This allows us to handle new segment events correctly; either by dropping
buffers or inserting silence; for example if the offset is changed on an srcpad
connected to audiomixer.
This reverts commit d387cf67df.
The analysis was wrong: The first 20ms of latency are introduced by the source
already and put into the latency query, making it only necessary to cover the
additional 20ms of audiomixer inside audiomixer.
Let's assume a source that outputs outputs 20ms buffers, and audiomixer having
a 20ms output buffer duration. However timestamps don't align perfectly, the
source buffers are offsetted by 5ms.
For our ASCII art picture, each letter is 5ms, each pipe is the start of a
20ms buffer. So what happens is the following:
0 20 40 60
OOOOOOOOOOOOOOOO
| | | |
5 25 45 65
IIIIIIIIIIIIIIII
| | | |
This means that the second output buffer (20 to 40ms) only gets its last 5ms
at time 45ms (the timestamp of the next buffer is the time when the buffer
arrives). But if we only have a latency of 20ms, we would wait until 40ms
to generate the output buffer and miss the last 5ms of the input buffer.
When we modify a GList (via g_list_delete_link), always reassign the
new head to the original GList. Otherwise we end up with
filtered_errors being corrupt (the head might have been the element
removed)
This function is static, and only ever called with the expose lock
taken. It thus has no reason to take this lock itself.
This was introduced by one of my locking fixes from 741355.
https://bugzilla.gnome.org/show_bug.cgi?id=741355
Check if dbin->decode_chain is NULL before running drain_and_switch_chains()
because if it is, we shouldn't run that function or it will segfault.
CID #1271074
Otherwise if there are multiple parsers we would most likely break negotiation
of the stream-format/alignment wanted by the decoders as parsers generally
support all possible stream-formats and alignments.
If caps on a newly added pad are NULL, analyze_new_pad will try to
acquire the chain lock to add a probe to the pad so the chain can
be built later. This comes from the streaming thread, in response
to headers or other buffers causing this pad to be added, so the
stream lock is taken.
Meanwhile, another thread might be destroying the chain from a
downward state change. This will cause the chain to be freed with
the chain lock taken, and some elements are set to NULL here, which
can include the parser. This causes pad deactivation, which tries
to take the element's pad's stream lock, deadlocking.
Fix this by keeping track of which elements need setting to NULL,
and only do this after the chain lock is released. Only the chain
manipulation needs to be locked, not the elements' state changes.
https://bugzilla.gnome.org/show_bug.cgi?id=741355
There was a deadlock between a thread changing decodebin/demuxer
state from PAUSED to READY, and another thread pushing data
when starting.
From the stack trace at
https://bug741355.bugzilla-attachments.gnome.org/attachment.cgi?id=292471,
I deduce the following is happening, though I did not reproduce the
problem so I'm not sure this patch fixes it.
The streaming thread (thread 2 in that stack trace) takes the demuxer's
sink pad's stream lock in gst_ogg_demux_perform_seek_pull and will
activate a new chain. This ends up causing the expose lock being taken
in _pad_added_cb in decodebin.
Meanwhile, a state changed is triggered on thread 1, which takes the
expose lock in decodebin in gst_decode_bin_change_state, then frees
the previous chain, which ends up calling gst_pad_stop_task on the
demuxer's task, which in turn takes the demuxer's sink pad's stream
lock, deadlocking as both threads are now waiting for each other.
https://bugzilla.gnome.org/show_bug.cgi?id=741355
Also improve the waiting condition for stream switches, which was assuming
before that the condition variable will only stop waiting once when it is
signaled. But the documentation says that there might be spurious wakeups.
https://bugzilla.gnome.org/show_bug.cgi?id=736655
Change the GAP events that are currently sent from the chain function of
the current pad to all other EOS pads. They should instead be sent from
their own streaming threads.
https://bugzilla.gnome.org/show_bug.cgi?id=736655
Wait in the event function when EOS is received until all pads are EOS
and then forward the EOS event from each pads own event function.
Also send a new GAP event for EOS pads from the event function whenever
going from PLAYING->PAUSED by shortly waking up the GCond. This is needed
to allow sinks to pre-roll again, as they did not receive EOS yet because
we blocked that, but also will never get data again.
https://bugzilla.gnome.org/show_bug.cgi?id=736655
There's no reason why audiomixer should override the segment
base of upstream with whatever value it got from a SEEK event,
or even worse... with 0 if there was no SEEK event yet. This
broke synchronization if upstream provided a segment base other
than 0, e.g. when using pad offsets.
Also that this code did things conditional on the element's state
should've been a big warning already that something is just wrong.
If this breaks anything else now, let's fix it properly :)
Also don't do fancy segment position trickery when receiving a
segment event. It's just not correct.
In gst_video_scale_fixate_caps () it can goto done without freeing the memory
of the tmp GstStructure. This makes it go out of scope and leak.
CID #1265766
Instead of using the GST_OBJECT_LOCK we should have
a dedicated mutex for the pad as it is also associated
with the mutex on the EVENT_MUTEX on which we wait
in the _chain function of the pad.
The GstAggregatorPad.segment is still protected with the
GST_OBJECT_LOCK.
Remove the gst_aggregator_pad_peak_unlocked method as it does not make
sense anymore with a private lock.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Reduce the number of locks simplify code, what is protects
is exposed, but the lock was not.
Also means adding an _unlocked version of gst_aggregator_pad_steal_buffer().
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Ignore chroma subsampling and color matrix transformations like the
old videoscale used to do. This is to make the performance like it was
before.
See https://bugzilla.gnome.org/show_bug.cgi?id=741987
Speex may decide not to consume any samples because it can't write any. I've
seen a hang during draining caused by the resample loop never terminating.
In that case, resampling happened as normal until olen was 0 but ilen was
still 1. _process_native then reduced ichunk to 0, so ilen never decreased
below 1 and the loop never terminated.
Instead of reverting 684cf44 ({audioresample: don't skip input samples),
break only if all output samples have been produced and speex refuses
to consume any more input samples.
https://bugzilla.gnome.org/show_bug.cgi?id=732908
VideRate keeps 1 buffer in order to duplicate base on closest buffer
relative to targeted time. This extra buffer need to be request
otherwise the pipeline may stall when fixed size buffer pool is used.
https://bugzilla.gnome.org/show_bug.cgi?id=738302
Consider pipeline: gst-launch-1.0 playbin uri=http://example.com/a.ogg
Consider 128kbit audio stream.
