How many bits, how fast, just how much resolution is enough?

[audiventory]

In theory, it ended with 44.1/16-bit.

 

It's only part of theory. Ideal case. But in reality we have non-ideal processing and hardware.

 

 

44.1/16, maybe 44.1/24 or 48/24. In extreme cases with exemplary gear and young terrific hearing 48/24 might be needed.

 

Ultrasound playback is not target of high resolution audio.

[audiventory]

16/44.1 wasn't a limitation of the technology back then. It was an understanding of where human hearing hits it's limits and they designed the technology around that.

 

Absolutelly. Base of audio design is 16-20 kHz.

 

16/44.1 was taken as minimal available that time (invention of 16/44).

 

All further increasing of sample rates was based on pragmatical technical reasons (real analog filtration abilities in DAC) on newer technologies.

 

Inceasing of bit depth give expanding of signal/noise ratio in listened dynamic range.

 

DSD's sample rate is matter of signal/noise ratio in audible frequency range in complex with analog filters of DAC.

[audiventory]

Where did I mention sample rate?

 

Sample rate and bit-depth is linked values. If bit-depth is constant, but sample rate increased 2 times, then noise floor decreased 2 times.

[audiventory]

Don't get confused by the stair step of the wave form.

 

Analog filter would be ideal interpolator, if it will have "brick-wall" amplitude and linear phase responses.

 

But there our abilities are very limited.

[audiventory]

Doesn't higher bit depth lower the noise-floor?

 

Absolutelly.

 

And if the noise-floor is lower, don't we get increased resolution?

 

Lower noise is result of higher resolution. I.e. target of any increasing of resolution is decreasing of the noise.

 

For same decreasing of noise floor, resolution may be increased differently in two domains: time and frequency.

 

For decreasing noise floor 2 times, adding of 1 bit is more effective than increasing of sample rate 2 times.

 

Example:

Original bitstream is 16bit x 44100kHz = 705600 bit per second

 

1. For adding 1 bit bitstream: 17bit x 44100kHz = 749700 bit per second (bitstream increased about 6%)

 

2. For increasing of sample rate 2 times bitstream: 16bit x 44100kHz x 2 = 1411200 bit per second (bitstream increased 100%).

 

First case have 6% increasing of resolution, second - 100%. But noise floor decreased 6 dB for both.

 

I.e. resolution and noise floor are separate physical values.

[audiventory]

Mathematically 24 bit is utterly superior to 16... but those extra 8 bits aren't being detected by my ears, so they are wasted.

 

There is physically no way to correctly compare 16 and 24 bit from technical point of view.

[audiventory]

What if we reduce the gain of a recording that maxes at 0dB by 36dB on both Redbook and 24/44.1?

Won't this give us an idea of 16 bit limitations?

 

Perhaps we could do this with Mario's PlayClassics files...

 

Signal noise ratio of 16 bit is worse 24 bit, of course. It is technical fact. Everybody can see it.

 

1. I want to said when we compare 16 and 24 bit we compare not only signal noise ratio but different modes of apparatus.

 

The different mode issue give additional ambiguity that distort picture of signal noise ratio.

 

2. Also there added resolution-conversion distortions.

 

3. But worst case is comparing two separate records produced in different resolution.

 

There is music production/post-production issues may be added to previous items.

[audiventory]

Just about no one in these forums is actually listening to analog that's been reconstructed straight from RedBook.

 

"Straight" is mean "without oversampling"?

 

There is no problem. Simply need invent analog filter with stop band 0 ... -120...200 dB in 20 ... 22 kHz band ;-)

[audiventory]

The authors report on a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz "bottleneck." The tests were conducted for over a year using different systems and a variety of subjects. The systems included expensive professional monitors and one high-end system with electrostatic loudspeakers and expensive components and cables.

 

Compared different systems but not resolutions as resolutions.

 

I don't know how to create identical hardware for 44/16 and 192/24 that need for comparing of resolutions.

