INTERACT FORUM
More => Old Versions => Media Center 15 (Development Ended) => Topic started by: JazzDoc on March 03, 2011, 05:39:23 pm
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I thought I would share an interesting post by blaine78 on the Computer Audiophile board regarding settings for MC :
"Hi,
I'll cut to the chase. I'm a computer audiophile for about 7 years now. Been tweaking hardware and software during this time to get the maximum fidelity from a computer to a DAC. So thought I'd share my findings with you. Here is, to me, the best setup of J River to get the best sound from it. I've posted this on another forum, so most is a copy and paste job...
My recommended findings on USB computer audio have been tested with Ayre Qb-9, Lavry DA11, Weiss Minerva.
J River Media centre, definitely THE best sounding player out there and I've used many.
In the audio configuration of J River, I tried the new WASAPI 'event style' built into J River, I didn't like it! It made the sound just a tad tight and congested and music was stripped of emotion. I went back to original J river's WASAPI mode and wont be changing.
This next tweak to me is THE most important and never really spoken about on forums I've noticed - Increase the WASAPI audio buffer in J River to 1.99 seconds!!. This works for my current Ayre DAC, some other DAC's may have different maximum buffer sizes. The Idea is to increase the buffer size as much as possible without the audio to stop working. This will make the sound much more open, airy, emotional and less harsh. Also very important in the audio options change the pre-buffering from the default 6 seconds to 20 seconds. Make sure Playback from memory is also enabled. In my set up, this makes a world of difference. Please don't ask why or dismiss this. it works and sounds the best to me every time. Also, Wireworld USB cable works amazingly compared to generic and other higher end cables. USB cables make a very big difference.
Also, another one to make a difference on sound for the better is using AIFF or WAV files and not FLAC ALAC or other compressed lossless. WAV and AIFF have more space around instruments and energy than ALAC or FLAC. FLAC and ALAC seem to be just a little flat sounding and lifeless in comparison. It's not immediate night and day, but notice after 30 seconds or so of listening. I know not logical, and know of the debates, and I would much prefer saving the disk space. But my ears and Ayre and ATC equipment don't lie (well i hope not). The only theories i can think that this may be is, the extra CPU processing on uncompressing Lossless in real-time adds noise downstream or affects timing on data.
Please give it a try and let me know what you find."
I tried out blaine78's recommendations, namely:
1. Revert to WASAPI rather than WASAPI Event Style
2. Increase the WASAPI audio buffer in J River to 1.99 seconds (I had to reduce it to 1.7 to prevent stutter on playback)
3. Change the pre-buffering from the default 6 seconds to 20 seconds
4. I also had to tick the 'flush device buffers on startup' in the WASAPI audio options (This was necessary to prevent stuttering)
I think that blaine78 may be on to something. The sound is certainly different. I need to do more listening but it would appear that the sound is more musical and open. This was particularly noticeable when playing a 24-Bit/88.2kHz version of the album 'Home' by Elvis Costello.
Any comments or experience?
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Please don't ask why or dismiss this.
This seems to be a requirement of being a good audiophile these days. ::)
I think MC would be better off without such endorsements.
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This seems to be a requirement of being a good audiophile these days. ::)
I think MC would be better off without such endorsements.
Yes and no. I tried all of the settings. None of them changed the sound though I had the slight impression that increasing the prebuffer size did something. But I can´t be sure without proper tests. Which means I could be listening to a placebo. So one thing is sure: this blaine78 is clearly listening to a placebo.
BUT: if one compares JRiver for example to foobar2000, it sounds decidedly different with similar settings (without any DSP of course). You won´t even need a DBT for it, the difference is almost frightingly clear. In theory it shouldn´t sound different. Or should it? I never was a believer that a software player changes the sound but obviously it does. Don´t know why. I even made some measurments only to find out that the output of both players technically is perfect (aside from some curious noise patterns around several testtones with foobar which weren´t present when played back with JRiver).
Sure, all of this is not mystical. It is not audiophile either, it´s just a question of technology since we play back our music on a closed, controlled system.
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Hi,
Sorry some don't have the same result as I'm getting, i guess it's system dependant. regards the endorsement response, I'm clearly just stating my findings and being honest about it, as i don't know why this would sound better than event style with 6 second pre buffer, although i have my theories, and just don't dismiss it as being rubbish. I have an extremely transparent ATC system, i know it very very well. Event style i lose interest in the sound, it sounds too dry and uninvolved on my system. the trick i found in my posting above works really well for me. i've tried numerous setting and configs, this one work best for me. and definitely NOT a placebo. I guess that's why i have this hobby, i take it seriously and look for the best results.
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Hi,
Sorry some don't have the same result as I'm getting, i guess it's system dependant. regards the endorsement response, I'm clearly just stating my findings and being honest about it, as i don't know why this would sound better than event style with 6 second pre buffer, although i have my theories, and just don't dismiss it as being rubbish. I have an extremely transparent ATC system, i know it very very well. Event style i lose interest in the sound, it sounds too dry and uninvolved on my system. the trick i found in my posting above works really well for me. i've tried numerous setting and configs, this one work best for me. and definitely NOT a placebo. I guess that's why i have this hobby, i take it seriously and look for the best results.
My dear, what you are describing sounds an awful lot like a perfect placebo. You could very well want to hear those differences and therefore you hear them. Furthermore, sentences like "I have an extremely transparent ATC system..." sorry, but that´s just arrogant rubbish. For all that matters I could say the same, because I´m not even using a loudspeaker system (which is a very insecure way of testing audio auditorally), I use a Sennheiser HD-600 instead, connected to an ASUS Xonar Essence ST. As you may well remember from a Stereophile review, this soundcard comes quite close in sound as a source to your Ayre. The combination of my little system points the sound directly into my ears - without room involvement.
But I´ll never talk about superiority of any system because in my experience, so called high-end-audio is nothing more than a smokescreen where the ultimate goal is not perfect playback but sonically spiced up playback. I´ve listened to a lot of high end stuff over the years and found that all of them more or less coloured the sound. Even my Sennheiser does this. It does this very good but it still does. So you could very well be listening to a placebo. Furthermore, Charles Hansen of Ayre doesn´t know half the time what he´s talking about. He mixes some technical knowledge with myths - a very dangerous combination IMO. I wouldn´t trust one piece of hardware coming from his company. Well, I could be going on and on forever about this but I´ll make it short.
If you are happy with your settings go use them. But please don´t bother other people with them. I´m ashamed of myself that I even tried to recreate them when I read sentences such as yours.
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Wow. All I can say is wow.
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wowie zowie?
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wowie zowie?
;D
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Hunter, is that you?
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When my music is too dry I crank up the reverb.
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My head just exploded.
These setting just blew my freaking mind.
I took it one step further.
Where does good audio start? At the source.
I re downloaded some tracks, but this time i used my $300 internet cable.
Awesome, just awesome.
I just ordered some nipple clamps that im going to modify into ground wires
for my body so that static electricity doesn't interfere with my listening experience.
I ordered enough that i can replace them every 8 hours of listening because new wires
make a huge difference. Most people don't know this but a good nipple clamp body ground
needs constant maintenance.
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Wrong crowd JazzDoc. thanks all the same and glad it worked for you.
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My dear, what you are describing sounds an awful lot like a perfect placebo.
how do you know that, are you in my house using my Hi-Fi? You should have called i would have made some h'orderves. Not just me who likes the sound, friend of mine hears it, and few others that have tried it and liked it too.
But I´ll never talk about superiority of any system because in my experience, so called high-end-audio is nothing more than a smokescreen where the ultimate goal is not perfect playback but sonically spiced up playback.
ATC are renowned for their Studio Monitors and Amplifiers as they are accurate and very uncoloured, so prefect for mastering, which is my goal profession.
Furthermore, Charles Hansen of Ayre doesn´t know half the time what he´s talking about. He mixes some technical knowledge with myths - a very dangerous combination IMO.
Thats a pretty bold statemnet, do you make hi end audio equipment?
If you are happy with your settings go use them. But please don´t bother other people with them. I´m ashamed of myself that I even tried to recreate them when I read sentences such as yours.
sorry your ashamed of yourself. if doesn't work for you, move on. Maybe it's the $200 soundcard inside your computer with the coloured sounding headphones that you mentioned?
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My head just exploded.
These setting just blew my freaking mind.
Really glad it worked for you!
I took it one step further.
Where does good audio start? At the source.
I re downloaded some tracks, but this time i used my $300 internet cable.
Awesome, just awesome.
Don't believe a $300 internet cable will do much, as the timing (analogue domain) of the 1 and 0's aren't important in this event. Also, I'm not a big believer in expensive cabling, but the wire world USB cable for the computer to DAC did tick all the boxes and made a nice improvement. $300 for internet cable? You do take this seriously! WOW
I just ordered some nipple clamps that im going to modify into ground wires
for my body so that static electricity doesn't interfere with my listening experience.
I ordered enough that i can replace them every 8 hours of listening because new wires
make a huge difference. Most people don't know this but a good nipple clamp body ground
needs constant maintenance.
is worth a try if it gets you closer to the music.
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I know that most of this is probably a placebo effect. At least such things as buffer. I personally think most of such things is utter nonsense, but I do activate some of the alternatives either way. It can't hurt. It's the same thing with drugs. If people thinks it will help, and it works, then it's a good thing. Placebo or not. No matter what the doctors say, it helps you.
The biggest problem here though is that people have way to little acceptance for each others opinions, and you all attack each other as a bunch of preteen girls on the comment space of your blogs.
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I know that most of this is probably a placebo effect. At least such things as buffer. I personally think most of such things is utter nonsense, but I do activate some of the alternatives either way. It can't hurt. It's the same thing with drugs. If people thinks it will help, and it works, then it's a good thing. Placebo or not. No matter what the doctors say, it helps you.
The biggest problem here though is that people have way to little acceptance for each others opinions, and you all attack each other as a bunch of preteen girls on the comment space of your blogs.
Very true. all in all, if it works for you great. I think if someone has given the generosity of their experiences, it is decent for others to respect that. as i've mentioned numerous times , it maybe be system dependant and so others mile will vary. ultimately this is a share and help site.
