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Is lossy outdated in 2019 & onwards?

Julf

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The digital world is all math. Everything done to the original wave file has an affect on the signal due to quantization or how the math is rounded. Even a simple level change has some effect. Bit depth minimizes this quantization effect. Most daws process at 32 bit depth with some plugins process at 64 bit.
They process at 32 or 64 bit floating point, mainly because that is the most efficient format for modern processors. 32 bit floating point only has 24 bit precision ("resolution").
 

Blumlein 88

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It sounds horrible to me too. Typical of satellite transmission schemes, a single, 4 Mhz channel is multiplexed dynamically into the number of channels broadcast. Each channel can get from 4 to 64 kbits/sec dynamically. Likely the channels that are core programming get the most bitrate, not necessarily what needs it. I had heard that most common rate for music just 32 kbps. SBR is used together with preprocessing to make the content easier to encode. Great example of quantity over quality.
I can't listen to XM. Even voice from talk shows gets under my skin.

An example of quantity eventually being at so low a quality it doesn't matter. I don't see how they stay in business. Yucky sound even for a few minutes. Pocket transistor radio AM from the 1960's blows it away in satisfaction for music listening.
 

Julf

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I am a little confused about the term "dynamic range". When I mix I watch my Dynamic Range Meter pretty close as this shows how much the signal is being compressed or squashed. Depending on the genre I see somewhere between 6db and 12db. Hopefully I am not showing my ignorance:)
As others have explained, there are two different dynamic ranges involved. When we talk about, for example, what 16 bit digital can represent, we are talking about absolute dynamic range - the ratio between the noise floor and the loudest possible signal ("FS" or Full Scale). What dynamic range meters show is calculated/perceived dynamic range (often called "program loudness"), based on ignoring (short-timed) extremes and averaging, as a weighted analog meter would do. There are different algorithms such as AES-17, the most recent standard is EBU R-128 that introduces a standard units LU (Loudness Unit) and LUFS (Loudness Unit Full Scale).
 

Julf

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Thank you for explaining. Is there a tool or site that shows the dB range as you describe in terms of actual min/max amplitude rather than the ratio to peak RMS?
The max is always Full Scale in digital. The ratio tells you the minimum.
 

Julf

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How many stations are analogue only these days? The BBC was using something like 14 bit nicam* in their FM chain a while ago for all their stations, that may have improved.

* The FM sounds better than CD advocates were perplexed when they discovered that.
They get even more perplexed when told that BBC, often considered the "golden standard", has been using digital since the early 1970s. Apart from very small local stations that don't use syndicated content, pretty much all analog FM content is stored and distributed digitally.
 

Soniclife

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I don't know. All my FM tuners are analog so that is what I had in mind. Not digital FM.
I've never heard of digital FM, maybe it's what we call DAB, which sounds awful because of the bandwidth chosen.
I was referring to the chain between the studio and the broadcast mast being in part digital, quite probably a mess of different standards of digital as well.
 

Julf

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I've never heard of digital FM, maybe it's what we call DAB, which sounds awful because of the bandwidth chosen.
Digital FM is an American system that overlays digital on standard FM. Very limited bandwidth, and much more primitive than DAB/DAB+.

I was referring to the chain between the studio and the broadcast mast being in part digital, quite probably a mess of different standards of digital as well.
Yes, used to be mostly NICAM, now migrated to various Audio-over-IP schemes.
 

Willem

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I did not even know there was such a thing as digital fm. Europe luckily escaped.
However I wonder about our cable television systems. I installed a new set top box yesterday (Ziggo Mediabox Next) and it is fortunately much more energy efficient/quieter but I also had the impression that the image was inferior (darker with more contrast) and that the sound of the optical output was tinny. Could they now be using more compression? Or is this my imagination?
 
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Julf

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I did not even know there was such a thing as digital fm. Europe luckily escaped.
Instead we have DAB - a solution to a problem that doesn't really exist any more. :)

However I wonder about our cable television systems. I installed a new set top box yesterday (Ziggo Mediabox Next) and it is fortunately much more energy efficient/quieter but I also had the impression that the image was inferior (darker with more contrast) and that the sound of the optical output was tinny. Could they now be using more compression? Or is this my imagination?
I am sure they are under constant pressure to squeeze in more channels...
 
