Sennheiser PC37X randomly goes bad after disconnecting the cable ?
Greetings, Yesterday I was using my headset like normal with my macbook, just listening to music and on a call with people like usual, and the headset was perfectly fine. The stock wire that came with the headset is extremely long and yesterday it annoyed me very much that it kept getting tangled with itself, so I decided to see if the cable is replaceable. I pulled out the cable from the headset and saw the adapter, and looked online for a replacement. Upon plugging it back in, the audio sounded extremely muffled and washed out. Im not sure what I did wrong to make it mess up like that as I've always taken good care of it, ive had it for about 2 years and its always just been chilling on my desk, but anywho I thought the cable just went bad and ordered a replacement. The replacement came, and the issue is still persistant, so I am not sure what the issue is I've tried multiple different headsets and the issue is not with the port, and I also tried it with my windows laptop and...
Apr 23, 2024
If you are not already using a "player" app; as Marlowe and Kxrider have said put the PC master volume, depends on actual PC DSP hardware, to 80% + or the on-board PC DAC basically gets "dumbed-down" to a lower bit rate. 100% is fine as your are not trying to actually use the PC's op-amp; jack out. I used to set it at 95% but that is just me, now I am using Audiogate.
" Reducing volume in software is basically equivalent to reducing the bit depth. In digital audio, the signal is split up into distinct samples (taken thousands of times per second), and bit depth is the number of bits that are used to describe each sample. Attenuating a signal is done by multiplying each sample by a number less than one, with the result being that you're no longer using the full resolution to describe the audio, resulting in reduced dynamic range and signal-to-noise ratio. Specifically, every 6 dB of attenuation is equivalent to reducing the bit depth by one. If you started with, say, 16-bit audio (standard for audio CDs) and reduced the volume by 12 dB, you'd effectively be listening to 14-bit audio instead. Turn the volume down too much and quality will start to suffer noticeably.Another issue is that these calculations will often result in rounding errors, due to the original value of the sample not being a multiple of the factor by which you're dividing the samples. This further degrades the audio quality by introducing what's basically quantisation noise. Again, this mostly happens at lower volume levels. Different programs might use slightly different algorithms for attenuating the signal and resolving those rounding errors, which means there might be some difference in the resulting audible signal between, say, an audio player and the OS, but that doesn't change the fact that in all cases you're still reducing bit depth and essentially wasting a portion of the bandwidth on transmitting zeroes instead of useful information. "
Death by PowerPoint here, actually its fine: http://www.esstech.com/files/3014/4095/4308/digital-vs-analog-volume-control.pdf
To get all the perfect little bits out though, you need to get past the on-board mixer/kernel altogether in the PC by using a player app such as JRiver or FooBar. A free trial is available, https://jriver.com/download.html , and you will want some high rez music like 24/192 and now you can also use DSD 64 (DoP transfer) files with the AuneT1SE. A white paper is here covering why you want to bypass your PC DSP :
http://m.marantz.co.uk/DocumentMaster/master/Marantz_Whitepaper_PC-Audio.pdf
What you are trying to make, by bypassing some normal PC stuff, they were not made for this... is a full music player system using your PC and an out-board better DAC, but a bit cheaper, well most have a PC or Mac handy so that is paid for already, and to keep control over components/software selection. The-all-in one, minus an amp, is done like this; little known fact, Sony speakers move your hair even at low volume, watch video:
https://www.sony.com/electronics/audio-components/hap-z1es
What you are doing with a "player", or open the white paper above for the whole story, well written, looks like boiler plate a few are using:
"Audio handling inside a PC To better understand this, take a look at the audio-handling diagram below – From the Media Player the audio will be send by Direct Sound, the default setting, through the Mixer, while Kernel Streaming, Asio and Wasapi bypasses it. The dedicate driver installed will allow the USB DAC to take the lead for the data request – this is called Asynchronous Mode. Just for the PC environment the Media player, for example JRiver, default setting is Direct Sound and uses the Mixer. So it is necessary to change the setting to Kernel Streaming, WASAPI or ASIO streaming. "