Commit graph

10160 commits

Author SHA1 Message Date
Olivier Crête
7a317ff732 rtpstorage: Add more debug messages 2019-03-26 19:41:06 -04:00
Olivier Crête
785219a317 rtpstorage: Make debug category available to sub objects 2019-03-26 19:41:06 -04:00
Olivier Crête
9b0a373eac rtpstorage: Add debug funcptr to chain function 2019-03-26 18:08:57 -04:00
Nicolas Dufresne
79fd0af152 gstrtpsession: Remove set but not use running-time 2019-03-22 20:01:52 +00:00
Olivier Crête
7ecbd7271d rtpmanager: Register chain functions to debug 2019-03-22 16:44:41 +00:00
Nicolas Dufresne
2ff7519d73 rtpbin: Allow reusing the sender AUX bin
This is needed for the case you don't know in advance all the sessions
you will be using, but would like to place all the related AUX element
in the same GstBin. As per current implementation, each time an sender
AUX bin is requested and returned, RTPBin will walk the src pads and
create sessions for these pads.

In the current implementation, if a src pad already have a sessions, it
returns an error and stops. As a side effect, if an AUX bin is reused in
a following AUX bin request, it can only work if the pads are created on
the last request.

This change simply relax the restriction in order to keep walking, and
just ensure that all newly created pads have a sessions.
2019-03-21 21:10:43 +00:00
George Kiagiadakis
d5ce10240a gstrtpsession: improve stats about rtx requests 2019-03-21 13:40:31 -04:00
George Kiagiadakis
db647ee55b rtprtxsend: Improve looging of not found RTX packet
When an RTX packet is not found, display a message that say if the
packet have not arrived yet or if it was already removed from the RTX
packet queue.
2019-03-21 13:19:52 -04:00
Nicolas Dufresne
0aff8a7d30 rtpsession: Remove unused rtp_session_create_source 2019-03-21 13:19:52 -04:00
Seungha Yang
63bb1e3a4d qtdemux: Don't pass zero to denominator for framerate
Need to respect return of gst_video_guess_framerate() to ensure
non-zero denominator.

This patch is to fix below error with an abnormal (but has valid frame) file.
(gst-play-1.0:17940): GStreamer-CRITICAL **: passed '0' as denominator for `GstFraction'
2019-03-19 12:35:08 +09:00
Charlie Turner
39d32b2394 qtdemux: Find mp4a esds atoms in protected streams sample description tables.
This problem was found in Test. 2 of the YouTube 2018 EME
tests[1]. The code was accidentally not finding an mp4a's esds atom in
the sample description table when the stream was encrypted. It assumed
that if the stream is protected, then only an enca atom will be found
here. What happens with YouTube is they often provide protected
content with a few seconds of clear content, and then switch to the
encrypted stream.

The failure case here was an incorrect codec_data field being sent
into aacparse. The advertisement of stereo audio @ 44.1kHz for the
mp4a (unprotected) stream was incorrect. As usual, the esds contained
the real values here which were mono at 22050 Hz.

Here's what the MP4 tree looks like for these types of files,
demonstrating why the code was making a wrong assumption (or maybe
YouTube is being unusual),

[ftyp] size=8+16
...
[moov] size=8+1571
...
  [trak] size=8+559
...
          [stsd] size=12+234
            entry-count = 2
            [enca] size=8+147
              channel_count = 2
              sample_size = 16
              sample_rate = 44100
              [esds] size=12+27
                ...
            ...
            [mp4a] size=8+67
              channel_count = 2
              sample_size = 16
              sample_rate = 44100
              [esds] size=12+27
                ...

In addition to fixing this, the checks for esds atoms in mp4a and mp4v
have been made symmetrical. While I haven't seen a test case for video
with the same problem, it seemed better to make the same checks. This
also fixes a crash reported from another user[2], they also noted the
asymmetry with mp4v and mp4a.

[1] https://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2018.html?test_type=encryptedmedia-test
[2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/398
2019-03-15 12:41:33 +00:00
Andreas Frisch
3160713abf flvmux: Fix scale of time values in warning message 2019-03-15 09:55:32 +00:00
Sebastian Dröge
a676c17259 rtspsrc: Don't remove udpsrc/sink from rtspsrc if they were not added to it
This can happen in various error cases that could happen between the
creation of the element in question and the adding to the rtspsrc.

It causes an ugly critical warning right now but is otherwise harmless.
2019-03-15 08:21:11 +00:00
Antonio Ospite
8c26e33f20 imagefreeze: add a num-buffers property
The imagefreeze element can be handy for benchmarking downstream
elements because it re-uses the same buffer memory and introduces less
overhead compared to always creating new frames with videotestsrc.

However it's not possible to make imagefreeze send EOS when using
gst-launch-1.0.

Add a num-buffers property to make it look more like a source in the
above scenario.
2019-03-14 09:12:28 +01:00
Guillaume Desmottes
fcd568dd56 matroskamux: add support for new color primaries 2019-03-12 16:52:45 +01:00
Antonio Ospite
2dfe228740 docs: fix typos s/recieve/receive/ 2019-03-07 12:41:40 +01:00
Antonio Ospite
30db93e3a4 rtpsource: fix documentation of rtp_source_send_rtp parameters
In commit 28e5f9098 (rtpbin: use PacketInfo for the sender, 2013-09-13)
the rtp_source_send_rtp signature changed but the documentation was not
adjusted to match the new one.

Update the documentation to match the function signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
38285e5bcf rtpsession: fix typo in a comment, s/SESSION_LOCK/RTP_SESSION_LOCK/
Fix a typo in a comment, mainly to avoid confusing autocompletion in
text editors.
2019-03-07 12:41:40 +01:00
Antonio Ospite
43e4226844 rtpsession: fix typos and update parameters names in comments
Some functions now accept a generic 'gpointer data' parameter because
they can work either on a single buffer or a buffer list.

However the comments were still referring to the old 'GstBuffer *buffer'
parameter, so update the comments to match the actual functions
signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
b2b60c4d8f rtpstats: fix some fields names in the RTPSourceStats documentation
Fix documentation of RTPSourceStats to use the actual fields names.
2019-03-07 10:36:11 +01:00
Mathieu Duponchelle
0da8f111e6 rtpulpfdecdec: only put recovered packet back into storage if not recovered from there 2019-03-06 19:40:10 +00:00
Mathieu Duponchelle
f9b49aef09 rtpulpfecdec: fix buffer leak when packet is recovered from storage
Exposed by rtpulpfecdec_recovered_from_storage test.
2019-03-06 19:40:10 +00:00
Tim-Philipp Müller
c79cf179cc rtph264depay: fix caps leak
Exposed by rtp_h264depay_bytestream() unit test.
2019-03-06 18:21:20 +00:00
Tim-Philipp Müller
899d0c4b3b matroskademux: fix AV1 caps when there's no codec_data
There is no "byte-stream" format for AV1 in Matroska, this
was probably cargo-culted from H.264. codec_data / CodecPrivate
is now mandatory for AV1 in Matroska[*], but there are sample
files out there which don't have it (e.g. some Elecard ones).

[*] https://github.com/Matroska-Org/matroska-specification/blob/master/codec/av1.md#codecprivate-1
2019-03-01 17:37:55 +00:00
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
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
Thibault Saunier
2e0a45d7df aspectcropration: Fix potential unref of NULL pointer 2018-11-26 08:11:57 -03:00
Thibault Saunier
eb2b58cc0b aspectcropratio: Set caps from the streaming thread on property changes
Otherwise it might lead to deadlocks

See https://gitlab.gnome.org/GNOME/pitivi/issues/2259

Closes #518
2018-11-26 07:14:09 -03:00
Nicolas Dufresne
21378d83c2 rtpssrcdemux: Forward serialized events to all pads
While forwarding serialized event, we use gst_pad_forward() function.
In the forward callback (GstPadForwardFunction) we always return
TRUE. Returning true there will stop the dispatching procedure. As a
side effect, only one events is receiving the events. This breaks
when sending EOS from the applicaiton, it also breaks the latency
tracer.
2018-11-24 13:01:25 +00:00
Alicia Boya García
753b7c17f3 matroskademux: Defer seeks received before GST_MATROSKA_READ_STATE_DATA
This patch enables matroskademux to receive seeks before it reaches
GST_MATROSKA_READ_STATE_DATA.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/514

This also enables receiving seeks in the element READY state.

When such a seek is received, it is stored to be later handled when
GST_MATROSKA_READ_STATE_DATA is reached.
2018-11-15 08:01:29 +00:00
Linus Svensson
8fc8b7ee33 rtpsession: Implement reset
Reset RTPSession when rtpsession changes state from PAUSED to READY.
Without this change, a stored last_rtptime in RTPSource could interfere
with RTP timestamp generation in RTCP Sender Report.

Fixes #510
2018-11-13 12:30:35 +00:00
Mathieu Duponchelle
fd560bcb27 rtpfunnel: Stop using G_DECLARE_FINAL_TYPE
Fixes #516
2018-11-13 00:37:11 +01:00
Matthew Waters
40fc8aea8f matroska: implement preliminary support for the bitrate query
Return the size / total duration as a ballpark estimate.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60
2018-11-07 15:07:18 +00:00
Matthew Waters
8a7074f748 isomp4: add preliminary support for the bitrate query
Return the upstream size over the duration as a first estimate.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60
2018-11-07 15:07:18 +00:00
Sebastian Dröge
87202cc03d rtpbin: Sink jitterbuffer/storage before passing as parameters to signals
Otherwise signal handlers from bindings will take ownership of them as
they are still floating, and we won't own a reference inside rtpbin
anymore.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/515
2018-11-07 09:11:16 +00:00
Olivier Crête
fea0d0b1a4 flvmux: Force timestamps to always be increasing
https://bugzilla.gnome.org/show_bug.cgi?id=796382
2018-11-05 18:17:01 -05:00
Seungha Yang
5d542030db qtdemux: Ignore corrupted CTTS box
If ctts (CompositionOffsetBox) has larger sample_offset
(offset between PTS and DTS) than (2 * duration) of the stream,
assume the ctts box to be corrupted and ignore the box.

https://bugzilla.gnome.org/show_bug.cgi?id=797262
2018-11-01 16:03:12 +02:00
Sebastian Dröge
a03d29420b scaletempo: Implement SEGMENT query
https://bugzilla.gnome.org/show_bug.cgi?id=797313
2018-10-28 17:52:18 +00:00
Sebastian Dröge
2415d517f1 wavparse: Implement SEGMENT query
https://bugzilla.gnome.org/show_bug.cgi?id=797313
2018-10-28 17:52:18 +00:00
Olivier Crête
486044063a dtmfsrc: Declare output as interleaved
This element doesn't support planar audio yet.
2018-10-28 17:12:59 +00:00
Olivier Crête
cc69c876fe rtpsession: Allow changing the SDES at runtime
Make it possible to modify the SDES in a packet at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=763502
2018-10-28 12:10:36 +00:00
Alicia Boya García
5fcb7f715a qtmux: round to nearest when computing mehd and tkhd duration
This fixes a bug where in some files mehd.fragment_duration is one unit
less than the actual duration of the fragmented movie, as explained below:

mehd.fragment_duration is computed by scaling the end timestamp of
the last frame of the movie in (in nanoseconds) by the movie timescale.

In some situations, the end timestamp is innacurate due to lossy conversion to
fixed point required by GstBuffer upstream.

Take for instance a movie with 3 frames at exactly 3 fps.

$ gst-launch-1.0 -v videotestsrc num-buffers=3 \
  ! video/x-raw, framerate="(fraction)3/1" \
  ! x264enc \
  ! fakesink silent=false

dts: 999:59:59.333333334,  pts: 1000:00:00.000000000, duration: 0:00:00.333333333
dts: 999:59:59.666666667,  pts: 1000:00:00.666666666, duration: 0:00:00.333333334
dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333

The end timestamp is calculated by qtmux in this way:

end timestamp = last frame DTS + last frame DUR - first frame DTS =
  = 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 =
  = 0:00:00.999999999

qtmux needs to round this timestamp to the declared movie timescale, which can
ameliorate this distortion, but it's important that round-neareast is used;
otherwise it would backfire badly.

Take for example a movie with a timescale of 30 units/s.

0.999999999 s * 30 units/s = 29.999999970 units

A round-floor (as it was done before this patch) would set fragment_duration to
29 units, amplifying the original distorsion from 1 nanosecond up to 33
milliseconds less than the correct value. The greatest distortion would occur
in the case where timescale = framerate, where an entire frame duration would
be subtracted.

Also, rounding is added to tkhd duration computation too, which
potentially has the same problem.

https://bugzilla.gnome.org/show_bug.cgi?id=793959
2018-10-27 13:12:56 +01:00
Marc Leeman
827d70daee udpsrc: print information about bind_error socket error
In some cases, a bind error occurs during operation. Printing
the information about the problem is critical for finding the
conflict

https://bugzilla.gnome.org/show_bug.cgi?id=797340
2018-10-27 13:12:53 +01:00
Johan Bjäreholt
e736f29376 matroska-demux: Fix caps memleak
https://bugzilla.gnome.org/show_bug.cgi?id=797326
2018-10-27 10:48:38 +01:00
Johan Bjäreholt
abfc7da345 matroska-ids: Fix uninitialized memory in contexts
https://bugzilla.gnome.org/show_bug.cgi?id=797327
2018-10-24 09:54:20 +01:00
Sebastian Dröge
01a2119ad0 qtmux: Add property for providing a threshold after which we create an edit list for gaps at the start
https://bugzilla.gnome.org/show_bug.cgi?id=797290
2018-10-22 12:29:23 +01:00
Sebastian Dröge
324f8c7f3c qtmux: Correctly set tkhd width/height to the display size
It was previously set to the display aspect ratio, e.g. 4x3, 16x9, etc.
but should be set to the display size.

