Vincent Penquerc'h [Thu, 10 Apr 2014 14:55:57 +0000 (15:55 +0100)]
xvimage: remove dead code
matching_attr can not be NULL here, we've tested that away a few
lines beforehand.
Coverity 1139655
Vincent Penquerc'h [Thu, 10 Apr 2014 14:51:05 +0000 (15:51 +0100)]
videotestsrc: bail out on unsupported caps
This avoids using uninitialized data (and properly rejects caps).
Coverity 1139898
Vincent Penquerc'h [Thu, 10 Apr 2014 14:16:03 +0000 (15:16 +0100)]
typefind: remove pointless checks for data being NULL
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
Vincent Penquerc'h [Thu, 10 Apr 2014 12:34:58 +0000 (13:34 +0100)]
decodebin: consider "no demuxer" case to not have dynamic pads
This fixes a possible NULL dereference.
Coverity 1195146
Vincent Penquerc'h [Thu, 10 Apr 2014 12:28:30 +0000 (13:28 +0100)]
encodebin: guard against gst_pad_get_peer returning NULL
If it does, the pad may be leaked if it's a request pad, though.
Coverity 1139799
Vincent Penquerc'h [Thu, 10 Apr 2014 12:26:42 +0000 (13:26 +0100)]
encodebin: guard against pathological NULL dereference
Coverity 1139798
Vincent Penquerc'h [Thu, 10 Apr 2014 11:32:24 +0000 (12:32 +0100)]
audioresample: reject 0 denominator when creating resampler
Coverity 1195140, 1195139, 1195138
Vincent Penquerc'h [Thu, 10 Apr 2014 11:14:48 +0000 (12:14 +0100)]
video-overlay-composition: guard against NULL pointer dereference on error
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
Vincent Penquerc'h [Thu, 10 Apr 2014 11:10:47 +0000 (12:10 +0100)]
videometa: fix texture_type memcpy size
Coverity 1139589, 1139588
Vincent Penquerc'h [Thu, 10 Apr 2014 10:19:26 +0000 (11:19 +0100)]
sdpmessage: fix multi statement macros
Wasn't playing nice with an if statement below.
Coverity 1139767
Vincent Penquerc'h [Thu, 10 Apr 2014 10:14:25 +0000 (11:14 +0100)]
audiocdsrc: guard aginst overflow
An audio CD may contain about a tenth of the samples 32 bit can
represent, so it doesn't seem likely this will be hit in practice.
Coverity 1139805
Tim-Philipp Müller [Thu, 10 Apr 2014 11:30:50 +0000 (12:30 +0100)]
pbutils: descriptions: default to systemstream=false for partial video/mpeg caps
Assume systemstream=false for video/mpeg caps where that field
is missing.
Vincent Penquerc'h [Thu, 10 Apr 2014 09:57:53 +0000 (10:57 +0100)]
audiobasesink: avoid possible sample count overflow
At 48 kHz, 2<<31 samples is reached before 13 hours so it
sounds plausible this would be hit.
Coverity 1139800, 1139801
Vincent Penquerc'h [Thu, 10 Apr 2014 09:45:21 +0000 (10:45 +0100)]
theoraenc: fix comparison to unset timestamp
Also rejects negative timestamps that aren't GST_CLOCK_TIME_NONE.
Coverity 1139797
Vincent Penquerc'h [Thu, 10 Apr 2014 09:33:46 +0000 (10:33 +0100)]
oggstream: fix a few left shifts operations on 32 bits cast to 64 bits
This should not cause any actual bug since Theora and Daala have
a maximum shift of 31, and a packet duration of 2^31 seems very
implausible. But it fixes:
Coverity 1139804, 1139803, 1139802
Vincent Penquerc'h [Thu, 10 Apr 2014 09:29:34 +0000 (10:29 +0100)]
oggstream: remove NULL test after dereference
And add NULLness asserts at top of function. The only call
to this passes local variable pointers, so non NULL.
Coverity 206375
Vincent Penquerc'h [Thu, 10 Apr 2014 09:25:46 +0000 (10:25 +0100)]
oggmux: test for failure to return tag
It should really not happen unless the tag list it corrupt,
but the API returns a failure code so we may as well use it.
