What I related to was with normal consumer product you can’t go higher then in 16bit/48khz
Last year we tested it with an oscilloscope form these consumer sources,cd, dvd-audio, Blu-ray audio.
Never made the statement that it not technically possible.
So if we make a 18bit source then we know it for sure 
In near future we probably will.
I’m sure it was a mistake but there’s a difference between bitrate and sample rate. The bitrate for DCC is always 384kbps (though MP1 allows other bit rates), and the supported sample rates for DCC (and MP1) are 32000, 44100 and 48000 samples per second, usually expressed as 32. 44.1 and 48 kHz.
The bit depth for DCC was 16 bits (per sample) for the first and second generation recorders. For third generation recorders, the bit depth was 18 bits and some Japanese recorders feature 20 bit encoders and decoders. I think I already mentioned that an increased bit depth basically means that the recorder has more dynamic range. 16 bits represents a dynamic range of about 97dB, 18 bits is about 108dB and 20 bits is about 120dB. See Audio bit depth - Wikipedia.
If you look at the specifications of your amplifier, you will probably notice that it has a noise level that’s usually above -97dB, unless you have a really expensive one. Many high quality amplifiers don’t do any better than -80dB or so. So most users probably won’t hear the extra bits just because their equipment is not good enough, and we’re not even talking about how the frequency range of your ears gets worse as you get older.
The point of using the 96kHz sample frequency has to do with filtering. The Nyquist theorem says that an analog waveform can be represented by digital samples as long as the sample rate is twice as high as the maximum frequency in other original analog waveform. Even a waveform that has frequencies that are really close to half the sample rate can be accurately represented because (according to the math, which I admit I don’t fully understand) for the decoder, there will only be one way to generate an analog waveform. However it’s important that, for this to work, the decoder has to have a filter that completely allows all frequencies up to the Nyquist frequency and completely cuts off everything about that frequency. Without that filter, the output will have distortions in the high frequencies. Such a filter is impossible to make because the sharper the cutoff curve is, the more phase distortion it introduces.
So the idea behind the 96kHz sample rate is to make the Nyquist frequency so high that it doesn’t matter if the filter isn’t good (or doesn’t even exist), because any distortion will have a frequency so high that nobody can hear them anyway. This is also how (and why) oversampling works: by artificially inserting extra samples during decoding, the Nyquist frequency is increased, and the filter can be simpler or can be left out completely.
If a source is converted between 44.1 and 48 kHz sample rate or vice versa, there’s a little bit of quality loss because of quantization errors: The conversion tries to guess what the sample should be, based on surrounding source samples. But for 96kHz to 48kHz, there is no need for guessing (interpolation) because all there are exactly two input samples for each output sample. So if the DCC museum ever wants to release a commercial tape based on a 96kHz master, conversion to 48kHz and then encoding in PASC is the best option.
=== Jac
Thanks for the great explanation, this good for the forum.
If you record digital from a DVD-a
That’s original 24bit/48khz, you have to put the down sampler on on your player. (We tested it on two type of reference players)
This is not because of the 24bit, but because a consumer bit that is high.
(In the time of DCC, there where no consumer players that where 24bit)
And the output results is then 16bit.
If you would remove the consumer bit 24bit should be possible.
I think what Jorn is saying is that if you want to record a DCC from a DVD or Bluray player or something like that, you’ll have to change the setup of the player so it outputs 48kHz PCM, not Dolby Digital and not DTS.
If the player outputs 24 bits audio, the DCC recorder should be able to deal with that (it will just ignore the bits it doesn’t use, in accordance with the SPDIF standard). But DCC recorders only accept 32, 44.1 or 48kHz PCM SPDIF.
===Jac
Is the DCC175 the only machine that has a KHz display? I can’t see it on my DCC730 or DCC300?
Hi,
None of the portables have a Khz mention in the display.
