Commit graph

7969 commits

Author SHA1 Message Date
Nicolas Dufresne
e7b962d9b5 alphacombine: Ignore all events coming from the alpha_pad
As per usage of this element, everything from this pad is a
duplicate. Instead of implemented needless aggregation, simply
drop all events from this pad and let the one from the main stream
passthrough. Also stop proxying some queries from the alpha pad_too.

This fixes racy test failure:
- validate.file.playback.scrub_forward_seeking.opus_vp9-alpha_webm

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2247>
2021-05-14 14:11:39 -04:00
Nicolas Dufresne
0484d658a8 codecalphademux: Do not set a GstFlowReturn from a boolean
This was a small overlook, gst_pad_send_event() returns a boolean,
so setting it into ret could confuse the flow combiner. Though,
it didn't bug, since both 0 and 1 are success (though 1 being
undefined).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2247>
2021-05-14 14:11:39 -04:00
Nicolas Dufresne
35775f1aec codecalphademux: Remove eos flow return workaround
It turns out that downstream returning OK after EOS is a bug in
multiqueue. As we moved to queue, we no longer have this issue.
Let's keep the code clean and just assuming that downstream will
keep returning EOS and allow convergence of flow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2247>
2021-05-14 14:11:39 -04:00
Nicolas Dufresne
c63b2f2712 alphadecodebin: Use normal queues instead of multiqueue
The multiqueue was too flexible for our need, allowing to queue passed
the configured threshold. It also didn't work well when trying to
propagate EOS flow return.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2238>
2021-05-13 14:38:47 +00:00
Nicolas Dufresne
1229257ad4 alphacombine: Implement flow return propagation
The EOS handling was not the problem way. Instead of this, implement
proper prorogation of the flow return for the alpha chain function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2238>
2021-05-13 14:38:47 +00:00
Nicolas Dufresne
ea08442699 codecalphademux: Fix handling of flow combine
As the alphacombine is simplified to received matching pair of buffers,
we can't just stop streaming when we receive EOS from downstream. Due
to usage of queue, the moment we get this return value may differ.

Though, by continuing pushing, we override the last_flowret on the pad
which can make us miss that we effectively can combine all flow into
EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2238>
2021-05-13 14:38:47 +00:00
Thibault Saunier
61a04cf51f testbinsrc: Handle setting URI on the fly
Reusing existing streams when possible

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2210>
2021-05-13 02:03:57 +00:00
Nicolas Dufresne
b884bcb93e vp9parse: Manually fixate codec-alpha field
This is a newly introduced field, and we interpret it as false when missing in
the caps. Otherwise, a simple capsfilter will just add the missing field and
keep going, despite the upstream caps being a superset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Nicolas Dufresne
e08c0803e1 doc: codecalpha: Add plugin documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Nicolas Dufresne
ba4053a2b9 alphadecodebin: Add wrappers to decode VP8/VP9 alpha
This includes base class with wrappers bin that will create a static
pipeline capable of handling the VP8/VP9 alpha channel decoding
using two instances of vp8/vp9dec element each.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Nicolas Dufresne
2cd927435c codecalpha: Implement alphacombine element
This element will merge video buffers in order to use the alpha stream
luma plane as the alpha of the video stream. The implementation is zero-copy
and currently only support merging I420 stream with an I420, NV12 or GRAY8
alpha stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Nicolas Dufresne
4dbf61d1ef alphacodecdemux: Implement meta demuxing
Produce two streams from a buffer that has GstVideoCodecAlphaMeta
attached.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Nicolas Dufresne
f3114d4d7e Introduce CODEC Alpha plugin
This plugin contains a set of utility elements allowing to extract,
decode and combine CODEC (typically VP8/VP9) alpha stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2199>
2021-05-11 16:06:56 -04:00
Jan Alexander Steffens (heftig)
3d02559002 rtpsrc: Plug leak of rtcp_send_addr
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2226>
2021-05-07 12:34:20 +00:00
Jan Alexander Steffens (heftig)
88d7141ba4 rtpsink: Return proper pad from _request_new_pad
Bizarrely, it returned a pad from the child rtpbin. I noticed because
our application leaked the implicitly created ghost pad. Make an
explicit ghost pad so this works properly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2227>
2021-05-07 12:07:05 +00:00
Jan Alexander Steffens (heftig)
ddac6ab91d rist: Plug leak of rtcp_send_addr
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2225>
2021-05-07 11:06:53 +00:00
Nirbheek Chauhan
4c4f031207 h265parse: don't invalidate the last PPS when parsing a new SPS
This is a port of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2019
to h265parse.