[audiventory]

Easy. Using a high-res source, apply the same low-pass filter and dither you'd use for a conversion while leaving the result at 192/24. Feed the original and the filtered data to the same DAC with the same settings.

 

Dither it is variable that depend on implementation.

 

If compare sample rates only and don't use dither: for artificaly synthesed sources your scheme give almost identical result for both sources.

 

Except level of noise. Synthesed source lo-res will have higher level of noise than synthesed hi-res one due different bands.

[audiventory]

Most complex in bit depth issue, what we don't know exactly where threshold of our perception of "qualitative"/"non-qualitative" sound.

 

1. We can measure/calculate dynamic range. It's objective.

 

Objective here is "same for everybody, if use identical way of measurement/calculations".

 

2. Dynamic range impact to signal noise ratio of audio signal. We can calculate/measure it too. It's objective too.

 

3. But we can't say exactly what signal/noise ratio is allowable as qualitaive for us.

 

And subjective aspect of objective parameter are discussed.

 

Esldude before wrote:

 

Some revisionist history here I think. Before the CD became a standard various digital recording formats were around being experimented with for music. They ranged from 13 bits to 16 bits. Sample rates ran from 32 khz to 50 khz. Some used various pre and deemphasis schemes. When Philips and then Sony decided to back a new standard Philips had decided 14 bits was enough while Sony insisted it be 16 bits.

 

It fine illustrate what filst current technical abilities are considered.

 

And after it improvement of subjective perception of objective changes are discussed.

[audiventory]

It is again (very) harmful because of extra high sample rate?

 

High sample rates may be harmful if traditional resampling or full band playback used.

 

But if used optimized mode (cutting ultrasound) with combination of high sample rate, there we have both:

 

1. minimized intermodulation products in audible range and

 

2. preferable mode of low frequency analog filter of DAC.

 

In the video we can hear how the optimizing (removing ultrasound) impact to audible range for some conditions. I don't know exactly where point of intermodulation in the scheme software-driver-hardware.

The experiment may be considered as model of hardware playback too.

 

[audiventory]

Can you show spectrum from this noisier file?

 

Unfortunatelly, I can't find the researched file. I suppose there was DSD's spectrum limited by half of sample rate.

[audiventory]

I would never do that, because it removes hires from the hires - slows down transients.

 

I have invested money and effort to have wide band capable system and don't want to spoil it.

 

But each to their own.

 

I can fill infrasound clearly (as vibration). But things linked with ultrasound are not so obvious for me.

 

You know, I don't consider hi-res for playback ultrasound, but as better way of applying of analog filter only.

 

May be future science works give us new knowledges about system ultrasound-body-ears-brain that we will use.

[audiventory]

Yuri and Jussi: Based upon the last two posts, and without asking either of you to disclose any trade secrets, is it fair to assume that how you treat these ultrasonics is a difference between using HQPlayer and Audiventory for upsampling files to DXD or DSD256+ resolutions?

 

Hi Sdolezalek,

 

For conversion either PCM to DSD or PCM to PCM (DXD is one kind of PCM) used upsampling or downsampling.

 

There used filtering anyway.

 

AuI have 2 upsampling/downsampling mode (by filter mode):

 

1. Traditional (non-optimised)

1.1. PCM to PCM: 0…[Fs/2 - transient band]. Where Fs is minimal sample rate between input and output ones.

1.2. PCM to DSD and back: 0…[20…27 kHz + transient band, depend on input and output sample rates]

Width of the band defined by noise level only, allowable into output band (in output file).

Traditional mode currently have linear phase filters only.

 

 

2. Optimised

0…[20 kHz + transient band]. Rest band is filtered.

I recommend it as «DAC helper».

Optimised mode have linear and minimal phase filters.

 

As default, used optimised mode. I consider it as more «right».

However for users, who prefer wider band, traditional mode available.

 

I suppose, optimised mode is single case, where I recommend perform hearing test before spectrum analisys :-)

[audiventory]

How about compression affecting the mood of the music?

 

For me more "qualitative" (by my subjective perception) music cause lesser tiredness during listening.