Bringing back the check removed in the previous commit but have that check be a
g_assert. Changing the function to static void since return can never be False,
because audio format will never be unkown.
func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
The conditional checking if func_index is >= 0 and < 8 will always be true.
Removing it.
CID 1226442
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=740505
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.
Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=741783
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).
https://bugzilla.gnome.org/show_bug.cgi?id=741398
Instead of constantly querying upstream, just cache the last duration,
and in the unlikelyness we might have gone over query again before
deciding we are EOS.
Cut 15% cpu off matroskademux streaming thread (srsly...)
This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
it is marked as a draft, this part was confirmed to be correct on
IRC), and allows one to determine whether a demuxed stream is
multistream or not, and thus set the multistream caps field
accordingly. In turn, this means downstream does not have to guess.
https://bugzilla.gnome.org/show_bug.cgi?id=740744
It's like rendering a buffer list, just with one buffer.
Has the added advantage that if there are multiple clients
we can send the buffer to all the clients in one go.
We unlock and re-lock the client lock while emitting the
removed signal, which causes inconsistencies in the client
list vs. the client counts. Instead, remove the client from
the list already before emitting the signal and put it into
a temporary list of clients to be removed. That way things
look consistent to the streaming thread, but signal callbacks
can still do things like get stats from removed clients.
Add prototype for a render_list() function that can use a
sendmmsg-style g_socket_send_messages() function once it lands
in GLib. We can use this infrastructure to send multiple buffers
made up by multiple memories to multiple clients in one go, which
drastically reduces the number of syscalls made when sending
high-bitrate video streams.
https://bugzilla.gnome.org/show_bug.cgi?id=732152
Use the refcount for memory management and keep track
of the number of duplicate clients in a separate
variable. This will be useful later, and means we
don't have to hold the OBJECT_LOCK all the time.
https://bugzilla.gnome.org/show_bug.cgi?id=732866
Since "basetransform: Fix caps equality check" commit a7f357,
set_info() will not be called anymore if crop didn't change
the caps. This is fixed by setting "need_update" boolean when
cropping properties has been changed, and then applying these
if they where not applied before rendering the next frame. This
patch also fixed the locking, dropping un-needed custom lock,
and no holding needless lock while doing the operation as we
already hold the streaming lock.
https://bugzilla.gnome.org/show_bug.cgi?id=740787
In some cases the currently set GstVideoInfo is not interlaced, but
upstream caps are interlaced and the info is passed in the filter,
we should take that info into account and make sure that we do not
consider that case as a "pass through" case.
https://bugzilla.gnome.org/show_bug.cgi?id=741407
A race condition in the state change function may cause buffers
to be unreffed while they are still used by the streaming thread
in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
up to the parent class first in the state change function to
make sure streaming has stopped and only then free those buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=741381
When dealing with fragmented files, we will get more accurate duration
information via the mfra and moof atoms.
In order for playback to not stop at the initial duration (from the
moov atom), we need to check and update the various duration variables
when we find more information.
Fixes playback of fragmented files in pull mode
Adds a new set of properties to make pushfilesrc output a TIME SEGMENT
(instead of the filesrc BYTE SEGMENT).
When time-segment is set to True the following will happen:
* Seeks are refused (data starts from the beginning of the file)
* The BYTE segment will be replaced by a TIME segment with the values
specified in the various properties
* The first outgoing buffer will have a timestamp set on it (by default
it has a value of GST_CLOCK_TIME_NONE)
When seeking or finding the previous keyframe, do
comparisons against targets and segments using composition time
to correctly decide which sample times match.
We used to setup an iterator with 1 GValue set with a NULL object
pointer which is not the normal way to do that. Instead we should make
sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
Currently during header parsing, we scan through the entire file
and skip every moof+mdat chunk for fragmented mp4s, which makes
start-up incredibly slow. Instead, just stop at the first moof
chunk when have a moov, and start exposing the streams, so we
can go and start handling the moofs for real.
