Commit graph

8003 commits

Author SHA1 Message Date
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
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
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
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
Mark Nauwelaerts
d0587467fc avidemux: resurrect some flow return handling 2015-03-07 20:22:33 +01:00
Nicolas Huet
5ead23a14a aacparse: fix LOAS parsing issue
Fix missing index in syncword searching

https://bugzilla.gnome.org/show_bug.cgi?id=745585
2015-03-06 14:34:08 -03:00
Jan Schmidt
b0ce43cde3 splitmuxsink: Protect property variables with the object lock.
Use the object lock instead of the splitmux lock to protect
internal property variables, so they're not locked when
switching to a new file.

https://bugzilla.gnome.org/show_bug.cgi?id=744420
2015-03-07 00:55:47 +11:00
Sebastian Dröge
c34a7cb90d rtspsrc: Fix handling of interleaved (TCP) streams
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.

https://bugzilla.gnome.org/show_bug.cgi?id=745599
2015-03-05 12:15:04 +01:00
Sebastian Dröge
b4aaa11f97 rtspsrc: Don't unref caps we don't own 2015-03-05 09:56:37 +01:00
Sebastian Dröge
297d808acc rtspsrc: Push RTCP caps on the RTCP pads
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
2015-03-05 09:47:29 +01:00
Sebastian Dröge
788074733c rtspsrc: Make sure to send SEGMENT events on all pads 2015-03-05 09:47:29 +01:00
Santiago Carot-Nemesio
e05378ec16 rtp: Add Full Intra Request (FIR) packets to statistics
https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:40 +01:00
Santiago Carot-Nemesio
22791413f9 rtp: Add Packet Loss Indication (PLI) to statistics
This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.

https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:07 +01:00
Nicola Murino
c4e542de69 matroskamux: Remove duration accumulation logic
Duration accumulation can cause rounding errors and generate wrong
duration with different buffers that share the same timestamp.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:37:48 +01:00
Nicola Murino
f727762c1f matroska: Add an helper method to get buffer timestamps
... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
that return PTS or DTS based on stream type.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:36:24 +01:00
Sebastian Dröge
8984e18ef7 rtpsession: Add explanation why we have space for 32 hash tables
And also create only one, there's no need yet to create all 32 until
we implement RFC2762.
2015-03-04 11:30:43 +01:00
Sebastian Dröge
af2bdd6e15 Revert "rtpsession: Do not use an array of maps if they are not being used"
This reverts commit 1591adf4cd.

https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
2015-03-04 11:26:57 +01:00
Santiago Carot-Nemesio
1591adf4cd rtpsession: Do not use an array of maps if they are not being used
rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession

https://bugzilla.gnome.org/show_bug.cgi?id=745586
2015-03-04 11:25:30 +01:00
Jimmy Ohn
42599eab76 avidemux: remove not needed code
In gst_avi_demux_handle_src_query, there is not needed code.
We already check about stream is vbr or not at the upper line.
o, we don't need to check this condition becase stream is not
vbr 100% in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=745276
2015-03-04 10:08:21 +01:00
Matej Knopp
f75e443a7a qtdemux: fix key unit seek
Unlike many other seek flags, the KEY_UNIT seek
flag is not copied over into the GstSegment,
since it's only relevant for the seek itself,
so we need to pass it explicitly to the seek
handler here.

https://bugzilla.gnome.org/show_bug.cgi?id=745339
2015-03-01 13:06:55 +00:00
Nicola Murino
e676b8ba9c matroskamux/demux: initialize dts_only
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Nicola Murino
09b8f0efc3 matroskamux: store DTS for V_MS/VFW/FOURCC streams
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Tim-Philipp Müller
f5b511b42b multifile: attempt to fix docs build issue on build bot 2015-02-26 19:48:33 +00:00
Arun Raghavan
0c06553fb2 interleave: Drop custom latency query handling
This is implemented by the default query handler now.
2015-02-27 00:59:43 +05:30
Arun Raghavan
dbc142afec videomixer: Drop custom latency querying logic
This is now implemented in the default latency query handler.
2015-02-27 00:59:43 +05:30
Sebastian Rasmussen
d331d931db rtpvorbispay: fix payloader description and author e-mail
https://bugzilla.gnome.org/show_bug.cgi?id=745226
2015-02-26 15:57:08 +00:00
Matej Knopp
fa283f407f matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
When such stream is present demuxer should set DTS on buffers instead
of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
streams.

Sample file
https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-26 11:12:34 +02:00
Krzysztof Kotlenga
e3ca4d1c86 rtspsrc: improve error message when unauthorized
Make use of NOT_AUTHORIZED error code instead of falling back to generic
READ error.

https://bugzilla.gnome.org/show_bug.cgi?id=601733
2015-02-24 11:08:27 +02:00
Thibault Saunier
fa0870658d qtdemux: All segment resulting from a seek should have the same seqnum
https://bugzilla.gnome.org/show_bug.cgi?id=744983
2015-02-23 20:05:20 +01:00
Vincent Penquerc'h
dc73d153cb rtpvp8pay: default encoding name to VP8
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:29:02 +00:00
Vincent Penquerc'h
b88ea286d2 rtpvp8pay: make caps writable before truncating them
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:06:51 +00:00
Vincent Penquerc'h
b866c989f5 rtpvp8pay: negotiate encoding name
Chrome uses a different one than gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 13:52:29 +00:00
Sebastian Dröge
939a95d44b rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
Otherwise we will just send buffers on the pad without any events beforehand
and will get g_warnings() about that.
2015-02-19 13:34:47 +02:00
Thiago Santos
84b7cf6795 qtmux: remove not needed condition
gst_buffer_replace can handle NULL inputs by itself
2015-02-18 10:36:06 -03:00
Thiago Santos
a12e41c106 qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
The tfdt should be more accurate as the buffer timestamp is provided
by the fragmented format manifest and it might just be an approximation.
2015-02-18 09:57:48 -03:00
Sebastian Dröge
735c6c40f8 rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
confuse downstream with buffers that come before such events.
2015-02-17 16:57:55 +02:00
Edward Hervey
6798dc7912 isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
2015-02-17 12:31:06 +01:00
Luis de Bethencourt
ea1d67abe3 goom2k1: use fractional part of float division 2015-02-16 14:31:05 +00:00
Luis de Bethencourt
4af5a2b760 splitmuxsin: remove dead code
Every instance of goto beach has buf_info equal NULL. Don't check
for a condition that never happens.

CID #1268399
2015-02-16 13:59:17 +00:00
Nicolas Dufresne
b8142bde07 spectrum: Fix min and max for bands property
The number of FFTs is calculated with the following formula:

  guint nfft = 2 * bands - 2;

nfft is passed to gst_fft_f32_new() as the len argument and is of type
unsigned integer. This method required that len is at leas 1, then
maximum G_MAXINT, as other values would be negative. If we extrapolate
from the formula above it means we need "bands" to be between 2 and
((guint)G_MAXINT + 2) / 2).

https://bugzilla.gnome.org/show_bug.cgi?id=744213
2015-02-15 21:34:28 -05:00
Thiago Santos
afa5481c50 qtdemux: do not use sparse streams in push-based seeking
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
 if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.

https://bugzilla.gnome.org/show_bug.cgi?id=742661
2015-02-14 11:36:11 -03:00
Tim-Philipp Müller
3f5b690e78 splitmuxsink: flag as sink from the start 2015-02-13 20:40:48 +00:00