As soon as uridecodebin detects the bitrate, it configures its input
queue2 max-size to 32000 bytes.
The 2MB buffer in multiqueue is nearly 2 orders of magnitude bigger.
This non-deterministically drives queue2 buffer anywhere from
100% to 0% until multiqueue is filled.
This patch sets multiqueue size to 5 buffers early in no_more_pads_cb.
Partly reverts commit db771185ed.
https://bugzilla.gnome.org/show_bug.cgi?id=740689
Decodebin has already added the element to the bin and should only
select caps compatible pads. It should disable the pad link checks
to avoid doing those again.
https://bugzilla.gnome.org/show_bug.cgi?id=742885
This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until the an output buffer should've been produced according
to the latency.
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
Create a function to do the pad cleanup of the GstSourceCombine struct
and use it to not forget to also cleanup the sink pad and fix a memory
leak.
https://bugzilla.gnome.org/show_bug.cgi?id=741198
In some cases, the user might want the stream outputted by encodebin to
be in the exact same format during all the stream. We should let the
user specify when this is the case. This commit add some API in the
GstEncodingProfile to determine whether the format can be renegotiated
after the encoding started or not.
API:
gst_encoding_profile_set_allow_dynamic_output
gst_encoding_profile_get_allow_dynamic_output
https://bugzilla.gnome.org/show_bug.cgi?id=740214
Before we were setting them to PAUSED and (much) later connecting to
their source pad caps notify signal.
There was a race where that demuxer was pushing a caps and later a buffer
on its source pad when we were not even connected to its source pad caps notify
signal leading to decodebin missing the information and not keeping on
building the pipeline on CAPS event thus the demuxer was posting an ERROR
(not linked) message on the bus. This need to be done for 'simple
demuxers' because those have one ALWAYS source pad, not like usual demuxers
that have several dynamic source pads.
A "simple demuxer" is a demuxer that has one and only one ALWAYS source
pad.
https://bugzilla.gnome.org/show_bug.cgi?id=740693
There was a race where:
1) we would put the element to PAUSED
2) It would get data sent to it from upstream
3) It would thus send caps
3) caps_notify_cb would continue autoplugging
4) caps would flow downstream, the last pad would get exposed
5) we were still not done sending the sticky events
Taking the stream lock on the new element's sinkpad and only
releasing it when sticky events have all been sent prevents
the caps from reaching the source pad of the element before
we're all set.
https://bugzilla.gnome.org/show_bug.cgi?id=740694
Otherwise the following can happen:
1. set mute=true
2. play media1 (Ok)
3. play media without audio (audiochain removed)
4. play media2 (audiochain created, mute=*false*)
https://bugzilla.gnome.org/show_bug.cgi?id=740675
There's no reason why we would have to wait for the next buffer to decide
whether to output the current one or not. We just have to check if the
current one is earlier than our expected next time, which is the previous
frame timestamp plus the expected frame duration.
https://bugzilla.gnome.org/show_bug.cgi?id=740018
If there are two parser elements available for the same media format,
then decodebin is autoplugging an extra capsfilter and parser irrespective
of caps and rank. So restrict the decodebin from autoplugging multiple parser
elements back to back in adjacent positions with in a single DecodeChain
for the same media format.
https://bugzilla.gnome.org/show_bug.cgi?id=738416
timestamp_offset is being declared as an int64 variable,
for which the min
value of G_MININT64 is -9223372036854775808
Changing the minimum and maximum limit for the offset variable.
https://bugzilla.gnome.org/show_bug.cgi?id=738568
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same
https://bugzilla.gnome.org/show_bug.cgi?id=738845
The "iradio-mode" property used to have a default FALSE value in HTTP
source elements but now it should default to TRUE or just do not exist
as a property so it is not really needed to set it any more in
uridecodebin.
Apart from that this code could've never worked as uridecodebin looks for a
string-typed iradio-mode property, but it's a boolean in all sources.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725383
It is better to use storel to splat the variable into the destination.
ORC doesn't know when a variable is last written to so it can't yet optimize
away the copy operation.
Support lanczos scaling method for NV12 and NV21 formats.
Scale the 'Y' plane and scale 'NV' plane.
Implementation for submethods - int16, int32, float and double
https://bugzilla.gnome.org/show_bug.cgi?id=737400
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
audioresample and videoscale is something the application will have to do if
required, but we can at least help here by adding the
audioconvert/videoconvert elements.
https://bugzilla.gnome.org/show_bug.cgi?id=735748
When switching URI from about-to-finish, playbin starts decoding the new
URI and the queue2 inside uridecodebin starts emitting buffering messages
immediately. However, the queue(s) inside playsink still have buffers to
play and the pipeline doesn't need to pause for buffering, so we should
not send those buffering messages up to the application, otherwise there
is an audible glitch caused by pausing the pipeline for a very short time.
https://bugzilla.gnome.org/show_bug.cgi?id=727255
when downsampling, the output buffer can be filled before all the input
samples are consumed. this is correct: when downsampling, several input
samples are needed for each output sample, so when only a small number of
input samples are available the number of output samples produced can be 0.
the resampler, however, was discarding those extra input samples instead of
clocking them into its filter history for the next iteration. this patch
fixes this by removing the check that the output buffer is full. the code
now always loops until all input samples are consumed, and relies on the
calling code to have provided a suitably sized location for the output.
note that there are already other checks in place in the calling code to
ensure that this is the case.
https://bugzilla.gnome.org/show_bug.cgi?id=732908
If we had plugins and an error occurred we only include the error message
caused by this, otherwise we will include the codec description as generated
from the caps.
This allows to detect which exact codec was missing instead of getting a
generic "no suitable decoders found" error message.
Otherwise we might change some capsfeatures from ANY to the specific
value from the filter and do not filter those out in case the
sink doesn't support them
https://bugzilla.gnome.org/show_bug.cgi?id=734822
Gracefully handle switching groups that all pads are deadend.
This can happen when quickly switching programs on mpegts as the
output is unaligned it can happen that not enough data was accumulated at
parsers to generate any buffers, causing the stream to receive EOS before
any data can be decoded.