When an caps-event is received, we must immediately change the crop
to videocrop correctly changed caps-event dimension, otherwise the
videocrop will first use the previous value of the crop that when
resizing video to a smaller resolution may cause an error.
https://bugzilla.gnome.org/show_bug.cgi?id=740671
Empty segments in an edit list have a media_start time of -1,
as they don't actually play any media. Allow for that when
aligning to the reference stream in reverse play.
Put a 0-byte at the end of the event string. Does not break ABI because
old depayloaders will skip the 0 byte (which is included in the length).
Expect a 0-byte at the end of the event string or a ; for old
payloaders.
See https://bugzilla.gnome.org/show_bug.cgi?id=737591
Both Firefox and Chrome uses VP8 as the encoding in their SDP.
Adding this now defacto standard name removes the need for special
case in SDP parsing code.
https://bugzilla.gnome.org/show_bug.cgi?id=737810
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.
There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.
After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.
Remove the condition for the immediate feedback for now.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.
https://bugzilla.gnome.org/show_bug.cgi?id=738722
left, right, top, bottom can be set from range of -2147483648 to 2147483647
when i launch the videobox element with that values, it gives a critical error
(gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.
https://bugzilla.gnome.org/show_bug.cgi?id=738838
The loop in zoomFilterSetResolution is meant to change the values in the
zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
but no conditions that change the value of decc are ever met and the array is
filled with zero for each element. Which is the initial state of the
array before the loop begins.
The loop does nothing.
https://bugzilla.gnome.org/show_bug.cgi?id=728353
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
In order to have a full mapping between channel positions in the audio
stream and loudspeaker positions, the channel-mask alone is not enough:
the channels must be interleaved following some Default Channel Ordering
as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.
As a Default Channel Ordering use the one implied by
GstAudioChannelPosition which follows the ordering defined in SMPTE
2036-2-2008[2].
NOTE that the relative order in the Top Layer is not exactly the same as
the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
using so may channels are already aware of such discrepancies.
[1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
[2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
Otherwise the CAPS event will be dropped and we never configure any caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.
https://bugzilla.gnome.org/show_bug.cgi?id=737735
This is about converting the format, not about converting any widths and
heights. Subclasses are expected to handler different resolutions themselves,
like the videomixers already do properly.
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_new (&sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
'GstRTSPResult' [-Werror,-Wenum-conversion]
res = gst_sdp_message_parse_uri (uri, sdp);
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Remove pads from flow combiner and reset last
flow return to FLOW_OK by resetting the flow combiner.
This prevents FLOW_FLUSHING when trying to re-use the
demuxer after setting it back to NULL/READY state.
https://bugzilla.gnome.org/show_bug.cgi?id=737359
DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
https://bugzilla.gnome.org/show_bug.cgi?id=737095
In 1.0, we pass the complete caps to transform_caps to allow for better
optimizations. Make this function actually work on non-simple caps
instead of just ignoring the configured filter caps.
We have to skip 12 bytes of data for the chunk, and the data size
passed to the sub-chunk parsing functions should have 4 bytes less
than the data size.
Also when parsing the sub-chunks, check if we actually have enough
data to read instead of just crashing.
https://bugzilla.gnome.org/show_bug.cgi?id=736266
Drop use of g_socket_get_available_bytes() which is
not useful on all systems (where it returns the size
of the entire buffer not that of the next pending
packet), and is yet another syscall and apparently
very inefficient on Windows in the UDP case.
Instead, when reading UDP packets, use the more featureful
g_socket_receive_message() call that allows to read into
scattered memory, and allocate one memory chunk which is
likely to be large enough for a packet, while also providing
a larger allocated memory chunk just in case the packet
is larger than expected. If the received data fits into the
first chunk, we'll just add that to the buffer we return
and re-use the fallback buffer for next time, otherwise we
add both chunks to the buffer.
This reduces memory waste more reliably on systems where
get_available_bytes() doesn't work properly.
