Pinging James on media players

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doctorcilantro

Re: Pinging James on media players
« Reply #80 on: 15 Jul 2009, 03:23 am »
I'm going to stick with the internal clock for now on the 1616M. I looked into a lot of other cards both PCI usb/firewire. I've owned quite a few cards and seems like RME and EMU get talked up most (of the prosumer cards) by recording engineers.

While I'd like to try an external clock, and may in the future, I am going to have the 1616M which runs on it's own AC/DC switching power outside the PC, modded. I have read a few threads online and Lavry comes to mind in particular is someone suggesting internal clock is usually the best route, so I'd like to first have some BNC connectors replace the RCA jacks so I can try a clock down the road if I choose. I use the ADC in the 1616M so I want the breakout box looked at for possible mods to help benefit ADC.

DC

racerxnet

Re: Pinging James on media players
« Reply #81 on: 15 Jul 2009, 05:53 pm »
 Just thought I would post a picture of the low latency I achieve with Foobar .083 and SRC @ 88.2. No DSP and ASIO setting is at 0. This is on a everyday use PC for audio and DVD playback. Files are stored on a striped array, card is from 3Ware. OS on separate drive. Chunk size is 256 on the raid, as I use to do video editing with a Canopus DVrex at one time. I did not need the speed, but opted to go this route to use multiple drives for more storage space of the audio files.






doctorcilantro

Re: Pinging James on media players
« Reply #82 on: 15 Jul 2009, 06:16 pm »
That's great but I still don't buy the argument about buffering being a negative; loading into memory etc.

The lower the latency, the closer to the "edge" you are of introducing error. Given various systems this is why people are having to adjust their ASIO buffers to avoid pops/clocks. Granted, this isn't as much an issue as it used to be.

Again, we're not talking about pushing our PSUs to max wattage here, audio decoding and playback even 192kHz upsampling doesn't ramp up the CPU. I think it's a red herring and the concept of introducing noise due to unfiltered power and "noisy" components should be addressed first. I just got through discussing RAID 0 for audio playabck on another forum. I have used it in the past and found no matter how seemingly stable, an issue always cropped up eventually. I can serve 1080P from 7200 rpm Sata II drives without issue and the effective bandwidth there is much greater than that of .flac or .ape.

A good test would be to run a 2-thread or 4-thread analyze on 10K .flac files using J. River while running a Prime95 stress test, and playing back audio. ABX this and see if one can hear a difference.
 
DC

doctorcilantro

Re: Pinging James on media players
« Reply #83 on: 15 Jul 2009, 06:31 pm »
Quote
Just trying to lighten up this sometimes tense thread.

I think most people do not thrive on challenge to ones beliefs. It becomes personal for no reason, and the mind becomes closed at that point. Call me skeptical, I'd rather base my knowledge on proven concepts in the field. This forum is a cake walk compared to Hydrogen Audio. Most people not having data to back up assertions are handed their a$$ on a plate. There are many very qualified people who frequent many forums, I just prefer to digest information with a GRAIN of SALT all the time.

It's about enjoying the music.

MAK

Oh man, HA forums are hardcore. I'm still curious what the actual source material was. Given all the comparisions you did Sasha, to end with "I liked the tray better" is a subjective, qualitative assessment, which is fine, but I would to have the other half, the quantitaive results of an ABX to look at too.

I could not access the AES paper btw.

thanks
Jon aka dr. c :green:

racerxnet

Re: Pinging James on media players
« Reply #84 on: 15 Jul 2009, 08:04 pm »
Hi Jon,

I don't have a problem with buffering the data either. So I am not disputing what others may prefer. I am aware of the edge as I have gone over it several times to verify where I have drop outs. Looking at the latency is just another tool to see how well through-put on the buss is without causing drop outs. The 3ware has its own on-board processor, so I am not sure if this is better than software based solutions of latter times sonic wise. I have not had any hic-ups for over a year due to latency settings in the 70ms. MadVr for HD and DVD playback is GPU based, and most others are taking advantage of GPU based decoding to lighten the CPU load. I am sure you are aware of this as well. Other than that I am just trying to share some information on what works for me at the moment. I've done quite a bit of software engineering/programming and reverse engineering with IDA pro. I'm not a professional, but try to look at this from an engineering perspective to produce positive results.

