Some commonsense

[Rexp]

6 hours ago, mansr said:

I took a 48 kHz music file and resampled it to 96 kHz and back 16 times using SoX at the highest quality setting. The spectrogram of the resulting difference looks like this:

 

As expected, there is slight change in the transition band of the filter. Below this, the level of the difference is about -150 dB RMS, -140 dB peak. In other words, the "loss" is confined to a few LSBs for all content below a certain frequency. Using higher precision arithmetic would reduce this further.

 

Music and difference files attached.

file1.flac 4.76 MB · 0 downloads file2.flac 4.75 MB · 0 downloads diff.flac 923.96 kB · 0 downloads

Will there be information present on a 24/96 master that is not present on the CD, yes or no? 

[Rexp]

3 minutes ago, tmtomh said:

 

Then why bother responding? And why expect any level of civility or respect in others' responses to your comments? Why participate in a discussion forum if you don't give a damn what others say?

From a guy that called others comments moronic, get a grip man! 

[sandyk]

5 minutes ago, fas42 said:

 

Audacity works well enough to be able to pick differences that are audible - I've mentioned a number of times a simple technique that makes pinpointing audible variations quite straightforward, that I've used for years. Roughly align the the two tracks if necessary, select a promising area in the clips of a few seconds or less, and solo one track on repeat - a monotonous pattern of sound is created, like a mantra; as soon as you've tuned into that pattern, switch to the other track, using solo again. It's immediately obvious that the second pattern has very different qualities - if in fact there's an audible variation; going back and forth makes it easy to confirm  that a true difference exists.

 Frank

 The vast majority of Audiophiles are not in the least interested in doing what you and several others  are insisting that they should do, and this is supposed to be an Audiophile forum. other than the huge anti MQA thread, the REAL action in this forum is happening in other areas of the forum than the General Forum area, where many members are finding ways to get improved musical enjoyment from their equipment .

Isn't this what this forum is supposed to be about ?

 

Audiophiles mainly listen to whatever Music genre and format they get the most enjoyment from using their EARS.

 Like it or not, high res LPCM and DSD is here to stay, and there is NOTHING that any member here can say to  change people's  minds about the improvements that they hear, for whatever reason, including the more relaxed HF filtering of high res material.

 

Alex

[sandyk]

18 minutes ago, tmtomh said:

 

Then why bother responding? And why expect any level of civility or respect in others' responses to your comments? Why participate in a discussion forum if you don't give a damn what others say?

 I will repeat this again.

 Like it or not, high res LPCM and DSD is here to stay, and there is NOTHING that any member here can say to  change people's  minds about the improvements that they hear, for whatever reason, including the more relaxed HF filtering possible with high res material.

 

 Perhaps you should be fighting a battle that hasn't already been lost yet, like attempting to kill of the insidious MQA format !

 

[Rt66indierock]

10 hours ago, Paul R said:

Common sense says that the higher sample rate file gives you a choice and thus, even if there is no audible difference to you between 24/192k and 16/44.1k,  preserve the choice for the future. There is no longer any justification in terms of size or bandwidth for 16/44.1k. I do not know of any professional studios recording in 16/44.1k these days, and not many amateurs do either. 

 

Paul, for there to be high resolution albums to purchase a high-resolution format must be specified in the recording contract. I asked Tommy Faragher, Rafa Sardina and Tony Braunagel last Saturday at t.h.e. Show in Long Beach if the labels often specified high-resolution deliverables. They said no.

 

The RIAA May 6^th press release said there are almost hi-res 400,000 tracks available in the United States. 7digital in their June 7^th press release outlining their survival plans said they have more than 70 million tracks. The 400,000 number includes studio production format, 24/44.1. Professionals give the labels what they specify.

 

The reason is simple economics. According to Tony Braunagel, an additional hi-res master costs between $18,000 and $60,000 more to other costs of making a record. That additional $18,000 would have to result in about a million more streams to breakeven.

[tmtomh]

1 hour ago, Rexp said:

From a guy that called others comments moronic, get a grip man! 

 

Actually, you make a good point there. I made that remark out of frustration and should not have used the term moronic.

[fas42]

1 hour ago, Rexp said:

Will there be information present on a 24/96 master that is not present on the CD, yes or no? 

 

There may be some data related to the musical content, but it will normally be so low level, or buried in the sound data that lies below 20kHz, of other instruments, etc - that it doesn't matter. This is where the term "masking" is indeed relevant ...

 

I have not come across a single instance, to date, where worrying about information above 20kHz makes sense - it certainly has absolutely nothing to do with achieving realistic playback ... information "missing" from a 24/96 master is angels dancing on a pin head stuff.

[Paul R]

2 hours ago, Rt66indierock said:

 

Paul, for there to be high resolution albums to purchase a high-resolution format must be specified in the recording contract. I asked Tommy Faragher, Rafa Sardina and Tony Braunagel last Saturday at t.h.e. Show in Long Beach if the labels often specified high-resolution deliverables. They said no.

