Commit graph

9 commits

Author SHA1 Message Date
Enrique Ocaña González
92a4cfe20f qtdemux: Don't emit GstSegment correcting start time when in MSE mode
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3867>
2023-02-06 12:42:49 +00:00
Thibault Saunier
6b30a5d987 adaptivedemux2: Generate proper stream-id taking into account upstream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3160>
2023-02-01 22:26:34 +00:00
Sebastian Dröge
9c0d9882e3 gst-integration-testsuites: Update cenc_audio_esds_property_overrides expected output file
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
2022-12-17 19:30:51 +02:00
Thibault Saunier
1c1b0380cb dashdemux2: Fix the way we determine current_position after seeks
Without that the current_position was off after seeks, potentially
leading to not properly push a last fragment when a `.stop` time was
set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Thibault Saunier
31acfcd875 decodebin3: Do not try to plug a decoder on raw formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3123>
2022-10-06 08:41:49 +00:00
Andoni Morales Alastruey
ffdc8634e8 test: update tests to include the new meta
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1458>
2022-06-03 08:29:05 +00:00
Vivia Nikolaidou
cb8da91f7a h264parse: Include coded-picture-structure info in caps
This reverts commit 652773de36 and
modifies it to rename the caps field name to coded-picture-structure.

It was previously removed because it confuses the decoder and we didn't
have a valid use case for including it in the encoded caps at this
stage. We now do have such a use case but still don't want to confuse
the decoder, so the field is renamed.

However, it is still not accurate without looking at the SEI picture
structure of each frame, so it was named coded-picture-structure. If its
value is "frame" it is most likely progressive, if it's "field" it is
most likely interlaced or mixed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2177>
2022-04-18 11:00:29 +00:00
Thibault Saunier
49055f1cd5 rtph264pay: Handle 'profile' field
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
2021-12-12 10:59:00 -03:00
Thibault Saunier
098b876985 Import gst-integration-testsuites 2021-09-24 16:29:33 -03:00