Coverity 1139595
Vincent Penquerc'h [Thu, 10 Apr 2014 09:22:43 +0000 (10:22 +0100)]
oggdemux: do not dereference NULL pad in warning message
Coverity 1197695
Sebastian Dröge [Thu, 10 Apr 2014 07:18:05 +0000 (09:18 +0200)]
video-event: Update the running times in the force-keyunit events from the pad offsets
Sebastian Dröge [Wed, 9 Apr 2014 14:03:15 +0000 (16:03 +0200)]
decodebin: In adaptive streaming mode, only have a fixed buffer limit for the non-buffering multiqueue
Wim Taymans [Tue, 8 Apr 2014 13:43:50 +0000 (15:43 +0200)]
sdp: guard against address parse errors.
Mathieu Duponchelle [Tue, 25 Mar 2014 16:11:34 +0000 (17:11 +0100)]
adder: rework the logic to check if eos has to be sent.
Checking the size available was incorrect, and the infos
for per-pad EOS are available.
Same logic as audiomixer.
fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727025
Josep Torra [Tue, 8 Apr 2014 10:46:21 +0000 (12:46 +0200)]
audioringbuffer: parse channels field from compressed audio caps
Also parse channels as an optional field in the caps for compressed
audio formats.
Jan Schmidt [Sun, 6 Apr 2014 12:26:20 +0000 (22:26 +1000)]
subtitleoverlay: Consider all caps for overlays, not just the first.
Check all supported caps on the overlay video pad, not just the
first of (possibly) many.
Tim-Philipp Müller [Sat, 5 Apr 2014 12:25:46 +0000 (13:25 +0100)]
tools: update gst-play-1.0 man page
Thiago Santos [Wed, 2 Apr 2014 10:20:43 +0000 (07:20 -0300)]
videodecoder: do not deactivate the bufferpool, just unref
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
Vincent Penquerc'h [Mon, 24 Feb 2014 16:17:05 +0000 (11:17 -0500)]
audiobasesink: clip start samples to match clipped start time
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.
Wim Taymans [Fri, 4 Apr 2014 15:36:04 +0000 (17:36 +0200)]
mikey: Fix the KEMAC payload
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.
Jan Schmidt [Thu, 3 Apr 2014 15:14:50 +0000 (02:14 +1100)]
playbin: Drop reference to any source element in NULL state
Drop the reference instead of waiting for either finalize(), or
for a new source when reused. Everyone else already forgot about
the old source.
Göran Jönsson [Tue, 1 Apr 2014 08:38:23 +0000 (10:38 +0200)]
rtspconnection: Added gst_rtsp_watch_set_flushing to list.
Added gst_rtsp_watch_set_flushing to list in file
libgstrtsp.def
Sebastian Dröge [Sun, 30 Mar 2014 16:26:59 +0000 (18:26 +0200)]
videodecoder: Always drain the decoder after a discont group in reverse playback mode
Sebastian Dröge [Sun, 30 Mar 2014 15:54:11 +0000 (17:54 +0200)]
videodecoder: Flush the decoder once per discont group, not once per keyframe
Sebastian Dröge [Sun, 30 Mar 2014 15:54:11 +0000 (17:54 +0200)]
videodecoder: Handle reverse playback with multiple GOPs per discont group properly
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.
Sebastian Dröge [Sat, 29 Mar 2014 09:23:05 +0000 (10:23 +0100)]
videodecoder: Log event types of events that are pushed downstream
Sebastian Dröge [Thu, 27 Mar 2014 19:15:01 +0000 (20:15 +0100)]
videodecoder: In reverse playback mode we need to finish the subclass after passing all frames to it
Wim Taymans [Fri, 28 Mar 2014 08:32:20 +0000 (09:32 +0100)]
rtspconnection: add flush method
Add a method to set/unset the flushing state that makes _wait_backlog()
unlock.
See https://bugzilla.gnome.org/show_bug.cgi?id=725898
Nicolas Dufresne [Thu, 27 Mar 2014 20:43:10 +0000 (16:43 -0400)]
ximagesink: only extrapolate alpha mask for 32-bit depth
Instead of passing bogus alpha mask values when there's no alpha.
https://bugzilla.gnome.org/show_bug.cgi?id=727188
Wim Taymans [Tue, 25 Mar 2014 10:14:51 +0000 (11:14 +0100)]
mikey: fix return values of g_return_*
Wim Taymans [Tue, 25 Mar 2014 10:07:34 +0000 (11:07 +0100)]
rtsptransport: UDP is also default for SAVP and AVPF
Wim Taymans [Thu, 20 Mar 2014 11:29:33 +0000 (12:29 +0100)]
docs: add MIKEY docs
Wim Taymans [Sat, 15 Mar 2014 17:46:52 +0000 (18:46 +0100)]
mikey: add MIKEY parsing helpers
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.