When a SPS is received then any previous PPS remains valid. So don't clear
the PPS flag from the parser state.

This is important because there are encoders that don't generated a PPS after
every SPS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2217>
2021-05-05 10:02:28 +00:00
François Laignel
ad3d7d34cc Use gst_element_request_pad_simple...
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2180>
2021-05-05 06:17:14 +00:00
Stéphane Cerveau
eb96f50c45 mxf: check EOS cond with any segment's flag
The previous test was preventing the pad to be in EOS
when the segment position was greater than segment stop.
It ended up consuming all the data before getting in EOS.

Regarding GST_SEEK_FLAG_SEGMENT it seems to be
correctly handled later in the method.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2173>
2021-05-04 19:11:27 +00:00
Stéphane Cerveau
c32f455b7b mxfdemux: fix keyframe detection in index
An index entry should be considered as a keyframe
if the flags allow a random access only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2173>
2021-05-04 19:11:27 +00:00
Olivier Crête
4b47b96ae1 jpegparse: Don't generate timestamp for 0/1 framerates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2194>
2021-04-23 17:54:30 +00:00
Thibault Saunier
788dfdbfa6 rtpsrc: Fix wrong/NULL URI handling
We can reset the URI to NULL and this fix a deadlock in that case or
when the URI was invalid.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2132>
2021-04-23 01:23:03 +00:00
Mathieu Duponchelle
0fb7392131 tsdemux: fix truncated output segment when seeking with a stop
In disabling the stop adjustment for negative rates in
03031037fa , two instructions
were inverted resulting in the stop always being adjusted by
0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2182>
2021-04-21 21:40:26 +00:00
Jan Schmidt
bd9f675318 switchbin: When collecting srcpad caps, don't intersect with path caps.
The path caps describe the input caps that will select each path, don't
intersect those with the srcpad caps, which could be completely
different. Instead, when querying allowed caps for the srcpad, just
construct the union of all possible output caps from all path srcpads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2018>
2021-04-12 14:27:00 +00:00
Jan Schmidt
1f865246c1 switchbin: Don't report sink pad caps for src pad queries.
When handling a caps query on the src pad, don't return the union
of input caps. Even when not active, a path element can be queried
for srcpad template caps, or for dropping paths the allowed downstream
caps is anything - as data will be dropped anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2018>
2021-04-12 14:27:00 +00:00
Stéphane Cerveau
891be51105 gst-plugins: allow per feature registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2110>
2021-04-11 16:16:55 +00:00
Helmut Januschka
7f60138ef6 allow NetStream.Play.PublishNotify Message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2154>
2021-04-10 20:34:26 +02:00
Philippe Normand
933ebba435 debugutils: Add fakeaudiosink element
This element can be useful for CI purposes on machines not running any system
audio daemon. The element implements the GstStreamVolume interface.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2125>
2021-04-09 08:13:12 +00:00
Haihua Hu
e69d0151d2 jpeg2000parse: fix critical log when play one gray colorspace video
Need guess color space based on number of components when cannot
got it from sink caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1955>
2021-04-08 01:06:12 +00:00
Doug Nazar
81d4ccdc44 rtmp2: Use correct size of write macro for param2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2146>
2021-04-07 07:48:57 -04:00
Jan Schmidt
3a3c80e7be mpegtsmux: Respect the start-time-selection property.
Use the start time provided by the aggregator base class for output
times.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2105>
2021-03-31 13:34:40 +00:00
Sebastian Dröge
5bcfb2dda0 avwait: Don't reset time tracking when receiving the same segment again
This causes avwait to go back into "dropping" mode until audio and video
are synced again, which is unnecessary when the segment didn't actually
change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2121>
2021-03-30 08:38:03 +00:00
Jan Alexander Steffens (heftig)
0b916e7cec rtmp2/connection: Separate inner from outer cancelling
The connection cancels itself when it is closed. To avoid the
cancellable passed to `gst_rtmp_connection_new` from being unexpectedly
cancelled, separate inner from outer cancellation by holding two
cancellables.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1558

