This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.
https://bugzilla.gnome.org/show_bug.cgi?id=703111
The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.
The code can be used as follows
```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink
gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```
rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate
GStreamer 1.16 does not yet support the newer GLib templates, so revert.
rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources
for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.
rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches
beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.
rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even
According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.
rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters
Locking is added because the URI allows to access the properties too.
rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction
In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
The start of each segment is relative to the Period start, minus
the presentation time offset.
As specified in section 5.3.9.6 of the MPEG DASH specification:
The value of the @t attribute minus the value of the
@presentationTimeOffset specifies the MPD start time of
the first Segment in the series.
Several tests use a Period@start value of 10 seconds, which either
needs to be taken into account when calculating expected timestamps
or have that attribute removed.
This commit uses a mix of updating the timestamps and removing the
start attribute, so that both the case of its presence and absence
is tested.
The start of each segment is relative to the Period start, minus
the presentation time offset.
As specified in section 5.3.9.6 of the MPEG DASH specification:
The value of the @t attribute minus the value of the
@presentationTimeOffset specifies the MPD start time of
the first Segment in the series.
dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.
Fixes#841
If we renegotiate, then it is currently possible for an added stream to
be added to webrtcbin before the SDP is complete. This causes an
internal inconsistency as there is a 'pending sink transceiver' without
a corresponding media section in the sdp. It also does not have an
associated transport stream and will fail in _connect_input_stream().
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice. Use an atomic add instead.
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
We add the signal watch in testSeekPreTestCallback so
remove it in testSeekPostTestCallback and not deep inside
some if clause in some other callback somewhere.
Based upon the souphttpsrc tests, add unit tests for the curlhttpsrc
element. The souphttpsrc tests are able to use an HTTP server that
is provided as part of the soup library. This does not exist in the
curl library, therefore these tests provide a very simple HTTP server
using the GIO library.
These curlhttpsrc tests contain one new test that does not come from
the souphttpsrc tests. The test_multiple_http_requests test tries to
reproduce the way in which GstAdaptiveDemux makes use of URI source
elements. GstAdaptiveDemux creates a bin with the httpsrc element
and a queue element and sets the locked state of that bin to TRUE,
so that it does not follow the state transitions of its parent. It
then moves this bin to the PLAYING state to start each download and
back to READY when the download completes.
It depends on the framerate how many cc_data byte pairs are allowed per
frame, and the framerate is also needed for converting into the CDP or
MCC format as the framerate is part of the header metadata.
With prenegotiated channels, the data-channel protocol is not used and
instead the channel's negotiation is intended to be performed out of band in
some application-specific manner.
Comes with test!
This means that we will reject all operations before we've transitioned
into READY.
This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread. Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
It works like a valve in front of the actual avwait. When recording ==
TRUE, other rules are then examined. When recording == FALSE, nothing is
passing through.
https://bugzilla.gnome.org/show_bug.cgi?id=796836
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
Try prioritizing downstream's caps over upstream's if possible so the
parser can configured in "passthrough" if possible and save it from
doing useless conversions.
https://bugzilla.gnome.org/show_bug.cgi?id=790628
Sending an event can accepted event if the caps were rejected
because the event could be queued and processed later.
Also send a drain query in the caps test to make sure that the
event has been processed.
https://bugzilla.gnome.org/show_bug.cgi?id=781673
Since insertion of aud landed, we need to change some testcases
accroding to the change.
Note that counting frames are changed in parser.c,
due to generated frames, AUD.
https://bugzilla.gnome.org/show_bug.cgi?id=736213
For duration queries on live streams, adaptivedemux ignores the query.
The problem then is that the query is answered by the downstream
qtdemux element, with the duration of the currently passing fragment.
This commit changes the behaviour of adaptivedemux to answer the duration
queries for live streams, returning GST_CLOCK_TIME_NONE.
https://bugzilla.gnome.org/show_bug.cgi?id=753879
See https://bugzilla.gnome.org/show_bug.cgi?id=773666
This would ideally be solved in baseparse but that requires further
thought at this point, and in the meantime it would be good to have
rawbaseparse not assert on this but handle it gracefully instead.
Make the unit tests handle the fact that pads don't appear
immediately. Before, the test assumed pads are exposed before the
internal source element is created, which is no longer true.
To satisfy follwing restriction of HLS spec 6.3.3,
select startup fragment sequence to 4th from end of playlist.
Also, seek range should exclude last three fragment in playlist.
"the client SHOULD NOT choose a segment which starts less than
three target durations from the end of the Playlist file."
https://bugzilla.gnome.org/show_bug.cgi?id=777682
If they were not ported after 4+ years it seems unlikely that anybody is
ever going to need them again. They're still in the GIT history if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=774530
Section 5.3.3 in ISO/IEC 23009-1:2014 defines that invalid references
(e.g., invalide URI or cannot be resolved) specified by "@xlink:href" attribute
shall be removed. That means, we should play it without error,
and just ignore the corresponding element.
