Commit graph

8581 commits

Author SHA1 Message Date
Stefan Sauer
3d23ceebae monoscope: update README
We can already create multiple instances.
2016-02-12 21:01:03 +01:00
Stefan Sauer
daea0540fd monoscope: code cleanup
Use constants more often. Cleanup comments and add more to explain how things
work.
2016-02-12 21:01:03 +01:00
Luis de Bethencourt
3738ce8ba1 deinterlace: remove check for impossible condition
Commit bd27a1f30b added a few error handling
memory management checks. These check srccaps to see if it needs to be
unreferenced before returning, in the case of invalid_caps this goto jump
always happens before srccaps is set, so it will always be NULL in this
error label.

CID #1352035
2016-02-08 23:48:28 +00:00
Tim-Philipp Müller
f301e3f236 matroska: get rid of _stdint.h include 2016-02-08 00:11:55 +00:00
Sebastian Dröge
e244b9be87 rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
For APP/JPG markers the size is following and we have to skip that. This is
not really a problem unless the marker contains e.g. a preview JPEG or
something else that we might interprete as another marker.
2016-01-31 11:05:05 +11:00
Seungha Yang
7873bede31 qtdemux: fix framerate calculation for fragmented format
qtdemux calculates framerate using duration and the number of sample.
In case of fragmented mp4 format, however, the number of sample can
be figure out after parsing every moof box. Because qtdemux does not
parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
framerate calculation.

This patch will triger gst_qtdemux_configure_stream() for every new moof.
Then, framerate will be calculated by using duration and n_samples of the moof.

https://bugzilla.gnome.org/show_bug.cgi?id=760774
2016-01-29 11:01:44 +01:00
Seungha Yang
0391a93a35 qtdemux: handling zero segment-duration edit list
Based on document ISO_IEC_14496-12, edit list box can have
segment duration as zero. It does not imply that media_start equals to
media_stop. But, it just indicates a sample which should be presented
at the first. This patch derives segment duration using media_time
and duration of file. And set derived duration to segment-duration.

https://bugzilla.gnome.org/show_bug.cgi?id=760781
2016-01-29 10:57:05 +01:00
Seungha Yang
d8bb6687ea qtdemux: expose streams with first moof for fragmented format
In case of push mode, qtdemux expose streams after got moov box.
We can not guarantee that a moov box has sample data such as sample duration
and the number of sample in stbl box for fragmented format case.
So, if a moov has no sample data, streams will not be exposed until get the first moof.

https://bugzilla.gnome.org/show_bug.cgi?id=760779
2016-01-29 10:53:39 +01:00
Sebastian Dröge
3edf0737d6 deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS 2016-01-27 18:48:17 +01:00
Sebastian Dröge
c7d90c1112 deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
Prevents double-negotiation during startup and in some other cases.
2016-01-27 18:44:23 +01:00
Vivia Nikolaidou
bd27a1f30b deinterlace: Do passthrough in auto mode if downstream only supports interlaced
If the following conditions are met:
1) upstream and downstream caps are compatible
2) upstream is interlaced
3) downstream doesn't support progressive mode
then deinterlace will just do passthrough instead of failing to link.

This is done with the following scenario in mind:

videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
queue ! deinterlace name=dein_desktop ! autovideosink
In this case, dein_src will do the deinterlacing. However,

videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
"video/x-raw,interlace-mode=interleaved" ! fakesink

In this case, caps auto-negotiation will make dein_file and dein_desktop do
the deinterlacing, while dein_src will be passthrough.

https://bugzilla.gnome.org/show_bug.cgi?id=760995
2016-01-27 16:45:29 +01:00
Sebastian Dröge
46735f8de9 deinterlace: Add mode=auto-strict
In this mode we will passthrough all progressive caps but interlaced caps must be
caps where we actually support deinterlacing.

This is the only difference between auto and auto-strict, auto would
passthrough all unsupported interlaced caps.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
2016-01-27 16:45:29 +01:00
Sebastian Dröge
2e8d4e8c7a deinterlace: Implement reconfiguration a bit better
And e.g. consider reconfiguration caused by RECONFIGURE events too.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
2016-01-27 16:45:29 +01:00
Sebastian Dröge
8c1c091439 deinterlace: Rewrite caps negotiation
Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
of caps were last set, and e.g. if we last had interlaced caps or not. That's
just broken.

Also previously the handling of non-sysmem caps features was rather random and
unusuable.

