The top performers from AKM, Cirrus, ESS, and TI are all multi-level designs. Achieving comparable performance with a single-bit design would be very difficult if not impossible.
This is true.
The top performers from AKM, Cirrus, ESS, and TI are all multi-level designs. Achieving comparable performance with a single-bit design would be very difficult if not impossible.
I’m surprised no one has produced a vacuum tube multivibrator that does this. Two small tubes and a power supply. Cheap to build, and you could charge a fortune.It is not just about the bitrate: DSD is very inefficient at information storage, despite the compression. So at the same bitrate, PCM is going to hold more information.
Where DSD wins is if you build a true single bit “chipless” DAC. Just a flip flop, an output filter, a buffer. Then you potentially get all the advantages in terms of sound quality — if subjective reports are to be believed.
You think so?I’m surprised no one has produced a vacuum tube multivibrator that does this. Two small tubes and a power supply. Cheap to build, and you could charge a fortune.
Yes, you would not normally store data in that encoding, if you even have access to it. I'm aware of only a few chips that expose these signals externally.
Is it not sometimes 5 level sigma delta some internal format that does not at all look like any consumer or pro format just something suitable for the inner workings of the chip ?
Yes, you would not normally store data in that encoding, if you even have access to it. I'm aware of only a few chips that expose these signals externally.
Many, if not most, DAC chips do indeed process DSD input digitally. I still consider DSD an inferior format to PCM since creating it in the first place is difficult to do without adding distortion. Even three-level sigma-delta is much easier to deal with..... then also DSD input to these DAC chips will eventually be turned into these internal formats for further processing I must assume , but I would not know I don't design these ? Would not a well designed modern DAC chip be input agnostic as everything ends up the same way just before it turns to analog ?
Be careful about interpreting such numbers. Those bits may not represent a normal two's complement encoding. 6 bits could also mean 7 levels represented as the number of ones in each value.Some chips have way more than 5 levels. I think Sabre chips use 6 bits, so they have 64 levels, for instance. AKM has now also 7-bit solutions.
That chip does digital oversampling and sigma-delta modulation. It is intended to be used with the AK4498EQ DAC chip.How many are there? I know only of the AK4191, which is in fact a pure modulator.
Simple rounding error, in all likelihood.If anyone could explain this last part with DR slight increase after normalisation for DSD track's I would love to hear/learn about it.
EBU R128 - 23 LUFS (standard).Simple rounding error, in all likelihood.
To what level were you normalizing?
The noise-shaping is inherent to the process and will be present even if you record straight to DSD. It's got nothing to do with format conversions, though these just add to the problem. This is because oversampling is less efficient in terms of increasing SNR than adding bits: each bit gives you another 6dB of SNR, but you need to increase the sample rate 4 times to achieve the same result. For raw DSD to have the same SNR as redbook CD it would need a sampling rate over 16000 times higher. Noise-shaping is absolutely vital for DSD and the only reason the format works at all.Note that every single DSD release out there (unless it's some kind of single-take affair like direct-to-disc vinyl recordings were once hyped) will have massive noise-shaped ultrasonic noise due to the fact that the bitstream needs to be converted to PCM for editing, mixing and mastering, and then reconverted to DSD.
Be careful about interpreting such numbers. Those bits may not represent a normal two's complement encoding. 6 bits could also mean 7 levels represented as the number of ones in each value.
I really don't follow people talking nonsense around hire. Bit is bit no matter if it's written in word or as a stream. DSD 64 bit stream is equal regarding data amount as PCM 176400 Hz 16 bit. DSD as a format is only relevant for music labels because it provides them a good protection. DSD can be compressed as any other data. However PDM stream is great for transport protocol to DAC or from ADC in DSD or any other form because simplicity and efficiency. DSD is not editable without converting and it's stiff (regarding it's layout) and proprietary format. I think another transport protocol based on PDM is needed which will be more flexible and open source but I don't mean repackaged DSD like DoP but fundamentally different regarding unfolding/processing.
An idea for another unnecessary audio encoding format?
It's called dynamic element matching, and most high-performance DACs use it.True. I have seen designs where the bits are just in parallel instead of in a ladder, and to reduce noise and distortion the allocation is randomised once the circuitry has determined the weight to use.
It's called dynamic element matching, and most high-performance DACs use it.