In a multimedia streaming scenario, incoming UDP packets
are almost never fragmented and thus almost always smaller
than the MTU size, which is also why we don't try to do
something smarter with more fallback memory chunks of
different sizes. The fallback scenario is just for when
someone built a broken sender pipeline (not using a
payloader or somesuch)
https://bugzilla.gnome.org/show_bug.cgi?id=610364
This makes sure that also properties like the pixel-aspect-ratio are the same
between both streams and that the output caps contain all fields necessary for
complete video caps.
https://bugzilla.gnome.org/show_bug.cgi?id=735804
gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.
replacing the same with gst_buffer_copy as the functionality is same.
https://bugzilla.gnome.org/show_bug.cgi?id=735880
Adding an extra condition while calling gst_caps_unref (templ)
and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
gst_caps_copy (caps) in line 177, since the functionality is same.
https://bugzilla.gnome.org/show_bug.cgi?id=735795
We return EOS after the first buffer, and GstPad will make sure now that we
won't get any other buffer afterwards until a flush happens. No need to check
for it ourselves.
https://bugzilla.gnome.org/show_bug.cgi?id=735581
FLV documentation stipulates that timestamp must start at zero.
In order to respect this rule, keep the first timestamp around
and offset the timestamp from this value. This allow for longer
recording time in presence of timestamp that does not start
at 0 already.
https://bugzilla.gnome.org/show_bug.cgi?id=731352
The tags in FLV are DTS. In audio cases, and for many video format this makes
no difference, but for AVC with B-Frames, PTS need to be computed from
composition timestamp CTS, with PTS = DTS + CTS.
https://bugzilla.gnome.org/show_bug.cgi?id=731352
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
https://bugzilla.gnome.org/show_bug.cgi?id=734322
udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
been removed. This patch replaces those references to socket and close-socket
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=734987
The old default timescale of 1 millisecond produces irrational
numbers for a lot of framerate/audio-packet-duration multiples.
1/1800 is a nicer number, as it tends to produce better fractions
and therefore slightly higher accuracy overall
Change the way the output framerate is calculated
to ignore the first sample (which is sometimes truncated
in my testing) and use the new gst_video_guess_framerate()
function to recognise common standard framerates better.
Remove the code that was sorting the first 20 sample
durations and then ignoring the result.
Commit 2b9493b5 broke this in two ways: a) we should only
pass duration queries in TIME format upstream (or at least
not those in DEFAULT or BYTE format), and b) we mustn't
overwrite the default value of 'res' from TRUE to FALSE
and not set it again later. This led to bogus durations
being reported for FLV playback from file, because TIME
queries would fail (as 'res' had been set to FALSE) and
parsers then do a BYTE query as fallback and try to
guesstimate something in return, which of course goes
horribly wrong since the BYTE size returned is for the
muxed file.
When changing the properties to not be in passthrough mode anymore,
we will only accept caps we can process ourselves, potentially causing
a not-negotiated error.
https://bugzilla.gnome.org/show_bug.cgi?id=720345
This makes sense in DASH reverse playback, where the upstream dashdemux
will download DASH segments in reverse order, but push their buffers
forward to qtdemux and mark each segment start as DISCONT. This needs
to be forwarded downstream to the parser/decoder, otherwise it won't work.
https://bugzilla.gnome.org/show_bug.cgi?id=734443
When writing out a trak with an edit list, make sure the
overall file duration is also updated to reflect the
lengthening of the stream.
Add some more debug to qtdemux to warn about streams that
are longer than the file and get truncated.
We only want to unlock if we push an event downstream and
jump to done_unlock label afterwards. We would also unlock
in case of a segment seek and then unlock again later, and
nothing good can come of that.
(This code looks a bit dodgy anyway though, shouldn't it
also bail out with FLOW_EOS here in case of a segment seek
scenario, just without the event?)
gst_matroska_parse_take() would return FLOW_ERROR instead of
FLOW_EOS in case there's less data in the adapter than requested,
because buffer is NULL in that case which triggers the error
code path. This made the unit test fail (occasionally at least,
because of a bug in the unit test there's a race and it would
happen only sporadically).