To handle this scenario, the _expose function now also gets if there is
any next group to be exposed along with the list of endpads. If there are
no endpads and there is another group to expose it will switch to this next
group and then retry exposing the streams.
Also, the requirement to only switch from the chain that has the endpad had
to be modified to care for when the drainpad is NULL
https://bugzilla.gnome.org/show_bug.cgi?id=733169
Set up a fakesink with a pad probe to replace the missing encoder to detect
if encoding was really required and only error out in this case. Otherwise
just let passthrough branch work.
This delays the error posting from the set_state function to when buffers
are really flowing. Unit test updated accordingly
https://bugzilla.gnome.org/show_bug.cgi?id=650652
Unsetting DISCONT flag means we need to copy the buffer. This copy operation
mandates that all GstMemory should be copy-able which is not always the case
https://bugzilla.gnome.org/show_bug.cgi?id=727409
We now add all our elements to uridecodebin *after*
GstBin::change_state(READY->PAUSED), so we need to post async-start
and async-done messages ourselves if we want to work async.
https://bugzilla.gnome.org/show_bug.cgi?id=733495
We now add all our elements to uridecodebin *after*
GstBin::change_state(READY->PAUSED), so we need to post async-start
and async-done messages ourselves if we want to work async.
https://bugzilla.gnome.org/show_bug.cgi?id=733495
otherwise we're going to
a) start Parser/Converter before they are linked to their capsfilter,
breaking their negotiation of a proper stream format
b) start demuxers without having connected to their pad-added signals. We
miss pads and in the worst case don't link any pads at all
... and if this fails for whatever reason we skip the element and instead
try with the next element. This allows us to handle elements that fail
when setting caps on them by just skipping to the next alternative element.
They might fail to go to PAUSED, and when connecting them further
we might already expose their srcpads on decodebin if we're unlucky.
This prevents us to handle failures going to PAUSED gracefully.
If the caps query returned us fixed caps this doesn't mean yet
that these caps are actually complete (fields might be missing).
It allows to do us some decisions, but the selection of the next
element should be delayed as only complete caps allow proper selection
of the next element.
Otherwise we might try to continue autoplugging e.g. for a specific
stream-format although the parser could convert to something else, thus giving
us potentially less options for decoders.
We can't convert to ANY capsfeatures, they are only there so that we
can passthrough whatever downstream can support... but we definitely
don't want to return them to upstream.
Canceling the accept/select happens when the source is shut down. This is
not an error and the GST_FLOW_ERROR causes problems when only part of the
pipeline is shut down.
https://bugzilla.gnome.org/show_bug.cgi?id=731567
When playing RTSP streams there will be one decodebin per stream. If some of
them fail because of a missing plugin we should not fail completely but play
the supported streams at least.
https://bugzilla.gnome.org/show_bug.cgi?id=730868
Aggregate buffering messages to only post the lower value
to avoid setting pipeline to playing while any multiqueue
is still buffering.
There are 3 scenarios where the entries should be removed from
the list:
1) When decodebin is set to READY
2) When an element posts a 100% buffering (already implemented)
3) When a multiqueue is removed from decodebin.
For item 3 we don't need to handle it because this should only
happen when either 1 is hapenning or when it is playing a
chained file, for which number 2 should have happened for the
previous stream to finish
https://bugzilla.gnome.org/show_bug.cgi?id=726423
Otherwise we might end up inside the callback without having stored
the probe id... then try to remove that probe (not!) from the callback
and wait forever for the pad to unblock.
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.
It was already checked in an early out, and as it's only
incremented for at most the size of the passed buffer, it
can only become NULL in an address wraparound.
While there, don't cast away const on a pointer.
Coverity 1139845
This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will
e
applied if possible -- for non-raw sinks, the filters will be skipped.
If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.
https://bugzilla.gnome.org/show_bug.cgi?id=679031
This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will be
applied if possible -- for non-raw sinks, the filters will be skipped.
If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.
https://bugzilla.gnome.org/show_bug.cgi?id=679031
2 seconds might be too small for some container formats, e.g.
MPEGTS with some video codec and AAC/ADTS audio with 700ms
long buffers. The video branch of multiqueue can run full while
the audio branch is completely empty, especially because there
are usually more queues downstream on the audio branch.
Usually these buffers are multiple seconds large, and having a maximum
of 5 buffers in the multiqueue there can use a lot of memory. Lower
this to 2 for adaptive streaming demuxers.
The typefinder returns LIKELY for as little as one possible
sync and no bad sync (not even taking into account how much
data was looked at for that). It's generally just not fit
for purpose, so should just not return anything like LIKELY
at all ever, even more so since it only recognises one out
of ten H263 files, and likes to mis-detect mp3s as H263.
https://bugzilla.gnome.org/show_bug.cgi?id=700770https://bugzilla.gnome.org/show_bug.cgi?id=725644
If we have the peer caps and a caps filter, return peer_caps +
intersect_first (filter, converter_caps) instead of
intersect_first (filter, peer_caps + converter_caps) and preservers
downstream caps preference order.
https://bugzilla.gnome.org/show_bug.cgi?id=724893
If we are using an adaptive stream demuxer, which outputs a non-container
stream, we are putting another multiqueue after the *parser* following
the adaptive stream demuxer. We do not want to add another instance of
the same parser right after this multiqueue.
Otherwise we will emit buffering messages not just from the last
multiqueue but also from previous multiqueues... confusing the
application with different percentages during pre-rolling.
For adaptive streaming demuxer we insert a multiqueue after
this demuxer. This multiqueue will get one fragment per buffer.
Now for the case where we have a container stream inside these
buffers, another demuxer will be plugged and after this second
demuxer there will be a second multiqueue. This second multiqueue
will get smaller buffers and will be the one emitting buffering
messages.
If we don't have a container stream inside the fragment buffers,
we'll insert a multiqueue below right after the next element after
the adaptive streaming demuxer. This is going to be a parser or
decoder, and will output smaller buffers.
Adaptive streams should download its data inside the demuxer, so
we want to use multiqueue's buffering messages to control the
pipeline flow and avoid losing sync if download rates are low;
https://bugzilla.gnome.org/show_bug.cgi?id=707636
Otherwise there's an interesting race condition when we destroy
the inputselector (actually it will be destroyed later when its state
change message gets destroyed) and afterwards release its sinkpad.
