Comments (4)
mjnk9xw
commented on June 4, 2024
2
I did it. I use 2 video and audio streams to add to the ts file.
from media-server-go-demo.
AlexanderKozhevin
commented on June 4, 2024
@notedit can you contact me?
from media-server-go-demo.
benpeterswake
commented on June 4, 2024
@mjnk9xw hey, could you share your solution?
from media-server-go-demo.
mjnk9xw
commented on June 4, 2024
@benpeterswake I convert 2 separate audio and video streams. video: h264parse, audio: opusparse, convert opus to voaac
pipeline of me : appsrc name=video is-live=true do-timestamp=true
! h264parse disable-passthrough=true ! queue
! mpegtsmux name=mux ! queue
! hlssink sync=false playlist-length=5 max-files=10 target-duration=5 playlist-location=path/index.m3u8 location=path/fragment%05d.ts appsrc name=audio is-live=true ! opusparse ! opusdec ! audioconvert ! audio/x-raw,rate=16000 ! voaacenc ! queue ! mux.
from media-server-go-demo.
Related Issues (20)
- ../include/media-server/include/rtpsession.h:10:10: fatal error: srtp2/srtp.h: No such file or directory #include <srtp2/srtp.h> HOT 15
- rtmp-to-webrtc demo run fail HOT 7
- rewrite the gstreamer part with gst HOT 1
- Bitrate Change HOT 1
- RTMPS support ? HOT 2
- [webrtc-to-hls] Destination path for hls files. HOT 1
- BEWARE OF MISSING PLUGINS! webrtc-to-hls files not being created. HOT 1
- Panic runtime error with VP8 HOT 2
- 编译错误
- panic: create pipeline error HOT 2
- webrtc-to-rtmp demo can't work HOT 5
- No video HOT 1
- demo raw-rtp-input cannot be played using chrome
- send raw-rtp-input use rtp
- problem with NAT HOT 1
- We don't have an DTLS setup yet
- webrtc to rtmp demo error: Required gstreamer plugin faac not found
- webrtc-to-rtmFp
- Required gstreamer plugin faac not found
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