Updating caps results in downstream elements potentially reconfiguring themselves
(such as decoders). If we do this in the middle of keyframes, we would result
in those elements being reconfigured and handling garbage until the next keyframe.
Instead of this only send (potentially) new codec_data when we have *both* SPS and
PPS.
https://bugzilla.gnome.org/show_bug.cgi?id=705333
If ever we lose sync, we were just checking for the next 0x47 marker ...
which might actually happen within a mpeg-ts packet.
Instead check for 3 repeating 0x47 at the expected packet size interval,
which the same logic we use when we initially look for the packet size.
We were only resetting the first 512 values of the lookup table instead
of the whole 8192.
This resulted in any PCR PID over 0x0200 ... ending up taking the first PCR
table around :(
ATSC ac3 streams are always guaranteed to be AC3 if EAC3 descriptor
is not present
If stream registration id is 'AC-3' then it's also guaranteed to be AC3.
Finally if AC3 descriptor is present it's guaranteed to be AC3.
Only silences a warning, but still.
We know we will not overflow 64 bits, therefore just use direct
multiplication/division instead of the scale method (trims usage from
50 instruction calls to 2/3).
Helps with debugging issues. And also remove unused variable (opcr)
This will also allow us in the future to properly detect:
* random-access location (to enable keyframe observation and
potentially seeking
* discont location (to properly handle resets)
* splice location (to properly handle new stream changes)
If a buffer was entirely clipped out (ie, it's out of the segment
entirely), we'll end up with a NULL buffer, which we don't want
to process/dereference.
The new seek handling re-creates the segment time information once it
has enough information after a seek.
The problem was that we'd completely ignore the requested rate. So store
that and use it in the newly created segment.
https://bugzilla.gnome.org/show_bug.cgi?id=694369
Make videotestsrc ! interlace ! $anything work again. Problem
was that upstream filter caps were passed which contained
interlace-mode=progressive, which doesn't intersect too well
with interlace's source pad template caps, leading to
not-negotiated errors.
The program_number attribute was overloaded, trying to indicate both
the currently playing program, and the program requested via the
"program-number" property. The end result was that setting the
property didn't work (see #690934).
I added a new requested_program_number field rather than reviving the
current_program_number field because it seemed this would result in
fewer changes overall and be less confusing. It breaks symmetry with
the "program-number" property, but it retains parallels with the likes
of program->program_number.
Because gst_ts_demux_reset is called after the properties have been
parsed, requested_program_number is initialised in gst_ts_demux_init.
Whether this is exactly the right place, I don't know.
Setting the program-number property does not affect which program
is actually being demuxed.
Moving the initialization of the program_number from
gst_ts_demux_reset to gst_ts_demux_init seems to fix this issue.
https://bugzilla.gnome.org/show_bug.cgi?id=690934
* Avoids handling twice the same seek (can happen with playbin and files
with subtitles)
* Set the sequence number of the segment event to the sequence number of
the seek event that generated it (-1 for the initial one).
The seeking start time is approximated from the seek offset in bytes
using the accumulated PCR observations, so on a VBR stream there might
be a big difference between the actual PCR and the estimated one after
the seek. This might result in a long wait to skip all out of segments
packets.
Instead we just recalculate the new segment to start at the first PTS
after the seek, so that playback starts immediatly.
The caps should always represent what the user is supposed to see.
So if there is a sequence_display_extension associated with the
stream then use the display_horizontal_size/display_vertical_size
to update the src caps (if they are less than the values provided
by sequence header).
https://bugzilla.gnome.org/show_bug.cgi?id=704009
This is actually a workaround (we'll be skipping the upcoming section)
This will only happen for sections where the beginning is located within
the last 8 bytes of a packet (which is the minimum we need to properly
identify any section beginning).
Later we should figure out a way to store those bytes and mark that
some analysis needs to happen. The probability of this happening is
too low for me to care right now and do that fix. There is a good chance
that section will eventually be repeated and won't end up on such border.
* packet.origts is no longer used since the PCR refactoring done ages ago
* known_packet_size is a duplicate of packet_size != 0
* caps was never used outside of the packetizer
Restore the original h264parser behaviour to report cropped dimensions
in size caps.
https://bugzilla.gnome.org/show_bug.cgi?id=694068
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
We had two issues with the previous code:
1) We were badly handling PUSI-flagged packets. We were discarding the
initial data (if pointer != 0) whereas we should have been accumulating
it with the previous data (if there was a continuity of course).
