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
Aleix Conchillo Flaqué [Mon, 17 Feb 2014 07:55:17 +0000 (23:55 -0800)]
rtspconnection: get rid of superfluous whitespaces
Stefan Sauer [Tue, 18 Feb 2014 19:48:57 +0000 (20:48 +0100)]
encodebin: simplify tests
Also use the profile helper for the ogg profile here.
Nicolas Dufresne [Tue, 18 Feb 2014 18:08:09 +0000 (13:08 -0500)]
video: Fix NV12_64Z32 default offset and size
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.
Sebastian Dröge [Tue, 18 Feb 2014 14:02:57 +0000 (15:02 +0100)]
playbin: Keep inputselector around until we release its pads
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
Tim-Philipp Müller [Tue, 18 Feb 2014 10:42:04 +0000 (10:42 +0000)]
audioconvert: never do mixing for 1->1 channel conversions
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
Rafał Mużyło [Tue, 18 Feb 2014 10:32:46 +0000 (10:32 +0000)]
audio: map channels=1,channel-mask=0 to MONO instead of NONE
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
Stefan Sauer [Sun, 16 Feb 2014 20:24:29 +0000 (21:24 +0100)]
encodebin: refactor tests
Add a new test to demo how to get missing plugin message.
Split some tests that unneccesarily munge unrelated checks into one test.
Sebastian Dröge [Sun, 16 Feb 2014 14:32:47 +0000 (15:32 +0100)]
playsink: Only remove the complete text chain if the text pad goes away
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
Tim-Philipp Müller [Fri, 14 Feb 2014 20:16:04 +0000 (20:16 +0000)]
tools: gst-play: add volume control
Thiago Santos [Thu, 13 Feb 2014 19:03:01 +0000 (16:03 -0300)]
oggmux: properly flush when seeking at the beginning
Reset all internal status when collect pads forwards a flush-stop
from the pads to be able to start the stream again.
Sebastian Dröge [Wed, 12 Feb 2014 16:34:32 +0000 (17:34 +0100)]
uridecodebin: Don't leak pad references
Sebastian Rasmussen [Sun, 2 Feb 2014 22:59:36 +0000 (23:59 +0100)]
tests: Don't build disabled plugins' check tests
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723492
Sebastian Dröge [Tue, 11 Feb 2014 15:35:45 +0000 (16:35 +0100)]
playbin: First try to get the pad's current caps, then query caps
The caps query might give us ANY caps while the pad has fixed caps
configured currently.
Sebastian Dröge [Mon, 10 Feb 2014 15:33:50 +0000 (16:33 +0100)]
playbin: Fix memory leak in autoplugging code
We should not leak element factories ideally.
Sebastian Dröge [Mon, 10 Feb 2014 15:33:35 +0000 (16:33 +0100)]
playbin: Fix memory leak in unit test
Sebastian Dröge [Sun, 9 Feb 2014 22:17:03 +0000 (23:17 +0100)]
subtitleoverlay: Remove unused function
Sebastian Dröge [Sun, 9 Feb 2014 10:28:48 +0000 (11:28 +0100)]
audiosrc: Fix typo in docs
We read *from* the audio device, not to it.
Sebastian Dröge [Sat, 8 Feb 2014 16:11:54 +0000 (17:11 +0100)]
videoscale: Fix compiler warning in unit test
error: implicit conversion from enumeration type
'GstFormat' to different enumeration type 'GstVideoFormat'
Sebastian Dröge [Sat, 8 Feb 2014 16:11:04 +0000 (17:11 +0100)]
videoconvert: Fix compiler warning in unit test
error: implicit conversion from enumeration type
'GstFormat' to different enumeration type 'GstVideoFormat'
Sebastian Dröge [Sat, 8 Feb 2014 16:07:15 +0000 (17:07 +0100)]
playback-test: Fix types for comparisons
Storing a 64 bit integer in a 32 bit integer and then checking
for the error cases might not be ideal.
error: comparison of constant -
9223372036854775808 with
expression of type 'guint' (aka 'unsigned int') is always true
Sebastian Dröge [Sat, 8 Feb 2014 16:02:27 +0000 (17:02 +0100)]
oggmux: Fix typo in header include guard
clang does not like this.
Sebastian Dröge [Sat, 8 Feb 2014 16:01:38 +0000 (17:01 +0100)]
alsa: Make clang happy with our g_strdup_vprintf() wrapper
Wim Taymans [Fri, 7 Feb 2014 14:33:34 +0000 (15:33 +0100)]
playback-test: allow seeking outside of the range
For download buffer, allow seeking outside of the already downloaded
area.
Thiago Santos [Fri, 7 Feb 2014 05:09:10 +0000 (02:09 -0300)]
basetextoverlay: use correct segment for text
video time uses the 'segment' and the text time should use
the 'text_segment'.
If different segments are used for video and text it would
lead to out of sync video/subtitles.
Wim Taymans [Tue, 4 Feb 2014 13:31:29 +0000 (14:31 +0100)]
check: add some more checks
Add header and payload length check in case of CSRCs.
