A/D Converter High Res No high pass

Yeah, but taking 384,000 samples in one second will give you a much higher detailed picture of the sound that can not be reproduced digitally.
The detail that you think it is capturing doesn't exist. There is no more detail within the human range of hearing, the mics capability of capturing, or your playback systems ability to reproduce for you to capture...again, that "detail" doesn't exist.
 
Yeah, but taking 384,000 samples in one second will give you a much higher detailed picture of the sound that can not be reproduced digitally.

Except that isn't true for low frequency sounds. Read up on the Nyquist frequency to see why that is the case. Higher sample rates will only allow the accurate reproduction of higher frequencies, and a sample rate of 44.1 is a standard because it can reproduce frequencies up to the effective limits of human hearing (about 22 kHz).

If you're having issues with the low end reproduction of your system, then the cause is either in the setup, the connections, or the room acoustics. Any affordable interface running at a 44.1 hKz sample rate can recreate booming bass. This is evidenced in the tendency for beginners to create overly muddy and/or boomy mixes that nearly make my subwoofer explode. I've made these same mistakes myself with entry-level interfaces running at 44.1 kHz.

If you're not getting enough bass in your recordings, then either your equipment is faulty or there are mistakes being made somewhere in the signal chain.
 
"Thermionic valves", to give them their proper title, "still exist" because people want them but the reasons for that are manifold and NOTHING to do with 'resolution'. Quite the opposite in fact.

A valve stage will always be noisier than a well designed transistor or even op amp stage and, by definition, you stop extracting useful information at the noise floor.

Valve distort (one of the reasons people like them). Even the best valve power amps have THD figure of 0.1% at rated output. A modest transistor amplifier will better that by an order and top class designs get 0.001% at 10kHz, at 1kHz and below the distortion cannot be differentiated from even the best test gear.

Valve PAs have a MUCH worse power bandwidth than solid state especially at your beloved bass end.

It is the tweaky, beardy audiophools that keep the valve factories afloat. I read that many times more valves are sold into the 'hi fi' industry than into their next biggest customer, instrument, mostly guitar amplifiers. THE biggest reason for the continued existence of valve gear is MARKETING BS.

Valve microphones are not intrinsically 'better' just a different set of compromises ALL analogue circuits are heir to.

Valve gear, a mic pre say CAN have massive headroom but when pro tape machines operate at +4dBu and A/D converters hit 0dBfs at +24dBu, even +18dBu, what use an output capabilty of +30dBu?

Dave.
 
Ryan Murphy said:
You guys do realize I am sending a 16 channel analog mixer with 12 ananlog VCO's a bass and one DCo w/sub oscillator through to 1/8th ananlog in mobo mic input right?

I suspected something like that but wasn't sure until you told us.

The 1/8 mic input on your computer is suitable for connection to a computer microphone with a 1/8 phone plug. Your mixer is probably sending a +4 dBu line level signal to an input that wasn't designed for that kind of signal. If your computer has a line input, it's more than likely going to want to see a -10 dBV (consumer audio) line level signal such as you would have with home stereo equipment. Some mixers can do a -10 line out. If yours can't, your best bet is to get any kind of recording interface designed for musical instruments. Even if the mixer can do -10, an interface is more than likely going to give you better quality. I'm not aware of any that have a high pass filter that would neuter your bass response, but an impedance mismatch from making the wrong connection can cause all kinds of strange stuff.
 
"impedance mismatch from making the wrong connection can cause all kinds of strange stuff. "

Aha! Yes! Some designers are extremely parsimonious with the value of output coupling capacitors and feeding such an output into to low an impedance can cause loss of LF. Most 'domestic' gear quotes outputs into 10k or higher.

But! If you are REALLY bothered just get a MOTU M4. Those outputs go down to DC...Good enough for ya?

Dave.
 
The detail that you think it is capturing doesn't exist. There is no more detail within the human range of hearing, the mics capability of capturing, or your playback systems ability to reproduce for you to capture...again, that "detail" doesn't exist.

Then why have I haven't heard any of this stuff before. he acoustic guitar is from 2000's. The amp and bass from the 60's. The synth from 80's. But I have never heard any other recording with that much detail except maybe a vynil. I think you guys are confused on how digital actually works.

Tape sounds good to the 1/8 ananlog mic on computer. This is something I recorded to a 4 track 15 IPS cassette a long time ago. I sampled it at 384khz and it came through really well. the tape did a good job at compressing my signal to be sent through the 1/8th ananlog.

Sure They Can Try.flac - Google Drive
Come Off.flac - Google Drive

and this one at 96khz. Same equipment and port smaller sample rate. It is ever so slightly less sharp.

Oh brick wall why.flac - Google Drive
Jam On.flac - Google Drive
 
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Wait. How are you sampling at 384khz with the 1/8 jack on built in sound?

What program are you using to record with?

What format are you recording to? (wav?)

None of what you are saying adds up.
 
Ryan Murphy said:
I recorded to a 4 track 15 IPS cassette

A standard cassette runs at 1 7/8 IPS. Tascam and Fostex made a number of 4 track machines (eg. 244) that run twice as fast - 3 3/4 IPS. I wasn't aware you could run a cassette at 15 IPS. What make and model is that?
 