This is the code path when the last channel is removed from the
input selector.
Gave this warning sometimes, for chained oggs or whenever else
we change decode groups:
GStreamer-CRITICAL **: Padname '':sink_0 does not belong to element inputselector0 when removing
MONO and NONE position are the same, for example, but in
general there isn't much to do here for such a conversion.
Fixes problem in audioconvert, which would end up using
a mixmatrix when converting between different mono format
because it thinks MONO positioning is different from
unpositioned channels, which is not the case in this
special case. The mixmatrix would end up being 0.0 so
audioconvert would convert to silence samples.
https://bugzilla.gnome.org/show_bug.cgi?id=724509
If the text pads does not go away we just set the overlay to silent, which
allows us to immediately re-enable subs later again. However before this
change we also released the streamsynchronizer text pads, which deadlocked
because there was still dataflow going on. Just do this only if we remove
the complete chain.
https://bugzilla.gnome.org/show_bug.cgi?id=683504
Change the way autoplug-select is accumulated so that it's possible to have
multiple handlers. The handlers keep getting called as long as they keep
returning GST_AUTOPLUG_SELECT_TRY.
One practical example of when this is needed is when hooking into playbin's
uridecodebin, which is perhaps not very elegant but the only way to influence
which streams playbin autoplugs/exposes.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723096
Discussion on IRC indicated that the main reason for this list was to
prevent demuxers that can trigger a lot of seeking from using
progressive buffering using queue2 (which due to being seekable triggers
that behaviour).
However given that upstream can indicate seeks are possible but should
be avoided via a scheduling query, this extra whitelisting shouldn't be
necessary for well-behaved demuxers.
https://bugzilla.gnome.org/show_bug.cgi?id=704933
Make a little table of conversions and manually score them. Use this
info to define better weights for the scoring algorithm.
give separate scores for doing changes and the impact of the change,
This allows us to avoid conversion when we can but still allow fairly
lossless changes.
The old code did not penalize GRAY conversions, PAL conversions were
punished too low and depth conversions too high.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722656
Don't try to interpolate the chroma samples, the used algorithm only
works for horizontal cositing. Let's switch to a faster and safer
version until we handle chroma siting correctly in the fastpaths.
Rework the orc code to be around 10% faster and support arbitrary matrices.
Pass the matrix parameters to the YUV->RGB functions to make them work
for all matrices. This enables more and faster fastpath conversions.
See https://bugzilla.gnome.org/show_bug.cgi?id=721701
This fast-path was adding 128 to every component including
alpha while it should only be done for all components except
alpha. This caused wrong alpha values to be generated.
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.
If a pipeline fails to preroll, it might happen that the sinks are
put into READY state from playbin's sink activation, but they are never
set to playsink, so they aren't being managed by a GstBin and will keep
their READY state until they are unreffed, leading to a warning.
Prevent this by always forcing them to NULL when deactivating a group
https://bugzilla.gnome.org/show_bug.cgi?id=708789
Fix component ordering, it's wrong in both the scanline and merge
function so it cancels eachother out and isn't really a except for
loss of precision of the green component.
Fix calculation of the filter weight
Some of the fastpath function can only work with aligned widht/height
so make sure we check this as well when choosing a fastpath.
Add fastpath for I420/YV12 -> BGRx
This commit adds detection of the "dash" and "avc3" compatible brands
in qt_type_find.
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV
box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in
the first sample of every fragment (i.e. the first sample in each mdat
box). The principal reason for avc3 is to make it easier for client
implementations, because it removes the requirement to insert the
SPS+PPS in to the decoder pipeline every time there is a representation
change.
https://bugzilla.gnome.org/show_bug.cgi?id=702004
Increase the number of temporary lines that we need, it is possible that the
up and downsampling offsets are out of phase and that we need to keep some
extra lines around. Also copy the unhandled output lines for the next round
instead of overwriting them.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706823
When playing mp3 files from a smb server, we get 64k read requests
that mostly overlap. Without using the cache to partially satisfy
these, we send these requests straight to the server, resulting in
a lot more network traffic than necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=705415
Each write will update the last_activity_time and otherwise we would
compare against a too old current time and immediately timeout because
current time is smaller than last activity time (overflow).
Each write will update the last_activity_time and otherwise we would
compare against a too old current time and immediately timeout because
current time is smaller than last activity time (overflow).
Remove dodgy code that detects mp3 with as little as
a valid frame sync at the beginning. This was only used
in some unit tests in -good where there were only a few
bytes after the id3 tag. We now require at least two
frame headers.
Fixes mis-dection of text files with UTF-16 LE BOM as mp3.
https://bugzilla.gnome.org/show_bug.cgi?id=681368
We have to hold the streams-lock when iterating over all pads,
also the stream-lock of the pad is already locked when we receive
EOS.
Call gst_pad_event_default() for the correct default handling of
events.
This commit adds a streamcombinerpad with an is_eos field.
When streamcombiner receives an EOS on one of its pads, it
forwards it all its other pads are EOS.
This commit also removes the notion of "stream-switching-eos".
In gst_sub_parse_dispose() parser_type will be UNKNOWN,
so these deinit calls were never executed. And we should
clean up the parser state in the downwards state change
anyway.
To celebrate 2013.gnome.asia, updated sami parser for gstreamer 1.x. :D
Remove conditional block for check libxml usage and
implement a simple html markup parser for the sami
parser.
https://bugzilla.gnome.org/show_bug.cgi?id=693056
Otherwise we will remove the bus that would proxy messages to playsink
and never set it again. If the sink is already in playsink, all failures
are fatal anyway as it's either a sink that worked before or one that
was set by the user.
https://bugzilla.gnome.org/show_bug.cgi?id=701997
playbin will now only activate the sinks in a single place and
will never change the states of any sinks that are owned by
playsink.
Also handle text-sinks the same way as audio/video sinks inside
playbin.
With the current test, we get into problems when we try to typefind
a MPEG stream from a small amount of data, which can happen when
we get data pushed from a HTTP source. We thus make a second test
to give higher probability if all the potential headers were either
pack or pes headers (ie, no potential header was unrecognized).
