Le 11/05/2011 10:36, Invisible a écrit :
> 
> I would suggest, however, that 4 bits/sample would be inadequate for
> audio applications.

For a linear PCM, yes.
But one of the latest format (SACD) use only 1 bit/sample (at a silly
high sampling rate) in a non-PCM storage.
(in fact the bit is simply a up/down delta, and it make easy to get ride
of the digital to analogue converter in the mass production of player:
use the bit to drive digitally a transistor, accumulate output in a
capacitor, the connection to the ear-set is done directly on the
capacitor (volume is set by resistor on the transistor's upstream))
2 cheap components (or even less if you can fake the capacitor with
circuitry) to replace an expensive DAC.

Delta Sigma encoding is really painful to convert back accurately in PCM.
And it is even worse with Adaptive Delta Sigma (where the unit of the
current sample is dependent of the previous samples), but they requires
far less bits than PCM.
(Adaptive Delta Sigma need more than 1 bit)

-- 
Software is like dirt - it costs time and money to change it and move it
around.

Just because you can't see it, it doesn't weigh anything,
and you can't drill a hole in it and stick a rivet into it doesn't mean
it's free.

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On 11/05/2011 13:21, Le_Forgeron wrote:
> Le 11/05/2011 10:36, Invisible a écrit :
>>
>> I would suggest, however, that 4 bits/sample would be inadequate for
>> audio applications.
>
> For a linear PCM, yes.

Linear PCM is the standard way to describe digital sound signals.

I should think that even logarithmic PCM would sound poor at 4 bits/sample.

> But one of the latest format (SACD) use only 1 bit/sample (at a silly
> high sampling rate) in a non-PCM storage.

You mean pulse density modulation.

> (in fact the bit is simply a up/down delta, and it make easy to get ride
> of the digital to analogue converter in the mass production of player:
> use the bit to drive digitally a transistor, accumulate output in a
> capacitor, the connection to the ear-set is done directly on the
> capacitor (volume is set by resistor on the transistor's upstream))
> 2 cheap components (or even less if you can fake the capacitor with
> circuitry) to replace an expensive DAC.

I'm not sure whether this is true.

I was also unable to discover what encoding SACD uses for its data. I 
rather doubt that raw PDM written directly to the disc surface, any more 
than a normal CD contains raw PCM.

> Delta Sigma encoding is really painful to convert back accurately in PCM.

Actually it's trivial. You just need a digital low-pass filter.

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Le 11/05/2011 15:10, Invisible a écrit :
> On 11/05/2011 13:21, Le_Forgeron wrote:
>> Le 11/05/2011 10:36, Invisible a écrit :
>>>
>>> I would suggest, however, that 4 bits/sample would be inadequate for
>>> audio applications.
>>
>> For a linear PCM, yes.
> 
> Linear PCM is the standard way to describe digital sound signals.
> 
> I should think that even logarithmic PCM would sound poor at 4 bits/sample.
> 
>> But one of the latest format (SACD) use only 1 bit/sample (at a silly
>> high sampling rate) in a non-PCM storage.
> 
> You mean pulse density modulation.

You can call it like you want. It's a Delta Sigma encoding on 1 bit at
2.8+ MHz: to replay it, you just have to integrate (sigma).

> 
>> (in fact the bit is simply a up/down delta, and it make easy to get ride
>> of the digital to analogue converter in the mass production of player:
>> use the bit to drive digitally a transistor, accumulate output in a
>> capacitor, the connection to the ear-set is done directly on the
>> capacitor (volume is set by resistor on the transistor's upstream))
>> 2 cheap components (or even less if you can fake the capacitor with
>> circuitry) to replace an expensive DAC.
> 
> I'm not sure whether this is true.

It is. At least, it is possible.

> 
> I was also unable to discover what encoding SACD uses for its data. I
> rather doubt that raw PDM written directly to the disc surface, any more
> than a normal CD contains raw PCM.

Raw, no. You need a container for specifying the data rate, the mixing
of the channels. And at lower layer, a bit of self-correcting error
detection encoding (just like the 512/4096 sectors of harddrive are in
fact a bit more bits on the medium, usually with a 8/14 encoding and
some CRC.

A bit like movie containers: the streams must be identified, and
multiplexed. Once separated, each stream can be pushed to its own decoder.

And IIRC, SACD likes to play with DRM (which means a bit of encryption:
the key for the disc is on the disc, but only readable by licensed SACD
device.)

> 
>> Delta Sigma encoding is really painful to convert back accurately in PCM.
> 
> Actually it's trivial. You just need a digital low-pass filter.

The issue is computing the right parameters of the filter and
*accurately*. (If many PCM to/from SACD encoding+decoding was performed
in sequences, some difference might appears between the various PCM
stages... ). Fine tuning that part was really out of my capability.

-- 
Software is like dirt - it costs time and money to change it and move it
around.<br/><br/>


Just because you can't see it, it doesn't weigh anything,
and you can't drill a hole in it and stick a rivet into it doesn't mean
it's free.

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> On 10/05/2011 20:15, Alain wrote:
>
>> Maybe that the original recording was done in a lossy format, or even a
>> non-lossy format but with a sample rate set to low and a sample
>> resolution also to low... Like 4000 kHz (or even less), 4 bits...
>> (I had a single CD that contained the whole Beatles discography encoded
>> as .wav at that level or about...)
>
> A normal CD is 40 kHz, so 4000 kHz would be 10x *higher* resolution than
> normal. And 4 bits per sample would be almost unrecognisable.

I missed the decimal point.
When I got that CD, I was realy surprised that they managed to cram so 
many tracks on it. I looked attentively at the files and their 
formating. I was rather incredulous at the 4 bits, but after checking 
with another programm, the result was the same.
And the sound was very surprisingly good.

>
>>> 2. If I can tell that it's compressed, despite not having the
>>> uncompressed original to compare to, doesn't that mean that there's more
>>> redundancy in the signal than the codec is taking advantage of?
>>
>> It's just that you have reasons to expect a higher chromatic range than
>> the one you have.
>
> Chromatic range? I think perhaps you meant dynamic range.

Dynamic range is about the range from low volume to high, chromatic 
range is about the low to high frequency, in particular, the fidelity of 
the harmonics.
At least, according to the texts that I have read over over 40 years...

>
>> Even the best codec set at the highest quality can't do miracle if the
>> source is bad...
>
> In this case, that's unlikely to be the problem.

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>> Chromatic range? I think perhaps you meant dynamic range.
>
> Dynamic range is about the range from low volume to high, chromatic
> range is about the low to high frequency, in particular, the fidelity of
> the harmonics.
> At least, according to the texts that I have read over over 40 years...

"Chromatic" usually refers to musical notes, not sonic frequencies. The 
term more usually applied to the range of frequencies that something 
supports is "bandwidth".

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