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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: DiggerinVA on February 28, 2015, 08:26:57 PM
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I am in the process of transferring some shows. I will be doing it DAp1 to my PMD661. My question is should I convert to 44.1?
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IMO no - why bother today - no one uses CD's anymore - besides I am a purest - the less processing the better IMO
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Thanks that was what I was thinking. Though I will test it with the Airport.
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I would keep/archive a copy at the original sample rate. Downsampling to 44.1 is an option after that.
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Heck I keep the WAV still. Mostly for posting. if I get a PSA then I will fix it for them.
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I downsample to 44.1. I can't hear any loss, and the people who download and want to burn CDs will have no idea how to convert sample rate.
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I downsample to 44.1. I can't hear any loss, and the people who download and want to burn CDs will have no idea how to convert sample rate.
Try 44.1 v. 48 on a USB DAC and you'll hear a difference. USB clock rates are incompatible with 44.1K and unless the DAC buffers and rebuilds the clock, there will be audible interference.
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I have, can't hear a difference.
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I downsample to 44.1. I can't hear any loss, and the people who download and want to burn CDs will have no idea how to convert sample rate.
Try 44.1 v. 48 on a USB DAC and you'll hear a difference. USB clock rates are incompatible with 44.1K and unless the DAC buffers and rebuilds the clock, there will be audible interference.
Seems odd to me that 44.1 would be incompatible with a USB DAC; they would be worthless to me considering how many recording's are at 44.1. I could understand 48 Being not compatible.
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I can't find the original article, but I found it quoted in a thread on DIYAudio (http://www.diyaudio.com/forums/pc-based/150343-computer-based-hifi-15.html#post1981493)
Quoting from John Swenson:
Synchronous: in this mode the readout clock is directly derrived from the 1KHz frame rate. There is a PLL that takes in the start of frame signal and genrates a clock. Using this scheme its rather difficult to generate 44.1, but very easy to generate 48KHz. This is a primary reason why many early USB audio devices only supports 48KHz, they used this mode. As you can guess this mode is very susceptible to jitter on the bus, pretty much anything that causes the output from the host to be jittered (PS noise, vibrations, interference etc) AND things that can cause jitter on the interconnect (interference, reflections, ground noise etc) will wind up with jitter on the readout clock. This is a VERY poor mode to use for decent quality audio.
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The key here is a USB DAC and maybe a early one at that. So I will not bother. Thank You.
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I can't find the original article, but I found it quoted in a thread on DIYAudio (http://www.diyaudio.com/forums/pc-based/150343-computer-based-hifi-15.html#post1981493 (http://www.diyaudio.com/forums/pc-based/150343-computer-based-hifi-15.html#post1981493))
Quoting from John Swenson:
Synchronous: in this mode the readout clock is directly derrived from the 1KHz frame rate. There is a PLL that takes in the start of frame signal and genrates a clock. Using this scheme its rather difficult to generate 44.1, but very easy to generate 48KHz. This is a primary reason why many early USB audio devices only supports 48KHz, they used this mode. As you can guess this mode is very susceptible to jitter on the bus, pretty much anything that causes the output from the host to be jittered (PS noise, vibrations, interference etc) AND things that can cause jitter on the interconnect (interference, reflections, ground noise etc) will wind up with jitter on the readout clock. This is a VERY poor mode to use for decent quality audio.
I would not suspect this is as much of an issue for Modern DAC's using more modern chips such as a the ESS Sabre32 chips which most modern higher USB enabled DAC's leverage today.
They typically are Asynchronous and do NOT pull any DA clocking from the USB bus - coupled with larger buffers all help to lessen the effect of jitter.
Also most of the content people listen through DAC's is going to be CD rips which are all 44.1khz - So I completely do not buy that jitter is going to be noticable unless the DAC is complete junk.
I could see this as an issue many years ago when computers had limited CPU and memory and USB was slower - not today.
I still do not see ANY advantage to downsampling though
I personally only post raw quality - if I record at 24/96 then that's all I will seed...
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believe it or not lots of folks still burn to cd. I would much rather seed 16/44.1 that I did myself correctly than let the average end user try and figure out how to re-sample and dither. for bands on the archive I upload both 16/44.1 and 24/48
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believe it or not lots of folks still burn to cd. I would much rather seed 16/44.1 that I did myself correctly than let the average end user try and figure out how to re-sample and dither. for bands on the archive I upload both 16/44.1 and 24/48
On Archive, I do have the site convert too MP3 - so you have the option to listen to MP3 as well - No in between though - IMO if you don't want hi-res then MP3 is good enough for you
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I downsample to 44.1. I can't hear any loss, and the people who download and want to burn CDs will have no idea how to convert sample rate.
Not all re-sampling methods are created equal.
In the late 90s / early 00s, I used to use a Zefiro ZA2 sound card for DAT transfers which was a great card (I thought) because it provided on-the-fly hardware resampling. This was a huge advantage vs. having to perform software resampling on computers of the late 90s. A single show could take overnight to resample in Cool Edit Pro with the quality set to 999.
I listened to those transfers a few years later and was appalled by how bad they sounded vs. an unsampled transfer. I ended up having to toss all of those transfer and re-do dozens of DATS. Modern software resampling sounds much, much better and is much faster than the days of old, so I would agree that when done properly, no one should be able to hear the difference.
In general, resampling is not a processing step I recommend unless absolutely necessary. For a show recorded and posted in 24 bit, it would be foolish to resample for any reason other than maybe because 96k or 192k files are enormous, but that begs the question why record in those resolutions in the first place. If you want to put out 16/44 red book compatible releases, totally fine and worthwhile, but I would never resample an archival master as a standard practice.