Add a read source on write socket when lost tunnel.
To be able to detect when clint closes get channel.
This is already done in gst_rtsp_source_dispatch_write but
only when the queue is empty.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730368
By re-using the uri argument for storing local data, we could end up in
a situation where we would free uri ... which would actually be the
string passed in argument.
Instead explicitely use a local variable. Fixes double-free issues.
CID #1212176
Buffer pool set_config() may return FALSE if requested configuration needed small
changes. Reget the config and try setting it again. This ensure we have a configured
pool if possible.
Currently the API is far from optimal and the user has to work around
our badly defined API to simply install missing plugins.
API:
new:
gst_discoverer_info_get_missing_elements_installer_details
deprecated:
gst_discoverer_info_get_misc
gst_discoverer_stream_info_get_misc
https://bugzilla.gnome.org/show_bug.cgi?id=720596
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.
We were returning in various places without unreffing the caps, and
we were also leaking (overwriting) the caps we got from _get_current_caps()
Spotted by Haakon Sporsheim in #gstreamer
This should allow for more meaningful errors. Dereferencing NULL
is more useful information than dereferencing a random address
happened to be on the stack.
If gst_video_overlay_rectangle_apply_global_alpha is called with
a rectangle with unsuitable alpha, expanding the alpha plane will
fail, and thus lead to dereferencing a NULL src pointer. It's not
certain this will happen in practice, as the function is static
and callers might ensure suitable alpha before calling, but there
is no apparent explicit such check.
Add prologue asserts for proper alpha to explicitely prevent this.
Coverity 1139707
Videodecoder does late renegotiation, it will wait for the next
buffer before renegotiating its caps and bufferpool. It might happen
that downstream element switched from passthrough to non-passthrough
and sent a reconfigure upstream (that caused this renegotiation).
This downstream element will ask the video sink below for the bufferpool
with an allocation query and will get the same bufferpool that
videodecoder is holding, too.
When renegotiating, if videodecoder deactivates its bufferpool it
might be deactivating the bufferpool that some element downstream
is using and cause the pipeline to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=727498
Clock slaving can clip start time to zero, giving us a shorted
duration than we originally got. To keep in sync, we must then
discard the samples falling before that zero timestamp.
This possibly fixes random distortion caused by constant PA
underflows which are never resynced.
The KEMAC payload actually needs to have subpayloads and the key should
go into the KEY_DATA subpayload. Add support for subpayloads and
implement the KEY_DATA payload.
Add some pointers to the conversion functions that allow us to add
encryption and decryption later.
baseparse will reverse each GOP for us already, so the segment events can
be after our keyframe. Make sure to get it and all other relevant sticky
events before starting to decode.
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
* Only check for conditions we are interested in.
* Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
will always be reported if they are true.
* Do not create timed source if timeout is NULL.
* Correctly wait for sources to be dispatched, context_iteration() is
not guaranteed to always block even if set to do so.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
Previously the sequence number kept track of by GstRTPBasePayload would
only be set when going from READY to PAUSED state. This meant that a
downstream element that attempted to configure a basepayloader by
setting seqnum-offset e.g. in its sinkpad's caps template would have
trouble configuring the basepayloader. The reason was that the caps
event which arrives with the desired value for seqnum-offset did not
arrive at the basepayloader until caps negotiation took place,
significantly later than the transition from READY to PAUSED.
The result after this patch is that the default value for the
seqnum-offset property, or later set values for this property, will take
effect when going from READY to PAUSED like before. In addition the an
arriving caps event will also affect the basepayloaders configured
sequence number as the event arrives.
The payload type field in an RTP packet header is 7 bits wide, hence the
boundary values ought to be 0x00 and 0x7f, not the previously stated
values 0x00 and 0x80.
Two new functions have been added,
gst_rtsp_connection_set_tls_database() and
gst_rtsp_connection_get_tls_database(). The certificate database will be
used when a certificate can't be verified with the default database.
https://bugzilla.gnome.org/show_bug.cgi?id=724393
This was a regression introduced by f52fd7a68, where we started using
the stride to encode the dimensions in tiles. This patch simply updates
offset and size calculation as described in the documentation,
part-mediatype-video-raw.txt.
