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Author SHA1 Message Date
Jan Schmidt 46cc64e09f mpegtsmux: Fix handling of MPEG-2 AAC 
The audio/mpeg,mpegversion=2 caps in GStreamer refer to
MPEG-2 AAC (ISO 13818-7), not to the extended MP3 (ISO 13818-3),
which is audio/mpeg,mpegversion=1,mpegaudioversion=2/3

Fix the caps, and add handling for MPEG-2 AAC in both ADTS and raw
form, adding ADTS headers for the latter.
 2020-07-08 12:24:13 +00:00
Tim-Philipp Müller 4c4b3dfc76 meson: Fix up update-orc-dist target for the case where there are no orc targets 
All those plugins might have been disabled, in which case meson
would complain about alias_target() needing at least two arguments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1418>
 2020-07-08 11:39:58 +01:00
Tim-Philipp Müller b428b7357e pkgconfig: fix meson warning about waylandlib not being in the config data 
meson.build:58: WARNING: The variable(s) 'waylandlibdir' in the input file 'subprojects/gst-plugins-bad/pkgconfig/gstreamer-plugins-bad-uninstalled.pc.in' are not present in the given configuration data.

We don't provide a .pc file for this lib nor install its headers,
so no need for this path to be in the uninstalled .pc file really.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1417>
 2020-07-08 10:54:49 +01:00
Tim-Philipp Müller f3fdd76683 rtmp, transcodebin: fix i18n header includes 
Fixes #1351

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1416>
 2020-07-07 19:55:00 +01:00
Nicolas Dufresne af741f0723 rist: Use g_signal_connect_object() 
rtpbin can still emit signals when it is being disposed, and while
rtpbin is inside ristsrc/ristsink it can still live longer.

So we either have disconnect all signals at some point, or let GObject
take care of that automatically.

Related to !1412

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1413>
 2020-07-07 15:37:57 +00:00
Josep Torra 7346e7c1e2 scenechange: use orc to compute score 
Add an orc implementation for SAD operation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1024>
 2020-07-07 15:06:55 +01:00
Sebastian Dröge b812d1c743 rtpsrc/sink: Use g_signal_connect_object() 
rtpbin can still emit signals when it is being disposed, and while
rtpbin is inside rtpsrc/rtpsink it can still live longer.

So we either have disconnect all signals at some point, or let GObject
take care of that automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1412>
 2020-07-07 12:42:36 +00:00
Jan Alexander Steffens (heftig) 9c2982d22c tests: mpegtsmux: Test we don't crash releasing unused pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1411>
 2020-07-07 14:05:04 +02:00
Jan Alexander Steffens (heftig) 076189e2dc tests: mpegtsmux: Avoid use-after-unref 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1411>
 2020-07-07 14:05:04 +02:00
Jan Alexander Steffens (heftig) cba9ba9b38 mpegtsmux: Avoid crash releasing pad with NULL prog 
If we release a pad while the muxer is running which has never been used
for aggregation (thus it does not have an assigned program), `prog` is
NULL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1411>
 2020-07-07 14:05:04 +02:00
Haihao Xiang 9e977832c1 msdkh265enc: let MSDK select the encoding mode by default 
MSDK may support lowpower and non-lowpower modes, some features are
available only under one of the two modes, which is hard to know for
user, so let MSDK select the mode by default.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1405>
 2020-07-06 14:43:31 +00:00
Matthew Waters d6635346a2 build: remove obsolete 'bad-transcoder' pc file 
Replaced by 'transcoder' pc files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1409>
 2020-07-06 14:17:34 +00:00
Tim-Philipp Müller 7b2c3a984c meson: add update-orc-dist target 
Add target to update backup orc -dist.[ch] files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1408>
 2020-07-04 15:05:23 +01:00
Vivia Nikolaidou 31d5d04bb1 videoparseutils: Only add a single closed caption meta 
Otherwise, having a stream go through a parser multiple times would
result in duplicate closed caption meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1396>
 2020-07-03 08:25:54 +00:00
Matthew Waters c94675f1d4 decklinkvideosink: write the cdp timecode data correctly 
We were mixing up the tens part with the unit parts all over the place.

