Commit graph

308 commits

Author SHA1 Message Date
Thibault Saunier
1cb4c050d0 rtpbin: Avoid holding lock GST_RTP_BIN_LOCK when emitting pad-added
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2411>
2022-05-13 06:25:03 +00:00
Sebastian Dröge
1223324246 qtdemux: Don't use tfdt for parsing subsequent trun boxes
The timestamp in the tfdt refers to the first trun box and if there are
multiple trun boxes then the distance between the first timestamps will
grow.

At some point this distance reaches a threshold and triggers the
resetting of the first sample's timestamp of this trun box to be reset
to the tfdt.

This threshold is implemented for files where there is a jump in the
timeline between fragments and where this can be detected via a jump
between the end timestamp of the previous fragment and the tfdt of the
next. This behaviour is preserved.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2409>
2022-05-13 04:19:36 +00:00
Guillaume Desmottes
aa3b6a11e0 vpxenc: enforce strictly increasing pts
From vpx_codec_encode() documentation:
  "The presentation time stamp (PTS) MUST be strictly increasing."

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Guillaume Desmottes
10b837ae5e vpxenc: conver input pts to running time
The input pts needs to be strictly increasing, see vpx_codec_encode() doc, so convert it to
running time as we don't want to reset the encoder for each segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Guillaume Desmottes
1e829696e8 vpxenc: fix crash if encoder produces unmatching ts
If for some reason the encoder produces frames with a pts higher than
the input one, we were dropping all the video encoder frames and ended
up crashing when trying to access the pts of a NULL pointer returned by
gst_video_encoder_get_oldest_frame().

I hit this scenario by feeding a decreasing timestamp to vp8enc which
seem to confuse the encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2405>
2022-05-12 13:00:53 +02:00
Nicolas Dufresne
522f19e013 v4l2videoenc: Setup crop rectangle if needed
Hantro H1 and Rockchip VEPU2 drivers will pad the width/height to a
multiple of 16. In order to obtain the right JPEG size, the image needs
to be cropped using the S_SELECTION API. This support is added as best
effort since older drivers may emulate this by looking at the capture
queue width/height.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2329>
2022-05-07 11:35:14 +00:00
Sebastian Dröge
d2c6f21fc1 mp4mux: Disable aggregator's default negotiation
mp4mux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Sebastian Dröge
841cba4182 flvmux: Disable aggregator's default negotiation
flvmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.

By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2372>
2022-05-05 17:41:58 +00:00
Matthew Waters
f4f342aa78 wavparse: ensure that any pending segment is sent before an EOS event is sent
Specifically fixes seqnum handling when an aggregator-based element
(audiomixer et al) is downstream and a seek is performed that
immediately causes an EOS from wavparse.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2356>
2022-05-04 08:00:02 +00:00
Sebastian Dröge
7466444b63 rtpjitterbuffer: Free CNAME/SSRC mappings on finalize and PAUSED->READY
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:33:47 +03:00
Sebastian Dröge
2c405da921 rtpmanager: Refactor RTCP packet loops to fix control flow
Mixing C loops with switch statements is a bad idea as break has a
different meaning in both. Breaking inside the switch statements wrongly
caused further loop iterations.

Instead use goto to get out of the loop and continue to do another loop
iteration, and never ever use break except for the end of a case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2336>
2022-04-29 23:13:15 +03:00
Seungha Yang
6619f1611f rtpjitterbuffer: Initialize variables
Avoid use of uninitialized variable
Fixing MSVC warning
gstrtpjitterbuffer.c(4733) : warning C4700: uninitialized local variable 'have_sdes' used

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2315>
2022-04-28 12:37:13 +00:00
Edward Hervey
7c9eb0335f mssdemux2: Don't expose/use streams we can't handle yet
Avoids issues further down

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2319>
2022-04-28 10:45:37 +00:00
Edward Hervey
2ec79418df mssdemux2: Ensure stream/track uniqueness
If there is more than one track of the same type (say audio), we would end up
creating several stream/types with the same name.