This is a regression from e655d47dfc
(1.5.1) and was correct before that.

https://bugzilla.gnome.org/show_bug.cgi?id=797318
2018-10-22 12:23:05 +01:00
Seungha Yang
7bce030be3 qtdemux: Fix build with GLib versions < 2.54
g_ptr_array_find_with_equal_func was introduced in glib 2.54
which is a higher version than our minimum required one.

https://bugzilla.gnome.org/show_bug.cgi?id=797239
2018-10-20 12:38:32 +01:00
Seungha Yang
05bd25ea35 qtdemux: Don't switch active streams and old streams ...
... before the old streams is not exposed yet for MSS stream.

In case of DASH, newly configured streams will be exposed
whenever demux got moov without delay.
Meanwhile, since there is no moov box in MSS stream,
the caps will act like moov. Then, there is delay for exposing new pads
until demux got the first moof.

So, following scenario is possible only for MSS but not for DASH,
STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET)
-> STREAM-START-> CAPS (configure stream again).

In above scenario, we can reuse old stream without any stream reconfigure.

https://bugzilla.gnome.org/show_bug.cgi?id=797239
2018-10-19 14:44:43 +02:00
Seungha Yang
b2876ad8a4 qtdemux: Use GPtrArray to store QtDemuxStream structure
GPtrArray has less overhead than linked list and the length also
can be auto updated by using it.

https://bugzilla.gnome.org/show_bug.cgi?id=797239
2018-10-19 14:44:43 +02:00
Seungha Yang
1600323119 qtdemux: Make QtDemuxStream refcounted structure
This a prework for porting GPtrArray.
Refcounting will help the use of g_ptr_array_new_with_free_func()
with QtDemuxStream structure

https://bugzilla.gnome.org/show_bug.cgi?id=797239
2018-10-19 14:44:43 +02:00
Seungha Yang
72123e3da3 qtdemux: Make function foreach method friendly
https://bugzilla.gnome.org/show_bug.cgi?id=797239
2018-10-19 14:44:43 +02:00
Olivier Crête
20d5f92b28 qtdemux: Only set width/height in caps if they're non-0
If they are not valid, then let a downstream parser complete them.

https://bugzilla.gnome.org/show_bug.cgi?id=796878
2018-10-19 14:19:27 +02:00
Wim Taymans
7db251f214 avidemux: fix misleading debug line 2018-10-18 15:50:19 +02:00
Philippe Normand
56669205eb qtdemux: Avoid warning when reporting about decryptors
https://bugzilla.gnome.org/show_bug.cgi?id=796652
2018-10-17 15:51:32 +01:00
Tim-Philipp Müller
cac9aab107 meson: Replace empty configuration_data() with copy keyword
Use 'copy' keyword to avoid meson warning message.
Note that 'copy' keyword in configure_file() is available
since meson 0.47.0

https://bugzilla.gnome.org/show_bug.cgi?id=797298
2018-10-17 14:15:33 +01:00
Vivia Nikolaidou
af0e30d545 splitmuxsink: Do not hardcode frames_of_daily_jam
Apart from the obvious drawbacks of hardcoding, the drawback here was
that, if we subtracted 2 frames (instead of 2.6) from the target running
time, we'd request the next keyframe a bit too far into the future,
which would make our files split at the wrong position.

https://bugzilla.gnome.org/show_bug.cgi?id=797293
2018-10-16 16:06:47 +03:00
Vivia Nikolaidou
09904e59df qtmux: Allow up to 1% of frame rate for lateness
https://bugzilla.gnome.org/show_bug.cgi?id=797290
2018-10-16 16:05:46 +03:00
Mathieu Duponchelle
ee461fb326 rtpfunnel: fix shutdown
By disposing of the ssrc_to_pad map in finalize instead of
dispose.
2018-10-15 14:20:58 +02:00
Havard Graff
53a45b1222 Initial commit of GstRtpFunnel
For funneling together rtp-streams into a single session.
Use-cases include multiplexing and bundle.
2018-10-15 14:20:58 +02:00
Yeongjin Jeong
bd6a4aa10d flvdemux: Use aac codec-data to adjust channels if needed
Flv does not support various channels in AAC stream format, for example
flvdemux detect an audio channels of 2(stereo) when the AAC really is 1(mono).

https://bugzilla.gnome.org/show_bug.cgi?id=797275
2018-10-12 14:35:37 -04:00
Yeongjin Jeong
8cae95a22d flvmux: Don't refuse caps changes after starting to write headers in streamable mode.
Flv does support changing the stream type and stream properties
after the headers were started to be written, and for example H264
codec_data changes can be supported.

https://bugzilla.gnome.org/show_bug.cgi?id=797256
2018-10-11 15:35:24 -04:00
Vivia Nikolaidou
faee020994 splitmuxsink: Fix if condition in drop-frame timecode wrap-around
Was previously: if ( x | y && a == b). Changed it into if ((x & y) && (a
== b)).
2018-10-11 13:58:34 +03:00
Vivia Nikolaidou
1219712da0 splitmuxsink: Subtract daily jam offset when day wraps around
For drop-frame framerates, when the expected next max timecode wraps
around at the end of the day, we have to subtract the offset of the
daily jam, otherwise we end up with a duration that's a few frames too
long.

https://bugzilla.gnome.org/show_bug.cgi?id=797270
2018-10-11 13:51:08 +03:00
Havard Graff
6c05180dc5 rtpmux: respect downstream "timestamp-offset" in caps.
https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:39:02 -04:00
Havard Graff
6f37bd8f19 rtpmux: cleanup ssrc-handling code a bit
And add some better logging.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:38:57 -04:00
Havard Graff
18a1dc4ab6 rtpmux: protect against NULL caps
Due to state-changes deactivating the pad from another thread,
this can happen.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Havard Graff
7cd36d2914 rtpmux: property should overrule both upstream and downstream
https://bugzilla.gnome.org/show_bug.cgi?id=762213

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Peter Körner
4b570026aa splitmuxsink: accept pads named 'sink' on the muxer, handle static pads as well
https://bugzilla.gnome.org/show_bug.cgi?id=797241
2018-10-03 23:24:26 +03:00
Thibault Saunier
defae35035 matroskdemux: do not use MapInfo.data after unmapping
And minor gst-indenting
2018-10-03 17:39:33 +02:00
Yacine Bandou
0432826950 matroska: Add the WebM encrypted content support in matroskademux
This commit:

1. Reads the WebM and Matroska ContentEncryption subelements.

2. Creates a GST_PROTECTION event for each ContentEncryption, which
   will be sent before pushing the first source buffer.
   The DRM system id field in this event is set to GST_PROTECTION_UNSPECIFIED_SYSTEM_ID,
   because it isn't specified neither by Matroska nor by the WebM spec.

3. Reads the protection information of encrypted Block/SimpleBlock and
   extracts the IV and the partitioning format (subsamples).

4. Creates the metadata protection for each encrypted Block/SimpleBlock,
   with those informations: KeyID (extracted from ContentEncryption element),
   IV and partitioning format.

5. Adds a new caps for WebM encrypted content named "application/x-webm-enc",
   with the following new fields:

   "encryption-algorithm": The encryption algorithm used.
                           values: "None", "DES", "3DES", "Twofish", "Blowfish", "AES".

   "encoding-scope": The field that describes which Elements have been modified.
                     Values: "frame", "codec-data", "next-content".

   "cipher-mode": The cipher mode used in the encryption.
                  Values: "None", "CTR".

https://bugzilla.gnome.org/show_bug.cgi?id=765275
2018-10-03 16:59:14 +02:00
John Nikolaides
6fe214e7a9 splitmuxsink: Added a split-at-running-time action signal
The video file can now be split at an arbitrary time, given by the user
as an argument to the action signal.

https://bugzilla.gnome.org/show_bug.cgi?id=787922
2018-09-28 16:53:29 +03:00
Tim-Philipp Müller
506e080a15 rtpmp4gdepay: detect broken senders who send AAC with ADTS frames
Strip ADTS headers if we detect any, apparently some Sony cameras
send AAC with ADTS headers. We could also change the stream-format
in the output caps, but that would be unexpected to pipeline builders
and would not exactly be backwards compatible.
2018-09-26 12:25:24 +01:00
Tim-Philipp Müller
f255ea99f4 rtpmp4gdepay: factor out pushing of output buffer 2018-09-26 12:20:13 +01:00
Sebastian Dröge
d51139ad16 imagefreeze: Allow ANY capsfeatures 2018-09-26 13:30:04 +03:00
Philippe Normand
babf4210f0 qtdemux: PIFF track encryption box support
The PIFF track encryption box is a UUID box containing the default encryption
values that should be used for PIFF sample encryption.

https://bugzilla.gnome.org/show_bug.cgi?id=796647
2018-09-25 09:53:31 +01:00
Alicia Boya García
bc0ea0dbbb qtdemux: turn impossible condition into an assert
qtdemux_update_streams() is only ever called after checking
`qtdemux->streams_aware` is TRUE. There is no need to check for that
condition again.

`qtdemux->streams_aware` is only modified when the demuxer is
hard-resetted, which is mutually exclusive with demuxing, so it cannot
be modified during the call.

https://bugzilla.gnome.org/show_bug.cgi?id=797191
2018-09-24 08:33:02 +01:00
Alicia Boya García
7ceefec714 matroskademux: Emit no-more-pads after parsing Tracks
Currently matroskademux does not emit no-more-pads until the first
Cluster is parsed, even though the Tracks have already been parsed and
from that point on there can be no more tracks.

This is important in MSE because the browser needs to know when the MSE
initialization segment has been completely parsed so that it can expose
the tracks to the user. Some applications depend on this been done
before they feed frames to the demuxer.

As a consequence, historically WebKit has relied on hacks such as
listening to the `pad-added` event, which made impossible to support
multiple tracks in the same file. Let's fix that.

https://bugzilla.gnome.org/show_bug.cgi?id=797187
2018-09-21 17:41:57 -03:00
Alicia Boya García
0e60076a39 matroskademux: Parse successive Tracks elements
This patch allows matroskademux to parse a second Tracks element,
erroring out if the tracks are not compatible (different number, type or
codec) and emitting new caps and tag events should they have changed.

https://bugzilla.gnome.org/show_bug.cgi?id=793333
2018-09-21 17:27:57 -03:00
Alicia Boya García
f279bc5336 matroskademux: Refactor track parsing out from adding tracks
This splits gst_matroska_demux_add_stream() into:

* gst_matroska_demux_parse_stream(): will read the Matroska bytestream
  and fill a GstMatroskaTrackContext.

* gst_matroska_demux_parse_tracks(): will check there are no repeated
  tracks.

* gst_matroska_demux_add_stream(): creates and sets up the pad for the
  track.

https://bugzilla.gnome.org/show_bug.cgi?id=793333
2018-09-21 17:27:57 -03:00
Alicia Boya García
9dc7859184 matroskademux: Allow Matroska headers to be read more than once
This is necessary for MSE, where a new MSE initialization segment may be
appended at any point. These MSE initialization segments consist of an
entire WebM file until the first Cluster element (not included). [1]

Note that track definitions are ignored on successive headers, they must
match, but this is not checked by matroskademux (look for
`(!demux->tracks_parsed)` in the code).

Source pads are not altered when the new headers are read.

This patch has been splitted from the original patch from eocanha in [2].

[1] https://www.w3.org/TR/mse-byte-stream-format-webm/
[2] https://bug334082.bugzilla-attachments.gnome.org/attachment.cgi?id=362212

https://bugzilla.gnome.org/show_bug.cgi?id=793333
2018-09-21 17:27:57 -03:00
Mathieu Duponchelle
9605028a71 splitmuxsink: Implement split-after
The behaviour of split-now is to output the current GOP after
starting a new file.

The newly-added split-after signal will output the current GOP
to the old file if possible once a new GOP is opened.

https://bugzilla.gnome.org/show_bug.cgi?id=796982
2018-09-21 16:31:34 +02:00
Seungha Yang
916ad09009 flvmux: Don't leak codec_data buffer
Use gst_buffer_replace() to prevent buffer leak

https://bugzilla.gnome.org/show_bug.cgi?id=797179
2018-09-20 11:35:31 +03:00
Sebastian Dröge
c6e07a6eee qtmux: Set Closed Caption track width/height to that of the first video track
Otherwise software like Premiere or Final Cut Pro won't like our files.

https://bugzilla.gnome.org/show_bug.cgi?id=797111
2018-09-20 11:35:31 +03:00
Sebastian Dröge
54a5b3f845 qtmux: Initialize caption track language code to 0 instead of "und"
Without this, Final Cut considers it "non-standard" and 0 (english) is a
good default for closed captions.

https://bugzilla.gnome.org/show_bug.cgi?id=797111
2018-09-18 17:32:46 +03:00
Vivia Nikolaidou
94f8603411 qtmux: Allow up to 1 trak timescale unit of lateness in prefill mode
For 59.94 FPS, it's common to set 60000 as timescale. For that
timescale, if the audio is late by as little as 0:00:00.000016666
(definitely less than one audio sample), lateness gets rounded to 1.
Added a safeguard that allows lateness up to 1 sample with the specific
trak's timescale, to make sure that values less than e.g. one audio
sample won't break the prefill mode. What will happen in this case is
that the audio will get squeezed back to the video's timestamp, which in
practice means that the audio will be 0.000016666 seconds early (with
the patch).

https://bugzilla.gnome.org/show_bug.cgi?id=797133
2018-09-13 13:51:18 +03:00
Tim-Philipp Müller
0d88e3553a wavpackparse: fix handling of correction streams
Accept wavpack correction streams (.wvc) on sink pad, so
that wavpackparse can also be used to packetise correction
streams.

Fix parsing of subblock ID tags - the higher bits are
flags and are not part of the ID. This resulted in
correction blocks not being recognised properly and
the output not having the right (correction) caps.
2018-09-12 01:05:05 +01:00
Seungha Yang
1cd5a5241f flvmux: Don't omit streamheader from caps on downstream reconfigure
The reconfigured downstream elements (e.g., dynamically added sink element)
most likely require the flv streamheader

https://bugzilla.gnome.org/show_bug.cgi?id=797089
2018-09-06 15:52:09 -04:00
Martin Kelly
be05515da7 matroskamux: don't store used UIDs
Currently, whenever we generate a 128-bit UID, we store it in a list and
return 0 if we ever encounter a collision. This is so mathematically
improbable that it's not worth checking for, so we can save memory and
time by not tracking the UID. Even if a collision happened, a list of
only 10 UIDs would be unlikely to detect it.

