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64 lines
2.1 KiB
Plaintext
64 lines
2.1 KiB
Plaintext
- rtmp2sink: Should look into reconnecting and resuming stream without
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deleting and recreating stream, which drops clients.
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- Move AMF parser/serializer to GstRtmpMeta?
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- Move AMF nodes from g_slice to GstMiniObject?
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- First video frame that comes from Wowza seems to be out-of-order; librtmp
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does not have this problem
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- Refactor connection, pull out the ad-hoc read and write handling and put it
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with the chunk layer into GBuffered{In,Out}putStream subclasses
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- Refactor elements and pull out the common connection+mainloop handling code
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into a context object
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- Change the location properties into something with less boilerplate?
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Perhaps a GstStructure-based prop, custom GValue transforms or GstValue
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(de)serializing
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- Use glib-mkenums to generate GEnumClasses
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- Post-connect onStatus handling (needed for src EOS and async errors?)
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- Better mux/demux, at the cost of losing compatibility with flvmux/demux.
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Something like (a/x = application/x-rtmp-messages):
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rtmp2src ! a/x ! rtmp2demux ! a/x,type=video ! rtmp2videodecode ! h264parse
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! a/x,type=audio ! rtmp2audiodecode ! aacparse
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x264enc ! rtmp2videoencode ! a/x,type=video ! rtmp2mux ! a/x ! rtmp2sink
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fdkaacenc ! rtmp2audioencode ! a/x,type=audio !
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And also, in case no muxing is required:
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x264enc ! rtmp2videoencode ! a/x,type=video ! rtmp2sink
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fdkaacenc ! rtmp2audioencode ! a/x,type=video ! rtmp2sink
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Proper GstBuffer timestamps need proper timestamp wraparound handling
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- Better client element, which generalizes the existing sink/src to allow
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multiple streams over one connection
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- Request src pad to play a stream
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- Request sink pad to publish a stream (base it on GstAggregator?)
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- rtmp2sink/src just specialize the client element with a static pad
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- Server implementation
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- Support more protocols
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- rtmpe (App-layer encryption)
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- rtmpt (HTTP tunneling)
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- rtmpte (HTTP tunneling + App-layer encryption)
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- rtmpts (HTTPS tunneling)
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- rtmfp (UDP)
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Needed testing:
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- AMF parsing
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- connection closure by peer
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- connection timeouts
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