screa a rolla doesn't care if cops get killed because they couldn't get hlep due to  digital not working

On Jan 29, 2011, at 12:19 PM, Steve Dresser wrote:

> I'm always amazed that in the rush to embrace digital communication the 
> biggest "dirty little secret" of digital audio is forgotten:  It only works 
> in a near perfect world.  Analog audio can withstand all sorts of signal 
> degradation from selective fading to weak signals to interference, but it 
> doesn't take much for a digital signal to dissolve into total 
> unintelligibility.
> 
> Steve
> 
> ----- Original Message ----- 
> From: "Martin McCormick" <[log in to unmask]>
> To: <[log in to unmask]>
> Sent: Saturday, January 29, 2011 09:22
> Subject: Re: Anyone used d star
> 
> 
>> The audio quality with digital voice communications
>> depends on a ton of other factors. Straight PCM digital is what
>> you have on CD's and movie sound tracks. We all know that that
>> can be as good as it gets. Digital cell phones and two-way
>> radios including DSTAR start out by digitizing your voice in
>> straight PCM and then they get it ready for the limited
>> bandwidth of sending it over the air. There are lots of schemes
>> for compression and all of them loose sound quality, sometimes,
>> lots of sound quality. One scheme I do know a little bit about
>> is called LPC or Linear Predictive Coding. On the transmit end,
>> your voice is analyzed and turned in to numbers representing
>> frequency, amplitude and wave form information. At the receive
>> end, a synthesizer turns those numbers back in to audio at the
>> cost of sound quality due to the inevitable mathematical issues
>> of rounding and timing that are part of anything this complex.
>> 
>> Sure, you could send the straight PCM over the air but
>> every voice channel would be about 100 KHZ wide or more. LPC
>> encoding actually uses about 100 bits per second during silence
>> and may go up to 9600 bits during sound which is much, much
>> better as far as bandwidth.
>> 
>> Also, you probably remember that the minimum band width
>> for the human voice is 300 to 3000 HZ. That is what the
>> telephone industry designs all their circuits to handle. You can
>> use 8,000 samples per second for voice and it is pretty decent.
>> If some system like LPC is built around that frequency range,
>> the mathematical compromises plus occasional damage to the
>> signal due to radio dropouts will make the sound pretty bad.
>> Some of those digital systems fail interestingly in that a voice
>> will suddenly turn in to weirdly musical sound like R2D2 from
>> "Star Wars" because the receiving filter only got part of the
>> information it should have gotten so it sends out the last good
>> sample it got and sort of hangs. It is the audio equivalent of
>> the picture freezes we get on digital television systems.
>> 
>> One other thing to remember. You've also got to keep Mr.
>> Nyquist happy. I'll let you ponder that last statement.
>> 
>> 73
>> 
>> Steve Dresser writes:
>>>> And don't forget, your ts2000 is all digital as well for the audio
>>> output,
>>>> so don't be too quick to discount digital audio.
>>> 
>>> I know that the noise reduction is digital processing in the audio
>>> circuit,
>>> and the high and low cut filters are digital as well, but I'm not sure
>>> about
>>> the rest of the audio circuitry.
>>> 
>>> Steve
>>> 
>>> 
>>