[LAU] Audio streaming work flow

I have been fooling around with an Internet radio station setup. Just to see what kinds of trouble might show up. There seem to be two sides to broadcasting software. Audio and admin. I don't know if I can really cover the admin side well, both rivendell and airtime seem to work for that. For that matter I have heard of some people using windows apps for doing admin and playlist creation while using Linux for audio streaming. I will cover the audio part for now. Audio is divided into two parts, studio and server. A number of people use a streaming service rather than their own server because most people have a dynamic IP and so name service is problematic. It is often cheaper to use a streaming service. In any case the server side install seems to be trivial, install icecast2, configure and run. I found that part easy. I set it up on a 10 year old 1.2Ghz Celeron box with Ubuntu Server installed. Quick and easy, basically set up the passwords, enable and start the init.d/icecast2 script. Done. The studio side takes a bit more. I will note here that the studio side setup also makes a good podcast recorder. I am not so sure about live DJ work as many live DJs consider video a large part of their show. I don't know if they would consider the video part as truly separate or expect to have one application that does it all. The response I have gotten to mixxx is that it is "not ready" for prime time use. My personal experience is that: a) it doesn't like the open nvidia driver (crashed xorg on me). b) it's window is too big for at least one of my screens (netbook). I think the first problem is known and may even be fixed by now. But the second is more troubling. Podcast recording or remote broadcast might well want to be done on a small screen computer like a netbook or notepad. The window is not even shrinkable to screen size with scroll-bars. Though once the user got used to kb shortcuts that might not be an issue. IDJC seems to be much better. It is the recommended studio solution for airtime and interfaces well with jack. That is what I used for my setup and at least one of our users uses it for their internet radio station as well. It is small enough to be used on any screen... well maybe not a smart phone :) It allows the unattended streaming of prerecorded content as well as mixing live audio with that. All audio content (including MP3s, OGGs etc.) are decoded to audio at the rate jack is running and remixed with whatever audio the users wishes to add. It has dedicated jack ports for VOIP or landline audio available. It has jack ports for insert purposes after mixing and before streaming. This allows the use of a compressor or eq or even outboard equipment. This is also the place to get audio out for live use or live broadcast (think transmitter, though I would think it may go through some other stuff before that). So far great. There is an online setup and use guide for IDJC that covers about everything. The last bit is still the roughest. Integration of desktop sounds with jack. The most common use for this is VOIP generally skype, though the problem is the same with almost all of these kinds of apps. The most common thing is doing remote interviews and the most common work around is to route audio outside the computer via pulse and then back in though audio ports. The PA-jack bridge is getting better, but has two limitations at this time. The PA that ships is 2.*, the new one I can find is 3.0 and both of these have the same problem. The first is a bug where jack will not start while PA is streaming. There is a fix, but it is not in any repo yet... there is also a workaround... but that is not easy to set for install. The second problem for those with a sound card with more than two ports, is that PA defaults to as many ports as the physical device has. The more ports PA has to deal with the more CPU it uses... with an ICE1712 based audio device that is 12/10. A default of stereo would be better. I am told the version currently in git has the ability to set the number of channels. I have asked for a default of two, but being able to set the number is second best. It is possible to set the PA-jack bridge not to auto connect and the method of doing so should be documented for this workflow. Using a jack session seems ideal with this workflow. The pa bridge can be connected to the correct ports as they appear. The nice thing about using the pa bridge for these things is that once PA is running it is always there unlike many desktop apps that only show up when they are running and therefore have to be manually connected after they connect to the sound server. Some good open conferencing SW that works with Jack and has OSX win clients would be really nice. The other possibility would be to have a logon or mode that is set up just for this work flow. Jack would start first and run as the default sound server. Pulse would be started after with sink and source modules started from script rather than jack detect. These module do allow setting channels. Other audio work flows might benefit from this kind of setup too. Len is out of breath. -- Len Ovens www.OvenWerks.net