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Blaine78,
Have you or anyone else tried Kernel Streaming mode?
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Yep tried Kernel streaming, found it a little light on the bass and too airy, so not a solid foundation to the music as WASAPI (original), this is on my system and DAC. ASIO i found to sound a bit bleached and too clean sounding 'processed' and robbed the emotion from music, a little like WASAPI event style, which is funny as found out later ASIO and WASAPI Event Style use the same buffer processes.
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ultimately this is a share and help site.
That's exactly what it is. And that's why you're original comments resulted in a backlash. You didn't share your experience of selecting certain settings and your perception of better results. You stated doing what you did would result in better sound and that we should not question you. If you wanted to be helpful, you would provide facts, preferably backed with evidence. That provides others with something they can try to reproduce. It can also include your subjective assessment doing something seems to have a particular result. In that case, the "fact" is weak one, but at least you're not misleading anyone.
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That's exactly what it is. And that's why you're original comments resulted in a backlash. You didn't share your experience of selecting certain settings and your perception of better results. You stated doing what you did would result in better sound and that we should not question you. If you wanted to be helpful, you would provide facts, preferably backed with evidence. That provides others with something they can try to reproduce. It can also include your subjective assessment doing something seems to have a particular result. In that case, the "fact" is weak one, but at least you're not misleading anyone.
Hi,
Hard to back fact with evidence when you're not listening to it on my system. i did state at the beginning 'My recommended findings on USB computer audio'. i did not proclaim 'do it, or die' and did not ever say once 'fact.' Again, this is trail and error. I'm not trying out for a college exam, I'm just giving my thoughts on my findings, and to me, work absolute. Nobody knows everything, but everyone wants to be heard.
I have quite a number of sound improvements that i've found to work well that are software, bios, and hardware based. all these tweaks add up to an immense improvement 'on my system'. this is just a couple of findings. I love music, and i love electronics and all the voodoo (spirit) that encompasses it. i find it quite like a living being. if others don't get it, switch off.
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JazzDoc, Blaine,
Don't give up on these guys. It's good for all of us to be exposed to the beliefs of others, even if we don't agree.
Thanks for posting this. We really appreciate the support audiophiles have given us on sites like ComputerAudiophile.com.
Jim
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Thanks Jim.
Appreciate it. it's just one of those things that is hard to describe to others unless experiencing it with you. Thanks to you guys for the effort to make this software great! Computer Audio is where it's at, and glad there is a software developer for PC that take it seriously too.
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how do you know that, are you in my house using my Hi-Fi? You should have called i would have made some h'orderves. Not just me who likes the sound, friend of mine hears it, and few others that have tried it and liked it too.
ATC are renowned for their Studio Monitors and Amplifiers as they are accurate and very uncoloured, so prefect for mastering, which is my goal profession.
Thats a pretty bold statemnet, do you make hi end audio equipment?
sorry your ashamed of yourself. if doesn't work for you, move on. Maybe it's the $200 soundcard inside your computer with the coloured sounding headphones that you mentioned?
Are you proposing a date? Why, thank you! I´ll bring the wine to the h'orderves. I live in Germany, where are you?
And no, I do not make high end audio equipment. Still, regarding the quality of Ayre´s products: at least some of his products are fairly standard technology inside a fancy aluminum case. Take the http://www.ayre.com/cx7e.htm (http://www.ayre.com/cx7e.htm) for example, its parts consists of a very cheap (and old) drive from Sanyo, not even the D/A converter is anything better than those you´d find in 200,- Euro players. Gosh, even my "colouring" ASUS soundcard has more expensive D/A converters. Despite these cheap parts this player costs 3600,- Euro here in Germany. I think it´s a bit much for an - albeit nice looking - aluminum case and a bit of fancy DSP programming.
Furthermore, his recent BluRay player has been revealed to be a simple OEM model with a revamped energy supply and output stage. But that´s about it. While the OEM players costs around 500,- Euros Ayre wants 10.000,- Euros. I won´t say anything more.
Yes, my headphone colours. I made that pretty clear from the beginning. I also suspect my soundcard to be colouring but I don´t know to which extent. I just wanted to say that you can´t be sure if your equipment colours.
Hard to back fact with evidence when you're not listening to it on my system. i did state at the beginning 'My recommended findings on USB computer audio'. i did not proclaim 'do it, or die' and did not ever say once 'fact.'
Well, we cannot listen to your system, can we? And as it happens, I too own a USB solution. Well, it´s "only" a E-MU 0202 USB which was by the way the first USB audio interface to use the asynchronous protocol (yes, Ayre wasn´t the first as it is stated so often) but I couldn´t find anything of the things you proposed. But then... maybe it´s not that revealing. Furthermore, your Ayre interface could very well have some issues with its USB driver. For all I know you could be listening to an error in the hardware chain when you change the output configuration. I did not say that you do but it could be.
JazzDoc, Blaine,
Don't give up on these guys. It's good for all of us to be exposed to the beliefs of others, even if we don't agree.
Thanks for posting this. We really appreciate the support audiophiles have given us on sites like ComputerAudiophile.com.
Yes, exactly my mind. You know Jim, if Blaine wouldn´t have been so absolute about his findings I would be completely on his side. He might think that I´m one of those proof loving guys from Hydrogenaudio - but I´m not. Just like him, I love music very much and I love the best possible playback solution for it. I also love to tweak my system. I own fancy RCA cables, power cables etc. but I will never say that my solution is the only one that makes someone happy and I certainly will not elevate them over everything else. Which exactly is what he did IMO.
I think that we are here to discuss, while not being absolute. Blaine, if I insulted you I´m truly very sorry. I mean that.
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That's exactly what it is. And that's why you're original comments resulted in a backlash. You didn't share your experience of selecting certain settings and your perception of better results. You stated doing what you did would result in better sound and that we should not question you. If you wanted to be helpful, you would provide facts, preferably backed with evidence. That provides others with something they can try to reproduce. It can also include your subjective assessment doing something seems to have a particular result. In that case, the "fact" is weak one, but at least you're not misleading anyone.
I think it's important to try different things out on our own systems. There are many factors not under the control of normal users, so most people just have to try things out. For example, the capability to diffifferentiate a more accurate sounding playback polarity is one of my basic criteria for audio. Not everyone will experience this. I chose the Kernel Streaming mode just because it seemed to be the most direct communication with the soundcard, and it sounds fantastic to me. Unless I know a technical reason to change this, it will probably be my default selection so that I can explore what other technical improvements can be made in a system.
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JazzDoc, Blaine,
Don't give up on these guys. It's good for all of us to be exposed to the beliefs of others, even if we don't agree.
Thanks for posting this. We really appreciate the support audiophiles have given us on sites like ComputerAudiophile.com.
Well said, Jim. Well said. I agree completely.
Hunter, is that you?
Can't stop here, this is bat country.
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Yes, exactly my mind. You know Jim, if Blaine wouldn´t have been so absolute about his findings I would be completely on his side.
Never mistake absolute fact with passion, only absolute passion :)
I think that we are here to discuss, while not being absolute. Blaine, if I insulted you I´m truly very sorry. I mean that.
Not insulted, but thanks. Made the ride more enjoyable ;)
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And no, I do not make high end audio equipment. Still, regarding the quality of Ayre´s products: at least some of his products are fairly standard technology inside a fancy aluminum case. Take the http://www.ayre.com/cx7e.htm (http://www.ayre.com/cx7e.htm) for example, its parts consists of a very cheap (and old) drive from Sanyo, not even the D/A converter is anything better than those you´d find in 200,- Euro players. Gosh, even my "colouring" ASUS soundcard has more expensive D/A converters. Despite these cheap parts this player costs 3600,- Euro here in Germany. I think it´s a bit much for an - albeit nice looking - aluminum case and a bit of fancy DSP programming.
I'm actually intrigued now. looking under the hood of my Ayre qb-9 dac, it looks to me of well thought out top notch quality components, a little old school with some components which i love as they are not going for newest and supposedly greatest (i prefer a lot of earlier electronics, more warmth) and that is what makes ayre sound so good. it's the combining of components that make the whole. out of all my DAC's including weiss, lavry, benchmark, i think ayre has the most engulfing and seductive sound. end of the day, that is what matters. yes it is expensive, but the sound i'm getting is worth it.
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The good news is that audiophiles are using and tweaking MC
The bad news is that audiophiles are using and tweaking MC and commenting about it as audiophiles do
Subjective listening vs. scientific measurements. Audio Asylum vs. Hydrogenaudio.
It's all good though. The audio world needs a mix of both. And most of us end up configuring or tweaking MC and our audio gear to our own personal preferences and biases and likes. I can't justify every decision or configuration I've got in MC and my computer setup with ABX tests or hard numbers. It's just not practical to do things that way. In many cases you just have to go with what sounds right subjectively whether it always makes sense or not.
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I can't justify every decision or configuration I've got in MC and my computer setup with ABX tests or hard numbers. It's just not practical to do things that way. In many cases you just have to go with what sounds right subjectively whether it always makes sense or not.
I understand and agree. But it's explanation I'm looking for, not justification. It would be helpful if those who understand more about these things try to respect the fact most of us know less. I don't have to be an audiophile to want the best sound available—in my particular circumstances. If a claim of better sound is made without any explanation at all, I'll probably have no choice by to reject it. I have no way of knowing if a wild deviation from a default "recommended" setting might cause unforeseen problems. I may not understand whether the result reported is something I might experience too, or something only possible using something like the poster's $20,000 system.
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I understand and agree. But it's explanation I'm looking for, not justification. It would be helpful if those who understand more about these things try to respect the fact most of us know less. I don't have to be an audiophile to want the best sound available—in my particular circumstances. If a claim of better sound is made without any explanation at all, I'll probably have no choice by to reject it. I have no way of knowing if a wild deviation from a default "recommended" setting might cause unforeseen problems. I may not understand whether the result reported is something I might experience too, or something only possible using something like the poster's $20,000 system.
