Commit graph

9718 commits

Author SHA1 Message Date
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
Olivier Crête
d857522237 rtpjitterbuffer: Run all timers immediately on EOS
When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.
2018-12-14 12:10:16 +00:00
Nicolas Dufresne
3de2c28fc1 rtpjitterbuffer: Stop waiting after EOS
After EOS is received, it is pointless to wait for further events,
specially waiting on timers. This patches fixes two cases where we could
wait instead of returning GST_FLOW_EOS and trigger a spin of the loop
function when EOS is queued, regardless if this EOS is the queue head or
not.
2018-12-14 12:10:16 +00:00
Jochen Henneberg
7824e87c5b flacparse: On sink caps change restart parser
Draining the parser is not enough here, on caps change we need to
reset it so it is ready to accept new caps.
2018-12-14 09:22:33 +00:00
Jochen Henneberg
9b6dcc7f1b rtpgstdepay: Update pad caps if inline caps change
If the inlined caps change while using the same CV we need to update the
source pad caps.
2018-12-14 09:22:33 +00:00
Sebastian Dröge
c50be8f146 qtdemux: Put framerate into the closedcaption caps if it can be calculated from the stream
Using the same calculation used for video streams.
2018-12-06 16:05:50 +00:00
Sebastian Dröge
830e7dc14b qtmux: Set timescale of closedcaption tracks to the one of the main video track 2018-12-06 16:05:50 +00:00
Maciej Wolny
ec655de288 Remove duplicate declarations
This causes 'redefinition of typedef ...' errors on GCC 4.5.3
2018-12-04 11:13:02 +00:00
Alicia Boya García
38b553dda7 qtdemux: set need_segment after a second moov
stream.segment should be updated with the values of the current edit
list, also when a new `moov` is received. Unfortunately this was not
being the case because of an early return.

As a consequence of this bugs, no end of movie clipping was being
performed on the new moov and no segment event was being emitted.

When performing stream switching (e.g. in MSE) the new moov may have a
different edit list. This is often the case when switching between
baseline H.264 (which lacks B-frames) and more demanding profiles. For
this reason it's important to emit a new segment in order to be able
to get matching stream times.
2018-11-30 20:44:57 +00:00
Alicia Boya García
26cc201c8a qtdemux: Initialize QtDemuxStream.segment in its constructor
This patch moves the initialization of QtDemuxStream.segment from
gst_qtdemux_add_stream() to _create_stream(). This ensures the segment
is always initialized when the stream is created.

Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case
were a track is reparsed and qtdemux_reuse_and_configure_stream() is
called instead of gst_qtdemux_add_stream(). (See
qtdemux_expose_streams() in the non streams-aware case.)
2018-11-30 20:44:57 +00:00
Miguel Paris
48a4fd4e50 rtpsession: properly handle rtcp_feedback_retention_window
- Consider GST_CLOCK_TIME_NONE as not to be used.
- Complete "rtcp-feedback-retention-window" property getter/setter
  implementation.
2018-11-30 10:55:26 +00:00
Miguel Paris
458741e4b2 rtpsource: properly prune RTCP packets out of feedback_retention_window
Closes #522
2018-11-30 10:55:26 +00:00
Miguel Paris
53f03d4cc1 rtpsource: properly compare buffer PTSs 2018-11-30 10:55:26 +00:00
Miguel Paris
57829c3352 rtpsource: retain_rtcp_packet: warning if invalid running_time 2018-11-30 10:55:26 +00:00
Miguel Paris
36f55b03e8 rtpsession: properly set the running_time for rtcp packet info 2018-11-30 10:55:26 +00:00
Nicolas Dufresne
d637567ab3 rtpssrcdemux: Rename confusingly name lock macros
This is an extra internal recurisve lock use to avoid having to take
both sink pad streams lock all the time. This patch renamed it
INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream
GST_PAD API.
2018-11-29 15:34:47 -05:00
Nicolas Dufresne
40daf6322d rtpssrcdemux: Hold on internal stream lock while pushing sticky
This reverts "6f3734c305 rtpssrcdemux: Only forward stick events while
holding the sinkpad stream lock" and actually hold on the internal
stream lock. This prevents in some needed case having a second
streaming thread poping in and messing up event ordering.
2018-11-29 15:33:57 -05:00
Matej Knopp
e9495c55f4 matroskademux: fix handling of MS ACM audio
Pass riff codec-data as strf, not strd, which is where
gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE
data.

https://bugzilla.gnome.org/show_bug.cgi?id=757583
Fixes #234
2018-11-28 11:55:14 +00:00
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00