About The Applicability of Over-Sampling Theory

One fact which I have described in my blog, is that when Audio Engineers set the sampling rate at 44.1kHz, they were taking into account a maximum perceptible frequency of 20kHz, but that if the signal was converted from analog to digital format, or the other way around, directly at that sampling rate, they would obtain strong aliasing as their main feature. And so a concept which once existed was called ‘over-sampling’, in which then, the sample-rate was quadrupled, and by now, could simply be doubled, so that all the analog filters still have to be able to do, is suppress a frequency which is twice as high, as the frequencies which they need to pass.

The interpolation of the added samples, exists digitally as a low-pass filter, the highest-quality variety of which would be a sinc-filter.

All of this fun and wonderful technology has a main weakness. It actually needs to be incorporated into the devices, in order to have any bearing on them. That MP3-player, which you just bought at the dollar-store? It has no sinc-filter. And therefore, whatever a sinc-filter would have done, gets lost on the consumer.

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A Note on FLAC -Compressing 24-bit

One note which I had commented about before my blog began, was that if authors decide to capture sound at 96k samples /second, the resulting sound should compress well using FLAC.

But now that I have experimented with ‘QTractor‘ and an external sound card, I have realized that we will probably also be capturing that sound in 24-bit sample-format, instead of 16-bit. And the sad fact is, that FLAC will not compress the 24-bit format as well, as it did 16-bit.

The reason seems clear. Using ‘Linear Predictive Coding’ means that FLAC will be able to predict the next sample in a set of so-many, to maybe 8 bits of precision, except that the next sample will always deviate from this prediction by a small residual. So 8-bit sound should compress brilliantly.

But then with 16-bit, the accuracy of the encoding stays the same. So again, the ‘LPC’ is really only 8-bits accurate at best, meaning that we get a larger residual. The size of that residual is what makes up most of a FLAC File.

Well at 24-bit, again, the LPC will only predict the next sample, accurately to within 8 bits. And so the residual is likely to be twice as large, as it was with 16-bit, completing 24-bit accuracy this time. We are not left with much compression then.

When I recorded my 14-second sound session the other day, I selected FLAC as my capture file format. I had a noisy air-conditioner running in the background. Additionally, the compression level defaults to Fastest, because the file needs to be written in real-time, and not chewed on.

At 96 kHz, 24-bit stereo, raw audio will take up about 4.6 mbps. At 44.1 kHz, 16-bit stereo, raw audio takes up about 1.4 mbps.

Well I was capturing to a stereo FLAC File, but was only using one channel out of the two. So the FLAC File that resulted, had a bit-rate of 2.3 mbps. This means that FLAC recognized the silent track and used ‘Run-Length Encoding’ on it, but that was about all this CODEC could do for me.

Now, we do have a command-line tool which will-re-compress that file:


$ flac -8 infile.flac -o outfile.flac
$ flac -8 infile.flac --channels=1 -o outfile.flac
$ flac -8 infile.flac --channels=1 --blocksize=8192 -o outfile.flac

The -8 means to use maximum compression.

For me, the bit-rate went down to 2.2 mbps either way.

It beats using a raw format, because using the latter would have meant, nothing would have detected my silent stereo channel, and the file would have been twice as large.

Dirk

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USB Sound Card

One of the recent developments in Computing is, that the actual PCs and laptops have relatively poor sound-chip-sets inside, but that we can add an external sound card via USB. I refer to these as ‘USB Sound Cards’, but think that most people just refer to them as ‘USB Sound Devices’. An actual sound card, used to refer to a PCIe interface card, which we could physically insert into our PC bus, inside the case.

When people buy a USB microphone, because the USB connection is digital, they are in fact buying the Analog / Digital converter inside that microphone, which also makes it the logical equivalent to a sound card. And the fact that it would be a USB mike, does not imply worse quality than an external sound card. To the contrary, users can expect their USB mikes to outperform the internal sound on their devices, which is the whole point in buying them.

I have embarked on yet another project, which is to buy an external sound card that is physically separated from any actual mike or sound source, and to buy a quality mike as well. Hence, I have received my USB sound device already, that has 2 output channels and 2 input channels.

Mine is a “Focusrite Scarlett 2i2” USB Sound Device, even though I usually try not to make endorsements or indictments of commercial products. It is stated to be capable of sampling at 48 and 96 kHz, and stated to be capable of 24-bit precision. It requires a USB 2 connection.

Because sound is taken very seriously with such devices, its only available inputs are a combined XLR / TRS jack each (not a 3.5mm mini-cable). This means that I am still waiting for my XLR-jack microphone to arrive, without which I cannot test the Focusrite. ( :1 )

A plausible question which some readers might ask would be, Why did Dirk not just buy a USB mike? And my answer would be, Because what I pictured wanting was closer to a USB Sound Card, hence an Analog / Digital converter, that can accept a variety of input devices.

But this would also be the context, in which it might make sense to switch my laptop ‘Klystron’ into JACK sound-mode, which supports real-time 48 kHz at 24 bits, and which also supports 96 kHz…

After all, not long ago I was pondering what the settings should be, with which JACK will start, in terms of sample-rate etc..

A key point of this project is again, to test whether the device will work properly under Linux. ( :2 )

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