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[pault]

Hi, I need to convert dff files to 24 bit, 96kHz WAV. (This is original music production to be sent to a mastering house.)

How does one convert to 24 bit, 96kHz WAV with JRiver? The conversion options for WAV allow me to select only the bit depth, but not the sample rate. I've been searching for an hour and can't find anything.

Thank you!!
Paul

[blgentry]

You seem to have found the Audio Conversion Options dialog.  (It comes up when you press options in the convert window.)

In this dialog, under Audio, you'll see a check box for Apply DSP.  Check that on.  Under that, you'll see DSP settings.  Click that and you'll get the entire DSP Studio window.  Use Output Format to convert from the DSD rate you are using, down to 96kHz.  Select the 352,000 rate to be converted down to 96kHz.  Heck, set ALL the rates to 96kHz, then you'll be sure! 

Brian.

[dtc]

When converting from DSD to PCM you need to apply a low pass filter to get rid of the noise in a DSD file. By default, MC uses a 24 KHz@48dB/octave filter. This is pretty aggressive. Unfortunately, the higher sample rate filters (30K and 50K) are 24dB/octave which is not steep enough to fully eliminate the noise. The Sony standard is 50 KHz@48db/octave if I remember correctly. You can set a filter like that in convert option DSP Studio - Parametric Equalizer (the same place you set the sample rate conversion).  If you do that you need to turn off the default filter in Tools - Options - Audio - Advanced - Configure Input Plug-Ins - DSD.

If none of this means anything to you, you might ask the mastering house or just use the default.

[pault]

Use Output Format to convert from the DSD rate you are using, down to 96kHz.  Select the 352,000 rate to be converted down to 96kHz.  Heck, set ALL the rates to 96kHz, then you'll be sure! 

Brian.

Thanks Brian- I did see this dialogue, but was completely stumped, because this dialogue makes the assumption that the user would know that a 2.8mHz DSD file has an effective input sample rate of 352,800 Hz. Actually, that seems counter-intuitive, since 2.8mHz = 2,800,000 Hz. So why would the user select 352,800 Hz under the input settings??

[pault]

When converting from DSD to PCM you need to apply a low pass filter to get rid of the noise in a DSD file.

This is interesting, are you suggesting that a DSD capture will create low-end noise that did not actually exist in the audio source? Because the audio source I've recorded in DSD (the stereo output of a mixing console) has already been engineered to provide exactly the amount of low end desired for the final recorded product (including usage of HPF's on individual tracks of the mix.)

If the DSD recorder captures exactly what I gave it, then I suspect no additional HPF's should be required, yes?

BTW, the "DSP Studio" options seem like a very bad idea for audiophiles. By engaging these EQ settings, etc, a user would effectively be altering the sound of the music in a manner that was not intended by the mixing or mastering engineers, and downgrading the quality of the audio in the process.

[ferday]

He said low pass, in order to cut out the >48 kHz signals that may be present in the DSD.  I wouldn't worry about it personally but he's saying he feels the stock filters are too aggressive.

The DSD settings can be turned off completely for "classic audiophiles"

[pault]

He said low pass, in order to cut out the >48 kHz signals that may be present in the DSD.  I wouldn't worry about it personally but he's saying he feels the stock filters are too aggressive.

The DSD settings can be turned off completely for "classic audiophiles"

Oh I see thank you- I misread his post the first time:)

[dtc]

This is interesting, are you suggesting that a DSD capture will create low-end noise that did not actually exist in the audio source? Because the audio source I've recorded in DSD (the stereo output of a mixing console) has already been engineered to provide exactly the amount of low end desired for the final recorded product (including usage of HPF's on individual tracks of the mix.)

If the DSD recorder captures exactly what I gave it, then I suspect no additional HPF's should be required, yes?

....

The problem is the high frequency quantization noise inherent in DSD. It has nothing to do with your source. It is part of the format. When you play DSD files, it gets filtered out. But when you convert to PCM, you need to remove it.

Imo, for the default filter the frequency is too low (24 K) and the for the others (30K and 50 K) the slope is not steep enough (24 dB/octave). Weiss uses a filter close to the MC default whereas Sony specified one at 50K. There is no right answer here. Just wanted you to be aware of the issue. But, some filter should be applied so as to not let the quantization noise through.

If your software has a spectral analysis option, try several filters and look at the difference.

Here are some examples from one of my vinyl files. The quantization noise is clearly visible when no filter is applied.

http://yabb.jriver.com/interact/index.php?topic=97740.msg675589#msg675589