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Cookie Marenco, recording engineer in San Francisco on poor
state of Audio....DSD files are the way to go for 'reproduced'
music today.
Any thoughts?
Follow Ups:
Started impressive at the beginning of the video, but when I saw her in front of that massive mixing console “doctoring” the audio signals that raised the red flag, DSD or whatever, once the producer starts to mess with the sound we are back to the main problem of audio fidelity with pop recordings these days.
Vahe
with formats, which is often due to equipment not up to snuff and understanding the entire chain.
All things being equal have heard some great DSD and red book. Depends upon the decoder etc. I just wish somehow the improvements would be easier for all of use to enjoy without working so hard at it, and wishing so hard they would just get the recording right.
iBasso DX100,DX50. HiFiman 901 balanced. AK100 modified. fi.Q amp, RSA Intruder, The Lightning, Fostex TH900 balanced, JH13 Pro balanced. etc.
Photo gallery: www.pbase.com/jamato8
If you don't hear the difference between MP3 and DSD you can save a whole lot of money on your stereo system. ;-)
Of the MP3 straw man.
No one has defended MP3's on sound quality for a decade now.
"The problem with quotes from the internet is that many of them just are just made up."
-Abraham Lincoln
"No one has defended MP3's on sound quality for a decade now."
If you search the Asylum you will find posts and posters who go so far as to say that many people prefer the sound of MP3 compared with the original lossless version. There is even a thread presently discussing this.
If a recording has little or no soundstage and if it suffers from electronic grunge the MP3 process may remove more electronic artifacts than musical information, making a presentation that many people prefer. The effect is similar to that of digital noise reduction that is sometimes used when remastering noisy analog recordings, but cruder since the goal is to reduce the number of bits rather than to "improve" the sound. However, if a recording is clean and has a natural soundstage the electronic slight of hand will fail as there will be no grundge to clean up but there will be a soundstage that the psycho-acoustic encoding can not retain. (Of course this assumes one has a decent system that offers clean sound and is using properly set up speakers that can project a soundstage and not a pair of ear buds.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
On PC Audio. :)
Yeah, I was wondering why they were comparing DSD to MP3. I thought that was rather ridiculous. I have no doubt that high-resolution PCM can compare directly with DSD, but not MP3.
Strange!
Best regards,
John Elison
That's what iPods are for!
I think the issue comes down to high resolution PCM like 24bit/96KHz or 24bit/192KHz vs DSD64 DSD128. If the original recording is excellent both PCM and DSD will sound excellent but possibly a little different.
I've compared some DSD to PCM and although it's not always possible to know how the source recordings were created, in general for the small samplings that I've tried, I find that DSD is a little smoother and more polite. This is fine if you want smooth and polite. ;-)
If you want to compare DSD to PCM, the best way is to use a PCM digital recorder and make a recording from the analog outputs of your DSD DAC. That way you will be able to compare the same musical performance. Quite likely, you will not hear a difference. I base this judgement on my 24/96 PCM recordings of vinyl while making A/B comparisons using the tape-monitor switch on my Pass Labs X1 preamp. If there is a difference, it is extremely small and totally insignificant to my ears.Best regards,
John Elison
Edits: 05/01/14
No such thing, and hasn't been in the last 20 years, since the good old days when 44.1KHz 16-bit word PCM sounded really bad. All today available A/D converters are front ended with a delta-sigma modulator that produces a Pulse Density Modulation bit stream, of either a 1-bit two level (DSD), or multi-bit two level PDM.
For PCM conversion, that 1-bit, or multi-bit PDM bit stream(s) is/are first decimate filtered so that no energy exists above the half frequency of the converted to PCM rate, then re-modulated into PCM words.
Other than that tiny detail, you're listening comparisons are dependent completely on the nature of the analog source, and transparency of your listening system. I will say that recording engineers who record acoustic music originating from a natural space (unlike a studio recording) prefer DSD to converted to PCM as more like the analog mix they hear.
. . . why do I always see (via spectrographic software) the ultra high-frequency dusting of noise from DSD sourced recordings, whereas most of the PCM sourced recordings look clean up in the same regions. It may be all PDM at some level, but, obviously something is causing this difference - right?
BTW, your statement "recording engineers who record acoustic music originating from a natural space (unlike a studio recording) prefer DSD to converted to PCM as more like the analog mix" is one I think no one could possibly take issue with. But by the same token, I'd think that most engineers would prefer PCM (if the recording started out that way) to converted-to-DSD. I mean, just on general principle, that's the way I feel.
The issue is whether or not the high frequency noise has been filtered out or not. When converting from DSD to PCM it is possible to filter out all of the high frequency noise and produce a clean looking spectrograph and some conversion software does this. However, this comes at the cost of steep filtering and this usually creates worse artifacts than the high frequency noise assuming one's analog amplifiers are suitably designed.
When DSD is converted to PCM there will be noise starting just above 25 kHz from the original modulator, creating this dilemma. If the source was one bit DSD at double speed this noise will be moved up the f-scale and filtering will be less critical. If multi-bit sigma delta modulation is used then the DSD noise will be reduced (6 dB per bit) and so will require less steep filtering.
Keep in mind that the only problem with the PCM format is its (generally) low sampling rate. It is this low sampling rate that creates the need for filtering on recording. Conversely, the only problem with DSD is the limited number of bits, creating a lot of high frequency noise that will be difficult to filter out, especially since the SACD standard for DSD was set by Sony and Philips at too low a sampling rate. They made the same mistake with SACD that they made with 44.1, namely setting the sampling rate based on technology in place at the time of their decision rather than optimizing the system for technology expected to arrive shortly, according to Moore's Law.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Hi Chris,
Tony hit the nail right on the head. It's the necessary decimation filter when converting PDM to PCM at a lower sampling rate that diminishes the DSD originated high frequency noise. But with this comes the phase shift effects and non linear time delays of any low pass filter, exacerbated by the steepness of the filter.
The point I was trying to make was if all A/D converters begin with PDM, and the type recording (classical acoustic music and jazz) requires only editing and level balancing, with no mixing or effects processing (all of which require PCM conversion), then leaving the recording and listening chain all PDM (DSD) less deteriorates the sound quality available from the microphones.
Thanks! for sharing.
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