Commit graph

9736 commits

Author SHA1 Message Date
Marc Leeman
8737e29a49 rtpsource: small spell correct 2019-02-27 16:14:22 +01:00
Nicolas Dufresne
e72ef633a6 rtpsession: Fix EOS forwarding
So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.

So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
2019-02-25 17:06:50 +00:00
Jan Schmidt
098f936be8 wavparse: Declare support for RF64
RF64 encode support was added to wavenc quite some time
ago, but not declared in wavparse. It seems wavparse can
decode it though, so add it to the sink pad.

The RF64 support was added in
https://bugzilla.gnome.org/show_bug.cgi?id=735627
2019-02-24 14:29:27 +00:00
Nicolas Dufresne
06c340edd4 rtp: Add property to disable RTCP reports per internal rtpsource
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
2019-02-13 15:07:39 -05:00
Olivier Crête
b88a3abf46 rtpsession: Emit on-new-sender-ssrc for RTX ssrc also 2019-02-13 15:07:39 -05:00
Olivier Crête
bf00ee46de rtpjitterbuffer: Limit size to 2^15 packets
If it goes over 2^15 packets, it will think it has rolled over
and start dropping all packets. So make sure the seqnum distance is not too big.

But let's not limit it to a number that is too small to avoid emptying it
needlessly if there is a spurious huge sequence number, let's allow at
least 10k packets in any case.
2019-02-11 23:41:14 +00:00
Olivier Crête
086bad4643 rtpjitterbuffer: There is no automatic reorder threshold 2019-02-11 11:33:36 -05:00
Ilya Smelykh
6db7bb1539 flvmux: Use 8kHz sample rate for alaw/mulaw audio 2019-02-08 20:33:55 +00:00
Ilya Smelykh
b9c4c8bca5 flvdemux: set sample rate to 8KHz for G.711 audio 2019-02-08 20:33:55 +00:00
Vivia Nikolaidou
92272b5e5c qtmux: Only write timecode trak for video
Recent changes in ccextractor were attaching timecode meta to the closed
caption track. We shouldn't write timecode information for the closed
caption trak.
2019-02-08 14:13:46 +02:00
Edward Hervey
f5f1de54d2 qtdemux: Remove trailing '\n' in debug 2019-02-05 11:01:21 +01:00
Mathieu Duponchelle
6ed7ddebf9 rtspsrc: use the correct segment seqnum 2019-02-04 13:14:37 +00:00
Mathieu Duponchelle
a6d681ad09 rtpjitterbuffer: use the correct segment seqnum 2019-02-04 13:14:37 +00:00
Mathieu Duponchelle
5e92f7d208 rtpsession: use the correct segment seqnum 2019-02-04 13:14:37 +00:00
Thibault Saunier
bc8af2cca5 flvdemux: Do not error out if the first added and chained pad is not linked
And let it the oportunity to get its other pad linked

Example:

```
$ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error.
Additional debug info:
../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
```
2019-02-02 18:36:09 +00:00
Christopher Snowhill
818428ce9c webmmux: allow resolutions above 4096
Modify the caps string to allow width and height greater than 4096.
There is no need to restrict it since the matroska format allows the
width and height values to be up to eight bytes long, and this also
applies to the webm subset of the format.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550
2019-02-02 15:40:53 +00:00
Nicolas Dufresne
6d3859bf70 rtph265depay; Fix handling of marker on aggregated packet
When multiple nals are aggrgated, the marker bit should be associated only
with the last NAL of the packet. Otherwise we may break rendering in with
AU alignment.
2019-01-31 19:30:14 +00:00
Nicolas Dufresne
98251f0158 rtph264depay: Fix handling or marker on STAP-A
Only forward the marker for the last NAL of the STAP-A. Otherwise each NAL
endup being assumed to be a full frame which may break rendering.

Fixes 557
2019-01-31 19:30:14 +00:00
Vincent Penquerc'h
a329a3a2c6 deinterleave: Allow switching between 1 channel configs
regardless of whether they're positioned, since positioning
with a 1 channel stream doesn't change anything.
2019-01-28 23:23:41 +00:00
Patrick Radizi
d3662bae00 rtspsrc: send GstRTSPSrcTimeout message on timeout
The GstRTSPSrcTimeout message is sent by the rtspsrc when it receives
the on-timeout signal from rtpsession. This can be used by an
application for error handling.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/499
2019-01-14 08:15:23 +00:00
Sebastian Dröge
ab8100e664 flvdemux: Handle the encoder metadata the same as metadatacreator
And store it in our ENCODER tag.
2019-01-13 13:22:41 +00:00
Sebastian Dröge
c28a9d5d9c flvmux: Add encoder metadata to the header
And also add a property for setting this. By default it has the same
value as the metadatacreator metadata.

Various software is using encoder instead of metadatacreator, others are
using them both for different purposes. As such it's useful to have
support for setting both here.
2019-01-13 13:22:41 +00:00
Jan Alexander Steffens (heftig)
06b2bbd8c7 rtph265pay: Only mark the last fragment of an AU
Commit e721071dca removed the check for
the end of fragmentation. As a result, all fragments of an AU's last
NALU were marked.
2019-01-09 15:36:40 +00:00
Jan Alexander Steffens (heftig)
798f320ba7 rtph264pay: Only mark the last fragment of an AU
Commit 4add820cce removed the check for
the end of fragmentation. As a result, all fragments of an AU's last
NALU were marked.