Decoder complains about "notification: Invalid mode encountered.
The stream is corrupted" though, even if it works, so there's
probably something wrong with the generated codec headers.
Implement 3 different cases for handling the SR:
1) we don't have enough timing information to handle the SR packet and
we need to wait a little for more RTP packets. In that case we keep
the SR packet around and retry when we get an RTP packet in the
chain function.
2) the SR packet has a too old timestamp and should be discarded. It is
labeled invalid and the last_sr is cleared.
3) the SR packet is ok and there is enough timing information, proceed
with processing the SR packet.
Before this patch, case 2) and 1) were handled in the same way,
resulting that SR packets with too old timestamps were checked over and
over again for each RTP packet.
This patch adds supports for the incoming key management parameters for
encryption and authentication key lengths.
It also adds a new signal request-rtcp-key that allows the user to
provide the crypto parameters and key for the RTCP stream.
https://bugzilla.gnome.org/show_bug.cgi?id=730473
Use a different variable name to make it clear that we are calculating
the header size.
Correctly check that we have enough bytes to read the header bits. We
were checking if there were 5 bytes available in the header while we
only needed 3, causing the packet to be discarded as too small.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).
We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=730563
Pre-allocate buffer list of the right size to avoid re-allocs.
Avoid plenty of double runtime cast checks and re-doing the
same calculation over and over again in rtp_vp8_calc_payload_len().
Only call gst_buffer_get_size() once.
Collect buffers to send out in buffer lists instead of
pushing out single buffers one at a time. For HD video
each frame might easily add up to a couple of thousand
packets, multiply that by the frame rate and that's a
lot of push() and sendmsg() calls per second.
A good reason to push out buffers as early as possible is
latency, so we don't accumulate the whole frame in a single
buffer list, but instead push it out in a few chunks, which
is hopefully a reasonable compromise.
Make sure that if AYUV is received it will detect that it can produce
both RGB and YUV formats
Signed-off-by: Ravi Kiran K N <ravi.kiran@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=725248
The code would previously crash trying to insert a NULL string
into a hash table.
It does seem a little broken that indexing is done by MIME type
and not by index though, unless the spec says there cannot be
two parts with the same MIME type.
https://bugzilla.gnome.org/show_bug.cgi?id=659573
This event was not sent. Send it before caps, this requires the pad to
be parented. This removes warning like: "Got data flow before
stream-start event".
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475
If the wav header contains an extended chunk, we want to keep
the codec_data field, but not for raw audio.
This fixes some elements (such as adder) from failing to intersect
raw audio caps which would otherwise be intersectable.
Handle the transformation matrix cases where there are only simple rotations
(90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario
when recording with mobile devices
https://bugzilla.gnome.org/show_bug.cgi?id=679522
tentacle3d.c:268:7: error: using integer absolute value function 'abs' when
argument is of floating point type [-Werror,-Wabsolute-value]
if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) {
^
tests.c:161:16: error: taking the absolute value of unsigned type
'unsigned long' has no effect [-Werror,-Wabsolute-value]
t->diff += labs (GST_BUFFER_TIMESTAMP (buffer) - t->expected);
1) sources that have sent BYE in the past cannot be senders, since
they would have timed out to being receivers in the meantime...
2) sources that have sent BYE are now being removed earlier inside
this function
If we are inserting a packet into the jitter queue we need to keep
looping through the items until the right position is found. Currently,
the code stops as soon as an event is found in the queue.
Regarding events, we should only move packets before an event if there
is another packet before the event that has a larger seqnum.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078
If two streams request a retranmission for the same SSRC, ignore the second
one if the first oen is less than one second old, otherwise time out the first
one and ignore the second.
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.
Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
If we're missing part of the clut, do not try to use it. It seems
very likely the break was meant to break out of the switch rather
than from the loop.
