Commit graph

11 commits

Author SHA1 Message Date
Thibault Saunier 019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00
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
Sanchayan Maity 248d2bb795 audiobuffersplit: Add support for specifying output buffer size
Currently for buffer splitting only output duration can be specified.
Allow specifying a buffer size in bytes for splitting.

Consider a use case of the below pipeline
appsrc ! rptL16pay ! capsfilter ! rtpbin ! udpsink

Maintaining MTU for RTP transfer is desirable but in a scenario
where the buffers being pushed to appsrc do not adhere to this,
an audiobuffersplit element placed between appsrc and rtpL16pay
with output buffer size specified considering the MTU can help
mitigate this.

While rtpL16pay already has a MTU setting, in case of where an
incoming buffer has a size close to MTU, for eg. with a MTU of
1280, a buffer of size 1276 bytes would be split into two buffers,
one of 1268 and other of 8 bytes considering RTP header size of
12 bytes. Putting audiobuffersplit between appsrc and rtpL16pay
can take care of this.

While buffer duration could still be used being able to specify
the size in bytes is helpful here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1578>
2020-09-21 15:17:18 +00:00
Sebastian Dröge 79e65951a9 audiobuffersplit: Perform discont tracking on running time
Otherwise we would have to drain on every segment event. Like this we
can handle segment events that don't cause a discontinuity in running
time to be handled without draining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
2020-05-11 07:25:39 +00:00
Sebastian Dröge 20756e3387 audiobuffersplit: Keep incoming and outgoing segments separate
We might have to drain already queued input based on the old segment
before forwarding the new segment event. The new segment is only
forwarded after a discont as otherwise we might cause unnecessary
timestamp jumps as we output buffers timestamped based on sample counts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
2020-05-11 07:25:39 +00:00
Vivia Nikolaidou ce0be4d1ac audiobuffersplit: Added max-silence-time property 2019-02-21 15:16:37 +00:00
Sebastian Dröge f19edc8c83 audiobuffersplit: Add a gapless mode which inserts silence/drops samples on disconts
The output is always a continguous stream without any gaps.
2018-08-17 16:40:16 +03:00
Sebastian Dröge b2602a459b audiobuffersplit: Keep track of resync time separately
If we drain after a discont, the discont time given by the stream
synchronizer is already the time after the discontinuity. But we need to
drain all pending data based on the previous discont time instead.
2018-08-17 16:40:16 +03:00
Sebastian Dröge dd490e1555 audiobuffersplit: Use new GstAudioStreamAlign API
https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:13:17 +03:00
Vivia Nikolaidou 668c44072b audiobuffersplit: Add strict-buffer-size property
If set to TRUE, any last audio samples too small to fill a buffer will
be discarded.

https://bugzilla.gnome.org/show_bug.cgi?id=779064
2017-02-22 21:01:46 +02:00
Sebastian Dröge 0acb3d87bb audiobuffersplit: New element that splits raw audio buffers into equal-sized buffers
This is useful e.g. if audio buffers should be exactly the duration of a
video frame, or if a audio buffers should never be too large because of
latency constraints.

The element is taking a fractional buffer duration, to allow working
with e.g. 1001/30000 as output duration and it accumulates rounding
errors in the buffer durations and compensates for them by making some
buffers one sample larger than the others.

https://bugzilla.gnome.org/show_bug.cgi?id=774689
2016-11-23 18:18:46 +02:00