Create output caps from input caps, so we maintain any fields we
might get on the input caps, such as codec_data or rate and channels.
Set channels and rate on the output caps if we don't have input caps
or they don't contain such fields. We do this partly because we can,
but also because some muxers need this information. Tagreadbin will
also be happy about this.
Sending the flush-start event forward before taking the stream lock actually
works, in contrast to deadlocking in downstream preroll_wait (hunk 1).
After that we get the chain function being stuck in a busy loop. This is fixed
by updating the minimum frame size inside the synchronization loop because the
subclass asks for more data in this way (hunk 2).
Finally, this leads to a very probable crash because the subclass can find a
valid frame with a size greater than the currently available data in the
adapter. This makes the subsequent gst_adapter_take_buffer call return NULL,
which is not expected (hunk 3).
The problem is that after a discont, set_min_frame_size(1024) is called when
detect_stream returns FALSE. However, detect_stream calls check_adts_frame
which sets the frame size on its own to something larger than 1024. This is the
same situation as in the beginning, so the base class ends up calling
check_valid_frame in an endless loop.
Baseparse internaly breaks the semantics of a _chain function by calling it with
buffer==NULL. The reson I belived it was okay to remove it was that there is
also an unchecked access to buffer later in _chain. Actually that code is wrong,
as it most probably wants to set discont on the outgoing buffer.
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.
Fixes#646800
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.
In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.
See bug #646397.
This option allows the videomixer2 element to output a valid alpha
channel when the inputs contain a valid alpha channel. This allows
mixing to occur in multiple stages serially.
The following pipeline shows an example of such a pipeline:
gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=\(fourcc\)AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink videotestsrc ! video/x-raw-yuv,format=\(fourcc\)AYUV ! mix2.
The first videotestsrc in this pipeline creates a moving ball on a
transparent background. It is then passed to the first videomixer2.
Previously, this videomixer2 would have forced the alpha channel to
1.0 and given a background of checker, black, or white to the
stream. With this patch, however, you can now specify the background
as transparent, and the alpha channel of the input will be
preserved. This allows for further mixing downstream, as is shown in
the above pipeline where the a second videomixer2 is used to mix in a
background of an smpte videotestsrc. So the result is a ball hovering
over the smpte test source. This could, of course, have been
accomplished with a single mixer element, but staged mixing is useful
when it is not convenient to mix all video at once (e.g. a pipeline
where a foreground and background bin exist and are mixed at the final
output, but the foreground bin needs an internal mixer to create
transitions between clips).
Fixes bug #639994.
Previously the chain function was working sample frame based. In each cycle it
was checking if it is time to run a fft or if it is time to send a message.
Now we changed the data transform functions to work on a block of data and
calculate the max length until either {end-of-data, do-fft, do-msg}. This allows
us also to avoid the duplicated code for the single and multi-channel case (as
the transformers have the same signature now).
Even though we wrap around the accumulated second, we still need to add the
error in the same cycle. Increase the todo in the same conditional as afterwards
the accumulated error will be below one second.
AUTHOR only existed in an old version of the spec and ARTIST is
the new replacement for this. We are still reading both to still
be compatible with old files.
Fixes bug #644875.
Before it was possible that we run an extra fft when the time for sending a new
message is due. Only do this if we have not run the fft for the interval at all.
Don't check the format for each sample frame to read. We can make that decission
in _setup already. This is still not ideal as we call the function per frame.
Ideally we determine how many samples we can copy and have a loop in the input
reader. As an alternative we might also consider to use the fft variants for the
various formats and not convert to float for all cases - we would still need to
mix or deinterleave though.
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process. So, query position prior
to seek and use upon failed seek.
Add a boolean multi-channel property with a default of FALSE. When set to TRUE
the element won't mix all input channels to mono, but instead run a FFT on each
channel. In that case the result message would contain a 2 dimensional array
of channel x data for magnitude and phase.
API: GstSpectrum:multi-channel
https://bugzilla.gnome.org/show_bug.cgi?id=593482
Use a separate function to read a sample frame into a ringbuffer slot. In the
future we can use format-specific function pointer to avoid the reoccuring
format checks.
We now keep the fft data that is related to one channel in a separate structure
to prepare for multichannel support. We also refactor the code to operate more
often on the channel context.
