solderdude
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Post by solderdude on Dec 22, 2013 18:52:27 GMT
You must have cloth ears ! When you cannot hear any differences between the DVD layer and a FLAC then it is most likely the same master is used for both 'files'. When differences are heard (even between DVD and hybrid red-book layer) the masters may be different or one of them is manipulated to sound better or the other one to sound worse. It is not because of different formats. Rumour has it (can't confirm this) that Sony is notorious for using below par masters for their red-book layer in their hybrid SACD's. Sometimes different masters are used for 'HD' downloads or DSD.
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Deleted
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Post by Deleted on Dec 22, 2013 19:25:09 GMT
i was expecting to notice a difference . i know my sacd player shares the same dac chips as my stagedac. i was expecting to notice some unwanted noise from my pc system. but no difference at all as far as i can tell.
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Javier
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Post by Javier on Dec 23, 2013 8:24:40 GMT
Modern electronics's (sources and amps) SQ is orders of magnitude better than what headphones/speakers can reproduce, so in properly done comparisons with matched levels differences tend to be very difficult, if not impossible, to spot. The same could be said about changing components within any given piece of kit, unless the rplaced parts were incorrectly specified or dammaged chances are the final sound won't be audibly changed even if the difference can be actually measured. Still, many think otherwise.
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solderdude
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Post by solderdude on Dec 23, 2013 13:45:48 GMT
I don't have to look for another USB DAC anymore unless I have to go above 24/192.
It seems there is new firmware for the FiiO X3 which turns it into an asynchronic USB DAC/headphone amplifier that can do up to 24/192. The Wolfson DAC used is quite good ... no galvanic separation though unless I buy one of those isolators but in my setup I don't need to.
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Deleted
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Post by Deleted on Dec 24, 2013 21:21:41 GMT
i cant hear anything, feel anything or smell anything, but i am seeing R E D ! it kind of distracts from everything
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Deleted
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Post by Deleted on Dec 25, 2013 12:38:35 GMT
Frans have you decided on the unit from jlsounds yet? or is that now shelved after your last post?
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Deleted
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Post by Deleted on Jan 21, 2014 23:53:05 GMT
i cant hear anything, feel anything or smell anything, but i am seeing R E D ! it kind of distracts from everything just in case anyone is wondering what this post was all about. i think Ian was experimenting with different colour schemes for the forum. at the time of this post everything was on a horrible red background.
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Rabbit
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Post by Rabbit on Jan 22, 2014 6:06:19 GMT
I think he was imagining it!
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solderdude
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Post by solderdude on Jan 22, 2014 6:08:44 GMT
At this moment I do not have the need for another DAC (the ones I have work just fine) but if I were to buy one at this moment it would be the one from JLsounds jlsounds.com/i2soverusb-with-ak4396.htmlIt has received a lot of positive reviews also by golden-eared wonders. I don't think I will ever need DSD nor bitrates above 192/24 etc. but it is a good thing it can play those formats and more. Maybe I would make another analog filter for it though the one already in it seems to meet the requirements of the DAC chip manufacturer. I also considered this cheapy: hifimediy.com/index.php?route=product/product&product_id=87That one has mixed results but is isolated from USB. Personally I think it will work well but my preference would be for the JL sounds. All that one needs is an enclosure and a decent linear power supply. The cheapy also needs a decent linear power supply which is not included.
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Dave
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Post by Dave on Jan 22, 2014 11:04:49 GMT
At this moment I do not have the need for another DAC (the ones I have work just fine) but if I were to buy one at this moment it would be the one from JLsounds jlsounds.com/i2soverusb-with-ak4396.htmlIt has received a lot of positive reviews also by golden-eared wonders. I don't think I will ever need DSD nor bitrates above 192/24 etc. but it is a good thing it can play those formats and more. Maybe I would make another analog filter for it though the one already in it seems to meet the requirements of the DAC chip manufacturer. I also considered this cheapy: hifimediy.com/index.php?route=product/product&product_id=87That one has mixed results but is isolated from USB. Personally I think it will work well but my preference would be for the JL sounds. All that one needs is an enclosure and a decent linear power supply. The cheapy also needs a decent linear power supply which is not included. And what say you Javier? Not doubting anything Frans says - far too much respect for that - but wondered how it compares technically with Amanero and your 'project'. TIA, Dave.