This article has a good description of how improbable a collision is:
https://en.wikipedia.org/wiki/Universally_unique_identifier#Collisions

https://bugzilla.gnome.org/show_bug.cgi?id=797086
2018-09-06 14:14:56 -04:00
Sebastian Dröge
d9b52d1f5e qtmux: Use existing helper function to create "und" language code 2018-09-06 20:07:57 +03:00
Alicia Boya García
794dcfbfee qtdemux: Keep sample data from the current fragment only (push mode)
This patch clears the sample table whenever the demuxing of a new
fragment begins. This avoids increasing memory usage for long videos.
This behavior was already present when upstream_format_is_time; this
patch extends it to all push mode operation (e.g. Media Source
Extensions).

https://bugzilla.gnome.org/show_bug.cgi?id=796899
2018-09-03 12:38:17 -03:00
Sebastian Dröge
abf2af1a66 deinterlace: Reset frame/tc/caption pointer to NULL after moving it in the history 2018-08-31 14:26:32 +03:00
Edward Hervey
8b4db06733 rtpmp4vpay: Increase ranking
Both rtpmp4vpay and rtpmp4gpay support MPEG4 elementary streams. But
the most supported variant is the video-specific one (rtpmp4vpay),
therefore increase the rank of that one so that auto-plugging of
payloaders for MPEG4 elementary streams ends up picking that one
and not the generic one.
2018-08-29 09:51:42 +02:00
Tim-Philipp Müller
9d6621a30d matroskademux: implement keyframe search also without cluster prev size
If we have cluster prev size (GStreamer muxer will write it by default),
we can go back to the previous cluster efficiently, but if we don't then
just search backwards until we find a cluster ebml identifier, like we
do when searching for clusters in the bisection loop.
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
2d6efbbae2 matroskademux: make max backtrack distance for keyframe search configurable
Add property instead of hardcoding it in the code.

In some scenarios such as CCTV variable fps and extra long GOPs are
used to minimise storage space, for example. In those cases there might
not be any keyframes for many minutes, so provide a property to override
the max allowed distance.

https://bugzilla.gnome.org/show_bug.cgi?id=790696
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
2990f0730a matroskademux: set limit how much to backtrack to find a keyframe
If we seek without an index and land on a cluster that starts
with a delta frame.

https://bugzilla.gnome.org/show_bug.cgi?id=790696
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
ffb4533137 matroskademux: no need to search for keyframes for intra-only streams
If the video streams are all I-frame only then we don't need to look
for a cluster with a keyframe, we can just assume there will be one.

https://bugzilla.gnome.org/show_bug.cgi?id=790696
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
694631520d matroskademux: figure out if we have prev_size when starting up
This is useful to know in case someone initiates a seek or
direction change before we reach the second cluster.
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
43ce85f794 matroskademux: try to ensure keyframe when seeking without index
When seeking in pull mode without an index (because there is no index
or the file is still being written to) we bisect to find the right
cluster to jump to. However, it's possible the cluster we found doesn't
start with a keyframe, which leads to decoding errors, so if we know
that the found cluster starts with a delta frame try to scan back to
previous clusters until we find one that starts with a keyframe or
we are back at the beginning. Theoretically it's possible that all
clusters but the first one do not start with a keyframe and the
keyframes are in the middle of clusters, but this is extremely
unusual, so we will cover this case with a basic sanity check.

This problem is especially problematic with content recorded with
dynamic GOP and FPS, where long GOP lengths and low FPS may cause a
large set of clusters to lack key frames. Playback would then be
started on a non-keyframe cluster, and the large number of such frames
would make the content impossible to decode fo a long stretch of time.

Based on patch by: Mats Lindestam <matslm@axis.com>

https://bugzilla.gnome.org/show_bug.cgi?id=790696
2018-08-27 23:01:21 +02:00
Tim-Philipp Müller
93ddea2a70 matroskademux: extract cluster prevsize if available
This is useful for reverse playback/trickmodes
without an index, and will also be useful in the
seek handler if we need to scan back to find a cluster
that starts with a keyframe.

https://bugzilla.gnome.org/show_bug.cgi?id=790696
2018-08-27 23:00:53 +02:00
Tim-Philipp Müller
fd0afe033b matroska: fix handling of FlagInterlaced
This is an enum not a boolean, and a value of 2 signals
that the video is progressive, but we would mistakenly set
interlace-mode=mixed on the output caps.

https://bugzilla.gnome.org/show_bug.cgi?id=787206
2018-08-23 23:32:06 +02:00
Nicolas Dufresne
cf34d51318 udpsrc: Fix build when SO_RCVBUFFORCE is not defined
This shoudl fix the mingw build.
2018-08-17 17:35:43 -04:00
Nicolas Dufresne
c61bcb1fd2 udpsrc: Balance Linux value of get/set_rcvbuf
On Linux, the kernel returns twice the size as it will allocate extra
space for accouting. We devides this value by two in order to ensure
that get/set value now match. This fixes the set buffer size validation
and allow having a nice warning when the size if surpassed and the
process does not have CAP_NET_ADMIN capabilities.

https://bugzilla.gnome.org/show_bug.cgi?id=727067
2018-08-17 14:24:25 -04:00
Guillaume Desmottes
9b4a022dee updsrc: set udp buffer size forcibly
The udp buffer size is limited to a maximum of around 100K.
Some apps need to set the force bufsize for their own operation.
Use the SO_RCVBUFFORCE option in order to override the rmem_max limit
of linux kernel. Require user to have the CAP_NET_ADMIN privilege to
work.

Original patch from Kyungnam Bae <kyungnam.bae@lge.com>

https://bugzilla.gnome.org/show_bug.cgi?id=727067
2018-08-17 13:46:42 -04:00
Guillaume Desmottes
58a5f59fe9 udpsrc: factor out gst_udpsrc_get_rcvbuf()
No semantic change.

https://bugzilla.gnome.org/show_bug.cgi?id=727067
2018-08-17 13:46:42 -04:00
Mathieu Duponchelle
bfcddb7125 mp4 robust muxing: improve documentation and logging 2018-08-16 19:43:50 +02:00
Jan Alexander Steffens (heftig)
20758215b5 rtph26*pay: Update param set timestamp even if parameters unchanged
rtph264pay and rtph265pay skip updating the parameter set timestamp if
the units they see contain no new configuration. This can result in
them injecting duplicate parameters.

https://bugzilla.gnome.org/show_bug.cgi?id=796748
2018-08-16 16:49:16 +03:00
Ulf Olsson
0f6a4e7c98 rtspsrc: Add support for SET_PARAMETER and GET_PARAMETER using signals
https://bugzilla.gnome.org/show_bug.cgi?id=792131
2018-08-16 09:03:42 +03:00
Jan Schmidt
5da08a21c6 splitmuxsink: Don't leak old muxer/sink in async mode
Make sure to clear the reference taken earlier in the function
when switching muxer/sink asynchronously so they don't leak
2018-08-15 02:28:20 +10:00
Jan Schmidt
f8076e208b splitmuxsink: Fix reference counting loop
The stream context was holding a reference to the
internal queue and pads, with pad probes that were
in turn holding references to the stream context.

This lead to a leak if the request pads weren't explicitly
released.

https://bugzilla.gnome.org/show_bug.cgi?id=796893
2018-08-15 02:12:53 +10:00
Roland Jon
7fe18db041 aacparse: fix codec_data buffer leak
https://bugzilla.gnome.org/show_bug.cgi?id=740101
2018-08-13 00:05:59 +01:00
Sebastian Dröge
5e4f215bfb qtdemux: Handle closed captions as subtitle streams 2018-08-02 16:13:26 +03:00
Sebastian Dröge
7a354d0507 qtmux: The sample size we have to reserve is 256+8 bytes for the header for CDP packets 2018-08-02 13:40:09 +03:00
Sebastian Dröge
8a3075fbae qtmux: Properly allocate 256 bytes per CDP packet in prefill mode
Instead of allowing 256 but only pre-allocating 100.
2018-08-02 12:27:45 +03:00
Sebastian Dröge
bbfd76e94e Revert "qtmux: Allow for CDP packets up to 320 bytes"
This reverts commit 5eed1d49bd.

255 is actually the maximum, there's a bug if more is arriving.
2018-08-02 12:27:17 +03:00
Sebastian Dröge
5eed1d49bd qtmux: Allow for CDP packets up to 320 bytes
Apparently they can be bigger than 256 bytes sometimes.
2018-08-01 16:51:11 +03:00
Nicolas Dufresne
2b11c62571 rtppayload: Fix VP8/VP9/OPUS dual encoding name handling
All these were copy pasted and would lead to assertion when chained with
rtpmux. This commit rewrite the negotiation with downstream. This also
drop the fallback to ancient names if the pad is unlinked. This was
completly arbitrary decision that made no sense.

https://bugzilla.gnome.org/show_bug.cgi?id=796809
2018-08-01 09:42:36 -04:00
Sebastian Dröge
7679412b69 qtdemux: Don't assert in prefill mode if a track has no samples at all
Just write it with a duration of 0, no samples, etc.
2018-07-31 12:47:47 +03:00
Sebastian Dröge
6ce2a5f7bf qtdemux: Don't assert if a file does not have any active streams
** (gst-play-1.0:9113): CRITICAL **: 12:31:54.360: qtdemux_is_streams_update: assertion 'qtdemux->active_streams != NULL' failed
2018-07-31 12:33:54 +03:00
Sebastian Dröge
e58dceec3a qtmux: Reserve 256 bytes for CDP packets in pre-fill mode
92 is sometimes too small and compared to the wasted space for other
codecs 256 bytes is small (and should be the maximum CDP packet size)
2018-07-30 13:35:02 +03:00
Nirbheek Chauhan
8f807477eb meson: Add feature options for all plugins
Checks for GL, Qt5, and C++ are still automagic. FIXMEs have been
added for these so they can be fixed later.

https://bugzilla.gnome.org/show_bug.cgi?id=795107
2018-07-27 18:42:54 +05:30
Vivia Nikolaidou
f2c2560db2 deinterlace: Closed caption pass-through
Pass through closed caption data when deinterlacing. When two
deinterlaced frames are created for the same interlaced frame (e.g.
fields=all), the second of the two frames will have no closed caption
data.

Also fixed memory leaks related to timecode meta pass-through.

https://bugzilla.gnome.org/show_bug.cgi?id=796876
2018-07-27 14:26:49 +03:00
Olivier Crête
6c478453a8 qtmux: Implement muxing of AV1 into MP4 files
According to
https://aomediacodec.github.io/av1-isobmff/
2018-07-26 14:14:53 -04:00
Olivier Crête
d641b37fcb matroskamux: Put codec_data as CodecPrivate for AV1 2018-07-26 14:14:53 -04:00
Olivier Crête
e3b14e345d matroskamux: Accept muxing AV1 2018-07-26 14:14:53 -04:00
Olivier Crête
e07ce9afb4 qtdemux: Recognize more AV1 atoms 2018-07-26 14:14:53 -04:00
Olivier Crête
415caf4b87 matroskademux: Extract codec_data for AV1
According to
https://github.com/Matroska-Org/matroska-specification/blob/av1-mappin/codec/av1.md
2018-07-26 14:14:53 -04:00
Olivier Crête
43ee4055c3 qtdemux: Extract AV1 codec_data and put it in the caps
Also extract the presentation-delay and put it in the caps.
2018-07-26 14:14:53 -04:00
Olivier Crête
b6e8c6323e qtdemux: Add initial support for AV1 demuxing
Following the spec at
https://aomediacodec.github.io/av1-isobmff/
2018-07-26 14:14:53 -04:00
Jan Schmidt
f067b50dd6 rtspsrc: Add a small configurable teardown delay
This causes rtspsrc to send a teardown and wait on
PAUSED->READY transition, with a configurable delay.
Otherwise, typically teardown never gets sent in
playbin / uridecodebin where the transition back to NULL
happens too quickly.

The timeout is set to 100ms default.

https://bugzilla.gnome.org/show_bug.cgi?id=751994
2018-07-27 00:41:57 +10:00
Sebastian Dröge
9a80cdbb40 rtpgstpay: Add support for force-keyunit events
This triggers immediate re-sending of the configuration data in-band.

https://bugzilla.gnome.org/show_bug.cgi?id=796877
2018-07-26 16:54:28 +03:00
Sebastian Dröge
f3631e6837 rtp: Use running_time instead of PTS for config-interval calculations
PTS can start again from a different offset while the running time is
increasing. The only thing that matters here is the running time.

https://bugzilla.gnome.org/show_bug.cgi?id=796807
2018-07-24 18:14:28 +03:00
Michael Olbrich
bd05ab8358 rtpL8pay: don't try to modify a read-only structure
Just remove the code. It's not doing anything useful anyways. The modified
caps are the result of a caps query, so either not used afterwards of a
reference to some internal caps of another element that should not be
modified.

https://bugzilla.gnome.org/show_bug.cgi?id=796837
2018-07-19 14:07:03 -04:00
Vivia Nikolaidou
ceac1b51b1 deinterlace: Timecode pass-through
When it is trivial to pass-through a timecode, by only removing the
"interlaced" flag, do pass-through. Otherwise, double the fps_n and
adjust the "frames" field.

https://bugzilla.gnome.org/show_bug.cgi?id=796818
2018-07-17 16:29:37 +03:00
Jan Schmidt
f672116c72 splitmux: Improve handling of repeated timestamps
When handling input with timestamps that repeat, sometimes
splitmuxsink would get confused and ignore a keyframe.

The logic in question is a holdover from before the cmd queue
moved the file cutting to the multiqueue output side and made
it deterministic, so it's no longer needed on the input
here.

https://bugzilla.gnome.org/show_bug.cgi?id=796773
2018-07-17 10:57:42 +10:00
Jan Schmidt
f667ab05d2 Revert "splitmuxsrc: Make sure events are writable"
This reverts commit 3ac5430311.