Coverity 1139878
Even if one woul hope one pixel can fit in a MTU, ensure we do not
overwrite a buffer if this is not the case.
Spotted while looking at Coverity 1208786
Rework the packet queue so that the most common action (insert a packet
at the tail of the queue) goes very fast.
Report if a packet was inserted at the head instead of the tail so that
we can know when to retry _pop or _peek.
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
gstdeinterlace.c: In function 'gst_deinterlace_output_frame':
gstdeinterlace.c:1537:57: error: 'pattern.length' may be used uninitialized in this function [-Werror=maybe-uninitialized]
This actually is always initialized before it is used there, but
let's just silence gcc here.
Make a new method to disable the jitterbuffer buffering.
Rework the update_estimated_eos() method. Calculate how much time
there is left to play. If we have less than the delay of the
jitterbuffer, we disabled buffering because we might never be able to
fill the complete jitterbuffer again.
If we receive an EOS event, disable buffering. We will drain the
buffer and eventually push the EOS event out.
When we reach the estimated NPT timeout and we didn't receive an EOS
event, make one and queue it so that it can be pushed.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
When the internal-ssrc property changes, we want to send a reconfigure
upstream to make payloaders use the new suggested ssrc.
Using the internal-ssrc property to change the SSRC of a stream is not a
good idea and doesn't work when there are multiple senders, we want to
set the SSRC directly on the payloaders. Therefore, deprecate this
property.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361
Rework the logic to make buffering messages a little, make sure we
don't make the same message multiple times.
Consider the buffer full when EOS was received.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
While it seems to keep a compile time selection, I traced it
to some code copied from videoconvert, where it was removed,
with the following comment:
Also remove the high-quality I420 to BGRA fast-path as it needs
the same fix, which causes an additional instruction, which causes
orc to emit more than 96 variables, which then just crashes.
This can only be fixed in orc by breaking ABI and allowing more
variables.
Thus, I remove it here as well.
Coverity 206064
When we are buffering, we can't block and wait for the serialized query
to complete because the jitterbuffer will not try to forward the query
while buffering. Instead, just refuse the query.
The caps query handling function for the sinkpads was called for
the srcpad, and the sinkpads had none. This commit moves it to the
right pad, but nonetheless the negotiation still looks wrong.
This makes the test pass again after the recent coverity fix
and also allows interleave to work again, but someone should
really review the negotiation code and fix it.
The marker bit isn't mandatory and we had in place code to guess AU
boundaries by detecting a new picture start. This guessing code
didn't work with interlaced content that has proper marker bits
to indicate the AU boundaries. It was leaking the first field buffer
and producing a corrupted output.
fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041
The code handles a -1 pattern index, and it seems plausible
that a pattern might be found later, so it seems best to not
send an element error here.
Coverity 1139766
Create and make a key for encrypting the RTCP packets back to the server
and wrap this in a MIKEY message that we send as a header in the SETUP
request.
... as sender should keep track of segment base accumulation.
Rather, it may have some adverse effects as a spurious segment event,
e.g. in collectpads.
Try to avoid using the request-pt-map to get caps but set them directly
on the udpsrc element. That way, the caps get nicely transformed as they
pass through the different elements in the rtpbin, including the AUX and
decoder/encoder elements.
Protect caps with the lock.
Don't push the caps event from the set_property function but mark the
pad for reconfiguration so that it will renegotiate and push the new
caps event in the streaming thread.
We should open the socket when going to NULL<->READY and not in the
start/stop vemthod, which is called in READY<->PAUSED. This makes it
possible to allocate a socket without going to PAUSED (and starting the
negotiation).
Instead the queued buffer might have an old caps while the pad
is already storing the information for a new caps. Mixing those
while handling buffers will often lead to issues
https://bugzilla.gnome.org/show_bug.cgi?id=725948
Remove caps restrictions that correspond to the default and are not
required in SDP. With the new usage of having pads require a subset
of the caps, they will make the negotiation fail.