Instead use the MSS stream name property to make them unique

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2319>
2022-04-28 10:45:37 +00:00
dongil.park
5b11e6a3d0 wavparse: Unset DISCONT buffer flag for divided into multiple buffers in push mode
In push mode (streaming), if the received chunk buffer size from _chain is bigger
than output buffer size, the flags of the divided-buffers are propagated to the
DISCONT flag from first received chunk buffer. This unexpected buffers contained DISCONT
flags are abnormally transformed when changing the sampling rate by audioresample element.
So unset unnecessary DISCONT flag before pad_push().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2305>
2022-04-27 14:29:10 +00:00
Sebastian Dröge
9d5179ad3f rtpjitterbuffer: add the reference timestamp meta in more situations
Previously, we only added it when actually performing synchronization
based on the NTP time.

The information can be useful downstream in other situations too, and
we can compute a NTP time as soon as we get a sender report with the
relevant information.

Co-authored-by: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2252>
2022-04-27 12:35:21 +00:00
Sebastian Dröge
ed425e2785 rtpgstpay: Don't push packets before the first input buffer is received
It's not possible to create a valid RTP timestamp for them, which would
cause a potentially very big RTP timestamp discontinuity between those
first packets (created from initial events) and the packet based on the
first input buffer.

As a side-effect, also simplify the packet aggregation code a bit and
work with only a single level of buffer lists.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2250>
2022-04-27 11:55:17 +00:00
Havard Graff
390ec99f1b rtptwcc: don't map the buffer twice
...and use the pt extracted rather than the one from RTPPacketInfo
when logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2271>
2022-04-26 10:27:25 +00:00
Thibault Saunier
d673a90aea rtpsession: Emit "notify::stats" when we update stats from RR or SR
Sensibily optimizing caching the pspecs and using them directly

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2266>
2022-04-26 08:49:42 +00:00
Mathieu Duponchelle
3391a7d499 rtpredenc: quieten warning about ignoring header extensions
Turn it into a FIXME, and only log once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2279>
2022-04-23 01:04:54 +00:00
Havard Graff
b7b71e6974 rtprtxsend: mark RTX buffers with GST_RTP_BUFFER_FLAG_RETRANSMISSION
It is useful for elements downstream from rtxsend to know if the RTP
buffer they are dealing with is an RTX buffer or not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2272>
2022-04-22 19:27:45 +00:00
Tristan Matthews
27dea62304 mp4mux: fix spelling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2241>
2022-04-22 14:07:57 +00:00
Jonas Bonn
2f6ad787b2 multiudpsink: allow binding to IPv6 address
When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6.  When binding to an IPv6 address, this
results in the following error:

gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)

This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1551>
2022-04-22 10:43:13 +00:00
Camilo Celis Guzman
5eadde319c rtphdrextsdes: fixup test trying to g_free a local variable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2235>
2022-04-22 08:41:59 +00:00
Edward Hervey
964ee0299d hls/m3u8: Fix starting segment for live playlist
RFC 8216 6.3.3 "Playing the Media Playlist File" : states that for live media
playlists "the client SHOULD NOT choose a segment that starts less than three
target durations from the end of the Playlist file"

This is an off-by-one error. Since we are looking for the "index" of the
segment, we need to subtract 1 from the searched position.

Ex: For a playlist with 12 entries, we want to start playback on the 9th segment
... which is at index 8.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2259>
2022-04-22 08:06:27 +00:00
Edward Hervey
8f2d347559 hls: Relax webvtt checks
If no hour field is present (which is allowed), the remaining data can be less
than 15 character.

Fix time translation failures if the hour field wasn't present

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2248>
2022-04-20 17:47:00 +00:00
Sebastian Dröge
02115a5efc rtpmanager: Move some duplicated constant and helper function to a single place
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
c7e12974ba rtpbin/rtpjitterbuffer: Don't parse RTCP SRs twice unless needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
82169aa140 rtpjitterbuffer: Add property to throttle handling of RTCP SR / NTP-64 syncing
This proxies the "rtcp-sync-interval" property of rtpbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
ce38614e1a rtpsession: Handle RTCP-SR-REQ (RFC6051) RTCP feedback message
This causes an RTCP SR to be sent at the earliest possible time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
0c819d2f31 rtpbin/rtpjitterbuffer: Allow syncing to an SR without CNAME if the CNAME is already known
The RTCP SR packet might be without SDES in case of a reduced-size RTCP
packet. For syncing purposes the CNAME is needed but it might be known
already from an earlier RTCP packet or out of band, via the SDP for
example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
cbaac3cdba rtpbin/jitterbuffer: Use inband 64-bit NTP timestamps according to RFC6051 for faster synchronization
When signalled via the caps that the header extension is used, it will
be read and used in the same way as the RTP/NTP time mapping from RTCP
SRs.

