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Hi, I'm trying t get the best from my configuration JRiver Media Center 24 (on a fanless i3, Windows 10) Audiolab M-DAC+ (using the provided IUSBAudio Driver ASIO) Since the M-DAC+ supports up to DSD256, I give it a try, using the options provided by JRiver Media Center 24, and, since then I'm converting on the fly my material from PCM to DSD128 (it seems that my i3 is unable to do more than that without stuttering). SQ is heavily modified with the standard material, in a way identically described all over the world. I asked to my my guitar teacher to compare the different versions, without telling him what to expect, and, when I asked to listen a 16/48 converted to DSD128 he jumped on the seat, saying: "Uh!?!". Me, I'm using DSP Studio to do the conversion. Let me guess that I'm bypassing the Windows Audio engine, since with this configuration I can disable the JRiver Media Center 24 and Windows 10 volume). Is it the better choice? By the way I tried also HQPlayer with identical results. Now it hangs, and I don't know the reason. Perhaps I have modified the wrong parameter among the 100's available (HQPlayer reminds me a Caterpillar, full of leverages. On the opposite there are cars with the automatic gear. Me, I like better my Alfa Romeo with a manual gear, and so my JRiver Media Center 24). Thanks

Hi, I'm trying t get the best from my configuration JRiver Media Center 24 (on a fanless i3, Windows 10) Audiolab M-DAC+ (using the provided IUSBAudio Driver ASIO) Since the M-DAC+ supports up to DSD256, I give it a try, using the options provided by JRiver Media Center 24, and, since then I'm converting on the fly my material from PCM to DSD128 (it seems that my i3 is unable to do more than that without stuttering). SQ is heavily modified with the standard material, in a way identically described all over the world. I asked to my my guitar teacher to compare the different versions, without telling him what to expect, and, when I asked to listen a 16/48 converted to DSD128 he jumped on the seat, saying: "Uh!?!". Me, I'm using DSP Studio to do the conversion. Let me guess that I'm bypassing the Windows Audio engine, since with this configuration I can disable the JRiver Media Center 24 and Windows 10 volume). Is it the better choice? By the way I tried also HQPlayer with identical results. Now it hangs, and I don't know the reason. Perhaps I have modified the wrong parameter among the 100's available (HQPlayer reminds me a Caterpillar, full of leverages. On the opposite there are cars with the automatic gear. Me, I like better my Alfa Romeo with a manual gear, and so my JRiver Media Center 24). Thanks

Now I finally found a solution to extract SACD ISO files on the fly. No need for Terminal, or the SACD-Xtract software... I just use AudirvanaPlus … This cool software has ALL I NEED!!!!!! There is an option: "Add Files To iTunes" in AudirvanaPlus. Be sure first to uncheck the two boxes in the iTunes Prefs->Advanced: "Keep iTunes Media folder organized" and "Copy files to iTunes Media folder when adding to library". I then drag my ISO file to the new opened window, set the "Quality of the proxy files to "Uncompressed" and whushhhh…. AudirvanaPlus extracts, and converts all the files I need. And adds them to iTunes too When finished, remember to tick the two boxes in iTunes again. Hours, or even weeks of conversion are just done within minutes now. Best Regards KK

Can audiophilia lead to suicide or mental institution ? This thread will be my chronicle ; feel free to bash or help. I was happy until my usb cable broke as a new DSD vs PCM debate broke as well. All that hatred... Then I was away from home for the holidays and when I came back last night, the computer full of files to test (upsampled dsd, SRCed PCM, what have you) I just found my system horrid. Was it the cold, the fatigue, the storm that cut the power for 36 hours (but a relative switched it on before my return) ??? or was it like when you haven't seen the loved one for several weeks and for a short while you know what you will know for sure only after the divorce, split, what have you : in fact she's very far from perfect, maybe not even your kind ; but after some warm up you're fulfilled again... just as I think my system is quite great today. Did I contemplate the truth or... was it fatigue, required breaking time, what have you ?? Anyway, I started playing with my files, wondering about crazy things such as should I feed my TEAC 501 with straight dff files or use Audiogate to either upconvert to 128 or convert to pcm so I can EQ (gosh ! isn’t 192 better while it should be 176.4? src ? eq? which settings ?where ? rx3, A+, offline, on flight?????)AAaaargh!! So far I have taken one decision : src no more, be it rx or A+, no upconversion at DAC level either, Slow filter (linear, almost no pre-ringing, aliasing at 25 K though), EQ (includes a Low Pass at 20K so I don’t care for the aliasing) To be followed… if I’m not institutionalized…

Hello, I'm working on a DSD to PCM conversion tool and I've just been looking into the click that you often get at the start and end of the converted tracks. I've written up a short entry on the topic here: https://code.google.com/p/dsf2flac/wiki/ClicksAtStartAndEnd which some may find interesting. All the best, Jack.

Hi there, new to the forums and will be the first one to tell you I'm an audio newbie, so bare with me ! I have a standard Creative 5.1 surround sound system for my PC ( http://img508.imageshack.us/img508/1731/inspirep580car9.jpg ) They are great for gaming, but I find they are no good for making music, headphones are good but I just dont like using them. And recently acquired a pair of Philips Hi Fi speakers and was wondering how I could use them with my PC .. Is it possible to replace the front left and right (5.1 speakers) with the Hi Fi speakers, or mount the speakers on to a headphone jack ? I've done a bit of reading and most suggest an amplifier but at the moment thats a tad expensive for me, and advice, info or suggestions welcome. Thanks for reading!