Is just one of those things. I have no explanation except what i hear, but i do know it's not a placebo for me. i just find this sounds better to me. yes, my post could be seem to read it was night and day experience that all will get, but in retrospect some might not notice at all the difference, because of their setup, and/or listening habits, and doens't affect it. fortunately i do and loved to let others know. the sound opens up a little more on my system. it does take me about 10 seconds into the swing of a song to hear what it's doing and is enough for me to stick with this setting and find it superior. i guess this is big for me. Others will think it's non existent and a waste of time, because if something doesn't sound instantly different first 2 seconds of a track, it doesn't exist to them, which is fair enough and will not say they are wrong. This hobby is about getting the last 1% of performance where you can, and all adds up in the end.
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I understand and agree. But it's explanation I'm looking for, not justification. It would be helpful if those who understand more about these things try to respect the fact most of us know less. I don't have to be an audiophile to want the best sound available—in my particular circumstances. If a claim of better sound is made without any explanation at all, I'll probably have no choice by to reject it. I have no way of knowing if a wild deviation from a default "recommended" setting might cause unforeseen problems. I may not understand whether the result reported is something I might experience too, or something only possible using something like the poster's $20,000 system.
There isn't a scientific explanation for some of the audiophile tweaks. They are what they are and have a folklore of explanation to try to justify them.
Expensive USB cables affecting the sound? No scientific explanation
WAV or AIFF vs. FLAC? No explanation. Trade-off if you want to go the WAV/AIFF route is that those formats have less tagging support so you end up making things more difficult on yourself to manage your library and larger files sizes can be inconvenient if it means you library must now span multiple hard drives. I go with FLAC.
More hardware buffering? Can be a good thing for home listening but a bad thing for professional audio where they need low latencies for editing and recording music.
More RAM buffering (playing from RAM)? No explanation based on software engineering or other engineering. No real downside though if you want to enable it.
Event style WASAPI vs. regular WASAPI? I don't know. Maybe it's something affected by the particular DAC. Maybe audio interfaces designed for studio work do event style better/differently than audiophile home DACs? I don't hear a difference on my gear (but my setup isn't all that transparent and is more mid-fi than audiophile). I just go with what is more robust against glitches for my setup. For me that's event style WASAPI or ASIO.
I'm not in the audiophile tweaker category. I do have fun though exploring what audio and audio gear has to offer. Later today I'm heading to a head-fi meet where I'll get to listen to some fancy gear. It's all fun and I'm looking for a new amp and some new headphones so it's a good way to explore different sound to find what I'm looking for.
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Normally, when there is an audible difference, it's possible to find the associated measurement. In software, due to windows real-time-clock resolution, it's hard to control the jitter of diginal signals. If there is sufficient buffering and control in the hardware, then this can be further controlled by the hardware clock. It's a pity that not enough information is available, which is the general trend in most commercial products. Whenever I have a question, support personnel most likely are using the same manual I look at, with no hands on experience or further data about the product.
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... In software, due to windows real-time-clock resolution, it's hard to control the jitter of diginal signals. If there is sufficient buffering and control in the hardware, then this can be further controlled by the hardware clock. ...
If you're playing to a receiver or a USB DAC, the description above isn't quite accurate.
The Windows software (MC in this case) sends the audio in a binary stream to the device. There is some starting and stopping of this stream, mainly to fill the buffer on the hardware device.
The device itself is then responsible for playback, taking the zeros and ones from its own buffer and converting them to analog form for playback.
So, no matter how irregularly the Windows software might feed the buffer on the device, there is no way that it could affect the sound output from the device. The software has no idea what the device is doing except when the device says "My buffer is getting low. Send me a few more ones and zeros." or the software says "Hey, can you take any more ones and zeros?"
In other words, the software is not responsible for the playback. It is only responsible for the delivery of these binary packages.
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If you're playing to a receiver or a USB DAC, the description above isn't quite accurate.
The Windows software (MC in this case) sends the audio in a binary stream to the device. There is some starting and stopping of this stream, mainly to fill the buffer on the hardware device.
The device itself is then responsible for playback, taking the zeros and ones from its own buffer and converting them to analog form for playback.
So, no matter how irregularly the Windows software might feed the buffer on the device, there is no way that it could affect the sound output from the device. The software has no idea what the device is doing except when the device says "My buffer is getting low. Send me a few more ones and zeros." or the software says "Hey, can you take any more ones and zeros?"
In other words, the software is not responsible for the playback. It is only responsible for the delivery of these binary packages.
I get that, and understand, but why does different buffer sizes sound different then, and why does WASAPI sound different to direct sound when it is bit perfect, also ASIO and Kernel streaming sound different. Foobar sounds quite different to J river yet is all bit perfect using WASAPI?
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If you're playing to a receiver or a USB DAC, the description above isn't quite accurate..............
In other words, the software is not responsible for the playback. It is only responsible for the delivery of these binary packages.
If so I wonder why so many people say MC sounds better than most other players?
Rod
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If so I wonder why so many people say MC sounds better than most other players?
The stream of bits the Windows software sends must be unaltered. Not all players or playback modes will do that.
Any two players sending this unaltered stream should sound identical when played by the same hardware.
There are a lot of ways to do this wrong.
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There isn't a scientific explanation for some of the audiophile tweaks.
I don't need a scientific explanation. A logical or meaningful explanation—as long as it's not evading an available scientific one—is fine. As you've illustrated, "no scientific explanation" can be the basis of a meaningful one. If an experienced engineer can explain why there can be no audible difference caused by different cables or lossless file formats, then that is a meaningful explanation. If you trust the engineer more than the listener's unsubstantiated claim, then it is a scientific explanation: There is no difference, and the question becomes, why does the listener believe there is a difference.
When these issues are discussed honestly and fairly, everyone is better served. On the question of file format, for example, knowing the facts, I can confidently choose FLAC. I can be certain it produces exactly the same result as any other lossless format, and has the advantages of being in common use and can be tagged according to some standards. Someone else can choose to believe—due to some unexplained reason—WAVE files sound better. In making that choice, they can weigh their perceived benefit against the cost (a collection that is a little more difficult to maintain).
The irony that seems to be present in much audiophile folklore is the "open minded" listener who hears some difference in sound quality cannot accept it as unexplained. So they make up something that's false and misleading. The need to do so only suggests a difficultly discerning real from imaginary. It also seems to deny the fact the most important (and variable) processing of the sound takes place in the listener's mind. Some have the confidence to simply say, "I use WAVE because I imagine it sounds better, and that makes me happy." There's no reason for anyone to argue with such a statement.
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The stream of bits the Windows software sends must be unaltered. Not all players or playback modes will do that.
Any two players sending this unaltered stream should sound identical when played by the same hardware.
There are a lot of ways to do this wrong.
I owned a Weiss DAC2, it has a bit perfect stream checker on it. It comes with WAV files that are played by the playback computer program you use. If the Wav files are played unaltered (no dsp, sampling rate, bit rate), three lights on the DAC will light up. if anything has affected the WAV file anywhere down the stream, the lights will not flash. My point being, foobar and J River where playing bit perfect according to this checker. yet sound different. There is more going on here than just getting the ones and zeros in the correct order. So, these 2 software are playing bit perfect with quite different sound.
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I don't need a scientific explanation. A logical or meaningful explanation—as long as it's not evading an available scientific one—is fine. As you've illustrated, "no scientific explanation" can be the basis of a meaningful one. If an experienced engineer can explain why there can be no audible difference caused by different cables or lossless file formats, then that is a meaningful explanation. If you trust the engineer more than the listener's unsubstantiated claim, then it is a scientific explanation: There is no difference, and the question becomes, why does the listener believe there is a difference.
When these issues are discussed honestly and fairly, everyone is better served. On the question of file format, for example, knowing the facts, I can confidently choose FLAC. I can be certain it produces exactly the same result as any other lossless format, and has the advantages of being in common use and can be tagged according to some standards. Someone else can choose to believe—due to some unexplained reason—WAVE files sound better. In making that choice, they can weigh their perceived benefit against the cost (a collection that is a little more difficult to maintain).
The irony that seems to be present in much audiophile folklore is the "open minded" listener who hears some difference in sound quality cannot accept it as unexplained. So they make up something that's false and misleading. The need to do so only suggests a difficultly discerning real from imaginary. It also seems to deny the fact the most important (and variable) processing of the sound takes place in the listener's mind. Some have the confidence to simply say, "I use WAVE because I imagine it sounds better, and that makes me happy." There's no reason for anyone to argue with such a statement.
The irony seems to be, people who don't hear any difference between audio players and formats, also seem to be the people who don't have a decent setups and/or an environment and/or the time to which to test these results. People who do hear the differences, seem to have much more revealing equipment and good environment and the time to come to conclusions.
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MP3 and WAV files sound the same? 128kbs CD Quality the encoder use to say. i remember arguments that went on forever that MP3 and CD sound exactly the same to a lot of listeners, even blind tests. you couldn't argue with them. But there was an explanations that proved that what you hear is not the identical to the CD. Yet they still didn't hear it?
May i ask the people who don't hear a difference, don't believe the crazy audiophiles, if you had a scientific explanation, would you suddenly hear a difference?
Would be interesting to see the people who don't hear a difference, suddenly hearing a difference only because they had an scientific explanation. Now that would be crazy
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My point being, foobar and J River where playing bit perfect according to this checker. yet sound different. There is more going on here than just getting the ones and zeros in the correct order. So, these 2 software are playing bit perfect with quite different sound.
The software is just sending the bits to the hardware, not telling it how to play it. If the hardware is getting the same bits (bit perfect) from both pieces of software how could it sound different? It can't...
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The software is just sending the bits to the hardware, not telling it how to play it. If the hardware is getting the same bits (bit perfect) from both pieces of software how could it sound different? It can't...
no it can't, but it does.
Jitter, EMI, Clocks, timing. there is more than just one's and zero's.
i Really would LOVE to invite people on here to my house, using my hi-fi to show you what i'm on about.
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I've removed a post and a reply to it. The post was needlessly argumentative in my opinion.
Please respect the rights of others to believe what they wish to believe. It's fine to say you don't agree, or to say why you think otherwise. It isn't OK to be disrespectful.
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no it can't, but it does.
