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'Dynamic range mis-match'?

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Does this exist/matter? For example, I was thinking of newer dacs that can do, say, 129db of dynamic range (redbook is ~100db, but 24/96 files would be >100db, right??) going into an amp that only does e.g. ~100db (normal?).

...also, this might be a naive question but do speakers have a rated dynamic range?

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...also, this might be a naive question but do speakers have a rated dynamic range?

Mine do have a maximum SPL rating of 110dB at 1 metre.

Considering that backgound noise level in my listening room is around 35dB the dynamic range of my system is about 75dB which is more than enough considering that most classical CDs have less than 55dB of DR and most pop/rock is below 30dB...

Cheers,

Ric

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Sometime ago, while discussing SACD’s and DVD-A’s aledged superiority in a webforum a recording engineer posted his description of real world 24 bit recording.

I don’t really feel like translating the whole thing but I’ll try to sumarize it in numbers as best as I can:

The DVD Audio has an S/N of 144 dB (6 x 24 = 144) but current recording and reproduction capabilities are far from that number. And let's not forget that the threshold of human hearing is somewhere around 120 dB (747 during take-off at 10 metres)...

MIC

Neumann’s most silent mic the TLM 103 has an S/N of 131 dB while the most common for “classical” and famous M150, when used in a Decca Tree configuration, lowerers this number to 119 dB

Assuming you are using the TLM 103 you have already lost 13 dB

MIC PREAMP

Next comes the mic preamplifier, let’s say, the excellent Millennia HV 3D with an S/N of 133 dB which is above the mic’s capabilities

A/DC

A good 24 bit AD like the APOGEE 16 X has an S/N of around 120 dB and this means removing 11 dB from the previous weak link, the mic, at 131 dB

You are now recording at 20 bit (120 / 6 = 20)

He goes on to say that after DSP, noise floor and mastering are considered you are down to around 18 bits.

Best,

Ric

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There is presently a thread going on at a certain highly inflammable forum, in search of '24 bit' recordings that have any real information in their lowest 8 bits. So far the yield is ... zero.

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There is presently a thread going on at a certain highly inflammable forum, in search of '24 bit' recordings that have any real information in their lowest 8 bits. So far the yield is ... zero.

Not a bit surprised. The background noise in even the best studios (NC15) means that even with an orchestra playing flat out,(say 110dB SPL), the S/N ratio is 95dB. In practice, when you have a room full of musicians, the noise level is a lot higher than 15dB SPL, so in practice the S/N ratio will be a fair bit lower. For the reasons given above by Ric, there's just too much noise in the system for 24 bit as a distribution format to be anything other than a marketing exercise. The bottom 6-8 bits will be capturing nothing other than traffic noise, aircon noise, mic and mixing desk noise, and musicians' shuffling their feet and turning music score pages over.

S.

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What about entirely or partly synthesised music created on a computer based set up and then converted into 24bit PCM? What's the lowest amplitude sine wave you can get out of a set of speakers with the volume turned right up? When Skalpol posted those dithered files we were getting beyond 16bit resolution then and if you whacked up your amp you could hear that albeit only just.

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What about entirely or partly synthesised music created on a computer based set up and then converted into 24bit PCM? What's the lowest amplitude sine wave you can get out of a set of speakers with the volume turned right up? When Skalpol posted those dithered files we were getting beyond 16bit resolution then and if you whacked up your amp you could hear that albeit only just.

If the music was entirely synthesised, then the minimum level depends on the synthesiser. What you can hear in terms of a very small signal then depends on the dither used. As Skalpol very interestingly demonstrated, it's possible to hear below noise level, but that's without any other signal playing. Masking ensures that in the presence of another signal, how far below that signal you can hear another depends on on masking thresholds, and how far apart they two signal are.

In general terms, if you have one programme playing at normal listening levels, you may be able to hear another programme behind that 20dB down, possibly even 30-40dB down, but after that it becomes damn-near impossible to hear the quieter programme except perhaps in the quiet bits of the main programme, if you see what I mean.

