| Commit message (Collapse) | Author | Age | Files | Lines |
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Also change error message to more accurately reflect cases in which
it can occur.
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In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth
header items and then passes them to gst_rtsp_connection_set_auth_param()
without freeing.
Fixes #594133
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Put the SDP attributes on the caps too so that they can be used by
depayloaders.
See #564437
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After a redirect we want to use the same protocols that we were using for the
current url.
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Add property to make the client suggest a blocksize to the server.
Fixes #585549
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We need to set the state of gstrtpbin to the same state as our source elements.
This fixes fallback to TCP again.
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Free messages correctly.
Fixes #577318
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Make the fakesrc that is responsible for sending dummy packets silent.
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Don't send a TEARDOWN request when we did not manage to successfully setup a
stream.
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Fix various version of find_stream_by_* by not trying to convert an int to a
pointer and vice versa, for portability reasons.
Fixes #581333
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Fix a typo in the dummy NAT packet sending code.
Fixes #581329
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Server eof (e.g. connection closed) is announced as connection closed,
so better record state and act accordingly to prevent (read/write)
errors during subsequent teardown/cleanup sequences. #Fixes 580851.(c).
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timestamps following flush/seek should be consistent between
UDP and TCP interleaved case. Fixes #580851.(b).
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A max range that overflows should not be trusted,
nor should a max range that equals the min range.
Fixes #580851.(a).
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We can use the SKIP seek flag to instruct the server to send data faster then
normal but with the same bandwidth.
Fixes #537609
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We need to release the state lock before trying to wait for the task to end
because the task might also take the lock.
Fixes #577671
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Fix some pad leaks.
See #577318.
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t is being overwritten after, before it's used.
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After a seek mark all streams with discont as it was said in the debug line.
Fixes that buffers after a seek are generated without a valid timestamp.
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Permit properly handle the EOS condition when server report it in a request.
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Allow and ignore http:// at the start of the proxy setting, like
souphttpsrc.
Fixes #573173
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Fix memory leak in rtspsrc because we didn't unref the udpsrc pad.
See #577318
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The on-npt-stop signals was added only recently to rtpjitterbuffer in
-bad, so check if the signal exists before g_signal_connect()ing to
it, to avoid warnings.
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Connect to the on_npt_stop signal of the session manager to schedule the EOS
actions.
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missing
Try to post a decent error message when it looks like we're failing
because the Real RTSP extension plugin is missing. Also add i18n
bits for rtspsrc so our error messages get translated.
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When timestamping in TCP mode, log the first timestamp we put on the buffers.
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don't send a PAUSE request when we are no longer connected.
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---
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Fix parsing of the range headers.
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Close the connection even when we fail to send the teardown message.
Use the connection url (which is a copy of the src url).
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---
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Add support for http tunneling and a new rtsph:// uri for it.
See #573173.
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Some old servers don't like us doing RTCP and thus we need a property to disable
it. See #573173.
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MS RTSP spec states that the UDP port pair used in subsequent SETUP
requests for various streams must be identical (since there will actually
be only 1 stream of muxed asf packets). Following traditional specs and
using different port pairs in the SETUPs for separate streams will result
in all but the first one failing and only one stream being streamed.
So, in appropriate circumstances, retry UDP SETUP using previously used
port pair. Fixes #552650.
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Call the receive_request extension methods so that extensions can handle the
server request if they want.
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Keep track of the state of the connection and don't try to send TEARDOWN when
the server has closed the connection.
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Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
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connect-failed socket by erroring out quickly....
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Prevent further read/write actions taken to the connect-failed socket by
erroring out quickly. See #562258.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (new_session_pad),
(gst_rtspsrc_parse_range):
Add some more debugging.
Use the reanges received from the server unconditionally.
Fixes #561625.
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Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes #559545.
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compatibility with some broken servers. See #53...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
(gst_rtspsrc_change_state):
Only send one transport at a time for improved compatibility with some
broken servers. See #537832.
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was playing.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_perform_seek):
Only pause/play in the seek handler when the source was playing.
Fixes #529379.
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