 I have many reactions and questions on my improve the sound quality of your DAC for free video. That's understandable, since where it works it can improve the sound quality clearly. Where it doesn't work, it raises a number of questions. Let's see if I can clarify some things. Before I begin, let me first explain that very long questions risk not being answered simply since I lack the time. Which one is better questions risk the same? I try to explain how good reviewed equipment is, if that doesn't suffice I'm afraid I can't help you further. And of course, rude questions or remarks, URLs, promotion of your equipment, irrelevant information and silly drunk 1am in the morning jokes won't even make it to the channel. As far as YouTube doesn't already block them, I certainly will. Let's keep the channel clean and the information density sufficiently high. Now I got that out of my chest, let's start with the job at hand. Let's start with a remark that was made several times. When you change the volume prior to the DAC, the signal is no longer bit perfect. This is true, but not a problem. I'll explain. Let's first see where the bit perfect problem had its origin. In the beginning of file based audio, computers were used as a digital source. Without taking special measures, the computer's operating system will convert all audio to one simple sampling frequency so that the music and the system beeps can be played together. Often this sampling frequency is 48 kW, the sampling frequency least used for music. Bit conversion can be done extremely good nowadays but this is not the case with operating systems. Thus the sound quality a computer sent out is lower. The answer is using BitPerfect music player software. They bypass the operating system sometimes by needing a special driver. This was called BitPerfect for obvious reasons. Today Linux and macOS support USB audio class 2 for years now and Windows does it since one or two years. Spitiff and Toslin connections are handled well by most BitPerfect music player software but these are not standard on most computers and if they are, in general, they offer lower sound quality. See my video choosing your digital connection. Many software players offer upsampling and in many cases that really improves the sound quality despite the fact that the information in the upsampled file at best is equal to the original. So there are no extra high frequencies nor is there more resolution. The improved sound quality has another reason. This has to do with the reconstruction filter used in the DAC. This filter has an easier job with the higher sampling frequency. You find a full explanation in my video the truth about Nyquist and why 192 kHz does make sense. These filters in two high end DACs, as in expensive, are of very high quality and their upsampling often has a very limited to no effect on the sound quality. As in DACs normal people can afford, these filters do have influence on the sound quality. The lower the sampling rate, the more these filters reduce sound quality. In that case, upsampling to the highest sampling frequency will make the job for the reconstruction filter in the DAC a lot easier, leading to a better sound. But only when the upsampling is done with high precision. The advantage of doing it in a computer or network player is that these have far more computational power than DAC chips have. Therefore the upsampling can be done better and thus can lead to a better sound quality. Since upsampling needs two, four or eight times as much bits, it is not bit perfect and yet might sound better. Don't be mistaken, there are upsamplers and upsamplers. In fact most DAC chips used in digital to analog converters also have upsamplers in them, but since the computational power of these chips is very low, corners have to be cut when writing the upsampling code, which usually leads to sound quality loss. Especially time resolution has to be sacrificed, leading to severe time smearing. Computers have far higher computational power and given good software based on high quality algorithms can yield a far better sound quality. Despite the bits are altered and thus the process is not bit perfect. As you might have guessed, right now real bit perfect isn't the holy grail. Altering the bits in the music files might be beneficial providing the altering is done using very good algorithms. The same goes for volume control. In the beginning volume control used 24 bits resolution or worse while rounding errors also had their impact. Nowadays up to 64 bits floating point volume control is used. See the specs in Rune for instance and it is no different in other quality software like ORIVANA and J-River, just to name the two I also have. That gives the digital volume control a very high precision with little or no impact on the sound quality or, better, reduce sound quality less than many analog volume control do. So reducing the level of a digital audio file by 3 dBs while using good software will have little to no impact on the sound quality, the overshooting of the reconstruction filter in the DAC chip wheel and that is queued by the minus 3 dB attenuation. You might want to watch improve the sound quality of your DAC for free again if you are lost a bit. Any alteration of MQA files will disable the MQA decoding. The DAC wheel referred to the normal 44.1 kHz 16 bit quality. So don't use the minus 3 dB trick when playing MQA files. By the way it is my experience that the problem the minus 3 dB trick solves doesn't occur when playing MQA files. Whether that is the case due to better mastering or better reconstruction filter, I don't know. There are DACs that cleverly have an equivalent of the minus 3 dB trick incorporated in their filtering. If you step through the comments below this video improve the sound quality of your DAC for free, you'll see brands like Benchmark and PS Audio mentioned. I have not checked them but they are made by brands that understand digital quite well. It's not so hard to incorporate an attenuation prior to or in the upsampling algorithm and compensate for the level change in the analog domain. The designer just has to understand digital well. If you use music player software that can send bit perfect music over the network to a network bridge or renderer, reducing the output level with 3 dBs will indeed do the same. Think of apps like Spotify, Tidal, Koboose and Amazon Music. But also apps that run on your computer like Audivana and J-Briver. You do need to set the playback level then on your amp of course. There are some remarks stating that a further level reduction sounds better. That can be the case but then it's not due to the phenomenon I discussed here. It might occur when the output voltage is clearly higher than the amp can handle. The defect of standard for digital consumer players is the so called Redbook spec. 2 volts RMS at 0 dBFS, the maximum output level. This is for single ended outputs, usually on RCA connectors. Balanced outputs, usually on XLR connectors, offer double dead voltage, so 4 volts. But there are also DACs that have a 6 dB higher output level, 4 volts single ended and 8 volts balanced. I have come across even higher levels. If your amp is designed for 2 volt sources and a source with 4 volts is connected, peaks might get clipped as well. There has been a remark by someone saying that minus 2 dB might be a better choice. To be honest, the 3 dB attenuation is a general rule of thumb. It depends on the hardware and the music what works best. If you play only heavily modulated dance music, minus 4 dB might even be a better choice. And if you find the result of a 2 dB attenuation better, that might be the case in your situation. But whether you use 2, 3 or 4 dB attenuation, the impact on the sound quality, apart from correcting the overshooting filter effect, will be negligible. Can you do the same trick with a network player with an integrated DAC? It depends, since the digital source, the streamer and the DAC are integrated. You can't interfere unless the volume control is done in the digital domain, so in front of the digital to analog converter. If you use the streamer with an external DAC and the volume control on the streamer influences the volume, you can. If the volume control is in the digital domain and you use that to set your listening level, you automatically have solved the problem since it's hardly unlikely you listen to the volume control fully opened. The effect of overshooting reconstruction filters might be subtle or non-existing in your case. Perhaps because it's masked by your playback equipment or is already solved by the manufacturer, as we have seen before. It is a simple problem if you understand it or, like the famous soccer player said, you'll get it when you understand it. Which seems to be a good line to end the program with. As usual, there will be a new video next Friday at 5 pm central european time. If you don't want to miss that, subscribe to this channel or follow me on the social media so you will be informed when new videos are out. Help me reach even more people by giving this video a thumb up or link to the video on the social media, it is much appreciated. 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 on YouTube. I am Hans Beekhuyzen, thank you for watching and see you in the next show or on the hbproject.com. And whatever you do, enjoy the music.