Commit graph

59 commits

Author SHA1 Message Date
Tim-Philipp Müller
0fc568c6b1 gst-plugins-good: re-indent with GNU indent 2.2.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
2023-03-17 03:18:54 +00:00
Rafał Dzięgiel
9d720554a0 dashdemux2: Improve initial representation selection
Do not always start with lowest quality possible. Use properties set
by user to select best allowed initial representation at startup too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3894>
2023-02-18 21:05:25 +00:00
Hosang Lee
0efb792fb4 tests: qtdemux: add test for MSS fragment wrong data offset compensation
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams. The samples will not be located and
eventually playback will error out. So compensate assuming data
is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Tim-Philipp Müller
491feead6e tests: qtdemux: use binary files for samples
Instead of hexdumping it in a 360k header file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Patricia Muscalu
c3e52d5c4f rtph264pay: Don't insert SPS/PPS before the second image slice
Only the first slice, for which fist_mb_in_slice is set to 0,
should trigger insertion of SPS and PPS buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3402>
2023-02-08 12:10:11 +00:00
Edward Hervey
0639f117cb hlsdemux2: Remove enable-llhls property
This was only used for testing purposes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Jan Schmidt
2b93dae59a hlsdemux2: Handle async playlist loading failures
Add failed variant playlists to a list and failover to other variants until
there is none left

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
9ae3978c72 hlsdemuxdemux2: Consider the hold-back when calculating seek range
When calculating the seek range for a live stream, use the same hold-back logic
as when choosing a starting segment, including low-latency segments if
enabled. Permits seeking closer to the live edge when re-synching or catching
up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
7496d2750e hlsdemux2: Parse EXT-X-SKIP tag
Parse the attributes from the EXT-X-SKIP tag

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:22 +00:00
Jan Schmidt
5baf5f4b1e hlsdemux2: Add a timestamp to the playlist
Store the timestamp for this playlist. If valid it represents the monotonic time
at which the data was retrieved, minus any proxy cache Age field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:22 +00:00
Jan Schmidt
43e042c4b7 hlsdemux2: Implement LL-HLS flag and part-hold-back/hold-back in live.
Add a flag to hlsdemux to enable or disable LL-HLS handling.

When LL-HLS is enabled and an LL-HLS playlist is loaded, use the part-hold-back
threshold to choose a starting segment.

For live streams that aren't LL-HLS, use the provided hold-back attribute, or
fall back to landing 3 segments from the end.

Make the gst_hls_media_playlist_seek() method able to choose a partial segment
within 2 target durations of the end of the playlist when requested.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:22 +00:00
Jan Schmidt
92e849070f hlsdemux2: Mark locations where partial segments need handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:21 +00:00
Jan Schmidt
43a8d45ac6 hlsdemux2: Add unit test for parsing LL-HLS playlist
Test parsing of partial segments (EXT-X-PART, EXT-X-PART-INF) and preload
hints (EXT-X-PRELOAD)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:21 +00:00
Tim-Philipp Müller
756a8986d0 good: tests: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Olivier Crête
46a6f72f03 rtopuspay: Ignore the stereo parameter in multiopus caps
Also add unit tests for the various variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
A. Wilcox
412eaf3526 tests: Cast drop-messages-interval type properly
The rtpjitterbuffer test drop_messages_interval uses a GstClockTime for
the message drop interval.  This property is defined as a guint.  On
systems with 64-bit time_t but 32-bit uint, this can cause the
g_object_set function to fail to read the arguments properly.

Fixes: #1656
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3580>
2022-12-16 01:36:07 -06:00
Johan Sternerup
9794c9bfd0 Use the correct SSRC(s) when routing a RTCB FB FIR
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
2022-11-23 11:31:23 +00:00
Sebastian Dröge
d815035e82 rtpjitterbuffer: Add test for rescheduling timers to negative times
This tests the changes introduced by 4d3b8d1129
for issue #571.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>
2022-11-16 08:26:41 +00:00
Rafał Dzięgiel
e93f391139 tests: Add DASH MPD baseURL with query test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
2022-11-14 23:45:53 +00:00
Tim-Philipp Müller
d132592423 xingmux: move from gst-plugins-ugly to gst-plugins-good
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/415

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3251>
2022-10-25 12:40:20 +00:00
Devin Anderson
4e03c5f885 wavparse: Fix crash that occurs in push mode when header chunks are corrupted
in certain ways.

