Make sure we don't read beyond the atom boundary. Note that the code
behaves slightly differently in the corner case where there is not
enough atom data for the specified number of samples (n_samples_time)
in the atom, but still enough data to fill the pre-allocated index of
n_samples entries: before we would just stop parsing the stts data
and continue, whereas now we will likely error out. This should not
be a problem in practice though. We could maintain the old behaviour
by doing reads with a size check inside the loop if needed.
Use GstByteReader to parse stsz and stsc chunks, and check size of
available data before parsing it, instead of blindly assuming there
will be enough data. Fixes crashes with some fuzzed/broken files.
At the end, Dirac streams have an EOS packet with 0 length.
Don't ever sit in an infinite loop when processing one. Allows
muxing Dirac into mkv to complete successfully.
Add a property to allow control over what event causes a file
to finish being written and a new file start. The default is
the same as before -- each buffer causes a new file to be
written. Added is a case where buffers are written to the
same file until a discontinuity in the stream.
Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.
Fixes#594490
When we receive a reordered packet with the same timestamp as the previous one
(which can happen for fragmented packets) don't consider the packet as lost but
instead wait for the reordered packet to arrive.
Switch the warning-level, so that a reordering does not get a warning, only
an actual produced lost-packet.
Fixes#594251
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
When receiving a sync-packet, all sessions with the same cname will be compared
and synced together. In this process, there could still be references to a
session that has been shut down in the meanwhile.
This patch makes sure that these references are removed when shutting down a
session, so that the syncing can be done safely.
Fixes#594283
The priv->clock_rate variable could become -1 between when its checked to not
be -1 and when its used, causing an assertion. Fixed by taking the mutex
earlier in the chain() function.
Fixes#593955
This reintroduces the fix for bug #593391. It also applies it in
gst_rtp_session_sync_rtcp() which has very similar code to
gst_rtp_session_send_rtcp().
When we construct a timestamp that would result in a timestamp that is earlier
than when the packet was received, reset the skew calculation as this is
probably a sign that the sender restarted or paused.
Fixes#593354
Recent changes in gst-plugins-good/gst/effectv prevents it from being compiled
with gcc 3. The problem is that the new code uses preprocessor conditionals
within a macro call which does not work with older versions of gcc.
Fixes bug #593688.
In case of non-interleaved (= sequentially payloaded) streams,
the AU-Index serves little purpose (that is not already covered by
RTP fields). (Broken) Payloaders might consider this field then
to be disregarded and have non spec compliant values, e.g. each
RTP packet having AU-Index 2 (rather than 0). As such, ensure/force
simple sequential sending of non-interleaved streams.
3GPP specs define a number of tags along with precise layout. While these
are normally expected to be found in a container whose major brand is a
3GPP brand, this may also happen when a 3GPP brand is only mentioned as a
compatible brand. Apply some checks, heuristic and fallbacks to extract
such tags as well.
Handle large, invalid or otherwise unusual chunk sizes.
Verify some chunk sizes to be at least the size they are
expected to be and round up some sizes to even number for
e.g. offset administration, which must also be properly
tracked in push mode.
After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.
Connect to the pad-removed signal of the ptdemux elements so that we remove the
ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
the sinkpads.
Fixes#561752
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
Free the session when all the request pads are released.
Don't mess with the session list in free_session as it is called from a foreach
on that list.
Set the state of the upstream element to NULL first.
See #561752
We usually only get SR packets in our chain function but if an invalid packet
contains the SR packet after the RR packet, we must not fail but simply ignore
the malformed packet.
Fixes#581375
Since neither rtpmanager nor any of the payloaders properly implement
pad allocation, there is no way for the rtpmanager to inform downstream elements
of the new SSRC if there is an SSRC collision. So the warning is emitted all the
time and it is confusing.
Fixes#580144
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
When the number of participants is less than 50, the RFC allows for sending the
BYE packet immediatly instead of using the regular BYE timeout.
Fixes#567828.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
When an SSRC is found on the caps of the sender RTP, use this as the
internal SSRC. Fixes#565910.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_getcaps_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_schedule_bye_locked), (rtp_session_schedule_bye):
* gst/rtpmanager/rtpsession.h:
Rename a method to better reflect what it really does.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp):
Use method to get the internal SSRC.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_set_property), (rtp_session_get_property):
Add property to congiure the internal SSRC of the session.
Fixes#565910.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
Only change the SSRC of the session and reset the internal source when
the SSRC actually changed. See #565910.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate):
* gst/rtpmanager/rtpsource.h:
When no payload was specified on the caps but there was a clock-rate,
assume the clock-rate corresponds to the first payload type found in the
RTP packets. Fixes#565509.
Original commit message from CVS:
Patch by: Arnout Vandecappelle <arnout at mind dot be>
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last outgoing timestamp and of the last sender-side
time. Timestamps can only go forward if they do at the sender
side, can only go back if they do at the sender side, and remain the
same if they remain the same at the sender side. Fixes#565319.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (obtain_source),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye):
Make obtain_source return an aditional ref so that we don't lose our ref
to it when a session cleanup occurs when we are emiting a signal.
Emit the on_new_ssrc signal for the CSRC, not the SSRC.
Fixes#562319.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync),
(gst_rtp_bin_clear_pt_map):
Reset the sync parameters when clearing the payload type map too.
Fixes#562312.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_client),
(gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream),
(gst_rtp_bin_class_init), (new_ssrc_pad_found):
* gst/rtpmanager/gstrtpbin.h:
Remove a lot of per stream state that is not needed and pass new info in
the method call.
Add signal to reset sync parameters.
Avoid parsing the caps to get a clock_base, we get this from the sync
signal now.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_set_property),
(rtp_session_get_property):
Add property to configure the RTCP MTU.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(copy_source), (rtp_session_create_sources),
(rtp_session_get_property):
Add G_PARAM_STATIC_STRINGS.
Add property to return a GValueArray of all known RTPSources in the
session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_create_sdes), (rtp_source_set_property),
(rtp_source_get_property):
Remove properties to set the various SDES items, an application is never
supposed to change the RTPSource data.
Change the SDES getter properties to one SDES property that returns all
SDES items in a GstStructure.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Release the right pads on rtpbin. Fixes#561752.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain_rtcp):
Initialize return value to fix compiler warning about uninitialized
variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Fix GST_DEBUG call to only have as many arguments as required
by the format string. Fixes a compiler warning.
