 In the past years I stated that audiophile network switches reduced jitter in the signal by using better crystals and other components. I was wrong. Let me start by saying that my knowledge of networks is sufficient to get swimming equipment played properly. I was already involved in networking in the mid 80s, but I have had no training as a network engineer. It was also in the mid 80s that I heard jitter the first time, although we didn't know it was called jitter. So when I heard the effect of audiophile switches, I thought I recognized the jitter fingerprint. Since the only variant in that setup was a switch, it was logical to come to the conclusion the switch caused jitter. But so I was told a network can't transfer timing information, at least not through the position needed for audio. The audio data is packed into packets that are sent from one IP address to the other and not even necessarily in the correct order. On the receiving side, the packets are buffered and, if necessary, put in the right order again. Then they are read from a buffer against the internal clock of the receiving device. So if there is jitter, it is due to the qualities of the clock crystal in the receiving device. And certainly not due to devices like the network filters I reviewed over the last year. But if it is caused by the crystal clock oscillator in the receiving device, there should not be a difference between an audiophile switch or network filter in place or not. But there is. There clearly is. And how come that many of the streamers, network bridges, network players and streaming DAX perform better with an audiophile switch or network filter in place? I know of only one digital player that doesn't sound different with or without an audiophile switch in the network, the Grim Audio Mewon. And even this player changes slightly when the costly network acoustics Mewon Pro filter is in between the non-oreophile switch and the player. My understanding of the phenomenon started while reviewing the Optone Aether region switch. Optone Audio founder Alex Kraspy sent me a copy of Aether region designer John R. Swenson's white paper called Understanding how perturbations or digital signals cannot find the sound quality without changing bits and how these issues are addressed by the Optone Aether region. It is now published on the Optone Audio site, a link in the description below this video on YouTube. Although being rather open about his theories, Swenson did not publish measurements to support it. It requires equipment that costs a fortune to buy and maintain. Products like the Aether region can only be developed by smaller companies and independent electronic designers that have spent enormous time and effort to acquire their knowledge. Their main drive is understanding what's going on, not what money there is to be made. I'm rather thankful to Swenson and some unmentioned designers that confirm his findings and were so kind to help me forward. In short, it is the contamination of the analog signal that is used to transport the digital information. That has the same effect on audio as play in the steering rack of a car has on the steering. It makes it less precise. And as the less accurate steering in a car is immediately clear to people that are used to driving a new or well-maintained car, the contaminated analog signal used to transport the digital information is easily identified by people that are used to listening to quality audio. For some reason, people with technical education often rather believe their measurement gear than their ears. Where they need a considerable time to measure an artifact, I normally hear it within minutes. Of course, I can't explain then why the artifacts occur, but to be fair nor can they with the exception of some really brilliant man like Swenson. And to be honest, being a reviewer, having bought about 3 generations of audio precision over the last 30 years, I think for a reviewer of digital equipment, the relevance is relatively low. Of course, I still measure everything I review, but only to see if there is something drastically wrong with the device on the test. I more and more have the feeling that if you want to do relevant measurements, especially with digital equipment, it should be in the time domain and at a far higher resolution than is done today. Think picoseconds. Not only in the digital domain, but also in the analog. My good colleague Jav Veenstra of alfaaudio.net has set up a measurement test for network switches. See the description for a link. He not only spent a lot of money on equipment, filters, certified converter plicks, a device to isolate the device on the test from the network and so on. He even bought a third hand to heal the test probe on the clock input of the DAC chip inside a Volumio Primo, since holding the probe by hand already had influence on the measurement. Why would you want to measure the clock input on the DAC chip? Well, as we have seen, digital data remains intact, even if the analog signal that transports the digital data is contaminated by a dirty power supply, ground loop, wrong cable impedance or other mishap. And as I have described in my