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2111>
2021-03-28 11:07:33 +00:00
Stéphane Cerveau
176a00985a mpegpsdemux: fix accurate seek
In an accurate seek, the segment start should be
the same as the one requested in the seek.
The start should be kept as the one from the
segment if its inferior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2048>
2021-03-22 12:54:14 +01:00
Stéphane Cerveau
497e88ae88 mpegpsdemux: Keep seqnum events
Keep the same seqnum of the new segment events for each
of the streams.
Keep the segment to send the EOS event.
Keep the seek seqnum for segment and flush event.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2048>
2021-03-22 12:54:14 +01:00
Stéphane Cerveau
96467f581e mpegpsdemux: avoid early EOS
In a case of a scr different from 0, after a seek,
the src_segment.stop has been updated with the duration
not including the base_time (scr). The segment position
needs to be tested upon segment.stop + base_time (scr)
to check for an EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2048>
2021-03-22 12:54:14 +01:00
Matthew Waters
640a65bf96 gst: don't use volatile to mean atomic
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead.  GCC 11 has started warning about using volatile
with atomic operations.

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719

Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2098>
2021-03-22 14:34:36 +11:00
Jan Schmidt
18a095ca63 mpegtsmux: Add PMT_%d support to prog-map.
Support a PMT_%d field in the prog-map, that's optionally used
to set the PMT for each program in the mux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2039>
2021-03-18 15:07:53 +00:00
Jan Schmidt
5e4a11bf36 mpegtsmux: Don't write PCR until PAT/PMT are output.
Make sure streams start cleanly with a PAT/PMT and defer the first PCR
output until after that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
2021-03-18 13:57:27 +00:00
Mathieu Duponchelle
b7e9e606a9 tsmux: finalize PCR timing for complete accuracy
In order to always insert a PCR packet right on time, we need to
check whether one is needed when outputting any packet, not only
a packet for the PCR stream. Most of the PCR packets will remain
data-carrying packets, but as a last resort we may insert stuffing
packets on the PCR stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
2021-03-18 13:57:27 +00:00
Jan Schmidt
3201575d2a mpegtsmux: Improve PCR/SI scheduling.
Change PCR / SI scheduling so that instead of checking if
the current PCR is larger than the next target time, instead
check if the PCR of the next packet would be too late, so PCR
and SI are always scheduled earlier than the target, not later.

There are still cases where PCR can be written too late though,
because we don't check before each output packet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
2021-03-18 13:57:27 +00:00
Jan Schmidt
49c61338d6 tsmuxstream: Fix comment typo
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2073>
2021-03-18 13:57:27 +00:00
Stéphane Cerveau
b0a9ba4ccf mpegvideoparse: do not clip the frame
If the current buffer is delta unit such as P or B
frame, the buffer should not be clipped and need to
let the decoder handle the segment boundary situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2070>
2021-03-11 15:01:38 +01:00
Sebastian Dröge
80c1722cba avwait: Don't post messages with the mutex locked
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2063>
2021-03-09 13:01:45 +02:00
Vivia Nikolaidou
cde4e74eca interlace: Discard stored_frame on EOS and PAUSED_TO_READY
Would otherwise leak it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
2021-03-08 21:02:01 +02:00
Vivia Nikolaidou
cb55d30b3c interlace: Specify interlace-modes in the sink pad template
Especially specify the field-order in the interleaved mode. Otherwise it
might cause the negotiation to fail, because
GST_PAD_SET_ACCEPT_INTERSECT is not set on the sinkpad, and the
field-order is missing in the sink template but can be present in the
outside caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
2021-03-08 21:01:50 +02:00
Tim-Philipp Müller
766bd655fc interlace: add more formats, esp 10-bit, 12-bit and 16-bit ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2054>
2021-03-03 18:34:26 +00:00
Jan Alexander Steffens (heftig)
b9cc83ccf8 mpegtsparse: Fix switched DTS/PTS when set-timestamps=false
Fixes 30ee21eae3.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2047>
2021-03-01 16:29:58 +01:00
Tim-Philipp Müller
438449db69 sdpsrc: fix double free if sdp is provided as string via the property
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1532

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2025>
2021-02-19 00:05:41 +00:00
Michael Olbrich
5a03862fca h264parse: don't invalidate the last PPS when parsing a new SPS
When a SPS is received then any previous PPS remains valid. So don't clear
the PPS flag from the parser state.

This is important because there are encoders that don't generated a PPS after
every SPS.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/571

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2019>
2021-02-17 16:22:18 +00:00
Vivia Nikolaidou
66bfd0e8ae h265parse: Detect height change on field-based interlaced files
The first time update_src_caps is called, there's no frame parsed yet,
therefore we don't know whether the file has alternate-field interlacing
mode. If we run it again after we have a frame, it might be that now we
have the SEI pic_struct parsed, and therefore we know that it's
field-based interlaced, and therefore the height must be multiplied by
two. Earlier on this was not detected as a change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2022>
2021-02-17 13:46:41 +00:00
Vivia Nikolaidou
21347e13f5 h265parse: Fix FPS/duration for interlaced files
There can be h265 files with frame-based, not field-based, interlacing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2020>
2021-02-17 13:46:03 +02:00
Mathieu Duponchelle
8ae56d60a3 h264parse: fix timestamping of interlaced fields in output
Instead of relying on GstBaseParse default behaviour of computing
the duration of a parsed buffer based on the framerate passed
to gst_base_parse_set_framerate(), we instead compute the duration
ourselves, as we have more information available.

In particular, this means we now output buffers with a duration
that matches that of raw interlaced buffers when each field is
output in a separate buffer.

This fixes DTS interpolation performed by GstBaseParse, as the
previous behaviour of outputting each field with the duration of
a full frame was messing up the base class calculations.

When not enough information is available, h264parse simply falls
back to calculating the duration based on the framerate and hope
for the best as was the case previously.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1973>
2021-02-16 17:15:27 +01:00
Vivia Nikolaidou
ae66a5772c h265parse: Support for alternate-field interlacing
Also don't set interlacing information on the caps, see #1313

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1996>
2021-02-03 16:09:45 +02:00
Jan Alexander Steffens (heftig)
0f084d4624 h264/h265parse: Add VideoTimeCodeMeta to the outgoing buffer
The parsers attempted to add the meta to the incoming buffer, which
might not be the outgoing buffer or may not have been writable yet.

To fix this, call `gst_buffer_make_writable` earlier and make sure to
use the `parse_buffer` to add the meta.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1521

