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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2149>
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Split plugin into features including
dynamic types which can be indiviually
registered during a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2038>
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Reintroduce socket descriptor parameter removed in 327ad84e to
"caller-added" and "caller-removed" signals, just set it always to zero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2004>
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The caller's IP and port is enough for unique identification. Don't leak
the socket handle since using it in unadvised libsrt calls from the
application could break the SRT element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1772>
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In listener mode, gst_stats() returns an independent set of
statistics for every connected caller. Having the caller's IP and port
present in each structure allows to correlate the statistics with a
particular caller that has been announced by "caller-added" signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1772>
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Allows compiling the plugin against old headers.
For SRTO_BINDTODEVICE there's nothing we can do, since the value depends on
configuration options of the library. Nice.
Fixes build with libsrt < 1.4.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1945>
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Use GST_RESOURCE_ERROR_NOT_AUTHORIZED code in posted error messages
related to SRT authentication (e.g. incorrect or missing password) so
that the application can recognize them more easily.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1943>
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On an error event, epoll wait puts the failed socket in both readfds and
writefds. We can take advantage of this and avoid explicitly checking
socket state before every read or write attempt.
In addition, srt_getrejectreason() will give us more detailed
description of the connection failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1943>
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Any socket option that can be passed to libsrt's srt-live-transmit
through SRT URI query string is now recognized.
Also make the code that applies options to SRT sockets more generic.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1842>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1541>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1541>
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The pipeline now gets stuck in gst_srt_object_write_one() until the
receiver comes online, which may or may not be desired based on the use
case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1836>
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So that the application gets notified may react to it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1935>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1725>
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This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.
Thread A (Holds object lock, wants sock lock):
#2 gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
#3 gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
#4 gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
#5 g_object_get_property from libgobject-2.0.so.0
Thread B (Holds sock lock, wants object lock):
#2 gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
#3 gst_element_post_message at gstreamer/gst/gstelement.c:2123
#4 gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
#5 gst_element_message_full at gstreamer/gst/gstelement.c:2298
#6 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
#7 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#8 gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#9 gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
#10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179
Fixes d2d00e07acc2b1ab1ae5a728ef5dc33c9dee7869.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
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`delay` should be a GstClockTimeDiff since SRT time is int64_t.
All values are in local time so we should never see a srctime that's in
the future. If we do, clamp the delay to 0 and warn about it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
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A zero srctime is a missing srctime. Apparently this can happen when
["the connection is not between SRT peers or if Timestamp-Based Packet
Delivery mode (TSBPDMODE) is not enabled"][1] so it may not apply to us,
but it's best to be defensive.
[1]: https://github.com/Haivision/srt/blob/v1.4.2/docs/API.md#sending-and-receiving
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
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If we don't have a clock, stop the source instead of asserting in
gst_clock_get_time. This can happen when the element is removed from the
pipeline while it's playing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
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Instead of leaking it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1668>
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The connection might be broken, which we should detect instead of just
aborting the write.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1669>
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SRT provides the original timestamp of a packet (with drift/skew corrected for
local clock), which is what should be used for timestamping the outgoing
buffers. This ensures that we output the packets with the same timestamp (and by
extension rate) as the original feed.
Also detect if packets were dropped (by checking the sequence number) and
properly set DISCONT flag on the outgoing buffer.
Finally answer the latency queries
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1658>
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Now that SRT no longer needs the family when creating the socket, this
code has become useless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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See https://github.com/Haivision/srt/blob/73ee1e1a3e3adc2702a9a5057d101ef80447b38c/docs/API-functions.md#srt_socket
`srt_create_socket()` was added in
https://github.com/Haivision/srt/commit/4b897ba92d34f1829a1c6e419eeab17f0763a0fc
and srt `v1.3.0` is the first release that has it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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This would provoke error messages from SRT.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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`srt_startup` can also return 1 if it was successful. Avoid warning in
this case.
Avoid a race when checking whether we need to call it at all.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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The [SRT documentation][1] specifies exact types for the socket options.
Make sure we match these.
This reverts the linger workaround in commit 84f8dbd932029220ee86154dd
and extends srt_constant_params to support other types than int.
[1]: https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1615>
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To propagate error, this commit sets a reason. Otherwise, the function
caller should check if `error` is NULL when the return value is not normal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
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If an address can't be parsed as IP address, try resolving it via
GResolver instead. SRT URIs more often than not contain hostnames and
without trying to resolve them we won't be able to handle such URIs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1493>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
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This makes `gst_srt_object_validate_parameters` work properly since
`localaddress` and `localport` will be missing if the URL did not
provide them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
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This fixes `gst_srt_object_set_string_value` in particular because the
value might not be a static string.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
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The doc system gets confused when we register the exact same
class as multiple elements, so make a subclass for each.
Also wrap registration of deprecated elements with #ifndef GST_REMOVE_DEPRECATED.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1354>
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fixes #1277
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1260>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1273>
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Attempting to reach an inactive SRT peer in caller mode
was leaking an fd every few seconds in the gst_srt_object_close()/open()
loop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1293>
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The stream id starts with '#!::' according to SRT Access Control[1],
but GstURI requires URI encoded string.This commit introduces additional
property to set the id by normal string.
[1] https://github.com/Haivision/srt/blob/master/docs/AccessControl.md
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/999>
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So we don't lose them. Split gst_srt_object_open_internal for internal
reconnections that don't reset the accumulated bytes.
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The field is a uint64_t.
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It seems overly complicated.
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Use GST_OBJECT_LOCK (srtobject->element) to protect only the fields
involved in property access.
Introduce a new mutex srtobject->sock_lock to go with
srtobject->sock_cond and protect the list of callers from concurrent
access.
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GstSRTObject is a structure that has an actual GstElement
which is extended to srt{src,sink}.
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Add `wait-for-connection` property to `srtsink` element. This
property allows the element to process packets even when no clients
are connected.
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Based on Stream ID, the application can accept or reject the connection,
select the desired data stream, or set an appropriate passphrase for the
connection. Example usage:
srt://127.0.0.1:1234?streamid=mystream
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