:00
Andre Moreira Magalhaes (andrunko)
25803d651b gamma: remove duplicate entries at format at caps 
Avoids extra caps/structures processing
 2012-09-27 15:50:49 -03:00
Wim Taymans
dbe941338d rtpvrawdepay: negotiate pool with srcpad caps 2012-09-27 14:15:50 +02:00
Tim-Philipp Müller
f5e0321dfc videomixer: clear video frame more correctly 
Make sure not to touch memory that doesn't belong to
our frame, we might be one part of a side-by-side 3D
frame, or in a picture-in-picture scenario.
 2012-09-26 09:28:59 +01:00
Tim-Philipp Müller
c203ce2dbe flvdemux: minor clean-up 
Use GstByteWriter, because we can, and g_value_take_boxed.
 2012-09-26 00:44:59 +01:00
Dmitriy Samonenko
7d4b6f655e flvdemux: fix speex audio decoding by creating fake stream header 
https://bugzilla.gnome.org/show_bug.cgi?id=683622
 2012-09-26 00:16:06 +01:00
Tim-Philipp Müller
626e0258e3 videomixer: fix warnings when using transparent background 
gst_video_frame_map() increases the refcount, which makes
the buffer not writable any more technically, so calling
gst_buffer_memset() on it will cause nasty warnings.

Unit test disabled because it very rarely (for me)
fails, possibly negotiation-related.

https://bugzilla.gnome.org/show_bug.cgi?id=684398
 2012-09-25 23:31:34 +01:00
Robert Swain
03e5376827 deinterlace: Add some useful debug logging 2012-09-25 17:05:37 +02:00
Robert Swain
33dd81569f deinterlace: Fix telecine 
This only affects behaviour in telecine cases with pattern locking
enabled. The default case should be untouched.

This works with the output from fieldanalysis at least, but the field
order looks swapped for telecine mixed buffers with the
David_slides_Schleef clip.
 2012-09-25 17:04:54 +02:00
Edward Hervey
ac9394de29 videomixer: Fix leak 2012-09-25 14:18:35 +02:00
Tim-Philipp Müller
ebe0b1887a smpte: send stream-start event 2012-09-23 16:51:31 +01:00
Tim-Philipp Müller
8e3c7fa799 multipartmux: send stream-start event 2012-09-23 16:51:24 +01:00
Tim-Philipp Müller
154404fa43 matroskamux: send stream-start 2012-09-23 16:33:35 +01:00
Tim-Philipp Müller
bc37b9f4fc qtmux: send stream-start event 2012-09-23 16:33:35 +01:00
Tim-Philipp Müller
ea7f8a919c interleave: add a bunch of FIXMEs 
Needs some more work, so stream-start, caps and tags are
sent in the right order.
 2012-09-23 16:33:35 +01:00
Tim-Philipp Müller
1c3c8c64e6 flvmux: send stream-start event 2012-09-23 16:33:34 +01:00
Tim-Philipp Müller
c3f62d7ead avimux: send stream-start event 2012-09-23 16:33:34 +01:00
Olivier Crête
0363c1cebf rtpdtmfdepay: Use 1.0-style caps negotiation and audio/x-raw 2012-09-22 15:00:27 -04:00
Tim-Philipp Müller
8b20603f8b rtspsrc: answer URI query 
Without this, something also answered the query
with TRUE but without setting a uri, not sure
what that was..
 2012-09-21 23:33:47 +01:00
Olivier Crête
bc252d29ee rtph264pay: Make sure the caps don't have duplicated sps/pps 2012-09-21 17:36:12 -04:00
Michael Smith
1026970347 qtdemux: 24 bit audio here is S24LE, not S24_3LE. 2012-09-20 18:01:52 -07:00
Robert Swain
480b894642 deinterlace: Remove incorrect logic 
I don't understand why these lines were added, they don't make sense to
me now and both David and I agree that removing them moves closer to
related logic being correct, therefore, they're being removed.

I've tested a few progressive, interlaced and telecine clips and they
all behave properly timestamp-wise and visually after these changes.
 2012-09-19 00:39:01 +02:00
Robert Swain
a35a931555 deinterlace: Fix field duration 
The frame rate fraction is correctly adjusted in the cases preceding the
field duration calculation and so the factor of 2 is incorrect.
 2012-09-19 00:17:49 +02:00
Michael Smith
63716151ef videobox: Fix U/V strides for a number of cases. 2012-09-18 10:34:03 -07:00
Mark Nauwelaerts
eda9c8b3cf videomixer: init videoinfo 
... to prevent random bogus caps fields.
 2012-09-18 12:15:17 +02:00
Mark Nauwelaerts
8c28a60eee videomixer: chain up to collectpads query function 2012-09-18 12:15:17 +02:00
Nicolas Dufresne
76da367ecd videomixer: Don't let GstCollectPad shadow custom sink pad query func 
In the current implementation, the custom pad query function is not called.
This patch, set that query function on the GstCollectPads to avoid this
shadowing.

See https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2012-09-18 12:14:43 +02:00
Mark Nauwelaerts
3eee42fdfc use gst_element_factory_get_metadata to replace obsolete API 2012-09-15 19:06:06 +02:00
Mark Nauwelaerts
0380de3f95 replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 17:27:49 +02:00
Wim Taymans
829c80ce6c fix more caps 2012-09-14 13:30:37 +02:00
Jan Schmidt
a27deda053 deinterlace: Don't treat every custom-downstream event as EOS 
Don't fall through to the EOS handling after receiving a
custom-downstream event.
 2012-09-12 12:23:08 -07:00
Stefan Sauer
f874922e1c collectpads: remove gst_collect_pads_add_pad_full 
Rename gst_collect_pads_add_pad_full() to gst_collect_pads_add_pad() and fix all
invocations.
 2012-09-12 21:05:44 +02:00
Mark Nauwelaerts
d6ca569c29 udp: add include for IPPROTO_* 2012-09-12 17:14:46 +02:00
Mark Nauwelaerts
58c96df0ae udp: properly match braces and cpp directives 
Fixes compilation where IPV6_TCLASS not defined.
 2012-09-12 16:39:08 +02:00
Edward Hervey
8498551692 shapewipe: Use default query handler where needed 
And clean up get_caps code while I'm at it
 2012-09-12 14:42:07 +02:00
Wim Taymans
1c64a91a50 deinterlace: improve framerate transform 
Handle G_MAXINT in the framerates better. If we cannot double or divide the
framerate, clamp to the smallest/largest possible value we can express instead
of failing.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683861
 2012-09-12 13:28:07 +02:00
Wim Taymans
6d9f9bf11a deinterlace: small cleanup 2012-09-12 13:17:54 +02:00
Youness Alaoui
c3d619be67 videomixer2: Adding nv12 and nv21 support 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683841
 2012-09-12 10:46:22 +02:00
Michael Smith
4f015c594c qtdemux: add support for prores 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683839
 2012-09-12 10:18:53 +02:00
Mark Nauwelaerts
f12ef67f56 ext, gst: only activate in pull mode if upstream is seekable 2012-09-11 17:44:51 +02:00
Wim Taymans
a374217786 qtdemux: don't reset segment 
Don't reset the segment because we need the values for accumulation. the segment
is reset at start and after a flushing seek. Fixes some problems with files with
quicktime segments.
 2012-09-11 11:59:54 +02:00
Mark Nauwelaerts
8d93246b93 gst: adjust comment style 2012-09-10 14:31:02 +02:00
Mark Nauwelaerts
ca36de1e8f avidemux: remove defunct commented code 2012-09-10 14:30:42 +02:00
Tim-Philipp Müller
6dc7b4c3c7 video/x-3ivx and video/x-xvid -> video/mpeg,mpegversion=4 
If it ever turns out that we really must use thoe specific
fourccs and not the generic one, we can still add a flavor
field to the caps later.
 2012-09-10 00:43:24 +01:00
Daniela
03fbd7ec6e rtspsrc: avoid leak 
When setup fails, make sure to cleanup afterwards.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509
 2012-09-07 16:33:18 +02:00
Mark Nauwelaerts
f24b58d19c rtpamrdepay: unmap rtp buffer 
... thereby plugging a memleak.
 2012-09-07 15:25:53 +02:00
Mark Nauwelaerts
fa90dfc4df rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 15:25:52 +02:00
Mark Nauwelaerts
8f4bfeb698 rtph263ppay: plug caps leak 2012-09-07 15:25:52 +02:00
Wim Taymans
ecaa2624d3 deinterlace: remove redundant _set_allocation call 2012-09-06 17:09:20 +02:00
Mark Nauwelaerts
1ce09d7ef9 deinterlace: plug some leaks 2012-09-06 17:05:49 +02:00
Wim Taymans
510482b01a deinterlace: reuse core function for GCD 2012-09-06 16:52:18 +02:00
Mark Nauwelaerts
9d4579b38a deinterlace: support filter in getcaps 2012-09-06 16:31:17 +02:00
Mark Nauwelaerts
a4458f5f74 deinterlace: do not leak getcaps result 2012-09-06 16:31:17 +02:00
Wim Taymans
45e5ec29ac deinterlace: add support for bufferpool 
Add bufferpool support to avoid a memcpy in the videosink when actively
interlacing.
Remove some commented obsolete code.
 2012-09-06 16:25:05 +02:00
Wim Taymans
f59fb16f58 deinterlace: proxy allocation query in passthrough 
We can let the allocation query pass when we are operating in passthrough mode.
 2012-09-06 13:38:52 +02:00
Wim Taymans
4efdbc97a5 deinterlace: use default event functions 
instead of blindly forwarding unknown events.
 2012-09-06 13:23:46 +02:00
Wim Taymans
a557282aaa deinterlace: small cleanups 2012-09-06 13:23:30 +02:00
Wim Taymans
f1ef3b4983 deinterlace: call default query handlers 
Call the default query handler instead of forwarding the query blindly. Fixes
issues of strides because of proxying the allocation query wrongly.
 2012-09-06 12:56:30 +02:00
Wim Taymans
6693a22875 videobalance: avoid deadlock 
_update_properties takes the object lock and should not be called when the
object lock is already taken.
 2012-09-04 12:35:53 +02:00
Tim-Philipp Müller
aeba106878 matroskamux: extract interlaced-ness of video track from interlace-mode field 
instead of the old boolean "interlaced" field.
 2012-09-03 12:46:03 +01:00
Tim-Philipp Müller
9bf90f47cf video/x-xvid -> video/mpeg,mpegversion=4 2012-09-03 02:51:24 +01:00
Tim-Philipp Müller
fb0f3c17f5 text/plain + text/x-pango-markup -> text/x-raw 2012-09-02 02:50:50 +01:00
Tim-Philipp Müller
b27ac94af2 gst_message_new_duration -> gst_message_new_duration_changed 2012-09-02 01:31:53 +01:00
Wim Taymans
5b394385b9 session: also stop probatation on existing sources 
Receiving an RTCP packet should also stop probation on sources we have seen
before.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065
 2012-08-30 22:07:24 +02:00
Aleix Conchillo Flaque
4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Mark Nauwelaerts
a2475a40a5 flacparse: fixup 0.11 port of suspect frame checking 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682959
 2012-08-30 11:30:01 +02:00
Mark Nauwelaerts
e1881d1e44 avidemux: avoid invalid H264 bytestream codec_data 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681369
 2012-08-28 19:01:11 +02:00
Mark Nauwelaerts
e523b42d41 qtdemux: port segment event creation to 0.11 2012-08-28 19:01:11 +02:00
Mark Nauwelaerts
748304ced7 qtdemux: release extra event ref when replacing pending newsegment event 2012-08-28 16:28:29 +02:00
David Corvoysier
d0eed20428 isomp4: add DASH tfdt box support 
MPEG DASH has defined a set of new boxes to specify duration, indexes and
offsets of ISOBMFF fragments.

The Track Fragment Base Media Decode Time (tfdt) Box can in particular be
included inside a traf box to specify the absolute decode time, measured on the
media timeline, of the first sample in decode order in the track fragment.

This information can be used by the isomp4 demux to find out the current position of
an MP4 fragment in the timeline.

This patch adds code to isomp4 to:
- parse the tfdt box
- adjust the time/position member of the new segment sent when playback starts

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677535
 2012-08-28 16:28:27 +02:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Tim-Philipp Müller
e4cb67fad8 docs: gst-launch-0.11 -> gst-launch-1.0 2012-08-27 21:20:29 +01:00
Tim-Philipp Müller
045c4b6ec8 deinterlace: the field in caps is "interlace-mode" not "interlace-method" 
Fix deinterlace unit test. Need to set right field on output caps.
Also remove right field (not old 0.10 "interlaced" boolean field)
from caps in unit test before comparing old and new.
 2012-08-27 21:20:29 +01:00
Michael Rubinstein
6ea5d31456 videomixer: fix endianness check on systems where non-glib endianness defines are not set 
On Windows LITTLE_ENDIAN without the G_ in was not defined,  so the
test comes out wrong.
 2012-08-24 19:45:11 +01:00
Wim Taymans
916e4c86fa udpsink: don't crash on NULL error 
Check if there is an error before retrieving its message.

See https://bugzilla.gnome.org/show_bug.cgi?id=682481
 2012-08-22 17:27:27 +02:00
Aleix Conchillo Flaque
8d864dbbfc rtspsrc: make jitterbuffer drop-on-latency available (fix #682055) 
Conflicts:

	gst/rtsp/gstrtspsrc.h
 2012-08-22 10:39:19 +02:00
Tim-Philipp Müller
bce47066ca video/x-dvd-subpicture -> subpicture/x-dvd 2012-08-20 23:30:38 +01:00
Tim-Philipp Müller
6ee9a7d228 multifilesrc: fix example pipeline in docs 2012-08-17 20:52:42 +01:00
Stefan Sauer
1f255a585b equalizer: enable presets for the n-band equalizer 
Add a test for saving and restoring the preset.
 2012-08-17 15:01:40 +02:00
Tim-Philipp Müller
0d148d9c6f deinterlace: fix not-negotiated errors on variable or missing framerate in input caps 
Remove some bogus code I added during porting that would error out
on missing or variable framerates in input caps. Handle this like
we do in 0.10

Fixes test_mode_disabled_passthrough unit test check.
 2012-08-14 01:20:19 +01:00
Sjoerd Simons
b19b914d3a law: Filter layout caps field 
The layout caps field shouldn't be passed through to the sink pad
of {mu,a}lawdec.

https://bugzilla.gnome.org/show_bug.cgi?id=681677
 2012-08-13 08:52:58 +02:00
Olivier Crête
264bcf7d6f rtph264pay: Make it actually work after cleanups 2012-08-08 19:49:05 -07:00
Sebastian Dröge
6586e42384 gst: Set alignment at the correct place of GstAllocationParams 2012-08-08 17:41:42 +02:00
Sebastian Dröge
6f74b2afb7 gst: Set alignment at the correct place of GstAllocationParams 2012-08-08 17:41:31 +02:00
Tim-Philipp Müller
0e6b66a2a0 gst: update disted orc files 2012-08-08 15:10:37 +01:00
Tim-Philipp Müller
787c314ec3 Silence some 'variable may be used uninitialized' compiler warnings 
When compiling with -DG_DISABLE_ASSERT
 2012-08-08 11:31:59 +01:00
Tim-Philipp Müller
4de8bd004c No code with side-effects inside g_assert() please 2012-08-08 11:07:55 +01:00
Olivier Crête
b4ff570532 multiudpsink: Return FLUSHING instead of ERROR on unlock 
If the base class asks multiudpsink to unlock, then it should return
FLUSHING, not ERROR
 2012-08-07 11:31:32 -07:00
Mark Nauwelaerts
2d179ebf90 flacparse: generate empty vorbiscomment for complete streamheaders if needed 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681335
 2012-08-07 12:24:42 +02:00
Olivier Crête
2e21ace12c rtpssrcdemux: Block pad while it is announced. 
Block the RTP pad and associated RTCP pads while they are being
announced. This it to prevent a race where one is announced and
before the callback has connected it, the other one gets a buffer.

We can't use the "padlock" of ssrcdemux because it causes deadlocks.
 2012-08-06 18:04:58 -07:00
Mark Nauwelaerts
1547fdbe5a rtpmparobustdepay: set correct data_size for generated dummy frame 
... which prevents getting stuck in a loop if such one is needed.
 2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
3e1832f5a4 rtpmparobustdepay: improve and fix debug statement 
... so it really informs about next rather than past frame.
 2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
31a1cb0a11 rtpmparobustdepay: update available bytewriter space when repositioning 
... and add some more assert to catch potential surprises early on.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558
 2012-08-06 14:58:21 +02:00
Sebastian Dröge
7b5925b5a4 gst: Add stream-id to stream-start events 2012-08-06 13:43:57 +02:00
Sebastian Dröge
46255d6ada matroskademux: Chain up to the parent class' query handler if no pad is provided 2012-08-06 10:59:18 +02:00
Olivier Crête
2aa360c936 rtpssrcdemux: Release lock before signalling new pad 
This prevents a deadlock where something would try to push an event
through the SSRC demux from the callback, causing the pads to be iterated
and the lock taken.
 2012-08-04 18:14:28 -07:00
Tim-Philipp Müller
c074bfd0b9 gst_tag_list_free -> gst_tag_list_unref 2012-08-04 16:10:16 +01:00
Mark Nauwelaerts
a549b0bf2c rtspsrc: manage race between connection closing and flushing 
... where the former can happen in task thread and the latter in mainloop
upon downward state change.
 2012-08-03 14:10:32 +02:00
René Stadler
75ee20ec67 qtdemux: fix double unref of private tag buffer 2012-08-01 12:16:41 +02:00
Anton Belka
86c236a5f6 wavparse: create TOC as needed 
Avoid creating the toc if the wav has no or empty cue chunk.
Also a small code cleanup.
 2012-07-30 20:39:19 +02:00
Tim-Philipp Müller
1ddb71e5b6 wavparse: update for TOC API changes 2012-07-28 11:26:01 +01:00
Tim-Philipp Müller
5b4eb723b6 matroska: update for TOC API changes 2012-07-28 11:22:43 +01:00
Tim-Philipp Müller
1d5ed57cfa flacparse: update for TOC API changes 2012-07-28 11:20:08 +01:00
Sebastian Dröge
0827f54b93 tag: Update for taglist/tag event API changes 2012-07-28 00:19:51 +02:00
Mark Nauwelaerts
dd25411161 qt(de)mux: pass private blob tags in a sample 
... rather than a buffer, and the detailed info in the sample info
rather than caps.
 2012-07-27 12:12:13 +02:00
Robert Swain
af7fee714d videocrop: Don't return NULL from _transform_caps 
If _transform_caps () returns NULL, the basetransform _transform_caps
tries to call gst_caps_is_subset () with a NULL subset which hits an
assertion.
 2012-07-27 11:33:12 +02:00
Mark Nauwelaerts
0bf9d8c6a6 rtpmparobustdepay: modify buffer data rather than buffer itself 2012-07-26 16:34:52 +02:00
Mark Nauwelaerts
c40807f6aa rtpmparobustdepay: avoid leaking bytewriter instance 2012-07-26 16:34:52 +02:00
Robert Swain
cc4941797d deinterlace: Fix timestamp adjustment and caps 2012-07-26 16:04:23 +02:00
Robert Swain
01016109d0 deinterlace: Fix/simplify telecine state checks 2012-07-26 16:03:57 +02:00
Robert Swain
db5bb81e36 deinterlace: Improve debug output 2012-07-26 12:31:52 +02:00
Robert Swain
f20d8f59c8 deinterlace: Fix low-latency pattern locking 2012-07-26 12:31:52 +02:00
Robert Swain
30a61f26ba deinterlace: RFF should be ignored in deinterlace 
RFF only occurs on progressive frames in telecine sequences. For
deinterlace, we don't want these repeated fields as we will simply be
pushing the progressive frame and then moving on.

However, we need to consider RFF in order to correctly identify patterns
and adjust the timestamps.
 2012-07-26 12:31:52 +02:00
Robert Swain
7c0af11fca deinterlace: Improve process logic 
The logic now works better if we filter orphans, then progressive, then
telecine interlaced fields which need to be woven and fall through to
interlace. Telecine interlaced fields will be regularly deinterlaced if
there is no pattern lock for us to be sure that we have a telecine
pattern.

Telecine sequences that aren't 24fps progressive with RFF flags can't
really be tested until fieldanalysis is ported.
 2012-07-26 12:31:52 +02:00
Wim Taymans
ef38efc2d7 rtsp: go and stay in the loop function on PLAY 
When we have a PLAY request, go into the LOOP function next. When we are
looping, keep on looping until we are told otherwise.
This fixed rtsp and TCP connections.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551
 2012-07-25 12:50:01 +02:00
Wim Taymans
943b56ff8e rtsp: set caps after activating the pad 2012-07-25 12:49:35 +02:00
Wim Taymans
0ed9e07c5d h264depay: small cleanups 2012-07-25 12:49:07 +02:00
Wim Taymans
0cb11943e5 xqtdepay: fix buffer refcount error 
After pushing the buffer into the adapter, we should not let the baseclass push
it out anymore. This error was introduced while porting to 0.11.

See https://bugzilla.gnome.org/show_bug.cgi?id=680540
 2012-07-25 10:11:29 +02:00
Stefan Sauer
242321e376 level: remove obsolete liboil comment 2012-07-24 21:42:40 +02:00
Mark Nauwelaerts
1a46572aaa matroskademux: push mode: increase segment accuracy following seek 
Conflicts:

	gst/matroska/matroska-demux.c
 2012-07-24 21:15:49 +02:00
Mark Nauwelaerts
ea0729ff32 matroskademux: perform proper KEY_UNIT seek also in push mode 
Conflicts:

	gst/matroska/matroska-demux.c
 2012-07-24 21:15:49 +02:00
Tim-Philipp Müller
d6f4f1e01f udpsrc: don't crash dereferencing NULL error when leaving multicast group on shutdown 
Strangely enough, if we do pass an error variable to be filled, we
no longer get an error on leaving.
 2012-07-24 20:06:07 +01:00
Mark Nauwelaerts
6cc2ad4744 avidemux: rearrange some checks to avoid NULL use 2012-07-24 16:05:32 +02:00
Mark Nauwelaerts
6cb106d690 avidemux: use same fourcc to determine caps in determining uncompressed-ness 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673898

Conflicts:

	gst/avi/gstavidemux.c
 2012-07-24 16:05:31 +02:00
Mark Nauwelaerts
e5369901ad Revert "avidemux: Don't consider 0 fcc_handler as uncompressed." 
This reverts commit c6b9f5b25a.

fourcc GST_RIFF_rgb = 0 still leads to raw uncompressed rgb caps.

See also https://bugzilla.gnome.org/show_bug.cgi?id=673898
 2012-07-24 16:05:31 +02:00
Mark Nauwelaerts
7e9dffa226 matroskademux: avoid NULL access when checking subtitle 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680388
 2012-07-24 12:33:41 +02:00
Edward Hervey
538c131b37 aacparse: Reset parser when we have caps without codec_data 
This ensures the detection (and proper downstream caps settings) will
actually happen when we have new incoming caps without codec_data.

This was easily triggered by streams from matroskademux which initially
provided caps with a constructed codec_data, but then pushed new caps
without the codec_data once it detected the stream was adts.
 2012-07-24 12:24:43 +02:00
Wim Taymans
f44808338f videomixer: prefix orc functions with video_mixer_orc_ 2012-07-24 09:17:09 +02:00
Wim Taymans
29743c3ed2 videobox: prefix orc functions with video_box_orc_ 2012-07-24 09:13:48 +02:00
Mark Nauwelaerts
d6ef204190 matroskademux: generate correct segment stream time 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680275
 2012-07-23 17:38:43 +02:00
Wim Taymans
4b92022120 rtp: always use buffer lists 2012-07-23 16:42:56 +02:00
Patricia Muscalu
3dd99f06f4 rtpmp4vpay: always enable buffer-lists 2012-07-23 16:17:37 +02:00
Patricia Muscalu
15cce2dd26 rtpjpegpay: always enable buffer-lists 2012-07-23 16:15:59 +02:00
Wim Taymans
7fdd607561 deinterlace: get frame flags correctly 
Also move the deinterlace plugin to ported status
 2012-07-23 15:50:18 +02:00
Mark Nauwelaerts
a5dfa3d689 matroskademux: proper parse recovery after seek 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680427
 2012-07-23 15:45:33 +02:00
Mark Nauwelaerts
33091e2bf5 flvdemux: clear old segment event when requesting new one 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680283
 2012-07-23 12:50:21 +02:00
Alban Browaeys
7b16eb49b8 wavparse: convert all non GST_FORMAT_BYTES to format bytes. 
Convert all non GST_FORMAT_BYTES to format bytes:
fixes:
GStreamer-CRITICAL **: gst_query_set_duration: assertion `format ==
g_value_get_enum (gst_structure_id_get_value (s, GST_QUARK (FORMAT)))'
failed
when playing more than one wav stream.
gst-plugins-base/tests/icles/playback/test7 uri1.wav uri2.wav
 2012-07-23 09:49:51 +02:00
Sebastian Dröge
cbf3c2bac0 wavparse: Don't fail if more data then needed is available when parsing cue chunks 
Fixes bug #680328.
 2012-07-23 09:26:40 +02:00
Sebastian Dröge
e7977d2d64 wavparse: Some minor cleanup to the cue/labl parsing 2012-07-23 09:26:40 +02:00
Robert Swain
eac172c433 deinterlace: Port to 1.0 
This requires the additional INTERLACED buffer flag recently added to
-base
 2012-07-20 23:23:42 +02:00
Wim Taymans
ec7f7264dc interleave: convert the output segment to time 
Convert the stored input segment to time before pushing it out.

Conflicts:

	gst/interleave/interleave.c
 2012-07-20 16:09:33 +02:00
Wim Taymans
4dfb796527 interleave: try to fix segment handling 
Conflicts:

	gst/interleave/interleave.c
 2012-07-20 15:54:38 +02:00
Sebastian Dröge
b4621cbb3a matroskademux: Non-update seeks should still make sure that reverse playback status is reset 
Conflicts:
	gst/matroska/matroska-demux.c
 2012-07-20 15:33:43 +02:00
Sebastian Dröge
9a83a0749e matroskademux: Properly initialize from_offset and from_time 2012-07-20 15:33:04 +02:00
Sebastian Dröge
b02034dda1 matroskademux: We need an index and index entry for reverse playback 
Reverse playback does not work with index-less files yet.
 2012-07-20 14:28:37 +02:00
Mark Nauwelaerts
d90686f722 wavparse: clean up push mode segment handling 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680277
 2012-07-20 14:10:41 +02:00
Mark Nauwelaerts
7247d136e5 qtdemux: properly transform incoming segment event 
... which is really useful for proper push mode seeking.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680278
 2012-07-20 13:35:29 +02:00
Sebastian Dröge
6dbc6ad3cf matroskademux: Fix reverse playback for seeks without stop position 
Conflicts:
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-demux.h
 2012-07-20 11:23:16 +02:00
Sebastian Dröge
42b5065cc4 matroskademux: Only take the stream_start_time into account for SET seeks 
For other seeks the stream_start_time is already added to the
segment values.

Conflicts:
	gst/matroska/matroska-demux.c
 2012-07-20 11:18:27 +02:00
Anton Belka
cc6d533521 wavparse: Add TOC support 
Add support for:
 * Cue Chunk
 * Associated Data List Chunk
 * Label Chunk

https://bugzilla.gnome.org/show_bug.cgi?id=677306
 2012-07-20 09:55:50 +02:00
Maria Giovanna Chiossa
561b131e1a rtspsrc: also set UDP buffer size in multicast 
Also set the UDP buffer size in multicast mode.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448
 2012-07-19 15:26:36 +02:00
Tim-Philipp Müller
f879e4e0f0 avidemux: fix header parsing in push mode 
Fix 'break' that got warped to the wrong place,
probably as part of a merge. Fixes GST_IS_BUFFER
criticals in parse_idit() when being accidentally
passed a NULL buffer because of the missing break.

gst-launch-1.0 playbin uri=http://docs.gstreamer.com/media/sintel_trailer-480i.avi
 2012-07-18 23:43:59 +01:00
Wim Taymans
ac2a366a12 update for ghostpad changes 2012-07-18 18:07:02 +02:00
Sebastian Dröge
9fdcad4aee matroskademux: Pass seek rate to upstream seek events in push mode 
Fixes bug #679435.

Conflicts:
	gst/matroska/matroska-demux.c
 2012-07-18 11:40:56 +02:00
Wim Taymans
3371297afc update for RTP buffer api changes 2012-07-17 16:39:02 +02:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Patricia Muscalu
d38ac43a27 rtph264pay: use buffer lists 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994
 2012-07-17 10:10:14 +02:00
Sebastian Dröge
b01cf1561c flacparse: Fix parsing of ISRC from the cuesheets 2012-07-17 10:01:54 +02:00
Anton Belka
ffc204e6bd flacparse: add TOC support 
Add support embedded cuesheets in flac files.
Parsing METADATA_BLOCK_CUESHEET as TOC.

https://bugzilla.gnome.org/show_bug.cgi?id=540891
 2012-07-17 09:58:07 +02:00
Mark Nauwelaerts
a94d5d9f3b flacparse: avoid some more frame misparsing by additional header sanity check 
... using a required constant blocking_strategy bit.

https://bugzilla.gnome.org/show_bug.cgi?id=679807
 2012-07-13 15:37:18 +02:00
Edward Hervey
f063e40af7 demux: Push STREAM_START event when needed 2012-07-13 13:51:48 +02:00
Stefan Sauer
0cff483bd7 qtmux: avoid warning if both ts are equal 2012-07-11 13:54:00 +02:00
Tim-Philipp Müller
80245e2a70 multiudpsink: check the right size when warning about too large udp packets 
What matters is the total size, not the size of any of the
individual memory chunks that make up the packet.
 2012-07-11 12:31:13 +01:00
Wim Taymans
ab77c424be autodetect: proxy ts-offset properties 
Proxy the ts-offset property in the audio*sink elements.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679343
 2012-07-10 14:38:21 +02:00
Wim Taymans
2052cabdc4 fix for allocator API changes 2012-07-09 16:28:41 +02:00
Mark Nauwelaerts
f1b435d1b5 update for riff field rename 2012-07-09 12:53:47 +02:00
Tim-Philipp Müller
945ed74ebe dtmfsrc: pass unhandled non-custom events to the base class 
https://bugzilla.gnome.org/show_bug.cgi?id=666626
 2012-07-08 00:08:55 +01:00
Tim-Philipp Müller
c6224443a4 rtph264pay: avoid some relocations 2012-07-06 19:11:02 +01:00
Tim-Philipp Müller
3ef35ecdbc rtpmp4vpay: remove deprecated send-config property 
Use config-interval instead.
 2012-07-06 14:49:18 +01:00
Tim-Philipp Müller
cd1da84bcc rtph264depay: remove deprecated "byte-stream" and "access-unit" properties 
These will be picked automatically based on downstream caps now, so
if you want the depayloader to output a specific format, make sure
the element downstream advertises that preference or use a capsfilter
after the depayloader to force it.
 2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
cffbf8cfc3 rtph264pay: remove deprecated and non-functional "profile-level-id" property 
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.
 2012-07-06 14:46:22 +01:00
Mark Nauwelaerts
400bdee601 aacparse: perform additional sanity check before confirming ADTS format 
... and tweak confusing debug message.
 2012-07-06 15:29:37 +02:00
Mark Nauwelaerts
986286a8ea aacparse: remove unhelpful stray debug message 2012-07-06 15:29:28 +02:00
Tim-Philipp Müller
c22268b5d3 rtpsession: remove deprecated and unused "ntp-ns-base" property 2012-07-06 13:16:00 +01:00
Tim-Philipp Müller
c60625a5e4 docs: update isomp4 docs for gppmux -> 3gppmux change as well 2012-07-06 12:57:34 +01:00
Tim-Philipp Müller
cf9b2149dd isomp4: remove gppmux, which was deprecated in favour of 3gppmux 2012-07-06 12:54:02 +01:00
Tim-Philipp Müller
1cb8295bb0 smtp: remove deprecated "fps" property 2012-07-06 12:49:54 +01:00
Tim-Philipp Müller
080cbf322f multipartdemux: remove deprecated and unused "autoscan" property 
Replaced by boundary=NULL.
 2012-07-06 12:46:30 +01:00
Tim-Philipp Müller
48706beb70 rtph263ppay: accept any h263 input unless downstream forces specific requirements 
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.

rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes

  videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink

work.
 2012-07-06 11:57:38 +01:00
Wim Taymans
8eadb9c12c update for query api changes 2012-07-06 11:26:46 +02:00
Sebastian Dröge
aeafc3a093 gst: Implement segment-done event 2012-07-05 13:13:09 +02:00
Sebastian Dröge
2e90ff9bb9 matroskademux: Remove the TOC query handling 2012-07-05 12:35:49 +02:00
Sebastian Dröge
04e0bbef17 matroska: Update for new GstToc API 
TOC support in matroskamux is disabled for now as it was broken anyway.
 2012-07-05 12:28:59 +02:00
Tim-Philipp Müller
8098a2f0b2 imagefreeze: clear 0 DTS on buffers output, as sinks will prefer DTS over PTS for syncing 
Since the initial decoded still image buffer will have dts=pts=0, and
we only set PTS on buffers we push out, all buffers pushed out would
have a DTS of 0. Sinks, however, will prefer DTS over PTS if both are
set, and will therefore always see a timestamp of 0 no matter what
the PTS is set to.

Fixes unit test too.
 2012-07-04 19:03:12 +01:00
Tim-Philipp Müller
42cc0d1e48 deinterleave; downgrade caps change failure debug message 
Add some more info and downgrade to warning, so
it doesn't look like the unit test failed.
 2012-07-03 19:44:26 +01:00
Tim-Philipp Müller
0fa3992e37 audiopanorama: fix negotiation and unit test 
Must remove a possibly-fixed channel-mask field if
we're going to set unfixed channels on the structure,
or a different channel count.
 2012-07-03 17:54:22 +01:00
Sebastian Dröge
407bf06dc4 matroskademux: Only push the TOC event, the message is handled by the sinks 2012-07-03 17:34:10 +02:00
Javier Jardón
c740490c26 rtp: remove some outdated comments 
https://bugzilla.gnome.org/show_bug.cgi?id=679301
 2012-07-03 08:58:26 +01:00
Tim-Philipp Müller
b9d020ac4f rndbuffersize: add push mode support 
https://bugzilla.gnome.org/show_bug.cgi?id=656317
 2012-06-28 20:05:09 +01:00
David Corvoysier
c06cb7c145 isomp4: Try to seek upstream before processing seek push event 
When it receives a seek in push mode, the qtdemux should first try to push the event upstream, and only if upstream fails fall back to
its own seek logic.
 2012-06-28 14:44:58 +02:00
David Corvoysier
998534a2a1 isomp4: Allow duration queries to be forwarded upstream 
When receiving a duration query for TIME format, try to query upstream, and only if upstream fails fall back to qtdemux duration handling.
 2012-06-28 14:44:58 +02:00
Wim Taymans
6d158775bb rtph264pay: cleanups 
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.
 2012-06-28 12:00:09 +02:00
Wim Taymans
429bda6923 h264pay: prefer AVC, it's easier to parse etc 2012-06-28 11:32:03 +02:00
Tim-Philipp Müller
83cb4c63c3 matroska: update for GstToc API additions 2012-06-26 18:48:11 +01:00
Wim Taymans
e565f0d1ff matroska: set interlace-mode 2012-06-26 17:04:41 +02:00
Tim-Philipp Müller
2c04c30ec3 matroska-mux: update for GstTocSetter changes 2012-06-25 20:11:53 +01:00
Sebastian Dröge
dff2fec970 matroskademux: Return FALSE from queries if we can't answer POSITION/DURATION queries 2012-06-25 13:33:57 +02:00
Anton Belka
c3061f434b matroskademux: Return FALSE from TOC query if no TOC exists instead of an empty TOC 2012-06-25 09:47:59 +02:00
Tim-Philipp Müller
296783908c matroska: update for GstToc API changes 2012-06-24 22:51:16 +01:00
Tim-Philipp Müller
456847c66b rtspsrc: update for gst_element_make_from_uri() changes 2012-06-23 14:57:28 +01:00
Wim Taymans
30d3dfee36 update for task api change 2012-06-20 10:33:42 +02:00
Wim Taymans
dc04908412 update for clock api changes 2012-06-20 10:01:57 +02:00
Matej Knopp
c55e492e80 matroska-demux: Send gap events for subtitle streams 2012-06-19 11:21:52 +01:00
Tim-Philipp Müller
b6da022417 splitfilesrc: fix up docs for 0.11 2012-06-17 01:00:40 +01:00
Tim-Philipp Müller
3b94e44571 splitfilesrc: small uri handler fixup and some more docs 
Get URI location using gst_uri_get_location(), so any
escaped bits get unescaped.

https://bugzilla.gnome.org/show_bug.cgi?id=609049
 2012-06-17 00:59:54 +01:00
Tim-Philipp Müller
1d659d8e41 splitfilesrc: re-port to 0.11 2012-06-17 00:59:21 +01:00
Bastien Nocera
9b13a29f91 splitfilesrc: Implement splitfile:// URI scheme 
https://bugzilla.gnome.org/show_bug.cgi?id=609049

Conflicts:

	gst/multifile/gstsplitfilesrc.c
 2012-06-17 00:58:54 +01:00
Wim Taymans
540245894f theoradepay: fix buffer memory 
The memory was added to the input buffer instead of the output buffer.
 2012-06-14 10:43:56 +02:00
Wim Taymans
694be55c05 rtspsrc: Don't reset time in flush-stop 
Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.
 2012-06-14 08:58:58 +02:00
Vincent Penquerc'h
fe45881a0f deinterlace: send QoS messages when dropping a frame 
https://bugzilla.gnome.org/show_bug.cgi?id=657941
 2012-06-12 15:40:37 +01:00
Wim Taymans
935472aba7 rtspsrc: Rework the async state handling 
Always send the flushing events to the udp elements now that basesrc supports
this. This makes sure a segment event is sent correctly after a flush.
Keep track of the currently executing command and make it possible to specify
what command you want to cancel when starting a new async command.

See https://bugzilla.gnome.org/show_bug.cgi?id=677905
 2012-06-12 16:05:40 +02:00
Stefan Sauer
ea17c457f9 childproxy: update api use 2012-06-11 18:24:20 +02:00
Mark Nauwelaerts
8b1da8adb2 matroskademux: always perform full seek if seek is flushing 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677838
 2012-06-11 13:12:26 +02:00
Tim-Philipp Müller
17b422137a rndbuffersize: printf format fix for long -> int change 2012-06-11 11:20:18 +01:00
Tim-Philipp Müller
98e415dc9d debug: change rndbuffersize properties from long to int 
These should all be int instead of long, to avoid bugs
when passing these as varargs with g_object_set(), and
there was no reason to use long in the first place here.
Fixes FIXME.
 2012-06-09 16:53:54 +01:00
Sebastian Dröge
a1948e34d2 elements: Use gst_pad_set_caps() instead of manual event fiddling 2012-06-08 15:54:42 +02:00
Wim Taymans
f65495d405 update for audio api change 2012-06-08 10:11:12 +02:00
Wim Taymans
eb982e4bbe rtspsrc: only reset the manager object when we did a seek 
Only reset the manager object when we used a Range header, ie. when we did a
seek. Otherwise we just paused and we can resume just fine.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677475
 2012-06-07 12:11:14 +02:00
Sebastian Dröge
91ca34a0bb matroskademux: Update for TOC event API change 2012-06-06 14:17:08 +02:00
Wim Taymans
b5df4f0e62 update for tag event change 2012-06-06 13:02:12 +02:00
Wim Taymans
37df608fdc fix Y800 format 2012-06-06 13:00:58 +02:00
Thiago Santos
78ec03e32f Some printf variable format fixes 
The osx compiler complains about those
 2012-06-05 17:53:57 -03:00
Sebastian Dröge
ca4b5d795b audioparsers: Fix GstBaseParse::get_sink_caps() implementations 
They should take the filter caps into account and always return
the template caps appended to the actual caps. Otherwise the
parsers stop to accept unparsed streams where upstream does not
know about channels, rate, etc.

Fixes bug #677401.
 2012-06-05 09:21:08 +02:00
Wim Taymans
b8c08838bb qtdemux: set the palette size correctly 2012-05-31 13:44:46 +02:00
Wim Taymans
72b7d4884f video: remove duplicate format 2012-05-29 17:52:11 +02:00
Edward Hervey
5294edded2 flvdemux: Post error message if EOS before pads were created 
Happens with some files with only headers
 2012-05-29 16:59:06 +02:00
Tim-Philipp Müller
3986174aa9 flv, matroska: don't use GstStructure API on tag lists 2012-05-27 00:02:08 +01:00
Edward Hervey
923be8a85b rtpmp2tdepay: Only output integral mpeg-ts packets 
From RFC 2250

2. Encapsulation of MPEG System and Transport Streams
...
   For MPEG2 Transport Streams the RTP payload will contain an integral
   number of MPEG transport packets.  To avoid end system
   inefficiencies, data from multiple small MTS packets (normally fixed
   in size at 188 bytes) are aggregated into a single RTP packet.  The
   number of transport packets contained is computed by dividing RTP
   payload length by the length of an MTS packet (188).
....

Since it needs to contain "an integral number of MPEG transport packets", a
simple fix is to check that's the case, and strip off any leftover data.

Fixes #676799

Conflicts:

	gst/rtp/gstrtpmp2tdepay.c
 2012-05-26 12:04:54 +02:00
Alessandro Decina
51c8cd805d matroskademux: increase NEWSEGMENT accuracy after seeking 
demux->common.segment is populated during seek handling with the target
start/stop positions. Don't override them when sending out a NEWSEGMENT.

Conflicts:

	gst/matroska/matroska-demux.c
 2012-05-24 14:31:55 +02:00
Alessandro Decina
66d95d808c matroskademux: don't discard the incoming seek segment on push based seeking 
The incoming seek segment was being discarded leading to push based seeking
being potentially inaccurate.
 2012-05-24 14:26:23 +02:00
Luis de Bethencourt
c81fff0471 rtp: fix build issue in gstrtph264pay.c 2012-05-24 09:29:25 +01:00
Jonas Holmberg
7bf3a1bf95 rtph264pay: Add unrestricted caps 
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.

Fixes bug #672019
 2012-05-24 10:01:19 +02:00
Maria Giovanna Chiossa
ff019d05f6 rtsp: add the Scale header when needed 
Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should
set the "Scale" field in the rtsp PLAY header.
Because the boolean "src->skip" is set after the call, "Speed" instead
of "Scale" is always set. Move the assignment before issuing the _play
request.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618
 2012-05-24 09:57:31 +02:00
Sreerenj Balachandran
f400a06ba5 videobox: Fix the sample pipeline. 2012-05-23 10:14:16 +02:00
Anton Novikov
eba7494ab0 icydemux: warning if setting srcpad caps fails 2012-05-23 10:05:41 +02:00
Anton Novikov
6c31088adc icydemux: activate srcpad before setting caps 
Before gst_pad_set_active() is called, the pad has
FLUSHING flag set, so setting the caps fails
 2012-05-23 10:04:09 +02:00
Thiago Santos
46083803d7 avimux: fix assertion when handling a date tag as a string 
Date tags are GDate, not strings. Add a special case to convert
it to the exif date format representation in string to avoid
the assertion
 2012-05-21 10:34:20 -03:00
Mark Nauwelaerts
182596b3ab rtpmp2tpay: respect mtu and packet boundaries 
See #659915.
 2012-05-18 12:53:44 +02:00
Youness Alaoui
7703a11073 rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data 
This allows some cameras (Logitech C920) that specify different quant
tables but both with the same data, to work.
Bug reported by Robert Krakora
 2012-05-16 09:49:08 +02:00
Tim-Philipp Müller
aef0ad44d4 rndbuffersize: only send flush-stop if it was a flushing seek 2012-05-09 15:14:55 +01:00
Tim-Philipp Müller
338286cedf rndbuffersize: must send flush-stop after acquiring the stream lock 
Otherwise the streaming thread might just keep on going and we
might never get the stream lock.
 2012-05-09 12:24:37 +01:00
Tim-Philipp Müller
7e03f5f004 rndbuffersize: port seeking code to 0.11 2012-05-09 11:39:34 +01:00
Tim-Philipp Müller
84c842cfe9 rndbuffersize: add support for seeks 
Useful for e.g. filesrc ! rndbuffersize ! queue2 ! ...
 2012-05-09 11:39:34 +01:00
Tim-Philipp Müller
920e91e072 rndbuffersize: send SEGMENT event before pushing buffers 
Conflicts:

	gst/debugutils/rndbuffersize.c
 2012-05-09 11:39:34 +01:00
Wim Taymans
354e35a6ee interleave: fix compilation again 2012-05-09 11:19:10 +02:00
Pascal Buhler
8161daef4a rtpsession: creation should be signaled before validation 
https://bugzilla.gnome.org/show_bug.cgi?id=667850
 2012-05-09 10:36:18 +02:00
Alban Browaeys
a56361623c isomp4: set layout=interleaved on raw audio caps 
This fixes a not-negotiated error at least on mov files with
twos audio with two channels and video dvcp. As playbin and gst-launch
sample coming from the qtdemux.c file uses audioconvert and the latter
require format interleaved.

https://bugzilla.gnome.org/show_bug.cgi?id=675326
 2012-05-03 23:28:50 +01:00
Tim-Philipp Müller
2d249dcc29 videomixer: change sink pad template name from sink_%d to sink_%u 2012-05-01 18:58:03 +01:00
Wim Taymans
01db5dbff0 interleave: handle EOS on all pads 
When all pads go to EOS immediately, we are not negotiated and our collected
function is called (without any available data). Handle this case gracefully.