Now the behaviour is the following, depending on the mode property:
1) mode=disabled
  Completely do passthrough of everything
2) mode=interlaced
  Only accept formats we can actually deinterlace, and accept interlaced
  and progressive content and always run the deinterlacer and output
  progressive content
3) mode=auto (i.e. playbin)
  Accept all progressive formats as passthrough, accept all formats that we
  can deinterlace ourselves (which we do then), but also accept everything
  else for which we then just passthrough. In auto mode, deinterlacing is best
  effort: If we can, we deinterlace, if we can't we just output interlaced
  content.

https://bugzilla.gnome.org/show_bug.cgi?id=720388
https://bugzilla.gnome.org/show_bug.cgi?id=760553
2016-01-27 16:45:29 +01:00
Sebastian Dröge
1053af6d0c deinterlace: Remove unused, obsolete bufferalloc code 2016-01-27 16:45:29 +01:00
Matej Knopp
e7460d9c06 matroskamux: use A_AAC instead of A_AAC/MPEGx/y
Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete

https://bugzilla.gnome.org/show_bug.cgi?id=761144
2016-01-27 13:50:21 +01:00
Víctor Manuel Jáquez Leal
e1834d1512 gst: Fix unintialized variable warnings
While cross-compiling with Linaro GCC 5.1-2015.08, it complained
about a couple unitialized variables.

This patch initializes them to zero.

https://bugzilla.gnome.org/show_bug.cgi?id=761094
2016-01-27 13:46:07 +01:00
George Kiagiadakis
eafa9f08f7 splitmuxsrc: print potentially negative offset with a sign 2016-01-25 15:36:29 +01:00
Tim-Philipp Müller
5d14746792 taginject: fix sample pipeline in docs
https://bugzilla.gnome.org/show_bug.cgi?id=679571
2016-01-21 15:30:42 +00:00
Tim-Philipp Müller
aeed2e550c rtp: fix compiler warnings with gcc-6
In file included from gstrtpL16depay.h:27:0,
                 from gstrtp.c:73:
gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
 static const GstRTPChannelOrder channel_orders[] =
2016-01-19 13:04:39 +00:00
Sebastian Dröge
7927f49ca0 wavparse: Don't play anything after the end of the data chunk even when seeking
Especially in push mode we would completely ignore the size of the data chunk
when not stop position is given for the seek. Instead make sure that the end
offset is at most the end of the data chunk if known.

Without this we would output anything after the data chunk, possibly causing
loud noises if the media file is followed by an INFO chunk or an ID3 tag.
2016-01-19 14:57:03 +02:00
Sebastian Dröge
322bdf5136 wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
We use that to signal "infinity", taking the difference between that and some
other value is not going to give us any useful result for the end offsets of
segments.
2016-01-19 14:55:57 +02:00
Sebastian Dröge
366bbffcd8 Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
This reverts commit 271501f657.

It wasn't meant to be pushed yet as the commit message indicates.
2016-01-18 11:30:45 +02:00
Aleix Conchillo Flaqué
665d14a2a0 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
We check the stream profile and use the proper RTCP caps:
application/x-srtcp if we are using a secure profile and
application/x-rtcp otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=760556
2016-01-18 11:29:25 +02:00
Sebastian Dröge
271501f657 WIP: rtpjitterbuffer: Add RFC7273 media clock handling 2016-01-18 08:58:59 +02:00
Sebastian Dröge
53c797d604 wavparse: When flushing on EOS, don't process more data than the "data" size
Even if we have more data queued up when flushing than the size of the data
chunk, don't process and output it. If the data size is known, this likely
contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
outputting them as if they were data is going to cause unexpected behaviour
and unpleasant audio noises.
2016-01-13 23:42:31 +01:00
Antonio Ospite
bdcc0390af interleave: Fix the example by setting channel-masks in the sink pads
The current example does not work, it fails with:

ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
streaming task paused, reason not-negotiated (-4)

This is because negotiation with wavenc gets messed up by the missing
channel positions configuration.

The proper way to define the channel layout when using the interleave
element in code would be to set the channel-positions property, but
gst-launch-1.0 does not know how to deal with arrays; so the example
pipeline works around the issue by setting the channel-masks in the sink
pads.

Also fix a repetition in the deinterleave example description

https://bugzilla.gnome.org/show_bug.cgi?id=735673
2016-01-12 22:11:30 +00:00
Tim Sheridan
205565ccd9 sbcparse: Fix frame length calculation
SBC frame length calculation wasn't being rounded up to the nearest byte
(as specified in the A2DP 1.0 specification, section 12.9). This could
cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
calculated frame lengths.