The "encoding-params" is optional in the SDP, because we now require
a subset of the caps, it would fail caps negotiatioin if it wasn't present.
So removed it from the template caps.
Keep track of what streams we did the SETUP for. We only need to
configure caps, wait for pads and push events on setup streams. We can
remove the disabled state of the stream and simplify some checks.
After we setup a stream, skip the other streams that have the same
control url. Use a skipped flag to mark streams that should be skipped.
Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
non-flushing before sending PAUSE and PLAY with the new npt range. Without this
patch, those commands would fail with EINTR as the connections were still
flushing.
It is placed inside a 'vids' struct, so it was being exposed on
a pad named video_%d. XSUB are subtitles and this patch adds
an special case for it to be exposed in a subpicture_%d pad
A media stream can have multiple payload types. Parse all the payload
types and collect the caps information. We then have to store the
pt<->caps mapping instead of 1 pt and 1 caps.
Parse the profile from the SDP and use that to negotiate the transport
instead of always using AVP.
Rework how we do some tweaks for ASF and Realmedia.
It is possible that the DTS is invalid (when we receive RTP packets from
TCP, for example). As a fallback, use the reconstructed PTS value to
calculate the buffer level.
gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
while (result == GST_FLOW_OK);
^
Several conditional statements perform comparison on RTP sequence
numbers without taking the sequence number rollover into account.
Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
comparison.
https://bugzilla.gnome.org/show_bug.cgi?id=725159
This fixes an issue with gst-rtsp-server where no sps and pps are
sent for the first intra frame, because the payloader starts working
already when receiving DESCRIBE but there is no transports so it tries
to send sps and pps, but that fails with a FLUSHING flow. But the time
for last sent sps and pps would still be set, so when PLAY arrives and
the first intra frame is to be sent there is no sps and pps sent due to
that time since last sps pps is less than spspps_interval.
https://bugzilla.gnome.org/show_bug.cgi?id=724213
In the case where we have no suitable candidate we post a warning and plug a
fake-element. Do the same when non of the candidate work.
This is more consistent and plugin the fakesink as a fallback is probably
helpful for running unit tests without requiring hardware src/sink elements.
Fixes#722981
Adds two extra checks:
- Sampling frequency on header can't be 15.
- Frame size should be at least 9 or 7, depending
on whether CRC protection is present.
https://bugzilla.gnome.org/show_bug.cgi?id=724638
Add support for a new property: tls-database. If the property is set,
the certificate database will be given to the rtsp connection if TLS
protocol is being used. If the server certificate can't be verified with
the default database, this additional database will be used.
https://bugzilla.gnome.org/show_bug.cgi?id=724396
It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
the only 4, we're fixing them instead.
recv_rtp_sink: allow proxying of the allocation query.
send_rtp_sink: allow proxying of caps and allocation. This allows us to
query caps downstream as well as get an allocator from downstream.
send_rtp_src: allow proxy of caps, this makes the caps query do
upstream.
See https://bugzilla.gnome.org/show_bug.cgi?id=723850
It was used in the past in 0.10 when there was no explicit DTS
field in buffers, now we have it in 1.x series and we can
check it directly with GST_BUFFER_DTS_IS_VALID
Do not try to use subsequent buffer timestamps to calculate
sparse streams durations because the stream is sparse and
the buffers might not be 'time adjacent'. So rely on the
duration and give the option to the pad to provide
custom 'empty' buffers to represent the gaps in the
stream, this can vary on how the data is represented.
Right now, the only sparse stream supported is tx3g subtitles.
This reverts commit 9f7b1128b1.
This should be handled automatically be rtspsrc if the AVPF profile
is used, and manual enabling of it can be done with the new-manager
signal.
If the expected packet (do_next_seqnum is TRUE) is the one we requested
for retranmission earlier, do the logic to update the retransmission
statistics as well before setting up the timers for the next expected
packet.
Also reset the retransmission counter if the timer is reused for another
seqnum.