Conflicts:

	gst/interleave/interleave.c
 2012-05-01 13:35:56 +02:00
Wim Taymans
e0636feff8 interleave: improve debugging 2012-05-01 13:34:32 +02:00
Tim-Philipp Müller
b072c78270 alpha: don't set up stuff before the input and output formats are known 
Fixes crash on startup.
 2012-05-01 00:23:14 +01:00
Peter Seiderer
175f666293 multifilesink: don't write stream header twice for first file 2012-04-30 22:53:42 +01:00
Peter Seiderer
7112b93a97 multifilesink: fix buffer list size calculation in render_list 
Fix uninitialized 'size' variable in call to gst_buffer_list_foreach().
 2012-04-30 22:00:59 +01:00
Luis de Bethencourt
54c63dac31 multifile: unnecessary size check 2012-04-30 21:58:00 +01:00
Luis de Bethencourt
c7f124c8a8 avi: fix build errors 
fix redundant declarations
and also style/indent issues
 2012-04-30 21:30:56 +01:00
Vincent Penquerc'h
93ce50f9b9 matroska: implement forward snapping keyframe seeking 
Requires an index.
 2012-04-30 10:37:57 +01:00
Vincent Penquerc'h
cfd0da4146 avi: implement forward snapping keyframe seeking 
In pull mode with an index.
 2012-04-30 10:20:40 +01:00
Tim-Philipp Müller
9c236b290d matroska: update for media type changes 2012-04-28 19:57:51 +01:00
idc-dragon
e0945d0a2d celtdepay: calculate size correctly 
The summation was done wrong, causing the de-payloader to exit its loop too
early, before all frames are processed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472
 2012-04-25 10:29:56 +02:00
Chris Pankow
6042bb1e6b audiofxbasefirfilter: Fix time-domain convolution for multichannel input 
Fixes bug #674025.
 2012-04-23 10:08:59 +02:00
Wim Taymans
ad5c3cd3dd multipartdemux: first activate pad then set caps 2012-04-20 16:49:56 +02:00
Wim Taymans
fcfe6d9e28 matroskamux: set caps on srcpad 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674219
 2012-04-20 13:35:35 +02:00
Sebastian Dröge
04b70571e5 video: Update for libgstvideo API changes 2012-04-19 12:20:59 +02:00
Mark Nauwelaerts
67e168aef4 collectpads2: rename to collectpads 2012-04-17 15:14:27 +02:00
Mark Nauwelaerts
04b4d30f2c misc: chain up to collectpads event handler 2012-04-16 16:37:49 +02:00
Mark Nauwelaerts
6d9a84b1cf smpte: use some more boilerplate 2012-04-13 17:24:38 +02:00
Mark Nauwelaerts
93f61c47b9 flxdec: improve segment handling 
... to send a proper TIME segment downstream.
 2012-04-13 17:24:38 +02:00
Mark Nauwelaerts
40cfe6787b flxdec: port to 0.11 2012-04-13 17:24:38 +02:00
Mark Nauwelaerts
64045ba909 videobox: adjust to deprecated GMutex setup 2012-04-13 17:24:38 +02:00
Mark Nauwelaerts
edf3139e22 videobox: port to 0.11 2012-04-13 17:24:38 +02:00
Mark Nauwelaerts
8bf26fa7dc alpha, smpte: adjust to removed color-matrix caps field 2012-04-13 17:24:38 +02:00
Sebastian Dröge
d99eb6d2cb Update everything for the removal of the interface library and mixer/tuner interfaces 2012-04-13 13:15:11 +02:00
Edward Hervey
71fc25849e rtp: Use unchecked variant of GstByteWriter where applicable 
The size was checked before
 2012-04-12 15:50:16 +02:00
Edward Hervey
4aef223db0 matroska: Check return value of GstByteReader/Writer 2012-04-12 15:49:44 +02:00
Edward Hervey
97591c1e77 isomp4: Check return value of GstByteWriter 
And use unchecked variant of GstByteReader where applicable
 2012-04-12 15:48:57 +02:00
Edward Hervey
eb0cdfe20f flvdemux: Use unchecked variant of GstByteReader 
We know there's at least 7 bytes (checked above)
 2012-04-12 15:48:00 +02:00
Edward Hervey
4bd694d2cd avi: Check return value of GstByteWriter 2012-04-12 15:47:49 +02:00
Edward Hervey
ba7569028c audioparsers: Check return value of GstBitReader/GstByteReader 2012-04-12 15:47:24 +02:00
Sebastian Dröge
4784e83938 Release 0.11.90 2012-04-12 10:27:31 +02:00
Mark Nauwelaerts
ea397f60e4 Merge remote-tracking branch 'origin/0.10' 
Conflicts:
	gst/flv/gstflvdemux.c
	gst/matroska/matroska-demux.c
 2012-04-10 11:57:53 +02:00
Mark Nauwelaerts
dfda34ea24 matroskademux: some more segment handling tweaking 2012-04-10 11:38:08 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Tim-Philipp Müller
fa5edd2680 interleave: make channel-poisitions property a GValueArray again 
Or perhaps it should just be a guint64 channel mask, which would
be nicer in C, but more awkward for bindings (even more so since
we can't add a flags type for it, since that only supports guint
size flags). Fixes wavenc unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=669643
 2012-04-09 11:13:05 +01:00
Mark Nauwelaerts
e90c67b3a9 matroskademux: cleanly initialize and set needed segment 
Fixes #673165.
 2012-04-06 16:12:36 +02:00
Nicolas Dufresne
628816784f flvdemux: Fix threading issue in index handling 2012-04-06 09:15:13 +02:00
Sebastian Dröge
acca0e77f1 flvdemux: Don't use static variables to hold index associations 
This not really threadsafe in any way.
 2012-04-06 09:14:28 +02:00
Mark Nauwelaerts
31edc9f7c0 updsrc: clear error 2012-04-05 19:17:29 +02:00
Sebastian Dröge
9c8944ca89 gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 18:02:56 +02:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
a2ac7554ee gst: Update versioning 2012-04-04 14:44:34 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans
cdb905efe0 avidemux: avi only knows about DTS 
Only set DTS on outgoing buffers unless we have a keyframe and then we can set
the PTS to DTS as well.
 2012-04-03 11:50:00 +02:00