Ognyan Tonchev [Sun, 16 Mar 2014 16:04:44 +0000 (17:04 +0100)]
rtspconnection: Fix minor memory leaks in error handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
Ognyan Tonchev [Sun, 16 Mar 2014 16:06:02 +0000 (17:06 +0100)]
rtspconnection: Fix connection_poll()
* 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
Wim Taymans [Thu, 20 Mar 2014 08:18:31 +0000 (09:18 +0100)]
rtpbasepayload: add pt and ssrc to stats
Thiago Santos [Sun, 16 Mar 2014 11:34:30 +0000 (08:34 -0300)]
tests: decodebin: port old decodebin2 test for parser and decoder linking
They were in the old decodebin2.c tests file and were never ported.
Now we can get rid of decodebin2.c
Arun Raghavan [Sun, 16 Mar 2014 16:00:38 +0000 (17:00 +0100)]
playback: Add video-/audio-filter properties
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
Sebastian Dröge [Sun, 16 Mar 2014 17:38:25 +0000 (18:38 +0100)]
Revert "playback: Add video-/audio-filter properties"
This reverts commit
fb8fdedb4f4649aa33700bbc720131c1678df49f.
Arun Raghavan [Sat, 15 Mar 2014 15:05:22 +0000 (16:05 +0100)]
playback: Add video-/audio-filter properties
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
Руслан Ижбулатов [Sat, 15 Mar 2014 20:21:32 +0000 (20:21 +0000)]
rtspconnection: Silence a compiler warning
Cast the argument into (const char *) on W32, as winsock2 expects it.
https://bugzilla.gnome.org/show_bug.cgi?id=726433
Arun Raghavan [Sat, 15 Mar 2014 10:24:23 +0000 (11:24 +0100)]
playsink: Fix documentation for what the audio chain looks like
https://bugzilla.gnome.org/show_bug.cgi?id=679031
Tim-Philipp Müller [Tue, 11 Mar 2014 21:58:49 +0000 (21:58 +0000)]
docs: update plugin docs and remove old properties and signals
Re-generate .args and .signals file from scratch so that
old signals that no longer exist (such as the 'new-decoded-pad'
signal on decodebin) no longer show up in the documentation.
Stefan Sauer [Tue, 11 Mar 2014 21:15:13 +0000 (22:15 +0100)]
adder: set a group-id on the stream-start event
Set a default group-id to fix a warning printed by the sink.
Christian Fredrik Kalager Schaller [Tue, 11 Mar 2014 16:39:54 +0000 (17:39 +0100)]
Add new header file
Thiago Santos [Thu, 6 Mar 2014 15:59:08 +0000 (12:59 -0300)]
oggmux: implement vp8 granulepos function
Add an extra function to the oggstream map to inform it about
the incoming buffers. This way oggmux can keep a count on the
vp8 invisible frames and calculate the granulepos correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=722682
Thiago Santos [Wed, 5 Mar 2014 19:34:42 +0000 (16:34 -0300)]
oggmux: create vp8 header data if not provided in caps
vp8 stream header shouldn't be assumed to be provided in caps always
as this would repeat the same code in all demuxers/encoders. Instead,
make oggmux generate them if they are not supplied.
https://bugzilla.gnome.org/show_bug.cgi?id=722682
Göran Jönsson [Thu, 6 Mar 2014 12:55:17 +0000 (13:55 +0100)]
rtspconnection: gst_rtsp_watch_wait_backlog
New method that wait until there is room in backlog queue.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
David Svensson Fors [Thu, 6 Mar 2014 12:50:27 +0000 (13:50 +0100)]
rtspconnection: GstRTSPWatch func for tunnel GET response
Add a callback in GstRTSPWatch where the response to HTTP GET for
tunneled connections can be modified.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
Wim Taymans [Thu, 6 Mar 2014 14:34:47 +0000 (15:34 +0100)]
rtspdefs: add RFC 4567 headers and status code
This new Header and status code is used for SRTP
Sebastian Dröge [Fri, 7 Mar 2014 16:09:24 +0000 (17:09 +0100)]
decodebin: Buffer up to 5 seconds in multiqueue buffering mode
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.