MP2, MP3, AAC etc all work because of these masking thresholds that make anything below that threshold more or less inaudible and consequently doesn't need to be coded.

S.

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Not a bit surprised. The background noise in even the best studios (NC15) means that even with an orchestra playing flat out,(say 110dB SPL), the S/N ratio is 95dB. In practice, when you have a room full of musicians, the noise level is a lot higher than 15dB SPL, so in practice the S/N ratio will be a fair bit lower. For the reasons given above by Ric, there's just too much noise in the system for 24 bit as a distribution format to be anything other than a marketing exercise. The bottom 6-8 bits will be capturing nothing other than traffic noise, aircon noise, mic and mixing desk noise, and musicians' shuffling their feet and turning music score pages over.

S.

how refreeshing to hear propper conversation about the various ways some high companys try to big up there recordings etc, the bottome line for cd is , they end up as beaten up 16 bit maximue capturre, a good many of them have very little dynamic breathing space on them , as far as i can hear , there are some other variations that come on a silver disk but i dont know much about them ,

the master file 24 bit b''LOCKS ETC, is not too much to do with what ou get as a end product in fact i would imagine most master files are runn at 24 bit and much higher capture rates as a matter of course.

i long ago gave up on getting good disks , if i get a good one it is by good luck , i have also sent disks back to companys that are bad i have had a few thnak you return letters but also mostly ignored etc,

i sent a recording back that was a classical fiddle concerto colection , full price , two cds , re issue , said on the pack in very small writting anouluge then digital , got it home it was a record ??? that had been ttransfered , clicks and all , in fact not a very good record set up i should say , the company rwote back giving me bollocks etc, saying no refumd , so i wrote back giving them bollocks and told em to keep the disk as it was no use to listen to ,but i would be happy to take a cassestte copy if they had one ,

no reply .

i men it was a propper company , actually if any one has the various head office addresses , it would be handy , i had trouble getting my letter to a actual person who gave a flying cuss, .

regards

lowendall .

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Not a bit surprised. The background noise in even the best studios (NC15) means that even with an orchestra playing flat out,(say 110dB SPL), the S/N ratio is 95dB. In practice, when you have a room full of musicians, the noise level is a lot higher than 15dB SPL, so in practice the S/N ratio will be a fair bit lower. For the reasons given above by Ric, there's just too much noise in the system for 24 bit as a distribution format to be anything other than a marketing exercise. The bottom 6-8 bits will be capturing nothing other than traffic noise, aircon noise, mic and mixing desk noise, and musicians' shuffling their feet and turning music score pages over.

S.

I'd always presumed that 24 vs 16 bit (for e.g.) resulted in an increased resolution within a dynamic range. I was relating it in my head to the A/D conversion of microscopes at work where increasing to 12 from 8 bit increases the resolution (in terms of intensity values at a given point) within the dynamic range (set by offset and gain, at a given laser power). Is this actually more related to the sampling *frequency* i.e. 48 vs 96khz? The bit depth (?) in audio solely changes the dynamic range?

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I'd always presumed that 24 vs 16 bit (for e.g.) resulted in an increased resolution within a dynamic range. I was relating it in my head to the A/D conversion of microscopes at work where increasing to 12 from 8 bit increases the resolution (in terms of intensity values at a given point) within the dynamic range (set by offset and gain, at a given laser power). Is this actually more related to the sampling *frequency* i.e. 48 vs 96khz? The bit depth (?) in audio solely changes the dynamic range?

Sorry, i'm tired. 12 vs 8 *does* increase the dynamic range of the 'scope data...and the pixel dimensions for a defined area relative to the objective resolution is the sampling frequency. End-of-day retardation.

So...if,in the case of serge's hypothetical orchestra, the sensitivity/offset of recording was set so it didn't record anything below 15db...would 24 bit give more resolution than 16bit (i.e. both set to capture the same real world dynamic range)?

...and what would happen in an extreme situation...eg 95db d range source into 50db amp... Would the bottom end always just be chopped off?