This fixes an issue with a MPEG1/MP2 stream being properly discovered
as video/mpeg from a file, but as audio/mpeg from souphttpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=703256
This makes sure the application gets any context related messages and
can do whatever is required to a) get the sink a context or b) share
the context with other elements in the pipeline.
The proxying is necessary because the sink is not a child element of
playbin, but instead will at a later point be a child of some bin
inside playsink.
https://bugzilla.gnome.org/show_bug.cgi?id=700967
Otherwise we're going to deadlock forever because no autoplugging
happens without having caps, but caps can never be send because
we're blocking.
Serialized queries before caps should never be sent unless really
necessary.
We found a case where untranslated values were being passed from the
proxy to the underlying channel, causing bad color balance values
in some setups.
Thanks to Sebastian Dröge for clarifying how the code works, and
suggesting the fix.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701202
This allows to chose something else than input-selector
for multiple audio/video/text streams, e.g. an adder could
be used for audio.
It is needed for example to implement some of the more
advanced HTML5 video features.
https://bugzilla.gnome.org/show_bug.cgi?id=698851
Add the actual decoder/parser/etc caps at the very end to
make sure we don't cause empty caps to be returned, e.g.
if a parser asks us but a decoder is required after it
because no sink can handle the format directly.
Otherwise we will only block after the serialized, non-sticky event
after the CAPS event or the first buffer. If we're waiting for another
pad to finish autoplugging after we got final caps on this pad, it
will mean that we will let the ALLOCATION query pass although the
pad is not exposed yet.
Otherwise we accumulate more and more queue2 elements, and let each
of them start a thread doing nothing but waiting each time uridecodebin
goes to PAUSED.
https://bugzilla.gnome.org/show_bug.cgi?id=699794
This makes it possible to take advantage of the O(log n) lookups
of GSequence on the ~1000 element lists and only do iterations
on <10 element lists. Previously the code iterated over ~1000 element
lists multiple times.
Autoplug the decoder elements and sink elements based on
the number of common capsfeatures if the ranks are the same.
This will also helps to autoplug the h/w_decoder and h/w_renderer.
https://bugzilla.gnome.org/show_bug.cgi?id=698712
Remove the byte limit for adaptive http streaming. Because some fragments might
be very big, we might need a lot of buffering. I also suspect another problem
where data is actually missing and things go out of sync somehow.
When we disable buffering in the more upstream multiqueue elements,
we need to also update the queue limits. In particular, the max_size_time should
be set to 0 or else we might simply deadlock.
When we have a scenario of demuxers linked to demuxers, decodebin2
will create multiqueue at different levels of the pipeline. The problem
is that only the lowest multiqueue's should do the buffering messaging,
as they will handle with the raw streams data.
When all multiqueues are doing buffering, the upper ones can handle
large buffers that easily fill them, moving from 0% to 100% from
buffer to buffer, causing too much buffering messages to be posted.
This hangs the pipeline unnecessarily and might lead to deadlocks.
Decodebin2's chains store a next_groups list that was being handled as
it could only have a single element. This is true for most of the
chaining streams scenarios where streams change not very often.
In more stressfull changing scenarios, like adaptive streams, those
changes can happen very often, and in short time intervals. This could
confuse decodebin2 as this list was always being used as a single
element list.
This patches makes it handle as a real list, using iteration instead
of picking the first element as the correct one always.
Even if the chain hasn't been 'handled' in this switching round,
report it as drained so upper chains/groups know abou it.
This makes switching happen on upper levels of the groups/chain
trees
Checks if the received XML is a smoothstreaming manifest
in both UTF8 and UTF16 formats. The check is made for a
SmoothStreamingMedia top level element.
Conflicts:
gst/typefind/gsttypefindfunctions.c
If a source element could be created for a URI, but all elements rejected
the URI for some reason, propagate the error from the URI handler instead
of reporting a 'no uri handler found for protocol xyz' error, which is
confusing. Fixes error reporting with dvb:// URIs when the channel config
file could not be found or not be parsed or the channel isn't listed.
https://bugzilla.gnome.org/show_bug.cgi?id=678892
Use a scheduling query to check if the source element has some
bandwidth limitations. If this is the case on-disk buffering might be
used. If the source element doesn't handle the scheduling query then
fallback to checking the URI protocol against the hardcoded list of
protocols known to handle buffering already.
Fixes bug 693484.
The compare_factories_func() should return negative value
if the rank of both PluginFeatures are equal and the name of
first PluginFeature comes before the second one (== ascending order).
The _decode_bin_compare_factories_func() should return negative
value if the rank of both PluginFeatures are equal and the name of
first PluginFeature comes before the second one (== ascending order).
This allows getting a pad for a specific encoding profile, which can
be useful when there are several stream profiles of the same type.
Also update the encodebin unit tests so that we check that the returned
pad has the right caps.
https://bugzilla.gnome.org/show_bug.cgi?id=689845
Before it was done the other way around and that can trigger the assert that
already is in place. This also makes more sense; when seeking to time x, we want
then sample that is <= that pos.
Try to select the conversion that would result in the minimal amount of quality
loss. Quality loss is calculated rather arbitrarily but it avoids doing
something really stupid in most cases.
This reverts commit adc9694ed7.
No need to restrict the conversion, we can handle interlace correctly. We
basically unpack each field, then convert each field to the target colorspace
and pack and interleave each field to the target format. We also disable any
fast path that can't deal with interlaced formats.
Do not use the buffer start offset when it is invalid, otherwise a
discontinuity is detected on the next buffer, and the subtitle parser
reset and some subtitle lines are not shown.
Also remove unused next_offset field.
https://bugzilla.gnome.org/show_bug.cgi?id=693981
subtitleoverlay handles any caps, not just the ones
for which a subtitle parser/renderer exist. It will
just ignore any unsupported streams instead of causing
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=688476
Add all the caps that we can convert to to the filter caps,
otherwise downstream might just return EMPTY caps because
it doesn't handle the filter caps but we could still convert
to these caps, causing us to return EMPTY caps although
conversion would be possible.
https://bugzilla.gnome.org/show_bug.cgi?id=688803
changed: gst_video_scale_set_info in gst/videoscale/gstvideoscale.c
DAR on sink side now calculated with PAR on sink side
ratio of output width/height now calculated with inverse PAR
additional condition that borders are 0:0 for passthrough mode
https://bugzilla.gnome.org/show_bug.cgi?id=696019
Ensure the detection of svc and mvc as a part of h264 stream.