I'm starting to think this guy is just trolling.

None of what he says is making any sense. His arguments are nonsensical. Somehow he runs cassettes at 15ips and records at 384khz with motherboard sound.

I call bullshit.
 
I think you guys are confused on how digital actually works

I rather think it's the the other way around. From your posts it seems you hail from the "digital is what I think it is" school, rather than you having actual knowledger of the physics involved. That's why you are copping a lot of criticism here. There are people here who do understand the physics and have been trying to unravel your statements,which, frankly, sound like techno-babble. But maybe that's a bit harsh. Perhaps it's a language thing. Perhaps you do have a degree in electronics. But it's not coming across that way.
 
I rather think it's the the other way around. From your posts it seems you hail from the "digital is what I think it is" school, rather than you having actual knowledger of the physics involved. That's why you are copping a lot of criticism here. There are people here who do understand the physics and have been trying to unravel your statements,which, frankly, sound like techno-babble. But maybe that's a bit harsh. Perhaps it's a language thing. Perhaps you do have a degree in electronics. But it's not coming across that way.
It is techno-babble from where I sit. OP has a "hearing problem" if it's not language, because there are plenty of folks here that really understand this stuff, so when they "talk," the proper response is to shut up, sit up and listen. (I do try!)

I get a vague feeling I've met someone like this on the board before, but there's no shortage, either.
 
A standard cassette runs at 1 7/8 IPS. Tascam and Fostex made a number of 4 track machines (eg. 244) that run twice as fast - 3 3/4 IPS. I wasn't aware you could run a cassette at 15 IPS. What make and model is that?

I think it was 9 IPS. It was a special cassette recorder for studio purposes.

And people people I am not rolling here. i don't know what you guys have against my hard evidence. But micro processors are capable of sampling analog audio 384,000 times in a second. This shouldn't be so confusing. The audio sampled at 384khz is better than the audio from the 4 track studio cassette at 96khz. Why is this so difficult for you people to get. 384>96 OMG OMG.... How dose this not add up. Did you peiople all go to retard school where you threw your brains away and learned how to become retarded becasue that is what I am seeing.

384>96

384khz sampled audio = more detailed audio than 96khz.

384k>96k
???

How is this even a little confusing. It's so simple yet you guys don't understand.

I do recall doing it back in 2014 on my HP G70 using Ubuntu 8 or whatever i probably had..... Get the tape ready. Hook up to mic input. Set the encoder to 384,000khz. Take a couple trys becasue sometimes it fails. I think I had to do it twice. Oh by the way Audacity is really good at sampling analog.

I am listening to these people talk, but w/o any hard evidence to contradict my hard evidence ther argument is null. And as far as I can go the hard evidence posted via that Red Hat video goes hand in hand with exactly what i am saying. For some reason you all are failing to understand simple mathematics of 384k>96k.

I don't really really do not understand what the problem is here,

Don't worry i know you guys have no hard evidence. Do you want to know why. Simple mathematics will solve this issue right now.

96k is not greater than 384k being true. It is impossible to prove something that is completely false.


Contradicting me at this point is just insanity.
 
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Again, answer these questions.

I think it was 9 IPS. It was a special cassette recorder for studio purposes.

And people people I am not rolling here. i don't know what you guys have against my hard evidence. But micro processors are capable of sampling analog audio 384,000 times in a second. This shouldn't be so confusing. The audio sampled at 384khz is better than the audio from the 4 track studio cassette at 96khz. Why is this so difficult for you people to get. 384>96 OMG OMG.... How dose this not add up. Did you peiople all go to retard school where you threw your brains away and learned how to become retarded becasue that is what I am seeing.

384>96

384khz sampled audio = more detailed audio than 96khz.

384k>96k
???

How is this even a little confusing. It's so simple yet you guys don't understand.

I do recall doing it back in 2014 on my HP G70 using Ubuntu 8 or whatever i probably had..... Get the tape ready. Hook up to mic input. Set the encoder to 384,000khz. Take a couple trys becasue sometimes it fails. I think I had to do it twice. Oh by the way Audacity is really good at sampling analog.

I am listening to these people talk, but w/o any hard evidence to contradict my hard evidence ther argument is null. And as far as I can go the hard evidence posted via that Red Hat video goes hand in hand with exactly what i am saying. For some reason you all are failing to understand simple mathematics of 384k>96k.

I don't really really do not understand what the problem is here,

Don't worry i know you guys have no hard evidence. Do you want to know why. Simple mathematics will solve this issue right now.

96k is not greater than 384k being true. It is impossible to prove something that is completely false.


Contradicting me at this point is just insanity.


It says Audacity in the above text. Did you not get the FLAC files I sent? Proof is in the pudding. I understand if your hardware fails, but it'll try it's best and if it dose fail the higher resolution wins except in deficiency in stereo amplifiers and speakers occur. I recommend running a wire from your analog out to a very capable stereo system. The edirol 24bit 192k recordings are from Ardour, and I forget what that uses bmimp or bwav something.... I can't recall. i just export stuff to FLAC or OGG.

