Re: Bridging the digital voice and data gap

"Tony Langdon, VK3JED" <vk3jed@...>
 

At 01:43 AM 8/2/2012, you wrote:


Maybe your all confused... audio doesn't have to come out a speaker to be analog. To be perfectly clear, the dv-dongle is still converting a digital signal to an analog one... and a second chip is required to go into any another digital code. Even if a lossless digital codec is used in between (like wave or similar), the conversion to a lossy format like AMBE is the troublesome one... not the conversion to analog. Yes, less signal is lost vs going out a speaker and into a microphone, but the principles I stated before are largely the same, and I'm not convinced the resulting stream of bits will be legible
No, the DV dongle converts a PCM bitstream to an AMBE coded bitstream and vice-versa. There is no analog audio near a DV Dongle (have you ever managed to get an analog signal down a USB bus? ;) ).

Conversions between codecs are better when there is no intervening A/D and D/A pair to go through, so running a DV Dongle (or equivalent) back to back with an IMBE equivalent will give the best result possible. You are right in one thing, the result won't be perfect, may not be pretty, but it may be functional. Only one way to find out! :) If the number of transcodes is limited to the bare minimum (i.e. 1), then you may get along with it.

Oh, and a case in point, telephony systems (e.g. Asterisk) do convert between codecs when necssary, so that you can attach any phone to the system, regardless of what code the network is running. The phone just needs to be capable of using at least one of the codecs Asterisk supports. Asterisk also decodes and recodes any participants in a conference call. Of course, most of the codecs used in telephony are a lot less "aggressive", but the principle works rather well.


73 de VK3JED / VK3IRL
http://vkradio.com

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