Hello all, perhaps some minds here can help. I am 40 years experienced electronics technician, having specialised in the Audio electronics world in all flavours. I have developed a mixture of materials that can be applied within cable connectors or as an enclosure fitted around electric/audio cables exactly in the manner of the now common ferrite clip-on filter assembly. I have been exploring using a disused Android phone as a music player, ie iPod substitute. The standard process is to transfer Wav audio files from PC hard drive to Phone flash memory via a USB cable, and then the phone can be disconnected from the PC and the phone headphone/line output signal connected to a standard hi-fi system for playback of the stored Wav file. I have found that the playback is changeable according to the particular USB cable used to transfer the Wav file to the phone. I have developed a protocol which is as follows. On the PC hard drive I create a set of folders - Orig, 01, 03, 05, 07, 09 etc. On the Phone I created a set of folders - 00, 02, 04, 06, 08, 10 etc. I can choose any particular Wav file and copy this into the Orig folder. Using a standard off the shelf USB cable I copy this Wav file to the Phone folder 00. I then change out the standard USB cable and replace it with a custom USB cable incorporating the filter mixture contained in both connectors. I then copy the Wav file from folder 00 to folder 01, and then copy this newly created file to folder 02 and repeat this process such that the folder number designates the number of transits that the Wav file has traveled the custom USB cable. I can then disconnect the phone and use it as a player, and select whichever Wav file at will. So the result is that Wav file 00 playback sound is different from the 'filtered' wav files contained in folders 02, 04, 06, 08, 10, and that the subjective difference increases according to the number of transits. I can run AB comparison tests perfectly easily with A being the #00 file, and B say the #10 file. So to cut to the chase, I am finding that particular Wav file copies alter the behaviour of the playback system, inducing a 'set' to the system, and according to the peak level encountered. To clarify, I can play the 00 (A) file multiple times and after the first play, subsequent plays do not change. If I then switch to say #10 (B) file, the system retains the #00 set/sound until the #10 file reaches higher/peak amplitude and the system then takes a new 'set' corresponding to the #10 file. Replaying the #00 file restores the system 'set' back to the original condition. So to summarise, A and B playback sounds are different, and further ABABAB playback sound is different to AABBAA playback sound. All files are returned as being identical when compared in binary file checker softwares. The subjective differences between the files is great enough to be perfectly reliably identified in ABX testing protocol. The subjective differences include rather better centre image placement and stability, greater clarity of depth/distance information and overall distinctly better clarity and musicality. So, it seems that there is some 'deeper' information contained in the Wav file that dictates/modifies the excess and/or 1/f noise behaviour of the playback/transducer system. Some informal external soundcard loopback testing showed significant reduction (6dB+) in VLF 1/f noise. Applying versions of the clip-on filter to all analogue systems such as phono and guitar amplifier significantly transforms such amplified sound. Applying filters to the signal multicore cable and power feeds of large scale stadium PA system produces same subjective improvement, but includes significant (5-10 secs) delay. Applying filters to transmit and/or receive antennas also alters subjective sound. Anybody have ideas of what's going on here ?. Eric.