There's no need to make a freshly created event writable,
and the other half of this patch was already fixed
and pushed in f2f15a1
2018-07-17 01:33:55 +10:00
Jan Schmidt
3ac5430311 splitmuxsrc: Make sure events are writable
Before setting the seqnum on events sent downstream,
make sure they are writable.
2018-07-17 00:17:18 +10:00
Havard Graff
ac6e77acad rtpsession: Don't start the RTCP thread until it's needed
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.

We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-07-12 18:37:33 +02:00
Thibault Saunier
57be4d2da3 scaletempo: Mark as Audio in classification 2018-07-08 20:10:12 -04:00
Edward Hervey
9be72b585d qtdemux: Store and propagate SEGMENT sequence numbers
* When receiving a segment in TIME, use that seqnum
* Only reset the stored sequence number when doing HARD reset
  (and not when we get a FLUSH event from upstream)
2018-07-06 15:21:33 +02:00
Philippe Normand
4c10a1ba27 matroskademux: Set subtitle tag title from TrackName field
GUI applications can then use the title tag to set menu items or labels
representing the track.

https://bugzilla.gnome.org/show_bug.cgi?id=796567
2018-06-29 17:11:28 +01:00
Alicia Boya García
025a430d08 qtdemux: rework segment event pushing, again
This patch aims at fixing the recent regressions in the adaptive test
suite.

All segment pushing in push mode is now done with
gst_qtdemux_check_send_pending_segment(), which is idempotent and
handles both edit lists cases and cases where the upstream TIME segments
have to be sent directly.

Fragmented files that start with a non-zero tfdt are also taken into
account, but their handling has been vastly simplified: now they are
handled as implicit default seeks so there is no need to extend the
GstSegment formulas as was being done before.

qtdemux->segment.duration is no longer modified when
upstream_format_is_time, respecting in this way the durations provided
by dashdemux and fixing bugs in reverse playback tests where mangled
durations appeared in the emitted segments.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-06-29 09:19:31 -04:00
Alicia Boya García
2c394304b7 qtdemux: Don't send EOS during upstream reverse playback
Upstream driving elements such as dashdemux often do reverse playback by
feeding qtdemux with the fragments containing the requested playback
range in reverse order.

But the requested playback range stop may be somewhere in the
middle of a fragment. In that case, a naive pts >= segment.stop
condition may declare end of segment prematurely when demuxing this
first fragment.

This used not to happen because there were places in moov parsing where
segment.stop was overwritten to GST_CLOCK_TIME_NONE even if
upstream_format_is_time -- resulting in this case in a segment with rate
< 0 and stop == -1 and hence not triggering the EOS check, but that was
likely an accident.

This patch modifies the EOS check to take this case into account, not
sending EOS when upstream_format_is_time if rate < 0.

This fixes adaptive.dash.playback.seek_end_live.DASHIF_livestream_testpic_2s

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-06-29 09:19:31 -04:00
Tim-Philipp Müller
238a37295c Update for g_type_class_add_private() deprecation in recent GLib
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
2018-06-23 23:44:19 +02:00
Sebastian Dröge
f2f15a119f splitmuxsrc: Make sure events are writable before setting their seqnum 2018-06-18 12:13:48 +03:00
Thibault Saunier
1840159d0b qtdemux: Do not set INVALID seqnum on events 2018-06-06 11:44:33 -04:00
Seungha Yang
6b29a4507e qtdemux: Forward upstream time-format segment without mapping
Sample table based segment event (genereted by qtdemux) could break
presentation timeline. For example, qtdemux should not modify upstream
time format segment (e.g., adaptivedemux use case)

https://bugzilla.gnome.org/show_bug.cgi?id=796480
2018-06-06 10:54:55 -04:00
Edward Hervey
418a6270e5 rtspsrc: Seek handling is always done with a valid event
Remove the checks
2018-06-06 07:48:32 +02:00
Edward Hervey
14a9cc603e wavparse: Don't set invalid seqnum on events
Some codepath will call gst_wavparse_perform_seek without an event
and therefore without a valid seqnum
2018-06-06 07:48:32 +02:00
Thibault Saunier
0fbde2a079 qtdemux: Clarify field name about stream-encryption-system
This field is actually only informatory and the user can potentially
choose something else. EME tests in WebKit testsuite actually doesn't
take it into and force another encryption system to be used, and expects
to be given the occasion to do so.

This basically also reverts 3e063703b3.
2018-05-28 16:52:10 -04:00
Thiago Santos
ae7b531902 qtdemux: mark segment as sent after pushing when moov is received
Otherwise we would try to send it a second time if the same moov is
received or in any other situation that might trigger segment sending.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-05-28 11:26:36 -07:00
Thiago Santos
364a921d3f qtdemux: do not update segment.stop is it is not a valid time
Otherwise it overflows and starts having a meaningful and wrong value.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-05-25 08:34:29 -07:00
Thiago Santos
9c514f9c7c qtdemux: offset edts segments by the min timestamp of the stream
Otherwise if the stream is starting at timestamp=X it would wait
'X' to start playing.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-05-25 08:34:29 -07:00
Thiago Santos
bfd0e0225e qtdemux: rework segment event pushing
Instead of always keeping a safe segment (start=0) event from the beginning,
delay the creation of this event to when we really know the timestamp of the
first sample. This is important to properly start fragmented streams that
we might join in the middle or to play isolated fragment files that might
have an advanced tfdt.

https://bugzilla.gnome.org/show_bug.cgi?id=752603
2018-05-25 08:34:29 -07:00
Thibault Saunier
43a540b1cd qtdemux: Do not unref a NULL stream_tags
stream->stream_tags is reset to NULL once we expose the stream and
these have been consumed, we need to check that when cleaning up
the stream.
2018-05-25 10:54:54 +02:00
Thibault Saunier
3e063703b3 qtdemux: Do not run the preferred decryptor context query if no decryptor avalaible
Ultimately this avoids a segfault as the code expect a non NULL array
here.
2018-05-25 10:53:16 +02:00
Alicia Boya García
c2a0da8096 qtdemux: Allow edit lists on fragmented files on push mode
Fragmented files often use elst.duration=0 which before
ee78825eae was wrongly interpreted as
having no frames.

Since that issue has now been fixed, there is no reason to disable edit
lists in fragmented files. This commit enables them, therefore producing
correct stream time for files containing edit lists.

https://bugzilla.gnome.org/show_bug.cgi?id=793058
2018-05-24 15:53:48 +02:00
Alicia Boya García
5388d6e6a2 qtdemux: fix computation of first_duration for fragmented files in push mode
Since ca068865c3 the duration of the first
frame is not used for estimating the frame rate.

For this purpose, stream->first_duration was initialized with the
duration of the first frame. In fragmented files, this was previously
done by peeking the first moof, but that can only be done in pull mode.

Fortunately, we don't really need to do that, at least with the current
design: When we are estimating the frame rate we already have the
sample table, regardless of the scheduling mode and whether the file is
fragmented or not, so we can obtain first_duration there much more
reliably.

This fixes frame rate estimation for fragmented files in push mode.

https://bugzilla.gnome.org/show_bug.cgi?id=796384
2018-05-24 15:53:48 +02:00
Vivia Nikolaidou
d11339d616 splitmuxsink: Added new async-finalize mode
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.

This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.

https://bugzilla.gnome.org/show_bug.cgi?id=783754
2018-05-24 12:47:24 +03:00
Alicia Boya García
d35f893715 qtdemux: Don't send gaps bigger than 1 second (now in push mode too)
This applies the same workaround to gaps that is being used in pull
mode.

https://bugzilla.gnome.org/show_bug.cgi?id=778426
2018-05-24 11:09:53 +02:00
Seungha Yang
f61c2bc301 qtdemux: Properly handle edit list in push mode
If there are empty segments in edit list, demux should
adjust "accumulated_base" to apply it into running time.

https://bugzilla.gnome.org/show_bug.cgi?id=778426
2018-05-24 11:09:53 +02:00
Mathieu Duponchelle
d9235404aa matroska-mux: write colorimetry
This is a straightforward translation of 5dd39d8, can be trivially
checked by running:

gst-launch-1.0 -v videotestsrc ! video/x-raw, colorimetry=2:4:7:1 ! \
matroskamux ! matroskademux ! fakesink

and verifying that the colorimetry is correctly preserved.

https://bugzilla.gnome.org/show_bug.cgi?id=796344
2018-05-23 15:14:23 +02:00
Alicia Boya García
ee78825eae qtdemux: fix buggy duration in edits with duration=0 in fragmented files without a mehd
https://bugzilla.gnome.org/show_bug.cgi?id=794858
2018-05-23 14:25:56 +02:00
Tim-Philipp Müller
db688c5504 docs: fix typos 2018-05-23 13:14:27 +01:00
Seungha Yang
c69ccf60d8 qtdemux: Clarify variable name
As defined by spec, use "empty edit". It's more straightforward.

https://bugzilla.gnome.org/show_bug.cgi?id=778426
2018-05-22 11:29:44 +02:00
Xabier Rodriguez Calvar
ee4b45da24 qtdemux: add context for a preferred protection
qtdemux selected the first system corresponding to a working GStreamer
decryptor. With this change, before selecting that decryptor, qtdemux
will check if it has context (a preferred decryptor id) and if not, it
will request it.

The request includes track-id, available key system ids for the
available decryptors and even the events so that the init data is
accessible.

[eocanha@igalia.com: select the preferred protection system even if not available]

Test "4. ClearKeyVideo" in YouTube leanback EME conformance tests 2016 for
H.264[1] uses a media file[2] with cenc encryption which embeds 'pssh' boxes
with the init data for the Playready and Widevine encryption systems, but not
for the ClearKey encryption system (as defined by the EMEv0.1b spec[3] and with
the encryption system id defined in [4]).

Instead, the ClearKey encryption system is manually selected by the web page
code (even if not originally detected by qtdemux) and the proper decryption key
is dispatched to the decryptor, which can then decrypt the video successfully.

[1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false
[2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/car_cenc-20120827-86.mp4
[3] https://dvcs.w3.org/hg/html-media/raw-file/eme-v0.1b/encrypted-media/encrypted-media.html#simple-decryption-clear-key
[4] https://www.w3.org/Bugs/Public/show_bug.cgi?id=24027#c2

https://bugzilla.gnome.org/show_bug.cgi?id=770107
2018-05-21 17:36:17 +02:00
Enrique Ocaña González
844423ff99 qtdemux: also push buffers without encryption info instead of dropping them
Test "17. PlayReadyH264Video" in YouTube leanback EME conformance tests 2016
for H.264[1] uses a media file[2] with cenc encryption whose first two 'moof'
boxes have no encryption information (no 'saiz' and 'saio' boxes).

Those boxes are actually not encrypted and the current qtdemux implementation
was just dropping them, breaking the test use case.

This patch detects those kind of situations and just lets the unencrypted
buffers pass. Of course, this needs some collaboration by the decryptors,
which should also do the same and not to try to decrypt those clear buffers.

[1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false
[2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/oops_cenc-20121114-142.mp4

https://bugzilla.gnome.org/show_bug.cgi?id=770107
2018-05-21 17:36:17 +02:00
Alicia Boya García
c097289725 qtdemux_parse_segments: remove superfluous variable
https://bugzilla.gnome.org/show_bug.cgi?id=793751
2018-05-18 11:29:47 -04:00
Olivier Crête
87b2b35fac flvmux: Remove custom get_next_time implementation
GstAggregator now does the same thing in the simple implementation.

https://bugzilla.gnome.org/show_bug.cgi?id=795486
2018-05-16 22:23:36 +02:00
Havard Graff
4d54673cb4 rtpsession: make "clear-pt-map" action signal actually work
Needed for PLI + FIR unit tests in follow-up commit.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:52:14 +01:00
Mikhail Fludkov
40eb462591 rtpsession: Avoid unnecessary copy of stats structure
The code before copied GstStructure twice. The first time inside
gst_value_set_structure and the second time in g_value_array_append.
Optimized version does no copies, just transfers ownership to
GValueArray. It takes advantage of the fact that array has already
enough elements preallocated and the memory is zero initialized.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:33:01 +01:00
Tim-Philipp Müller
d5d6e6cfdd Revert "BugFix : Change peak value to normalize audio file with fallback gain"
This reverts commit 36e49fd6f8.

Breaks unit test, someone needs to investigate if it's the
patch's fault or if the test needs adjusting/updating.

https://bugzilla.gnome.org/show_bug.cgi?id=673970
2018-05-15 10:35:09 +01:00
Stian Selnes
457fdf95c4 rtpsession: Drop packet if trying to send from non-internal source
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 10:34:29 +01:00
Tim-Philipp Müller
adc315978c matroskademux: tag disabled streams with FLAG_UNSELECT
So they're never picked as default, only by explicit
user action.

https://bugzilla.gnome.org/show_bug.cgi?id=690911
2018-05-15 10:25:35 +01:00
Sebastian Dröge
6e7b0d6061 qtmux: Print expected/actual values in debug log on mismatch in prefill mode
This helps debugging a lot.
2018-05-14 21:06:55 +03:00
Havard Graff
b43ee8f5b1 rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc
Some RTP stacks out there does not set the sender_ssrc. In order to be
more robust, try to lookup the media_ssrc before dropping the PLI.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:41:39 +01:00
Mikhail Fludkov
386ca1d378 rtpsession: Fix on-feedback-rtcp race
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.

The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:33:56 +01:00
Stian Selnes
29f26e8768 rtpsession: Add missing lock around sess->ssrcs iteration
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:17:02 +01:00
John-Mark Bell
0a2b55ac3c rtpsession: do not emit RBs for internal senders.
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:16:59 +01:00
Vivia Nikolaidou
498ebc5503 splitmuxsink: Added caption_%u pad template
For closed-caption-enabled muxers (e.g. qtmux)
2018-05-11 16:39:46 +03:00
Edward Hervey
4253c51c5f qtdemux: Initialize riff library
Avoids debugging message issues. Also just use the main riff header
2018-05-10 13:59:36 +02:00
Seungha Yang
7651036530 qtdemux: Protect _expose_streams() from flush event
Flush during stream change can break autoplugging or the
flush event could be dropped.