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Javier
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Post by Javier on Jan 22, 2014 14:40:02 GMT
Unlike Frans, I like technology for the sake of it so we have different views on some things. I agree with him 100% on the relevant stuff but have my own ideas about gear, specifically when it concerns digital. He likes "good enough" from the perceptual POV and I prefer "technically better" from the conceptual POV. IMO he is spot on when he asserts that there is not much to gain from going beyond a, nowadays, certain easy to reach level but I enjoy taking the those extra steps and strive for more, regardless of its perceptibility. For example, if I have a Delta Sigma DAC I want a DSD capable system because to make it makes sense to do the mandatory PCM to DSD conversion in software using a high quality and flexible app like HQPlayer which will allow me to decide how it is done instead of using the resource constrained thus lower performing DAC chip. If I were only listening to PCM like he does I'd probably will get myself a true PCM DAC based on a ladder (R2R)or current segment chip. With that in mind, no, I wouldn't buy the JLSounds part. In a PCM only scenario I would get either the Amanero+isolator board or the JLSounds (or any XMOS based adapter for that matter) and partner it with either a PCM1704 or a TDA1541a DAC with a good I/V stage and PS.
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Post by deireleire on Jan 24, 2014 14:25:42 GMT
What about the Shiit Loki? "Can I use Loki with all of my music, including non-DSD music, AKA PCM? Yes, you can convert all your music to DSD on the fly with JRiver and Foobar if you want to use Loki as your only DAC." got this from the FAQ
I don't understand the full meaning, but I smell something rotten in Denmark...
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solderdude
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Post by solderdude on Jan 24, 2014 20:21:10 GMT
Most modern DAC's (those with Delta Sigma chips) basically ALL convert PCM to DSD on the fly. That is it converts the value of a PCM multibit signal to a 1 bit (or a just a few bits) by up-sampling/digital filtering. Basically the output of a Delta Sigma DAC is a (VERY CLOSE) approximation of the original sample and a lot of interpolated samples right to the next bit value of the PCM data.
Of course you can also do this in software and send a DSD (1 bit signal) to a 1 bit DAC. In this case the output signal is also a close approximation of the original sample and a lot of interpolated samples right to the next bit value of the PCM data.
There are but few 'real' PCM DACs around (affordable ones) and they need a sharp reconstruction filter unless they upsample a LOT. Also in this case the output signal of an upsampling DAC is a close approximation of a 'sample' and there is an analog 'interpolation' between that sample and the next sample (there are no steps in a well designed DAC, only in unfiltered NOS DAC's). In case of an upsampling ladder DAC the output signal is also a close approximation of the original sample and a number of interpolated samples right to the next bit value of the PCM data. There is an analog 'interpolation' between that sample and the next interpolated sample just like in a NOS DAC.
So indeed you can use the Loki for all file formats WHEN converted. BUT you will always need to use those conversion programs in that case. This is NOT the case when you have a DAC that can 'natively' play all formats. If, however, a Delta Sigma DAC chip is used in that DAC the PCM is still converted on the fly to a 'DSD' signal but DSD signals may potentially be 'untouched'.
The 'optimum' DAC would have a (as much as possible) upsampling ladder DAC chip with a well executed fixed analog reconstruction filter PLUS a real DSD DAC chip that simply 'converts' the L-R separated DSD stream + steep enough analog filter.
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Post by deireleire on Jan 27, 2014 12:15:23 GMT
I had to read it trice to understand what you wrote... Still letting it process in the background processes not to overheat my single core... I'll be digging in the wiki's during the lunch breaks to fully technically understand what you're saying.
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solderdude
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Post by solderdude on Jan 27, 2014 23:23:42 GMT
I will try to explain.
There are 2 basic ways to convert an analog signal into a digital one.
