Commit graph

40 commits

Author SHA1 Message Date
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
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
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
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
Mathieu Duponchelle
5dc280de9f rtp/redenc|ulpfecenc: add support for TWCC
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.

In ulpfecenc we add one in that case to our protection buffers.

This makes TWCC functional when UlpRed is used in webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
2021-12-14 03:26:56 +00:00
Mathieu Duponchelle
4412198c05 rtpfunnel: fix extmap handling on accept-caps
Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.

I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
2021-11-23 20:26:30 +00:00
Mathieu Duponchelle
97d83056b3 rtpfunnel: don't enforce twcc during upstream negotiation
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.

Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
2021-11-12 18:40:32 +00:00
Tim-Philipp Müller
ea8dc0c737 Couple more g_memdup() -> g_memdup2() fixes
Fixes deprecation warnings with newer GLib versions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>
2021-10-30 10:37:37 +01:00
Olivier Crête
0dbe0e21fe rtphdrext-clientaudiolevel: Rename RFC 6464 element
Multiplying elements named after RFC numbers is confusing,
so let's give them meaningful names.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1125>
2021-10-20 00:03:09 +00:00
Philippe Normand
c3455def2e soup: Runtime compatibility support for libsoup2 and libsoup3
The src and sink elements no longer link against libsoup. It is now loaded at
runtime. If any version is resident already, it is used. Otherwise we first try
to load libsoup3 and if it's not found we fallback to libsoup2.

For the unit-tests, we now build one version of the test unit file per libsoup
version found. So if both libsoup2 and libsoup3 are available on the host, the
CI will cover them both.

Based on initial patch by Daniel Kolesa <dkolesa@igalia.com> and
Patrick Griffis <pgriffis@igalia.com>.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1044>
2021-10-13 08:32:25 +00:00
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00