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
d72a7c038c qtdemux: Try to expose whenever got new moov or new stream-start
Whenever got new moov or new stream-start,
demux will try to expose new pad by following rule.

Comparing stream-id in the current moov with previous one, then
* If matched stream-id is found from previous one,
  reuse existing pad (most common case)
* Otherwise, expose new pad with new stream-start
* No more used stream will be freed

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
0bd3459053 qtdemux: Remove duplication of initializing member variables
Most initialization of variables in gst_qtdemux_init() are duplicated in
gst_qtdemux_reset() function.

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
553607756b qtdemux: Create stream whenever got new moov
Whenever demux got moov, demux will create new stream. Only exception is
duplicated track-id in a moov box. In that case the first stream
will be accepted. This patch is pre-work for rework of moov handling.

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
96a6aab6a1 qtdemux: Store stream-id to manage streams
In order to figure out stream change such as
track-id change or stream-id change, demux will store
 stream-id per QtDemuxStream structure.

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
b5245be1b1 qtdemux: Use GList to manage QtDemuxStream
* Move to GList from static array
* Logging track-id instead of array index. It's more meaningful.

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
aca8d32437 qtdemux: Adjust the number of args of some functions
To be used with g_list_free_full in the next patch

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Seungha Yang
80f391aa3b qtdemux: Add parentheses in macro
https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Marinus Schraal
c2cce8a9fc isomp4: Use full date time if available
The ©day tag contains a full date time, use it for the DATE_TIME tag
instead of just the DATE tag. This overrules the unreliable qt creation
time.

https://bugzilla.gnome.org/show_bug.cgi?id=731029
2018-05-09 15:49:30 +02:00
Jan Schmidt
9cbde904e8 rtspsrc: Fix doc comment markers 2018-05-07 01:40:05 +10:00
Kyrylo Polezhaiev
70a55f90a3 icydemux: avoid timestamp field initialisation for tag event
This field is not used and will be removed in 2.0 API.

https://bugzilla.gnome.org/show_bug.cgi?id=761462
2018-05-07 01:25:25 +10:00
Matej Knopp
61ba778347 dcaparse: do not accept header with invalid channel count
https://bugzilla.gnome.org/show_bug.cgi?id=737928
2018-05-06 11:08:20 +02:00
Anthony Violo
36e49fd6f8 BugFix : Change peak value to normalize audio file with fallback gain
https://bugzilla.gnome.org/show_bug.cgi?id=673970
2018-05-05 16:49:57 +02:00
Tim-Philipp Müller
5b8e775d1c rtpvrawpay: don't use buffer lists if everything fits into one buffer
People might use very large mtu sizes where every payload
fits into a single output packet.

https://bugzilla.gnome.org/show_bug.cgi?id=795758
2018-05-05 16:32:59 +02:00
Olivier Crête
c2c7d110e5 flvmux: Don't wake up the muxer unless there is data
https://bugzilla.gnome.org/show_bug.cgi?id=795332
2018-04-26 15:41:54 -04:00
Olivier Crête
11297c3337 flvmux: Save the current position in the output segment
https://bugzilla.gnome.org/show_bug.cgi?id=795332
2018-04-26 15:41:54 -04:00
Olivier Crête
168fae813b flvmux: Wait for caps from both srcs before writing header
Wait for caps on all pads to start writing data even when source is live.

Includes unit test by Havard Graff that simulates it.

https://bugzilla.gnome.org/show_bug.cgi?id=794722
2018-04-26 15:41:54 -04:00
Xavier Claessens
edd9c8f6b8 Meson: Generate pc file for all plugins in good
https://bugzilla.gnome.org/show_bug.cgi?id=794568
2018-04-25 11:07:06 +01:00
Edward Hervey
90244da780 qtmux: Read caption from input buffer
And not from unallocated output buffer GstMapInfo

CID #1435131
2018-04-25 10:37:40 +02:00
Edward Hervey
7378f1b4fd isomp4: qtmux: Add Closed Caption support
Supports CEA 608 and CEA 708 CC streams

Also supports usage in "Robust Prefill" mode if the incoming caption
stream is constant (i.e. there is one incoming CC buffer for each
video frame).

https://bugzilla.gnome.org/show_bug.cgi?id=606643
2018-04-23 15:58:10 +02:00
Edward Hervey
76e32ef414 isomp4: Make 'gmhd' atom usage more generic
Only the 'gmin' atom is required. Any other entry within it are
optional.

https://bugzilla.gnome.org/show_bug.cgi?id=606643
2018-04-23 15:58:10 +02:00
Mathieu Duponchelle
90f5ae8f45 ulpfecdec: output perfect seqnums
ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.

rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:

* The protection packets don't make it as far as the depayloader,
  which means it will mark buffers as DISCONT every time the previous
  packets were protected

* While we could work around the previous issue by looking at
  the protection packets ignored and dropped in rtpptdemux, we
  would still mark buffers as DISCONT when a FEC packet was lost,
  as we cannot know that it was indeed a FEC packet, even though
  this should have no impact on the decoding of the stream

With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.

https://bugzilla.gnome.org/show_bug.cgi?id=794909
2018-04-19 18:17:39 +02:00
Sebastian Dröge
6d92fcd043 Revert "rtspsrc: Fix up sendonly/recvonly attribute handling"
This reverts commit af273b4de9.

While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.

Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.

https://bugzilla.gnome.org/show_bug.cgi?id=793964
2018-04-17 17:58:01 +03:00
Sebastian Dröge
336a97157c flacparse: Drain the parser when a CAPS event is received
After a CAPS event, in theory a new stream can start and it might start
with the FLAC headers again. We can't detect FLAC headers in the middle
of the stream, so we drain the parser to be able to detect either FLAC
headers after the CAPS event or the continuation of the previous stream.

This fixes for example

gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \
    audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \
    concat name=c ! rtpgstpay ! udpsink host=127.0.0.1 port=5000

gst-launch-1.0 udpsrc multicast-group=127.0.0.1 port=5000 \
    caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \
    rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin
2018-04-16 21:29:57 +03:00
Mathieu Duponchelle
fc9db36a1f flvmux: unref return of aggregator_pad_peek_buffer
We ended up leaking every single buffer going through the
muxer, which is far from ideal
2018-04-13 23:01:20 +02:00
Mathieu Duponchelle
ee99cd7037 qtmux: Fix leak
gst_qt_mux_can_renegotiate () gets called everywhere following
that pattern:

return gst_qt_mux_can_renegotiate (ref(self));

This means the reference must be released both in the success
and failure cases, it was only done in the success case.
2018-04-13 22:49:43 +02:00
Mathieu Duponchelle
acb6090e47 flvmux: aggregate should not push EOS itself
Instead it is expected to return GST_FLOW_EOS, and let the
base class handle that.
2018-04-13 22:44:14 +02:00
Alicia Boya García
a5e93b7062 matroskademux: Add comment about Opus clipping
https://bugzilla.gnome.org/show_bug.cgi?id=793523
2018-04-13 09:50:35 +01:00
Sebastian Dröge
c84f4e0491 monoscope: Only fixate pixel-aspect-ratio if the field exists 2018-04-11 21:42:16 +02:00
Vivia Nikolaidou
7d7c90bf5c splitmuxsink: Don't send fragment-opened-closed message if the reference ctx is NULL
It can happen during teardown that the reference context becomes NULL.
In that case, trying to send the fragment-opened-closed message would
lead to a crash.
2018-04-11 17:59:21 +03:00
Sebastian Dröge
6d7242925b splitmuxsink: Run gst_iterator_foreach() as long as it returns GST_ITERATOR_RESYNC
CID 1434160
2018-04-11 09:21:19 +02:00
Edward Hervey
2662f58cc9 qtdemux: Fix comparision for extra caption atom
We want to make sure we have *enough* data for the potential 2nd
caption atom.

CID #1434161
2018-04-11 08:51:32 +02:00
Edward Hervey
8213fc2edf qtdemux: Handle bogus caption samples
Corrupted files could potentially have multiple cdat/cdt2 atoms in
a sample entry, which is unclear how to handle.

Ignore repeated ones.

CID #1434162
CID #1434159
2018-04-11 08:45:06 +02:00
Sebastian Dröge
21eb9b49be monoscope: Fixate pixel-aspect-ratio too and make sure the final caps are completely fixated
Otherwise e.g. this fails with assertions:
gst-launch-1.0 audiotestsrc ! audioconvert ! monoscope ! videoconvert ! \
    videoscale ! video/x-raw,width=800,height=600 ! ximagesink
2018-04-10 21:17:30 +02:00
Edward Hervey
7229bfd068 qtmux: Add comments and doc about prefill mode 2018-04-09 15:25:50 +02:00
Edward Hervey
cc0c27850a qtmux: Refactor pad re-negotiation code
It was similar for all pads

https://bugzilla.gnome.org/show_bug.cgi?id=606643
2018-04-09 15:25:50 +02:00
Edward Hervey
2869edeea2 qtdemux: Detect and expose CEA 608/708 Closed Caption tracks
https://bugzilla.gnome.org/show_bug.cgi?id=606643
2018-04-09 15:25:50 +02:00
Mathieu Duponchelle
8270cbacb4 rtxsend: fix wrong memory layout assumption
The code responsible for creating retransmitted buffers
assumed the stored buffer had been created with
rtp_buffer_new_allocate when copying the extension data,
which isn't necessarily the case, for example when
the rtp buffers come from a udpsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=794958
2018-04-06 20:25:04 +02:00
Mathieu Duponchelle
4f8b34ab85 rtpbin: new signal "get-storage"
Similar to the get-session and get-internal-session signals,
we expose a get-storage signal in addition to the
get-internal-storage signal to give access to the actual
element for applications that need to set properties on the
element, in particular "size-time"

https://bugzilla.gnome.org/show_bug.cgi?id=794910
2018-04-06 20:21:43 +02:00
Sebastian Dröge
0f1770f2bd splitmuxsink: Add new reset-muxer property
With this the muxer is not set to NULL after each segment but instead
only flush events are sent to it to reset the EOS state.

As a result, the muxer will keep stream state and e.g. mpegtsmux will
keep the packet continuity counter continuous between segments as needed
by hlssink2.

https://bugzilla.gnome.org/show_bug.cgi?id=794816
2018-04-03 14:37:44 +03:00
Mathieu Duponchelle
893d39cef7 rtspsrc: reject segment seeks
https://bugzilla.gnome.org/show_bug.cgi?id=784681
2018-03-26 21:13:12 +02:00
Edward Hervey
d7038b9bb0 qtdemux: Handle variant of vorbis in mp4
Comes from gpac apparently. The codec_data uses the same packing
mechanism as matroska.

https://bugzilla.gnome.org/show_bug.cgi?id=738244
2018-03-22 15:22:33 +01:00
Edward Hervey
60bc77cc35 qtdemux: Check sample count is valid in PIFF parsing
The value stored in cenc_aux_sample_count wasn't in sync with the
parsing code that followed which checks whether all entries are
valid and present.

Only write the actual sample count when we know for sure.

CID #1427087
2018-03-22 15:22:24 +01:00
Petr Kulhavy
738eb0d8ed udpsrc: switch to using a buffer pool
This exposes a new property, mtu, which is used to determine the
initial size of buffers from the buffer pool. If received data
exceeds this, the element gracefully handles that in a manner similar
to what we had previously: a large memory gets filled and reallocated
at the next call to "fill".

The default size is set to 1500, which should cover most use cases.

With contributions from Mathieu Duponchelle <mathieu@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=772841
2018-03-21 17:47:27 -04:00
Petr Kulhavy
589019d8f5 udpsrc: optimize GstUdpSrc object for cache performance
Optimize GstUdpSrc for cache performance.

Move the hot properties, which are used by the read function, to the top:
@used_socket, @addr, @cancellable, @skip_first_bytes, @timeout,
@retrieve_sender_address.

Remove the unused property @ttl.

Where needed reorder so that holes are avoided (the 64-bit @timeout)

https://bugzilla.gnome.org/show_bug.cgi?id=772841
2018-03-21 17:47:26 -04:00
Sebastian Dröge
850e678813 qtdemux: Fix seeking on streams with frame reordering
The samples table is sorted by DTS, not PTS. As such we can only get the
correct result when using a binary search on it, if we search for the
DTS.
Also if we only ever search for the frame, where the following frame is
the first one with a PTS after the search position, we will generally
stop searching too early if frames are reordered.

In forwards playback this is not really a problem (after the decoder
reordered the frames, clipping is happening), in reverse playback
it means that we can output one or more frames too few as we stop too
early and the decoder would never receive it.

https://bugzilla.gnome.org/show_bug.cgi?id=782118
2018-03-20 12:08:28 +02:00
Sebastian Dröge
ed2ccb1a60 rtp: Fix compilation with non-C99 compilers
By moving variable declarations out of loop headers.
2018-03-20 12:08:28 +02:00
Tim-Philipp Müller
56577236f8 rtpulpfecdec: fix build with older gcc
As on Ubuntu Trusty.

https://bugzilla.gnome.org/show_bug.cgi?id=794493
2018-03-19 18:39:08 +00:00
Sebastian Dröge
42f5f3d1be splitmuxsink: Allow splitting at exactly the time/bytes threshold
76e458a119 changed the conditions from
"queued > threshold" to "queued >= threshold", which broke hlssink2 and
resulting in too small fragments being created although keyframes would
be at *exactly* the configured threshold.

https://bugzilla.gnome.org/show_bug.cgi?id=794440
2018-03-19 12:23:47 +02:00
Tim-Philipp Müller
47ff21ea3b rtpulpfec: fix unconditional use of __attribute__ ((packed))
Fix compilation with MSVC. We still assume that attribute
is supported by all other relevant compilers, which seems
to be the case since we haven't had any complaints about
similar code in rtpsbcpay.
2018-03-17 20:29:35 +00:00
Tim-Philipp Müller
65ede0b565 rtpulpfec: don't use non-portable notation for 64-bit int constants
Use GLib macro instead, even if it's a bit unwieldy.
2018-03-17 13:04:47 +00:00
Tim-Philipp Müller
c21765e88e rtpulpfecdec: don't use __builtin_ctzll unconditionally
Fixes build with MSVC, and possibly other compilers too.
2018-03-17 12:55:57 +00:00
Tim-Philipp Müller
b387989bc6 docs: rtpbin: add some Since markers for new properties 2018-03-12 13:21:08 +00:00
Olivier Crête
96261ce220 flvmux: Duration & unit tests
The muxed buffers will not carry the duration of the
incoming buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=793457
2018-03-01 18:25:02 -05:00
Olivier Crête
c0bf793c05 flvmux: Set PTS based on running time
https://bugzilla.gnome.org/show_bug.cgi?id=793457
2018-03-01 17:15:02 -05:00
Sebastian Dröge
f29fe76d7e rtspsrc: Ignore sendonly/recvonly attributes unless a backchannel is configured
This works around a bug in various ONVIF cameras that implement the
attributes the wrong way around. They still won't work with a
backchannel but at least normal playback will work for the time being.
It restores pre-1.14 behaviour where we would fail to preroll on any SDP
that lists a recvonly stream. For 1.16 a better solution should be
found.

