Commit graph

13635 commits

Author SHA1 Message Date
Tim-Philipp Müller
8f8f1f9de1 tests: fix vararg handling in rtpbasedepayload unit test
Makes it pass on 32-bit systems.
2014-06-23 01:02:22 +01:00
Tim-Philipp Müller
ec6b178339 tests: fix vararg handling in rtpbasepayload unit test
Makes it pass on 32-bit systems.
2014-06-23 00:33:18 +01:00
Sebastian Dröge
9d33b38969 playsinkconvertbin: Filter out ANY capsfeatures from the converter caps
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.
2014-06-22 20:42:43 +02:00
Sebastian Dröge
78317be45d Back to development 2014-06-22 19:36:14 +02:00
Sebastian Dröge
988f53ed18 Release 1.3.3 2014-06-22 18:07:57 +02:00
Sebastian Dröge
cc429be8eb Update .po files 2014-06-22 17:25:42 +02:00
Sebastian Dröge
aabfca91d2 po: Update translations 2014-06-22 14:23:32 +02:00
Sebastian Dröge
909dd7831b audiodecoder: Don't be too picky about the output frame counter
With most decoder libraries, and especially when accessing codecs via
OpenMAX or similar APIs, we don't have the ability to properly related
the output buffers to a number of input samples. And could e.g. get
a fractional number of input buffers decoded at a time.

Previously this would in the end lead to an error message and stopped
playback. Change it to a warning message instead and try to handle it
gracefully. In theory the subclass can now get timestamp tracking
wrong if it completely misuses the API, but if on average it behaves
correct (and gst-omx and others do) it will continue to work properly.

Also add a test for the new behaviour.

We don't change it in the encoder yet as that requires more internal logic
changes AFAIU and I'm not aware of a case where this was a problem so far.
2014-06-20 11:02:55 +02:00
Michael Olbrich
5ebfe5b26b tcpserversrc: close the server socket after accepting a connection
g_socket_accept() is only called once for a server socket. So
keeping the socket open ist just confusing possible clients.

https://bugzilla.gnome.org/show_bug.cgi?id=731566
2014-06-13 11:18:10 +01:00
Tim-Philipp Müller
79cf59b7dd tcpclientsrc: return FLUSHING when select() is canceled
https://bugzilla.gnome.org/show_bug.cgi?id=731567
2014-06-13 10:04:47 +01:00
Michael Olbrich
593a52a656 tcpserversrc: return FLOW_FLUSHING instead of an error when accept/select is canceled
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
2014-06-13 09:19:49 +01:00
Edward Hervey
072fa3543e mikey: Fix Wall to NTP conversion
We are scaling from a unit in microseconds to a unit in ((1 << 32) per seconds).

We therefore scale the microseconds values by:

         value of a second in the target unit (1 << 32)
  --------------------------------------------------------------
  value of a second in the origin format (1 000 000 microsecond)
2014-06-12 11:55:59 +02:00
Vincent Penquerc'h
c1b42827c1 oggdemux: allow unset seek stop time in push mode 2014-06-11 17:59:47 +01:00
Tim-Philipp Müller
55b4d8f6af docs: add streamsynchronizer to documentation 2014-06-11 12:50:23 +01:00
Tim-Philipp Müller
c5a790b178 docs: add playsink element to documentation 2014-06-11 12:43:35 +01:00
Tim-Philipp Müller
adf02f83c5 docs: add navigation interface to docs 2014-06-11 10:53:50 +01:00
Thiago Santos
bbf226d9d6 appsrc: add send_event handler for flushing
Adds a send_event handling for allowing appsrc to flush its internal
data, allowing users to flush the pipeline without setting it to null.

https://bugzilla.gnome.org/show_bug.cgi?id=724231
2014-06-10 12:59:53 -03:00
Thiago Santos
3202938369 videoscale: vs_image: strides are a gsize
The strides that are set from the GstVideoInfo structs are
a gsize. Using an int can cause overflows when dealing with large
enough images

https://bugzilla.gnome.org/show_bug.cgi?id=731195
2014-06-09 23:38:39 -03:00
Thiago Santos
fb3a9d1bc5 video: avoid overflows when doing int operations for size
size is a gsize, so cast the operands to it to avoid overflows
and setting wrong value to the video size.

Includes tests.

https://bugzilla.gnome.org/show_bug.cgi?id=731195
2014-06-09 23:38:39 -03:00
Edward Hervey
8810ada4cd theoraenc: Remove unneeded check
running timestamps are guaranteed to be positive and valid since the
GstVideoEncoder base class will clip incoming buffers

CID #1139797
2014-06-09 10:53:03 +02:00
Edward Hervey
23996adf5c vorbisenc: add missing va_end in variadic function
Coverity 1139944
2014-06-09 10:38:53 +02:00
Vincent Penquerc'h
0a49b7050f tests: fix uninitialized variable use in video decoder test 2014-06-06 10:42:11 +01:00
Sebastian Dröge
1f41a1606a uridecodebin: Also catch CODEC_NOT_FOUND errors and delay them until all decodebins are done 2014-06-05 15:35:31 +02:00
Sebastian Dröge
0f43e801f2 uridecodebin: Ignore missing-plugin messages unless all decodebins post one
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
2014-06-04 17:00:43 +02:00
Sebastian Dröge
393f090197 decodebin: Do async-done on expose errors too 2014-06-04 17:00:43 +02:00
Michael Olbrich
42623c995e dmabuf: fix checking mmap flags
A simple '&' is not sufficiant. With mmapping_flags == PROT_READ and
prot == PROT_READ|PROT_WRITE the check produces the wrong result.
Change the check to make sure that prot is a subset of mmapping_flags.

