- This topic has 13 replies, 6 voices, and was last updated 14 years, 9 months ago by
.
-
Posts
-
While were on the subject of this…..
What is the most reliable best sounding encoder at low bit rates?
Basically, to cut a long story short….there’s a radio station that want’s to use a private 1-to-1 stream as an STL but here’s the catch….there’s no broadband connection in the studio and the only way is a mobile broadband 3G dongle.
So….what type of encoder? and what is the lowest bitrate possible without killing the quality?
Any ideas? 🙂
I guess you better start a new topic for that, i am also going a little bit offtopic… 😳 And the answer to your question is AAC+V2 i guess. The lowest bitrate without hearing loss of audio quality.. hmm, 64 kbps?
in the 0 to 48kbps area, I would go with the latest Coding Technologies PS-AAC codec.
In the 48-96 area it’s pretty much a tie between CT’s latest HE-AAC, and the current aoTuV tuning of Vorbis (v5.7)… Vorbis has the advantage the higher the bitrate goes (by around 64kbps), and sits very pretty between the weak point of the transition from HE-AAC to LC-AAC, in the 80 to 160 kbps area.
What I would do is go with Vorbis at 128-ish kbps if you have the STABILITY at that bitrate. Otherwise go with PS-AAC at 48 kbps.
[quote author=”JesseG”]in the 0 to 48kbps area, I would go with the latest Coding Technologies PS-AAC codec.
In the 48-96 area it’s pretty much a tie between CT’s latest HE-AAC, and the current aoTuV tuning of Vorbis (v5.7)… Vorbis has the advantage the higher the bitrate goes (by around 64kbps), and sits very pretty between the weak point of the transition from HE-AAC to LC-AAC, in the 80 to 160 kbps area.
What I would do is go with Vorbis at 128-ish kbps if you have the STABILITY at that bitrate. Otherwise go with PS-AAC at 48 kbps.[/quote]
OK nice one, I’m going to give the aoTuV Vorbis 5.7 a try at 128 kbps and see how it goes…..
What do I have to do (or not do)or download differently from Leifs settings on this thread to get this going? any info or weblinks would be great. 😉
http://www.claessonedwards.com/forum/viewtopic.php?f=5&t=247
This is what I get at the studio end when I send material from home.
This is just sent with my playout system playing WAV files and feeding into Opticodec and then to my shoutcast server (all on the same machine). My machine at the station end has Winamp connecting back to my Shoutcast Server.
This is with Playout -> Breakaway Live -> Opticodec -> Shoutcast. Opticodec set to 48kbps AACPlus Stereo V2.
This is with Playout -> Breakaway Live -> Opticodec -> Shoutcast. Opticodec set to 64kbps AACPlus Stereo.
I tried to capture the same track at the same spot for each screenshot. They’re pretty close but you get the picture.
There is the usual 16 second delay between send and receive but for what I use it for that makes no difference to me. It is just a much better signal than I get with either AudioTX or the Comrex Access system.
Scott
[quote author=”Lee XS”]OK nice one, I’m going to give the aoTuV Vorbis 5.7 a try at 128 kbps and see how it goes…..
What do I have to do (or not do)or download differently from Leifs settings on this thread to get this going? any info or weblinks would be great.[/quote]
just EdCast and preferably IceCast v2 for Tx, and a vorbis codec for EdCast… which you can get here:
http://www.rarewares.orgOptionally you can try FAAC with the LD profile, short for "Low Delay"… which works in EdCast. But FAAC developers admit it’s not the best tuned AAC codec out there, and you also would have to optimize the Tx and Rx buffering as well.
What is also a viable option, if you have some skills, is to use the Linux version of Nero’s free AAC codec to hook into a low delay streaming app via std-in/std-out… Nero’s codec supports the LD profile.
There’s also Nullsoft’s transcoder for Linux with CT’s codec, but I’m not sure if you can input directly from a soundcard’s input via std-in or what. If it accepts RIFF WAV format on the input, then that might be an option too, if you configure the shoutcast server to have very low Tx delay, and use a very small buffer on the Rx. However… I’m not sure it even supports the LD profile. I think the profile might be based on presets picked via the desired bitrate, same as Nullsoft’s Windows build of the codec.
Food for thought if you ever want to go with LD-AAC.
Thanks…I’m gonna try the Vorbis for now…..
Already got Edcast and Icecast hooked up…just sorting the Vorbis side of things out now, see how it goes. 🙂
May I ask something too ?
If you have enough bandwidth (like me) is it wise to use OGG FLAC ?Also if I finally choose ogg vorbis what is the best settings in edcast (Quality setting or bitrate) ?
[quote author=”bamp”]May I ask something too ?
If you have enough bandwidth (like me) is it wise to use OGG FLAC ?Also if I finally choose ogg vorbis what is the best settings in edcast (Quality setting or bitrate) ?[/quote]
OggFLAC is very stable over a good network connection. That’s what Slim Server uses, for all of the Computer/Network -> Slim Devices connection of audio.
for Vorbis, quality based setting is better in general, but for streaming you generally will want to have a predictable bitrate with a codec that has longer frames. Of course you could always see how a quality-based setting works on your connection & software used.
Ok so the Mobile Broadband thing didn’t work out as it would of cost around £200 per month just in credit for the bandwidth and that’s even if it was reliable enough with a constant 3G signal!!! 🙁 🙁 🙁
On the plus side, I’ve now got an Ogg vorbis stream set up doing nothing……so if any of you guys are curious to see what an OGG stream sounds like at 128mbps then check this out…..(copy & paste URL into your winamp etc)
http://87.98.177.199:8700/stream.ogg
🙂 🙂 🙂
- The forum ‘Breakaway Professional Products – [discontinued]’ is closed to new topics and replies.