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Showing results for tags 'upsampling'.
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New free real-time plugin for foobar2000 (64bit version) We are happy to announce a 'free as in beer' real-time plugin (foo_pggb_rt) for foobar2000 (64bit version) that will allow you to resample local tracks using the same core engine as PGGB 256, the limitations are that the processing is done only at 64-bit precision, and you cannot save them using 'convert' feature. The plugin was developed in collaboration with Audiowise and licenses PGGB-RT SDK. Guide: foo_pggb_rt - ZB's Guide (remastero.com). FAQ: PGGB-RT - FAQ (remastero.com) Download: foo_pggb_rt - ZB's Guide (remastero.com) There are two processing modes: 'Maximize Quality': will process the whole track but you will incur longer startup delay on the first track but process the whole track in one go just like PGGB 256 would at 64-bit precision. 'Maximize Speed': will process the track in about 23 second chunks (equivalent to 16M taps in terms of length @hanshopf) Features: Just like PGGB 256, the component supports both upsampling and downsampling PCM up to 64fS rates. It supports bit depths between 16 and 64 bits, so should work with most DACs. It also has the same noise shaping and HF filtering options of PGGB 256. EQ is possible too, but EQ filters need to be imported using PGGB 256 app (trial license will work). EQ is still a beta feature. Uses the same algorithms as PGGB 256 for all processing except only 64 bit processing is done due to real-time nature. How does this compare to PGGB 256? PGGB 256 when set at 64 bit precision will match the real-time processing of foo_pggb_rt when set at 'Maximize for accuracy' and all other settings are set the same. But please note that foo_pggb_rt is meant to just provide a glimpse of what PGGB is truly capable of since, PGGB 256 at 128bit or 256 bit precision is still more accurate.
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- foo_pggb_rt
- PGGB
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Why are there debates if "bits are bits"? Can you hear a difference in streaming audio software? Bits are Bits! Foobar2000 was lighter on its feet, more accurate tone/timbre, quieter, and had better timing of decays. There are a delicacy and precision about its sound. JRiver had a “fatter” sound with more bleeding edges and smeared textures. It sounded flatter (by a good margin) and more veiled in a thicker, unnatural warmth.
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A number of DAC manufacturer's are limitng the output for DSD with -6dB, compared to output for PCM. I have personal experience with Holo Audio Spring DAC and iFi micto DAC's. Other models from these manufacturers do the same, as specified in their manuals. I understand that theoretically DSD can go up to 0dB, and even uo to +3dB for short peaks. But in practice, this turns out to be counterproductive, at least when upsampling both PCM and DSD (DSF) to DSD256/DSD512. From the same SACD album, upsampling the SACD layer (DSF file) to DSD256 results in (+/-) 6dB lower output level, compared with the redbook layer (AIFF file) . This means I have to crank up the volume by 6dB (on my preamp) to keep same sound level. Using a DAC from one of the forementioned brands, the output level is pushed down another 6dB, as the DAC receives DSD data. As a result, I am receiving a -12dB level with DSD, compared with PCM. So, why are these (and probably other) DAC manufacturer's do this? I find this rather annoying to say the least (one of the reasons why I have sold my Holo Audio Spring dac) Dirk
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When I buy SACDs, how can I be sure they aren't just Redbook CDs that have been upsampled?
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Just read this article on Digital Audio Review shared by Srajen Ebaen of 6moons. Very interesting to say the least. http://www.digitalaudioreview.net/2017/07/kih-46-mqas-missing-link/ Best. Mev
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I wonder if anyone either using Roon or HQPLAYER to upsample to 384 to a (Hugo) DAC via Raspberry Pi could check the following for me. Basically at this upsampling rate I get left to right speaker inversion. Totally reproducible and doesn't happen at any lower rate. It also doesn't happen with direct connection between PC and HUGO. Searching doesn't seem to come up with anyone else having this issue, I'm unsure whether it is due to an interaction between Hugo and Raspberry Pi or just a Pi issue as I have no other DAC to check. So I'd be obliged if anyone upsampling to 384 with Raspberry Pi can check if Ringo moves from left to right speaker or whatever track you have that will display the left to right inverted nature. Im using dietpi. 384 kernel activated. Thanks in advance. .sjb
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JRiver has implemented upsampling using the open source SoX codec for a while now, but it is only implemented as a checkbox "on or off" on the local playback. The full parameter set of SoX enables the user to choose between linear o minimal phase, steep or slow roll off options. This is important to upsample in what one can consider a more musical way, improving the sound reproduction (optional!). In the link below I have compiled a few graphs of different hardware implementations. I invite everyone who thinks this is important to post a +1 on their forums for the feature request. Thanks! https://yabb.jriver.com/interact/index.php/topic,113602.0.html
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I am really looking forward to taking delivery of a Schiit Bifrost Multibit in the next couple of weeks (as soon as the next batch is scheduled to go out). I use Audirvana to play everything (including my iTunes library). For 176.4kHz and 192kHz input sample rates, the Bifrost runs in non-oversampling (NOS) mode. What does this mean for input sample rates that are less than 176.4Hz? Most of my music files are 24/96 or 24/192, but I also have ripped all of my CDs in 16/44 lossless. I know this is probably a personal preference, but for those of you with this DAC, do you do any upsampling in Audirvana for any of your music files? If so, what settings? My apologies in advance if this is a stupid question with an obvious answer. I am still very much a newby. Thanks!
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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…
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- conversion
- dsd
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Hello folks, this is my first post around here. I am new to computer audio. Recently I put together this desktop system: - Macbook Pro Retina (Late 2013) >> Audirvana Plus 2 >> AudioQuest Pearl USB cable >> DAC TEAC UD-301 >> XLR cables >> Active Speakers JBL LSR305. I am pretty happy with it, since I used to listen to music in cheap earbuds, and now, with less than 600 bucks, my experience have changed so much! I have a few questions for the community: 1) I am using my DAC to control the volume, but in the transport area on Audirvana I can't see the volume knob representation. Is this normal with the TEAC? 2) I use macOS Sierra 10.12.1 in my MBP, and when I try to enable Direct Mode plus Integer Mode, there is an error when trying to playback music. Just Integer Mode enabled is fine, no problem. Is this normal? 3) Will I have an audible improvement by upsampling Tidal 16/44.1Khz music to 176.4Khz using the Teac feature? Or should I do it within the Audirvana Plus software? Or not at all? 4) Lastly, what setting tweaks do you recommend on A+ for my hardware configuration? Thank you for all the useful info I got from this community already! Diego
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- audirvana plus
- tweaks
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