Commit graph

21 commits

Author SHA1 Message Date
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
Jan Schmidt
1b3ba87d13 audiobuffersplit: Calculate the correct size for fixed size buffers
Fix the output-buffer-size property to do what it says by calculating
the correct audio buffer size for that target size, rounded down to
the nearest whole number of samples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1887>
2020-12-17 04:41:18 +11:00
Marijn Suijten
dc90a3d3cf audio: Use new AudioFormatInfo::fill_silence function
The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
2020-11-26 10:06:42 +02:00
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
ea5f38440d audiobuffersplit: Specify in the template caps that only interleaved audio is supported
Needs special support for non-interleaved audio and e.g. use the
GstPlanarAudioAdapter.

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/779

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1363>
2020-06-23 10:03:53 +03:00
Sebastian Dröge
bd67ef18e9 audiobuffersplit: Unset DISCONT flag if not discontinuous
And also set/unset the RESYNC flag accordingly.

It can happen that the flag is preserved by GstAdapter from the input
buffer. For example if a big input buffer is split into many small ones,
each of the small ones would have the flag set.

All other buffer flags seem safe to keep here if they were set,
including the GAP flag.

Also ensure that the buffer is actually writable before changing any
flags or metadata on it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1298>
2020-05-25 12:41:32 +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
Johan Bjäreholt
962ca01596 audiobuffersplit: Fix format string warning
We have a GST_DEBUG_OBJECT, which prints a guint64 with %lu which gave a
compiler warning. Used G_GUINT64_FORMAT instead.

https://bugzilla.gnome.org/show_bug.cgi?id=797036
2018-08-28 11:12:45 +03:00
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
2f761b89df audiobuffersplit: Always set DISCONT flag after resyncs 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
186870d10a audiobuffersplit: Update output buffer size after each buffer to compensate for accumulated errors
https://bugzilla.gnome.org/show_bug.cgi?id=796981
2018-08-17 16:40:16 +03:00
Sebastian Dröge
d01724a9ab audiobuffersplit: Drain pending samples if the caps are changing
https://bugzilla.gnome.org/show_bug.cgi?id=787560
2017-09-28 14:13:44 +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
Sebastian Dröge
6026d12d90 audiobuffersplit: Allow changing the output-buffer-duration at any time
Previously this was only allowed before the CAPS event arrived.
2017-05-11 18:41:08 +02:00
Sebastian Dröge
971a481a8c audiobuffersplit: Install srcpad query function on the srcpad, not the sinkpad 2017-03-22 13:23:11 +02:00
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
5cff156f2c audiobuffersplit: Don't try to output anything if samples_per_buffer == 0
This happens if we had no CAPS event yet but e.g. got an EOS event. We
would then try to output a 0-sized buffer, but getting that from the
adapter will give an assertion, return NULL and then crash.
2016-12-23 13:27:42 +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