 

The RIAA May 6^th press release said there are almost hi-res 400,000 tracks available in the United States. 7digital in their June 7^th press release outlining their survival plans said they have more than 70 million tracks. The 400,000 number includes studio production format, 24/44.1. Professionals give the labels what they specify.

 

The reason is simple economics. According to Tony Braunagel, an additional hi-res master costs between $18,000 and $60,000 more to other costs of making a record. That additional $18,000 would have to result in about a million more streams to breakeven.

 

Whether they sell sell the high res tracks or not is rather irrelevant here. They record them in high res, if for no other reason than it does less damage to the sound when it is mixed and mastered.  The delivered file, be that 16/44.1k or higher, is downsampled from the mixing format. 

 

 

[esldude]

2 hours ago, sandyk said:

 Frank

 The vast majority of Audiophiles are not in the least interested in doing what you and several others  are insisting that they should do, and this is supposed to be an Audiophile forum. other than the huge anti MQA thread, the REAL action in this forum is happening in other areas of the forum than the General Forum area, where many members are finding ways to get improved musical enjoyment from their equipment .

Isn't this what this forum is supposed to be about ?

 

Audiophiles mainly listen to whatever Music genre and format they get the most enjoyment from using their EARS.

 Like it or not, high res LPCM and DSD is here to stay, and there is NOTHING that any member here can say to  change people's  minds about the improvements that they hear, for whatever reason, including the more relaxed HF filtering of high res material.

 

Alex

Does Hires material use more relaxed filtering?  It is available to do so.  The bandwidth is there to use it.  But for the most part Hires files don't use more relaxed filtering.  They use the same steep filtering at a higher frequency. 

[sandyk]

3 minutes ago, esldude said:

Does Hires material use more relaxed filtering?  It is available to do so.  The bandwidth is there to use it.  But for the most part Hires files don't use more relaxed filtering.  They use the same steep filtering at a higher frequency.  

That's Nit Picking Dennis. 

The "steep"  filtering is then well above the normal human hearing range where it far less likely to cause audible problems, and it does NOT need to be quite as a severe rolloff either.

[PeterSt]

4 hours ago, esldude said:

But for the most part Hires files don't use more relaxed filtering.

 

At least the DXD from 2NL does not make use of filtering at all (unless they changed their mind by now).

[mansr]

2 hours ago, PeterSt said:

At least the DXD from 2NL does not make use of filtering at all (unless they changed their mind by now).

At the very least, the ADC will use a filter in its internal resampling from the raw sigma-delta output to DXD. If you were referring to filters in the analogue front-end of the ADC, there probably isn't one.

[Jud]

9 hours ago, Rt66indierock said:

Paul, for there to be high resolution albums to purchase a high-resolution format must be specified in the recording contract.

 

A possibly significant exception is analog masters. Half of all music streamed last year was "back catalog," so the companies know there's a market.

[mansr]

10 hours ago, Rexp said:

Will there be information present on a 24/96 master that is not present on the CD, yes or no? 

If the 96 kHz master has spectral content above 22 kHz, it will obviously not be on the CD. Nobody has ever claimed otherwise.

[Jud]

17 hours ago, mansr said:

The term "lossless" has a widely accepted meaning.

 

Yes, there are any arbitrary number of mathematically lossless conversions that can be made between AIFF, ALAC, FLAC, and WAV at a given sample rate, for instance. Conversion between sample rates isn't a mathematically lossless operation, though as you've pointed out it can be done so as to be "perceptually lossless."

[Jud]

16 hours ago, mansr said:

I took a 48 kHz music file and resampled it to 96 kHz and back 16 times using SoX at the highest quality setting. The spectrogram of the resulting difference looks like this:

 

As expected, there is slight change in the transition band of the filter. Below this, the level of the difference is about -150 dB RMS, -140 dB peak. In other words, the "loss" is confined to a few LSBs for all content below a certain frequency. Using higher precision arithmetic would reduce this further.

 

Music and difference files attached.

file1.flac 4.76 MB · 0 downloads file2.flac 4.75 MB · 0 downloads diff.flac 923.96 kB · 0 downloads

 

Would be interesting to see one of the Native DSD files recorded live to DSD256 compared to the same file decimated to Redbook resolution with a filtering chain that might typically be used for mass market product.

[mansr]

39 minutes ago, Jud said:

Yes, there are any arbitrary number of mathematically lossless conversions that can be made between AIFF, ALAC, FLAC, and WAV at a given sample rate, for instance. Conversion between sample rates isn't a mathematically lossless operation, though as you've pointed out it can be done so as to be "perceptually lossless."

No, that is not what I'm saying. Sample rate conversion can be done with arbitrarily high precision. Pick any non-zero number, and the error can be made smaller. The band limiting filter merely needs to be made sufficiently close to the ideal sinc filter. It might require 1024-bit precision and 10 billion filter taps, but it can be done.

 

In practice, the SRC precision need only be high enough to fully utilise the resolution of the target format, typically 24 bits. Taking into account realistic DAC noise levels, the requirements can be relaxed further still.