When using gstrtpbin with ignore-pt=true, the free_stream function tries to
call gst_element_set_locked_state and gst_element_set_state on a stream->demux
which is NULL.
fixes#642412
Fix slightly confused tag handling in some places: make it clear when
we're taking ownership of a tag list and when not. For example,
gst_icydemux_tag_found() was taking ownership when the source pad
existed, but otherwise not (leak). Also, gst_event_parse_tag() does
not return a newly-allocated taglist, but a tag list that belongs to
the tag event, so don't give ownership of it away.
While we're at it, some minor clean-ups: don't re-invent g_strndup()
and simplify gst_icydemux_parse_and_send_tags() a bit, and don't
leak the tag list in case no valid tags where found.
https://bugzilla.gnome.org/show_bug.cgi?id=641330
* gst/qtdemux/qtdemux.c (gst_qtdemux_src_convert): Unref the qtdemux; we
weren't doing so before.
(gst_qtdemux_handle_src_event, gst_qtdemux_chain): Fix some error
cases which would leak a ref to the qtdemux.
Extract MusicBrainz tags added by MusicBrainz's Picard
tagger application. These tags (esp. the album id) are
helpful for rhythmbox et.al. to automatically downloads
cover art.
https://bugzilla.gnome.org/show_bug.cgi?id=642205
Images might have framerate=0/1 in the caps, which caused an
assertion on deinterlace. I don't know of interlaced image formats
but deinterlace might be hardcoded on some generic pipelines and
it shouldn't assert.
The fix was to set field_duration to 0 if the input has a framerate
with a 0 numerator.
This patch also adds checks for this situation on the unit tests.
https://bugzilla.gnome.org/show_bug.cgi?id=641400
Theora can only use the last frame (or the keyframe) as a reference, so in
practice. If we receive a buffer that references an unknown codebook, request
new headers. It probably means that headers were lost.
Functions that process the rtcp buffer could decide to keep a ref
on the buffer for further processing. So make the metadata writable
only after they are done.
In particular, this avoids missing the intended keyframe when first converting
from the frame's mov time to global segment time, and then back from global
time to mov time when activating the segment.
Make win32 build bot happy again, and nicefy output while we're at it.
qtdemux.c: In function 'qtdemux_parse_trun':
qtdemux.c:2162:3: error: format '%lu' expects type 'long unsigned int', but argument 9 has type 'guint32'
Check that the WAVEHEADER node is present instead of blindly using it.
If not present we won't be able to provide a more refined caps, but at
least we won't crash.
https://bugzilla.gnome.org/show_bug.cgi?id=640028
Old code was difficult to understand exactly how the neighboring
scan lines are calculated, and it appeared that some were off by
+2 or -2, depending on the field flag. Fixes#639321.
Set caps from the start so discoverer doesn't blow up on
seeing no negotiated caps between elements on preroll,
which might happen if no subtitle buffers have been
pushed yet at the time. See file from bug #603308.
The previous default, greedyh, takes 4 times as long as MPEG-2
video decoding, and is unlikely fast enough on any current CPU
to play 1080i video in real-time. greedyl isn't much faster.
linear was chosen over vfir, since the quality advantage of vfir
is minimal compared to the occasional visual artifacts and slower
processing.
Improve parsing of the samplerate.
Parse the framelen so that we can calculate timestamps.
When interpollate the incomming timestamp on outgoing buffers when there are
multiple subframes.
fixes#625825
It was an arbitrary limit from the start, meant as a basic sanity check,
so may just as well increase it a little. Would be good to provide
progress reporting while completing the block in any case..
https://bugzilla.gnome.org/show_bug.cgi?id=637060
Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise
floating point numbers, instead of ugly hacks that switch locale
before and after calling libc functions (which is not a good idea
in a multi-threaded application).
atof() converts strings according to the current locale, but the
framerate string will likely always use a dot as floating point
separator, so use g_ascii_strtod() instead (but also canonicalise
the string before, so we can handle both formats as input).
Include all possible stats of a source in the stats structure because we might
be interested in what happened in the past.
Document the stats property and the fields.
Using this in a demuxer will cause deadlocks if there's
a pad with a pending pad-block downstream, no matter if
there is a queue between the pad or not. Queues pass
bufferalloc downstream from the same thread and only
act as a thread boundary for events and buffers.
