Q. How do I verify that I am listening to the FM stations properly?
We are assuming that you are using an MP3Pro compatible client. If not, then refer to the “MP3 Player Recommendation” section of this FAQ.
To verify that you are tuned in correctly, check the sampling rate displayed by your client. If it’s 44Khz, then it means it recognizes the highest sampling rate, so you are all set! However, it if says that the sampling rate is 22 KHz, it mean that the client is not kicking into MP3Pro mode. Check your client’s documentation. Winamp users, refer to this “Winamp MP3Pro plug-in info” page.
If you are still having problems and are on a PC, we sincerely recommend using jetAudio. Some clients, such as “jetAudio,” will automatically tell you what audio format it recognizes. If it says “mp3Pro”, then you are all set. If it says “mp3”, then you will need to refer to the product’s documentation for additional information.
Q. I'm having various problems with the mp3Pro plug-in for Winamp. Suggestions?
The most common problems are .pls links not working and song info not streaming. Both are problems with the plug-in itself and aren't fixable until a new version of the plug-in is released. Actually, the .pls file problem can be corrected if you run the plug-in in Winamp 2.
Q. I understand ‘ZRN supports song title streaming. So why doesn’t my client display them?
This problem is client dependent. Some clients, like WMP, only display the station name, while clients like jetAudio displays everything. We are using an old version of Icecast, so my guess is that some clients can’t handle the old meta data format properly.
Winamp users, refer to the previous question for issues with title stream using the MP3Pro plug-in.
Q. Why does it take so long to buffer? Other Internet radio stations are almost instantaneous!
For those that unfamiliar with “buffering”, what it does is capture several seconds of audio to a chunk of memory before starting. Without buffering, if there is heavy network traffic between you and the source, the data packets from the audio stream will be lost and need to be resent. On your end, you will experience pauses in your audio streams as the client attempts to catch-up. With buffering, the client can absorb some of this “abuse” to prevent those annoying pauses from occurring in your audio stream.
Now, back to your question. We are using an older version of IceCast, which does not have “instant buffering” as I like to call it. Basically, Shoutcast and (I believe) Icecast 2 have the ability to “flood” the client with audio data when they connect to fill the buffer instantaneously.
The downside to this is an increased delay between when a song is started at the source to when it is heard at the client. With Shoutcast, it has been observed to be a minute or longer. With Icecast 1, it’s only about 10 seconds! When listening to automated program, this delay is unimportant. However, it’s extremely important when doing live programming, which we do a lot of.
Q. The audio stream sometimes pauses to rebuffer. What can I do to stop this?
Typically, it means that your ISP or network is congested with traffic. Either reduce the amount of Internet activity at your location or switch ISPs. Of course, those recommendations are of a last resort and there are other things I suggest you try.
First, try increasing the buffer size in your program. In theory, doubling the buffer size should double the time which the stream will need to rebuffer. You could also try another MP3 player. It has been reported that some MP3 players will rebuffer more often then others. (Perhaps some handle audio streams better than others?) Finally, if all else fails, try the station’s AM counterpart.
Q. The audio stream sometimes drops, forcing me to reconnect. What's going on?
This symptom is typically attributed to network traffic. See above question for suggestions.