Commit graph

14 commits

Author SHA1 Message Date
Jan Alexander Steffens (heftig)
0596da1966 mpegtsparse: Don't assert the packet_size when filling for EOS
If the packetizer got reset for any reason (failure to find PCR?) then
the packet_size can be zero here even though we already enqueued some
packets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1038>
2021-10-05 09:38:27 +00:00
Seungha Yang
90e5e0efea mpegtsmux: basetsmux: Don't try to return value from void function
gstbasetsmux.c(1508): warning C4098: 'free_splice': 'void' function returning a value

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1018>
2021-10-04 13:37:09 +00:00
Marc Leeman
58d4a5b449 ristsink: set sync to FALSE on RTCP sink
See commit 921e9a54: rtpsink: set sync off on rtcp_sink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993>
2021-10-01 22:57:02 +00:00
Marc Leeman
b7820a0de7 rtpsink: set sync off on rtcp_sink
When using the following setup (the error can be reproduced using
simpler sender pipelines), the receiver resynchronises the clock on RTCP
packets. The effect was that a couple seconds were cut out of the
playback because an initial RTCP packet was dropped.

When sending out all RTCP packets (setting sync=FALSE on the RTCP
updsink), the playback is fine.

This syncs rtpsink with rtpsrc (where this property was already set).

gst-launch-1.0 filesrc location=899-en.mp3 \
    ! mpegaudioparse \
    ! mpg123audiodec \
    ! audioconvert \
    ! audioresample \
    ! avenc_g722 \
    ! rtpg722pay
    ! rtpsink uri=rtp://239.1.2.3:1234

gst-launch-1.0 uridecodebin rtp://239.1.2.3:1234?encoding-name=G722 \
    ! autoaudiosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993>
2021-10-01 22:57:02 +00:00
Marc Leeman
a774dfb18f rtpmanagerbad: do not set iface on sink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/994>
2021-10-01 20:31:17 +00:00
Mathieu Duponchelle
f0a158407c mpegts: add missing Since comments after SCTE 35 work
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:38 +00:00
Mathieu Duponchelle
555a5ea9dd basetsmux: use private copy of g_ptr_array_copy
This function is only present since glib 2.62

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:38 +00:00
Mathieu Duponchelle
c2eeb639b0 basetsmux: fix SCTE pts_adjustment with offsets
When there are elements between the demuxer and the muxer that
introduce an offset to the running time, or when offsets are
set on pads by the application, this shift must be taken into
account when calculating the final pts_adjustement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:38 +00:00
Mathieu Duponchelle
c3a161f287 basetsmux: rework SCTE section handling to handle passthrough
mpegtsmux can receive SCTE sections from two origins: events
created by the application, and events forwarded downstream by
mpegtsdemux, containing sections that may not have been fully
parsed, and additional data to help tsmux translate times to
the correct domain, both for requesting keyframes and calculating
an accurate pts_adjustment.

The complete approach is documented further in a comment above
the relevant function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:38 +00:00
Mathieu Duponchelle
4af003bc02 tsdemux: switch SCTE 35 sections handling to a passthrough model
Instead of modifying the splice times in the incoming sections
to running time and expecting eg mpegtsmux to convert those back
to its local PES time domain, which might be impossible when
those splice times are encrypted or the specification is extended,
transmit the needed information to the muxer as separate fields in
the event:

* A pts offset field can be used by the muxer in order to calculate
  a final pts_adjustment

* A rtime_map can be used by the muxer to determine the correct
  running times at which it should request keyframes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:38 +00:00
Mathieu Duponchelle
be4d0fff23 basetsmux: extend SCTE 35 support
Makes it possible to support passing SCTE 35 cue points from
demuxer to muxer, while preserving correct timing.

This will also improve ex nihilo cue points injection, as splice
times and durations are now interpreted as running time values,
and may trigger key unit requests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:37 +00:00
Mathieu Duponchelle
1ca08bff57 tsdemux: Expose send-scte35-events property
When enabled, SCTE 35 sections (eg ad placement opportunities)
are forwarded as events donwstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:37 +00:00
Mathieu Duponchelle
b2718ed6cf mpegtsbase: expose vmethod to let subclass handle sections
This can be used by tsdemux to handle and forward SCTE 35
sections.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
2021-09-25 01:29:37 +00:00
Thibault Saunier
019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00