Potential fix for https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/540
2019-01-09 15:36:40 +00:00
Sebastian Dröge
3537c4d217 splitmuxsrc: Refactor part preparation code and remove "prepared" signal from reader helper object
We don't need a special signal anymore but can directly work with
async-done
2019-01-09 13:35:58 +02:00
Sebastian Dröge
99bb6f44ba splitmuxsrc: Implement state change asynchronously instead of blocking
Blocking in change_state() is a recipe for disaster, even more so if
we wait for another thread that also calls into various element API and
could then lead to deadlocks on e.g. the state lock.
2019-01-09 13:35:58 +02:00
Sebastian Dröge
ec931601a6 qtdemux: Split CEA608 buffers correctly so that each output buffer represents a single frame 2019-01-02 10:29:46 +00:00
Sebastian Dröge
aa65ea85f9 qtdemux: Refactor buffer pushing into its own function 2019-01-02 10:29:46 +00:00
Sebastian Dröge
d471be4f3a qtdemux: Extract CEA608 framerate from the (first) video stream
EA608 closed caption tracks are a bit special in that each sample
can contain CCs for multiple frames, and CCs can be omitted and have to
be inferred from the duration of the sample then.

As such we take the framerate from the (first) video track here for
CEA608 as there must be one CC byte pair for every video frame
according to the spec.

For CEA708 all is fine and there is one sample per frame.
2019-01-02 10:29:46 +00:00
Seungha Yang
022fbe9a46 matroskademux: Don't leak allocated index memory
Don't forget to free returned memory from _search_pos()
2018-12-26 20:31:10 +09:00
Tim-Philipp Müller
f480261815 audiofx: add stereo element which was moved from -bad to build
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457
2018-12-25 16:10:49 +01:00
Tim-Philipp Müller
d0a5e9d8b0 Move stereo plugin from -bad
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457
2018-12-25 13:07:23 +01:00
Philippe Normand
ce96d6dcd4 qtdemux: Offset correction for track language code parsing
The duration field being a uint64, is stored in 8 bytes, not 4. So the offset of
the following field, language code, needs to be updated accordingly so that the
parsed language code is not garbage.
2018-12-22 20:05:34 +01:00
Juan Navarro
5dfd12b64c rtspsrc: Accept NULL for "port-range" property
The documentation of "port-range" implies that passing NULL should be
valid, but currently it is not. Without this check, the sscanf() call
will crash.
2018-12-21 10:59:22 +01:00
Mathieu Duponchelle
f52e16ceb8 Revert "rtpbin: receive bundle support"
This reverts commit dcd3ce9751.

This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.

This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.

Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.

Fixes #537
2018-12-20 13:25:10 +00:00
Nicolas Dufresne
05059ce16b rtph264pay/rtph265pay: Fix use after free
We can't assume a buffer that has been pushed in the adapter is still
valid. This fixes a use after free detect when running test on jenkins.
2018-12-19 13:54:57 -05:00
Nicolas Dufresne
d397cf6d1f rtph265pay: Don't wait for next nal when input is aligned
This is the same as what was done on rtph264pay in the patch
d5d28055c1
2018-12-18 13:39:54 -05:00
Nicolas Dufresne
0524e6f8cd rtph265depay: Drain on EOS event 2018-12-18 13:39:54 -05:00
Nicolas Dufresne
65b01d5f02 rtph265depay: Factor out the code that push
This will be needed to implement draining on EOS.
2018-12-18 13:39:53 -05:00
Nicolas Dufresne
e694e2752a rtph264depay: Drain on EOS event 2018-12-18 13:39:46 -05:00
Nicolas Dufresne
d12128f527 rtph264depay: Factor out the code that push
This will be needed to implement draining on EOS.
2018-12-18 13:39:46 -05:00
Nicolas Dufresne
5e8cab71ea rtph26xpay: Remove unused IS_ACCESS_UNIT macro
This macro is not longer used. It was secretly checking if that nal was
a slice, and confusingly name to that one may think it was checking if
the nal is an AUD.
2018-12-18 13:39:46 -05:00
Nicolas Dufresne
0a6e5e439c rtph265pay: Fix reading timestamps from adapter
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
ff2e5b94b9 rtph265pay: Forward the marker bit as buffer flag
We have a buffer flag to represent the marker bit (when present).
Forward this bit by setting the buffer flag accordingly.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
e721071dca rtph265pay: Properly set the marker bit
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.

So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
1f72131781 rtph264pay: Fix reading timestamps from adapter
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
4add820cce rtph264pay: Properly set the marker bit
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.

So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
e4f38c986e rtph264depay: Forward the marker bit as buffer flag
We have a buffer flag to represent the marker bit (when present).
Forward this bit by setting the buffer flag accordingly.
2018-12-18 13:30:05 -05:00
Nicolas Dufresne
13278fbcf5 rtph264pay: Protect against use of reserved NAL types
Don't allow external encoder to use one of the reserved NAL type
implicated in NAL aggreation. These out-of-spec NAL types, if passed
from the outside world will lead to an invalid RTP payload being
created.
2018-12-18 13:30:05 -05:00
Sebastian Dröge
4f7ef56c53 isomp4: Replace GST_VIDEO_CAPTION_TYPE_CEA608_IN_CEA708_RAW with CEA608_S334_1A
For the demuxer we have to select line offset 0 for the time being as
this information is not passed over MOV.
2018-12-15 21:31:20 +00:00