I'm having a hard time getting this higher resolution stuff to compress to .ogg or something that is more shareable. It makes some of the stuff like VCO's that are cutting glass in real life sound like complete and utter garbage in recording. If you can understand my metaphor. The glass being an illusion in your brain of 44.1k being all you need.
 
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Of course he's having problems, he somehow has a cassette 4track that runs at 9 ips, a computer with built in sound that records at 348khz that cuts the low end by 30db...

Not only does he have equipment no one else has even heard of, but also has a problem that no one else has.

Not understanding the physics of digital sampling is really the least of his problems. Thinking that sampling a cassette recording at a stupid high sample rate would sound any better than a normal high sample rate is just ludicrous.

Thinking that just because 384 is more than 96 it would make any audible difference is silly too.

The only way it would make a difference is if your converters were broken...which might be why your low end is lacking.
 
First off, Audacity is a DAW and it has nothing to do with "sampling analog". In order to sample analog, you need an A/D converter such as you would find in a computer sound card, or your Edirol or other audio interface. Once the audio reaches Audacity or any other DAW, it's already digital. Personally I don't like using Audacity because when I've tried it, it didn't have the ability to run processing effects in real time and didn't use ASIO drivers for controlling the interface. Others who may not need those functions might like it well enough. I'm okay with that.

Next, if you look at the silkscreening on the top of your Edirol, in the bottom right corner it says "10 in 10 out 24 bit 192 kHz". The maximum sample rate of that unit is 192 kHz. If you want to sample at 384 kHz, you need a converter that can operate at that rate. The Edirol can't do it. If you have a unit that can, name the make and model.

Also, understand that Flac and Ogg Vorbis are compressed audio formats. MP3 and Ogg are lossy compression formats. Depending on how you encode it, I've heard that Flac can give you more of a "lossless" compression, similar to the quality of a raw (uncompressed) wave file, but with a much smaller file size. Analog to digital converters output raw, uncompressed wave data. On a Windows machine that would be a .wav file. A Mac might use .wav or .aiff, but they are essentially very close to the same thing minus maybe some header information for the operating system. A DAW runs on uncompressed wave data. Compressing the data (essentially throwing most of it away) has nothing to do with basic sampling theory. It's a final process to make smaller files for convenience only.

Ryan Murphy said:
I think it was 9 IPS. It was a special cassette recorder for studio purposes.

Again, name the make and model of the tape recorder please. 9 IPS sounds very unusual.
 
Basic Digital Sampling:

The Nyquist/Shannon Sampling Theorem developed by mathemetician Harry Nyquist and later proven and implemented by Claude Shannon at Bell Labs circa the 1940's is a very complicated piece of mathematics. It's not "simple math", but the basics of it aren't too complicated in laymans terms.

Basically in order to sample a frequency and reproduce it accurately in all its detail, you need to sample at twice the frequency of interest. In other words, any frequency that you want to accurately reproduce needs at least 2 samples per cycle. At what is normally considered the upper limit of 20 kHz for human hearing, that means 20,000 Hz or 20,000 cycles per second. So to capture 20k, we need to sample at 40k, minimum to get those 2 samples per cycle. The "Nyquist Frequency" is the half way point, so again if we're doing 40k sampling, the Nyquist limit is 20k.

A reconstruction filter is needed upon playback through the D/A converter to get rid of any frequencies above Nyquist. If there are less than 2 samples per cycle you get aliasing, which is a form of digital garbled nonsense that will ruin everything. We need the samples, but we need to filter out any frequencies above Nyquist to prevent aliasing. This is done with a very steep low pass filter, or reconstruction filter. When Sony and Philips developed the standard for (absolute bare minimum) full range digital audio CD's, they specified 16 bit, 44.1 kHz audio. So at 44.1 kHz, the Nyquist limit becomes 22.05 kHz. The space between 20k (end of the road for human ears) and 22.05 (Nyquist) is used to build the filter.

Since we can capture 20,000 cycles per second with 2 samples (very slightly more with the filter), lower frequencies are no problem because they end up having more than the required 2 samples per cycle. It has no effect on the accuracy of the reproduction. So at 10k we have 4 samples per cycle. 5k, 8 samples. By the time you get to bass frequencies, there's hundreds of samples per cycle if not more. It doesn't mean those frequencies are "better" because there's "more samples". Increasing the sample rate will increase the theoretical capture range of frequencies above human hearing, but we can't hear it. In some cases (certainly at 192 kHz) it makes for huge file sizes and a lot of data to process which normally has negative effects on system performance.
 
I think you guys are confused on how digital actually works.

FTW!!! :D ;)

That made my night...especially since you're probably convinced that you do .

Your concept of how digital sampling works, and your argument points...sound the way people thought about digital 20 years ago, when digital was still new to most folks and misunderstood, and there was all kinds of mythical and illogical stuff being tossed out as fact....but it's not unusual for misconceptions to be formed, as some of it can be confusing, especially when you start comparing to analog or to video...it's just that all that stuff has been explained 1000 times by now.

Go read up on it some more...the answers are out there, pretty much the same ones you are getting here.
 
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