Buffer is added to the internal cache, and pushed only when accumulated
buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
is not freed. Freeing the cache when the state changes from PAUSED to READY.
https://bugzilla.gnome.org/show_bug.cgi?id=754212
By not doing this, the muxer is not effectively a rtpmuxer, rather a
funnel, since it should be a single stream that exists the muxer.
If not specified, take the first ssrc seen on a sinkpad, allowing upstream
to decide ssrc in "passthrough" with only one sinkpad.
Also, let downstream ssrc overrule internal configured one
We hence has the following order for determining the ssrc used by
rtpmux:
0. Suggestion from GstRTPCollision event
1. Downstream caps
2. ssrc-Property
3. (First) upstream caps containing ssrc
4. Randomly generated
https://bugzilla.gnome.org/show_bug.cgi?id=752694
If seeking targets an empty segment skip it as there is no media
offset to get from it. Instead look for the next one.
This doesn't make seeking in push-mode work if you seek to an
empty segment but at least won't get you to wrong offsets.
https://bugzilla.gnome.org/show_bug.cgi?id=753484
mux_start_time refers to the running_time of the buffer
that goes first in the output file. Normally this time is
0, so this variable is initialized to 0 during the state
change to PAUSED.
However, when dealing with dynamic pipelines and starting
a recording while the pipeline has already run for a while,
the running_time of the first buffer is > 0 and this causes
a problem with detecting the end of the first file(s) when
splitting by duration, because the code will later compare
the threshold_time with (last buffer running_time - mux_start_time)
and will get it wrong until mux_start_time advances enough
to make this difference < threshold_time, creating empty files
in the meantime.
https://bugzilla.gnome.org/show_bug.cgi?id=753624
During reverse playback, the media should stop playing at segment.start
This does not happen, and avidemux continues to process data even when
current timestamp is less that segment.start.
https://bugzilla.gnome.org/show_bug.cgi?id=755094
Avoid using default accept-caps handler that will query downstream
and is more expensive. Just check if the caps is compatible with
the template and check if the channels are the same.
Caps from the pad template are being leaked. In any case it is
from a static pad template and will 'leak' in the end, just doing
the cleanup for the good practice.
On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
"Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
'channels' are pointing to uninitialized values can end on setting random values as
rate and channels on src caps.
https://bugzilla.gnome.org/show_bug.cgi?id=755611
Otherwise we end up considering the values did not change and we wrongly
work with the old video format (which will lead to wrong
behaviour/segfaults).
https://bugzilla.gnome.org/show_bug.cgi?id=755621
eceb2ccc73 broke segment seeks by always
accumulating segments manually when activating a segment. This is only
needed when handling edit lists, not when activating a segment because of a
seek. Do the accumulation when switching edit list segments instead.
This fixes segment seeks again, while keeping edit lists playback working.
https://bugzilla.gnome.org/show_bug.cgi?id=755471
* use g_list_free_full(), don't iterate elements maually when freeing
* call gst_rtp_*_pay_clear_packet(), don't duplicate its code
* use gst_buffer_unref() to clarify that it is buffers being released,
instead of refering directly to gst_mini_object_unref()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
It's normal when dropping into the middle of a stream to
not always have the config available immediately, so skip LOAS
until a valid config is seen without either setting invalid
caps or erroring out.
https://bugzilla.gnome.org/show_bug.cgi?id=751386
flac contains the sample offset in the frame header, so after a seek
without index flacparse will know the exact position we landed on and
timestamp buffers accordingly. It only set the pts though, which means
the baseparse-set dts which was set to the seek position prevails, and
since the seek was based on an estimate, there's likely a discrepancy
between where we wanted to land and where we did land, so from here on
that dts/pts difference will be maintained, with dts possibly multiple
seconds ahead of pts, which is just wrong. The easiest way to fix this
is to just set both pts and dts based on the sample offset, but perhaps
parsed audio should just not have dts set at all.
https://bugzilla.gnome.org/show_bug.cgi?id=752106
One-line removal of tags_written++
This should fix rtmp output to crtmpserver, and hopefully
noone is expecting that the element count includes the end
element, as different bits of documentation say different
things about whether it should or not.
https://bugzilla.gnome.org/show_bug.cgi?id=661624
Apparently the Microsoft Azure RTMP server requires that the
videodatarate and audiodatarate metadata be provided, so
set those, even if it's to 0. Use the actual input bitrate
tags if available.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.
Fix a memory leak handling Mikey data.