1: take samples as often as one can (for left and right) and convert the momentary value of the sample (in time) into a digital word that represents that value. well known speeds are 44,100, 48,000, 88,200, 96,000, 176,400, 192,000, 352,800, 384,000 x per second. The samples can be divided in 'steps' which can be described in a number of bits. Common amounts of bits are 16, 24, 32 (steps of 8 bits). This is called PCM. 16 bits describes a sample in 65,536 steps (90dB difference), 24 bits in 16,777,216 (138dB difference) steps and 32 bits in 4,294,967,296 steps (187dB difference) These bits can be transmitted behind each other which creates a stream of bits. Such a stream consists of 2xbitsxsamplespeed+errorcorrectiondata+tagdata in 'chunks' that can easily be handled. There are many types of 'containers', protocols, types of compression depending on storage type, developer, brand e.t.c. Well known is WAV, FLAC, APE etc. there are tens of types of containers and compression methods all representing the SAME digital 'word' that describes the sample.
2: One can also convert an analog signal to a serial 'stream' at a VERY high frequency so at each moment in time a 'slice' is represented by an 'average' of pulses. This can be done by varying the pulse-width (digital amplifiers) or PWM (Pulse Width Modulation) by using lots of similar width pulses but varying the density of them so more pulses behind each other averages out at a higher voltage level then when fewer pulses are present. This is called PDM (Pulse Density Modulation). So NO samples as in PCM but average analog levels represented by lots of very short pulses. left and right can be 'woven' in each other and pulled apart later. This too provides a stream of 1's and 0's and can be stored and transmitted like PCM can. The well known DSD stream is a PDM 'format' for such a stream
DXD is a format that sort-of mixes DSD and PCM technologies.
DAC conversion:
In the case of PCM converting the samples back to voltage levels takes receiving the bits, pulling left and right apart and converting them to parallel data words at the speed dictated by the sample speed and described with the said number of bits (44100 x 16 x 2 for CD). This can be done in a few electronic ways but as the smallest step is MUCH smaller than the biggest one this presents physical problems due to tolerances occurring in the real world. The most 'real' presentation of this is the ladder or R2R DAC which can put out signal levels corresponding to the described levels. For this reason it is very difficult to create signals that actually represent the samples accurately. This is expressed in ENOB (Effective Number Of Bits) and is actually more applicable for ADC's but DAC's have the same properties. Alas we mortals cannot make something that can 'resolve' more than 22 bits this way and noise floors are lurking in that area as well. The 'steps' though are not seen in the analog signals as they are smoothed out to the next value by the obligatory reconstruction filter. The complexity of this filter is another bottleneck. so NO steps in the output signal just a VERY close representation of the original signal. VERY expensive to make IF the output signal is to be accurate.
DSD is much easier to convert back. All it takes is pulling L and R apart and running those pulses through a low pass filter which gives back the average analog level. In practice the DAC needs to be fed from a VERY clean power signal but is easy to make as well as the analog filter.
It is also possible to convert a PCM signal to a DSD alike signal with certain technologies. These can be software or hardware. Each manufacturer uses their own 'technology' to do the conversion in order to avoid patent claims. They all use fancy words and beautiful descriptions as to why theirs is best. Once converted to this high frequency stream it is quite easy to convert by simply routing it through a simple low pass filter that costs little. Such a DAC is better in linearity than ladder DAC's achieve but have more noise which is the bottleneck here. VERY cheap to make.
Most physical DAC's (so not the chips but the complete device) uses chips that convert PCM to a DSD alike stream so when you think you are buying a DAC that can do '24 or even 32 bits' you buy a simple DAC that can reach about 20 bits in step values and is a (VERY close) approximation but NOT a 1 to 1 representation of the original sample (only NOS DACs can truly represent steps but have LOTS of serious issues) and only true DSD DACs can truly reproduce the DSD stream. Some DACs can convert both 'formats' and do this by converting PCM to a DSD-alike stream and doing real DSD or use ladder DAC's and convert DSD to PCM. Very few DAC's will have 2 different chips that do their job, not very economical and difficult to make electronically.
I like the Schiit approach. the Modi and Loki which do their own thing at low cost. As Modi cannot do DSD and Loki cannot do PCM (what some DAC's can do) it is also possible to use software to convert DSD to PCM and use a PCM DAC or convert PCM to DSD and use a DSD DAC for those formats.
An elaborate way to explain it is possible.
Of course in all cases the limits will be determined by the used topology and electronic components, noise levels, timing accuracy, tolerances etc.
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