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2002>
2021-02-02 18:44:49 +01:00
He Junyan
db134d27a0 av1parse: set the default alignment for input and output.
1. Set the default output alignment to frame, rather than current
   alignment of obu. This make it the same behaviour as h264/h265
   parse, which default align to AU.
2. Set the default input alignment to byte. It can handle the "not
   enough data" error while the OBU alignment can not. Also make it
   conform to the comments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
2021-01-26 12:22:31 +00:00
He Junyan
5abf4ad4dd av1parse: Reset the annex_b when meet TU inside a buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
2021-01-26 12:22:31 +00:00
He Junyan
d83f253258 av1parse: Output each OBU when output is aligned to obu.
The current behaviour for obu aligned output is not very precise.
Several OBUs will be output together within one gst buffer. We
should output each gst buffer just containing one OBU. This is
the same way as the h264/h265 parse do when NAL aligned.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
2021-01-26 12:22:31 +00:00
He Junyan
ee1f6017ac av1parse: Always copy the OBU to cache.
The current optimization when input align and out out align are
the same is not very correct. We simply copy the data from input
buffer to output buffer, but we failed to consider the dropping of
OBUs. When we need to drop some OBUs(such as filter out the OBUs
of some temporal ID), we can not do simple copy. So we need to
always copy the input OBUs into a cache.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
2021-01-26 12:22:31 +00:00
He Junyan
a9c8aa4788 av1parse: Improve the logic when to drop the OBU.
When drop some OBU, we need to go on. The current manner will make
the data access out range of the buffer mapping.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
2021-01-26 12:22:31 +00:00
He Junyan
7196abf7a3 av1parse: Fix some issues in the src caps.
1. Add the mono_chrome to identify 4:0:0 chroma-format.
2. Correct the mapping between subsampling_x/y and chroma-format.
   There is no 4:4:0 format definition in AV1. And 4:4:4 should
   let both subsampling_x/y be equal to 0.
3. Send the chroma-format when the color space is not RGB.

Fixes: #1502
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1974>
2021-01-23 10:53:44 +00:00
He Junyan
1029c84dbf vp9parse: Fix the subsampling_x/y to chroma format mapping.
The chroma format 4:4:4 needs both subsampling_x and subsampling_y
equal to 0.

Fixes: #1502
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1974>
2021-01-23 10:53:44 +00:00
He Junyan
fe19bc0a2e videoparsers: av1: Add the AV1 parse.
This AV1 parse implements the conversion between alignment of obu,
tu and frame, and the conversion between stream-format of obu-stream
and annexb.

TODO:
1. May need a property of operating_point to filter the OBUs
2. May add a property to disable deep parse.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1614>
2021-01-19 18:38:03 +00:00
Raju Babannavar
7e7e54d089 dvbsuboverlay: Add support for dynamic resolution update.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1487

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1897>
2020-12-21 15:34:46 +05:30
Jan Schmidt
1b3ba87d13 audiobuffersplit: Calculate the correct size for fixed size buffers
Fix the output-buffer-size property to do what it says by calculating
the correct audio buffer size for that target size, rounded down to
the nearest whole number of samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1887>
2020-12-17 04:41:18 +11:00
Edward Hervey
83e4310da1 tsparse: Don't use non-object for debugging statement
Use the pad instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
2020-12-14 17:57:40 +01:00
Edward Hervey
fe6ae27046 mpegts: Don't add non-padded streams to collection on updates
When carrying over existing GstStream to a new GstStreamCollection we need to
check whether they *actually* were being used in the previous collection.

This avoids adding unknown streams (metadata, PSI, etc...) to the collection on
updates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
2020-12-14 17:57:40 +01:00
Lim Siew Hoon
3ce1086b14 intervideosrc: fix negotiation of interlaced caps
In 1.0 the field in caps is called "interlace-mode", not "interlaced".

Fixes #1480

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1869>
2020-12-13 13:25:13 +00:00
Vivia Nikolaidou
82dcb27401 basetsmux: Don't send the capsheader if src pad has no caps
That means we're shutting down, so there's no point in the streamheader
being sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1864>
2020-12-09 13:14:40 +00:00
Matthew Waters
1f7515100c rtmp2/connection: pass the parent cancellable down to the connection
Otherwise, when rtpm2src cancels an inflight operation that has a queued
message stored, then the rtmp connection operation is not stopped.