If the CNAME of the stream's SSRC is provided out of band via e.g. the
SDP then this allows streams to be synchronized immediately on the first
packet instead of having to wait for the first RTCP SR to arrive.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/383

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7c796b3c05 rtpsession: Only add send latency to the running time if it is actually known
Otherwise we can't know the running time yet if rtcp-sync-send-time is
set, and have to wait until the latency is known later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
7ffc830959 rtpsession: Update 64-bit NTP header extensions with the actual NTP time in senders
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Sebastian Dröge
8980c35efe rtpmanager: Add header extension implementation for the 64-bit RFC6051 NTP header extension
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Xavier Claessens
e950095867 Always define ENABLE_NLS
GLib guarantees libintl API is always available, provided by
proxy-libintl as last resort. GLib itself unconditionally define
ENABLE_NLS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Xavier Claessens
82ca0e291b Delete unused i18n headers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Xavier Claessens
b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Tim-Philipp Müller
0dd04764f7 tests: dash_mpd: fix linker issues with non-optimizing compilers
undefined reference to `download_request_take_buffer'

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2117#note_1344646

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2228>
2022-04-19 10:35:30 +00:00
Ruben Gonzalez
70579285a8 gst_plugin_load_file: force plugin reload if diff filename
If a file includes a new version of a plugin that exits in the
registry, the output of gst-inspect is incorrect. The output has the
correct version but incorrect filename, and element description.

This seems to have also fixed some documentation issues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1344>
2022-04-19 14:26:08 +05:30
Edward Hervey
af78c16dd5 New HLS, DASH and MSS adaptive demuxer elements
This provides new HLS, DASH and MSS adaptive demuxer elements as a single plugin.

These elements offer many improvements over the legacy elements. They will only
work within a streams-aware context (`urisourcebin`, `uridecodebin3`,
`decodebin3`, `playbin3`, ...).

Stream selection and buffering is handled internally, this allows them to
directly manage the elementary streams and stream selection.

Authors:
* Edward Hervey <edward@centricular.com>
* Jan Schmidt <jan@centricular.com>
* Piotrek Brzeziński <piotr@centricular.com>
* Tim-Philipp Müller <tim@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2117>
2022-04-18 14:11:23 +00:00
Hou Qi
8dcb8a28af v4l2videodec: copy colorimetry values to output_state caps
This is to avoid transcoding negotiation fail between v4l2h265dec
and v4l2h264enc caused by colorimetry mismatch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2192>
2022-04-18 13:17:55 +00:00
Brad Hards
488b760e7e tests: rename 'icles' subdir to be more descriptive
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2178>
2022-04-14 11:57:11 +00:00
Havard Graff
71891e5647 qtdemux: fix leak of channel_mapping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2179>
2022-04-14 19:41:36 +09:00
Ming Qian
030d749019 doc: Update cache after NV12_8L128 and NV12_10BE_8L128 addition
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2158>
2022-04-13 07:20:58 +00:00
Ming Qian
dce02a870e v4l2: Add NV12_8L128 in gst_v4l2_object_get_caps_info
It should be included in
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1379>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2158>
2022-04-13 07:20:58 +00:00
Ming Qian
6af66167d0 v4l2: Add a missed break
Fix a typo that miss a break in the switch statement

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2158>
2022-04-13 07:20:58 +00:00
Robert Rosengren
e4a6521ac7 rtpbin: Fix division by zero when using ts-offset-smoothing-factor
avg_ts_offset may cause division by zero when calculating potential
overflow protection. This fix will avoid the division.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2151>
2022-04-11 15:29:49 +02:00
Tristan Matthews
86f0f8b67f rtpopusdepay: assume 2 channels if sprop-stereo is missing
Fixes #1064

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2125>
2022-04-08 13:11:25 +00:00
Matthias Fuchs
42ec223f94 qmlglsrc: Fix deadlock when stopping
This fix makes sure that streaming thread stops waiting when the
qmlglsrc element transitions from playing to paused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2115>
2022-04-06 10:54:51 +00:00