vps/sps/pps in codec_data shouldn't be considered as inband data.
Otherwise, h26{4,5}parse never insert them to nal when transform
(packetized to byte-stream) use case
Similar change as the on I did in h264parse. We want to process SEI
recovery point as keyframe so muxers will mark them as seek points and
decoders will be able to start decoding from them rather than waiting
for an IDR.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/790
The spec states that "recovery point SEI message assists a decoder in
determining when the decoding process will produce acceptable
pictures for display after the decoder initiates random access or after the
encoder indicates a broken link in the coded video sequence."
Mark those as keyframes so muxers will mark them as seek points and
decoders will be able to start decoding from them rather than waiting
for an IDR.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/790
This removes the crossfade-ratio property and replaces it with an
operator property. Currently this implements the following operators:
- SOURCE: Copy over the source and don't look at the destination
- OVER: Default blending of the source over the destination
- ADD: Like OVER but simply adding the alpha instead
See the example for how to implement crossfading with this.
https://bugzilla.gnome.org/show_bug.cgi?id=797169
If the first audio buffer to be dropped started right between two video
buffers (after the end of the first but before the start of the second,
as is often the case with N/1001 video frame rates), we would miss
sending the dropping=true message.
https://bugzilla.gnome.org/show_bug.cgi?id=797248
tsdemux expects a custom descriptor (GST_MTS_DESC_AC3_AUDIO_STREAM)
to detect a stream as AC3 and not EAC3.
Note that tsdemux expects this descriptor because mpegtsmux writes
a stream with a HDMV registration descriptor.
Fixes:
gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! ac3parse ! mpegtsmux ! \
tsdemux ! ac3parse ! avdec_ac3 ! audioconvert ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=797220
Previously it was dispatched before the last video buffer, and audio
buffers would follow afterwards. It's misleading to send the
dropping=true message before both streams have really stopped, it can
lead to races when someone is e.g. waiting for that message to send EOS.
Also added some debug output.
https://bugzilla.gnome.org/show_bug.cgi?id=797145
Direct applying the commit 7bb6443. This could fix also unexpected
nal dropping when nonzero "config-interval" is set.
(e.g., gst-launch-1.0 videotestsrc ! x265enc key-int-max=30 !
h265parse config-interval=30 ! avdec_h265 ! videoconvert ! autovideosink)
Similar to the h264parse, have_{vps,sps,pps} variables will be used
for deciding on when to submit updated caps or not, and rather mean
"have new SPS/PPS to be submitted?"
See also https://bugzilla.gnome.org/show_bug.cgi?id=732203https://bugzilla.gnome.org/show_bug.cgi?id=754124
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.
