You have too much network bandwidth!!

@tomrk Be sure to consult these HQP discussion groups for help and guidance:

HQ Player - Head-Fi

HQ Player - Audiophile Style

EDIT: Yes, HQP can convert DSD to PCM, although I've not have a reason to do it, so I cannot comment.

OP

my 80M/Bit is more than enough. so I completely agree with you

There’s 4K TV and then there’s 4K streamed TV.

All digital TV is compressed using lossy algorithms historically devised by the Motion Picture Experts Group also known as MPEG.  MPEG sought compression techniques from the industry, and combined the best to produce compression standards, the best known being MPEG 3 and MPEG 4.  Within each standard, the producer of an mpeg file or stream can choose the degree of compression, and hence loss of quality.

I asked ChatGPT:  here are some vital statistics which I have re-written ;-).

Blu-ray 4K typically outputs between 92 and 144 Mbps.  Audio is lossless.

Netflix 4K is highly compressed down to 16 Mbps with many visible artifacts.  Audio is lossily compressed, even Dolby Atmos.

For further comparison, a Super Audio Compact Disk SACD 5.1 recording outputs DSD at about 2.8 x 6 = 16.8 Mbps.  This is more than Netflix uses for both sound and video in order to get the Netflix streaming bit rate manageable!

But wait, there’s more.

A Pure Audio Blu-ray with 9 tracks of 192-kHz 24-bit PCM outputs about 43.2 Mbps

Finally, you can convert DSD to PCM but there is a loss of detailed timing information.  PCM can be converted to DSD losslessly

@tomrk 

That is objectively true, but on a typical modern CPU or streaming device, the time taken to decode a level 8 FLAC file (highest level) can be considered to be negligible and often falls within milliseconds

I agree, but I don't think your comment has any bearing on the topic of bit-rates!

All digital content is buffered in multiple places during consumption.  If streaming, there is a trade-off between latency and accuracy.  Voice over Internet Protocol (VoIP) is a use-case where latency should be minimised if you want to have a conversation in real-time, whereas high resolution audio should favour accuracy and if it takes a few seconds to start, who cares?.

@richardbrand 

The FLAC discussion was base on a post saying PCM doesn't have to be compressed and implied it was better than compressed files for music.  I was pointing out the CPU even at the maximum compress of FLAC is so small as to not be a real consideration.

Buffering is a completely different topic and has been a discussion about whether it's good or bad.   The providers have already made this decision for us, so in my view it's a non-factor as to whether small bits of data like a music file is being harmed by latency and error correction in the IP protocol.   The short answer is "it's not".  So I agree with you on this point.  However, there is train of thought in the audiophile community that think like error correction is bad, so you must buy audiophile switches and ensure no magnetism gets close to your gear, and for the most part audiophile switches are  just a chinese house brand switches with a different paint job.

VOIP as you mention is sensitive to latency, but particularly jitter but that's a different issue entirely.  Downloading a FLAC file even 1M (which is extremely large for a flac file) will take approximately 1 second to download.

Video is an interesting topic;  I think many technical people believe it uses UDP, but in fact most video streaming is done using TCP because it gets priority on the net and it doesn't have to switch ports when it reaches the destination.  There are other protocols for video but in practice those have not been used by the majority of video providers like Apple or Netflix and have fallen by the wayside.

Apologies for the long response.