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Sorry, i'm tired. 12 vs 8 *does* increase the dynamic range of the 'scope data...and the pixel dimensions for a defined area relative to the objective resolution is the sampling frequency. End-of-day retardation.

So...if,in the case of serge's hypothetical orchestra, the sensitivity/offset of recording was set so it didn't record anything below 15db...would 24 bit give more resolution than 16bit (i.e. both set to capture the same real world dynamic range)?

...and what would happen in an extreme situation...eg 95db d range source into 50db amp... Would the bottom end always just be chopped off?

Good questions! If you set the noise level of a 24bit recording so that it just failed to capture the background noise in the studio, then the maximum loudness it could capture would be hugely in excess of what either the orchestra could generate or that the microphones could capture. Doing ti another way, if you set the maximum level (0dBFS) to be he loudest the orchasta could go, then the lowest few bits would just be capturing studio noise.

What this means is that the dynamic range of 24 bit recording is way in excess of what is required. Even 16 bit is more than sufficient for wide-range orchestral recording and as for compressed pop, even 16 bit is overkill.

Studio mixers usually mix internally at 32 bit or even more, not for the dynamic range of the finished recording, but to allow for multiple microphone channels all contributing their 3dB of noise and level. My own recordings are generally done using two or three microphones. I mix to stereo on-site and record only stereo. My mixer is analogeu and then I digitise to 16 bit after mixing. I don't have a noise problem nor do I have a noticeable resolution problem, but given that I allow some 10dB headroom, my recordings are probably only 13 bit between peak and noise floor. The ambient noise floor of the recording is always well above the digital noise floor, even leaving 10dBs of headroom, so my conclusion is that 16 bit is more than adequate as a distribution medium, and is even adequate for simple straight-to-stereo recordings.

S.

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Good questions! If you set the noise level of a 24bit recording so that it just failed to capture the background noise in the studio, then the maximum loudness it could capture would be hugely in excess of what either the orchestra could generate or that the microphones could capture. Doing ti another way, if you set the maximum level (0dBFS) to be he loudest the orchasta could go, then the lowest few bits would just be capturing studio noise.

What this means is that the dynamic range of 24 bit recording is way in excess of what is required. Even 16 bit is more than sufficient for wide-range orchestral recording and as for compressed pop, even 16 bit is overkill.

Studio mixers usually mix internally at 32 bit or even more, not for the dynamic range of the finished recording, but to allow for multiple microphone channels all contributing their 3dB of noise and level. My own recordings are generally done using two or three microphones. I mix to stereo on-site and record only stereo. My mixer is analogeu and then I digitise to 16 bit after mixing. I don't have a noise problem nor do I have a noticeable resolution problem, but given that I allow some 10dB headroom, my recordings are probably only 13 bit between peak and noise floor. The ambient noise floor of the recording is always well above the digital noise floor, even leaving 10dBs of headroom, so my conclusion is that 16 bit is more than adequate as a distribution medium, and is even adequate for simple straight-to-stereo recordings.

S.

Thanks. So am i right in my understanding that the 'resolution' is ultimately set by the microphones and 16bit is more than is needed to sample/capture that analogue signal? Is that related to the SNR noise of the mic?

I suppose in my case there must be a limiting SNR for the photomultiplier tubes?

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do you want me to record a 24bit sine wave at 0db and -128 so you can see ?

or even raise it in 1db steps from 0 so you can count.. and probably only hear the 60% of them anyway

It's a pointless exercise though in music.. you are taking something you probably wont be able to hear and have it go so loud you'll be deaf after 2 minutes. Why would you want that in music ?

think about it , stereo differences on vinyl is something like 35db, ie.. when you hear something only coming from the left speaker it is at best only 35db lower on the right speaker - yet you can't hear it.

[edit] Just done it in soundforge.. it only lets me create a 95db file , won't go any further even in 24bit.

anyone brave/stupid enough want to try it out? ha ha

73341732.jpg

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