It's similar to "urn:mpeg:dash:resolve-to-zero:2013".
https://bugzilla.gnome.org/show_bug.cgi?id=774463
External xml could have empty, one or multiple top-level "Period" elements.
Because xml parser cannot parse the multiple top-level elements
(i.e., no root element), we need to wrap a xml in order to make root element.
See also ISO/IEC 23009-1:2014 5.3.2.2
https://bugzilla.gnome.org/show_bug.cgi?id=774357
PlayReady being the one of the few DRM formats encoding its data with
base64 it was not consistent to have a special case for this. So the
base64 decoding operation now needs to be done by the protection event
consumer, if needed.
https://bugzilla.gnome.org/show_bug.cgi?id=774112
Add a test of the gst_mpd_client_get_maximum_segment_duration() function
to check that it first checks the MPD@maxSegmentDuration and then falls
back to checking all of the segment durations.
https://bugzilla.gnome.org/show_bug.cgi?id=753751
When the test involves doing a seek, only check for data size after
the seek. The final segment range after seek might be different/smaller
than the threshold for doing the seek and doing the check before
seeking would fail.
Following the Don't Repeat Yourself principle, define macros
for the structures that contain the request and response headers,
so that the name is not repeated in multiple places in multiple files.
https://bugzilla.gnome.org/show_bug.cgi?id=762144
Test content protection
Configure 3 content protection sources:
- a uuid scheme/value pair
- a non uuid scheme/value pair (dash recognises only uuid schemes)
- a complex uuid scheme, with trailing spaces and capital letters in scheme uri
Only the uuid scheme should be recognised. We expect to receive 2 content protection events
https://bugzilla.gnome.org/show_bug.cgi?id=758064
Test header download error.
Let the adaptive demux download a few bytes, then instruct the
GstTestHTTPSrc element to generate an error while the fragment header
is still being downloaded.
https://bugzilla.gnome.org/show_bug.cgi?id=762144
Moved testQuery after testFragmentDownloadError so that testDownloadError
and testFragmentDownloadError are grouped together.
The commit just moved the testQueryCheckDataReceived and
GST_START_TEST (testQuery) functions but git gets confused in matching the
lines and reports a lot of changes in the patch.
https://bugzilla.gnome.org/show_bug.cgi?id=762144
To allow the adaptivedemux live stream tests to run in non-realtime, use a
GstTestClock as the system clock. This allows the unit tests to complete
more quickly than if they had to complete in real time.
https://bugzilla.gnome.org/show_bug.cgi?id=762147
Deallocate GObject* with g_object_unref instead of gst_object_unref.
Even if it works now, it is confusing and in the future it might
not work if any GstObject specifics are added.
https://bugzilla.gnome.org/show_bug.cgi?id=762142
When the start_type is GST_SEEK_TYPE_NONE for a forward seek
(or stop_type for a reverse) is not set on a snap seeking operation,
the element should use the current position and then snap as requested.
Also fixes uninitialized variable complaint by clang about
'ts' variable.
All hlsdemux tests create a GstStructure called "state" that can be used
by test cases to store information during a test. The name of this
structure is arbitrary. When the code was written, the intention was
to use the name of the test, to aid debugging. However, during
development this was lost, so that the state GstStructure is always
given the name "setup_test_variables".
This commit changes this so that the name of the test is used.
https://bugzilla.gnome.org/show_bug.cgi?id=762684
The default one is 6 minutes, the test was using 5 minutes so just
resort to using the default.
For the non-valgrind test also use the default 20 secs instead of
reducing it to 6s. No real reason to set a custom value here.
When caps are already negotiated it should be possible to
select formats other than the one that was negotiated. If downstream
allows alpha video caps and it has already negotiated to a non-alpha
format, caps queries should still return the alpha caps as a possible
format as caps renegotiation can happen.
Includes tests (for compositor) to check that caps queries done after
a caps has been negotiated returns complete results
https://bugzilla.gnome.org/show_bug.cgi?id=757610
The function actually returns the segment availability start time (as defined by the standard).
That is at the end of the segment, but it is called availability start time.
Availability end time is something else (the time when the segment is no longer
available on the server). The function name was misleading.
https://bugzilla.gnome.org/show_bug.cgi?id=757655
The demux_sent_eos callback is unused in tests. It was also registered on
a wrong pad, so it actually triggered when demux received eos from a
fragment download.
https://bugzilla.gnome.org/show_bug.cgi?id=760328
Adds unit tests similar to the ones that we have for DASH and HLS.
Tests:
* manifest parsing finishes successfully
* some queries (duration, seekable, latency)
* seeking with various values and flags
Similar to HLS but DASH has the extra issue that it can have
multiple streams so snapping can be tricky as streams usually
won't be aligned.