https://bugzilla.gnome.org/show_bug.cgi?id=730559
2014-06-04 10:01:27 -04:00
Vincent Penquerc'h
b444d8ba97 alsasink: make gst-ident happy 2014-06-03 15:17:20 +01:00
Vincent Penquerc'h
3b2d583373 alsasink: fix occasional crash intersecting invalid values
When a pipeline using alsasink and push mode upstream fails
to preroll, the following state will be the case:
- A loop upstream will be PAUSED, pushing a first buffer
- alsasink will be READY, pending PAUSED, because async

On error, the pipeline will switch to NULL. alsasink is in
READY, so goes to NULL immediately. It zeroes its cached
caps. Meanwhile, the upstream loop can cause a caps query,
conccurent with the state change. This will use those cached
caps. If the zeroing happens between the NULL test and the
dereferencing, GStreamer will critical down in the GstValue
code.

Since it appears that such a gap between states (PAUSED
and pushing upstream, and NULL downstream) is expected, we
need to protect the read/write access to the cached caps.

This fixes the critical.

See https://bugzilla.gnome.org/show_bug.cgi?id=731121
2014-06-03 15:17:20 +01:00
Thibault Saunier
d2ea3260e9 videodecoder: Keep still meaningfull pending events on FLUSH_STOP
Only EOS and segment should be deleted in that case.

+ Add a testcase

https://bugzilla.gnome.org/show_bug.cgi?id=709868
2014-06-03 13:03:49 +02:00
Thibault Saunier
12df7fa49d audiodecoder: Keep still meaningfull pending events on FLUSH_STOP
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
2014-06-03 13:03:21 +02:00
Thibault Saunier
2843f358d9 videoencoder: Keep still meaningfull pending events on FLUSH_STOP
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
2014-06-03 13:03:19 +02:00
Thibault Saunier
42ecafec0c streamsplitter: Keep still meaningfull pending events on FLUSH_STOP
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
2014-06-03 13:03:18 +02:00
Thibault Saunier
967d1fb982 audioencoder: Keep still meaningfull pending events on FLUSH_STOP
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
2014-06-03 13:03:16 +02:00
Vincent Penquerc'h
716987063c oggstream: consider all opus packets as "keyframes"
This lets oggdemux determine they are not delta units, and removes
spurious per packet warnings about being unable to determine the
packet's keyframeness.
2014-06-02 12:40:27 +01:00
Edward Hervey
464611acc4 mikey: Free MikeyPayload in error cases
CID #1212136
2014-06-02 09:54:54 +02:00
Thiago Santos
783195ccef decodebin: aggregate buffering messages
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
2014-05-29 18:59:30 -03:00
Philip Withnall
ba87655628 audio: Add a missing precondition to gst_audio_format_from_string()
https://bugzilla.gnome.org/show_bug.cgi?id=730874
2014-05-28 11:34:01 +02:00
Thiago Santos
5d35675a74 tests: videodecoder: audiodecoder: add tests for eos after segment
Tests that pushing a buffer after the segment returns EOS
2014-05-26 23:24:56 -03:00
Thiago Santos
0cb5ea4396 videodecoder: actually return the push result in backwards playback
It was always returning _OK regardless of what downstream returned
2014-05-26 23:24:56 -03:00
Thiago Santos
ff9e37ea66 videodecoder: return EOS when segment is over
if a buffer is clipped by being completely out of segment, check if this
buffer is after the end of the segment and return EOS upstream

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:26:45 -03:00
Thiago Santos
09b8f902ea audiodecoder: return EOS when segment is over
if a buffer is clipped by being completely out of segment, check if this
buffer is after the end of the segment and return EOS upstream

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:26:45 -03:00
Thiago Santos
d3ca7271dc oggdemux: use new gstutils helper GstFlowCombiner
Fixes the handling of GST_FLOW_EOS by using the helper object
from gstutils that does the correct combination of flow returns.

https://bugzilla.gnome.org/show_bug.cgi?id=709224
2014-05-26 19:26:43 -03:00
Tim-Philipp Müller
64b4b2a20a tools: play: use cubic volume factor when adjusting volume
This is more natural and better-suited for a playback application.
2014-05-23 19:21:35 +01:00
Sebastian Dröge
cf4c70265b Back to development 2014-05-21 13:23:24 +02:00
Sebastian Dröge
1917400691 Release 1.3.2 2014-05-21 13:06:34 +02:00
Sebastian Dröge
ab756a6844 Update .po files 2014-05-21 12:01:15 +02:00
Sebastian Dröge
0b36e6960b Automatic update of common submodule
From 211fa5f to 1f5d3c3
2014-05-21 10:50:56 +02:00
Sebastian Dröge
7bdaa49fa3 video: And check comparison for real 2014-05-21 10:43:49 +02:00
Sebastian Dröge
f8b367212d video: Fix broken comparison in unit test
libs/video.c:540:50: error: comparison of constant 2 with boolean expression is always false
      [-Werror,-Wtautological-constant-out-of-range-compare]
            && !GST_VIDEO_INFO_N_PLANES (&vinfo) > 2) {
               ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^ ~
2014-05-21 10:40:32 +02:00
Wim Taymans
9a20920aa4 rtsp-transport: clarify port usage
Comment in the docs what the client_port and server_port fields are used
for in TCP mode (if the application wants to set those values).
2014-05-20 16:01:08 +02:00