Constantly updating the ts_offset results in audiable glitches
when streaming audio using ntp-sync=true. By requiring a minimum
offset before updating ts_offset this can be mitigated. Added a
parameter which can be used to set min_ts_offset in ntp-sync mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
Hotdoc should be able to extract and parse comments out of these. Just
need to be careful to only add the glob in directories that actually
contain *.m (objc) and *.mm (objcpp) files.
Also fix some doc comments and remove redundant ones.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1614>
This is listed as a public interface implemented by osxaudio, so we
need to mark it as a plugin API so that it's listed in the
documentation correctly.
This is an ancient symbol, so add it to the symbol index too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1601>
Starting with libsoup3, there is no attempt to handle thread safety
inside the library, and it was never considered fully safe before
either. Therefore, move all session handling into its own thread.
The libsoup thread has its own context and main loop. When some
request is made or a response needs to be read, an idle source
is created to issue that; the gstreamer thread issuing that waits
for that to be complete. There is a per-src condition variable to
deal with that.
Since the thread/loop needs to be longer-lived than the soup
session itself, a wrapper object is provided to contain them. The
soup session only has a single reference, owned by the wrapper
object.
It is no longer possible to force an external session, since this
does not seem to be used anywhere within gstreamer and would be
tricky to implement; this is because one would not have to provide
just a session, but also the complete thread arrangement made in
the same way as the system currently does internally, in order to
be safe.
Messages are still built gstreamer-side. It is safe to do so until
the message is sent on the session. Headers are also processed on
the gstreamer side, which should likewise be safe.
All requests as well as reads on the libsoup thread are issued
asynchronously. That allows libsoup to schedule things with as
little blocking as possible, and means that concurrent access
to the session is possible, when sharing the session.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/947
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1555>
This is due to an unsafe usage of the pad task. We didn't ensure proper
ownership of the task. That race involved the task being released too early,
and was detected, luckily, by the glib mutex implementationt that
reported the mutex being disposed while being locked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1478>
On Android (especially) and for static builds in general it is safer to link
against libsoup and have the dynamic custom loading disabled. For those cases we
can safely assume the application will use either libsoup2 or libsoup3 and not
both.
Fixes#939
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1536>
When multiple streams are bundled together, there may be more
than one red payload type to handle.
In addition, as the red decoder works by filling in gaps in
the seqnums, there needs to be one rtp_history queue per sequence
domain.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.
In ulpfecenc we add one in that case to our protection buffers.
This makes TWCC functional when UlpRed is used in webrtcbin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
XNextEvent() blocks indefinitely in absence of X11 events, which can
prevent the pipeline from stopping.
This can cause problems when ximagesrc is used in "remote desktop"
scenarios and the GStreamer application itself, through which the user
is viewing and controlling the machine, is the only source of input
events.
Replace the call with non-blocking XCheckTypedEvent().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1438>
since `gst_caps_replace()` and `gst_pad_set_caps()` both ref the caps and neither of them takes the ownership of the caps -> it must be unreffed in `gst_multi_file_src_set_property()`
to test the leak (on Unix): `echo coucou > /tmp/file.txt && GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7" gst-launch-1.0 multifilesrc location=/tmp/file.txt caps='txt' ! fakesink`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1436>
Also ignore 0x0 sizes in the fallback case and assume the size can be
anything between 1x1 and MAXxMAX.
This fixes the case where a width=0, height=0 caps are created. Whith
this patch the caps will contain width=[1,MAX], height=[1,MAX].
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1396>
Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.
I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
If qt asks us to redraw before we have both set a buffer and caps we
would attempt to use the new caps with the old buffer which could result
in bad things happening.
Only update caps from new_caps once the buffer has actually been set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1346>
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.
Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
Negative composition time offsets are only allowed with version 1 of the
box, however we parse it as a signed value also for version 0 boxes as
unfortunately there are such files out there and it's unlikely to have
(valid) huge composition offsets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
g_sequence_remove_range's end iter is exclusive, so if one
wants to remove that item as well, it should be called with
the next iter.