Add an accumulator that stops the signal emission as soon as a caps has
been retrieved. Otherwise the default handler would continue emitting
the signal and possibly overwrite the result with NULL again.
Uses information gathered during EBML parsing to
forge a more suitable set of caps instead of blindly
assuming everything is video/x-matroska.
For consistency, stream type reset was added to
matroska-demux too.
https://bugzilla.gnome.org/show_bug.cgi?id=722311
To do that, enqueue the EOS event to be sent from the srcpad task
thread and flush the queue right afterwards, so that no more rtx
buffers can be sent, even if there are more requests coming in.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370
The reason behind this is to minimize the retransmission delay.
Previously, when a NACK was received, rtprtxsend would put a
retransmission packet in a queue and it would send it from chain(),
i.e. only after a new buffer would arrive.
This unfortunately was causing big delays, in the order of 60-100 ms,
which can be critical for the receiver side.
By having a separate GstTask for pushing buffers out of rtxsend,
we can push buffers out right after receiving the event, without
waiting for chain() to get called.
Instead do it like all other demuxers and let parsers and decoders
handle that. The keyframe information inside the container might
be completely wrong like in the sample file of the bug report,
and if it is correct and we push no keyframes, then the parsers
and decoders will handle that properly anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=682276
Make sure empty segments are used and pushed with a gap event
to represent its data (or lack of it)
Each QtSegment is mapped into a GstSegment with the corresponding
media range. For empty QtSegments a gap event is pushed instead
of GstBuffers and it advances to the next QtSegment.
To make this work with seeks, need to keep track of the starting
'base' to make sure it remains consistently increasing when
pushing new segment events.
For example: if a seek makes qtdemux start from 5s, the first
segment will have a base=0. When the next segment is activated,
its base time will be QtSegment.time - qtdemux.segment_base so
that it doesn't include the first 5s that weren't played and
shouldn't be accounted on the running time
This purposedly will remove the fix made for
https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this
point it was decided to respect the gaps, even if they cause
a delay on playback, because that's the way the file was crafted.
https://bugzilla.gnome.org/show_bug.cgi?id=345830
1) pt can be lower than 96
2) there is no point in checking that because rtprtxsend will not
even store buffers for payload types that it doesn't know about,
so this case will never be reached
This patch moves the creation of rtx packets to be done early,
in the src_event() function, when they are requested. The purpose
is to run gst_rtp_rtx_buffer_new() with the object locked to
protect internal data, because if it is done at the pushing stage,
we would have to lock and unlock multiple times in a row while we
are pushing the rtx buffers.
Previously there was no locking at all, which was terribly wrong.
According to ISO/IEC 13818-7, "channel_config" field in ADTS header
may have value of 0, as in the case of frame with PCE.
gst_aac_parse_detect_streams() returned FALSE for those frames
and discarded them.
The need for rewriting apparently is obsolete 0.10 leftover.
We now have caps for subtitles when we create the headers,
so we always write the correct data in the first place.
This avoids issues with writing dummy data first, then having
to come back and write correct data later. Doing so prevents
the muxed stream from being actually streamable.
https://bugzilla.gnome.org/show_bug.cgi?id=712134
Mov spec says it uses a pascal style string, while isomedia uses
a null terminated one. Store the current atoms flavor into the HDLR
to be able to generate the correct output.
https://bugzilla.gnome.org/show_bug.cgi?id=705982
This reverts commit b3aa8755fe.
We are already using the running-time because they were placed on the
buffers with gst_collect_pads_clip_running_time(). Arguably it would be
better to not modify the incomming buffers but collectpads seems to want
to use absolute timestamps from the buffers for finding the best buffer
(this can be changed with a custom compare function..).
This property allows you to specify the amount of buffers
to keep in the retransmission queue expressed as time (ms)
instead of buffer count (which is the max_size_buffers property).
The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.
The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.
RTX is SSRC-multiplexed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.