When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
In gst_smpte_collected(), check upfront if input formats are same
or not. This avoids allocation of in1 and in2 buffers and
subsequent memory leak when input formats do not match.
https://bugzilla.gnome.org/show_bug.cgi?id=754153
When we haven't started yet, set the start_index when we set the index property,
so that we start at the right index position after the initial seek. The index
property was never really meant to be for writing, but it used to work, so let's
support it for backwards compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=739472
Commit 7d7e54ce68 added support for
DASH common encryption, however commit
bb336840c0 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.
The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.
This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"
https://bugzilla.gnome.org/show_bug.cgi?id=750101
Use constantDuration to calculate the timestamp of non-first AU in the
RTP packet.
If constantDuration is not present in the MIME parameters, its value
must be calculated based on the timing information from two consecutive
RTP packets with AU-Index equal to 0.
https://bugzilla.gnome.org/show_bug.cgi?id=747881
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
This commit adds support for ISOBMFF Common Encryption (cenc), as
defined in ISO/IEC 23001-7. It uses a GstProtection event to
pass the contents of PSSH boxes to downstream decryptor elements
and attached GstProtectionMeta to each sample.
https://bugzilla.gnome.org/show_bug.cgi?id=705991
GstRTSPMedia uses this classification to detect the real payloader
inside a dynpay bin and asserts if it doesn't find it, therefore
it is required
https://bugzilla.gnome.org/show_bug.cgi?id=753325
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.
Also use the PT constants from the rtp lib when possible
https://bugzilla.gnome.org/show_bug.cgi?id=747965
We need a proper caps event from upstream with the full RTP caps as we can't
create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
a filesrc or any other element that supports pull mode.
https://bugzilla.gnome.org/show_bug.cgi?id=753066
h264parse does the same, let's keep the behaviour consistent. As we now
include the codec_data inside the stream too here, this causes less caps
renegotiation.
The spec says:
When a picture parameter set NAL unit with a particular value of
pic_parameter_set_id is received, its content replaces the content of the
previous picture parameter set NAL unit, in decoding order, with the same
value of pic_parameter_set_id (when a previous picture parameter set NAL unit
with the same value of pic_parameter_set_id was present in the bitstream).
If the GOP is completed, pads have to start gathering for the
next one but it is possible that the the state might go to
COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
thread has a chance to wake up and proceed, leaving it trapped in
the check_completed_gop loop and deadlocking the other threads
waiting for it to advance.
To solve it, this patch also checks that tha input running time
hasn't changed to prevent this scenario.
h264parse does the same and this fixes decoding of some streams with 32 SPS
(or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
This looks like a mistake in the part of the spec about the codec_data.
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Don't hold the main splitmux part lock over
the parent state change function, as it prevents
posting error messages that happen. Since the purpose
is to prevent typefinding from proceeding, use a
separate mutex just for that.
Need to check that the number of bytes we want to copy from the adapter
actually is available and handle the error case gracefully. This error
may happen if malformed packets are received and we don't have a
complete frame.
https://bugzilla.gnome.org/show_bug.cgi?id=752663
The subtitle buffer we push out should not include a NUL terminator
as part of the data, we just add such a terminator for safety, but
it should not be included in the buffer size.
A NUL terminator is not valid UTF-8, so checks will fail if it's
included in the size, and the NUL will be replaced by the fallback
character specified when converting, i.e. '*'.
https://bugzilla.gnome.org/show_bug.cgi?id=752421
In certain applications, splitting into files named after a base
location template and an incremental sequence number is not enough.
This signal gives more fine-grained control to the application to
decide how to name the files.
https://bugzilla.gnome.org/show_bug.cgi?id=750106
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the mapping the base class has already done.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the map the base class has already done.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
we would produce wrong DTS. As now the estimated DTS is based on the clock,
don't store it in the jitterbuffer items as it would otherwise be used in the
skew calculations and would influence the results. We only really need the DTS
for timer calculations.
https://bugzilla.gnome.org/show_bug.cgi?id=749536
When a new time segment is received upstream is going to restart
with a new atom. Make the neededbytes and todrop variables
reflect that to avoid waiting too much or dropping the
initial bytes that contain the header.
The adapter might have data remaining from the previous segment,
push it all before clearing the adapter and starting a new segment.
It can accumulate data if it had pushed and got not-linked, returning
immediately without processing all the data. Before starting a new
segment this data should be handled.
The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.
The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)
Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.
https://bugzilla.gnome.org/show_bug.cgi?id=738363
This reverts commit 05bd708fc5.
The reverted patch is wrong and introduces a regression because there
may still be time to receive some of the packets included in the gap
if they are reordered.