Sebastian Dröge [Thu, 6 Mar 2014 21:37:44 +0000 (22:37 +0100)]
decodebin: Keep the number of buffers after an adaptive streaming demuxer lower
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.
Sebastian Dröge [Thu, 6 Mar 2014 21:28:46 +0000 (22:28 +0100)]
decodebin: Simplify adaptive streaming demuxer code a bit
Adrien Schwartzentruber [Thu, 6 Mar 2014 17:49:09 +0000 (17:49 +0000)]
pango: demote debug WARNING to LOG for variable framerate video input
No need why we need to warn about that, it's perfectly allowed.
https://bugzilla.gnome.org/show_bug.cgi?id=725837
Matthieu Bouron [Thu, 30 Jan 2014 15:41:49 +0000 (15:41 +0000)]
tests: add textoverlay passthrough with composition feature unit tests
https://bugzilla.gnome.org/show_bug.cgi?id=721953
Matthieu Bouron [Thu, 23 Jan 2014 12:20:05 +0000 (12:20 +0000)]
pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
https://bugzilla.gnome.org/show_bug.cgi?id=721953
Matthieu Bouron [Thu, 23 Jan 2014 12:19:13 +0000 (12:19 +0000)]
video-overlay-composition: add GST_CAPS_FEATURE_META_GST_VIDEO_OVERLAY_COMPOSITION
Andres Gomez [Tue, 4 Mar 2014 14:51:58 +0000 (16:51 +0200)]
docs: Removing GnomeVFS left bits
gnomevfs was removed time ago but there are still some left bits.
https://bugzilla.gnome.org/show_bug.cgi?id=725658
Tim-Philipp Müller [Wed, 5 Mar 2014 00:35:30 +0000 (00:35 +0000)]
typefindfunctions: lower H.263 typefinder max probability
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=700770
https://bugzilla.gnome.org/show_bug.cgi?id=725644
Ognyan Tonchev [Sun, 2 Mar 2014 10:58:58 +0000 (11:58 +0100)]
rtspconnection: Call closed() when GET is closed in tunneled mode
This patch adds read source on the write socket in tunneled
mode and we get a callback when client disconnects the GET
channel.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313
Sebastian Rasmussen [Sun, 2 Mar 2014 11:58:21 +0000 (12:58 +0100)]
videoformat: Remove duplicate/incorrect section
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
Sebastian Rasmussen [Sun, 2 Mar 2014 11:54:08 +0000 (12:54 +0100)]
docs: Add annotations for return values
Rephrase and clarify some return value descriptions
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
Sebastian Rasmussen [Sun, 2 Mar 2014 04:06:07 +0000 (05:06 +0100)]
docs: Fix argument and annotation typos
* colorbalance: Fix misspelled annotation
* rtsp: Replace incorrectly documented function argument
* sdp: Escape @ character to avoid gtk-doc warning
* video-*: Add missing annotation colon
* videodecoder/video-color: Fix function argument typos
* videoutils: Remove unknown annotation field
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
Sebastian Rasmussen [Sun, 2 Mar 2014 04:09:05 +0000 (05:09 +0100)]
.gitignore: Ignore gcov intermediate files
https://bugzilla.gnome.org/show_bug.cgi?id=725479
Sebastian Dröge [Fri, 28 Feb 2014 08:34:31 +0000 (09:34 +0100)]
Automatic update of common submodule
From fe1672e to bcb1518
Matthieu Bouron [Thu, 20 Feb 2014 20:01:30 +0000 (20:01 +0000)]
playbin: improve autoplug_query_caps return
Makes autoplug_query_caps return
downstream_caps + intersect_first(filter_caps, element_caps)
https://bugzilla.gnome.org/show_bug.cgi?id=724828
Stefan Sauer [Wed, 26 Feb 2014 21:11:01 +0000 (22:11 +0100)]
Automatic update of common submodule
From 1a07da9 to fe1672e
Tim-Philipp Müller [Wed, 26 Feb 2014 11:43:06 +0000 (11:43 +0000)]
rtsp: fix build with older GLib versions
The gio/gnetworking.h header is only available since glib 2.36
https://bugzilla.gnome.org/show_bug.cgi?id=725206
Ognyan Tonchev [Wed, 26 Feb 2014 10:45:24 +0000 (11:45 +0100)]
rtspconnection: Add missing include
https://bugzilla.gnome.org/show_bug.cgi?id=725206
Matthieu Bouron [Fri, 21 Feb 2014 14:01:37 +0000 (14:01 +0000)]
playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
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
Sebastian Rasmussen [Thu, 30 Jan 2014 23:06:18 +0000 (00:06 +0100)]
tests: Refactor RTP basepayloading test into pay/depay parts
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723328
Sebastian Rasmussen [Thu, 30 Jan 2014 23:19:16 +0000 (00:19 +0100)]
rtpbasepayload: Let caps event also configure seqnum-offset
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.
