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In Reply to: RE: Christie Winn - C'est Magnifique posted by Roseval on September 11, 2011 at 07:17:32
As converted by foobar2000 to 176.4/24 there is music related energy up to about 31 kHz. This can be seen in the screen shot of iZotope RX 2 Advanced. One can clearly see the musical energy in the two plots, one where intensity is encoded by color and the plot is time vs. frequency and the other a more traditional spectrograph at one part of the recording, where I have superimposed this separate window. (Click on the image if you want to see a higher resolution version.)
It looks like the foobar2000 DSDIFF module got the filtering approximately right. I note that the transition of the filter starts somewhat below 88.2 Khz, which prevents residual DSD noise from aliasing while still preserving audio signal. If one expects much more than 40 Khz bandwidth through DSD one is kidding oneself as the damn 1 bit noise creates problems. If left in it cruds things up and if filtered out it takes music with it. (In case you didn't guess, I am not a fan of DSD and DSD to PCM conversions are the worse of both formats as one loses S/N ratio thanks to DSD and one loses transient response and/or bandwidth thanks to the lower sampling rates of PCM.)
Unfortunately there is a problem with the foobar2000 converter. The gain used to convert from DSD to PCM is set a bit too high. While PCM has a hard limit on the maximum undistorted sample, this is not the case with DSD, where the engineer can go a few dB over "maximum" DSD level at the penalty of a small amount of distortion. As a result there is no "correct" setting for converter gain. However, with 24 bit PCM there is more dynamic range than one gets with DSD, so the conversion should be set conservatively. This appears not to be the case with the foobar2000 converter. Of the 12 tracks on "C'est Magnifique" the conversion clipped two of them. One track was clipped for about 10 samples (at 176.4 kHz) on a piano note and this was not audible. Another track had two clips that went on for 18 samples and there was definite vocal harshness which was audibly recognizable and which I had noted while listening to the music before looking at the actual samples with Soundforge. The audible clipping occurs on track 6 at approximately 3:22. From using the "declip" functionality of iZotope RX 2 Advanced it appears that this clipping could have been avoided if foobar2000 had used 0.52 dB of attenuation over and above whatever it has built-in.
I must emphasize that this is a problem with foobar2000, not with the recording. IMO the gain in foobar2000 for converting DSD to PCM should be taken down another 1 or 2 dB. Whenever a sample rate conversion is performed there is the possibility that peak PCM levels may be exceeded. In the case of PCM input this can easily be dealt with by the user by attenuating the input signal, but there is no way to attenuate DSD without converting it to PCM. Consequently to avoid possible distortion DSD to PCM converters should either provide very conservative built-in gain or come with a user-adjustable gain settings. (Weiss Saracon takes the later approach.) In the past some commercial downloads produced from SACDs have had clipping problems and these were eliminated by corrections to the conversion software.
Most audiophiles will probably not notice this clipping, by the way. However, any competent recording engineer will notice it immediately unless they have ruined their hearing by producing over-compressed music by prostituting themselves to the dollar in the loudness wars. The ability to hear common forms of distortion is an essential (learned) skill for a recording engineer as this can happen from misuse of tools or equipment defects.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Follow Ups:
free Audiogate; give 'pleasant' result but I think a little coloured.
A little additional investigation (including browsing into the dreadful HydrogenAudio) let to a post from the author of the Foobar2000 conversion software. Apparently, its output can be selected in 32 bit floating point as well as the 24 bit fixed point that I previously used. This means that the clipping problem can be avoided by the following process:
1. Set the conversion to output in 32 bit floating point.
2. Play the resulting file directly through a player that accepts 32 bit floating point and has a digital volume control or, better,
2a. Take the resulting file and put it through an editor and make any necessary gain adjustments.
Doing this indicated that the track 6 conversion resulted in a signal 0.54 dB louder than "maximum" that fits in 24 bit mode. There was no clipping distortion when playing the file this way. (There is no practical maximum in the Floating Point format, but a system that could play the maximum 32 bit floating point value undistorted would probably be suitable for destroying the entire planet, just what the Vogons would need to construct their inter-spatial bypass.)
In the course of further investigation I found a bug in the SoundForge 10.0 C Statistics function. It correctly finds and displays the (now unclipped) peak at 106.362% of full scale but if the dB display option is chosen it shows the peak at -.535 dB rather than the correct value of +0.535 dB. Idiots who write data conversion software that isn't adequately debugged should be taken out and shot. :-)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
@ Tony:
Try the new DSD decoder plugin for foobar:
http://sourceforge.net/projects/sacddecoder/
that one does allow for gain adjustments ... ;)
(or use AudioGate, which even has an automatic-scan setting for gain checks)
Cheers
Harald
After installing the program (.dll) foobar2000 crashes whenever any kind of file (e.g. flac, wave, mp3) is played. The built in foobar2000 diagnostic program falsely blamed the cue sheet creator, as a known "problem" add-on, but after removing all the add ons and reinstalling them one at a time it was apparent that it was the sacd decoder.
As this isn't something that I need anyhow, I just removed it.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Interesting. However, I don't have any SACD .iso files, just some DSDIFF files that I bought from Blue Coast Records, so I don't think this software will be helpful to me at this time. It's not clear where I would get (legitimate) SACD .iso files. I don't have a playstation or any other device that can read the special pit codes used with SACD DRM.
Thanks anyhow. I will enjoy reading the source code that was provided and learn how the converter was implemented.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
The Saracon allows you to adjust the gain. I first convert the files to aiff, then test the batch with the Foobar Dynamic Meter to get the correct gain setting. I then drop the gain .01 and rerun the Saracon. The sound of the conversion doing this for the Blue Coast Collection was outstanding.
Edits: 09/11/11
Ideally one should NOT adjust anything unnecessary such as gain. There is no clarity on how these things are DONE.
I doubt that you will lose any noticeable resolution by using a constant setting for all albums that is slightly conservative. But having the adjustable gain is a useful feature and provides the best possible results at no cost in speed if implemented cleverly. Note that if you wish to compare two transfers at different levels you will need a calibrated analog attenuator either in the DAC or downstream of the DAC. That way you can ensure that the two versions that you are comparing are rendered at the exact same level, otherwise the comparison will not be valid as the "louder" generally sounds better. (I assume you keep both the original and converted versions. :-) )
What was the speed of Weiss Saracon in doing the conversion? DSD to PCM conversion can be implemented very rapidly because the required filtering does not need any multiplications since one of the products involved is always +1 or -1. This leads to various possible computational shortcuts, depending on the DSP processor or computer CPU used. One would think because of the high sample rates involved that this conversion would be time-consuming, however this does not seem to be the case. (I don't know the performance of Saracon, but foobar2000 is quite fast, using less CPU to downsample than I use when upsampling 44.1/16 to 176.4/32. Of course the numbers will depend on the particular processor and motherboard.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
The speed of the Saracon was about 1.13-1.17 for 176.4/24
Sorry, don't understand what the numbers mean.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
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