Commit graph

14941 commits

Author SHA1 Message Date
Nicolas Dufresne
84725d62b5 rtspsrc: Fix segment in TCP mode
It is expected that buffers are time-stamped with running time. Set
a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
would do. Depayloaders will update the segment to reflect the playback
position.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
2015-03-26 17:54:08 -04:00
David Schleef
c3bb399fd3 wavparse: be more strict about typefinding DTS
Code now matches comments.
2015-03-26 12:22:43 -07:00
Nicolas Dufresne
32aed67144 rtspsrc: Remove useless function
This function didn't do anything special, let's not use a function for
that.
2015-03-25 15:28:24 -04:00
Nicolas Dufresne
12762ad1a5 rtpjitter: Account for rtx_retry in overflow check
As rtx_retry is part of the substraction, we need to take it into
account, otherwise we may endup with a big value.
2015-03-25 15:25:56 -04:00
Julien Isorce
d63c163335 osxvideosink: check for deprecated constants prior to OSX 10.10
cocoawindow.m:339:5: error: 'NSOpenGLPFAWindow'
is deprecated: first deprecated in OS X 10.9

cocoawindow.m:576:7: error: 'NSOpenGLPFAFullScreen'
is deprecated: first deprecated in OS X 10.6

cocoawindow.m:605:24: error: 'setFullScreen'
is deprecated: first deprecated in OS X 10.7
2015-03-24 23:16:26 +00:00
Nicolas Dufresne
8afc8c8f3b rtspsrc: Fix seeking query
The segment start/stop in the query is meant to represent the seekable
portion of the stream. It does not match the segment start/stop. Instead
export 0 to duration.
2015-03-24 16:51:12 -04:00
Sebastian Dröge
ac0141b6a0 flvdemux: Only set caps once if they don't change
Previously we were setting new caps with the same content for every H264 or
AAC codec_data we found in the stream, spamming everything and causing
renegotiations.
2015-03-24 16:18:53 +01:00
Sebastian Dröge
c9b42951fe flvdemux: Don't create AAC/H264 caps without codec_data
Instead delay creating the caps until we read the codec_data from the stream,
or fail if we get normal data before the codec_data.

AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad
without them is going to make negotiation fail most of the time. Even if we
later set new caps with the codec_data, that's usually going to be too late.

https://bugzilla.gnome.org/show_bug.cgi?id=746682
2015-03-24 16:15:04 +01:00
Sebastian Dröge
5e88b53212 flvdemux: Fix indention 2015-03-24 15:39:40 +01:00
Ilya Konstantinov
6af516b21f osxaudio: Fix string format warning on 32-bit
UInt32 (Darwin, not C99's uint32_t) is 'unsigned long' on 32-bit
platforms.
2015-03-23 19:48:43 +05:30
Sebastian Dröge
0e3609a6e1 rtpsession: Fix another instance of sticky event misordering warnings
Make sure that the sync_src pad has caps before the segment event.
Otherwise we might get a segment event before caps from the receive
RTCP pad, and then later when receiving RTCP packets will set caps.
This will results in a sticky event misordering warning

This fixes warnings in the rtpaux unit test but also in the
rtpaux and rtx examples in tests/examples/rtp

https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-03-21 19:30:32 +01:00
Sebastian Dröge
17d90b453f rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
Before we only started it when either:
- there is no send RTP stream
or
- we received an RTP packet for sending

This could mean that if the send RTP pads are connected but never receive any
RTP data, and the same session is also used for receiving RTP/RTCP, we would
never start the RTCP thread and would never send RTCP for the receiving part
of the session.

This can be reproduced with a pipeline like:

gst-launch-1.0 rtpbin name=rtpbin \
udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v

Before this change the rtcp_fakesink would never send RTCP for the receiving
part of the session (i.e. no receiver reports!), after the change it does.

And before and after this change it would send RTCP for the receiving part of
the session if the sender part was omitted (the last two lines).
2015-03-21 17:38:07 +01:00
Sebastian Dröge
1018aacb35 rtprtxsend: Add support for buffer lists 2015-03-19 11:54:37 +01:00
Sebastian Dröge
57ff27f8c8 rtprtxqueue: Implement support for buffer lists 2015-03-19 11:54:37 +01:00
Nicolas Dufresne
1c27002ebd rtspsrc: Improve trace readability
Change the command number into strings.
2015-03-18 17:32:36 -04:00
Jan Alexander Steffens (heftig)
be8e3196a3 flvdemux: Don't repeatedly warn after no_more_pads (v2)
This can get rather spammy for such a high log level.
Only warn once per stream.

https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Jan Alexander Steffens (heftig)
ac8a272381 flvdemux: Introduce constant for no-more-pads threshold
https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Jan Alexander Steffens (heftig)
f2a1f74cec flvdemux: Fix warning to contain 'video'
https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Nicola Murino
bb3d82ef04 matroskademux: for dts only stream set pts=dts for intra only formats
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-15 14:28:36 +00:00
Ramiro Polla
0fad053497 matroskademux: fix sending of tags
* Fix critical when new tags are found after segment event has already
  been sent.
* Send global tags before stream tags.
* Split sending of tags out of gst_matroska_demux_send_event() into its
  own function.