Jitter, EMI, Clocks, timing. there is more than just one's and zero's.
i Really would LOVE to invite people on here to my house, using my hi-fi to show you what i'm on about.
Thanks. I wish I could take you up on the invitation.
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MP3 and WAV files sound the same? 128kbs CD Quality the encoder use to say. i remember arguments that went on forever that MP3 and CD sound exactly the same to a lot of listeners, even blind tests. you couldn't argue with them. But there was an explanations that proved that what you hear is not the identical to the CD. Yet they still didn't hear it?
[edited by JimH]
Are you really enjoying the media experience any more?
I guess everyone has their thing!
All that said I'd love to see your set up. People's fascinations fascinate me!
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there is more than just one's and zero's.
Not as far as the software is concerned. If the hardware is getting the same bits from both software players it would mean that the hardware is playing the exact same data differently which would suggest a hardware problem.
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Not as far as the software is concerned. If the hardware is getting the same bits from both software players it would mean that the hardware is playing the exact same data differently which would suggest a hardware problem.
No, no problem with the hardware. all works perfectly. just sounds different with different settings and players. :)
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No, no problem with the hardware. all works perfectly. just sounds different with different settings and players. :)
Well if the software players are delivering the same bits to the hardware and it sounds different to you then there is something wrong with either your hardware or ears because it has nothing to do with the software at that point.
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Well if the software players are delivering the same bits to the hardware and it sounds different to you then there is something wrong with either your hardware or ears because it has nothing to do with the software at that point.
Once again No, no problem with the hardware. Ears work Great, my goal profession is audio mastering, you need golden ears for that.
Have to think of it as a whole. The PSU, hardware, ram, software, the file format, the decoding process, the operating system processes, USB hardware, USB cabling, the DAC. anywhere along this chain of events things could have slight change in the timing of the data or added EMI somewhere/anywhere from upstream form other processes, CPU clocks, Power fluctuations due to other processes demanding CPU time. any number of things can result in different variation of the final analogue sound.
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Using a PC for audio is PCM audio.
PCM is samples (the bits) and sample rate, the timing.
Both must be right.
The bits are probably the easiest part. When configured right players like MC can deliver bit perfect output.
The timing is a more complex affair.
We need some expensive gear to measure small deviations in timing properly.
Home Theater and High Fidelity tested the jitter on the SPDIF out of a blue-ray player in stop mode and running.
(http://www.thewelltemperedcomputer.com/Pictures/Hardware/Connect/jitter-spectrum-coaxial-connection.gif)
Indeed running the system doubles the jitter.
However, 5 or 10 ps are very low values.
The periodic jitter (the spikes) is probably more relevant.
One thing is obvious, indeed the higher electrical activity as a result of the system running maps into a measurable difference in jitter at the SPDIF out.
How this translates to a "PC" is another question.
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Using a PC for audio is PCM audio.
PCM is samples (the bits) and sample rate, the timing.
Both must be right.
The bits are probably the easiest part. When configured right players like MC can deliver bit perfect output.
The timing is a more complex affair.
We need some expensive gear to measure small deviations in timing properly.
Home Theater and High Fidelity tested the jitter on the SPDIF out of a blue-ray player in stop mode and running.
(http://www.thewelltemperedcomputer.com/Pictures/Hardware/Connect/jitter-spectrum-coaxial-connection.gif)
Indeed running the system doubles the jitter.
However, 5 or 10 ps are very low values.
The periodic jitter (the spikes) is probably more relevant.
One thing is obvious, indeed the higher electrical activity as a result of the system running maps into a measurable difference in jitter at the SPDIF out.
How this translates to a "PC" is another question.
Thanks Vincent,
makes a lot of sense to me. Timing and EMI are the main issues than just 1's and 0's. Been playing around with computer PSU's, clock speeds, power fllters, and software activity to reduce this type of thing. so far doing well. but still far to go. I look forward to the day where we have specially designed PC transports for hi end Hi Fi.
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I don't have a great feel for what those numbers mean but it does appear that the scale is "picoseconds" and "nanoseconds". If someone can hear differences on that scale then I think they should wear a cape and mask and have some kind of sidekick.
Also, please point me to the "EMI" setting in MC.
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Also, please point me to the "EMI" setting in MC.
Memory playback
Indeed no head movements during playback so reducing electrical activity.
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Timing and EMI are the main issues than just 1's and 0's.]
EMI will only impact audio quality when using analog outputs (post-DAC). Digital bitstreams on copper ARE certainly susceptible to EMI, like any electrical signal. The difference is that digital signals sent over a serial digital bus (like USB, 1394, or SPDIF) cannot "degrade" gracefully when they encounter EMI.
People think of EMI as impacting digital audio using the context they've learned over years of dealing with analog audio (ground loops and whatnot), but they're not the same thing at all. Any data coming over a digital bus will have a certain amount of redundancy built-in for error correction, so that if the bitstream becomes corrupted in route, the devices on the bus actually KNOW that it has been corrupted. It can either recover the data using the error correcting parity bits (and then when it plays it is still bit-for-bit perfect) or it will lose the data completely. At that point, either you'll hear a very audible "gap" in playback, or if there is enough time in the buffer (and the bus supports it) the "recipient" can request that the "sender" re-send the missing data. EMI literally cannot "color" the music or anything like that over a digital bus. It can corrupt it, but you'd absolutely KNOW that something was going on, because you'd get either nothing at all or horrible "screeching and static" noises. Either the interference isn't enough to change the data, or the data stream becomes corrupted (more than the error correction can handle) and it fails. Flipping even one "bit" from an EMI blast doesn't "color" the sound, it breaks it.
Timing is certainly something else, and that's why any DAC will buffer the input before sending it to output. The buffer on the DAC allows it to "smooth out" the jitter and other latency problems on the input signal. This latency can come from the sending device, or from the DAC itself (latency can be introduced by the DAC itself as it processes error correction). As long as the latency doesn't exceed the buffer size of the DAC, the latency also should not have an audible impact (unless the DAC is very poorly designed). If a buffer underrun occurs, again, the audio will actually just stop or "skip", not change in "quality".
Of course, computers are much more variable in the workload they have to process, so the jitter patterns are possibly substantially more complex than what you'd get off of a standalone component (though not always, I've seen standalone devices outrun their buffers plenty of times, especially when loading a particularly "complex" feature of some kind). So, it is theoretically possible (though unlikely), that you could end up with audible problems due to jitter if there is so serious that it is constantly "on the edge" of causing a buffer underrun on the DAC. If this is the case, you probably either have serious EMI problems or a very underpowered PC. Audio is just not that "difficult" of a computational task for modern CPUs to handle.
Likewise, computers are absolutely NOT the only devices that suffer from jitter. In fact, that BluRay player (or DVR, game console, DVD player, or even standalone CD player) you have really IS actually a computer inside. If you tear the case open and look, you'll see that they use many of the exact same components as you'd find on your PC's motherboard. There are only so many chip designs for things like DSPs and whatnot out there, and real custom silicon is incredibly expensive (we're talking Billions with a B just to get test spins of silicon designs which might not even work). I'm sure there are a few audio devices that use FPGAs (http://en.wikipedia.org/wiki/Field-programmable_gate_array) to do some processing, but those are still expensive and SLOW (and difficult to program effectively without a large team of highly-paid chip engineers). I'd be willing to bet that actual custom silicon (where they contract TSMC or TI to build them a real chip) in the audio processing space is either completely non-existent or incredibly rare and reserved for things way out of all of our price ranges. There just isn't a good reason to do it and the costs would price you out of even the audiophile market without economies of scale. Even if they DID do it, I would be very skeptical that they could do it well. Custom silicon design is an incredibly complex and expensive process to do correctly, and there are all sorts of things that can wrong. Custom is absolutely NOT better. Custom is generally LESS reliable.
Now, to be clear, that isn't to say that there isn't a difference between a high-quality DAC and a cheap consumer one! Far from it. It is just that the differences typically lie in features (amount of buffer, accuracy of timing clocks, etc) on the digital side, and on the ANALOG side (post-decode). And, it is entirely possible that different DACs will react to "unexpected conditions" in unpredictable ways post-decode. I can, at least, imagine that some very high levels of jitter and/or latency could conceivably cause problems with some specialized DACs that expect a pristine input signal (I'd call that poor design, but who knows). It is also possible that some DACs might react differently on the analog side to types of connections on the input side (even though the bits are the same, it might process the SPDIF input differently than it does the USB input or HDMI input). They probably shouldn't (the same digital bits coming in, once decoded, should route to the same exact system for output), but who knows. I can imagine that DACs probably have complex logic to work around many of the "errors" that are common on crappier input devices (improperly formatted streams and whatnot). This logic introduces complexity that could conceivably "color" the analog output side, I suppose (again, I'd blame the DAC though).
But, for a digital input bitstream sent over a digital serial bus, either the information gets there perfectly or it doesn't get there at all. There is no in-between like there is with an analog signal.
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[author=glynor link=topic=62784.msg421149#msg421149 date=1299519018]
EMI will only impact audio quality when using analog outputs (post-DAC)
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I disagree. EMI affects digital stream as it has a clock. A clock is analogous in nature, there are plenty of articles regarding 'sample rate' of digital audio. Whether the clock is in the computer or dac, EMI from electrical components ESPECIALLY computer affects this directly or indirectly through powered components. Computers have many clocks running at once at different frequencies. the psu in a computer is swtich mode. not sure if you have read how dirty and noisy a switch mode power supply is?
Digital bitstreams on copper ARE certainly susceptible to EMI, like any electrical signal. The difference is that digital signals sent over a serial digital bus (like USB, 1394, or SPDIF) cannot "degrade" gracefully when they encounter EMI.
USB have blocks of data sent analogously, 'timing' it is VERY susceptible from noise of EMI.
Timing is certainly something else, and that's why any DAC will buffer the input before sending it to output. The buffer on the DAC allows it to "smooth out" the jitter and other latency problems on the input signal. This latency can come from the sending device, or from the DAC itself (latency can be introduced by the DAC itself as it processes error correction). As long as the latency doesn't exceed the buffer size of the DAC, the latency also should not have an audible impact (unless the DAC is very poorly designed). If a buffer underrun occurs, again, the audio will actually just stop or "skip", not change in "quality".
The EMI from computers will reach the DAC, it's best to reduce computer activity to clean up the timing. power filters also work well, to some extent.
Of course, computers are much more variable in the workload they have to process, so the jitter patterns are possibly substantially more complex than what you'd get off of a standalone component (though not always, I've seen standalone devices outrun their buffers plenty of times, especially when loading a particularly "complex" feature of some kind). So, it is theoretically possible (though unlikely), that you could end up with audible problems due to jitter if there is so serious that it is constantly "on the edge" of causing a buffer underrun on the DAC. If this is the case, you probably either have serious EMI problems or a very underpowered PC. Audio is just not that "difficult" of a computational task for modern CPUs to handle.
but computers are not made to be Audio transports either, the auido over USB is very much an afterthought. DAC makers now have improved GREATLY with Asynchronous mode USB, and not dreaded Adaptive. I guess people were ok with, and had no issues with Adaptive mode until Asynchronous popped up? I like to keep an open mind on this subject and glad DAC makers do to. It's what improves the audio over the computer for us. When people say, no way, digital will just work, is when we get stuck in ruts and no desire to improve it. USB audio is improving, due to the open minds of people who want more from it. Funny enough, with these kinds of discussions, audio from a computer is improving!!
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wasapi events is the mode i use for my external DAC. I use JRiver only because it has this driver.
i Ive tried others software and drivers, the wasapi events is very close to aqvox and ploytec asio drivers (that are notre free).
i confirm the usb cable is relevant. I use, at the USB input of DAC a supply for produce the 5V to the input usb receiver circuit of the DAC (this supply replace the 5V of USB cable delivered by the computer, i placed a filtered power chord Lavardin for this supply for better results).
The differences with this software/drivers and hardware settings are (with my installation but NOT ONLY with MY ears) very substantials. The problem is to choose...
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I guess people where ok about Isochronous until Asynchronous popped up?
AV over USB is done in Isochronous transfer mode.
Isochronous transfer can be done with three possible types of synchronization in the USB audio device, synchronous, adaptive and asynchronous.
Asynchronous mode is considered best in terms of jitter reduction compared with the 2 other modes.
As usual results depends very much on the quality of the implementation.
http://thewelltemperedcomputer.com/KB/USB.html
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wasapi events is the mode i use for my external DAC. I use JRiver only because it has this driver.
i Ive tried others software and drivers, the wasapi events is very close to aqvox and ploytec asio drivers (that are notre free).
i confirm the usb cable is relevant. I use, at the USB input of DAC a supply for produce the 5V to the input usb receiver circuit of the DAC (this supply replace the 5V of USB cable delivered by the computer, i placed a filtered power chord Lavardin for this supply for better results).
The differences with this software/drivers and hardware settings are (with my installation but NOT ONLY with MY ears) very substantials. The problem is to choose...
I think the WASAPI event is a great development. I can see it working realy well in some DAC's. WASAPI orignal just has that bit of an edge in transparency to make me want to use it on the Ayre.
I'm very curious if the separate break in USB power supply will work well with my Ayre DAC. as the Ayre's USB controller is still powered by the computer, but the digital stream is isolated with optics. although Ayre have gone to great lengths to improve noise over the 5v, wonder if this will take it further.
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I’m afraid you are mixing up the USB transfer modes and the synchronization.
All DACs use Isochronous transfer mode.
Most of them use adaptive mode but if we look at recent offerings asynchronous is becoming more popular.
The QB9 is a USB class 2 audio device using isochronous transfer mode with asynchronous synchronization (DAC times the data out of the PC)
A bit more about transfer modes: http://www.beyondlogic.org/usbnutshell/usb4.shtml
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I’m afraid you are mixing up the USB transfer modes and the synchronization.
All DACs use Isochronous transfer mode.
Most of them use adaptive mode but if we look at recent offerings asynchronous is becoming more popular.
The QB9 is a USB class 2 audio device using isochronous transfer mode with asynchronous synchronization (DAC times the data out of the PC)
A bit more about transfer modes: http://www.beyondlogic.org/usbnutshell/usb4.shtml
Ah yes, i see!, Adaptive or Asynchronous i should have said. not Asynchronous or Isochronous. Have fixed up the post, Thanks
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I think the WASAPI event is a great development. I can see it working realy well in some DAC's. WASAPI orignal just has that bit of an edge in transparency to make me want to use it on the Ayre.
I'm very curious if the separate break in USB power supply will work well with my Ayre DAC. as the Ayre's USB controller is still powered by the computer, but the digital stream is isolated with optics. although Ayre have gone to great lengths to improve noise over the 5v, wonder if this will take it further.
the 5V supply i use can be seen on aqvox site http://www.aqvox.de/usb-power_en.html (http://www.aqvox.de/usb-power_en.html), ive tried (and bought) their usb cable. ive added flirerd power chord for the supply , i've choosed a lavardin power chord http://www.lavardin.com/lavardin-cablesE.html (http://www.lavardin.com/lavardin-cablesE.html).
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[author=glynor link=topic=62784.msg421149#msg421149 date=1299519018].
I disagree. EMI affects digital stream as it has a clock. A clock is analogous in nature, there are plenty of articles regarding 'sample rate' of digital audio. Whether the clock is in the computer or dac, EMI from electrical components ESPECIALLY computer affects this directly or indirectly through powered components. Computers have many clocks running at once at different frequencies. the psu in a computer is swtich mode. not sure if you have read how dirty and noisy a switch mode power supply is?
I think we have different understandings of what I meant by "impacting audio quality". I suppose I should have used more precise language. I meant that it cannot change the content of the audio stream.
I do agree that timing can be impacted, perhaps dramatically (but almost certainly not in most cases), by latency and jitter on the digital bus. I specifically exempted timing. But, because in the context of "home theater" audio playback there is no reason to require real-time playback, and timing issues for any digital bus can be handled by a buffer (assuming that the DAC can get or generate an accurate clock), I'm not sure it matters in this context. Audio interfaces used for real-time recording and synthesis, pro audio gear, obviously can't use a substantial input buffer or rely upon an external clock, and for them, latency matters a great deal. Also, latency is a big problem when you need to sync audio with a video track. For basic music playback, though, you have plenty of opportunity to use a buffer size so substantial that timing variations on the bus should be completely mitigated. The vast majority of the time, any failures caused by EMI interference on the bus are going to be so spectacular (like if you have a "broken" cable) that the audio stream will stop or stutter dramatically (and everyone in the room would know something is terribly wrong).
But either way... I do admit that I don't know a huge deal about high-end audiophile DAC design, so I could certainly be completely wrong. So I'll grant you, it is possible (though unlikely) that timing issues on the bus could, maybe, have an audible impact.
However, I do know quite a bit about how data transmissions work over a digital bus. If there are high-end DACs that depend upon the stream coming from the computer to handle timing without an adequate buffer and without a way to guarantee a clock accurate enough to exceed the human capacity to notice the variations, then I would call that a failing of the DAC design, not of the computer sending the data stream. Maybe they exist, and maybe they're everywhere, but that doesn't seem that it should be an insurmountable challenge of engineering. The impact of latency and EMI on CPU and GPU design is a much more difficult problem, but yet Nvidia and AMD put out new GPUs every 6-8 months that seem to be solving those same problems under much greater demands for precision. If audio devices really are as sensitive to these latency issues as some manufacturers would have you believe, then it would seem to be either (a) an unbelievable engineering failure on the part of the device manufacturers, or (b) done on purpose. And, even if it was true, I would also guess that the same exact issues would impact the vast majority of other audio consumer electronics devices manufactured today, because they all suffer from the same design problems (in fact, EM problems in the CE space are often magnified by smaller case designs and tighter thermal requirements). So what is the point of reference?
That was generally the point of my previous missive. EMI can impact timing, it cannot impact the data stream integrity in a way that doesn't cause spectacular and obvious failures. I suspect that you can mitigate the timing issues, but I grant that this is a troublesome area. But if the total latency is measured in values of less than 1ns? I'm open minded, but skeptical.
In the end, I'll say this: If the goal is to chase perfection for the sheer joy of perfection, then that's fine, it just does not appeal to me. There is no true perfection in this world, and I'm really okay with that. Perfection is boring.
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I thought I was done, but I wasn't quite. I'm gonna get philosophical, cause, you know... Why not? ;D
I don't, personally, equate perfection in audio accuracy with "quality". I don't even really fully understand the desire. The actual music you are listening to (with perhaps the exception of some drum and bass stuff made by engineers/physicists/crazy people) was almost certainly not made with the same tolerances in mind. There really is no perfection. Heck, many of the artists themselves can't even hear very well, and a good percentage of the best were stoned out of their minds when they made the music (including, and especially, those classical composers). And no recording system yet invented by man really accurately captures what it is to be there to experience the moment live. When we play back a recording, it is a convincing simulation of a moment which usually never actually really happened.
Music isn't about that for me. It isn't about perfection. The vocals for one of my favorite songs were recorded with an answering machine "because it was there". One of my most powerful memories of music was listening to a U2 song played on the radio, filled with static, tuned by a terrible GE alarm clock radio.
Music is about the experience, and how it affects you, personally.
And, that's why I'd say "I don't get it, but more power to you." If it really makes your experience better, who cares if it is psychosomatic or not? So long as you have the money to buy the stuff (you aren't risking your future or hurting anyone else) then what does it matter and who am I to judge?
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I thought I was done, but I wasn't quite. I'm gonna get philosophical, cause, you know... Why not? ;D
I don't, personally, equate perfection in audio accuracy with "quality". I don't even really fully understand the desire. The actual music you are listening to (with perhaps the exception of some drum and bass stuff made by engineers/physicists/crazy people) was almost certainly not made with the same tolerances in mind. There really is no perfection. Heck, many of the artists themselves can't even hear very well, and a good percentage of the best were stoned out of their minds when they made the music (including, and especially, those classical composers). And no recording system yet invented by man really accurately captures what it is to be there to experience the moment live. When we play back a recording, it is a convincing simulation of a moment which usually never actually really happened.
