Foobar 2000 for Dummies (Part 2)

SACD plugin configuration (version 0.6.2 or later) ASIO Driver Mode: Options are PCM or DSD. DO NOT use DSD if you don’t have a DSD capable DAC, you might break something!. PCM Volume: Compensates for the lower volume of most SACDs, Can be configured to any desired value from +0 to +6dB. Choose by personal preference unless you get distortion, recommended value is +0dB because it leaves more headroom but it may make DSD tracks sound quieter and can be annoying in a mixed PCM/DSD playlist. If only SACD ISOs or SACD-R is used, there is a replay gain database file that can be used to normalize volume levels. PCM Sample Rate: Options are 44100, 82000, 176400 and 352800Hz. Recommended value is at least 88200 but the higher the less processing involved (ie better) so try going as far as your soundcard/DAC allow. Solutions capable of 352.8KHz are starting to become available at affordable prices both as ready built commercial products or as DIY kits. DSD2PCM Mode: Options are: – Multistage (Fixed-Point): supposedly the best one SQ wise but it is the heavyest CPU hitter, needs a fairly recent PC or may produce “scratching” noises by hitting too high CPU usage. – Multistage (Floating-Point): This one is SSE coded so it is much easier on the CPU a the cost of slightly inferior SQ (if you can hear the difference that is) – Direct (Floating-Point, 30KHz LF): The name says it so no extra description needed this one is also SSE optimized. Being 30KHz low pass filtered means all DSD noise above that frequency is removed and does not reach the amp. A very good balance between SQ and CPU usage. – Multistage (Double-Precission) – Direct (Double-Precission, 30KHz LF) As the previous ones but using 64bit precission floating point instead of 32bit. Since version 0.6.0 two new modes have become available, these allow for custom filters in the DSD->PCM conversion. Some sample FIR filters with different Freq cut points are provided along with the plugin in the “Filters” folder inside the zip. Those with the required knowledge can write their own filters.

If any of the options inside the red rectangle is chosen the “load” becomes active and a filter can be loaded, the sample ones look like this: Preferable Area: Options are: – None: it will show all available tracks on the SACD – Stereo: only stereo tracks are displayed – Multichannel: limits shown tracks to multichannel ones Editable tags: Yes/No Edited Master Playback: Yes/No Store Tags With ISO: Yes/No (it actually doesn’t write to the ISO, it stores tag info in the plugin folder)

PLAYING NATIVE DSD IN FOOBAR: For those lucky enough to have a DAC capable of native DSD decoding these are the steps required to send the uncoverted stream to your DAC: If you have ASIO compatible drivers 1) run the ASIOProxyInstall.exe included in the plugin zip 2) Once the above is installed, a new ASIO device should appear on the list:

3) Double click the foo_dsd_asio to open the configuration dialogue window. It may open minimized so look at the bottom of the screen if it doesn’t come up on top, it should look like this:

4) In the “ASIO driver” drop down list look for and select your DSD capable DAC. 5) From the options available in the “DSD Playback Method” choose the one that the manufacturer of your DAC recommends. What if your DAC doesn’t have ASIO drivers?: You will need to take an additional step by installing ASIO4All (ASIO wrapper for KS) and telling foo_dsd_asio to use it as the output device. Needless to say that ASIO4All should be set up to use your DSD capable DAC as output device. If all is set up correctly, when playing DSD in Foobar ASIO4ALL should report 176400 Hz incomming sample rate (DSD64) or 352800Hz (DSD128). Real-time conversion from PCM to DSD (version 0.6.1 and later) This is a new feature introduced in version 0.6.1 of the SACD plugin. It allows the user to convert PCM from WAV. FLAC, etc. to be converted to DSD as the music is played (the played file is NOT altered in any way) and sent to a DSD compatible DAC. All modern Sigma Delta DACs convert PCM to a format very similar to DSD before converting to analogue, this plugin lets the user do this process in software offering a choice of filters with a higher sampling rate (DSD128) instead of relying on the fixed algorithm present in the DAC. This feature is enabled by changing from “none” to any of the 4 options provided under “PCM to DSD method” in the foo_dsd_asio configuration window: NEWVersion 0.6.4 has introduced the possibility of choosing either integer  or floating point (FP32) based coversion, the conversion algorithm should produce very similar results but with FP modes being easier on the CPU.

