Commit graph

13860 commits

Author SHA1 Message Date
Sebastian Dröge
d2ab5199bc amrparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
6f89b430ea ac3parse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
b3abbe3f5e aacparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Sebastian Dröge
c07424a534 mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
Olivier Crête
ada6ea668b rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78 rtpsession: Only count nacks when a nack packet is received
Not when any RTCP feedback packet is.
2013-12-13 16:08:35 -05:00
Olivier Crête
dc845c1899 tests: Initialize segment in rtpcollision test 2013-12-13 16:05:41 -05:00
Olivier Crête
7af9fdbca6 rtpsession: Process PSFB FIR requests which lack the media ssrc
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.

Fixes a regression introduced by commit 57c27ec3
2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.

This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
2013-12-12 16:44:27 +01:00
George Kiagiadakis
f9b7f44938 tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin
This test checks that when we have multiple internal sender sources
in rtpsession, SRs contain RBs for every other sender source, and that
they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
which is the max number of RBs that can fit in a SR.
2013-12-12 16:02:56 +01:00
Wim Taymans
b4ebf8ca50 docs: improve docs 2013-12-12 16:01:10 +01:00
Julien Isorce
a0df8b8fce doc: add design-rtpcollision.txt that explains when GstRTPCollision is created
It also talks about "BYE only the corresponding source, not the whole session."
2013-12-12 15:46:24 +01:00
Julien Isorce
d562263852 tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session
Conflicts:
	tests/check/elements/rtpcollision.c
2013-12-12 15:39:40 +01:00
Julien Isorce
7b001e35ed tests/check: add rtpcollision::test_master_ssrc_collision unit test
It checks the payloader changes its ssrc when collision happens
2013-12-12 15:39:39 +01:00
George Kiagiadakis
c78a115154 rtpsession: keep extra stats for scheduling BYE
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
2013-12-12 10:38:43 +01:00
George Kiagiadakis
282028e753 rtpsession: when we schedule BYE, only deal with BYE sources
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81 rtpsession: reset state after scheduling BYE
After we do RTCP, we are not scheduling bye anymore.
2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef rtpsession: also count NACKS when no signal was pending 2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0 session: ignore RTCP packets for the BYE sources
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
2013-12-12 10:09:25 +01:00
Julien Isorce
33b398e345 rtpsession: determine if the session is doing point-to-point
In this case T_dither_max is set to 0 according to RFC 4585
2013-12-10 16:57:56 +01:00
Wim Taymans
eee515cb2c rtpjitterbuffer: serialize events in the buffer
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00
Wim Taymans
36e78bc5ca rtpjitterbuffer: detect -1 seqnum
Keep the seqnum as a full guint so that we can check for -1 entries and
deal with them correctly.
Immediately try to push -1 seqnum.
2013-12-10 11:04:06 +01:00
Wim Taymans
4a2e0f4ff4 rtpjitterbuffer: reorganize jitterbuffer items
Keep the oldest item at the head and the newest items on the tail. This
makes it easier to deal with -1 seqnums.
2013-12-10 11:01:03 +01:00
Wim Taymans
ea2a222cef jitterbuffer: correctly check for invalid values
Check for -1 on the guint from the buffer item instead of on the guint16
or guint32.
Also insert -1 seqnum at the head of the jitterbuffer.
2013-12-09 23:34:10 +01:00
Alessandro Decina
fdf8ac40d8 osxvideosink: fix segfault when dealing with padded frames
Fixes crashes with vtdec ! osxvideosink where VideoToolbox outputs padded UYVY
2013-12-08 16:51:00 +01:00
Sebastian Dröge
f3c3dee148 mulawdec: Require caps to be set before accepting any data 2013-12-05 12:15:29 +01:00
Sebastian Dröge
d9deb95012 wavpackdec: Require caps to be set before accepting any data 2013-12-05 12:15:19 +01:00
Sebastian Dröge
c880e36779 speexdec: Require caps to be set before accepting any data 2013-12-05 12:13:33 +01:00
Sebastian Dröge
774e56b051 flacdec: Require caps to be set before accepting any data 2013-12-05 12:13:10 +01:00
Sebastian Dröge
5c8163a685 vpx: Use new gst_video_decoder_set_needs_format() API 2013-12-05 11:42:34 +01:00
Olivier Crête
dc782af133 pulsesink: Free device_info in accepts caps
https://bugzilla.gnome.org/show_bug.cgi?id=719811
2013-12-04 16:24:06 -05:00
Sebastian Dröge
d585bd7bbd rtptheorapay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d105de6e0f rtptheorapay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
0915d696c7 rtpvorbispay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
967280df42 rtpvorbispay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d87f6cf483 rtpstreamdepay: Add RFC4571 RTP stream depayloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Sebastian Dröge
c5284dc047 rtpstreampay: Add RFC4571 RTP stream payloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Thiago Santos
1fd094d96b qtdemux: improve fragment-start tracking
Some buffers can have multiple moov atoms inside and the strategy
of using the gst_adapter_prev_pts timestamp to get the base timestamp
for the media of the fragment would fail as it would reuse the same
base timestamp for all moofs in the buffer instead of accumulating
the durations for all of them.