If the cancellation occurs during rtmp connection start up, then
rtpm2src does not have any way of accessing the connection object as it
has not been returned yet.  As a result, rtpm2src will cancel, the
connection will still be processing things and the
GMainContext/GMainLoop associated with the outstanding operation will be
destroyed.  All outstanding operations and the rtmpconnection object will
therefore be leaked in this case.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1862>
2020-12-08 23:43:02 +00:00
Marc Leeman
102c60f82c rtpmanagerbad: allow setting caps on rtpsrc
rtpsrc tries to do a lookup of the caps based on the encoding-name. For
not so standard encodings, the caps can be set, avoiding the lookup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1406>
2020-12-04 14:51:38 +00:00
Edward Hervey
30ee21eae3 tsparse: Forward incoming timestamps
Ensure we properly forward the upstream PTS/DTS on the regular and program
source pads. All packets being processed will carry over the latest PTS/DTS (as
a reconstructed GstBuffer).

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1419

And properly forward PTS/DTS for program pads (which wasn't the case before)

Original patch by Vivia Nikolaidou <vivia@ahiru.eu>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1769>
2020-12-02 14:22:06 +00:00
Thibault Saunier
8eb0e637c7 transcodebin: Minor error message enhancement 2020-11-30 17:31:48 -03:00
Thibault Saunier
eb0d72f382 transcodebin: Unlock while setting decodebin caps
Otherwise it will deadlock recursing up to notify parent object property changes
2020-11-30 17:31:48 -03:00
Thibault Saunier
5ccaa595a9 transcodebin: Avoid plugin converter if filter handles ANY caps
For example identity or clocksync or this kind of elements can be
used with any data flow and we should not enforce decoding to row in
that case.
2020-11-30 17:31:48 -03:00
Thibault Saunier
878a196080 transcodebin: Add filter as soon as it is set
Instead of waiting so that we can simply use a clocksync element as
filter, otherwise we won't know the pipeline is live as it won't
return NO_PREROLL as one would expect in that case.

Adding it right away shouldn't create any issue, both ways are fine.
2020-11-30 17:31:48 -03:00
Thibault Saunier
530f694366 uritranscodebin: Add setup-source and element-setup signals
The same way as playbinX does it as it is often quite useful
2020-11-30 17:31:48 -03:00
Thibault Saunier
142e571c28 transcode: Port to encodebin2
This allows supporting muxing sinks like hlssink2 or splitmux
2020-11-30 17:31:48 -03:00
Marijn Suijten
dc90a3d3cf audio: Use new AudioFormatInfo::fill_silence function
The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
2020-11-26 10:06:42 +02:00
Edward Hervey
50e230a270 mpegtsdemux: Fix off by one error
Turns out timestamps of zero are valid :) Fixes issues with streams where the
PTS/DTS would be equal to the first PCR.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1807>
2020-11-13 17:50:03 +01:00
Mathieu Duponchelle
c969239c7c h264parse: try harder to update timecode
NumClockTS is the maximum number of timecodes the pic_timing SEI
can carry, but it is perfectly OK for it to carry fewer, and have
one of the clock_timestamp_flags set to 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1804>
2020-11-13 13:09:01 +00:00
Mathieu Duponchelle
e93558efac h264parse: fix installing of update-timecode property
Simply fixes a typo that did not have any adverse effect,
and avoid hardcoding initializer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1805>
2020-11-12 21:34:18 +00:00
Seungha Yang
7cec64499d mpegdemux: Set duration on seeking query if possible
Set duration on seeking query in the same way as duration query handler.
Otherwise application might get confused as if the duration is unknown.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1791>
2020-11-11 14:10:27 +00:00
Edward Hervey
a2a73c02ef mpegtspacketizer: Handle PCR issues with adaptive streams
A lot of content producers out there targetting "adaptive streaming" are riddled
with non-compliant PCR streams (essentially all the players out there just use
PTS/DTS and don't care about the PCR).

In order to gracefully cope with these, we detect them appropriately and any
small (< 15s) PCR resets get gracefully ignored.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1785>
2020-11-09 18:30:51 +01:00
youngh.lee
49df312086 aiffparse: Also set a channel mask for 2 channels
And only do add debug output at FIXME level when using the fallback
channel mask, not for those defined in the AIFF spec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1756>
2020-11-04 07:36:47 +00:00
Thibault Saunier
d1945de102 transcodebin: Create the decodebin in _init
This way user can request pads right from the beginning

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
2020-10-29 13:30:07 +00:00
Philippe Normand
88c96789bf transcodebin: Accept more than one stream
Look-up the stream matching the given ID also after building the stream list
from the received collection. Without this change the transcoder would discard
the second incoming stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
2020-10-29 13:30:07 +00:00
Thibault Saunier
b254c0d5fe transcodebin: Port to decodebin3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
2020-10-29 13:30:07 +00:00
Thibault Saunier
a5fd2a4bc3 uritranscodebin: Move to using a urisourcebin for our source.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
2020-10-29 13:30:07 +00:00
Seungha Yang
639fb6ac15 rtmp2src: Set buffer timestamp on output buffer
This timestamp information would be useful for queue2 element
when calculating time level and also it makes buffering decision
more reliable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1727>
2020-10-28 16:32:32 +00:00
Aaron Boxer
b2a0fd9e96 jpeg2000parse: sub-sampling parse should take component into account
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1653>
2020-10-27 08:26:23 +01:00
Stéphane Cerveau
7edff6e746 jpeg2000parse: no pts interpolation with subframe.
The jpeg2000parser must not interpolate PTS with subframes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1653>
2020-10-27 08:26:23 +01:00
Aaron Boxer
db13dc9d02 jpeg2000parse: support frame and stripe alignment in caps
forward alignment and num-stripes caps properties

Use caps height when setting caps for subframe

We want downstream to use full frame height, not subframe height

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1653>
2020-10-27 08:26:23 +01:00
Nicolas Dufresne
dcb3044478 rtpsrc: Cleanup on BYE, timeout or when pad is reused
In this patch, we enabled 'autoremove' feature of rtpbin and also call
'clear-ssrc' on the rtpssrcdemux element when a pad is being reused. This
ensure that the jitterbuffer is removed and no threads accumulates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1575>
2020-10-16 17:23:46 +00:00
George Kiagiadakis
2fcbb4386b rtpsrc: re-use the same src pad for streams that have the same payload type
Also use payload type when naming pads, this will make it easier to identify
pads and simplify the code.

Fixes #1395

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1575>
2020-10-16 17:23:46 +00:00
Seungha Yang
634eb1fc38 h265parse: Don't enable passthrough by default
SEI messages contain various information which wouldn't be conveyed
by using upstream CAPS (HDR, timecode for example).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1639>
2020-10-15 03:25:17 +09:00
Marc Leeman
0be59181d7 rtpmanagerbad: remove duplicate parent declaration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1689>
2020-10-12 13:56:50 +02:00
Tim-Philipp Müller
1ed969d276 rtmp2sink: fix since marker on new "stop-commands" property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1687>
2020-10-12 11:55:46 +01:00
Guillaume Desmottes
75dc98cc08 h265parse: set interlace-mode=interleaved on interlaced content
interlace-mode=alternate is a special case of interlace-mode=interleaved
where the fields are split using two different buffers.

We should use the latter instead of the former to no break compat with
elements supporting only 'interleaved'.
Decoders producing alternate, such as OMX on the Zynq, should change the
interlace-mode on their output caps.

Fix https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/825

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1655>
2020-10-09 10:19:52 +00:00
Jan Alexander Steffens (heftig)
5a1b56a0e0 mpegtsmux: Restore intervals when creating TsMux
Otherwise the settings from the properties would be overwritten with
the defaults.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1593>
2020-09-23 16:50:34 +00:00