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Showing results for tags 'foobar2000'.
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LoryWiv posted a topic in DAC - Digital to Analog ConversionI've not seen much written about this DAC/AMP here or on Head-Fi, but I wanted a portable to stack with my DAP and to run balanced output to my 64 Audio A6, that would also serve me well as a USB DAC for PC. I took a leap of faith on this unit, bought from Moon Audio, and thus far pleasantly surprised by it's quality. The unboxing and ergonomics are nothing special, rather plain and uncluttered in appearance: That said, implemented thus far as a USB DAC from PC (still waiting for proper interconnect to stack w/my DAP using the KEB-03 Toslink input) I am impressed with it's sound signature. Using the 2.5 mm TRRS, AK-type plug on my IEM's, the balanced output achieves wonderful, wide soundstage, although depth and height are less impressive. Layering, instrument placement and separation are also strong points. I do feel it lacks some of the detail retrieval of my iDSD Micro., and whereas the Micro. brought out the lower frequencies in my A6, KEB-03 synergy with these IEM's provides a more neutral frequency response overall. The iDSD / A6 combo. has visceral, impactful bass but bleeds sometimes into mids. With KEB-O3 I hear a bit cleaner, grain-free and more forward mids (vocals sound terrific) and greater top end sparkle. The Micro. doesn't have balanced out put, so this may account for the soundstage / separation difference in part, but the overall signatures sound different beyond that as described. Both have Sabre DAC's, which I sometimes find a little sterile, but as implemented in the KEB-O3 signature is overall more balanced across the frequencies, quite musical, and less "digital" sounding than I sometimes perceive with the iFi Micro. I'm not very well-acquainted with "tube" sound but from the auditioning I did last month at CanJam SoCal. of the Woo Audio and other products, the KEB-O3 sounds more "tube-like" to me than the Micro. and suits most of my material well. I listen to a lot of classical piano, and the sound is nicely balanced, engaging across the frequencies but never shrill or strident. For classic rock or acoustic jazz featuring standup string bass, I do miss the low end "punch" the Micro. coaxes out of my A6 that the the KEB-03 can't quite replicate. Overall, I'd conclude that the KEB-03 offers great value for a moderate price, and strengths are the balanced output producing wonderful soundstage and instrument separation. Weaknesses are that it falls short in detail retrieval and bass impact, and I would add that it is not a powerhouse (balanced output is 131mW/channel @22 ohm load) and may be best suited for IEM's or low impedance / high-sensitivity cans. Lastly, a question: The manual (here) indicates it can decode native dsd via USB but I can only get it to do DoP in Foobar. I have foo_dsd-ASIO installed, works fine with iDSD Micro. The manual appears to be translated from Japanese so I may be misreading it (it could be indicating that native DSD is implemented for Android USB OTG), or perhaps I have Foobar configured incorrectly. Any help would be appreciated.
I use Foobar2000 on a small Samsung laptop (2013, SSD, 8gb RAM, Windows 8.1, 64bit) to stream music from a Synology NAS to an Asus Essence One USB DAC (I know, not the best but a setting that fits my limited budget...). My 300 GB of music files include downloaded MP3s, lossless high-res and CD quality FLAC files and WAV files, as well as FLAC files ripped from my CD collection. Therefore, bitrate varies considerably between tracks. My problem is how to select the best audio output settings. The Foobar2000 website advises using the Windows settings which allow to select up to 24 bit / 192 kHz output. The problem is that I then need to manually select the bit / kHz settings each time, otherwise it would modify all files to fit the selected numbers (eg a 16bit / 44100 Hz file would undergo processing to become a 24bit / 192 kHz file which I obviously don't want). The alternative is using an ASIO plugin which I have done, however I read lots of people resent doing that for many reasons, and I am sometimes annoyed by "clics" at the beginning and end of tracks (an issue I have read others have also experienced). I am thankful for any comment / suggestion on which Foobar2000 audio output settings would best fit my needs and automatically adapt to different bit rates. I know some people may advise me to start using JRiver, but though I might do just that at some point, I still like Foobar2000 which is free and has served me very well so far. I also take advantage of this thread to thank Chris and all CA contributors for their hard work Thanks, Chris
Hello Computer Audiophiles! I need your help if you are failiar with MPD module development. I am almost new to this community. I have developed active crossover such as http://www.aedio.co.jp/download/LatestModules/foo_dsp_channeldividerF2|F2B|F3|F3B.dll for those who likes to use multi-way speaker with multi-channel amps using FIR technology. THESE ARE VOLUNTEER BASED DEVELOPMENT so this is distributed free of charge! See Down load from file Recently I am now asked by several users if we can apply this active crossover to MPD or LINUX based system. I have Fedora / OSS / Lynx Based standalone application for this objective , but I donot know how to interface to MPD ,only I know and can do is, - Utilizing pipe feature of MPD configuration and pipe in filter program ( I have already) and out to pipe out and play by aplay -D surround40 or similar command. In this case we have to fix sampling rate if I use pipe out feature of MPD. This seems OK if I can resample 192KHz24bit or 96KHz24bit in MPD and fixing sample rate of data which was output from MPD to pipe out. So I like to know Is anyone be able to pass data from MPD to pipe or special API to modify data from MPD to our filter program? What I can provide you : I can provide you with latest source code of foobar2000 version, LINUX from pipe to pipe filter program (requires Non Disclosure Agreement) and know how to work with them. If this is realized I will distribute these modules at low cost for supporting overhead. Is there anyone who likes to do this type of Voluteer or small money development? Actually I have ported foobar2000 version to LINUX/Lynx Studio AES-16 by 4 Front Technology. But this is NOT ALSA but OSS because of Lynx Studio policy limitations. So now I am considering to port foobar2000 version (32bit integer buffer to FFT'd FIR filter processing with 64bit double precision and re-convert ot 32bit integer) to complete MPD additional module. I like someone who likes to cooperate with us to develop MPD based FIR filter development. Thanx in advance, Yutaka IIDA, CTO / Owner , AEDIO Japan
I started using Foobar2000 for the first time today after having purchased a Cambridge Audio DacMagic Plus. At first I was utterfly confused by the program, but after some messing around I started to understand how it works. I spent a few hours creating my own layout, and I'm quite pleased with the results. I then became interested in knowing how everyone here has set up their Foobar2000 layout and thought people might be interested in sharing. Here is my current layout