Music isn't about that for me. It isn't about perfection. The vocals for one of my favorite songs were recorded with an answering machine "because it was there". One of my most powerful memories of music was listening to a U2 song played on the radio, filled with static, tuned by a terrible GE alarm clock radio.
Music is about the experience, and how it affects you, personally.
And, that's why I'd say "I don't get it, but more power to you." If it really makes your experience better, who cares if it is psychosomatic or not? So long as you have the money to buy the stuff (you aren't risking your future or hurting anyone else) then what does it matter and who am I to judge?
I completely agree with you here.
I love music, and i also love hi-fi and electronics ;) rare occasions the 2 do not go hand in hand. some days it's all about the music, and enjoy listening even on, dare i say cheap ipod headphones walking around the city. I almost sure i am deliberately using the shittiest of headphones just so i get involved with the music than tweaking it to get it better.
Then i get home to the hi-fi. it sounds so surreal like i'm actually in the same room as the musicians, as mentioned before i love audio mastering so i think about how it sounds.. a lot. when the bug hits, I want more, is like a drug. if you spend quite a bit of time with very transparent equipment, you realise that nearly anything you do, can change the sound just a tiny bit, or very noticeable. It is an obsessive hobby. It's not for everyone! ;D
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ive choosed the wasapi events driver and made many adjustments in hardware stuffs, coerds ans supplies to optimize without change the driver mode.So today, cause of this discussion, ive tried the wasapi dtriver (not the event one), and the sound is really another one, more mixed, soft, sweet, confortable, bath of sound, the wasapi events is more precise, clear, physical, punchy, real presence of musicians and air around them, 3d effects etc... the 2 are convenients with the different minds i could have when i go to listen music.Intersesting...
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ive choosed the wasapi events driver and made many adjustments in hardware stuffs, coerds ans supplies to optimize without change the driver mode.So today, cause of this discussion, ive tried the wasapi dtriver (not the event one), and the sound is really another one, more mixed, soft, sweet, confortable, bath of sound, the wasapi events is more precise, clear, physical, punchy, real presence of musicians and air around them, 3d effects etc... the 2 are convenients with the different minds i could have when i go to listen music.Intersesting...
Did you try it with the WASAPI buffer set at 1.99sec, and pre buffer to 20sec?
Yes i agree, the WASAPI original is softer, sweeter, golden, more analoge soundinging i would say and musical bit more depth. The event style, precise, little bleached, bit more bass presence. i guess my ears and system likes original wasapi. Am glad you heard what i heard ;D!!!
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not yet, i have to go now, ill try it
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I wanted to chime in with a recent experience.
I was listening to Pixies Trompe Le Monde and noticed some serious sibilance. I had never noticed it before and scratched my head. I skipped around through different albums and different artist and I heard it everywhere.
I approached the speaker with an ear at the tweeter, I panned the balance back and forth, etc... It was coming through in both channels and was actually sort of "scratchy", almost distorted.
I paused the music and gave all contacts a "cleaning" by repeated insertion, I made sure my cable dressing (rat's nest) was reasonable, checked all connections etc.... When I went to check the toslink output of my Juli@ soundcard i found the culprit. The Chord toslink cable was barely inserted. As soon as I touched it, it fell out.
I snapped the puppy back in tightly, and resumed playback. All was perfect again.
Many people seem to think that 0s, and 1s, either work or they dont, they are either there or they are not. I thought the same in a way I suppose. I figured if the cable either sent a signal or not. If it had a poor connection I thought I would hear drop-outs even if quick and minute. In this case the bad connection resulted in EASILY noticeable high frequency distortion or noise. Anyone who heard it would have assumed it was FM radio or a blown tweeter or something.
All I am saying is that sometimes there is more going on than what meets the (usually uneducated) eye. On top of that there are many conflicting experiences. I cannot discern a difference in FLAC vs WAV. I can though hear the difference in various digital cables. I can pick a 320mp3 from lossless EVERY time (in my kit). But I cannot hear any difference between KS and ASIO.
I am sure there is someone out there with the exact opposite experience with their audio player set-up, DAC, room, ears etc...
The crux of the biscuit though, as some have said above, if it sounds better to you than it is. To eschew anyone's honest experiences is pretty low.
-Patrick
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I wanted to chime in with a recent experience.
I was listening to Pixies Trompe Le Monde and noticed some serious sibilance. I had never noticed it before and scratched my head. I skipped around through different albums and different artist and I heard it everywhere.
I approached the speaker with an ear at the tweeter, I panned the balance back and forth, etc... It was coming through in both channels and was actually sort of "scratchy", almost distorted.
I paused the music and gave all contacts a "cleaning" by repeated insertion, I made sure my cable dressing (rat's nest) was reasonable, checked all connections etc.... When I went to check the toslink output of my Juli@ soundcard i found the culprit. The Chord toslink cable was barely inserted. As soon as I touched it, it fell out.
I snapped the puppy back in tightly, and resumed playback. All was perfect again.
Many people seem to think that 0s, and 1s, either work or they dont, they are either there or they are not. I thought the same in a way I suppose. I figured if the cable either sent a signal or not. If it had a poor connection I thought I would hear drop-outs even if quick and minute. In this case the bad connection resulted in EASILY noticeable high frequency distortion or noise. Anyone who heard it would have assumed it was FM radio or a blown tweeter or something.
All I am saying is that sometimes there is more going on than what meets the (usually uneducated) eye. On top of that there are many conflicting experiences. I cannot discern a difference in FLAC vs WAV. I can though hear the difference in various digital cables. I can pick a 320mp3 from lossless EVERY time (in my kit). But I cannot hear any difference between KS and ASIO.
I am sure there is someone out there with the exact opposite experience with their audio player set-up, DAC, room, ears etc...
The crux of the biscuit though, as some have said above, if it sounds better to you than it is. To eschew anyone's honest experiences is pretty low.
-Patrick
Thanks Patrick.
That's a perfect example. The 1's and 0's were there, but it sounds like the timing of these bit's was all a bit of a mess. sounds like what you heard was extreme jitter! the half unplugged cable with slight tiny vibration especially from a computer fan or speakers, and/or loss of focus of the light source from cable to the toslink input optic will definitely mess up the 44,100 or 48,000 (or whatever your output is) cycle per second timing.
Great album by the way...
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ive never heard a difference between wav and flac files.
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ive never heard a difference between wav and flac files.
some wont. for me, i could never go back.
reason i perceive it to be better, due to less CPU activity, which in turn less computer power, less electrical switching. WAV is a more direct PCM path. more direct path to the dac the better. Even stranger and more inconceivable, AIFF vs WAV. WAV sounds better to me, which really doesn't make sense as both are raw PCM. there are huge discussions on computeraudiophile website with many of us with similar results. these results weren't due to the fact of peer pressure, i found it out myself, then read the discussion which confirmed i wasn't hearing things.
This topic will really get on others nerves. :)
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reason i perceive it to be better, due to less CPU activity, which in turn less computer power, less electrical switching.
Maybe it isn't your convictions, but the way you arrive at them, or try to describe them.
You just stated a "reason" for hearing something. You might state a reason for the difference. But your language says these things are why it sounds better to you.
One would have a VERY hard time not attributing self-fulfilling prophecy or placebo to statements like that.
I would simply stick with reporting your experiences rather than postulate about why. You owe no one any explanation or justification.
In the end everyone will use their own ears. It isn't like it is hard to take the pepsi challenge oneself.
You prefer WAV (many do)
I cant hear a difference so I use FLAC for ease of tagging and its ubiquity (many do)
Lastly, many of these comparisons are done with quick A-B tests, switching back and forth trying to find a difference somewhere.
The ONLY way you will hear anything ULTRA-subtle like an alleged difference between FLAC and WAV will take long listening sessions. Listen with Cable A for a month. Then switch to Cable B for a month. This will give you much better (truer) results.
-P
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Maybe it isn't your convictions, but the way you arrive at them, or try to describe them.
You just stated a "reason" for hearing something. You might state a reason for the difference. But your language says these things are why it sounds better to you.
One would have a VERY hard time not attributing self-fulfilling prophecy or placebo to statements like that.
I would simply stick with reporting your experiences rather than postulate about why. You owe no one any explanation or justification.
In the end everyone will use their own ears. It isn't like it is hard to take the pepsi challenge oneself.
You prefer WAV (many do)
I cant hear a difference so I use FLAC for ease of tagging and its ubiquity (many do)
Lastly, many of these comparisons are done with quick A-B tests, switching back and forth trying to find a difference somewhere.
The ONLY way you will hear anything ULTRA-subtle like an alleged difference between FLAC and WAV will take long listening sessions. Listen with Cable A for a month. Then switch to Cable B for a month. This will give you much better (truer) results.
-P
I do see your point and language used. i wanted to say I do find it better, and to describe what is happening as to why it would sound better. BUT, it sounds better to me, and if i use wrong language that way, people will get annyoed they didn't hear it too. See how this can get confusing.
As for quick A and B, i rarely do that, i do take time listening to if it sounds great, but with more obvious stuff it really does take a 30 second A and B. WAV vs Flac thing, took me just one song to know for sure.
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This topic will really get on others nerves.
But perhaps not for the reasons you think. I have no issue with you claiming to hear some difference, or outlining the settings that might have some bearing on the cause (you have to at least describe what you know is different). What's annoying are the unfounded conclusions that one or more specific things are the direct cause of the difference you claim to hear. If you consistently hear "better" sound from WAV than FLAC then use WAV and be happy. There's no need to tell us WAV produces better sound when you have no real evidence of that. It's equally plausible you have some hardware issue that happens to manifest when playing FLAC. Or maybe the problem is triggered by playing WAV, but you perceive the result as "better."
Also annoying is the dismissal of constructive and informative contributions of others on the basis they must be triggered by some kind of nervous disorder. It's possible others just want to participate in what could be an interesting technical discussion—even if it has no clear resolution. The reasons for their interest likely have more to do with their own circumstance than yours. This is why I attempted to suggest providing some kind of reasoning (factual or theoretical) of cause and effect relationships would make the discussion more positive and useful. Looking back over this, I see glynor provided an excellent demonstration of what I'm talking about. You don't have to accept his conclusions to agree his contributions are enlightening and useful.