When the jitterbuffer contains -1 timestamps, make sure we still calculate the
buffer fill level by skipping the -1 buffers.
Try to be more resilient to weird input timestamps.
since we are using the clock for sync, we need to also provide a clock for good
measure. The reason is that even if downstream elements provide a clock, we
don't want to have that clock selected because it might not be running yet.
... thereby (partially) deprecating properties currently controlling whether
or not byte-stream output or NAL/AU alignment (though properties still determine
fallback if nothing specified in caps).
Fixes#606662.
Extra info can't hurt. Field names aren't necessarily consistent with
what's used elsewhere though (e.g. avidemux), but then neither are the
caps.
https://bugzilla.gnome.org/show_bug.cgi?id=623178
In particular, when streaming interleaved, this arranges for setting a new
timestamp on outgoing buffer so downstream can appropriate reset
to a change in (rtp)time.
Use 3 adapters, one to accumulate paketization units, another on to accumulate
tiles and a last one to accumulate the final frame.
Don't just blindly flush the adapter on DISCONT but only discard the current
packetization unit.
When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with
the new lenght.
In particular, accept unknown stream in track fragment, and only error out
if that raises problems later on with respect to offset tracking.
Fixes#620283.
The following keys will now be interpreted by navseek:
'f' means fast forward: the stream gets played at rate 2.0
'r' means rewind: the stream gets played at rate -2.0
'n' means normal: the stream gets played at rate 1.0
Fixes#631516.
On the one hand, it insufficiently checks whether it only updates a dummy
segment. On the other hand, only doing this at the time the last sampled is
prepared (and sent downstream) is too little too late.
That is, parse each moof in one pass (considering all contained streams'
metadata), and do so incrementally as needed for playback rather than
an initial complete scan of all moof (though all moov sample metadata
is fully parsed at startup).
... as some bogus files may indicate streams of 0 duration in moov,
while indicating the complete movie duration in mvhd (the latter should
be in mehd).
Avoid extra allocation in _parse_trun, add more checks for parsing errors,
add or adjust some debug statement, fix comments, sprinkle some branch
prediction.
The allocation of the samples can be placed out of the loop.
Makes the code clearer.
Also avoid relying on traf information as it is placed on the
end of the file and might not be acessible on push mode.
The fragmented mp4 format stores the tracks and samples information in the
'moof' boxes, which are appended before each fragment (fragment->'moof'+'mdat').
The 'mfra' box stores the offset of each 'moof' box and their presentation
time. The location of this box can be retrieved from the 'mfro' box, which is
located at the end of the file.
The 'mfra' box is parsed to get the offset of each 'moof' box and their
presentation time.
Each 'moof' box can contain information for one or more tracks inside
'tfhd' boxes. For each track in a 'moof', we have a 'trun' box, which
contains information of each sample (offset and duration) used to build
the samples table.
Based on patch by Marc-André Lureau <mlureau@flumotion.com>
https://bugzilla.gnome.org/show_bug.cgi?id=596321
Versions 0 and 1 of mvhd have different sizes of its values
(32bits/64bits). This patch makes it dump them correctly.
Also use the right node in the parameter and not the root node.
https://bugzilla.gnome.org/show_bug.cgi?id=596321
The DTS typefinder may return a lower probability for frames that start
at non-zero offsets and where there's no second frame sync in the first
buffer. It's fairly unlikely that we'll acidentally identify PCM data
as DTS, so we don't do additional checks for now.
https://bugzilla.gnome.org/show_bug.cgi?id=636234
When parsing the bitstream, look for SOP markers because we are allowed to split
packets on those marker boundaries.
Rework the parsing code a little so that we can pack multiple Packetization
units in one RTP packet.
When handling newsegment, flush out the buffer history in the
existing segment, not the new one. Fixes playback in some DVD
cases.
Partially fixes#633294
In a number of cases it is necessary to flush the field history by
performing 'degraded' deinterlacing - that is, using the user-chosen
method for as many fields as possible, then using vfir for as long as
there are >= 2 fields remaining in the history, then using linear for
the last field.
This should avoid losing fields being kept for history for example at
EOS.
This may address part of #633294
Only set the delta flag when all of the units in the packet are delta units.