Incorrect frame length calculation causes frame coalescing to fail, as
subsequent frames in the stream aren't found in the expected locations.

https://bugzilla.gnome.org/show_bug.cgi?id=742446
2016-01-12 21:52:12 +00:00
Reynaldo H. Verdejo Pinochet
0bb8000874 flacparse: demote warning on wrong reserved value to fixme
We are likely just parsing a backward-compatible stream we
don't fully support.
2016-01-10 22:54:12 -08:00
Thiago Santos
4ac0a49308 imagefreeze: simplify caps selection
The downstream caps query with a filter alraedy gives us the possible
intersection so there is no need to check it again with downstream
if it is supported. Just try to set it directly.
2016-01-08 16:29:29 -03:00
Tim-Philipp Müller
3aa0dd8629 rtph264depay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
6171b0a675 rtpdvdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
c75f94c8f5 rtpamrdepay: fix unnecessary sub-buffer creation
We create a sub-buffer just to copy over its metas and then
throw it away immediately, just use the original input buffer
directly.
2016-01-08 16:40:32 +00:00
Tim-Philipp Müller
a8b8643977 rtpvrawdepay: fix major memory leak and performance issue
We call gst_rtp_buffer_get_payload() which creates a sub-buffer
of each input buffer, just to copy over metas, and then leak it.

https://bugzilla.gnome.org/show_bug.cgi?id=760289
2016-01-08 16:40:28 +00:00
Tim-Philipp Müller
6dab3ece07 flacparse: don't map buffer multiple times when parsing 2016-01-07 16:24:09 +00:00
Steven Hoving
910d75ddaf matroska: Store subtitle stream count in the correct variable
And don't override the video stream count instead.
2016-01-07 18:20:30 +02:00
Sebastian Dröge
4917515342 equalizer: The child-proxy API is GObject based in 1.x
Not GstObject anymore.
2016-01-05 18:59:25 +02:00
Reynaldo H. Verdejo Pinochet
ba094b50e1 flacparse: add debug msg on CRC mismatch while validating frame header 2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
6b7675e4a2 flacparse: drop unneeded braces at _parse_frame() exit
Additionally, drop redundant comment & line break
2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
b6ebad0997 flacparse: minor grammar correction 2015-12-31 16:04:15 -08:00
Reynaldo H. Verdejo Pinochet
5234c7c2bd flacparse: update URLs on pointers to online spec 2015-12-31 15:34:57 -08:00
Reynaldo H. Verdejo Pinochet
5f4317843c flacparse: make buffer DTS setting explicitly unconditional
We are setting it to PTS regardless of block_strategy
2015-12-31 14:40:15 -08:00
Reynaldo H. Verdejo Pinochet
2c14f2fff1 flacparse: add actual invalid block type to warning
For someone that read the spec is clear the only *invalid*
data block type is 127. For the rest, its useful information.

Additionally. values 7-126 are currently reserved by the
spec so the situation might change in the future.
2015-12-31 14:21:40 -08:00
Reynaldo H. Verdejo Pinochet
c43f84abf3 flacparse: use shift instead of mask & comp
We are only interested on the first bit of the first
byte of the metadata block header to figure out whether
is marked as the last one. The shift makes it quite
clearer.
2015-12-31 14:12:36 -08:00
Reynaldo H. Verdejo Pinochet
8a745837aa flacparse: warn on wishful parsing of weird headers
If we get anything from 7 to 126 as type when parsing
a metadata block header, we are likely dealing with a
FLAC stream version we don't fully understand. Issue
a warning if so.

Document function assumptions regarding the passed-on
type while at this.
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
df6f0bc595 flacparse: show meaningful info on frame CRC check
As CRCs are calculated for the comparition already, we
might as well (cheaply) inform the user how the numbers
differ if a missmatched pair is found.

While at it:

Rephrase candidate-frame message to make more sense
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
395afed566 flacparse: drop remaining trailing whitespace 2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
a086ee6192 flacparse: drop superflous else clauses 2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
7286aae6e5 flacparse: factor out buffer time and offset resetting
Avoids multiple occurrences of the same resetting pattern
2015-12-31 13:04:23 -08:00
Reynaldo H. Verdejo Pinochet
5bf1f1ec9c flacparse: move block handling by type out of _parse_frame() 2015-12-31 13:04:23 -08:00