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But perhaps not for the reasons you think. I have no issue with you claiming to hear some difference, or outlining the settings that might have some bearing on the cause (you have to at least describe what you know is different). What's annoying are the unfounded conclusions that one or more specific things are the direct cause of the difference you claim to hear. If you consistently hear "better" sound from WAV than FLAC then use WAV and be happy. There's no need to tell us WAV produces better sound when you have no real evidence of that. It's equally plausible you have some hardware issue that happens to manifest when playing FLAC. Or maybe the problem is triggered by playing WAV, but you perceive the result as "better."
Also annoying is the dismissal of constructive and informative contributions of others on the basis they must be triggered by some kind of nervous disorder. It's possible others just want to participate in what could be an interesting technical discussion—even if it has no clear resolution. The reasons for their interest likely have more to do with their own circumstance than yours. This is why I attempted to suggest providing some kind of reasoning (factual or theoretical) of cause and effect relationships would make the discussion more positive and useful. Looking back over this, I see glynor provided an excellent demonstration of what I'm talking about. You don't have to accept his conclusions to agree his contributions are enlightening and useful.
see above, i did give my theories as to why.
why can't i say i prefer WAV. Someone said, they didn't hear a difference between FLAC and WAV, i answered to that 'i did' and gave theories as to why i did. this thread is about these discussions. i don't get your point, and don't get why you are annoyed. I can't show graphs, but Vincent kindly posted one up giving some idea that higher electrical activity can raise jitter.
I'm stating I hear a difference AND giving my reasons as to why. I feel i'm repeating myself.
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Thank you JazzDoc and Blaine78!
I've used WASAPI Event style since I ugraded to Arcam rDAC, but I haven't experimented with the output mode and buffer setting before I read this post. I think you're right, there is a (huge) difference between WASAPI and WASAPI Event style. I think WASAPI is better. The differences as I hear them are more space and air around singers and instruments, and more defined, natural sound with greater depth. With WASAPI Event the sound was more edgy and flat, but maybe more dynamic and punchy.
With WASAPI Event style I rated MC 3rd among my software players. First was XXHighEnd (0.9z-3 demo) and second Foobar2000, both with kernel streaming. Foobar was just a tad better than MC. But with WASAPI MC simply is the best sounding IMO ;D
MC is my number one player anyway, because of the features.
I tried FLAC against WAV, but I couldn't hear any difference. Can it be related to PC setup?
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Thank you JazzDoc and Blaine78!
I've used WASAPI Event style since I ugraded to Arcam rDAC, but I haven't experimented with the output mode and buffer setting before I read this post. I think you're right, there is a (huge) difference between WASAPI and WASAPI Event style. I think WASAPI is better. The differences as I hear them are more space and air around singers and instruments, and more defined, natural sound with greater depth. With WASAPI Event the sound was more edgy and flat, but maybe more dynamic and punchy.
With WASAPI Event style I rated MC 3rd among my software players. First was XXHighEnd (0.9z-3 demo) and second Foobar2000, both with kernel streaming. Foobar was just a tad better than MC. But with WASAPI MC simply is the best sounding IMO ;D
MC is my number one player anyway, because of the features.
I tried FLAC against WAV, but I couldn't hear any difference. Can it be related to PC setup?
yes exactly what i hear too! did you raise the buffer in wasapi setings to 1.99sec (or largest possible without it not working), and increase the pre-buffer to 20sec?
As for Wav vs Flac, if you don't hear a difference, i wouldn't worry about it. some do some don't, maybe you pc processes/decodes flac with no discernible difference.
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After some experiments i have to admit that i hear a real difference between flac and wav files, my hardware configuration have changed several months ago pecially cables) and to make new comparisons is instructive now... i have many, many flacs files in sub folders of a unique folder. Somebody could have an advice for the method for convert all flac files in wav (not subfolder one by one?)
Thank you
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Audio Myths Workshop: http://www.youtube.com/watch?v=BYTlN6wjcvQ
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Audio Myths Workshop...
Just another attempt to cloud the issue with facts. ;D
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Audio Myths Workshop: http://www.youtube.com/watch?v=BYTlN6wjcvQ
This is fantastic. Thanks Alex.
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Audio Myths Workshop: http://www.youtube.com/watch?v=BYTlN6wjcvQ
Nice video, pretty much knew all that and agree with a some, but disagree with much. The narrator of this video is obviously of the school 'measurement equipment only' to get his facts which is flawed, because specs and measurements mean nothing if you don't like how it sounds? While others use their ears to get the sound they like. still doesn't prove anything, just another school of thought. The proof is what the individual person hears and gets out if it. I'm one of those people who are both blessed and cursed with golden ears.
I'll give you an example. I did a course in Audio engineering. In one of the studios i knew immediately something was wrong with the sound that was coming out of the speakers. I mentioned this to the teacher who has been in the field for a number of years. He told me nothing was wrong. i said the speakers are out of phase and it is really obvious to me. Everyone else in the class either didn't notice it or too scared to say anything. he said to me, no they are not, they are set up professionally and are balanced wired speakers, they can't be out of phase. he believed there was nothing wrong because he had faith in the fact 'balanced speakers can never be out of phase' and a 'pro' set them up. turns out they were out of phase according to another teacher and they were fixed.
Would also like to add, i'm not such a audiophile nut as to endorse the use and purchase of expensive cabling. Contrary to some beliefs, they do make a difference. the reason they do, is adding or subtracting capacitance and resistance and are the main reasons of hearing differences. Is more like flavors than transparency and i don't subscribe to the crazy prices for what are essentially filters. BUT I believe in using good quality, not expensive copper cabling, I use $3 a meter Jaycar OFC copper and bought some nice quality copper (not brass) RCA plugs. it cost me $50 to make up a 50cm pair. My speaker cables are again good quality OFC $5 per meter.
My Power cables are the ones supplied with the equipment, generic.
The USB cable, unfortunately yes indeed made a great and noticeable difference that was genuine. I compared over and over again to the cheaper ones, the cheaper ones did sound scratchy with loss of definition with most instruments. i would have kept the cheaper ones in if they sounded as great and saved $100.
My point being, i'm not an over the top audiophile, and i'm not here to spread nonsense.
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It would be interesting if MC had an ABX testing module. Foobar has an ABX testing module. But theirs is designed for comparing different audio files. In MC we've got zones. That opens up additional testing possibilities. For example you could have one zone set with a DAC doing ASIO and another zone set with the same brand/model DAC doing WASAPI Event Style. Then the ABX test could compare/test the two. That would be neat, if it's even something that would be possible in MC. It would also be a neat way to compare two different DACs.
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It would be interesting if MC had an ABX testing module...
Especially if it only did double blind tests, the results of which could only be obtained by posting the data publicly here. 8)
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Especially if it only did double blind tests, the results of which could only be obtained by posting the data publicly here. 8)
Blind tests would be good. then again it will be needed in controlled environment with a setup that is excellent and everybody listening to that same setup. If this is not the possible, the case is not conclusive and would only serve those listening to their own systems in that environment.
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Great subject... and I liked the Audio Myths Workshop as it confirms that we each have different perceptions of what we think makes a difference. For me I absolutely believe I can hear a difference between:
* Lossy 5.1 DD/DTS Vs TrueHD/DTS-MA tracks on blu-ray discs on my HTPC
* 128K MP3 Vs CD on my HTPC but NOT from a Portable Media Player with $2 Ear Phones
I also hear a difference (and prefer) the HD 24bit/96khz 5.1 mix Vs the same track from a CD source up-mixed by MC to 5.1.... but as pointed out from Matt/Jim this can easily be due to the additional effort that went into the mastering of this version rather than just an increase in Sample Rate and Bit Depth. As a consumer I don't actually care why the HD version sounds better and as a result I now have a predisposition for HD Audio tracks regardless as I've not heard one that sounds "worse"! That said, I believe in the law of diminishing returns, so while increases in Sample Rate and Bit Depth will make a measureable difference, the noticeable effect will continue to decrease exponentially (even if the cost to chase the diminishing return increase exponentially!).
Do I think I can hear a difference on my setup between exotic cables, interconnects, lossless formats, power conditioning etc....Nope, and since I tend to think many of them are expensive snake oil, I'll stick to this belief even though I've never ABX it to check it out. For me the classic AB test I saw in store was a Monster Cable AB Box proving their product produced a better picture, and it did.... but the scam was it was switching between the $200 HDMI Monster Cable carring 1080p VS a Composite Cable carring 576i...really...
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why suspecting things that everybody can experiment? all persons who listen to my installation constat differences when changing power cable or driver type of wasapi for example, the same kind of difference, but they taste and appreciate differently, sure, thisd is interesting. Why doubt of that, its not religious or conviction affair, nobody loose something, what interest is threatened? weird...
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I guess I do see much of the audiophile debate as a religious argument. I have no more chance of convincing you that the "AQVOX USB Low-Noise Power Supply" you purchased makes no discernible difference as you will convincing me that it does. I'm sure you do hear a difference, and I'm equally sure I will not. However, like all such debates I'm also happy to come over to your place, share a beer (or wine) and debate it's relative merits :). After a few wines I may even be convinced (or not) but I'm sure it would be enjoyable either way!
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Blind tests would be good. then again it will be needed in controlled environment with a setup that is excellent and everybody listening to that same setup. If this is not the possible, the case is not conclusive and would only serve those listening to their own systems in that environment.
Having "everybody" participate in one test isn't what I had in mind. I imagine a system that allows you to pick the software or media-related variable you'd like to test, and then conducts a minimum of 10 trials with the same listener to get a significant result. That result would be reported here, so we would all know whether that listener really heard a difference or not. This, of course, wouldn't help decide who's got the better hardware, cables or ears—but that's not what this forum is for.