Hello, If anyone is interested I have just uploaded some code I've been working on for dsf to flac conversion. You can get the source at https://code.google.com/p/dsf2flac/ Although there are other conversion tools out ther (Audiogate and Saracon for example), I wanted to have something where I could play with filters, dither etc and see exactly what is going on. It is early days but it seems to work quite well, runs at around 14x on my laptop (core2duo [email protected]). I've only tested on 1bit stereo DSD64 files on linux X86_64 and there are no binaries yet but thought it might be interesting for the enthusiast. Cheers, Jack.

Hi audiophiles! I've some concerns about decompressed and lossless conversions using XLD, MAX and some other software. So for example I have some hi-res FLAC audio files. If I convert them to AIFF, they will add in size as expected. But if after I will compress these new AIFF files back to FLAC the new output files will be significantly smaller than original ones. And if you'll continue this routine the decrease in size of original files also will occur. Depend on software you're using (I've tried XLD and MAX as a mac user, but I know that some win software exhibits same properties) the exact decrease in size will differ, but will occur anyway. So my question is how that happens and in this case don't these conversions really affect quality?

An apple AAC file contains my apple-ID, email address and other personally identifiable information or details in every purchased AAC file somewhere. If I convert a purchased AAC music file to Lossless file format does the new lossless format contain any of this personal information ? The reason I want to do this is the risk of my files getting stolen or copied is high in the environment I am currently in and I do not want this personal info to go with files if that happens. If a purchased AAC file is converted to MP3 format I assume the personal information is carried over to the MP3? Is that correct ? I understand that after I convert the files, the audio quality will be roughly the same and that I most certainly will not gain any audio quality.

Note: This is going to be a little complicated, so I will try to cover every factor in this problem I've had. I purchased some DSD (352 khz) files of solo piano music that's fairly dynamic and detailed, with loads of gorgeous harmonics. I prefer to play my music in 24/88 or 24/96 format, and also be able to convert to 16/44 WAV to play on my 256 gb Apple devices. So I purchased a well-respected converter (Mac version) to make the 24/88 FLAC and WAV files, and carefully followed the instructions given by the music critic who promoted the album, as well as the detailed instructions given by the converter gurus. Those 24/88 conversions not only wouldn't play on my PC in Foobar2000, they corrupted Foobar so badly that I had to reinstall it. That process of conversion from DSD to 24/88 and attempts to play on Foobar followed by reinstallation was repeated many times, with input from the persons noted above, with no success. Note that I have over a thousand high-res (24/88 to 24/192) WAV and FLAC files from HDTracks and several other such sites, and I've never had a single problem playing those on my Foobar2000 player. And I tested a number of those again during this process. So I finally decided to convert the DSDs to 16/44 WAV files, and those played just fine on Foobar2000. But when I converted those WAV files to FLAC, the FLAC files (Level 5 FLAC) were smaller than MP3s, i.e. about 23 percent of the 16/44 WAV size. Using that same Foobar2000 and both FLAC converters I have (ca. 2007 and 2014), I've converted several thousand CD/WAV rips to FLAC over the years, and none of those FLACs were less than 60 percent of the WAV file size. I even went a step further to see how my trusty FLAC converter could make files smaller than the corresponding MP3s - I converted a few MP3s (320 kbps, 4 minutes and 10 mb each) to 16/44 WAV tracks (44 mb each), and then I restarted Foobar and converted those WAVs to FLAC. Those FLACs, which originated from the MP3 files, were now 24 MB each, and Foobar showed a bitrate around 760 kbps when playing. The "23 percent" FLACs noted above showed a bitrate playing in Foobar of 290 kbps. So my question is, given that I cannot make a FLAC that's 23 percent of a WAV file size from any of the thousands of WAV files that I've ripped or purchased, or even from up-rez'd MP3s, what could possibly be in the WAV files I got from the well-regarded Mac converter that I purchased, that would trick my FLAC converter into making FLACs smaller than MP3s? The converter software gurus say that the DSD recordings have "extremely low complexity", which makes no sense, and doesn't fit with all of the other experience I've described.

My father has 1000+ CDs spanning all genres and I have tasked myself with ripping them to a hard drive. I'm an audio engineer so I have a plan for format of conversion, though I am now considering PhileAudio because it could rip a lossless version and a compressed version from that, also may offer increased efficiency and multi-drive support. I have the following computers to rip with: PowerMac G5 (PowerPC, so likely iTunes rip) MacBook Pro Core2Duo MacBook Pro i7 iMac i5 Microsoft Surface i5 (if I decide to throw a PC in the mix) I have ordered one each of these extra drives to try in order to speed the process, was going to stick with brands I know but I went by Amazon rating barring any specific Mac incompatibility: LG Buffalo Coolead Samsung Looking to get a 4TB USB 3.0 and Firewire or Thunderbolt drive for lossless storage, should have enough space on the home computer for the compressed collection. My concept is to have some friends over who like music, bait them with booze and free music, and have a rip party. If anyone has any ideas for organization, workflow or anything else please let me know!

Interested in the facts? One of the world’s top converter designers Dan Lavry has written a new paper in simple language to demystify the subject. http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf See why many professional engineers still work at 96kHz years after 192kHz became available. Find out why “more” is not always “better!”