The first gen (900, 82, 92, 2000 and RSDC10) and 4th Gen, (RSDC8 909, ZDV919) do.
Ralf
Thanks Ralf!
Hi DCC Enthousiasts.
I since 2 years reinvented DCC for myself. 
When i was a boy (16 years) i was working at a local record store where they also sold DCC tapes and the DCC 900 set.
I did liked it very much but ofcourse i didn’t had the money at that time. 
Now i just bought myself some portable players and also i found a DCC 951 two years ago new in a box from a seller out of Belgium which never did opened it.
Now i saw this topic about 18 bits recording, and i have a question about that.
On the web you can ofcourse find a lot of 24 bits recordings.
If you play that recording via the output card of your pc via SPDIF with VLC player and connect it to the SPDIF input and record it directly to the 3rd generation DCC 951 player will it be indeed a 18-bit recording?
And if so, how can i verify that because you can’t see it in the display of the recorder?
Hi,
@Jorn and his team is working on a device to make this visible.
Right now, no recorder can actually show the number of bits. The 951 does not even show the frequency. You are recording in 18bit for sure that way.
All the chips in the 3rd generation recorders (DCC730, DCC951) support 18 bit PCM for recording as well as playback, analog as well as digital.
So yes, if your computer (or other device) outputs 18 bits (or more) and you record it with a DCC730 or DCC951, those bits will get encoded into the PASC signal.
A device to detect how many bits in an SPDIF (PCM) signal are used is not too difficult to make. I’m thinking of adding it to my SPDIF receiver (https://hackaday.io/project/24911).
However, if such a device is used on the output of a lossy decompression algorithm such as MP3 or MP1 or PASC, it’s not very useful: An audio CD has 16 bits of significant data and an 18 bit recorder will simply add two 0-bits to each sample. The lossy compression algorithm encodes the signal to a bit stream that describes how much of each frequency band has to be mixed when decoding. When playing a DCC tape, PASC decoder will generate an 18 bit signal based on the information that it has, and it’s possible that the information is inaccurate enough to set the 17th and 18 bits to 1 as needed. So a 16 bit recording may generate an 18 bit playback signal with 2 extra bits of inaccuracy. You shouldn’t be able to hear this; in fact you could say that the DCC recording might sound better than the original CD.
That being said, I haven’t actually verified this; it’s possible that (at least for some recordings, maybe for all) if you record a CD, the output of the PASC decoder will set the last 2 bits to 0 as long as it has enough bandwidth in the bit stream to accurately describe the 16 bit signal.
===Jac
I don’t know about the 20 bit recorders but yes that assumption was false: All the ICs in the audio chain in the Philips 3rd generation recorders are fully 18 bit: Digital, analog, recording and playback.
===Jac
Great update. I was always convinced that digital in only allowed 44.1kHz 16bit and analog would allow switching to 48kHz 18 bit only.
When using the analog input, all DCC recorders record at 44.1 kHz only.
===Jac
I didn’t know that, but that’s a special recorder 
.
So there is at least one exception to the “rule”.
There is no reason for DCC recorder not to have analog inputs at other frequencies (I think the DCC spec mandates that recorders at least have 44.1). Internally, the PASC encoder works at all 3 frequencies; Philips just decided it wouldn’t be important enough for consumers to have more than one analog sample frequency, and perhaps different filters for each?
It wouldn’t take much to put a different ADC on the I2S bus to record from, either. Or an entire different device such as a microcontroller with I2S interface.
=== Jac
I am really following this. I hope that, when a conclusion is reached, the correct information will be available on the website, or on the forum, or on a wiki or something.
Thanks for all the good updates after my first post here.
I am recording now via a 24bit audio file via SPDIF (computer) to the DCC195 SPDIF input.
I have only one issue.
I have to put the receiver almost on 80 procent of the max volume, to have some decent sound (volume) What do i have to change the have the proper volume on normale volume settings.
Change audio input on the player or do i have to change something on my pc’s output?