=> First series of information loss
2) We were not checking whether there were more sections after the end
of one (i.e. when the following byte was not a stuff byte).
This fixes those two issues.
Fixes#677443https://bugzilla.gnome.org/show_bug.cgi?id=677443
Until now we simply ignored those streams (since we couldn't do anything
with it anyway). Now that we have the mpegts library and we offload the
section handling to the application side we can properly identify and
extract them.
By default it is disabled for tsparse and enabled for tsdemux, but there is
a property to change that.
This should open the way to properly handle all private section streams,
including:
* DSM-CC
* MHEG
* Carousel data
* Metadata streams (though I haven't seen any of those in the wild)
* ... And all other specs/protocols making use of those
Partially fixes#560631
Migrate the code to use the new parser API based on GstMpegVideoPacket.
Also try to optimize gst_mpegv_parse_process_config() by using more of
GstMpegVideoPacket and determining the extension_start_code_identifier
prior to calling the parser function for that extension packet.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Since we now send all sections to the packetizer, we no longer need to do
anymore in-depth checks for the validity of a section.
The choice boils down to:
1) Is it from a known PES pid ? If so pass it on (which might be just pushing
downstream in the case of tsparse, or accumulating PES data for tsdemux)
2) Is it from a known SI pid ? If so pass it to the section packetizer
We still have some other stream types which haven't been ported, but
we will do so once we have defined the enums in the mpegts library.
Also add some FIXMEs regarding items discovered during analysis
* Only mpeg-ts section packetization remains.
* Improve code to detect duplicated sections as early as possible
* Add FIXME for various issues that need fixing (but are not regressions)
https://bugzilla.gnome.org/show_bug.cgi?id=702724
We use add_stream(stream_type:-1) to ensure a programs' PCR Stream is
also taken into account. For most programs this will re-use an
existing ES stream.
So only warn that we are re-adding a stream if it was already present
AND it is not to ensure the PCR stream is taken into account.
Only create subtables when needed. It was previously creating one every
single time ... to check if one was present.
And speed up code to detect whether a subtable was already present or not.
Overall makes section pushing 2 times faster.
In some cases (NIT on highly-populated DVB-C operator for example), there
will be more than one section emitted for the same subtable and version
number.
In order not to lose those updates for the same version number, we checked
against the CRC of the previous section we parsed.
The problem is that, while it made sure we didn't lose any information, it
also meant that if the same section came back (same version, same CRC) later
on we would re-process it, re-parse it and re-emit it.
This version improves on that by keeping a list of previously observed CRC
for identical PID/subtable/version-number and will only process sections if
they really were never seen in the past (as opposed to just before).
On a 30s clip, this brings down the number of NIT section parsing from 4541
down to 663.
https://bugzilla.gnome.org/show_bug.cgi?id=614479
First send stream-start, then caps, then segment.
The segment we push is from upstream in push-mode. If we work in pull-mode
then we initialize the base segment to BYTES.
https://bugzilla.gnome.org/show_bug.cgi?id=702422
Sync byte scan is incorrect for M2TS streams because the timestamp 4
bytes were not included in the flush size. This can result in an
infinite loop.
Rework the scan code to be clearer and work in all cases.
By adding the video-source-filter during construction time, rather then
patching it in later (*), we can greatly reduce the amount of caps involved
in negotation, speeding up pipeline creation.
I wrote this while working on speeding up the startup of cheese. My cheese
has been modified to add a capsfilter, filtering for only the configured
resolution, with that cheese patch + this patch, the pipeline creation time
goes from aprox 1.1 seconds to aprox 350ms. This is with a Logitech 9000
pro camera, which supports lots of different resolutions at many different
framerates per resolution, causing a caps "explosion" if not filtered.
*) Note the code for this is left in, as it is still necessary if the
video-source-filter is changed between a stop + re-start.
https://bugzilla.gnome.org/show_bug.cgi?id=701953
check_and_replace_src() was setting self->app_vid_src to NULL, which
means that an app setting the video-source property, and then starting,
stopping and re-starting the pipeline (ie to make changes to the
video-source-filter property) would after the restart no longer have
a video-source.