Once the typefinder detect a subset_sequence_parameter_set(ssps),
then each nal unit with type 14 or 20 should be detected as a
part of h264 stream thereafter.
https://bugzilla.gnome.org/show_bug.cgi?id=694346
Previously adder was only sending the flush-stop, when it saw the flushing seek.
If one sends a flushing see direcly to an element upstream of adder, it would
fail to unflush the downstream pads.
This ensures the ghost pad will not stay in flushing mode
when it receives a flush stop event, and generally behave
badly.
This fixes at least one case of a dynamic decodebin2 + encodebin
pipeline finding a source that has not prerolled when it should
have been (due to the ghostpad staying in flushing mode).
We were setting the query-func on the sink-pad, which got overwritten when
adding the new pad to collect pads. Instead register our query-func with the
collect pads object. This fixes filter caps. Add a test for it.
A return value of FALSE here indicates that we don't have control-values. In
0.10 we were returning the default value of the property. Now we don't fill an
array with defaults in the ControlBinding, but leave it up to the element to
handle this case.
The codec data blob we get from matroskademux with the SSA/ASS
init section is supposed to be valid UTF-8. If it's not, just
continue with the bits that are valid UTF-8 instead of erroring
out. We don't actually parse the init section yet anyway..
https://bugzilla.gnome.org/show_bug.cgi?id=607630
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)
+ Add tests that it behave correctly
When the input buffers for a stream don't have a duration set,
timestamp_end might still be GST_CLOCK_TIME_NONE. When advancing
EOSed streams via GAP events (with other streams not yet EOS), we
would then use the invalid timestamp_end to calculate the duration
of the gap. This in turn would make baseaudiosink abort, because it
would try to allocate memory for a trizillion samples.
So if buffers don't have a duration set, assume a duration of
one second for stream catch-up purposes, just so we can still
continue to catch up in those cases. And make sure that
timestamp_end is valid before doing calculations with it.
http://bugzilla.gnome.org/show_bug.cgi?id=678530
Make AAC LOAS typefinding a bit more reliable; don't report
a LIKELY probability already after just two sync points, but
scan for a few more consecutive frames and determine probability
based on how many we found. Fixes mis-detection of wavpack file.
https://bugzilla.gnome.org/show_bug.cgi?id=687674
Check for second block sync and return different
probabilities depending on what we found (trumping
the AAC loas typefinder's LIKELY probability after
finding a second frame sync in this particular case).
https://bugzilla.gnome.org/show_bug.cgi?id=687674
Previously we could've chosen another format with the same
depth even if the input format was possible.
Also make sure to chose according to the order in the
caps.
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
This is an adaptation of patch #3 from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008240.html ),
but without the NEON optimizations (these come in a separate commit).
The idea is to replace SATURATE32(PSHR32(x, shift), a) operations with a
combined SATURATE32PSHR(x, shift, a) macro that can be optimized for
specific platforms (and also avoids rare rounding errors).
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
There were two issues with the previous decodebin2 group switching algorithm:
Issue 1: It operated with no memory of what has been drained or not, leading to
multiple checks for chains/groups that were already drained.
Issue 2: When receiving an EOS, it only detected that a higher-level chain
was drained if it contained the pad receiving the EOS.
The following modifications have been applied:
- a new drained property has been added to GstDecodeChain
- both drained properties of chain/group are set as soon as they are detected
- the algorithm now tests agains these values
See https://bugzilla.gnome.org/show_bug.cgi?id=685938
Should fix "cannot register existing type `GstPlaybinSelectorPad'" warnings
and subsequent errors when creating multiple players at the same time.
Conflicts:
gst/playback/gststreamselector.c
GstId3Mux sink pad is an always (static) pad. Thus releasing it
as if a request pad triggers:
(sound-juicer:11826): GStreamer-CRITICAL **:
gst_element_release_request_pad: assertion `GST_PAD_PAD_TEMPLATE (pad)
== NULL || GST_PAD_TEMPLATE_PRESENCE (GST_PAD_PAD_TEMPLATE (pad)) ==
GST_PAD_REQUEST' failed
https://bugzilla.gnome.org/show_bug.cgi?id=685110
Need to store the old running time and frame numbers when renegotiating and
start from 0 again when a new caps is set, preventing that framerate changes
cause timestamping issues.
For example, if a stream pushed 10 buffers on framerate=2/1, its
running time will be 5s. If a new framerate of 1/1 is set, it would
make the running time go to 10s as it would count those 10 buffers
as being sent on this new framerate.
Fixes camerbin unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=682973
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c: In function 'gst_audio_resample_dump_drain':
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c:729:9: warning: variable 'in_len' set but not used [-Wunused-but-set-variable]
streams with non-TIME segments will not have timestamps ...
... and therefore will never unblock the other streams.
Fixes blocking issue when using playbin suburi feature
People expect audiorate to fix things up and not make things worse
by default, so let's default to a similar tolerance as audiosinks
do. Should help with transcoding and the like, though one might
possible still want higher values then.
We can't just make a vfunc that takes a union of int
and pointer as argument, and then set up subclass-specific
action signals and signals that take int (in multifdsink's
case) or a GSocket * (in multisocketsink's case), and then
expect everything to Just Work. This blows up spectacularly
on PPC G4 for some reason.
Fixes multifdsink unit test on PPC, and fixes aborts in
multisocketunit test (now hangs in gst_pad_push - progress).
* Update outgoing segment.base with accumulated time, ensuring all
streams are synchronized.
* Only consider streams as "new" is they have a STREAM_START event
with a different seqnum.
* Use GstStream segment.base instead of separate variable to store
the past running time.
* Disable passthrough
* Switch to glib 2.32 GMutex/GCond
* Avoid getting pad parent the expensive way
* Minor other fixes
Make sure to send a CAPS event downstream when we get our
first input caps. This fixes not-negotiated errors and
adder use with downstream elements other than fakesink.
Even gst-launch-1.0 audiotestsrc ! adder ! pulsesink works now.
Also, flag the other sink pads as FIXED_CAPS when we receive
the first CAPS event on one of the sink pads (in addition to
setting those caps on the the sink pads), so that a caps query
will just return the fixed caps from now on.
