![]() ![]() Last question: under that same SACD section, there's another option labeled "DSD processor" with the options "DSD processor". ![]() What do each of these do, and what's the best one? Online I found instructions saying to choose "DSD", but does that convert everything to DSD? Does PCM+DSD allow for both codecs to be transmitted without any kind of conversion between the two? The choices are "PCM", "DSD", and "PCM+DSD". Under preferences>tools>SACD, there's an option labeled "output mode". Under preferences>tools, there's a section called "DSD processor" with a box labeled "Use DSD Processor" that I can select. Will that make Foobar automatically upconvert anything that's not already 32-bit, and if so does that affect sound quality? Also, does this affect DSD files? Under preferences>playback>output, I have to choose between 8, 16, 24, and 32-bit for output data format. What do all those settings mean, and do I have it in the right one? I don't want to click through them all and test because last time I tried to, it screwed with playback and I started getting error messages, no matter output I chose after. I also have the choices of "ASIO: DSD Transcoder (DoP/Native)", "ASIO: Sony Audio Driver", "DS: Primary Audio Driver", and "DS: CAS-1 (Sony Audio)". Under preferences>playback>output, I have my output device set to "DSD: ASIO : Sony Audio Driver". Are any of those redundant/unnecessary? I've tried playing around but can't figure it out. I have ASIO support, DSD Processor, DSDIFF Decoder, and Super Audio CD Decoder installed into Foobar. I downloaded and installed a few Foobar components to support DSD playback. ![]() Will I affect anything by changing it to 192khz? In Foobar, the default maximum sample rate is 88.2khz. Is that the right setting to be in? When I run foobar, will it automatically change the output based on the track, or is everything going to get converted to 32/192? In Windows, I have the USB audio output of my Surface Pro set to 32-bit/192khz. Here's a list of issues I've tried to research on my own but can't find answers to: Unfortunately the CAS-1 doesn't support DSD playback beyond 2.8Mhz, so my DSD 128 files can't be played. I have a couple 32-bit WAV files too, but they're 48khz. Right now the best files I have are DSD 64/128 and WAV 24-bit/192khz. It supports DSD, FLAC, and PCM up to 32-bit/192khz. I have a Sony CAS-1 DAC/amp, which is powering a pair of LS50's. positive values) results in distortion known as "clipping."I'm trying to set up playback of hi-res files using Foobar2000, but I'm having a really hard time figuring out what all the settings mean, and what options to choose. Anything below that is expressed as a negative value below full scale. For PCM audio-most digital audio like CDs, DVDs of all stripes, Blu-rays, FLAC, MP3, etc.-program amplitude level is measured against a maximum limit of 0dBFS ("zero decibels full scale"). It has to do with a difference in reference level. ![]() I find -1dBFS to be a fairly safe normalization target. Instead, I would convert to PCM and then normalize to a sensible level on a per-album basis (to preserve inter-track dynamic relationships). I don't, however, recommend applying a blanket +6dB of gain to all PCM conversions, as many SACDs peak above 0dBFS, or -6dBFS In PCM terms. Therefore, when converting DSD to PCM, you may find that the resulting audio peaks well below what you might expect from native PCM. Because of this, it is possible to exceed "full scale" in DSD (not truly full scale) by nearly 6dB without clipping. positive values) results in distortion known as "clipping."įor DSD audio as found on SACD, Sony decided to set the 0dBFS reference level at the equivalent to -6dBFS in PCM. Click to expand.It has to do with a difference in reference level. ![]()
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