Heres a better explanation of the issue:
qtdemux receives a buffer where PTS(buf) = X
buf -> moofA | moofB | moofC

The problem was that PTS(buf) was used as the base timestamp for
all 3 moofs, causing all buffers to be X based. In this case we want
only moofA to be X based as it is what the PTS on buf means, and the
other moofB and moofC just use the accumulated timestamp from the
previous moofs durations.

To solve this, this patch uses gst_adapter_prev_pts distance
result, this allows qtdemux to calculate if it should use the
resulting pts or just accumulate the samples as it can identify
if the moofs belong to the same upstream buffer or not.

https://bugzilla.gnome.org/show_bug.cgi?id=719783
2013-12-04 10:36:38 -03:00
Julien Isorce
3c70741e45 v4l2bufferpool: add support for multi-planar V4l2 API in DMABUF mode
Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
2013-12-04 09:28:25 +00:00
Julien Isorce
303cec48db v4l2: refactor by emulating one v4l2_plane in non-MPLANE mode
so that the buffer informations can be retrieved the same way
in both MPLANE and non-MPLANE mode.

Here "emulating" means "manually fill in the plane".

Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
2013-12-04 09:28:08 +00:00
Julien Isorce
61ae84b50d v4l2: add support for multi-planar V4L2 API
This api is in linux kernel since version 2.6.39,
and present in all version 3.

The commit that adds the API in master branch of the
linux kernel source is:
f8f3914cf9

v4l2 doc: "Some devices require data for each input
or output video frame to be placed in discontiguous
memory buffers"

There are newer structures 'struct v4l2_pix_format_mplane'
and 'struct v4l2_plane'.
So the pixel format is not setup with the same API when using
multi-planar.

Also for gst-v4l2, one of the difference is that in GstV4l2Meta
there are now one mem pointer for each maped plane.

When not using multi-planar, this commit takes care of keeping
the same code path than previously. So that the 2 cases are
in two different blocks triggered from V4L2_TYPE_IS_MULTIPLANAR.

Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
2013-12-04 09:27:20 +00:00
Wim Taymans
0d55724a2b audioparsers: don't leak template caps 2013-12-04 09:12:07 +01:00
Wim Taymans
e0a5c07e8d audioparsers: use ACCEPT_INTERSECT flag
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.

This reverts commit 26040ee38c

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
2013-12-03 22:26:44 +01:00
Wim Taymans
e3f393f7e6 audioparsers: remove fields from filter
We need to remove the fields from the filter when we can convert
between them.
2013-12-03 21:39:57 +01:00
Wim Taymans
e8313a1e70 audioparsers: refactor code to remove caps fields 2013-12-03 21:29:13 +01:00
Tim-Philipp Müller
a424fb289b deinterlace: microoptimisation: avoid some unnecessary GValue copies 2013-12-02 00:10:43 +00:00
Tim-Philipp Müller
63b0e84add deinterlace: fix off-by-one crash when downstream caps contain a list of framerates
https://bugzilla.gnome.org/show_bug.cgi?id=719544
2013-12-01 23:33:04 +00:00
Thiago Santos
079dde49ed qtdemux: Use the timestamp of the moof as the base fragment start
In SmoothStreaming fragmented scenario, the timestamps are calculated
starting from the fragment buffer timestamp. When there is a not-linked
return from downstream, qtdemux will return upstream and will keep the
non-pushed data into its adapter.

On a new fragment buffer pushed to qtdemux, the new buffer timestamp
would overwrite the previous one that should be used on the still
to be pushed buffers. Because of this, this patch will also
update the fragment_start timestamp from the adapter last pts
to make sure the moof and timestamps are in sync and will result
in correct timestamps for all fragments.
2013-11-29 17:28:48 -03:00
Thiago Santos
45c16599ff qtdemux: avoid re-reading the same moov and entering into loop
In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
could read the moov again after the mdat because it was considering the
media as a fragmented one.

To avoid this loop this patch makes it store
the last processed moov_offset to avoid parsing it again.
And it also checks if there are any samples to play before
resturning to the mdat, so that it knows there is new data to be played.

https://bugzilla.gnome.org/show_bug.cgi?id=691570
2013-11-29 17:28:48 -03:00
Thiago Santos
fcc78aa3bd qtdemux: do not free streams if they were not created locally
When parsing a trak only free streams on failures if those streams
were created locally. They could have been created from a previous
fragment, in this case we they have valid info from the other fragment.
Including pads.

https://bugzilla.gnome.org/show_bug.cgi?id=691570
2013-11-29 17:28:48 -03:00