For now, those tests handle the case of only having a single
stream.
https://bugzilla.gnome.org/show_bug.cgi?id=759158
Handling the ghostpad and its internal pad was causing more issues
than helping because of their coupled activation/deactivation
actions.
As we have to install custom chain,event and query functions it is
better to use a floating sink pad internally in the demuxer and just
use those pad functions to push through a standard pad in the demuxer
https://bugzilla.gnome.org/show_bug.cgi?id=757951
The URI attribute from the EXT-X-KEY tag and the URI attribute from the
EXT-X-I-FRAMES-ONLY tag are both quoted-string attibutes that have their
quotation marks removed during parsing. The CODECS attribute of the
EXT-X-STREAM-INF is also a quoted-string attribute, but this attribute
was not being un-quoted.
This commit changes the parser to always unquote all quoted-string
attributes and adjusts the unit tests to this new bevahiour for the
CODECS attribute.
An additional test is added to check that parsing of all of the fields
in the EXT-X-STREAM tag is correct, including those that contain comma
characters.
https://bugzilla.gnome.org/show_bug.cgi?id=758384
Using the new GstAdaptiveDemux test framework, add tests that
exercise hlsdemux. The following tests are added:
simpleTest
A simple playlist that contains some media URLs
testMediaPlaylist
A master playlist with a variant playlist that contains media URLs
testMediaPlaylistNotFound
A master playlist that points to a missing variant playlist
testFragmentNotFound
A master playlist with a variant playlist that contains media URLs
There is a missing media file referenced from the variant playlist.
testFragmentDownloadError
A master playlist with a variant playlist that contains media URLs
During the download of one media file, the test simulates the network
connection being dropped.
testSeek
A simple test of trying to perform a seek on an HLS stream.
To allow code from dash_demux.c to be used by other elements
that are based upon GstAdaptiveDemux, the code has been
refactored into four new files:
adaptive_demux_engine.[ch]
adaptive_demux_common.[ch]
The code in adaptive_demux_engine.c provides a generic
test engine for elements based upon GstAdaptiveDemux.
The code in adaptive_demux_common.c provides a set
of utility functions that are common between the tests
for hlsdemux and dashdemux.
As part of the refactoring, variables in structures were
renamed from using camelCase to underscore_case to match other
GStreamer source code.
The fake_http_src was renamed test_http_src and changed to use
callbacks to provide input data and error conditions. Rather than
using an array of input data that tries to encode all the
possible use cases for the GstTestHTTPSrc element, use a struct of
callbacks.
Users of this element are obliged to implement at least the src_start
callback, which provides a way to link from a URI to the settings
for that URI.
gst_mpdparser_parse_utctiming_node does not validate the parsed values completely. The following scenarios are incorrectly accepted:
- elements with no schemeIdUri property should be rejected
- elements with unrecognized UTCTiming scheme should be rejected
- elements with empty values should be rejected
The last one triggers a division by 0 in gst_dash_demux_poll_clock_drift:
clock_drift->selected_url = clock_drift->selected_url % g_strv_length (urls);
because it urls is a valid pointer to an empty array.
https://bugzilla.gnome.org/show_bug.cgi?id=759547
The videoframe-audiolevel element acts like a synchronized audio/video "level"
element. For each video frame, it posts a level-style message containing the
RMS value of the corresponding audio frames. This element needs both video and
audio to pass through it. Furthermore, it needs a queue after its video
source.
https://bugzilla.gnome.org/show_bug.cgi?id=748259
This no longer does anything, and it was marked as CONSTRUCT_ONLY
which means someone would really have to go out of their way to
be able to set this, which would only be done in very custom
scenarios, if ever, and those will likely target a specific
version of GStreamer then, so probably not much point keeping
it deprecated for a while before removing it.
The Onvif Streaming Specification specifies that the NTP timestamps
in the Onvif extension header indicaes the absolute UTC time associated
with the access unit. But by using running time we can not achieve that,
since a frame's running time depends on the played interval, whether a
non-flushing is done, etc. Instead we have to use the stream time.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
It is now possible to update the currently used ntp-offset with a
custom serialized downstream event. The element will read the ntp-offset
property when doing the state transition from READY to PAUSED and
use that offset until it receives a "GstNtpOffset" event, which also
has a "ntp-offset" attribute in that it's structure. In case the
property is not set and no event has been received, the element will
guess the npt-offset with help of the clock. If no clock can be
retrieved, the element will error out and stop the data flow.
The same event is also used for updating the D/E-bits in the RTP
extension header. The discont flag in a buffer can be set whenver a
live/network source looses a frame, but that is not the type of
discontinuity that the onvif extension header should reflect. The
header is mainly used for playback of a track concept, in which
gaps can be present, and it's those kind of gaps that should be
highlighted with the D- and E-bits.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