Avoids accumulating all samples from a fragmented stream that could
lead to a 'index-too-big' error once it goes over 50MB of data. It
could reach that before 2h of playback so it doesn't take that long.
As upstream elements are providing data in time format they should
be the ones that have more information about the full media index
and should be able to seek if possible.
upstream_newsegment isn't really clear on what it means, it is set
to TRUE when the upstream element sends a segment in TIME format, so
rename it to be more clear about it.
It is important to know this because it means that upstream has
a notion of time and qtdemux is likely being driven by an upstream
element that is reading from a higher level abstraction than a file,
such as a DASH, MSS or DLNA element.
In fragmented streaming, multiple moov/moof will be parsed and their
previously stored samples array might leak when new values are parsed.
The parse_trak and callees won't free the previously stored values
before parsing the new ones.
In step-by-step, this is what happens:
1) initial moov is parsed, traks as well, streams are created. The
trak doesn't contain samples because they are in the moof's trun
boxes. n_samples is set to 0 while parsing the trak and the samples
array is still NULL.
2) moofs are parsed, and their trun boxes will increase n_samples and
create/extend the samples array
3) At some point a new moov might be sent (bitrate switching, for example)
and parsing the trak will overwrite n_samples with the values from
this trak. If the n_samples is set to 0 qtdemux will assume that
the samples array is NULL and will leak it when a new one is
created for the subsequent moofs.
This patch makes qtdemux properly free previous sample data before
creating new ones and adds an assert to catch future occurrences of
this issue when the code changes.
This reverts commit d46631c5c7.
pad only handle EOS events but not EOS flow, and will push the buffer again
resulting in an assertion error. So we should not handle the buffer
and return EOS flow.
goom_core.c: In function 'goom_update':
goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
^
goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
^
https://bugzilla.gnome.org/show_bug.cgi?id=752053
endpos variable does not correctly understand in the
4.6.3 GCC version. So compile error appears when we do
compile rtph261pay using jhbuild.
This patch is fixed the compile error in 4.6.3 GCC version.
https://bugzilla.gnome.org/show_bug.cgi?id=751985
Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
R: Bit reserved for future use. MUST be set to zero and MUST be
ignored by the receiver.
https://bugzilla.gnome.org/show_bug.cgi?id=751929
gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
GObjectClass *gobject_class;
Implementation according to RFC 4587.
Payloader create fragments on MB boundaries in order to match MTU size
the best it can. Some decoders/depayloaders in the wild are very strict
about receiving a continuous bit-stream (e.g. no no-op bits between
frames), so the payloader will shift the compressed bit-stream of a
frame to align with the last significant bit of the previous frame.
Depayloader does not try to be fancy in case of packet loss. It simply
drops all packets for a frame if there is a loss, keeping it simple.
https://bugzilla.gnome.org/show_bug.cgi?id=751886
If we have a clock, update "now" now with the very latest running time we have.
If timers are unscheduled below we otherwise wouldn't update now (it's only updated
when timers expire), and also for the very first loop iteration now would otherwise
always be 0.
Also the time is used for the timeout functions, e.g. to calculate any times
for the next timeouts and we would otherwise pass too old times there.
https://bugzilla.gnome.org/show_bug.cgi?id=751636
We always pushed one buffer into the adapter, then handled exactly that one
buffer and flushed it from the adapter. Now also don't memcpy() the actual
payload but just attach the input buffer's data to the output buffer.
This code still needs some serious refactoring/rewriting.
This reverts commit 0c21cd7177.
If we have multiple immediate timers, we want to first handle the one with the
lowest sequence number... which would be broken now.
Instead of this we should just use a GSequence for the timers, and have them
sorted first by timestamp, and for equal timestamps by sequence number. Then
we would always only have to take the very first timer from the list and never
have to look at any others.
Most files don't contain the values for transposing the coordinates
back to the positive quadrant so qtdemux was ignoring the rotation
tag. To be able to properly handle those files qtdemux will also ignore
the transposing values to only detect the rotation using the values
abde from the transformation matrix:
[a b c]
[d e f]
[g h i]
https://bugzilla.gnome.org/show_bug.cgi?id=738681
The media start has nothing to do with the shift we have applied
but with the value of the first PTS. This is defined as:
Dt(0) = 0
Ct(0) = Dt(0) + CTTS(0)
So the media start is always the first CTTS.
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Allows playing edts editted files with proper synchronization of
streams. This patch fixes the regression introduced by
bf95f93c01 that was added to fix
segment seeks handling.