[Jud]

2 minutes ago, mansr said:

No, that is not what I'm saying. Sample rate conversion can be done with arbitrarily high precision. Pick any non-zero number, and the error can be made smaller. The band limiting filter merely needs to be made sufficiently close to the ideal sinc filter. It might require 1024-bit precision and 10 billion filter taps, but it can be done.

 

In practice, the SRC precision need only be high enough to fully utilise the resolution of the target format, typically 24 bits. Taking into account realistic DAC noise levels, the requirements can be relaxed further still.

 

Except we were talking about a definition of "lossless." I would guess you could make a damn fine mp3 as well.

[mansr]

1 hour ago, Jud said:

Would be interesting to see one of the Native DSD files recorded live to DSD256 compared to the same file decimated to Redbook resolution with a filtering chain that might typically be used for mass market product.

Compared how?

[Jud]

Just now, mansr said:

Compared how?

 

As you did in your prior comment, comparing the initial file to the sample rate converted one. I'd like to see something quite typical of the usual filtering chain used. Since we're talking about common sense, I'd like to get an idea of what's common.

[mansr]

1 minute ago, Jud said:

Except we were talking about a definition of "lossless." I would guess you could make a damn fine mp3 as well.

No, not without limits. MP3 is 320 kbps at most. This alone limits what it can encode. That's basic information theory. The particulars of the encoding method further restrict the achievable accuracy. A pure tone at 500 Hz will probably be represented fairly well, a 5 kHz tone less so. This is because the format inherently quantises high frequencies with lower precision. That is a choice made by the designers of the format based on models of human hearing. Yes, that is perceptual coding.

 

Optimally encoding the full information content in a typical CD quality music file requires roughly 500 kbps, a bit more or less depending on the type of music. An MP3 encoder thus must discard nearly half the information in order to fit the remainder below the 320 kbps limit. Beyond a few basic assumptions, nothing in the format spec dictates which parts of the input get discarded in the encoding process. It would be perfectly valid, for instance, to apply a bandpass filter retaining only the 500-1500 Hz range. You probably wouldn't like the result. A better encoder is more selective, discarding primarily signal components deemed inaudible according a perceptual model.

 

The important point here is, in an MP3 encoding something is always lost, and there are no guarantees whatsoever. PCM, meanwhile, promises to preserve every frequency below Nyquist with the accuracy afforded by the bit depth of each sample value. Need higher frequencies, increase the sample rate. Need better accuracy, increase the bit depth. The only limitations are of a practical nature, such as storage space and computing power. For any given PCM encoding, say 44.1 kHz 16-bit, it is precisely defined what can and cannot be represented.

 

Getting back to the specific term "lossless," it is not normally applied to raw data formats such as PCM. Rather, the lossless/lossy distinction is used when characterising compression algorithms. FLAC is lossless, whereas MP3 is not. Both operate on PCM data. Similarly, PNG and JPEG are both compression methods for raster image data, one lossless and the other not. When FLAC and PNG are termed lossless, there is no implication that the files encode every shred of information hitting the microphone or camera lens, only that upon decoding the uncompressed form of the data will be recreated exactly.

[jabbr]

20 hours ago, mansr said:

I really don't understand your insistence over this. The term "lossless" has a widely accepted meaning. You are attempting to redefine it. Why?

 

True or false:

a) Downconversion from 24/192 to 16/44.1 is frequently lossy?

vs

b) Downconversion from 24/192 to 16/44.1 is always lossless?

 

My position, according to the common definition of “lossy” is that (a) is true and (b) is false. 

[mansr]

1 hour ago, Jud said:

As you did in your prior comment, comparing the initial file to the sample rate converted one. I'd like to see something quite typical of the usual filtering chain used.

How would you carry out this comparison. You can't, for a multitude of reasons, do a sample by sample subtraction between a DSD file and a downsampled version of the same.

 

1 hour ago, Jud said:

Since we're talking about common sense, I'd like to get an idea of what's common.

I'm not sure what's common, but it sure ain't sense.

[mansr]

2 minutes ago, jabbr said:

True or false:

a) Downconversion from 24/192 to 16/44.1 is frequently lossy?

vs

b) Downconversion from 24/192 to 16/44.1 is always lossless?

 

My position, according to the common definition of “lossy” is that (a) is true and (b) is false. 

Why are you even asking that question? Sample rate conversion isn't considered a compression algorithm (unless your name is Bob Stuart).

[PeterSt]

4 hours ago, mansr said:

6 hours ago, PeterSt said:

At least the DXD from 2NL does not make use of filtering at all (unless they changed their mind by now).

At the very least, the ADC will use a filter in its internal resampling from the raw sigma-delta output to DXD. If you were referring to filters in the analogue front-end of the ADC, there probably isn't one.

 

By now I am quite sure that I confused myself with their proposed playback means. Thus the DAC. Back at the time they used the first 384 capable DAC (there was an other brand and I suppose I was the third with one) and they were proud of that sampling rate not needing a digital filter (nor analogue). It was a DAC which was not officially on the market (yet). Btw, this must have been 13 years or so ago and I was talking about it with Morten Lindberg over the phone, because I thought it would be a good idea to have their 352.8 supported in a software player.

Those were the days.