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
* Change running time type to guint64
* Use GST_CLOCK_TIME_NONE() to check for invalid timestamps
* Name variables so ns-based and hz-based timestamps are evident
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
We call the _get_time function from the provided clock and we don't lock
the sink object for performance reasons. Make sure we only read and
check variables once so that they don't change while we are executing
the code.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720661
For default caps generation when handling gap events that are sent
before any buffer, try to use caps that are closer to what upstream
provided to avoid fixating rate or channels to 1 as default.
So there are the steps:
1) Try to set rate, channels and channel-mask from upstream if provided
2) Fixate the rate and channels to the default rate and channels from
audio lib
3) Fixate the caps just to be sure everything is fixed
4) If no channel-mask was provided and channels > 2, use a default
channel-mask (taken from audioconvert code)
https://bugzilla.gnome.org/show_bug.cgi?id=722144
Before trying to generate a default fixated caps when handling a gap
event, make sure that the same strategy that is used when handling
a buffer has been attempted. Otherwise audiodecoder will ignore
upstream caps settings such as rate and channels and will likely
end with a caps with channels=1 and rate=1.
https://bugzilla.gnome.org/show_bug.cgi?id=722144
Instead of using extra plane, we encode the number of tiles in x and y in the stride of
each planes (i.e. y_tiles << 16 | x_tiles) and introduce tile_mode, tile_width and
tile_height into GstVideoFormatInfo structure.
https://bugzilla.gnome.org/show_bug.cgi?id=707361
For reverse playback, the segment event will only be pushed when
the first buffer is actually pushed. But for decoding frames and storing
those into the list to be pushed the output_segment.rate value is used
to determine if it is forward or reverse playback.
In case a previous segment event (or none) is in use it will mistakenly
think it is doing forward playback and push the buffers immediatelly and
try to clip buffers based on an old segment (or an uninitialized one, leading
to an assertion)
This patch fixes this by copying the segment earlier if on reverse playback
https://bugzilla.gnome.org/show_bug.cgi?id=721666
Add a method to make a media-type from the transport. Deprecate the old
method that only used the mode.
Based on patch from Aleix Conchillo Flaqué <aleix@oblong.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720219
Port a change from audiobasesink from def07410, to ignore setcaps
when the caps don't actually change, and avoid a reconfiguration
and reset of the ringbuffer in that case.
And don't assume in other code that set_format() preserves any fields at
all. These assumptions were already made here for fields that were changed
by set_format().
If there are no caps from the audio decoder when handling a GAP
event - as when one is received right at the start on a DVD without
initial audio - then choose any default caps for downstream and
then send the GAP, so the audio sink has a configured format in
which to start the ringbuffer.
Also, make the audio sink reject a GAP without caps with a clearer
error message.
Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=603921
Fixes "Unitialized Scalar Variable" issues reported by Coverity.
Has the added advantage of detecting whether somebody *does* use those
fields (ending up with a invalid address).
https://bugzilla.gnome.org/show_bug.cgi?id=720810
This must only ever be used in caps in combination with a non-system
memory GstCapsFeatures, and where it does not make sense to specify
any of the other video formats. Examples of this would be in gst-vaapi.
This reverts commit 5fcdabd907.
Instead of making it impossible to use the ENCODED format we should
just document that it must not be used for capsfeature-less caps.
Also this commit broke API/ABI.
GST_VIDEO_FORMAT_ENCODED was added to support *extracting* video-related
information (like width, height, framerate,...) from caps.
It is __NOT__ intended to be used as a format field on video/x-raw caps.
So that it avoids to send an allocation query twice.
One from an early call to gst_audio_encoder_negotiate from a
subclass, then one from gst_audio_encoder_allocate_output_buffer.
Which means that previously gst_audio_encoder_negotiate was not
clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success.
Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684
So that it avoids to send an allocation query twice.
One from an early call to gst_video_encoder_negotiate from a
subclass, then one from gst_video_encoder_allocate_output_frame.
