In just a few easy steps, you can convert any type of audio files to any format, such as: MP3, M4A, FLAC, MP4, AVI, and many others. Very much appreciate guidance on the above.Convert songs and audio recordings in just a couple of clicks! Does Illustrate have an eye to revisit the DSD codec in terms of filter performance, or is the sound today 'best can ever be' and the only updates to the DSD codec would be to support higher and higher DSD rates? I ask this because we know over the years Illustrate made great strides in regular SRC evolution, and was likewise curious if a similar approach is active/planned for DSD conversion. If the end user selects 88.2 from the drop down above, is the conversion done (a) 'directly and immediately' to 24/88.2, or (b) does the conversion actually happen to 24/96 first then DBPA 'secretly' executes the SRC DSP function (even though it isn't explicitly selected) to downsample to 88.2?ģ. 24/88.2? In other words, is this a design decision, based on the design/performance unique to DBPA DSD codec conversion and filter process, that Illustrate recommends 24/96 and hence it's the default?Ģ. What is the reason DBPA defaults to 24/96 vs. So, when starting a conversion with DBPA using DSD codec, here's how it presentsġ. Note that I am not citing sources or trying to drive a debate here. Most of the narrative I've read elsewhere about DSD64 (ie, what's on SACDs) is that DSD64 equates best to 24/88.2, not 24/96. If you wish to convert to a different sample rate / bit depth, you can insert DSPs to do so (preferred: sample rate first, bit depth later).I'd like to revisit this point as a humble non-professional. I'm usually doing DSD64 to PCM 24bit-88.2kHz.ĭon't forget to dither to the final wordlength (16bit or 24bit) as final step.ĭBpoweramp turns DSD into 96KHz 24-bit PCM, which can then be written to any file format, WAV, FLAC or another. If you don't want to burn a CD I would stay at 24bit. sox with: gain +6 sinc -L -26k, also good for SRC, -b 24 rate -v -a 88.2k dither (complete cmd: >source file gain +6 sinc -L -26k rate -v -a 88.2k dither) TDR Ultrasonic Filter (VST2 32 and 64 bit, alpha state but great working, parameters 25kHz, 100%, 100%) I did a little research and found above 26kHz you can't distinguish the music signal from the noise shaped noise, so I lowpassing at 26kHz with a steep filter min. With source DSD64 and target fs at 44.1kHz or 48kHz there is no need for a seperate lowpass filter as the signal is steep lowpassed at fs/2 in the SRC process.įor target fs 88.2kHz and above a steep lowpass filter starting at 30kHz in front of SRC is advisable. But seeing as human hearing top end is around 20kHz it's not likely to be an issue.in truth it is fs/2 (nyquist theorem) You can't sample anything higher than the sample frequency. I doubt that I can hear any frequencies above 20 KHz, but I leave them in their native DSF format at home. I leave them in DSF for my home streamer. And they take up too much space on my iPhone. I bought the files of the masters that I wanted in the only format offered, DSF, and I would like to play them from my iPhone in the car and on my headphones, but moving the DSF files to the iPhone through the program Waltr2 seems to preserve their native size. And you, as the consumer must determine whether it is better, the same, or worse. If the actual audio has been manipulated in some manner, that is really an artistic matter, for better or worse. I do understand that some material re-released in DSD has been "remixed" or "remastered", whatever that may or may not mean. I included the word possible because it is very questionable whether you could actually hear any difference between 96/24 and 44/16 lossless files from the same source in a properly conducted A/B/X test. If you want to get any possible benefit from the DSD files, you would convert them to 96/24 lossless (PCM/FLAC or whatever) or better files. This does raise the question, if you spent more for the DSD sources, why did you spend the extra money if you are converting them to CD quality files? For all practical purposes you are ending up listening to a file that is the same as the material on the CD. So if playing CDs doesn't damage your tweeters, then playing the converted files won't. In practice, as the low pass filter already in your digital to analog converter has some "slope", the upper reproduced frequency slopes off, provably starting about 20 kHz. Assuming that your digital to analog converter is working properly (and it works the same for these converted files as it does for a ripped CD) there is nothing of significance above 22.05 kHz that will be reproduced, per the above mentioned Nyquist/Shannon theorem. 44 (presumably actually 44.1) thousand samples per second, 16 bits of resolution. A 44-16 file is the same as what is on a CD.
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