There's still a race between other upstreams checking if
caps are accepted and sending a first buffer with possibly
different caps than the first caps we receive on some other
pad, but such is life.
Also need to take into account optional fields better/properly.
https://bugzilla.gnome.org/show_bug.cgi?id=679545
Fix invalid memory access caused by broken pointer arithmetic.
If we have a uint16_t *tmpbuf and add n * dest->stride to it, we
skip twice as much as we intended to because dest->stride is in
bytes and not in pixels. This made us write beyond the end of
our allocated temp buffer, and made the unit test crash.
Make function pointers NULL when nothing needs to be done.
Pass target pixels to dither and matrix functions so that we can later make
them operate on the target buffer memory directly.
This allows the following use-cases to expose the group and pads
before an ALLOCATION query comes through:
* Single stream use-cases
* Multi stream use-cases where all streams sent the CAPS event before
the first ALLOCATION query
Some cases will still make the initial ALLOCATION query fail though,
which isn't optimal, but not fatal (it will recover when pads are
exposed, a RECONFIGURE event is sent upstream and elements can
re-send an ALLOCATION query which will reach downstream elements).
https://bugzilla.gnome.org/show_bug.cgi?id=680262
A caps event is also used to establish that a stream has prerolled.
Without this, we end up allowing negotiation queries to fail, ending
in decoders (and other elements) to not be configured right from the
start with the most optimal settings.
videoconvert.c: In function 'videoconvert_convert_new':
videoconvert.c:287:11: error: 'Kr' may be used uninitialized in this function
videoconvert.c:287:15: error: 'Kb' may be used uninitialized in this function
Fix the calculation of the offset and scale values for GRAY formats. We also
need to set the offset and base of the chroma values to match what the unpack
function creates.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679612
Might just be paranoia, but better safe than sorry. Make sure
the compiler really always passes a 64-bit integer to the
g_object_set() vararg function.
They are not added again by every code path, e.g. when switching
only the deinterlace flag and are missing then.
Fixes bug #678763.
Conflicts:
gst/playback/gstplaysink.c
...and in playbin2 additionally prefer sinks over parsers.
This makes sure that we a) always directly plug a sink if it supports
the (compressed) format and b) always plug parsers in front of decoders.
This avoids that bin being leftover and being found when reusing playbin2,
and fixes restarting on a new URI after failing to activate with a previous
URI.
https://bugzilla.gnome.org/show_bug.cgi?id=673888
For audio/video we should flush too for fastest stream switches but this
currently isn't possible because the flushes would need to go to the sink,
which then causes state changes and causes all timing information to be
changed.
Should work out of the box in 0.11 with the flush-stop that doesn't reset
the times.
Conflicts:
gst/playback/gstplaybin2.c
gst/playback/gstplaysink.c
gst/playback/gstsubtitleoverlay.c
Sending a non-flushing seek might not be enough for switching
to an external sub that has already been used because the flushes
are needed to reset the state of its decodebin's queue.
For example, if the subtitle is short enough, the queue might get
and EOS and keep its 'unexpected' return state. If the user switches
to another subtitle and back to the external one, the buffers
won't get past the queue.
This patch fixes this by adding the flush flag to the seek and
preventing that this flush leaves the suburidecodebin.
https://bugzilla.gnome.org/show_bug.cgi?id=638168
Conflicts:
gst/playback/gstplaybin2.c
RGB8_PALETTED -> RGB8P
Fix the definition of paletted formats, store the palette in the second
plane.
Make sure we copy the palette correctly in gst_video_frame_copy()
Don't do alignment on the palette in videopool
Add support for the I420_10 formats
Use the video frame api to get pixels and strides instead of our own
custom versions. Fixes the YVU9 format and probably some others.
Make the uri property getter return the next uri, like it was configured in the
setter.
Make a new current-uri and current-suburi property that reflects the currently
playing uri and suburi.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676665
This makes sure that we always prefer sinks that support a format without
decoding, independant of its rank. Previously we only sorted by rank.
Conflicts:
gst/playback/gstplaybin2.c
If a property is not found (for example last-sample when
gst_debug_bin_to_dot_file is used while the pipeline is
slightly broken (thus no last-sample) the unref of the item
gvalue which is not refed fails. Only unref if it was found.
They're hardly used, and probably more confusing than anything
else, and it's not clear that anyone would really need to be
able to tell them apart at the media type level.
The sinkpads are unblocked when going from PAUSED->READY, we need to block them
again when going READY->PAUSED. The blocking of the pad previously only happened
when it was freshly obtained with _request_pad or when the caps changed. If we
don't release the pad when going to READY it was previously never blocked again
causing not-linked errors.
For example the Sintel subtitles have this and without this change
they're detected as text/plain and not usable as subtitles. The
parser itself already handles this just fine.
For streaming sources a queue is added before the demuxer, which can not be
properly filled by live sources. As http source can be live sources, this
caused issues for example with http live sources.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674057
Adds a property for playsink to define how it should handle
events sent in send_event function. The default is the same as
GstBin's, sending events to all internal sinks. There is also
mode-first, that will send to sinks until the one handles the
event successfully.
https://bugzilla.gnome.org/show_bug.cgi?id=673211
When the video sink is a fakesink, which does not implement the
navigation interface, playsink will drop the navigation command.
In this case, send to the video sink as a fallback. It breaks
the interface abstraction, but is better than just dropping the
navigation event.
Turn _sink_event() into the collectpads event function and merge the logic from
the recently added gst_adder_event. Drop flush_start events as we allready
handle them on the src-pad side. Fixes#670850.
Now that we no longer support all methods for all formats, we
need to cater for that in the transform function: we can't
transform formats not supported by the currently-selected
mehod.
make check, folks. It's da bomb.
Only return LIKELY probability if we've seen an SPS, PPS and an
IDR slice nal, i.e. try harder to avoid false positives such
as with certain VC-1 files.
https://bugzilla.gnome.org/show_bug.cgi?id=668565
We need to call the default query handler of the proxy pad because only that one
will forward the query to the target pad in case of the allocation query.
After a PAUSED->READY change the sink pads are currently not set to
blocking state. When the element is set back to PAUSED, the change will
be done asynchronously, but as the _pad_blocked_cb() callback is now not
called, the state change never completes.