Sebastian Rasmussen [Thu, 30 Jan 2014 23:18:35 +0000 (00:18 +0100)]
rtpbasepayload: Fix payload type property boundary value
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.
Sebastian Rasmussen [Thu, 30 Jan 2014 23:06:30 +0000 (00:06 +0100)]
rtpbasedepayload: Fix typos in comments
Tim-Philipp Müller [Fri, 21 Feb 2014 19:28:55 +0000 (19:28 +0000)]
docs: add GstVideoPool to docs
Sebastian Dröge [Fri, 21 Feb 2014 08:53:09 +0000 (09:53 +0100)]
decodebin: If we have a demuxer without dynamic srcpads, just assume no-more-pads
Otherwise we will wait until the multiqueue after the demuxer will
overrun, which is clearly not needed then.
Sebastian Dröge [Fri, 21 Feb 2014 08:43:38 +0000 (09:43 +0100)]
decodebin: Also make sure to not duplicate an element factory after a group
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.
Sebastian Dröge [Thu, 20 Feb 2014 14:38:48 +0000 (15:38 +0100)]
decodebin: During pre-rolling always use the auto-preroll limits on multiqueues
Even if we're buffering in the multiqueues.
Sebastian Dröge [Thu, 20 Feb 2014 14:37:54 +0000 (15:37 +0100)]
decodebin: Pass through the seekability information when setting multiqueue limits
Sebastian Dröge [Thu, 20 Feb 2014 14:36:47 +0000 (15:36 +0100)]
decodebin: During exposing of pads don't set the multiqueue limits multiple times to different values
Instead just set them once in the very end to the correct values.
Sebastian Dröge [Thu, 20 Feb 2014 14:07:26 +0000 (15:07 +0100)]
decodebin: Only enable multiqueue buffering once we're pre-rolled
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.
Sebastian Dröge [Thu, 20 Feb 2014 14:02:09 +0000 (15:02 +0100)]
decodebin: Make sure that we always have a second multiqueue for adaptive streaming demuxers
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.
Sebastian Dröge [Wed, 19 Feb 2014 09:21:16 +0000 (10:21 +0100)]
uridecodebin: Always use buffering in multiqueue for adaptive streams
Sebastian Dröge [Wed, 19 Feb 2014 09:06:13 +0000 (10:06 +0100)]
uridecodebin: Only add a queue2 for buffering for non-adaptive streaming streams
Thiago Santos [Wed, 6 Feb 2013 11:46:58 +0000 (08:46 -0300)]
uridecodebin: pass on the buffering property for adaptive streams
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
Tim-Philipp Müller [Fri, 21 Feb 2014 19:07:59 +0000 (19:07 +0000)]
tests: add new unit tests to .gitignore
Ognyan Tonchev [Wed, 19 Feb 2014 12:54:17 +0000 (13:54 +0100)]
rtspconnection: New unit test
See https://bugzilla.gnome.org/show_bug.cgi?id=724720
Ognyan Tonchev [Wed, 19 Feb 2014 12:53:06 +0000 (13:53 +0100)]
rtspconnection: Remove read child source when POST is disconnected
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720
Aleix Conchillo Flaqué [Thu, 20 Feb 2014 00:10:25 +0000 (16:10 -0800)]
defs: update for new rtspconnection symbols
Thiago Santos [Wed, 19 Feb 2014 04:55:50 +0000 (01:55 -0300)]
oggdemux: allow file to go until the end in push mode
When seeking back to original state after duration seeks, let
upstream know that we want the whole file, including the last
byte that wasn't requested on the duration seeks.
https://bugzilla.gnome.org/show_bug.cgi?id=724633
Thiago Santos [Thu, 20 Feb 2014 02:54:59 +0000 (23:54 -0300)]
oggdemux: remove unused instance variable event
It is never set to anything
Aleix Conchillo Flaqué [Mon, 17 Feb 2014 01:39:35 +0000 (17:39 -0800)]
rtspconnection: allow specifying a certificate database
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