MQN

[sbgk]

my view is that I can only hear up to 16KHz, so any perceived improvement due to hirez is due to the increased no of samples. I like 24/192, but the data volumes are quite large, 24/96 seems a good compromise, 16/44 sounds quite rough in comparison.

I'm sure there are contesting views about the ability of the human ear/brain to hear the difference between the varying sample rates, but there you go.

[Aleg]

my view is that I can only hear up to 16KHz, so any perceived improvement due to hirez is due to the increased no of samples. I like 24/192, but the data volumes are quite large, 24/96 seems a good compromise, 16/44 sounds quite rough in comparison.

I'm sure there are contesting views about the ability of the human ear/brain to hear the difference between the varying sample rates, but there you go.

Quite so:
http://boson.physics.sc.edu/~kunchur/pa ... Foster.pdf

"Many misconceptions and mysteries surround the perception and reproduction of musical sounds. Specifications such as frequency response and certain common distortions provide an inadequate indication of the sound quality, whereas accuracy in the time domain is known to significantly influence audio transparency. [...] Our recent behavioral studies on human subjects proved that humans can discern timing alterations on a 5 microsecond time scale, indicating that the digital sampling rates used in consumer audio are insufficient for fully preserving transparency."

this article states that people can discern sounds that are only about 6 microseconds apart. According to the article the consequence of this capability is that sample frequency should be above 166 kHz to maintain the transparency of the original sound and avoid sound smearing which was not in the original sound.
So it has to do with the timing aspect of the signal and not with tone frequency represented by that number.

The test was done to generate a 7 kHz sound pulse and listen to that through two speakers. Then they moved one of the speakers by milimeters at a time and check when the person (doing the test blinded) could differentiate between the sound coming from each speaker.

A few other quotes:

Audio Quality in Networked Systems / Sampling Issues / Temporal resolution (http://www.yamahaproaudio.com/global/en ... o_quality/)
"In the professional live audio field, a 48 kHz sampling rate is adopted as standard, with some devices supporting multiples of this rate: 96khz and 192kHz. (Some devices also support 44.1 khz and 88.2 kHz for compatibility with the music recording field, eg. the Compact Disk). However, apart from the temporal resolution of a digital part of an audio system, the temporal characteristics of the electro-acoustic components of a system also have to be considered. In general, only very high quality speaker systems specially designed for use in a music studio are capable of reproducing temporal resolutions down to 6 microseconds assumed that the listener is situated on-axis of the loudspeakers (the sweet spot). For the average high quality studio speaker systems, a temporal resolution of 10 microseconds might be the maximum possible. Live sound reinforcement speaker systems in general can not support such high temporal resolutions for several reasons. "

http://recordinghacks.com/articles/the- ... ond-20khz/
"Many engineers have been trained to believe that human hearing receives no meaningful input from frequency components above 20kHz. I have read many irate letters from such engineers insisting that information above 20kHz is clearly useless ..."

"Human hearing is generally, I believe, misunderstood to be primarily a frequency analysis system. The prevalent model of human hearing presumes that auditory perception is based on the brain’s interpretation of the outputs of a frequency analysis system ..."

"The human hearing system uses waveform as well as frequency to analyze signals. It is important to maintain accurate waveform up to the highest frequency region with accurate reproduction of details down to 5µs to 10µs."

http://www.audiostream.com/content/qa-b ... recordings
"When I first heard properly done 24/192, it was a jaw dropper. For the first time in my experience, those reservations I have always had about digital, where I felt there were some things the best analog did better, simply evaporated. This is, to my ears, a bigger jump up in quality over 24/96 than that was over 16/44. It no longer feels like a great digital recorder or a great analog recorder. It feels like the recorder has been effectively removed from the equation and I am listening directly to the mic feed."

[sbgk]

uploaded mqnplay 9.11 avx2, found some more things I could leave out

set up for thysecon driver device so should be long enough for everyone.

