or Connect
New Posts  All Forums:

Posts by Roseval

Most of the time headphones have a separate left and right hot wire and a common ground. Of course on can wire them as one wires speakers to an amp using 2 pair of wires, one ground and one signal for each channel. Notice that nobody calls the way we wire speakers to an amp balanced. I would say understandably because there is nothing balanced about this wiring schema. It is symmetrical, that's all we can say about this wiring schema.   Hence the question is what...
I'm afraid it is a long, long time ago Plextor was making its own drives. This might be a useful resource: https://forum.dbpoweramp.com/showthread.php?37706-CD-DVD-Drive-Accuracy-List-2016   As ripping CDs is a tedious process, I suggest to rip to a lossless format. Most media player can do transcoding. This means if you sync to a portable, they convert to e.g. MP3 on the fly.
I’m afraid you are talking about something completely different. The OP asks about converting a file to a file with a different sample rate and/or bit depth.As this is from file to file, we do stay in the digital domain hence no jitter is added or removed. You are referring to ASRC, asynchronous sample rate conversion, that what the DAC1 is doing. This is decoupling the clocking of the input stream from the clocking driving the DA.Here indeed we talk about actual physical...
The standard says 5m max but those standards are a bit conservative. You might try 5+2 or buy a 7m. Alternatives are 5 > USB hub > 5m USB Booster cable USB extenders: http://www.thewelltemperedcomputer.com/HW/USB_Extender.htm
The only “domesticated” version I know is the Merging Technologies NADAC. At $10,500 it not only has an audiophile price tag but is more than the double of the pro-version (Hapi).  Better stick to the pro-version including all those things we don't need! http://www.thewelltemperedcomputer.com/HW/Connect/Ravenna.htm
It isn't: https://en.wikipedia.org/wiki/Oversampling
Maybe this link is of use: http://www.hardwaresecrets.com/how-on-board-audio-works/3/ It confirms what ProtogeManiac says, all analog input is digitized and converted back to analog by the soundcard. Even if you get it to work (USB audio out + soundcard out, maybe Virtual Audio Cable can do that) you are still using the DAC of the soundcard. As this is the case you might as well skip the USB part entirely or get yourself a USB DAC with a decent headphone out.
A CD contains PCM audio, 2 channels, 16 bit word, 44.1 kHz sample rate. If you rip it to any lossless format with the same characteristics (2/16/44.1) you have a one to one copy of the content. This is in theory. In practice there might be reading errors. Hence it is recommended to use ripping software with a secure mode. A nice feature is AccurateRip. The ripper calculates a checksum and compares it with the AccurateRip database.  An extra lock on the door.   Maybe...
What I understand is MQA can do two things: Lossy compression of Highres recording Compensating for time-domain errors   It compresses Highres audio into a 24 bit / 44.1 or 48 kHz PCM format. As the result is PCM, it can be treated like any PCM format. It can be contained in any lossless format like WAV, FLAC, ALAC, etc. The lossy compression is rather complex. It is based on the assumption that below -120 dBFS recordings contain random noise. Compress the part above...
If you do why not stick to Foobar? Tagging classical is a problem.The only tool I found doing is decently is MusiChiYou get your composers spelled right and this applies to a lot of compositions as well.http://www.thewelltemperedcomputer.com/SW/Players/MusiCHI.htm
New Posts  All Forums: