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Eyevinn / srt-whep

SRT to WHEP (WebRTC)
Apache License 2.0
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Compatibility with ffmpeg #37

Closed Wkkkkk closed 1 year ago

Wkkkkk commented 1 year ago

We noticed that the SRT stream from ffmpeg can not be played out (no SDP offer generated from whipsink).

Wkkkkk commented 1 year ago

Some example streams could be generated and played out with:

ffmpeg -f lavfi -i testsrc=size=640x360:rate=30 -f lavfi -i sine=frequency=440 -pix_fmt yuv420p -c:v libx264 -vprofile baseline -c:a aac -f mpegts "srt://127.0.0.1:1234?mode=listener"

gst-launch-1.0 srtsrc uri="srt://127.0.0.1:1234?mode=caller" ! queue ! decodebin name=d \
    d. ! queue ! autovideosink \
    d. ! queue ! audioconvert ! autoaudiosink

High video profile is supported too.

Wkkkkk commented 1 year ago

To play out the stream, it is necessary to transcode it first.

## Generate SRT stream with ffmpeg
ffmpeg -f lavfi -i testsrc -pix_fmt yuv420p -c:v libx264 -vprofile baseline -f mpegts "srt://127.0.0.1:1234?mode=listener"

## Transcode the stream
gst-launch-1.0 srtsrc uri="srt://127.0.0.1:1234?mode=caller" ! queue ! decodebin ! queue ! videoconvert ! \
    x264enc tune=zerolatency ! video/x-h264, profile=constrained-baseline ! \
    mpegtsmux ! srtsink uri="srt://127.0.0.1:1235?mode=caller" wait-for-connection=false

## Connect
GST_DEBUG=1 cargo run --release -- -i 127.0.0.1:1235 -o 127.0.0.1:8888 -p 8000 -s listener | bunyan

## Play with WebRTC Player, URL:
http://localhost:8000/channel
Wkkkkk commented 1 year ago

We noticed an incredibly high computing overhead when using ffmpeg to generate an SRT stream. And it runs into errorav_interleaved_write_frame(): Input/output error after 10 seconds.