Frequently Asked Questions and Answers

Please check here before asking for personal help.

• How can I Listen to the S-Meter Site Receivers and Watch NASA TV using Firefox?
• What is the Receiver Audio Streaming Bandwidth?
• Why is the receiver audio stream delayed several seconds?
• Do I need to install special codecs to listen to the receiver?
• How can I Make Suggestions?

How can I Listen to the S-Meter Site Receivers and Watch NASA TV using Firefox?
Firefox doesn't have native support for Windows Media Players embedded in web pages, but that capability can be added easily. See Adding Windows Media Player Support to Firefox Web Browsers for step-by-step instructions.

What is the Receiver Audio Streaming Bandwidth?
Some websites with receivers stream poor quality audio to reduce their Internet costs. All the receiver audio streams at the S-Meter site are high-quality 20Kbit streams. There is little discernable quality difference between what is heard at the receivers and what you hear via the Internet.

Why is the receiver audio delayed several seconds?
Communicating a signal from one place to another always requires time. The amount of time depends on several factors, including the distance, the propagation speed, the bandwidth, and the encoding/decoding methods used. Engineers call signal transmission delay latency.

Radio signals travel from senders to receivers at nearly the speed of light. Radio channel bandwidths usually are wide enough that bandwidth delays are relatively imperceptible to human listeners (that isn't always true). Most AM and FM radio transmission systems utilize analog encoding/decoding methods that are extremely fast. For all these reasons, radio listeners usually hear audio from live broadcasts only a small fraction of a second after origination (extremely unusual radio propagation conditions can cause longer delays).

In contrast, audio communicated via the Internet is subject to much longer delays. The delays have several causes, the most important being that digital rather than analog encoding and decoding is used. Anyone with a digital cell phone has experienced the problem of talking over another party because of digital encoding and decoding delays. Digital cell phone latency usually is less than a second, but even that short delay is annoying.

Digital encoding and decoding latencies are not fixed. They can be adjusted over a very wide range. They can be made smaller by sacrificing audio quality, sacrificing reliability, and increasing processing speeds and transmission bandwidths (in other words, by increasing costs). Because latency is critical to the practical use of digital cell phones, cell phone design engineers compromise as far as is reasonably possible in exchanging low-latency for quality and reliability. The result is marginal audio quality that is barely adequate to understand human voice, frequent drop-outs, but latency that is low enough to make cell phones usable.

Audio quality and reliability are much more important than latency in other applications, so different design compromises are made. Audio streamed from the Kenwood communications receiver at this website currently experiences the following delays:

Digital encoding on our end: 4 seconds
Internet transmission: 1/4 second (typical)
Digital buffering on your end: 1 second
Digital decoding on your end: 4 seconds

The typical total delay is therefore about 9.25 seconds. All those delays, except the Internet transmission delay, can be adjusted and may be changed in the future.

Being able to analyze the receiver audio output for 4 seconds before encoding makes it possible to send significantly higher audio quality at a given transmission bandwidth. The same is true on the decoding end of the circuit. Allowing your computer 4 seconds to reconstitute a replica of the original signal makes it possible to recover significantly higher-quality audio on your end. The one-second buffer results avoids audio dropouts even where Internet transmission is interrupted up to one second. Without that delay many short transmission gaps would occur over typical circuits.

The end-result is that in exchange for a 9.25 second typical delay you hear audio that in most cases is indistinguishable from the direct audio output of the receiver. If you record audio coming back via the web from your own transmitter, it will be representative of what others with communications receivers actually hear.

Do I need to install special codecs to listen to the receiver?
Receiver audio is streamed to your computer using the Windows Media Audio 9 Voice codec, which is a unique hybrid voice and music codec for AM radio-style audio streams over low-bandwidth networks. It is designed to do something no other audio codec does well-- process audio streams that are primarily voice, but that contain some music. Other voice compressors butcher music. Other music compressors are very poor with voice. S's are lispy and there is echo. The Windows Media Audio 9 Voice codec provides is a small, low-bandwidth stream with AM-quality audio that sounds great with either voice or music.

Prior to Windows Media 9 Audio, Microsoft shipped the same voice codec that RealNetwork markets as RealVoice. However, Microsoft's the Windows Media Audio 9 Voice codec provides a 20-percent compression improvement over RealNetwork's technology with better sound quality. The audio stream from this website can be decoded by the old RealVoice codec, but the sound quality is better with the new codec.

The Windows Media 9 Audio codec was automatically installed on your computer when you installed Windows Media Player 9 or Windows Media Player 10. However, it can be corrupted in some cases by the subsequent installation of other media players. If you think that may have happened, you can download and re-install the latest codecs from the Microsoft Download Center.

How can I Make Suggestions?
Post them at the User Forum. We are interested in any suggestions you have.