See https://bugzilla.gnome.org/show_bug.cgi?id=723196
Sebastian Rasmussen [Mon, 3 Feb 2014 01:35:57 +0000 (02:35 +0100)]
jsseek: Add missing HAVE_X check
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723507
Eric Trousset [Tue, 4 Feb 2014 12:55:49 +0000 (13:55 +0100)]
tagdemux: Forward TIME seeks upstream too, maybe upstream can handle that
https://bugzilla.gnome.org/show_bug.cgi?id=723597
Stefan Sauer [Fri, 31 Jan 2014 22:27:03 +0000 (23:27 +0100)]
docs: doc fixes for audio library
Add sections docs for audiometa. Fix sections docs for audiochannels. Remove old
mixerutil section.
Julien Isorce [Fri, 31 Jan 2014 13:40:36 +0000 (13:40 +0000)]
videotestsrc: ensure having caps when setting the buffer pool config
It happens if downstream does not propose a buffer pool.
GST_DEBUG=2 gst-launch-1.0 videotestsrc ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=723271
Sebastian Dröge [Thu, 30 Jan 2014 20:18:04 +0000 (21:18 +0100)]
gst-play: Support non-ASCII tags
By calling setlocale() to get us multi-byte/UTF-8 support.
https://bugzilla.gnome.org/show_bug.cgi?id=723164
Bastien Nocera [Tue, 28 Jan 2014 13:28:27 +0000 (14:28 +0100)]
gst-discoverer: Support non-ASCII tags
By calling setlocale() to get us multi-byte/UTF-8 support.
https://bugzilla.gnome.org/show_bug.cgi?id=723164
Edward Hervey [Thu, 30 Jan 2014 09:43:48 +0000 (10:43 +0100)]
Automatic update of common submodule
From d48bed3 to 1a07da9
Thiago Santos [Wed, 29 Jan 2014 16:58:07 +0000 (13:58 -0300)]
streamsplitter: push pending events before eos
Push any pending events downstream before pushing eos
Thiago Santos [Wed, 29 Jan 2014 15:33:21 +0000 (12:33 -0300)]
tests: audioencoder: add tests analogous to the videoencoder ones
Thiago Santos [Wed, 29 Jan 2014 15:32:16 +0000 (12:32 -0300)]
audioencoder: push pending events and tags before EOS
if there are tags or events pending and an EOS is received, push those
events and tags before the EOS.
Thiago Santos [Tue, 28 Jan 2014 18:25:05 +0000 (15:25 -0300)]
tests: videoencoder: check that tags are pushed before eos
Check that if a new tag event is received right before eos it
is pushed before the eos
Thiago Santos [Tue, 28 Jan 2014 18:30:35 +0000 (15:30 -0300)]
videoencoder: push tags and events before eos
if any tags or events are pending, push them before pushing eos
Thiago Santos [Tue, 28 Jan 2014 18:06:39 +0000 (15:06 -0300)]
tests: videoencoder: basic videoencoder base class test
Adds a single test for video encoding
Sebastian Rasmussen [Tue, 26 Nov 2013 00:13:45 +0000 (01:13 +0100)]
rtpbasepayload: Do cosmetic changes to rtptime calculations
* 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
Sebastian Rasmussen [Mon, 27 Jan 2014 23:40:38 +0000 (00:40 +0100)]
rtpbasepayload: Expose running-time of payloaded stream
https://bugzilla.gnome.org/show_bug.cgi?id=719415
Sebastian Rasmussen [Wed, 22 Jan 2014 16:47:02 +0000 (17:47 +0100)]
rtpbasepayload: Improve documentation for perfect-rtptime
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
Sebastian Rasmussen [Thu, 16 Jan 2014 15:58:43 +0000 (16:58 +0100)]
rtpbasepayload: Fix typos in documentation for properties
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
Alessandro Decina [Mon, 27 Jan 2014 13:19:07 +0000 (00:19 +1100)]
decodebin: make it possible to register multiple handlers for autoplug-select
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
Sebastian Rasmussen [Thu, 16 Jan 2014 20:49:59 +0000 (21:49 +0100)]
rtpbasepayload: Add statistics property
This property allows for an atomically retrieved set of properties that
can e.g. be used to generate RTP-Info headers.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719415
Sjoerd Simons [Fri, 26 Jul 2013 13:44:28 +0000 (15:44 +0200)]
uridecodebin: Drop hardcoded list of media suitable for download buffering
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
Wim Taymans [Fri, 24 Jan 2014 11:19:43 +0000 (12:19 +0100)]
videoconvert: tweak the scoring algorithm
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
Wim Taymans [Thu, 23 Jan 2014 09:45:00 +0000 (10:45 +0100)]
video-chroma: don't crash on NULL resamplers
Make dummy resamplers for all cases and only execute the horizontal
resampler instead of crashing.
See https://bugzilla.gnome.org/show_bug.cgi?id=722742
Wim Taymans [Tue, 21 Jan 2014 10:21:56 +0000 (11:21 +0100)]
audiobasesink: make _get_time more threadsafe
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
Sebastian Dröge [Mon, 20 Jan 2014 15:11:04 +0000 (16:11 +0100)]
audioresample: It's HAVE_EMMINTRIN_H, not HAVE_XMMINTRIN_H for SSE2
Antoine Jacoutot [Mon, 20 Jan 2014 14:44:09 +0000 (15:44 +0100)]
audioresample: Fix build on x86 if emmintrin.h is available but can't be used
On i386, EMMINTRIN is defined but not usable without SSE so check for
__SSE__ and __SSE2__ as well.
https://bugzilla.gnome.org/show_bug.cgi?id=670690