\\?\tusbaudio_enum#vid_20b1&pid_000a&ks#6&1e31c658&0&1#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\pcm_out_02_00

think fidelizer might be increasing the priority of audiosrv and audioendpoint, tried increasing audiosrv and there does seem to be a difference which some may care for, I'm happy with the default priority. I don't use fidelizer, was just investigating.

the plan is to get it working for all devices, a couple of things to try first though.

Great I look forward to an easy implementation. ;)
From the feedback it seems the latest MQn's are very good indeed, so I will wait patiently and check in now and again.

Personally, I think the time spent on DSD would be better invested elsewhere.
Only a minority of music lovers would be interested in such a restrictive format.
For me and all my audiophile mates, 44.1khz -96khz is king...

The ear can't discern above 88khz resolution anyway... so DSD seems a rather futile pursuit.

you would get dsd64 dop with 24/176 format, so it's not extra work, well maybe a little. Several professional musician/recording types seem to favour 24/192. Some people rave about upsampling everything to dsd512, I've never found upsampling to be that satisfactory in pcm world at least.

[sbgk]

Some good info there Aleg, maybe people should be doing a loudspeaker test to see if their speakers are resolving enough for hirez.

Will you get a chance to try the new version ? I think it's an improvement (it allows optimised loading of data into the play ram), am going to use that method to create a load.exe so that control hands off the loaded file into ram and the load.exe loads the play ram from the load ram, this way the load.exe can be as optimised as the play.exe, also gives me an easy way to pass the device name from control. Means one more exe to maintain, but hopefully it will be worth it.

[Ken Moreland]

After a struggle I got 4.24/9.11 working using Sebna's Hexidecimal editor.
(For those using Chord Hugo the string was \\?\usb#vid_245f&pid_1213#413-001#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\wave leaving out the last 0 as reported earlier.)
This worked fine and sound is excellent so I progressed to 4.25/9.12 and got sound but only for about 10 or 20 seconds, not sure if others are reporting the same.

KM

[sbgk]

After a struggle I got 4.24/9.11 working using Sebna's Hexidecimal editor.
(For those using Chord Hugo the string was \\?\usb#vid_245f&pid_1213#413-001#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\wave leaving out the last 0 as reported earlier.)
This worked fine and sound is excellent so I progressed to 4.25/9.12 and got sound but only for about 10 or 20 seconds, not sure if others are reporting the same.

KM

ok, I'll check that out, I was just pleased to get it working, sure I played it for more than 20s though it was late.

[Ken Moreland]

Just a small observation 4.25/9.12 starts to play very quickly even before the countdown has finished.

[Aleg]

After a struggle I got 4.24/9.11 working using Sebna's Hexidecimal editor.
(For those using Chord Hugo the string was \\?\usb#vid_245f&pid_1213#413-001#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\wave leaving out the last 0 as reported earlier.)
This worked fine and sound is excellent so I progressed to 4.25/9.12 and got sound but only for about 10 or 20 seconds, not sure if others are reporting the same.

KM

ok, I'll check that out, I was just pleased to get it working, sure I played it for more than 20s though it was late.

No problem listening to full albums here on the main system with the SonicWeld DiverterHR2.

Just created a second version for the office system using a Mutec MC-1.2, and no problem either.

Cheers

Aleg

[Octagon]

trying a new way of loading the data
uploaded control 4.25/ play 9.12, seems better clarity. only work with each other.

Sorry, but had a lot business on my plate the last days, couldn't test for a while and still pleased with 9.00 avx2 oct.

Tried to convert 9.12 but my FF UCX name is 4 positions longer. Gordon, is there a chance to use a longer dummy or do another individual version for me?

Many thank's
Thomas

[Aleg]

Just a small observation 4.25/9.12 starts to play very quickly even before the countdown has finished.

Yes, I noticed that too.
Rather like it that way.
You can lower the time out value from 5 sec to 2 sec now inside the .bat file that starts playback. The time out is used to allow mqnplay to load before adjusting the affinity and priority level.