The problem here is that the ONVIF spec has the meaning of the two
attributes the wrong way around in the examples, compared to RFC4566.

https://bugzilla.gnome.org/show_bug.cgi?id=793715
2018-03-01 20:27:10 +02:00
Mathieu Duponchelle
273d2c23d9 Port to latest GstAggregator segment API
The aggregator segment is now exposed on the src pad

https://bugzilla.gnome.org/show_bug.cgi?id=793945
2018-03-01 15:35:24 +01:00
Sebastian Dröge
80d5c43a81 matroskamux: Only mark new clusters as keyframe if they start on a keyframe or we're muxing only audio
Based on a patch by Nicola Murino <nicola.murino@gmail.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792775
2018-02-28 19:21:53 +02:00
Sebastian Dröge
15ae79838c matroskamux: Clip maximum cluster duration to the maximum possible value
Only up to timescale * G_MAXINT16 is possible as cluster duration, which
is already higher than our default value. Using higher values would
cause overflows and broken files.

Based on the investigation by Nicola Murino <nicola.murino@gmail.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792775
2018-02-28 19:20:23 +02:00
Sebastian Dröge
fc37bf70c5 matroska-mux: Refuse caps changes after starting to write headers
Matroska does not support changing the stream type and stream properties
after the headers were started to be written, and for example H264
codec_data changes can't be supported.

https://bugzilla.gnome.org/show_bug.cgi?id=782949
2018-02-28 10:38:29 +02:00
Tim-Philipp Müller
7b74816f07 rtp: fix another debug log printf format warning on 32-bit systems
rtpulpfeccommon.c:432:27: error: format ‘%lx’ expects argument of type
‘long unsigned int’, but argument 10 has type ‘guint64 {aka long long unsigned int}’

https://bugzilla.gnome.org/show_bug.cgi?id=793732
2018-02-27 13:13:49 +00:00
Mathieu Duponchelle
d2b51bd727 rtpptdemux: provide example usage for ignored-payload-types 2018-02-26 17:02:52 +01:00
Mathieu Duponchelle
55ecde7ee5 rtpbin, rtpptdemux: Add missing Since markers 2018-02-26 16:53:08 +01:00
Mathieu Duponchelle
3a754d51e0 FEC elements: document, remove irrelevant properties
The ulpfecenc "mux-seq" and "ssrc" properties were initially added
because the element did more than implement ULPFEC. As it was
decided that FLEXFEC would be implemented in a separate element,
both properties are now unneeded and confusing.

Change the default for the ulpfecenc multi-packet property,
as it is expected that most users of this element will be protecting video
streams.

Change the default property for the rtpredenc allow-no-red-blocks
property, as it should also be its default mode of operation.

https://bugzilla.gnome.org/show_bug.cgi?id=793843
2018-02-26 16:41:12 +01:00
Mathieu Duponchelle
efb4ee1919 rtpgstdepay: do not warn when caps were not yet received
It is expected that when connecting to a stream that has
already started, the caps will only arrive at the interval
specified on rtpgstpay, we shouldn't be warning as this is
a normal mode of operation.

https://bugzilla.gnome.org/show_bug.cgi?id=793798
2018-02-24 20:06:54 +01:00
Arnaud Bonatti
e3a3d4fb76 rtpulpfec: fix debug log printf format warning on 32-bit platforms
https://bugzilla.gnome.org/show_bug.cgi?id=793732
2018-02-23 10:02:23 +00:00
Tim-Philipp Müller
427033591c docs: hook up new RTP FEC elements
https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-22 15:56:49 +00:00
Tim-Philipp Müller
51177004d3 rtp: dist new header files
Fixes make distcheck
2018-02-21 20:45:31 +00:00
Tim-Philipp Müller
3789afd491 rtp: fec: fix build with gstreamer debug log system disabled 2018-02-21 19:01:15 +00:00
Mathieu Duponchelle
359b0a86f1 rtpptdemux: do no assume sink caps are non NULL 2018-02-21 19:59:04 +01:00
Mathieu Duponchelle
fdf64195ac rtpbin: Expose FEC support signals
Also slightly refactor complete_session_src

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Mikhail Fludkov
d5ad50bd61 rtp: Implement ULPFEC (RFC 5109)
We expose a set of new elements:

* ULPFEC encoder / decoder
* A storage element, which should be placed before jitterbuffers,
  and is used to store packets in order to attempt reconstruction
  after the jitterbuffer has sent PacketLost events
* RED encoder / decoder (RFC 2198), these are necessary to
  use FEC in webrtc, as browsers will propose and expect ulpfec
  packets to be wrapped in red packets

With contributions from:

Mathieu Duponchelle <mathieu@centricular.com>
Sebastian Dröge <sebastian@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Mathieu Duponchelle
82d0950254 rtpptdemux: Add ignored-payload-types property
Packets with these payload types will be dropped. A use case
for this is FEC, where we want FEC packets to go through the
jitterbuffer, but not be output by rtpbin.

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Mathieu Duponchelle
36b991f0b3 rtpptdemux: Add ssrc to output caps
It may be useful downstream

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Sebastian Dröge
a65a7ec05b monoscope: Forward the SEGMENT event from the chain function
Otherwise we'll break the event order and forward the SEGMENT event
before sending a CAPS event.
2018-02-18 21:38:13 +01:00
James Stevenson
d64ae7b01a rtspsrc: Fix missing read property of backchannel
Add missing read property code for backchannel

https://bugzilla.gnome.org/show_bug.cgi?id=793507
2018-02-16 12:53:07 +00:00
Sebastian Dröge
7512120e5d rtspsrc: Implement ONVIF backchannel support via TCP 2018-02-16 11:06:27 +02:00
Nirbheek Chauhan
befa41cdf6 rtspsrc: Implement ONVIF backchannel support
Set backchannel=onvif to enable, and use the 'push-backchannel-sample'
action signal with the correct stream id.
2018-02-16 11:06:27 +02:00
Jan Schmidt
d61066e6b5 splitmuxsrc: Improve not-linked handling.
Don't report not-linked unless all pads have
returned not-linked.
2018-02-16 17:02:31 +11:00
Patrick Radizi
364dbb5fc7 rtpjitterbuffer: allow timestamps to move backwards
The original solution for #784002 incorrectly assumed that timestamps
may not move backwards and changed timestamps that did so.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2018-02-15 10:05:39 +02:00
Tim-Philipp Müller
072c070e6c docs: remove pointless Since: 0.10.x markers 2018-02-15 00:58:38 +00:00
Alban Bedel
4e7ce28623 rtpvorbisdepay: fix unbounded memory usage
All received configurations are parsed and added to a list, this lead
to an unbounded memory usage. As the configuration is resent every
second this quickly lead to a large memory usage.

Add a check to only add the config if it is not already available in
the list. This fix only handle the typical case of a well behaved
stream, a malicious server could still send many useless
configurations to raise the client memory usage.
2018-02-14 18:04:56 +00:00
Edward Hervey
192b447e0b qtdemux: Fix comment typo in previous commit 2018-02-09 11:26:56 +01:00
Edward Hervey
e16be12e37 qtdemux: More 'meta' atom parsing fixes
Turns out everybody is doing it their own way, so peek into the
meta atom itself to figure out which spec it is following
2018-02-09 11:22:38 +01:00
Sebastian Dröge
e7177059e9 qtmux: Add support for muxing svmi atom for stereoscopic video information
https://bugzilla.gnome.org/show_bug.cgi?id=793120
2018-02-09 11:05:02 +02:00
Edward Hervey
4fd8635983 qtdemux: Minor cleanup
Just move variables to the blocks where they are used.

That function is massive, could do with some splitting up for
readability :(
2018-02-09 09:00:46 +01:00
Edward Hervey
357d77cf55 qtdemux: Cope with difference between QTFF and ISO BMFF specs
The 'meta' atom is defined differently in QTFF and BMFF, so try
to guess which spec the current stream applies to by looking
at the major file type.
2018-02-09 09:00:46 +01:00
Edward Hervey
b415358c7e isomp4: Make 'hdlr' atom dump more flexible
The smallest possible is 24 (and not 25) bytes.

The last "name" field can according to QTFF specifications not be present
at all. The parser will handle this fine and so will the rest of
the qtdemux code.
2018-02-09 09:00:46 +01:00
Edward Hervey
8239c67799 Update ORC files 2018-02-09 09:00:46 +01:00
Justin Kim
ad8a6cb639 qtmux: send stream warning when refusing video caps
If codec_data is changed, the stream is no longer valid.
Rather than keeping running when refusing new caps,
this patch send a warning  to the bus.

Also fix up splitmuxsink to ignore this warning while changing caps.

https://bugzilla.gnome.org/show_bug.cgi?id=790000
2018-02-01 10:12:46 +00:00
Justin Kim
dabeed52a9 rtph264depay: update output caps regardless format
`codec_data` should be transfered if any information of
SPS/PPS is changed.

https://bugzilla.gnome.org/show_bug.cgi?id=790000
2018-02-01 10:09:20 +00:00
Edward Hervey
0c56a80333 isomp4: Add gmhd/gmin debugging
* gmhd is a container, mark it as such so we can see/dump
  what is contained within
* Add dumping for the Base Media Information atom (gmin)
2018-01-31 19:12:48 +01:00
Philippe Normand
cbb9c31228 interleave: fix memory leak of GAP buffers
https://bugzilla.gnome.org/show_bug.cgi?id=793067
2018-01-31 15:16:46 +00:00
Edward Hervey
0511c4c225 qtdemux_dump: Demote verbose logging to TRACE level 2018-01-31 11:38:35 +01:00
Edward Hervey
40be9f4f85 qtdemux: Re-enable full debug logging of stsz entries
No idea why it was disabled (was the case since 2007)
2018-01-31 11:23:07 +01:00
Seungha Yang
75241cfa6b qtdemux: Remove white space at end of line
https://bugzilla.gnome.org/show_bug.cgi?id=791483
2018-01-29 18:42:47 +00:00
Seungha Yang
1551759cec qtdemux: Apply qtdemux debug category to gstisoff
.. instead of the use of default debug category.
And, make new header to declare the debug category

https://bugzilla.gnome.org/show_bug.cgi?id=791483
2018-01-29 18:40:02 +00:00
Tim-Philipp Müller
ddd4fe680f splitmuxsrc: properly set total duration on outgoing segment
We would accidentally pass through the duration value from the
demuxer from a single fragment, which causes problems when
feeding the stream from splitmuxsrc to rtsp-server. Streaming
would stop after one fragment due to that.

https://bugzilla.gnome.org/show_bug.cgi?id=792861
2018-01-29 11:29:24 +00:00
Tim-Philipp Müller
07ae64118d splitmuxsrc: don't respond to duration query with CLOCK_TIME_NONE
total_duration is initialised to CLOCK_TIME_NONE, not 0, so check
for that as well in order not to return an invalid duration to
a duration query. Doesn't fix anything particular observed in
practice, just seemed inconsistent.
2018-01-29 11:29:24 +00:00
Alicia Boya García
ac46d5c2a3 qtdemux: Add more prose to the comment of gst_qtdemux_find_sample()
https://bugzilla.gnome.org/show_bug.cgi?id=792910
2018-01-25 20:40:46 +00:00
Tim-Philipp Müller
a5b5451d93 multifilesrc: fix up uri handler a little
Fix path escaping when creating URI from location in get_uri().

Return FALSE with an error when URI can't be parsed in set_uri().

https://bugzilla.gnome.org/show_bug.cgi?id=783581
2018-01-23 18:38:26 +00:00
Dimitrios Katsaros
dccfaf2e91 multifilesrc: implement uri handler
With this patch we can now provide a set of files
created by multifilesink as a source for uri elements.

e.g. gst-launch-1.0 playbin uri=multifile://img%25d.ppm

Note that for the %d pattern you need to replace % with %25.
This is to be compliant with URL naming standards.

https://bugzilla.gnome.org/show_bug.cgi?id=783581
2018-01-23 18:05:49 +00:00
Vivia Nikolaidou
379059b1c7 qtmux: Make sure timecode uses the same timescale as video
Don't blindly derive it from the frame rate, but try to get the per-pad
configured timescale first (if it exists)

https://bugzilla.gnome.org/show_bug.cgi?id=792680
2018-01-23 16:40:36 +02:00
Sebastian Dröge
c71cd08d0e qtmux: Allow configuring trak timescale per pad/trak
It generally makes not much sense to configure it for all pads/traks at
once as this value is usually different for each of them. As such, add a
new property on the pads in addition to the existing property on the
whole muxer.

https://bugzilla.gnome.org/show_bug.cgi?id=792649
2018-01-23 16:40:23 +02:00
Tim-Philipp Müller
f459438278 Update for renamed aggregator pad API
https://bugzilla.gnome.org/show_bug.cgi?id=791204
2018-01-23 09:46:32 +00:00
Sebastian Dröge
af273b4de9 rtspsrc: Fix up sendonly/recvonly attribute handling
We can't handle recvonly streams, sendonly streams are perfectly fine.

