Hi, I was wondering why adaptive bitrate of LL-HLS is not working based on my setting.
█ Here’s Flash Media Encoder setting
URL: rtmp://172.16.46.143:1935/live
Stream: myStream
█ SMIL file
█ Also, enable the transcoder > Transrate (Default)
including source, 720p, 360p, 160p
█ On Incoming Stream page, can see Default Instance (definst) including streams above
streaming is working under
https://172.16.46.143/live/myStream/playlist.m3u8
replacing “myStream” with “myStream_160p”, “myStream_360p”, “myStream_720p” are also working
but not working with
http://172.16.46.143:1935/live/myStream/smil:myStream.smil/playlist.m3u8
first, downloading m3u8 file is passed “172.16.46.143:1935/live/myStream/smil:myStream.smil/playlist.m3u8”
2nd, dowloading chunk got 404 error “172.16.46.143:1935/live/myStream/smil:myStream.smil/chunklist_w877383523_b744100_sleng.m3u8”
Is anything wrong with current setting?
my test player is “akamai players/hlsjs”
thanks
after correct the myStream_source bitrate which is the same as myStream_720p
now I got another error
HTTPStreamerAdapterCupertinoStreamer.onPlaylist: Rendition is not supported [live/myStream/smil:myStream.smil/chunklist_w665802848_b744100_sleng.m3u8]: AUDIOVIDEO
PlaylistDelegate.getManifest[app[live] stream[myStream_720p] format[CMAF], type[AUDIOVIDEO]]: Manifest doesn’t exist: live/myStream/smil:myStream.smil/chunklist_w665802848_b744100_sleng.m3u8
Welcome to our community @jesper_lai. 
What is the audio codec you are using?
Usually we check to make sure your stream names and bitrates/resolutions match for all the renditions, which they look like they do. So, if it is saying rendition not supported, it can potentially be the codecs, usually the audio codec. Is it aac?
If it’s working in wowza test players or vlc and not in the akamai player, we’ll need a support ticket to run some tests to see why not.
many thanks, finally, the issue was fix automatically and I did nothing changed excluding correct the birate
btw, may I know while using LL-HLS, the shortest latency is just about 1.5s ~ 3s?
or… it’s possible to have shorter latency after changing something else?
thanks
Oh so sorry, missed your reply, I didn’t get a notification. I can tell you what we are consistently seeing on average with our extensive testing is 2 to 2.5 seconds. If you need anything lower, you might want to consider WebRTC? It also depends on if you are using a CDN or not since that is another potential latency add-on before it reaches the viewer.
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