This could in theory fix an issue where:
* The sequence isn't entirely trimmed, with an old item lingering
* Following FEC packets are immediately discarded because they
arrived later than corresponding media packets, long enough for
seqnums to wrap around
* We now try to reconstruct a media packet with a completely obsolete
FEC packet, chaos ensues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1341>
We need to hold onto the last buffer until the next buffer arrives.
Before, if a caps change comes we would remove the currently rendering
buffer. if Qt asks use to render something, we would render the dummy
black texture.
Fixes a period of black output when upstream is e.g. changing resolution
as in hls adaptive bitrate scenarios.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1338>
The `gst_v4l2_buffer_pool_dqbuf` function contains this ominous comment:
/* get our GstBuffer with that index from the pool, if the buffer was
* outstanding we have a serious problem.
*/
outbuf = pool->buffers[group->buffer.index];
Unfortunately it is common for buffers in _output_ buffer pools to be
both queued and outstanding at the same time. This can happen if the
upstream element keeps a reference to the buffer, or in an encoder
element itself when it keeps a reference to the input buffer for each
frame.
Since the current code doesn't handle this case properly we can end up
with crashes in other elements such as:
(gst-launch-1.0:32559): CRITICAL **: 17:33:35.740: gst_video_frame_map_id: assertion 'GST_IS_BUFFER (buffer)' failed
and:
(gst-launch-1.0:231): GStreamer-CRITICAL **: 00:16:20.882: write map requested on non-writable buffer
Both these crashes are caused by a race condition related to releasing
the same buffer twice from two different threads. If a buffer is queued
and outstanding this situation is possible:
**Thread 1**
- Calls `gst_buffer_unref` decrementing the reference count to zero.
- The core GstBufferPool object marks the buffer non-outstanding.
- Calls the V4L2 release buffer function.
- If the buffer is _not_ queued:
- Release it back to the free pool (containing non-queued buffers).
**Thread 2**
- Dequeues the queued output buffer.
- Marks the buffer as not queued.
- If the buffer is _not_ outstanding:
- Calls the V4L2 release buffer function.
- Release it back to the free pool (containing non-queued buffers).
If both of these threads run at exactly the same time there is a small
window where the buffer is marked both not outstanding and not queued
but before it has been released. In this case the buffer will be freed
twice causing the above crashes.
Unfortunately the variable recording whether a buffer is outstanding is
part of the core `GstBuffer` object and is managed by `GstBufferPool` so
it's not as straightforward as adding a mutex. Instead we can fix this
by additionally recording the buffer state in `GstV4l2BufferPool`, and
handle "internal" and "external" buffer release separately so we can
detect when a buffer becomes not outstanding.
In the new solution:
- The "external" buffer pool release and the "dqbuf" functions
atomically update the buffer state and determine if a buffer is still
queued or outstanding.
- Subsequent code and a new
`gst_v4l2_buffer_pool_complete_release_buffer` function can proceed to
release (or not) a buffer knowing that it's not racing with another
thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1010>
The pipeline flow for receiving looks like this:
rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
rtpptdemux ! stream_fec_decoder ! ...
There are two places where a fec decoder could be placed.
1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
for each ssrc/pt produced
2. after rtpssrcdemux but before rtpjitterbuffer: added for the
rtpst2022-1-fecenc/dec elements,
However, there was some cross-contamination of the elements involved and
the request-fec-decoder signal was also being used to request the fec
decoder for the session_fec_decoder which would then be cached and
re-used for subsequent fec decoder requests. This would cause the same
element to be attempted to be linked to multiple elements in different
places in the pipeline. This would fail and cause all kinds of havoc
usually resulting in a not-linked error being returned upstream and an
error message being posted by the source.
Fix by not using the request-fec-decoder signal for requesting the
session_fec_decoder and instead solely rely on the added properties for
that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>