e.g. 12 seconds would be encoded as 0x21 instead of the correct 0x12

Aligns the code with the same change applied to ccconverter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Matthew Waters ebd1b2c929 ccconverter: write the cdp timecode data correctly 
We were mixing up the tens part with the unit parts all over the place.

e.g. 12 seconds would be encoded as 0x21 instead of the correct 0x12

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Matthew Waters 327a79e982 ccconverter: output warning log if parsing a cdp packet fails 
Simplifies figuring out why there may be no output from ccconverter with
a cdp input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Matthew Waters 6fa4a8c3c3 ccconverter: fix cdp timecode parsing 
The first reserved bits are in the most significant bit.

i.e. 0xc0 vs 0x0c

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Ederson de Souza f8bf84307f avtp: Use g_strerror instead of strerror 
It should avoid some implicit declaration errors (and be utf-8 friendly).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1404>
 2020-07-03 04:04:39 +00:00
Tim-Philipp Müller 890df7ac8c Back to development 2020-07-03 02:03:56 +01:00
Tim-Philipp Müller 1408ffc6fa Release 1.17.2 2020-07-03 00:31:19 +01:00
Philippe Normand 8900f2d2f9 wpe: Update plugin's doc cache 
This was forgotten in !1392.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1402>
 2020-07-02 17:07:46 +00:00
Nicolas Dufresne 1bef43f9d4 v4l2decoder: Track pending request 
With the asynchronous slice decoding, we only queue up to 2 slices
per frames. That side effect is that now we are dequeuing bitstream
buffers in both decoding and presentation order. This would lead to
a bitstream buffer from a previous frame being dequeued instead of
the expected last slice buffer and lead to us trying to queue an
already queued bitstream buffer.

We now fix this by tracking pending requests. As request are executed
in decoding order, we marking a request done, we can effectively
dequeue bitstream buffer from all previous request, as they have been
executed already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne a88e63dd2f v4l2decoder: Improve debug tracing 
Add some missing traces and move per-slice operation to TRACE level to
reduce the noise level.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne d5a205cff4 v4l2decoder: Convert request pool to GstQueueArray 
The decoder is not being access from multiple threads, instead it is
always protected by the streaming lock. For this reason, a
GstAtomicQueue for the request pool is overkill and may even introduce
unneeded overhead. Use a GstQueueArray in replacement, the
GstQueueArray is a good fit since the number of item is predictable and
unlikely to vary at run-time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne a2eb1b57ff v4l2slh264dec: Wait on previous pending request in slice mode 
In slice mode, we'll do one request per slice. In order to recycle
bitstream buffer, and not run-out, wait for the last pending
request to complete and mark it done.