The direction is the one from the point of view of the SDP offerer
(i.e. the RTSP server), and a recvonly stream would be one where the
server expects us to send media.

RFC 3264, section 5.1:
   If the offerer wishes to only send media on a stream to its peer, it
   MUST mark the stream as sendonly with the "a=sendonly" attribute.

This is mixed up in the ONVIF streaming specification examples, but
actual implementations and conformance tools seem to not care at all
about the attributes.

https://bugzilla.gnome.org/show_bug.cgi?id=792376
2018-01-22 12:24:18 +02:00
Jan Alexander Steffens (heftig)
e273e5f7a6 aacparse: When parsing raw input, accept frames of any size
Raw AAC streams might have very small frames, e.g. 6 byte frames
when encoding silence. These frames are then smaller than aacparse's
default min_frame_size of 10 bytes (ADTS_MAX_SIZE).

When passthrough is disabled or aacparse has to output ADTS, GstBaseParse
will concatenate these short frames to the following frame before
handling them to aacparse, which processes each input buffer as a single
frame, producing bad output.

To avoid this problem, set the min_frame_size to 1 when receiving a raw
stream.

https://bugzilla.gnome.org/show_bug.cgi?id=792644
2018-01-18 19:09:19 +00:00
Mathieu Duponchelle
03dc22951b rtpbin: fix leak of elements requested by signals
When the signal returns a floating reference, as its return type
is transfer full, we need to sink it ourselves before passing
it to gst_bin_add (which is transfer floating).

This allows us to unref it in bin_remove_element later on, and
thus to also release the reference we now own if the signal
returns a non-floating reference as well.

As we now still hold a reference to the element when removing it,
we also need to lock its state and setting it to NULL before
unreffing it

Also update the request_aux_sender test.

https://bugzilla.gnome.org/show_bug.cgi?id=792543
2018-01-18 15:26:43 +01:00
Tim-Philipp Müller
dfb27b44bc multifilesink: document unit of "max-file-duration" property 2018-01-16 13:19:29 +00:00
Florent Thiéry
7c5e39f7a7 udpsrc: fix typo in documentation
https://bugzilla.gnome.org/show_bug.cgi?id=792458
2018-01-12 18:21:06 +00:00
Tim Allen
db07fc7e85 rtp: add L8 audio support 2017-12-24 13:19:56 +01:00
Mark Nauwelaerts
f3c3f8fb38 udpsrc: fix typo in multicast join error message 2017-12-23 12:48:20 +01:00
Mark Nauwelaerts
aad0faaf59 rtspsrc: also proxy multicast-iface property to RTCP udpsrc 2017-12-23 12:48:20 +01:00
Sebastian Rasmussen
5091b5f39f multiudpsink: don't try to set IPV6_TCLASS on IPV4 sockets
Avoids ERROR log message.

https://bugzilla.gnome.org/show_bug.cgi?id=757449
2017-12-23 12:45:17 +01:00
Vincent Penquerc'h
8b814f6351 flv: flvmux ported to the GstAggregator
This makes it possible to create a flv file from a live source and not stop
when there are packet drops.

https://bugzilla.gnome.org/show_bug.cgi?id=782920
2017-12-20 15:34:11 -05:00
Olivier Crête
6ea569841d multiudpsink: Call gst_base_sink_wait_preroll on unlock
This means that packets will not be lost on fast pause/playing cycles.

Also refactor the code a little to simplify it.

https://bugzilla.gnome.org/show_bug.cgi?id=774945
2017-12-19 16:47:52 -05:00
Olivier Crête
0a15d1782b multiudpsink: Remove unused variable 2017-12-19 15:50:14 -05:00
Edward Hervey
9a7dd45e57 rtspsrc: Fix two leaks
* gst_event_new_stream_start() does not take ownership of the stream_id

* the pipeline_request_id string that is created was not being freed
2017-12-19 11:57:52 +01:00
Nicolas Dufresne
3a17f52ca3 videocrop: Add GstVideoCropMeta support
If downstream supports this meta, it will add or update it from
the GstBuffer in-place rather then copying.

https://bugzilla.gnome.org/show_bug.cgi?id=791453
2017-12-17 15:39:53 -05:00
Sean DuBois
de8f080a5c Add AV1 to matroska plugin
https://bugzilla.gnome.org/show_bug.cgi?id=784160
2017-12-15 18:00:31 +01:00
fengalin
3464aac3c9 matroska: fix memory leaks due to toc related updates
https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
fengalin
a6702a76d5 matroska: re-activate and update TOC support
TOC support in mastroskamux has been deactivated for a couple of years. This commit updates it to recent GstToc evolutions and introduces toc unit tests for both matroska-mux and matroska-demux.

There are two UIDs for Chapters in Matroska's specifications:
- The ChapterUID is a mandatory unsigned integer which internally refers to a given chapter. Except for title & language which use dedicated fields, this UID can also be used to add tags to the Chapter. The tags come in a separate section of the container.
- The ChapterStringUID is an optional UTF-8 string which also uniquely refers to a chapter but from an external perspective. It can act as a "WebVTT cue identifier" which "can be used to reference a specific cue, for example from script or CSS".

During muxing, the ChapterUID is generated and checked for unicity, while the ChapterStringUID receives the user defined UID. In order to be able to refer to chapters from the tags section, we maintain an internal Toc tree with the generated ChapterUID.

When demuxing, the ChapterStringUIDs (if available) are assigned to the GstTocEntries UIDs and an internal toc mimicking the toc is used to keep track of the ChapterUIDs and match the tags with the appropriate GstTocEntries.

https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
Edward Hervey
6e21faffe7 qtdemux: Push a GAP event if there's a second *or more*
And not "more than a second"
2017-12-13 12:06:49 +01:00
Edward Hervey
2e45926a96 qtdemux: Don't push GAP event if first buffer is within 1s
If we saw empty segments, we previously unconditionally pushed a
GAP event downstream regardless of the duration of that empty
segment.

In order to avoid issues with initial negotiation of downstream elements
(which would negotiate to something before receiving any data due to
that initial GAP event), check if there's at least a second of difference
(like we do for other GAP-related checks in qtdemux) before
deciding to push a GAP event downstream.
2017-12-13 11:35:37 +01:00
Edward Hervey
5c341f0980 qtdemux: Don't set pared=True on underspecified audio/mpeg
This *really* needs to go through a parser to figure out what the
exact content type is.
2017-12-13 10:21:17 +01:00
Michael Catanzaro
d9235cdb49 equalizer: Fix -Wincompatible-pointer-types warning
This is caused by the new type propagation for g_object_ref.

https://bugzilla.gnome.org/show_bug.cgi?id=791494
2017-12-11 21:57:25 +00:00
John Nikolaides
91dfc20a04 splitmuxsink: added a "split now" action signal
Now, the video file can be split at an arbitrary time chosen by the user.

https://bugzilla.gnome.org/show_bug.cgi?id=787922
2017-12-08 21:16:15 +02:00
Alvaro Margulis
66f253ed9b multiudpsink: fix bind address leak
https://bugzilla.gnome.org/show_bug.cgi?id=790986
2017-12-08 00:31:32 +00:00
Tim-Philipp Müller
d4c04cb079 Revert "flacparse: fix header rewriting being ignored"
This caused broken metadata and also looks a bit dodgy.
Revert until we can figure out a solution that works for
all cases and doesn't break anything.

This reverts commit adeee44b07.

https://bugzilla.gnome.org/show_bug.cgi?id=727802
https://bugzilla.gnome.org/show_bug.cgi?id=785558
2017-12-07 11:17:32 +00:00
Matt Staples
ea1b10e4ca rtspsrc: Add a signal to allow outgoing messages to be modified or dropped
This feature allows applications to implement extensions to the RTSP
protocol, such as those defined in the ONVIF Streaming Specification.

https://bugzilla.gnome.org/show_bug.cgi?id=762884
2017-12-06 10:46:01 +02:00
Haakon Sporsheim
3c0d006c03 rtpsession: Handle zero length feedback packets
https://bugzilla.gnome.org/show_bug.cgi?id=791074
2017-12-02 13:58:34 +00:00
Florian Zwoch
b4ca81591c qtdemux: fix debug log for 'hvcC' codec_data
https://bugzilla.gnome.org/show_bug.cgi?id=784749
2017-12-02 13:46:29 +00:00
Sebastian Dröge
b02350bd62 flacparse: Request at least the full header size when parsing headers
Otherwise baseparse will incrementally send us bigger buffers until the
full header size is reached, which is not only pointless but also means
that baseparse will reallocate and copy into a bigger buffer for every
input buffers. In pull mode that's done in 64kb increments, in push mode
usually in much smaller increments, causing a lot of overhead for
example when parsing high-quality coverart.
2017-11-29 17:28:25 +02:00
Edward Hervey
10bc8fdfd2 rtspsrc: Do more checks for seekability
When receiving a seek event, check whether we can actually seek based
on the information the server provided.

Also add more documentation on what the seekable field means
2017-11-24 15:39:38 +01:00
Jan Schmidt
a260eb80fb qtmux: Always update reserved-duration-remaining
If a reserved-max-duration is set, we should always track
and update the reserved-duration-remaining estimate, even
if we're not sending periodic moov updates downstream for
full robust muxing.
2017-11-25 00:56:11 +11:00
Jan Schmidt
76e458a119 splitmuxsink: Use muxer reserved space properties if present.
If the use-robust-muxing property is set, check if the
assigned muxer has reserved-max-duration and
reserved-duration-remaining properties, and if so set
the configured maximum duration to the reserved-max-duration
property, and monitor the remaining space to start
a new file if the reserved header space is about to run out -
even though it never ought to.
2017-11-25 00:56:11 +11:00
Jan Schmidt
3a813a0dcc splitmux: Fix file switch-on-caps-change.
Switching to a new fragment because the input caps have
changed didn't properly end the previous file. Use the normal
EOS sequence to ensure that happens. Add a test that it works.
2017-11-24 16:56:03 +11:00
Tim-Philipp Müller
c8ff205089 rtph265depay: don't insert SPS/PPS inline for hvc1 output
Only for byte-stream or hev1. For hvc1 the SPS/PPS are in the
caps as codec_data field and in this case they shouldn't be in
the stream data as well. The output caps should be updated with
the new codec_data if needed, for hvc1.
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
8da79ca824 rtph265depay: store negotiated output format as enum
We keep the boolean byte_stream around since it's nicer for
readability and most of the code just cares about byte_stream
or not. This is useful for future-proofing the code for when
we add support for hev1 output as well.
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
46861027b9 rtph265depay: add support for hvc1 as output format 2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
b84201bf81 rtph265pay: don't add trailing zeros to VPS/PPS/SPS
This would happen if input is byte-stream with four-byte
sync markers instead of three-byte ones. The code that
scans for sync markers will place the start of the NALU
on the third-last byte of the NALU sync marker, which
means that any additional zeros may be counted as belonging
to the previous NALU instead of being part of the next sync
marker. Fix that so we don't send VPS/SPS/PPS with trailing
zeros in this case.

See https://bugzilla.gnome.org/show_bug.cgi?id=732758
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
311b9895ba rtph265depay: assemble AUs into downstream-allocated memory
When merging NALs into AUs, use downstream-provided allocator
to allocate memory and copy NALs directly into that memory when
assembling them.
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
2d0ea4d381 rtph265depay: try to negotiate an allocator with downstream 2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
528b7e01f1 rtph265depay: simplify buffer accumulation control flow
There is no difference between pushing out a buffer directly
with gst_rtp_base_depayload_push() and returning it from the
process function. The base class will just call _depayload_push()
on the returned buffer as well.

So instead of marshalling buffers through three layers and back,
just push them from one place in handle_nal() and always return
NULL from the process vfunc. This simplifies the code a little.

Also rename _push_fragmentation_unit() to _finish_fragmentation_unit()
for clarity. Push sounds like it means being pushed out, whereas
it might just be pushed into an adapter.

This change has the side-effect that multiple NALs in a single STAP
(such as SPS/PPS) may no longer be pushed out as a single buffer if
we output NALs in byte-stream format (i.e. not aggregate AUs), but
that shouldn't really make any difference to anyone.
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
289882497a rtph265depay: fix crash with empty sprops-parameters
https://bugzilla.gnome.org/show_bug.cgi?id=780040
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
0580efc6a6 rtph265depay: minor clean-up
Declutter caps update code a bit.
2017-11-23 09:36:15 +01:00
Philip Craig
ec11b228a4 rtph264pay: don't add trailing zeros to PPS/SPS
This would happen if input is byte-stream with four-byte
sync markers instead of three-byte ones. The code that
scans for sync markers will place the start of the NALU
on the third-last byte of the NALU sync marker, which
means that any additional zeros may be counted as belonging
to the previous NALU instead of being part of the next sync
marker. Fix that so we don't send SPS/PPS with trailing
zeros in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=732758
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
f3e4df72a1 rtph264depay: assemble AUs into downstream-allocated memory
When merging NALs into AUs, use downstream-provided allocator
to allocate memory and copy NALs directly into that memory when
assembling them.
2017-11-23 09:35:59 +01:00
Tim-Philipp Müller
b6f13ce4e9 rtph264depay: try to negotiate an allocator with downstream 2017-11-23 09:35:59 +01:00
Tim-Philipp Müller
44f70445b6 rtph264depay: minor clean-up
Declutter caps update code a bit.
2017-11-23 09:27:22 +01:00
Edward Hervey
22ab222dbc qtdemux: Run gst-indent 2017-11-23 08:00:58 +01:00
Edward Hervey
4e90993423 rganalysis: Fix left shift of signed values
left shifting signed values is undefined.