https://bugzilla.gnome.org/show_bug.cgi?id=745973
2015-03-14 18:17:48 +00:00
Ramiro Polla
90be7b4e1e rtspsrc: properly escape percent sign in documentation 2015-03-14 14:22:39 +00:00
Ramiro Polla
63944753b0 rtpdtmfmux: properly escape percent sign in documentation 2015-03-14 14:22:26 +00:00
Thiago Santos
0a945e7099 v4l2src: delay renegotiation until it is likely buffers were reclaimed
Allow renegotiation to happen when buffers have returned after an allocation
query. As the allocation query is serialized, all buffers from the pool
should have returned and we can stop it to create a new one for the
new format

https://bugzilla.gnome.org/show_bug.cgi?id=682770
2015-03-13 18:48:03 +00:00
Thiago Santos
6cfa6c0da8 v4l2object: add gst_v4l2_object_try_format
Similar to set_format but it uses TRY_FMT instead of S_FMT

https://bugzilla.gnome.org/show_bug.cgi?id=682770
2015-03-13 18:47:55 +00:00
Tim-Philipp Müller
3c595f308a multiudpsink: fix crash with GST_DEBUG enabled
g_inet_socket_address_get_address() does not give
us a ref to the address, so don't unref it.
2015-03-13 18:38:42 +00:00
Sebastian Dröge
7b90bf3215 level: Don't read over the end of the input memory
Previously we advanced the in_data pointer by bps for every channel, and then
later again for block_size*bps. This caused us to be one sample further than
expected if an input buffer covered two analysis frames. And in the end lead
to completely bogus values reported by level.

https://bugzilla.gnome.org/show_bug.cgi?id=746065
2015-03-12 13:51:56 +00:00
Jan Schmidt
c809bcd394 Remove a couple of superfluous trailing semi-colons 2015-03-12 01:37:08 +11:00
Tim-Philipp Müller
c4fa54da17 Fix double semicolons 2015-03-10 09:31:20 +00:00
Jan Schmidt
d441140cd6 splitmux: Shut down element before downward state change
Make sure the state change won't hang trying to shut down pads
by making sure the streaming has stopped before chaining up.
2015-03-10 15:49:33 +11:00
Ilya Konstantinov
b528c527b7 osxaudio: stream format is an SPDIF-only field 2015-03-10 09:14:57 +05:30
Ilya Konstantinov
7b365042f0 osxaudio: fix spaces 2015-03-10 09:14:57 +05:30
Ilya Konstantinov
8f62f50a98 osxaudio: add type check macro 2015-03-10 09:14:57 +05:30
Ilya Konstantinov
d450b1cac1 osxaudio: rename gst_core_audio_set_channels_layout()
to gst_core_audio_get_channel_layout().
2015-03-10 09:14:57 +05:30
Ilya Konstantinov
4637b3eb82 osxaudio: remove unused finalize 2015-03-10 09:14:57 +05:30
Luis de Bethencourt
3763f4057a vp9enc: remove duplicate declaration of function 2015-03-09 16:25:43 +00:00
Luis de Bethencourt
823194284c rtph264depay: remove unused value
CID #1226474
2015-03-09 16:22:33 +00:00
Luis de Bethencourt
5cd293fe76 rtph263pay: fix leak
CID 1212156
2015-03-09 16:17:45 +00:00
Luis de Bethencourt
e87113781a rtph263pay: remove uneeded variable
We just need to save the ebit information in case there is an error decoding.
2015-03-09 16:17:45 +00:00
Sebastian Dröge
a52e432fda vp[89]enc: Reset the encoder when flushing
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 16:47:57 +01:00
Luis de Bethencourt
db3ade5bfb matroska: error mode if can't push buffer
If gst_pad_push() fails, inform and return flow error.
2015-03-09 12:51:21 +00:00
Luis de Bethencourt
f494da89b4 matroska: unused value
Value set in ret will be overwritten just before exiting the function.

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

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2015-03-09 10:18:35 +01:00
Sebastian Dröge
38bf3d3808 rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams 2015-03-09 10:05:14 +01:00
Arun Raghavan
3751c87f00 pulsesink: Make sure to filter caps in all cases during CAPS query
We were skipping the filter step while returning template caps, for
example.
2015-03-09 11:55:40 +05:30
Nicolas Dufresne
eeb4d2e8b1 v4l2bufferpool: Don't update buffer for OUTPUT
For output device, we should not update the buffer with flags and
timestamp when we dequeue. The information in the v4l2_buffer is not
meaningful and it breaks the case where the buffer is rendered at
multiple places.

https://bugzilla.gnome.org/show_bug.cgi?id=745438
2015-03-08 21:15:53 +00:00
Sebastian Dröge
8965619f13 souphttpclientsink: Implement cookies property 2015-03-08 18:04:34 +01:00
Sebastian Dröge
b2bcb3d61f souphttpclientsink: Implement automatic-redirect property 2015-03-08 18:02:51 +01:00
Sebastian Dröge
814f741a28 souphttpclientsink: Implement proxy support
The properties were there before, but not used anywhere.
2015-03-08 17:54:42 +01:00
Mark Nauwelaerts
d0587467fc avidemux: resurrect some flow return handling 2015-03-07 20:22:33 +01:00