Which means that previously gst_video_encoder_negotiate was not
clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success.
Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684
... so subclasses can release a frame all the way (also from frame list)
without having to pass through _finish_frame or _drop_frame.
The latter may not be applicable, or may or may not have already
been called for the frame in question.
See https://bugzilla.gnome.org/show_bug.cgi?id=693772
allows configuration of whether GstVideoGLTextureUploadMeta is
added to buffers resulting from a buffer pool. This is sperate
to the caps feature in that an element may want to add the upload
meta itself rather than allowing the buffer pool to.
https://bugzilla.gnome.org/show_bug.cgi?id=712798
Raise an error in case no frames are decoded before EOS and we
have input, meaning that data was received but it was somehow invalid.
Based on the videodecoder change, merged here for consistency.
https://bugzilla.gnome.org/show_bug.cgi?id=711094
Allows using -1 to make audiodecoder never post an error message
after decoding errors.
Based on the videodecoder change, merged here for consistency.
https://bugzilla.gnome.org/show_bug.cgi?id=711094
We could have allocation query before caps event and even without caps inside
the query. In such cases , the downstream can return a bufferpool object with
out actually configuring it. This feature is helpful to negotiate the bufferpool
with out knowing the output video format. For eg: some hardware accelerated
decoders can interpret the o/p video format only after it finishes the decoding
of one buffer at least.
https://bugzilla.gnome.org/show_bug.cgi?id=687183
Accumulate buffers in an adapter instead of appending them because append causes
a lot of memcpys.
Keep track of the last tagsize and accumulate enough data before attempting to
parse more data.
This patch implements a minimal amount of changes in order to not change the
behaviour. We should really rewrite the tag handling and trimming using
the adapter API instead of merging and trimming into a buffer.
Added new functions gst_rtsp_connection_set_tls_validation_flags() to
allow setting the TLS certificate validation flags when establishing a
TLS connection.
A getter is also available, gst_rtsp_connection_get_tls_validation_flags().
https://bugzilla.gnome.org/show_bug.cgi?id=711231
gst_audio_ring_buffer_set_channel_positions() checks whether the given
positions are identical with the current setup and returns
immediately if so. But it also clears need_reorder flag before this
comparison, thus this flag might be wrongly cleared if the function is
called twice with the same channel positions.
Move the flag clearance after the check.
https://bugzilla.gnome.org/show_bug.cgi?id=709754
We're checking the caps to see if we got more caps details after a parser got
plugged. This will also have a flipped 'parsed' field. If the field was already
present before the parse the match will fail. Add a function that will do the
check while excluding this field.
Creating a GSource and not attaching it to a context will cause
a leak of it's child sources. That is why we create writesrc right
before attaching it to a context.
https://bugzilla.gnome.org/show_bug.cgi?id=708667
Makes it easier to track how many users there are
Also make it possible to create a dmabuf struct on systems without mmap,
it just won't be possible to map it.
https://bugzilla.gnome.org/show_bug.cgi?id=707793
The payload type can't be between 72 and 76 because with the marker bit set,
this could be mistaken for an RTCP packet then. We do a relaxed check and
only refuse 72-76 when the marker bit is set. The effect is that when
we try to map an RTCP packet as an RTP packet, we will certainly fail.
id3mux and id3v2mux expect GST_TAG_ID3V2_FRAME type to be stored in a
GstSample and not a buffer, which is also needed because we can't
attach extradata/caps to buffers any more. These are private tags
no one should be poking at, and also the extra info is missing.
https://bugzilla.gnome.org/show_bug.cgi?id=707765
Mark terms such as "planar", "packed", and "palettized" as
translatable, and re-arrange strings a bit to make them
better suited for translation.
Also fix bug in yuv descriptions, one plane is packed, more
is planar (or semi-planar).
https://bugzilla.gnome.org/show_bug.cgi?id=707789
Windows Media Video Screen (WMV Screen) are video formats that
specilise in screencast content. This provides a correct media type
for them instead of just video/x-asf-unknown.