Instead of doing "x << offs" which is undefined, do the equivalent
"x * (1 << offs)" which is well defined
2017-11-23 07:59:07 +01:00
Edward Hervey
d08e155ec8 qtdemux: Check presence of bitrate tags
Check whether the tag was present before printing it out

CID #1418501
2017-11-23 07:57:44 +01:00
Edward Hervey
92c3207463 rtspsrc: Use the proper maximum value for seekable
it's a gfloat, not a gdouble
2017-11-21 09:33:49 +01:00
Jan Schmidt
3ba556a54d qtdemux: Use new GST_SEQNUM_INVALID constant 2017-11-18 02:33:01 +11:00
Jan Schmidt
97e8fd8272 splitmuxsrc: Don't return FALSE from event handling.
Returning FALSE because we drop an event means that
internal sources like qtdemux might throw an error
and break the whole pipeline. The only time it can
happen is either flushing or shutdown, and those
will be handled anyway.
2017-11-18 02:33:00 +11:00
Jun Xie
7c8aeff262 qtdemux: reset reused QtDemuxStream while parsing a new 'trak'
if QtDemuxStream is reused, then we need to reset it.

https://bugzilla.gnome.org/show_bug.cgi?id=788759
2017-11-17 11:49:24 +02:00
Seungha Yang
8aa5c59b5f isomp4: Add official fourcc for VP8 codec
fourcc for VP8 codec is "vp08" defined by spec. To follow it,
add it to demux and change legacy VP8 fourcc "VP80" to "vp08" in mux.
Also, enable sync table in case of VP8 codec.
See also https://www.webmproject.org/vp9/mp4/

https://bugzilla.gnome.org/show_bug.cgi?id=790026
2017-11-13 10:08:14 -05:00
Seungha Yang
93f19b19d2 isomp4: Add support VP9 codec
Add fourcc for VP9 codec and support it by qtdemux and qtmux
See also https://www.webmproject.org/vp9/mp4/

https://bugzilla.gnome.org/show_bug.cgi?id=790026
2017-11-13 10:08:14 -05:00
Edward Hervey
f89d672eb7 matroskademux: Remove bogus error message
It's just informational
2017-11-13 13:51:20 +01:00
Edward Hervey
50c3733a89 rtpmpvpay: Don't create empty buffer list
If there's nothing to send, just return
2017-11-10 15:51:05 +01:00
Seungha Yang
5dd39d8946 matroskademux: Add parsing Colour element
... and forward colorimetry to downstream. The Colour element describes
various color information (similar to 'colr' box in isobmff).
Note that, due to the comparatively limited syntax for color information
in vpx codecs, the color information in mkv/wemb container level
should be used for sophisticated color handling (e.g., HDR video).

https://bugzilla.gnome.org/show_bug.cgi?id=790023
2017-11-08 11:01:55 +02:00
Youness Alaoui
593615de46 rtpg722pay: Add encoding-params to the src caps template
The G722 payload only accepts G722 audio with channels=1, so it must
specify the encoding-params=1 in its src caps, otherwise it causes issues
with farstream which thinks it supports 2 channels G722 and when
confronted with a remote that has G722/8000/2, it will negotiate it
and error out with a not-negotiated when the caps don't intersect
at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=789878
2017-11-03 17:20:31 -04:00
Sebastian Dröge
af3ce874a0 rtspsrc: Add "accept-certificate" signal for manually checking a TLS certificate for validity
https://bugzilla.gnome.org/show_bug.cgi?id=785024
2017-11-01 13:42:45 +02:00
Sangkyu Park
257f7c9063 rtspsrc: Print RTSP/SDP messages to gstreamer log instead of stdout
- 'debug' property is deprecated
- All RTSP messages are printed to gstreamer log with 'log' level.

https://bugzilla.gnome.org/show_bug.cgi?id=788917
2017-11-01 11:35:44 +02:00
Justin Kim
2a5aafe425 rtpsesson: downgrade message level to debug when detected XR
When XR packet is detected, warning message leads to misunderstandings.
Until RFC3611 is implemented in gst-plugins-base, the level needs to
be downgraded to avoid confusion.

https://bugzilla.gnome.org/show_bug.cgi?id=789746
2017-11-01 10:57:00 +02:00
Ashish Kumar
1aed3948a8 gst-plugins-good: atoms_recovery: Handled buffer mapping failure
https://bugzilla.gnome.org/show_bug.cgi?id=789413
2017-10-24 22:04:27 -07:00
Thiago Santos
923b83a48c atomsrecovery: read from mdat only what is on headers
It is possible that the mdat has more data than what was stored in the
headers file. If we put that to the output the file will have bogus data
at the end and some players will complain.

https://bugzilla.gnome.org/show_bug.cgi?id=784258
2017-10-24 21:33:25 -07:00
Thiago Santos
69605b6c61 isomp4: atomsrecovery: handle common and large atom headers
Do not assume all files are large files. Check and use the short or
extended atom size field only if needed.

https://bugzilla.gnome.org/show_bug.cgi?id=784258
2017-10-24 21:33:25 -07:00
Thibault Saunier
6d72ae13fe rtsp: Add missing Since marker 2017-10-16 14:21:56 +02:00
Sebastian Dröge
728a1629cf qtdemux: Fix compiler warning
qtdemux.c: In function ‘gst_qtdemux_configure_stream’:
qtdemux.c:7764:34: error: suggest parentheses around ‘&&’ within ‘||’ [-Werror=parentheses]
     if ((stream->n_samples == 1) && (stream->first_duration == 0)
         ~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2017-10-10 18:04:50 +03:00
Nael Ouedraogo
5874bafa4f qtdemux: fix assert when moof containing one sample
Avoid computing frame rate when a stream contain moof with only one
sample, to avoid an assert. The moof is considered as still picture.

The same is already done for one sample given in the moov.

https://bugzilla.gnome.org/show_bug.cgi?id=782217
2017-10-10 18:03:42 +03:00
Thibault Saunier
fb952e3470 rtspsrc: Avoid potentially dereferencing NULL pointer
CID 1418986
2017-10-09 14:39:14 +02:00
Tim-Philipp Müller
6cb51bd8cf rtpjitterbuffer: fix debug message on pt mismatch 2017-10-08 00:07:43 +01:00
Tim-Philipp Müller
d5f72418c8 rtpbin, rtspsrc: fix compiler warnings about 64-bit integer signednes
"warning: this decimal constant is unsigned only in ISO C90" with
gcc 4.8.4 (Ubuntu/Linaro 4.8.4-2ubuntu1~14.04.3)
2017-10-07 15:55:24 +01:00
Sebastian Dröge
263494f9c7 rtpsbcdepay: Fix potential NULL pointer dereference
CID 1418864
2017-10-07 14:06:38 +03:00
Sebastian Dröge
7d80fa4bbf audioecho: Micro-optimize
Gives 1.28x speedup in surround-delay=false mode
2017-10-07 03:03:46 +03:00
Sebastian Dröge
ad062e1879 audioecho: Don't do linear interpolation between samples
Linear interpolation adds quite some noise, and it's unlikely that
anybody will ever need sub-sample accurate delays. Proper resampling
before that will lead to better results.
2017-10-07 03:03:46 +03:00
Nicolas Dufresne
986f3e15b2 flvdemux: Only set pixel-aspect-ratio if specified
If it's not specified, we should let the decoder figure it out.
Apparently the code was already in place, all was to make the code
conditional.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Nicolas Dufresne
307018da89 flvdemux: Don't pull passed the EOS
When a truncated FLV is provided and processed in pull mode, we
may endup trying to pull passed EOS, causing a rather confusing
warning as the pull offset is an integer overflow.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Nicolas Dufresne
cfc1638ca5 flvdemux: Ignore invalid H.264 codec data
This code basically skip over codec_data with empty payload. In
this case, the codec_data variable is the size of the header for
the CODEC part of Video Tag. The remaining is supposed to be the
H.264 codec data, hence should not be empty.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Nicolas Dufresne
18dbd49fb8 flvdemux: Avoid integer overflow on invalid CTS
If the CTS is negative an would lead to a negtive PTS, clip
the CTS so the PTS will be 0.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Thibault Saunier
125e835045 rtspsrc: Fix build 2017-10-05 14:37:54 -03:00
Thibault Saunier
ffcd173c09 rtspsrc: Handle TCP as lower transport with RTSP 2.0
Meaning that the interleave fields have to be updated as
if streams setup was working when using pipelined setup
request. Otherwise there is a mismatch between the server
channel count and our own.

This also makes RTSP 2.0 over HTTP working.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:40 -03:00
Thibault Saunier
8121752887 rtsp: Start implementing support for RTSP 2.0
- Handle version negotation:

  Added a `default-version` property so that the user can configure
  what to use in case the server does not support version negotation
  (which actually exist)

- Handle pipelined requests, which allow avoiding full round trip to
  setup the RTP streams (request are sent in a raw, and response are
  handled as they arrive).

- Handle the new Media-Properties header

- Handle the new Seek-Style header

- Handle the new Accept-Ranges header

Handling of IPV6 should already be OK.

We are still missing (at least) the following features (which do not
seem really mandatory as they require a "persistent connection between
server and client"):

  - Server to Client TEARDOWN command (Not so usefull fmpov)
  - PLAY_NOTIFY (not needed for our server yet)
  - Support for the new REDIRECT features

and probably some more protocol changes might not be handled yet.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:40 -03:00
Thibault Saunier
5faad79049 rtspsrc: Use a macro to debug RTSP messages
Simplifying the code a little.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:39 -03:00
Reynaldo H. Verdejo Pinochet
0ddb65dec7 Use proper GtkDoc notation for NULL/FALSE/TRUE 2017-10-03 16:30:10 -07:00
Sebastian Dröge
26cf8c1dc9 rtspsrc: Ignore medias marked as sendonly
We're never going to receive anything from them, so don't create pads
for them. These medias are destinations where *we* could send something.
2017-10-01 16:09:13 +02:00
Sebastian Dröge
58f0eabd61 sbcdepay: Add property to ignore input timestamps
This then just counts samples and calculates the output timestamps based
on that and the very first observed timestamp. The timestamps on the
buffers are continued to be used to detect discontinuities that are too
big and reset the counter at that point.

When receiving data via Bluetooth, many devices put completely wrong
values into the RTP timestamp field. For example iOS seems to put a
timestamp in milliseconds in there, instead of something based on the
current sample offset (RTP clock-rate == sample rate).

https://bugzilla.gnome.org/show_bug.cgi?id=787297
2017-09-28 14:15:12 +03:00
Ponnam Srinivas
c0622addf6 rtph265depay: Fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=787937
2017-09-26 11:09:53 +03:00
Deepak Srivastava
49ed67c4f2 deinterlace: Fixed memory leak in error code path
https://bugzilla.gnome.org/show_bug.cgi?id=788041
2017-09-22 17:25:50 +03:00
Tim-Philipp Müller
a802f5df42 rtpjitterbuffer: implement basic chain_list function
Doesn't do anything fancy yet, but still avoids lots of
unnecessary locking/unlocking that would happen if the
default chain_list fallback function in GstPad got invoked.
2017-09-17 16:33:15 +01:00
Tim-Philipp Müller
07dd3dba6f multifilesink: use new gst_buffer_list_calculate_size() 2017-09-17 12:50:30 +01:00
Patrick Radizi
3de0244532 rtpbin: add option for sanity checking timestamp offset
Timestamp offsets needs to be checked to detect unrealistic values
caused for example by NTP clocks not in sync. The new parameter
max-ts-offset lets the user decide an upper offset limit. There
are two different cases for checking the offset based on if
ntp-sync is used or not:
1) ntp-sync enabled
   Only negative offsest are allowed since a positive offset would
   mean that the sender and receiver clocks are not in sync.
   Default vaule of max-ts-offset = 0 (disabled)
2) ntp-sync disabled
   Both positive and negative offsets are allowed.
   Default vaule of max-ts-offset = 3000000000
The reason for different default values is to be backwards
compatible.

https://bugzilla.gnome.org/show_bug.cgi?id=785733
2017-09-15 13:33:14 +03:00
Patrick Radizi
23f7739ba4 rtpbin: add option for increasing ts_offset gradually
Instant large changes to ts_offset may cause timestamps to move
backwards and also cause visible effects in media playback. The new
option max-ts-offset-adjustment lets the application control the rate to
apply changes to ts_offset.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2017-09-14 13:15:56 +03:00
Mark Nauwelaerts
cf3f195012 matroskademux: search_cluster should find preceding cluster before target
... since failing this constraint takes search_pos by surprise which might
then end up in an infinite loop.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=787538
2017-09-11 20:39:22 +02:00
George Kiagiadakis
286e1e62be rtprtx{send,receive}: improve the debug messages
* use INFO/DEBUG/LOG/TRACE equaly and meaningfully;
  previously rtprtxsend:LOG and rtprtxreceive:LOG would generate
  a totally different amount of log traffic and sometimes it was
  impossible to see the information you wanted without useless
  spam being printed around
* improve the wording, give a reasonable and self-explanatory
  amount of information
* print SSRCs in hex
* avoid G_FOO_FORMAT for readability (we are just printing integers)
2017-09-07 14:43:32 +03:00
Jan Schmidt
d191584531 qtmux: Allow MPEG layer 1/2, AC3 and Opus in qtmux
qtmux is supposed to be the muxer that allows all formats,
with others (mp4mux and friends) being profile-restricted.
2017-09-06 23:55:38 +10:00
Tim-Philipp Müller
6f9cb1716a rtph265depay: fix keyunit detection
https://bugzilla.gnome.org/show_bug.cgi?id=787254
2017-09-05 13:56:18 +01:00
Edward Hervey
5bc5e07531 dcaparse: Really fix "usage before unmap"
Previous patch would try to unref a buffer that was pushed downstream.

Instead only unref when/if needed and keep usage of the cleanup: goto
block
2017-09-04 17:06:39 +02:00
Arun Raghavan
5cf368b36c dcaparse: Don't unmap buffer before accessing data from it
The previous patch added a check for a substream header after
gst_buffer_unmap(), which is incorrect.
2017-09-03 15:23:10 +05:30
Matej Knopp
f7854411fa dcaparse: preserve DTS HD substream 2017-09-03 15:05:08 +05:30
Arun Raghavan
301e8d558e rtpsbcpay: Fix some tabs that crept in somehow 2017-08-29 22:12:35 +05:30