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I guess I do see much of the audiophile debate as a religious argument. I have no more chance of convincing you that the "AQVOX USB Low-Noise Power Supply" you purchased makes no discernible difference as you will convincing me that it does. I'm sure you do hear a difference, and I'm equally sure I will not. However, like all such debates I'm also happy to come over to your place, share a beer (or wine) and debate it's relative merits :). After a few wines I may even be convinced (or not) but I'm sure it would be enjoyable either way!
Amen. Nicely said.
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ok for wine, but im sure youll hear différerence before, you are not the first to doubt. I'm scientific teacher and ive long time refused to admit power supplies coud change sounds... you speak of the usb power, its the exception: i use 5m long cable, the 5V of PC is very "dirty", and this voltage supply the usb receiver circuit of the DAC, so its possible to understand that a filtered flat supply just for the usb receiver can influence something. For the others supply or the drivers wasapi, interconnect cables etc... the explanation is maybe interesting, the more interesting is the different sounds they give, no need to be a specialist to hear; i promise.
Im in france, Burgundy, leave me choose the wine, you are welcome....
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* Lossy 5.1 DD/DTS Vs TrueHD/DTS-MA tracks on blu-ray discs on my HTPC
Interestingly, I learned at NABShow this past year that this effect is often because they actually master the TrueHD track differently than the DD track for many big-budget movies. Apparently the BluRay "consortium" is currently leaning on the studios pretty hard to do exactly this in order to create a public perception that BluRay is "needed" over DVD, because they still haven't seen the uptake of the new format that they expected (which was always a pipe-dream, but that's another story).
In many cases, both the video and the audio on a BluRay are mastered completely separately from the DVD edition, and intentionally changed to make them look and sound different on purpose.
The editor who was speaking at the lecture I attended actually said that in most of his films (which included ones you would have heard of), his preference was for the DD version. They only remastered the TrueHD tracks to appease the studio, and he just let his audio engineer do it alone. He said he chose to focus on the DD version of the audio because that would see much wider distribution, and the TrueHD track was only for a niche market. In other words, he figured that if the studio was going to make him make two different versions for marketing reasons, he was going to make the TrueHD one the "bad one" because hardly anyone would hear it comparatively. I'm sure most editors don't do this, but it was interesting to hear that they're being pushed to "differentiate" the BluRay versions artificially.
Back in the day, I read rumors that many of the same shenanigans were happening with the "high-def" CD formats they were trying to push at one time, but this was the first I'd heard confirmation from someone in the industry who would absolutely know.
So... If what this guy said was true outside of his experience, you absolutely should hear a difference, but because they are different on purpose, not necessarily because of the compression.
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Hi Glynor, I have very much been pulled into that with the CD VS HD Audio tracks where they have been clearly mastered and the same could be true on movie tracks if you selected between say DD, PCM, and TrueHD wich are all independant tracks. DTS-MA however is one track with a DTS Core so I "guess" it was a single mastering then coded accordingly and I clearly "perceive" a difference between these. One "fact" I would put forward is the a 5.1 DD Track only has 640kbps to play with all up (so similar to a low bit rate MP3) while a decoded PCM stream (from any of the HD sources of LPCM, TrueHD, DTS-MA) is 5-6Mbps.
It could still all be in my head, and a good example is the drive in the HTPC community to ensure we get a "full" 24-Bit / 96KHz when such tracks have been encoded. Now I'm note sure I can hear the difference between this and 16-Bit/48KHz but I certainly want to extract it as is without downmixing if I can!
Thanks
Nathan
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He didn't go into DTS at all and no one asked, but you may be right in those cases (but who knows, really). Shady business.
The goal is, of course, to get people to re-buy their DVD collections on BluRay. Reselling the same content over and over and over is a huge part of Hollywood's business model.
PS. I wish I could remember the guy's name.... I can't, and I don't remember which movies he cut, but I do remember that I'd seen 3-4 of the ones listed. I went back a few minutes ago and looked through my Evernote notes for NABShow, but I didn't take any notes at that one (it was generally not an extremely technical talk, more conceptual). Plus, my laptop was probably dead.
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....Shady business. The goal is, of course, to get people to re-buy their DVD collections on BluRay. Reselling the same content over and over and over is a huge part of Hollywood's business model.
Yup I could not agree more (Lucus is one of the worst for this + the games they play with titles like LOTR just annoy everyone ... or lead to the three R's ;) as people wait for the super final Extended Edition Directors Cut with added flavor version). Thankfully I really bought very few VHS / DVD (would have only a couple of Dozen) so have very few double or tripple dips over the many '00s of Blu/Red. I had decided early in the piece to hold off buying content due to the shift to HD (and given the law of diminishing returns I figure this will be as good as it gets for CE formats for some time, well unless 3D magically takes off - ha ha).
FYI - The growth of the use of DTS-HD is been staggering with over 50% of all Blu-ray discs every released now having DTS-HD http://www.blu-raystats.com/Stats/Stats.php. I really don't see many new releases with any other Audio Codec for the primary track (I wonder what deal they have done). It's a pity really for the HTPC world as this is the one codec not currently decoded by freeware filters (the Arcost TMT filter can be made to work perfectly however).
EDIT: More Blu-ray stats for those that care. Of the discs released this year:
* Audio: 80% of Blu-ray Releases this year are DTS-HD (10% True-HD, 5% LPCM, 5% DD)
* Video: 94% AVC (x264), 6% VC-1, 0% MPEG2
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FYI - The growth of the use of DTS-HD is been staggering with over 50% of all Blu-ray discs every released now having DTS-HD http://www.blu-raystats.com/Stats/Stats.php. I really don't see many new releases with any other Audio Codec for the primary track (I wonder what deal they have done).
It was mentioned in a thread at HomeTheaterShack back in November that DTS had signed a deal to have DTS-HD on all (or almost all) new releases going forward. In releases in 2011, TrueHD has only been on 17 titles (9.94%). This dropped to 7.35% for Feb and 5.26% (only 2 titles) for March.
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audiophile considerations are less imprtant when looking images in same time of listening. If the sound is good in presence and frequency range its OK. The subject here is the 3dimensionnal feeling and timber precisions etc.. that can be perceived in "exclusive" listening act.
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Here is an article to Glynor's point http://www.hemagazine.com/node/Dolby_TrueHD_DTS-MA_versus_Uncompressed_PCM
Conclusions as follows
So Subtle
What impressed, or perhaps surprised, me most about these tests was how good the base codecs actually are. The difference between the original audio and the basic Dolby Digital and DTS is a lot subtler than you’d expect, given the extreme amount of compression (around 10:1, a similar ratio to that of 128 kbps MP3).
That said, I could definitely pick out the difference between the lesser (or perhaps it’s more accurate to say “better”) compressed versions and the higher compressed versions. The difference is mostly in the presence, or ambience. The lossless, Dolby Digital Plus, and DTS-HD High Resolution compressed tracks were just a little more open and airy. I hate to say it, but they just sounded more realistic and transparent. The 448 kbps Dolby Digital and standard DTS tracks were less so, a little more closed off. Between the 640 kbps Dolby Digital and the uncompressed, the difference was even less noticeable. Enough so that most people, even those trained to listen for it, probably won’t be able to hear the difference.
The core DTS call is a little harder, as there wasn’t the same blind system in place to A/B as precisely as at Dolby. Results were similar, though.So by all means go for the new codecs, as they definitely sound better than what was on DVD. Uncompressed PCM, on the other hand, is just a waste of space (though compatible with everything).
If you’ve been listening at home and are sure you can hear a difference on your favorite discs, be wary. There is absolutely no way to tell that compressed and uncompressed tracks on any disc have anything to do with each other. They could come from different masters, they could be mixed differently, or any number of other variables that makes an in-home test, unfortunately, impossible. That said, trust your ears, and go with the one that sounds best to you. –Geoffrey Morrison
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It was mentioned in a thread at HomeTheaterShack back in November that DTS had signed a deal to have DTS-HD on all (or almost all) new releases going forward. In releases in 2011, TrueHD has only been on 17 titles (9.94%). This dropped to 7.35% for Feb and 5.26% (only 2 titles) for March.
I'm not surprised at all. DTS had a massive presence on the floor at NABShow last year, and I heard that they did again at later shows. Besides, I've always preferred DTS processing when expanding stereo. Always much less harsh.
BTW, I know most people didn't interpret it this way, but to be blatantly clear, I wasn't saying that DD compression doesn't have an audible impact (it certainly does, it is fairly heavy compression using an old technique). I was just making the point that what is "right" is entirely dependent on the sources involved. Things aren't always as simple as they seem, and many times, the industry is actively trying to trick the consumer.
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after experiments, i have to admit the wav files sound different, principally are more "soft" in my config. Ive changed the HP cables (crimson electronics), differences are more relevant.
ive began to convert some flac files to wav. It appears that wav files dont support tags.
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Tag and Rename can tag the file (for ex. Track Number). After that if i Re-open the wav file with Tag and Rename, it can display the tag whiwh has be written before, but not in jriver.
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You would need to enable WAVE tag writing:
http://wiki.jriver.com/index.php/WAV_%26_AIFF_Tagging
WAVE tag reading is always enabled so probably Tag & Rename uses a different tag format. MC's WAVE tagging format is compatible at least with dBpoweramp and the now discontinued Musicmatch player (if someone is searching a new program that would support Musicmatch tags).
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thank you, it works, tags of tag and rename appears in jriver;
How to do to convert flac in wav format with tags of flac files copied in the wav files?
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thank you, it works, tags of tag and rename appears in jriver;
How to do to convert flac in wav format with tags of flac files copied in the wav files?
Use dBpoweramp. was the only program that converts flac to wav and retains original ID tags. It has lots of functionality and very useful program.
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Use dBpoweramp. was the only program that converts flac to wav and retains original ID tags. It has lots of functionality and very useful program.
MC will do the same if you enable WAV tagging in the plug-in. WAV tagging may break playback in other programs.
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MC will do the same if you enable WAV tagging in the plug-in. WAV tagging may break playback in other programs.
Ah good to know!
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It appears that wav files dont support tags.
It is just turned off by default in order to leave the wav files "pristine".
Tools > Plug-in Manager... > Input > WAVE & AIFF Plugin > Configure button > Check the checkbox.