This patch fixes this by making gst_camerabin_setup_default_element return a
ref to the passed in user_element, rather then returning the user_element as
is, so that that ref can be passed on to the bin, and the app_vid_src ref
stays valid.
https://bugzilla.gnome.org/show_bug.cgi?id=701915
Current fallback to lost_sync seems to impede a delay to restore
sync. Let the parser parse and skip the private stream.
Here it contains the digital camera brand (in 2010 bytes)
and is repeated twice.
https://bugzilla.gnome.org/show_bug.cgi?id=697283
descriptors are stored as a GValueArray of GString. The downside is
that there is no way to "pass" ownership of a GValue to a GValueArray
which previously resulted in expensive copy/free of the (already expensive)
GString.
Here we estimate first the size of the GValueArray, then create it,
then directly use the GValue of that array.
Speeds up total SI parsing by ~30%
If rfb_decoder_new() allocates the decoder sructure, rfb_decoder_free()
should free the structure. We should not free the decoder when an
error occurs during connection - it holds lots of configuration/state
and will be freed later in finalize.
prepare_func will allocate a new buffer to replace the original
one. Instead of using gst_buffer_replace (which causes an extra
refcount increment on the new buffer), we just unref the original
buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=699786
Don't send any source caps yet if we're still in
drop-buffers-until-we-get-a-sequence-header mode.
Fixes transmuxing of many MPEG-TS/PS streams into
formats which require things like width, height or
codec_data on the input caps.
Also fixes issues when using playbin with decoder
sinks that want width/height etc.
https://bugzilla.gnome.org/show_bug.cgi?id=695879
Since there is a conflict between the DCII stream type and BluRay
stream types, moved the processing of BluRay-specific stream types
to the beginning of the function. Only if a BluRay stream type
IS NOT found do we proceed to check the rest of the stream type
identifiers
Previous code was also "sort-of" handling a similar conflict between
BluRay AC3 audio and standard AC3 audio. Moved the special case BluRay
AC3 handling in the main switch statement to the new BluRay-specific
switch.
https://bugzilla.gnome.org/show_bug.cgi?id=697892
The src element may not include information about whether
the data is parsed or not. Hence do not require parsed=false.
Fixes multipartdemux ! jpegparse ! ...
https://bugzilla.gnome.org/show_bug.cgi?id=697884
And if we detect a discontinuity there (like... when losing packets
or having MPEGTS over raw UDP with out-of-order packets) we just
drop the corresponding packet.
A future version could try to implement a re-ordering algorithm based
on that, similar to what rtpjitterbuffer does.
Backport fix for crashes and invalid writes in totem from libvisual
in -base, to minimise differences to version in -base and to make
sure the bug doesn't sneak back in later when the base class is
made public.
The shader code looks like it makes assumptions that are not
necessarily always true, even if they're true for now for the
existing elements, namly that pixel stride is 4, for example.
See https://bugzilla.gnome.org/show_bug.cgi?id=683527
gst_query_set_nth_allocation_pool() requires there to be a pool in the
query already. This is not always the case when we get the query from
upstream. Use gst_query_add_allocation_pool() instead in such case.
https://bugzilla.gnome.org/show_bug.cgi?id=681719
When converting the incoming segment from byte to time format,
don't just convert the start/stop/time values, but also change
the segment format to TIME.
https://bugzilla.gnome.org/show_bug.cgi?id=696361
Rework things a bit so that we can run over the midi events and fire callbacks
for each of them. We can then use that for calculating the duration and also for
doing playback.
Only parse as many tracks as specified in the header.
Fix default tempo;
Send MIDI tick events every 10ms
This filter converts interlaced content that was originally
telecine'd from 24 fps (or similar) progressive content. It works
approximately like videorate, but with awareness of interlacing.
It merges neighboring fields in the input interlaced stream with
the goal of minimizing combing artifacts, while keeping the output
framerate constant. If it cannot avoid combing artifacts, it will
reconstruct the image from a single field. Note that this filter
does not autodetect the framerate, but will automatically fixate
at 24 fps.
A non-live element is supposed to continue streaming in the paused state so
don't stop the tasks when going to paused.
We also always want to start the update task after we prerolled enough data,
not only in the playing state.