video, why digital circuits influence the sound quality that can influence several components inside a DAC if the digital signal isn't clean, especially the digital through analog converter chip. Measuring on the clock input of that chip tells you how the converter can perform. To take it to the extreme, I didn't use an audio file switch but rather the network acoustic mu and probe filter. That has great impact on the sound quality but is fully passive. So no reclocking, just passive filtering. See my review, links at the usual places. Yab uses the Wavecrest SIA 3000 signal integrity analyzer, a prism sound, de-scope, a regal spectrum analyzer and many more. The set was as follows. The Volumio was opened up to give access to the DAC chip. It was powered by a regal laboratory power supply. The network connection was done over a com-power CDN T8 isolator to fully decoupled a network connection from external influences. Then a test probe was held against the clock input of the DAC chip inside the Volumio by a third hand. The probe was connected to a Matthews engineering VET M1 that converts the high impedance output of the probe to 50 ohms as needed by the Wavecrest. That was connected to an input on the Wavecrest. We first measured the frequency spectrum up to 500 MHz, after which the mu and probe was inserted and the measurement was repeated. This gave about the same result. It seemed as if the filter wasn't doing anything. So we measured the spectrum with a regal analyzer that goes up to 1.5 GHz to see if something happened at the lower frequencies. Again, first without a filter and then with the mu and probe in place. Again, no significant difference. So we went back to the Wavecrest to measure jitter and phase noise. This time we did see a difference. Without the mu and probe in place, the average jitter was 14.553 picoseconds and the phase noise minus 32.804 dB in the lower 10 Hz range. Then we inserted the mu and probe again. The measurements were done again and now the average jitter was 11.718 picoseconds and the phase noise minus 37.213 dB in the lower 10 Hz range. So the average jitter was down 2.835 picoseconds and the phase noise 4.409 dB. Especially the phase noise is clearly lower. Don't forget the dB is logarithmic. Please forgive me using the decimal comma on the slide. I already know the mu and probe. I have reviewed it and added it to my reference set of 1A. But Jaap was curious to hear the difference so we went to his listening room where a pair of THD Evolution 2 loudspeakers are driven by a PassLabs XP-12 and X150.8 pre-empower amp connected over Vendor Hull Novart speaker cables. The DA conversion was taken care of by the Sonnet Pasitia and the digital source was the Volumio Primo we used for the measurements. The switch was a D-Link. The difference in sound quality between no mu and probe and the mu and probe in place was immediately clear even though it was the first time I listened to this setup and Jaap's first time to listen to the mu and probe. The sound was more relaxed, voices sounded clearer, the stereo image was more natural and so on. It was in short just more natural. To quote Jaap, it takes only a second to hear the difference or rather one second. Remember bits are bits but the way they arrive at a DAC chip can have a profound effect on the sound quality. When using the right measurement equipment and method it can be made visible in measurements. Doing these measurements is quite elaborate and takes quite some time. The measurements we did for this video took close to two hours. For something you can hear in seconds. But fair is fair, it's not the switches that by using an improved clock oscillator that improves the sound quality. It's the contamination of the analog signal that carries the digital signal as I explained in my video why digital circuits influence the sound quality. Whether you hear the difference depend largely on the quality of your stereo setup. Not only the equipment but also the cabling, loudspeaker placement, acoustics and so on. And of course the quality of your auditory system. As whether this can be measured also depends on the measurements done and the quality of the measurement equipment. And that's rather money intensive. It's not something you can do with an audio precision, at least I can't with mine. But I love to learn. And on that bombshell we come to the end of this video. As usually there will be a new video next Friday at 5pm central european time. If you don't want to miss that, subscribe to my channel or follow me on the social media so you will be informed when new videos are out. I'll be reaching more people by giving this video a thump up or link to this video on the social media. It is much appreciated. Also visit alpha-audio.net, you'll find a lot of interesting publications there. Many thanks to those viewers that support this channel financially. It keeps me independent and lets me improve the channel further. If that makes you feel like supporting my work too, the links are in the comments below this video in youtube. I'm Hans Beekhuyzen, thank you for watching and see you next Friday. And whatever you do, enjoy the music.