In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000  52 49 46 46 e4 fd 00 00  57 41 56 45 66 6d 74 20  |RIFF....WAVEfmt |
00000010  12 00 00 00 01 00 01 00  80 3e 00 00 00 7d 00 00  |.........>...}..|
00000020  02 00 10 00 64 61 74 61                           |....data|
00000028
```

(Note that the original file is much larger.  This was the smallest sub-file
I could find that would generate the crash.)

Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
2022-10-13 08:56:49 +00:00
George Kiagiadakis
8dd512fd9f tests/check/rtpsession: extend test_internal_sources_timeout
to verify that rtx SSRCs do not BYE after timeout

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00
Jan Schmidt
3c2c4bbe2c dashdemux2: fix mpd unit test expectations
Update unit test for some mpd cases that were reporting
timestamps including the period start time, while
dashdemux2 expects that it needs to add the period
start time itself.

Fix the tests to not expect the period start time
to be included.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3025>
2022-09-27 00:00:49 +00:00
Tim-Philipp Müller
2ac5d687e1 tests: add a few more orc tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3029>
2022-09-15 12:14:56 +01:00
Edward Hervey
855dabb578 dashdemux2: Remove bogus limitation checks for duration fields
Just like for the seconds field, there are no limitations on the hours and
minutes fields. The specification for xml schema duration fields doesn't forbid
specifying durations with only (huge) minutes or hours values.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2951>
2022-09-07 12:48:28 +00:00
Sebastian Dröge
cc454f0fc3 rtpjitterbuffer: Add test for crash caused by removing timers twice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
2022-09-03 09:26:24 +00:00
Patricia Muscalu
3c9e4f4886 rtph265: keep delta unit flag
Without this patch all buffers that pass the payloader
are marked as non-delta-unit buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2969>
2022-09-02 08:56:13 +00:00
Raul Tambre
e1d3612321 rtpjitterbuffer: remove lost timer for out of order packets
When receiving old packets remove the running lost timer if present.
This fixes incorrect reporting of a lost packet even if it arrived in time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2922>
2022-09-01 09:01:31 +00:00
Seungha Yang
b233df3537 splitmuxsink: Don't crash on EOS without buffer
Fix a case where upstream pushed EOS without buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2174>
2022-07-05 11:33:35 +00:00
Thibault Saunier
339f950e79 rtprtx: Fix copying extension headers
There was a typo leading to reading memory from the buffer we were
writing to.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2696>
2022-07-04 19:20:57 +00:00
Tim-Philipp Müller
90090dc13b tests: udpsink: make test work in environments without IPv6
Part-fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/939

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2659>
2022-07-02 11:57:31 +00:00
Edward Hervey
787dbfd4e4 tests: check: Update hlsdemux2 tests for playlist changes
We no longer do auto-magic fallbacks when synchronizing a disconnected
playlist. It is handled at a higher level.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2679>
2022-06-28 17:59:24 +00:00
Tim-Philipp Müller
c895cdbec8 tests: skip unit tests for dependency-less elements that have been disabled
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1136

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2660>
2022-06-27 07:05:00 +00:00
Piotr Brzeziński
0a2c490723 adaptivedemux2: Prevent duplicate symbols on static builds
Uses prelude header files with #defines to rename DASH and MSS
symbols duplicated in their old standalone versions.
Also redefines soup-related functions when building it for
adaptivedemux2 to prevent symbol conflicts there.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2534>
2022-06-15 01:13:16 +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
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
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
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
Xavier Claessens
b004464ac6 Remove glib and gobject dependencies everywhere
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.

While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
2022-04-01 16:32:17 +00:00
Stéphane Cerveau
1170ab3c29 wavparse: handle query in any parse state
In order to create the stream_id, we need to
pass the query to the default query handler.

If the parse state is different from GST_WAVPARSE_DATA
the query should be passed to the default query
handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1987>
2022-03-22 16:25:35 +00:00
Matthew Waters
206021e4d4 rtpmanager/rtx: implement initial support for reading/writing rid extensions
Two RTP Header extensions are very relevant for rtprtxsend/receive.
1. "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id": will always be removed
2. "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id": will be written
    instead of the "rtp-stream-id" header extension.

Currently it's only a simple replacement of one header extension for
another however a future change would only add the relevant extension
based on some heuristics (like, video frames only on one of the rtp key
frame buffers, or only until the rtx ssrc has been validated by the peer)
in order to reduce the required bandwidth.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
2022-03-21 03:18:18 +00:00
Matthew Waters
33be3e5936 test: add tests for sdes-based RTP header extensions
mid, stream id and repaired stream id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1759>
2022-03-21 03:18:18 +00:00
Havard Graff
e5bd9839c4 rtprtxsend: don't require clock-rate in caps
For multiplexing, the rtpstreams you are multiplexing might not use
the same clock-rate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1881>
2022-03-15 19:05:00 +00:00
Havard Graff
4d31641302 rtprtxsend: don't start the task unless we are doing rtx
The rtxsend element can do pass-through when not enabled (no pt-map set)
and in those cases there is no point in starting an additional task
that does absolutely nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1880>
2022-03-15 12:03:27 +00:00
Carlos Rafael Giani
671c89c392 mpg123: Add gapless playback support
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
2022-03-14 10:32:15 +02:00
Carlos Rafael Giani
0431a0845c mpegaudioparse: Support gapless playback
Gapless playback is handled by adjusting buffer timestamps & durations
and by adding GstAudioClippingMeta.

Support for "Frankenstein" streams (= poorly stitched together streams)
is also added, so that gapless playback support doesn't prevent those
from being properly played.

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1028>
2022-03-14 10:32:15 +02:00
Mikhail Fludkov
815d279f2e rtprtxreceive: fix crash when RTX payload has zero length
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1875>
2022-03-08 09:07:41 +00:00
Havard Graff
2a26daee46 rtprtx: signed/unsigned and style fixes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1872>
2022-03-07 21:16:45 +00:00
Sebastian Dröge
b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00
Corentin Damman
31f4444724 rtpjitterbuffer: fix typo in tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1466>
2021-12-23 14:31:27 +00:00