Differences between 4 modes (ignore the compression column): It is possible to send DSD in the following formats (since V. 0.6.2):

- 2.822 MHz (DSD64@44.1KHz)

-  3.072 MHz (DSD64@48KHz)

-  5.644 MHz (DSD128@44.1KHz)

- 6.144 MHz (DSD128@48KHz)

- 11.289 MHz (DSD256@44.1KHz)

- 12.288MHz (DSD256@48KHz)

- 22.579 MHz (DSD512@44.1KHz)

- 24.576 MHz (DSD512@48KHz)

NOTE:  ASIO Proxy (aka foo_dsd_asio) included in the SACD plugin zip file upsamples both 44.1Khz based PCM (44.1K, 88.2K, 176.4K & 352.8K) and 48KHz based PCM family sampling rates (48K, 96, 192K & 384K)  though not all DACs support 48K based DSD.

NOTE 2: Version 0.6.5 of the plugin has introudced a new option to help reduce clicks and pops when changing from PCM to DSD and vice versa, it consists of a drop down list with delay values ranging from 10ms to 5 seconds (5,000ms), see the following screen capture:

NOTE 3:

The latest 0.6.6 version of the SACD plugin lists the following bug fixes and additions:

- Random channel rearrangement in DSD mode (for stereo<->multi transition) fixed.
– Installable filter description added.

Screen captures have been updated to reflect these changes.

Our warmest and most heartfelt thanks to Mr. M. Anisiutkin incredibly generous and talented creator of Foobar’s DVD-A and SACD plugins among many other things.

Comments
  1. Syd says:

    It’s great to see this info receiving attention again! Any chance, at some point, of a tutorial on the layout capabilities? I have messed mine up several times and it took me ages to recall how I had got it as it looks now. I think many who have never tried this aspect of Foobar would appreciate it.

    • Syd, I use a pretty basic interface myself, cero bells & whistles just sheer speed but can try to add some more part to the article covering non audio areas like setting it for remote control, interface improvement, etc.

      • Syd says:

        My setup is reasonably basic, but at first I didn’t even realise Foobar had so many possibilities. Javier guided me through the basics one to one regarding appearance and functionality. I have all of my music folder down the left side, album cover below, selected music to the right and a search function at the bottom. Finally a load of digital data on the current track. Its excactly as I want it. I actually managed to mess it up recently and it took me ages to recall what to do to get it back! Just thought it would be good to know there was a reference to come to, (when time permits of course!).

  2. Anonymous says:

    Fideliser is great with Foobar IMHO.

    • I personally use a little frre tool called TimerResolution.exe V1.2 to set Windows timer resolution to 0.5ms (max supported) instead of the default 1ms of W7, this reduces average RPC latency in my PC from 80-100ms to 40-50ms and smoothes out spikes so it very rarely goes above 100-120ms. (Fidelizer also does this but IMO it messes up more than what it fixes). It is possible configure ASIO or WASAPI to use high priority within Foobar, I have added how to do this to Part 1 of the guide.

      Javier

  3. Robert R Greene says:

    I have a Mytek Stereo 192 DSD DAC and like using pcm to dsd upsampling but I can’t use foo_dsd_asio 0.6.1 on up because Mytek can not do the math for 48KHz upsampling rate.
    It would be cool if upsampling for 48KHz could be possible. I would love having all my CD rips upsampled to DSD128 I really need to upsample 48KHz as I have music at that I do 96 and 192KHz but they don’t need to be up sampled

    • Hi Robert,

      I can think of at least four possible solutions for converting 48K based material to DSD128. One would be using the SOX plugin to previously convert 48K based to a 44.1 based sampling rate, for example 176.4K.

      Also you could ask Maxim at http://sourceforge.net/projects/sacddecoder/files/?source=navbar to add an option for choosing between converting everything to 44.1K based DSD like older versions of foo_dsd_asio did and converting using the same fundamental like current versions do. He is very open to suggestions and if he thinks it enhances his plugin he most probably will do it.

      A third option would be asking Mytek to include 48K based DSD in their firmware because I believe they use the ES9016 chip in their 192 DSD DAC which supports DSD based both rates so it should be doable, problem is whether they’d be interested as there is no commercial material in 48K DSD material and is not used in the Pro world.

      The fourth way to achieve what you want would be changing Foobar for HQPlayer (quite expensive) which can be set to output a specific sample rate like 5.6MHz with an amazing quailty and plenty of configuration options or changing to JRiver (resonably cheap) which can upsample to DSD128 in 5.6MHz.

      Newer DACs have no such problems so there is also the possibility of selling the Mytek and get something newer?

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