The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
and request-rtcp-decoder). The user will be able to provide encoders
or decoders dynamically. The encoders must follow the srtpenc API and
the decoders the srtpdec API. Having separate signals for RTP and RTCP
allows the user to use different encoders/decoders or provide the same
one (e.g. that would be the case for srtpenc).
Also, rtpbin now allows application/x-srtp in its pads.
https://bugzilla.gnome.org/show_bug.cgi?id=719938
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.
Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
Make it clear what should be handled purely by mss mode:
1) Expose the streams on the first moof as there are no moov atoms
2) Properly cleanup streams on flushes
Add a note about the meaning of upstream_newsegment and mss_mode
for future reference.
Make all other special fragment handling shared for both dash
and mss streams.
In a fragmented scenario, qtdemux is operating in push mode
and it gets a fragmented buffer. While processing its data
downstream gets unlinked (or a input-selector changes its
active pad and returns not-linked). Qtdemux stops processing
this fragment and returns not-linked upstream, leaving the
remaining data in its adapter.
When it gets an EOS it should make sure that all the data it
had received is pushed before pushing EOS.
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.
Fixes a regression introduced by commit 57c27ec3
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.
This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
Some buffers can have multiple moov atoms inside and the strategy
of using the gst_adapter_prev_pts timestamp to get the base timestamp
for the media of the fragment would fail as it would reuse the same
base timestamp for all moofs in the buffer instead of accumulating
the durations for all of them.
Heres a better explanation of the issue:
qtdemux receives a buffer where PTS(buf) = X
buf -> moofA | moofB | moofC
The problem was that PTS(buf) was used as the base timestamp for
all 3 moofs, causing all buffers to be X based. In this case we want
only moofA to be X based as it is what the PTS on buf means, and the
other moofB and moofC just use the accumulated timestamp from the
previous moofs durations.
To solve this, this patch uses gst_adapter_prev_pts distance
result, this allows qtdemux to calculate if it should use the
resulting pts or just accumulate the samples as it can identify
if the moofs belong to the same upstream buffer or not.
https://bugzilla.gnome.org/show_bug.cgi?id=719783
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.
This reverts commit 26040ee38c
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
In SmoothStreaming fragmented scenario, the timestamps are calculated
starting from the fragment buffer timestamp. When there is a not-linked
return from downstream, qtdemux will return upstream and will keep the
non-pushed data into its adapter.
On a new fragment buffer pushed to qtdemux, the new buffer timestamp
would overwrite the previous one that should be used on the still
to be pushed buffers. Because of this, this patch will also
update the fragment_start timestamp from the adapter last pts
to make sure the moof and timestamps are in sync and will result
in correct timestamps for all fragments.
In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
could read the moov again after the mdat because it was considering the
media as a fragmented one.
To avoid this loop this patch makes it store
the last processed moov_offset to avoid parsing it again.
And it also checks if there are any samples to play before
resturning to the mdat, so that it knows there is new data to be played.
https://bugzilla.gnome.org/show_bug.cgi?id=691570
When parsing a trak only free streams on failures if those streams
were created locally. They could have been created from a previous
fragment, in this case we they have valid info from the other fragment.
Including pads.
https://bugzilla.gnome.org/show_bug.cgi?id=691570
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
As for text subtitles and as suggested in #712643, throw
away the 2 byte terminator packets that some encoders insert.
This will make things better when remuxing and causes generation
of gap events.
Otherwise there were race conditions where we would send tags
on a flushing srcpad.
We have a test for that in GES, but this should be tested
systematically with harness in the future as I believe it
is useful for exactly that kind of cases.
https://bugzilla.gnome.org/show_bug.cgi?id=708165
Clean up the handling of mp4s streams. Use the generic esds
descriptor function to extract the palette, instead of hard coding
a wrong magic offset.
Add some more size safety checks when parsing ES descriptors, and
replace magic numbers with the descriptive constants that are already
defined.
Enhance dump output for stsd atoms.
Streams from both bug 712643 and historic bug 568278 now both work
correctly.
Fixes: #712643