Just my .02

MAK

doctorcilantro

Re: Pinging James on media players
« Reply #85 on: 15 Jul 2009, 08:09 pm »
Sounds good. I started a 1616M thread based on my planned upgrades to it and possability of adding an external clock.

thanks!
Jon

racerxnet

Re: Pinging James on media players
« Reply #86 on: 15 Jul 2009, 08:19 pm »
Jon,

Where are you posting the mods on the 1616M? I am not sure at the moment, but others may have posted this at DIY audio: http://www.diyaudio.com/forums/forumdisplay.php?forumid=9  I'd be interested in reading about the external clock and differences of the sonics.

MAK

doctorcilantro

Re: Pinging James on media players
« Reply #87 on: 15 Jul 2009, 08:22 pm »
I posted here on AC in the Discless Circle.

I talked to Bolder Audio about some mods to PSU and analog stage of 1616M box and we also discussed I2S which I am not going to pursue.

I did post about how the 1616M box interfaces with the PCI card and still trying to figure if an external clock, given that insight, may really make a difference.

http://www.audiocircle.com/index.php?topic=69855.0

best,
Jon

doctorcilantro

Re: Pinging James on media players
« Reply #88 on: 21 Jul 2009, 05:06 pm »

James Tanner

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Sasha

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Re: Pinging James on media players
« Reply #90 on: 21 Jul 2009, 08:30 pm »
Jitter study

http://www.jstage.jst.go.jp/article/ast/26/1/50/_pdf


dc aa

This study is complete nonsense.
250nS is according to them inaudible? What BS?



bob stern

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Jitter Audibility
« Reply #91 on: 21 Jul 2009, 09:50 pm »
First, note that footnote 1 of the Ashihara paper quotes another study that found much lower levels of jitter to be audible.

Second, Meridian's Bob Stuart, in an interview in the current issue of The Absolute Sound, described some hazards of ABX testing that may be relevant to the Ashihara study.  It is especially fascinating that Bob Stuart faults the traditional scientific ABX method because he is a scientist with many published papers.

Bob Stuart said he finds ABX comparisons unreliable because: (1) the brain responds to music in subtle ways that are disturbed by the pressure of an ABX test; (2) long listening sessions are required for the listener to appreciate differences because it takes time for the brain to acclimate to a new listening "environment".  (The Ashihara paper said the subjects controlled when and how often they switched, but the paper did not report the typical time duration chosen by the subjects.  They may have switched too fast knowing they had to undergo several iterations.)

Conversely, Bob Stuart also discussed hazards of long-term comparisons, i.e., the alternative to ABX tests.  In long-term comparisons, successive playings of the same musical passage are not equivalent because your brain interprets each repetition based on its memory of the previous repetitions.  You typically hear things the second time you didn't notice the first time.  Importantly, he says that if you hear a low resolution component, then a higher resolution component, then return to the low res component, the experience of hearing details on the high res component will enable you to perceive those details on the low res component that you missed the first time.  This can lead you to erroneously conclude that the low res component is just as good.

Bob Stuart also said that after listening to a passage a certain number times (he didn't suggest how many), it becomes useless and you need to move on to other music.

James Tanner

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Re: Pinging James on media players
« Reply #92 on: 21 Jul 2009, 10:46 pm »
Good stuff!-- thanks guys for all the input here.

I know about A/B/X testing first hand as I was involved in many tests at the NRC's Canadian facility in Ottawa run by Floyd Toole back in the days when he was developing his theories on what makes a good speaker and what most people perceive as quality sound - but hey that's another story.

Anyway, what I have found over the many years that I have been doing this is that my first response after I have been away from the system for a few hours or days is usually the correct one.  For instance when I do an audio show setup I will spend coutless hours the first day of setup and fine tune every little detail including speaker position, component choices etc. Many times it's 2 in the morning when I finally say "OK" got it!.

Then the next day after breakfast I walk into the demo room - fire up a song ...and in the first 30 seconds I know!---a pat on the back is appropriate or time to start all over!

james
 
« Last Edit: 22 Jul 2009, 12:42 pm by James Tanner »

doctorcilantro

Re: Pinging James on media players
« Reply #93 on: 23 Jul 2009, 06:00 pm »
Okay, dumb question, but I just had this thought pop in my head.

Can't we monitor the output at speaker with some method or device modeled on the human ear? We keep talking about measuring jitter "inside" the system, but shouldn't we able to somehow get a piece of technology to measure the effects as the come out of the speaker?

Could the tool be designed to look at a "microscopic" level (e.g. frequency response graphing) and also on a macroscopic level; e.g. "looking for obvious shifts/anomalies in the midrange). Random jitter may be hard to detect, but I can't help thinking if we can hear it with our ears, we should be able to see it on paper.

Here's the psu in my new PC.

DC