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P25 voice is way inferior compared to analog.

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prc117f

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On my work radio Analog sounded so much better (EDACS Analog) then switching to P25 the audio is pretty inferior people sound a little robotic etc..

Now why go digital if it offers an inferior experience? Why not just modernize the systems but keep the quality of analog voice? I never remembered all these dead spots etc.. prior to this P25 switch over.

In terms of spectrum savings, I am not so sure P25 really buys much of any spectrum savings when the channels are still spaced 12.5khz anyhow.

I could only imagine how bad Phase II quality will sound like if the stick to voice channels in one 12.5khz slot.
 

jim202

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On my work radio Analog sounded so much better (EDACS Analog) then switching to P25 the audio is pretty inferior people sound a little robotic etc..

Now why go digital if it offers an inferior experience? Why not just modernize the systems but keep the quality of analog voice? I never remembered all these dead spots etc.. prior to this P25 switch over.

In terms of spectrum savings, I am not so sure P25 really buys much of any spectrum savings when the channels are still spaced 12.5khz anyhow.

I could only imagine how bad Phase II quality will sound like if the stick to voice channels in one 12.5khz slot.
You haven't provided any info on the digital radio system your complaining about. There are poorly adjusted digital systems and there are some well adjusted digital systems. If the system is simulcast, then it's even more important for the system to be correctly adjusted.

I have listened to some of the simulcast, P25 systems and they sound good. I have listened to other P25 simulcast systems and they sound terrible.

Then again, you haven't mentioned what make and model radio your using to make your judgement on how the digital radio voice sounds.
 

Thunderknight

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In terms of spectrum savings, I am not so sure P25 really buys much of any spectrum savings when the channels are still spaced 12.5khz anyhow.
.
When two adjacent channel users are both P25, they can be spaced much closer than two analog users.
The occupied bandwidth of the 12.5 channel is less for P25 than analog voice.
 

N4KVE

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Just yesterday I tried an experiment with a friend. A local P25 UHF ham repeater was recently put up. We tried in analog first, & the signal was 80% noise. I was close to the repeater, but my friend who was transmitting was far away. He then switched to P25, & his signal was then 100%. No noise, or robotic sound. The difference was amazing. Both radios had the newest FW.
 

jonwienke

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Having listened to a variety of digital and analog systems, and having done head-to-head comparisons of analog FM vs digital (TDMA DMR), the OP reminds me of one of those people that insists that vinyl LPs sound "better" than CDs. While there may be isolated instances where this may be true, it is not generally the case. The same goes for digital radio vs. analog. Digital doesn't have the hiss and static of analog FM, and when the voice encoder is working properly, you don't sound "robotic" unless you're at the ragged edge of reception where analog FM probably wouldn't break squelch reliably.
 

kayn1n32008

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Just yesterday I tried an experiment with a friend. A local P25 UHF ham repeater was recently put up. We tried in analog first, & the signal was 80% noise. I was close to the repeater, but my friend who was transmitting was far away. He then switched to P25, & his signal was then 100%. No noise, or robotic sound. The difference was amazing. Both radios had the newest FW.


Experienced the same with DMR. LOVE digital, no static, and AGC keeps the volume even between different users(NOT amateur network) late entry RID display.

Active noise cancelling speaker mics and modern DSP do wonders to get rid of background noise as well.

AMBE+ vocoder runs circles around IMBE.
 

ElroyJetson

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You do understand that the whole idea behind digital voice is to increase useful range at the expense of some voice fidelity, don't you?

The objective is not greater range AND better voice quality, it's better range with ADEQUATE voice quality.

The reduction in voice quality is due to a loss of frequency definition. The way IMBE/AMBE works, fundamentally, is that the sounds of your voice are broken down into sounds occurring within certain ranges of frequencies. For example, one of those ranges may be 950 to 1050 Hz. If there is voice activity between these frequencies then the LEVEL of that voice activity is measured, but the codec
doesn't care if that sound is at 955 Hz or 1045 Hz. All it cares about is which RANGE the sound is in,
which is the 1K range, and the level.

So, it throws away the fine frequency information. Only the range and the level matters.

On the decode side, that same signal is produced as a 1000 Hz tone at the same level that the original signal was encoded at.

There are many frequency ranges that are sampled and reproduced this way, and it allows voice audio to be represented with much less data than if it were to be encoded like audio is on a CD, which is linear PCM. (Pulse Code Modulation)

The end result is that there is some loss of frequency information but the codec generates all the frequencies we need for recognizable voice quality. And less bandwidth needed to carry the voice.
 

jonwienke

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Your explanation of lossy audio compression is not quite correct. In no case will the vocoder change the pitch of a tone from 955Hz to 1000Hz. A more accurate description would be that the waveform is simplified by discarding low-level (low volume/energy, not low frequency) frequency components of the input waveform until it can be successfully encoded by the output bit rate. The vocoder analyzes the frequency components of the incoming waveform, encodes as much of the most prominent frequencies as it can within the constraint of the output bit rate, and discards the rest.

What distinguishes various vocoders is the way they break down the input waveform into its component parts, and what (if any) priority is given to one frequency over another to maximize voice intelligibility.

Also, you're ignoring the fact that analog static compromises voice fidelity, too. In most cases, more so than the vocoder lossy compression. Just like cassette tape hiss is generally a bigger killer of audio fidelity than MP3/AAC compression.

Digital gives you constant fidelity, right up to the edge of usable range. Analog may start out slightly better quality at close range, but as distance increases, so does the noise, until the signal is completely unintelligible. One can cherry-pick instances where analog is better than digital (lower signal/noise ratio and total harmonic distortion in the transmitted audio), but overall, digital is better.
 
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