Fix that by setting the sink pads to blocking state on a PAUSED->READY
change, which ensures that the _pad_blocked_cb() is called when needed
on any future READY->PAUSED change. The sink pads are already put to
blocking state on NULL->READY change, so this behavior is consistent.
Fixes bug #668097.
In order to allow for proper functionality when a decoder only supports
one instance at a time (dsp), we must block the demuxer pads when they
get created if they are not part of the active group, preventing buffers
from being sent to the decoder (and initializing it through setcaps),
then after we switch to a new group, we unblock the demuxer pads for
the active groups. In the callback for the unblock, we prune the old
groups, making sure the previous decoder instance is destroyed before
we push a buffer to the new instance.
Since caps are no longer 'shared' between two pads (but forwarded from
source pad to sink pad) we end up with the first chain pad not having
specified caps (i.e. typefind:src).
This solves the issues by getting the pad's peer caps.
It is not optimal since it will (for most demuxers) return the pad
template caps, which might contain non-fixed caps (ex : with
qtdemux "video/quicktime; video/mj2; audio/x-m4a; application/x-3gp")
https://bugzilla.gnome.org/show_bug.cgi?id=667337
... to avoid unnecessary spurious errors (upon e.g. shutdown).
If a real error is applicable in this unusual circumstance (missing other pad),
other (STREAM_LOCK protected) call paths can take care of that.
We have removed things like protocol=gdp in the tcp elements
in favour of explicit gdppay/depay elements, so there's no need
to keep a public API and library for now. We can still add it
back later. Someone needs to think hard about 0.11 and gdp
anyway one of these days.
Make a new method to allocate a buffer + memory that takes the allocator and the
alignment as parameters. Provide a macro for the old method but prefer to use
the new method to encourage plugins to negotiate the allocator properly.
Improve GstSegment, rename some fields. The idea is to have the GstSegment
structure represent the timing structure of the buffers as they are generated by
the source or demuxer element.
gst_segment_set_seek() -> gst_segment_do_seek()
Rename the NEWSEGMENT event to SEGMENT.
Make parsing of the SEGMENT event into a GstSegment structure.
Pass a GstSegment structure when making a new SEGMENT event. This allows us to
pass the timing info directly to the next element. No accumulation is needed in
the receiving element, all the info is inside the element.
Remove gst_segment_set_newsegment(): This function as used to accumulate
segments received from upstream, which is now not needed anymore because the
segment event contains the complete timing information.
Hide the GstStructure of the event in the implementation specific part so that
we can change it.
Add methods to check and make the event writable.
Add a new method to get a writable GstStructure of the element.
Avoid directly accising the event structure.
So run-time bindings can introspect the names correctly (we abuse this
field as description field only in elements, not for public API
(where the description belongs into the gtk-doc chunk).
https://bugzilla.gnome.org/show_bug.cgi?id=629946
Adds that warning to configure.ac
Includes a tiny change of the GST_BOILERPLATE_FULL() macro:
The get_type() function is no longer declared before being defined.
https://bugzilla.gnome.org/show_bug.cgi?id=611692
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
Don't write to the same region of memory as a uint64 and uint16
as this breaks strict aliasing rules and apparantly breaks on PPC
and s390. Thanks to Sjoerd Simons for analysing. Fixes bug #348114.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_packet_from_event_1_0):
When calculating GDP body CRC, use the correct pointer.
Fixes part of #522401.
Original commit message from CVS:
2006-08-11 Andy Wingo <wingo@pobox.com>
* configure.ac:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packetizer_new):
* tests/check/libs/gdp.c: (gst_dp_suite): GST_DISABLE_DEPRECATED
is only for users of API that don't want to see deprecated
functions in the headers; people that want to compile out
deprecated code should pass -DGST_REMOVE_DEPRECATED into the
CFLAGS. Fixes the build of multifdsink, or will soon..
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer_any), (gst_dp_packet_from_caps_any),
(gst_dp_crc), (gst_dp_header_payload_length),
(gst_dp_header_payload_type), (gst_dp_packet_from_event),
(gst_dp_packet_from_event_1_0), (gst_dp_buffer_from_header),
(gst_dp_caps_from_packet), (gst_dp_event_from_packet_0_2),
(gst_dp_event_from_packet), (gst_dp_validate_header),
(gst_dp_validate_payload):
Make debug category static
Constify the crc table.
Do some more arg checking in public functions.
Fix some docs and do some small cleanups.
* tests/check/libs/gdp.c: (GST_START_TEST), (gst_dp_suite):
Add some more checks to see if GDP deals with bogus input.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_event_from_packet_1_0):
Fixes#347337: failure to deserialize event packets with
empty payload (only event type)
Original commit message from CVS:
* docs/README:
* docs/images/gdp-header.svg:
add a gdp image
* docs/libs/Makefile.am:
* docs/libs/gdp-header.png:
* libs/gst/dataprotocol/dataprotocol.c:
add it to the API docs
* docs/manual/intro-motivation.xml:
fix typo
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
factor out CRC code
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
factor out some common header init code
Original commit message from CVS:
* docs/libs/gstreamer-libs-sections.txt:
* docs/libs/tmpl/gstdataprotocol.sgml:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_crc):
* libs/gst/dataprotocol/dataprotocol.h:
API: make gst_dp_crc() public
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event),
(gst_dp_event_from_packet):
Fixes in reading/writing events over GDP (not currently used?) -
dereferencing NULL events for unknown/invalid event types, memory
leak, and change g_warning to GST_WARNING.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
Fix docs for dataprocotol to not get the return types completely
wrong for a few functions.
Original commit message from CVS:
2005-10-13 Andy Wingo <wingo@pobox.com>
* libs/gst/dataprotocol/dataprotocol.c (gst_dp_packet_from_caps):
Fix Timmeke Waymans bug.
(gst_dp_caps_from_packet): Make sure we pass a NUL-terminated
string of the proper length to gst_caps_from_string. There's a
potential for, before this fix, that this could cause someone
connecting over the network to cause a segfault if the payload is
not NUL-terminated.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
* libs/gst/dataprotocol/dataprotocol.h:
* libs/gst/dataprotocol/dp-private.h:
It's about time we bump the version number.
Since event types don't fit in the guint8 anymore describing
the payload type, make payload type 16 bits wide.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event),
(gst_dp_event_from_packet):
Fix serialization of seek events.