We only wait after having queued the current slice in order to reduce
that potential driver starvation and maintain performance (using dual
buffering).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne b20c6fe6c4 v4l2slh264dec: Renew bitstream buffer after submitting slice 
Submitting a slice actually clears the bitstream buffer. Ensure we
have a newly allocated bitstream buffer for the next slice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne bc1a0323a9 v4l2slh264dec: Factor out bitstream allocation 
No functional changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne 779f331bd4 v4l2slh264dec: Add a helper to ensure output buffer 
In preparation of multi-slice decoding, we will decode multiple
slices into the same buffer. This will ensure we have a buffer to
decode to, queued into the driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne d65f7de650 v4l2slh264dec: Factor out request wait 
This will be reused to wait for previous slices to be complete
when dealing with following slices (in slice decoding mode).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne 176a860169 v4l2slh264dec: Remove double return in submit_bitstream() 
This is code cleanup, no functional changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Nicolas Dufresne 1f48e60bde v4l2slh264dec: Fix typo in debug trace 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1395>
 2020-07-02 12:21:51 -04:00
Mathieu Duponchelle b33f10e7e2 docs: remove gst prefix from plugin titles 2020-07-02 18:10:21 +02:00
Seungha Yang c72ccded6c docs: Update plugin cache for Windows plugins 2020-07-02 17:21:33 +02:00
Seungha Yang 8d0dc4fdd2 plugins: Update for documentation of Windows plugins 
* Add Since marks
* Make use of GST_PARAM_CONDITIONALLY_AVAILABLE flag
 2020-07-02 17:21:29 +02:00
Seungha Yang 76793ffabc nvcodec: Update for documentation 
* Add Since marks
* Make use of GST_PARAM_CONDITIONALLY_AVAILABLE flag
* Add documentation template caps
 2020-07-02 17:21:24 +02:00
Philippe Normand db0ab58e14 wpe: Set documentation caps 
As the caps template can vary depending on the WPEBackend-FDO version
found at build time, set a fixed template for the generate documentation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1392>
 2020-07-01 20:31:42 +00:00
Jan Alexander Steffens (heftig) afdde9fa40 videoparsers: Fix parsing ATSC bar data 
It rejected the case of all bars being disabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1394>
 2020-07-01 20:02:35 +00:00
Jan Alexander Steffens (heftig) 01896c11d2 videoparsers: Fix parsing of ATSC AFD data 
The test for 0x40 being set is repeated by
gst_video_parse_utils_parse_afd, which also extracts the low nibble
again, so we must not clear it here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1394>
 2020-07-01 20:02:35 +00:00
Jan Alexander Steffens (heftig) cedb07fe46 videoparsers: Give gstvideoparseutils.c a debug category 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1394>
 2020-07-01 20:02:35 +00:00
Matthew Waters c6c4d42c4a ccconverter: fail negotiation when framerate conversion is not possible 
Converting between anything but cdp will fail at converting
framerates and negotiation should reflect that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
 2020-07-01 19:33:56 +00:00
Matthew Waters 4f334234c8 ccconverter: fix missing output framerate on the caps 
A pipeline like this:

closedcaption/x-cea-708,format=cdp,framerate=30000/1001 ! ccconverter ! closedcaption/x-cea-708,format=cc_data

would produce a critical/assert:

GStreamer-CRITICAL **: 14:21:11.509: gst_util_fraction_multiply: assertion 'a_d != 0' failed

because there would be no framerate field on ccconverter's output.

Fixed by always fixating a framerate if the input has a framerate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
 2020-07-01 19:33:56 +00:00
Jan Alexander Steffens (heftig) 1e29c5d52a rtmp2: Set connect args like libavformat does 
To improve our compatibility. Critically, a server might elide data for
codecs we don't advertise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
 2020-07-01 18:33:42 +00:00
Jan Alexander Steffens (heftig) 2ad3aab1d4 rtmp2: Add support for AGGREGATE messages 
They're multiple frames (tags) of FLV data wrapped into a message.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
 2020-07-01 18:33:42 +00:00
Jan Alexander Steffens (heftig) 30b1187108 rtmp2: Move FLV tag header parsing into rtmputils.c 
To be shared with the AGGREGATE handling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
 2020-07-01 18:33:42 +00:00
Jan Alexander Steffens (heftig) 368c038ef0 rtmp2: Mark our memory singleton as leakable 
So it doesn't appear in the leaks tracer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
 2020-07-01 18:33:42 +00:00
Jan Alexander Steffens (heftig) edd3c4fadf rtmp2: Remove GST_ERROR from rtmputils.c 
This file does not have debug logging set up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
 2020-07-01 18:33:42 +00:00
Tim-Philipp Müller c229127b43 avtp: documentation fixes 
Unclear why hotdoc wants 'gstavtp' as the plugin name here,
that's just wrong.

Add since marker and mark private subclasses as plugin API
so hotdoc knows they belong to the plugin and aren't external.

Fix GstAvtpAafTstampMode get_type() function.
 2020-07-01 18:41:25 +01:00
Tim-Philipp Müller 22a00d78ce docs: update plugin cache with avtp plugin 
CI picks this up now because the wrap was re-added in gst-build.
 2020-07-01 11:17:51 +01:00
Seungha Yang 487f9a08de codecs: h264decoder: Fix for DPB size calculation 
Some bitstreams might require more DPB size than that of what we've
calculated.

This change should've been part of initial commit of h264 stateless
codec implementation but it was missed.

See also https://chromium-review.googlesource.com/c/chromium/src/+/760276/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1385>
 2020-06-30 22:20:29 +00:00