Based on patch from Olivier Crête <olivier.crete@collabora.co.uk>
Fixes#632945
If caps weren't negotiated, goom should return not-negotiated
from its chain functions instead of using bps unitialized, which
leads to a division by 0
https://bugzilla.gnome.org/show_bug.cgi?id=633212
GST_ELEMENT_ERROR must not be called with the object lock held,
since it will call gst_object_get_parent() internally, which
takes the object lock as well.
Only send newsegment events with new positions downstream when actually
needed, instead of sending multiple newsegment events with new seek
positions in a row. Also set the discont flag on buffers after a
discontinuity.
Re-use the existing 'pos' field maintained by ebml writer to set
buffer offsets. This also makes sure that we set the right offsets
on buffers after a seek (e.g. when writing an index at the end).
Incomming buffer is only pushed on the adapter at the end of the
handle_buffer function. But duration/timestamp of this buffer is already
taken into account for the current data in the adapter. This leads to
wrong rtp timestamps and extra latency.
Both history_count and fields_required count from 1. As per the while loop
condition that follows this code, to perform the deinterlacing method, we need
history_count >= fields_required fields in the history. Therefore if we have
history_count < fields_required (not fields_required + 1), we need more fields.
This fixes the assumption that DecoderSpecificInfo must be 2 bytes long
for AAC files. The specification allows HE-AAC to be explicitly
signalled in a backward compatible way. This is done by means of an
additional information after the regular AAC header. It is expected that
decoders that can play AAC but not HE-AAC will parse the header normally
and ignore extended bits, much as they do for the HE-AAC specific payload
in the actual stream.
https://bugzilla.gnome.org/show_bug.cgi?id=612313
Implement a latency query and report how much latency we will add to the
stream.
Alse make some defaults for the default width/height/framerate
Fixes#631303
This uses gstpbutils to extract the profile and level from the video
object sequence and adds this to stream caps. This can be used as
metadata and for fine-grained decoder selection.
https://bugzilla.gnome.org/show_bug.cgi?id=616521
This exports the AAC profile and level in caps for use as metadata and
(eventually) for more fine-grained selection of decoders at
caps-negotiation time. (Doesn't work for HE-AAC yet though.)
https://bugzilla.gnome.org/show_bug.cgi?id=612313
Using _foreach_remove on the hashtable, while releasing the lock protecting
that table inside the callback is not a good idea. The hashtable might
then change (a source removed or added) while signals like on_timeout
are being sent.
This solution makes a copy of the table, performs the _foreach without
actually removing any sources, but marks them for removal on a second
iteration with the real list, but this time not letting go of the lock.
Fixes#630452
Holding internal locks while potentially calling out is a source
of deadlocks, and in this case the application might subscribe to the
pad-added signal.
Fixes#630449
If the source has been inactive for some time, we assume that it has
simply changed its transport source address. Hence, there is no true
third-party collision - only a simulated one.
Fixes#630447
For H264, there is an extra header containing the CTS, which is a timestamp
offset that should be applied to the PTS. Parse this value and use it to adjust
the pts.
Fixes#630088
Parses uuid atoms in push mode when they are found, they might
contain xmp tags.
Also does a minor refactoring to put the global tags posting
into a single function instead of repeating it in 3 different
places.
Fixes#629839
xmp packet is placed into a top-level uuid atom for
isom/mp4 variants.
This patch makes qtdemux parse all top-level atoms
in pull-mode before starting to push data, making
it able to find those tags.
https://bugzilla.gnome.org/show_bug.cgi?id=629839
That is, if parse error occurs in state requiring to move to next cluster,
and doing so to the expected next position of cluster fails, then scan for a
next cluster from present position and resume from there.
Fixes#620790.
If some bits out of place in block(group) parsing, forego and move to next.
Also skip large blocks in pull mode, but need to give up in push mode.
Fixes#626463.
Improves #620790.
Add an ntp-sync property that will sync the received streams to the server
NTP time. This requires synchronized NTP times between the sender and receivers,
like with ntpd.
Based on patch from Thijs Vermeir.
Fixes#627796
Instead of adding multiple client structures for the same host/port pair, use a
refcount.
Add a send-duplicates feature that allows you to disable sending multiple copies
of the same packet to the same host when it was added multiple times. The
send-duplicates property is by default set to TRUE for backwards compatibility
although it is very likely that this is not desired behaviour.