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I give you an example of their over the line zeal. One of their diehard members challenged me on transparency of MP3 on an audio forum (not HA). I told him that it could not achieve such. He said it would in the case of LAME MP3 encoder. I showed him the graph from HA forum's own Wiki saying otherwise:

View attachment 32040

As you see, it shows that it never achieves 100% quality no matter how high you go in bit rate.

The poster storms away, goes to HA forum asking for help. You can read the saga here: https://hydrogenaud.io/index.php/topic,90464.0.html

Their final answer? Delete the graph from the wiki!

View attachment 32041

The graph is correct and should have been left as is.

There is a fine line in these arguments where we look like crooks instead of providing the science. I want to make darn sure we never cross it as they did there....
There own wiki and even FHG even admitted that MP3/AAC can't cover all content even at 320kbps. Some even showed 400kbps Wavpack hybrid or Q8 Musepack doing much better with harder to encode samples.

Audiophiles have shown different ways that MP3 can break, i tried yesterday cymbal and symphonic rich metal that sounds like crap at 160k mp3 and 128k AAC.

Edit: As per usual HA totally ignores that lo fi mastering, Noisy and chaotic soundscapes on modern rock/metal/some pop can even be enough for average joe to go this sucks. Because a guy showed proof that his black metal albums needed 160kbps Opus because of the noisy sound of Black metal and others needed 350kbps for there 8bit C64 game soundtracks using sine waves and noise gens.
 
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Frank Dernie

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I've never heard of digital FM, maybe it's what we call DAB, which sounds awful because of the bandwidth chosen.
I was referring to the chain between the studio and the broadcast mast being in part digital, quite probably a mess of different standards of digital as well.
AFAIK BBC still uses 32kHz/14 bit NICAM to supply the transmitters.
 

Frank Dernie

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In my amateur live recordings the “noise floor” on the recording is audience and room noise which I have never had below 40dB and usually nearer 60 dB. A soprano/piano will peak at around 100dB, an orchestra maybe 120dB.
16-bit is plenty IME.
If you make a completely electronic recording of in a very silent studio maybe more would be helpful, and 24-bit makes it a doddle to set levels.
Replay is another story. If you had a full orchestra in a silent studio there may well be 120dB between silent bits and peak but that would be completely useless for replay, even on a brilliant home system, never mind a car.
 
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I am sure they had a reason to remove the samples. Any links/pointers to where you posted them?
Since i got locked out my account by my stupidity. I can't remember it was way back, but i'm sure there were linked samples since my OP say's so. It's not the first time i was told i was invalid for easily ABX'ing high bitrate lossy because of my genre choices, The haters nikamu 2007 last track needs 320kbps because i get odd clicking that not in the lossless original.
 

Julf

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The haters nikamu 2007 last track needs 320kbps because i get odd clicking that not in the lossless original.
What encoder are you using? Lame? That sounds like a bug.
 
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What encoder are you using? Lame? That sounds like a bug.
The latest one lame 3.99 to 3.100. Sounds like to me VBR fails to use a 320kbps frame to aid quality, but when i tried prurient history of aids at 320kbps i can hear in the intro of the first track a click sounds wrong while the lossless one is fine.

Musepack at 192kbps doesn't have the issues i get with AAC/Lame/Vorbis. That haters track sounds clean with a bit rate of 180kbps, infact all of my problem samples sound great on 192kbps musepack.
 

Julf

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Musepack at 192kbps doesn't have the issues i get with AAC/Lame/Vorbis. That haters track sounds clean with a bit rate of 180kbps, infact all of my problem samples sound great on 192kbps musepack.
Interesting!
 

pwjazz

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I am playing a mind trick by ripping everything in FLAC, intellectually I know 320K MP3 is so close as to be indistinguishable for regular listening but just knowing a track is lossless fools me into thinking it must be better.
Archiving in lossless makes a lot of sense. I personally do some post processing (volume normalization, EQ, etc) and that requires re-encoding. Even if an initial MP3 rip is transparent, after reencoding it might not be anymore, so it's always good to start with FLAC.
 

Sal1950

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Even 12 years ago when I ripped all my CD and LP collection I could see no reason to use anything else but flac for archiving.
With todays speed of electronics and low cost of storage space, why would anyone chose anything else?
 

Kal Rubinson

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Even 12 years ago when I ripped all my CD and LP collection I could see no reason to use anything else but flac for archiving.
With todays speed of electronics and low cost of storage space, why would anyone chose anything else?
Agreed. FLAC for all PCM rips and downloads.
 
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