Having the accumulated_base separated from the main segment.base
allows handling both segment seeks and edts editted files.
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Buffers need not to start at running-time 0 so the last_dts needs
to be the value of the first buffer's dts as it is used to compute
the duration of the buffers. If it was left at 0 the first buffer
would have a larger duration when it shouldn't
https://bugzilla.gnome.org/show_bug.cgi?id=751361
Fix 2 startup races when things happen too quickly, and 1
at shutdown by holding a ref to the pads in use until the
loop functions exit.
Handle errors activating file parts and publish them on
the bus.
https://bugzilla.gnome.org/show_bug.cgi?id=750747
Sometimes, extra async-start/done from the internal sink
while the element is still starting up can cause splitmuxsink
to stall in PAUSED state when it has been set to PLAYING
by the app. Drop the child's async-start/done messages while
switching, so they don't cause state changes at the
splitmuxsink level.
https://bugzilla.gnome.org/show_bug.cgi?id=750747
Move the multiview caps calculations to the configure_stream()
function, so the rest of the video info is available, and
use the gst_video_multiview_guess_half_aspect() function to
determine if the half-aspect flag should be set on frame-packed
video.
The cslg atom provide information about the DTS shift. This is
needed in recent version of ctts atom where the offset can be
negative. When cslg is missing, we parse the CTTS table as proposed
in the spec to calculate these values.
In this implementation, we only need to know the shift. As GStreamer
cannot transport negative timestamps, we shift the timestamps forward
using that value and adapt the segment to compensate. This patch also
removes bogus offset of ctts_soffset, this offset shall be included
in the edit list.
https://bugzilla.gnome.org/show_bug.cgi?id=751103
We shift DTS forward to avoid negative timestamps which cannot be
represented with version 0 of the CTTS table. To stick with that
version (backward compatibility), the spec recommend using an
edit list entry to move back the presentation time to where it
should be.
https://bugzilla.gnome.org/show_bug.cgi?id=751242
No need to check for context availability while freeing. We are inside
inside a code block with a condition that dereferences context.
if (context->type == 0 ...
https://bugzilla.gnome.org/show_bug.cgi?id=751306
If update_receiver_stats() fails, we can't really do anything with this buffer
anymore and have to drop it. This happens if there's a big seqnum
discontinuity for example.
https://bugzilla.gnome.org/show_bug.cgi?id=751311
The new property allows to select the time source that should be used for the
NTP time in RTCP packets. By default it will continue to calculate the NTP
timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
clock time. The latter is especially useful for synchronizing multiple
receivers if all of them share the same clock.
If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
setting and continue to use the running time plus 70 years. This is only kept
for backwards compatibility.
In presence of a CTTS, the segment start/stop must be offset so
the segment start/stop include the PTS. This is needed since the
PTS cannot be negative in this format. This fixes issues where the
running time of the first buffer isn't at the start.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
This is done by using new feature of the CollectPad clip function
which sets the DTS as a gint64 in the collected data. It also simplify
the code a bit.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
When deciding whether it's time to switch to a new file, take into
account data that's been released for pushing, but hasn't yet
been pushed - because downstream is slow or the threads haven't been
scheduled.
Fixes a race in the unit test and probably in practice - sometimes
failing to switch when it should for an extra GOP or two.
Also fix a problem in splitmuxsrc where playback sometimes
stalls at startup if types are found too quickly.
https://bugzilla.gnome.org/show_bug.cgi?id=750747
Remove a custom specialized version of gst_buffer_new_wrapped by
using gst_buffer_new_wrapped_full inside a macro to simplify
parameters and give it a more meaningful name.
It is only used to create temporary buffers to have its data copied.
Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.
One extra feature that was added was the ability to add extension
atoms after set_caps as the AC-3 extension atom needs some data
that has to be extracted from the stream itself and is not
present on caps.
Implement support for the packed video formats WebM
uses, not all the values that Matroska might use.
In practice, it's really hard to find any samples in the
wild of any.
Supported in both the muxer and demuxer.
The MPEG-A format provides an extension to the ISO base media
file format to store stereoscopic content encoded with different
codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
atom declares the presence and storage method for the video.
Stereo video information for MPEG-A can also be supplied through
the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
is not implemented in this patch.
Also missing is support for stereo video encoded as separate video tracks
for now.
Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>
https://bugzilla.gnome.org/show_bug.cgi?id=611157
When performing seek, segment->start is being updated with desired_offset,
but in case of reverse playback segment->start should be 0 and
segment->stop should be updated with desired offset.
https://bugzilla.gnome.org/show_bug.cgi?id=750675