Extend the _push_event() function so that it can also send events to the udp
sources when asked.
Implement a custum send_event function that correctly dispatches the downstream
events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS
downstream.
Put a DISCONT event on the next output buffer when the input buffer had a
DISCONT.
Make sure we clear our adapter and reset our state before going to PAUSED.
Free the qtables.
Fixes#626869
Use 'input' instead of 'spectrum->input' which was intende already (variable
exists, but not used everywhere). Also use a local version of
'spectrum->input_pos'.
consistently only update if the property actualy changed the value. Do it
without reading the gvalue twice. No need to reset the spectrum analyzer for
threshold changes.
There's no need to call orc_init() unless you're using the Orc
API directly. All code created by orcc is guaranteed to work
without calling orc_init().
This is based on collectpads2 and is synchronizing
all streams based on the running time.
New features compared to old videomixer:
* Synchronizing frames on the running time
* Improved and simplified negotiation
* Full QoS support
* Variable framerate support
Fixes bug #626048, #624905.
Although the spec says that the clock-rate should always be 90000, some rtsp
servers send different clock-rates so we must accept then in order to handle
those streams too.
When we can't find any channel or encoding-params on the caps for dynamic
payload types, set the default number of channels to 1, as the spec says we
should.
See #623209
Don't reuse sockets but make the udpsrc element fail the state change when the
socket is already in use. If we don't prevent reuse, we might end up using the same
port for different streams in some cases.
Fixes#622017
When parsing the number of channels, use the encoding-params property from the
RTP caps because that is where we can find the channels according to the spec.
Fall back to the channels property in the caps when needed.
Fixes#623209
G729 packets may only occur intermittently (e.g. cn packets), and as such
do not allow for perfect-rtptime calculating rtp times based on frame or byte
count. In particular, do not use rtp audio base payloader as base class, but
rather base payloader directly.
To support setups with firewall/ipsec, it is useful for an rtsp client to be
able to set the range of ports that can be used for rtp/rtcp reception.
Allows this by adding a "port-range" property to the rtspsrc element.
Fixes#625153
The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
periodically as a ping. The code in gst_rtspsrc_handle_request forms an OK
response and sends, but doesn't call gst_rtsp_message_unset to free the memory
after sending the response. This results in a constant slow memory leak.
Fixes#624770
That is, in files that have no index (Cue), perform seek by scanning for
nearest cluster with timecode before requested position. Scanning is done
as a combination of interpolation and sequential scan.
Fixes#617368.
Timestamp rounding issues could lead to going backwards 2 keyframe periods
(rather than only 1). While this is not necessarily a problem, it might
potentially place additional (buffering) load on downstream and could be
avoided (because We Can).
Fixes#623629.
There seems to be a bug in libmp4v2 that generates a MPEG4BitRateBox as
(bufferSizeDB, avgBitrate, maxBitrate) instead of (bufferSizeDB,
maxBitrate, avgBitrate), according to the spec. I used the mp4file
output while writing this code, so the order is wrong. This patches
fixes that.
https://bugzilla.gnome.org/show_bug.cgi?id=623654
PluginInfo is quite a sizeable struct, let's not allocate it on the
stack, especially not if we're copying it over into another dynamically
allocated copy anyway.
Fixes#570761.
If we restart the Stream in the case of doing a transition from
PAUSED_TO_READY and back with READY_TO_PAUSED aso. the duration of the video
will get calculated even if we have a avi header with that information.
Signed-off-by: Michael Grzeschik <m.grzeschik@pengutronix.de>
On windows builds, sets source address for bind to INADDR_ANY, while
maintaining the original multicast group address for subsequent join.
Fixes#595978
Skip everything before the @ sign in the url location. VLC uses that as the
remote address to connect to (but we ignore it for now). This makes our udp urls
compatible with the ones used by VLC.
Fixes#597695
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
Subsequent entry time calculations use blockalign value to determine
number of frames per chunk, and blockalign == 1 is then most unlikely to result
in reasonable values (which also aligns with "spec").
So matroska's Block structure has no keyframe flag, only the SimpleBlock has it.
To detect keyframes in Blocks, it is just the BlockGroup container that needs
to have a ReferenceBlock attached if it is a delta frame in video.
