Commit graph

7372 commits

Author SHA1 Message Date
Matthieu Bouron
0bbdb9bb1d deinterlace: support any video formats and any caps features if deinterlace mode allows it
https://bugzilla.gnome.org/show_bug.cgi?id=719636
2014-01-03 11:22:01 +01:00
Wim Taymans
bb2d37b11d rtpbin: add some docs about AUX elements 2013-12-31 15:08:49 +01:00
Wim Taymans
d08e05b4ef rtpbin: add support for AUX sender and receiver
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.

The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
2013-12-31 15:08:48 +01:00
Wim Taymans
ae22c95881 rtpbin: make request_element method internally
We can use the same method to create encoder and decoder elements, they
are just internal elements that we create.
2013-12-31 15:08:48 +01:00
Stéphane Cerveau
e7912641c3 wavparse: Skip id3 tag
Skip id3 tag during wav parse.

https://bugzilla.gnome.org/show_bug.cgi?id=721241
2013-12-31 10:39:21 +01:00
Edward Hervey
711c73290c avimux: Add missing break
I guess no-one noticed we no longer could mux WMV3 ...

COVERITY CID 1139759
2013-12-30 17:23:22 +01:00
Edward Hervey
91c5b09fb4 rtpvrawpay: Add missing break
COVERITY CID 1139762
2013-12-30 17:20:37 +01:00
Wim Taymans
ee7f41ba2e rtpsession: internal-ssrc is no longer deprecated 2013-12-30 17:00:45 +01:00
Wim Taymans
e721d26c68 rtpbin: add Since tags 2013-12-30 16:59:20 +01:00
Wim Taymans
5a2bc1405e rtpbin: add signal for new jitterbuffer
Emit a signal when a new jitterbuffer is created so that the app can
have a chance to configure it.
2013-12-30 16:52:28 +01:00
Wim Taymans
3f3b2d0886 rtpbin: handle multiple encoder instances
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
2013-12-30 16:28:57 +01:00
Wim Taymans
05c8edc174 rtpbin: fix memory leaks 2013-12-30 15:17:05 +01:00
Wim Taymans
9345c2280a rtpbin: expect the pads on the encoders
Don't use request pads for the encoder elements, the signal handler
should request the pads and make sure they are available with the right
name.
2013-12-30 15:17:05 +01:00
Wim Taymans
cbc80d10dd rtpbin: request-rtp-encoder are no action signals
The request-rtp-encoder signals are not action signals so mark them
correctly and use an accumulator to collect the result value.
2013-12-30 15:17:05 +01:00
Stefan Sauer
2e277bb341 wavparse: emit midi-base-note tag from data in 'smpl' chunk
Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and
emit it as a tag.
2013-12-30 14:41:47 +01:00
George Kiagiadakis
5ddf6a5e32 gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
2013-12-30 14:03:05 +01:00
George Kiagiadakis
17517ca491 rtpsession: allow setting internal-ssrc again 2013-12-30 14:03:05 +01:00
Edward Hervey
e732b86b8e y4mencode: Remove dead code
set/get property isn't used
2013-12-30 13:50:35 +01:00
Edward Hervey
ac40045d0d rtpqcelpdepay: Remove uneeded variable 2013-12-30 13:50:35 +01:00
Aleix Conchillo Flaqué
47c65fc269 rtpbin: allow dynamic RTP/RTCP encoders/decoders
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
  added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
  and request-rtcp-decoder). The user will be able to provide encoders
  or decoders dynamically. The encoders must follow the srtpenc API and
  the decoders the srtpdec API. Having separate signals for RTP and RTCP
  allows the user to use different encoders/decoders or provide the same
  one (e.g. that would be the case for srtpenc).

  Also, rtpbin now allows application/x-srtp in its pads.

  https://bugzilla.gnome.org/show_bug.cgi?id=719938
2013-12-30 11:24:00 +01:00
Wim Taymans
f48bbabafc rtpjitterbuffer: dynamically recalculate RTX parameters
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.

Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
2013-12-30 11:18:51 +01:00
Wim Taymans
416bd9a2c3 rtpjitterbuffer: calculate average jitter 2013-12-30 11:18:51 +01:00
Wim Taymans
7181a21ca9 rtpsession: use RTT from the Retransmission event
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
2013-12-30 11:18:50 +01:00
Wim Taymans
e996f73d0c jitterbuffer: take more accurate running-time for NACK
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
2013-12-30 11:18:50 +01:00
Thiago Santos
c1cd2f81f9 qtdemux: improve mss_mode/fragmented special handling
Make it clear what should be handled purely by mss mode:
1) Expose the streams on the first moof as there are no moov atoms
2) Properly cleanup streams on flushes

Add a note about the meaning of upstream_newsegment and mss_mode
for future reference.

Make all other special fragment handling shared for both dash
and mss streams.
2013-12-27 12:04:49 -03:00
Thiago Santos
a82f3418fd qtdemux: drain the adapter before pushing EOS
In a fragmented scenario, qtdemux is operating in push mode
and it gets a fragmented buffer. While processing its data
downstream gets unlinked (or a input-selector changes its
active pad and returns not-linked). Qtdemux stops processing
this fragment and returns not-linked upstream, leaving the
remaining data in its adapter.

When it gets an EOS it should make sure that all the data it
had received is pushed before pushing EOS.
2013-12-27 12:00:27 -03:00
Wim Taymans
bf878d75d1 rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
Use the aggregate control instead of the original request url to perform
PAUSE/PLAY and TEARDOWN.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003
2013-12-26 11:27:30 +01:00
Sebastian Dröge
2f07b570f7 rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly 2013-12-24 14:40:25 +01:00
Nicola Murino
5b1108dd5f matroskamux: adpcm max block align is 8192 2013-12-24 10:00:16 +01:00
Sebastian Dröge
4baf8080f2 matroskamux: Use correct codec id for ADPCM/DVI 2013-12-23 15:46:48 +01:00
Sebastian Dröge
7cae8922cb matroskademux: Check for the correct size of codec_data in the ACM case 2013-12-23 15:46:43 +01:00
Nicola Murino
00ea1cb003 matroskamux: basic adpcm support
https://bugzilla.gnome.org/show_bug.cgi?id=664339
2013-12-23 15:31:04 +01:00
Sebastian Dröge
371482a90c qtdemux: Fix calcuation of descriptor length
https://bugzilla.gnome.org/show_bug.cgi?id=720813
2013-12-23 15:09:49 +01:00
Tim-Philipp Müller
9c9efffd8c udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped
coverity CID 1139866.
2013-12-19 20:35:03 +00:00
Tim-Philipp Müller
627109ce4d multiudpsink: fix misleading comment
Those are not allocated on the stack.
2013-12-19 12:47:22 +00:00
Todd Agulnick
8bab119af9 Some compiler warning fixes to satisfy XCode compiler
https://bugzilla.gnome.org/show_bug.cgi?id=720513
2013-12-16 16:52:40 +01:00
Sebastian Dröge
2927805749 wavpackparse: Post AUDIO_CODEC tag 2013-12-16 10:03:06 +01:00
Sebastian Dröge
753d3c23a2 sbcparse: Post AUDIO_CODEC tag 2013-12-16 10:03:06 +01:00
Sebastian Dröge
05e196cbb6 flacparse: Post AUDIO_CODEC tag
https://bugzilla.gnome.org/show_bug.cgi?id=720512
2013-12-16 10:03:06 +01:00
Sebastian Dröge
29f2cae129 dcaparse: Post AUDIO_CODEC tag 2013-12-16 10:03:05 +01:00
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
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
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