Use new dts audio typefinder from -base to check if the PCM data
contains a dts stream. This way we recognise more varieties more
reliably and also detect the dts stream if there isn't a frame
sync right at the start of the data.
Fixes#413942.
Start cluster at every keyframe or when we would overflow the previous
cluster's relative timestamp field. This would avoid as much as possible
starting clusters at non-keyframes.
Don't send them upstream because for upstream a BYTES seek
might make sense but is completely wrong because upstream
can't seek to a byte position of the audio or video stream.
Also don't build the index in push mode for non-TIME seeks,
things will go wrong here otherwise.
This allows us to skip delta units earlier and is a bit clearer in my
opinion. It also makes only video buffers ever be delta units, not
just for SimpleBlock as before.
When the keyframe bit of SimpleBlock Flags wasn't set, the buffer was being
marked with GST_BUFFER_FLAG_DELTA_UNIT, causing all buffers to be skipped
after a seek. This may be a problem with the Sorenson Squish encoder, but
arguably the keyframe bit should only be applied to video.
Fixes bug #620358.
Even though we don't use delivery-method in our payloader, older versions of
the theora payloader in gstreamer required it. As such we need to keep this
around in the caps for backwards-compatibility.
This reverts part of 49463a37cbFixes#618940
When we calculate the frame duration, we need to use the amount of
frames in the _previous_ packet, not the current packet. The frame duration is
needed to correctly de-interleave interleaved streams. This fixes the case where
there are a variable number of frames in a packet.
Fixes#620494
This commit basically puts _get_caps() in sync with accept_caps().
If we don't have a master pad OR the master pad caps aren't negotiated
then we just return the downstream allowed caps.
If we have a master pad with negotiated caps, we return those caps
with a free range of width/height/framerate
When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't
get stuck buffering forever when there isn't enough data left to fill the
buffer.
In demuxer and muxer use the gst_util_uint64 scaling functions rather than
standard integer division. Add warnings (to be changed to debug) for debugging
the timestamp and duration.
Specifically, this reduces pushing several small buffers for each
data buffer and also avoids a seek for each buffer altogether
(though a seek is still needed for each cluster).
Fixes#619273.
Before, vp8dec had no option but to decode all frames even if some/all
of them would be late. With this change, performance when keyframes are
frequent is helped a great deal. On my Thinkpad X60s, decoding a 20 s
1080p sunflower encode with keyframes every 10 frames went from taking
42 s with 5 frames shown to 21 s with 15 frames shown (still slow
enough to count by hand). When keyframes are more sparse, you will
still be able to catch up eventually, but the results won't be as
noticable.
Fix timestamp rounding to allow the correct index to be located.
The issue was that scaling from GStreamer time format to mov time format was
rounding down causing the timestamp of the newsegment event received after a
flushing keyframe seek to find the sample index before the one it should
causing further backward seeking to the keyframe prior until no rounding error
occurred.
Rounding up when scaling to mov format has the desired effect, and it is
not clear whether just the _round () variant would be sufficient.
Fixes bug #619105
Add webm typefinder just for the release, so webm works for
people whose distros don't patch gst-plugins-base as well.
We'll remove this again after the release.
In this day and age this should be safe. There's otherwise a risk people
will be creating unneccessarily big WebM files as they can't use
SimpleBlock in v1.
The original plan was to let WebM v1 be the same as Matroska v2 (with
extra constraints), but for simplicity it was decided to handle the
versions equally, such that e.g. SimpleBlock is only allowed in WebM v2.
Failure to do this for corrupt input can cause a subbuffer bigger
than the actual buffer to be created, quickly leading to segfault.
Test case:
bug_s222005751_r0.001____memcpy.webm
The comment says this cannot happen, but it did and I don't know
why. This is not the correct fix, needs investigation. Test case:
bug_s555010094_r0.0005:0.008____IA__g_assertion_message_expr.webm
This was triggering an UTF-8 assertion in gst_caps_set_simple for
corrupt files with garbage as codec id. Test case:
gstreamer_error_trying_to_set_invalid_utf8_as_codec_id.webm
Old gst_ebml_read_ascii renamed to gst_ebml_read_string, also used by
gst_ebml_read_utf8. Unlike for UTF-8, failure to validate is an error,
as gst_ebml_read_ascii is used for reading doctype and codec id and we
might just as well give up early in those cases.
