INTERACT FORUM
More => Old Versions => Media Center 11 (Development Ended) => Topic started by: Von on April 09, 2006, 10:43:21 am
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Is there a way to make Media Center buffer the entire file before playback? I play mostly lossless files, and every few seconds my pc reads from the hard drive, which makes just enough noise to bother me when I listen to music at a low volume. With file sizes around 20-40 Mb, I don't think this buffering should be a problem.
JimH replied to my question, saying that newer drives are more silent. Fair enough, but...
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I second this request for just another reason.
Scrolling through a file isn't smooth as it could be. MC has to buffer new portions of the file every few seconds.
Using winamp (where you can buffer 99999 kilobytes) there are no hickups while scrolling the file.
I would appreciate this feature in MC!
Fred
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I've come to a conclusion over time, that most drives start off almost
silent (Got almost every manufacturer here Samsung, Hitachi, WD, Maxtor), but after 6months of use, they're anything but....
I guess the fluid dynamic ball bearings or what ever they use stops being so fluidic...
Most people can't hear it due to fan noise, or ambient noise, but once you've eliminated them....
you realise just how loud they are. I've tried those mounting bars which
are supposed to reduce vibrations etc, but tbh it make no difference at all.
My current soloution involves an external drive on a very long USB cable
(Which introduces it's own problems).
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so how about that file buffering ? sure would save some battery life off of a laptop :)
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We'll consider it in a future version. Under normal circumstances, it shouldn't make any difference.
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Has there been a complete file buffer option considered for the new verison?
There is a growing number of people who are finding that putting the whole file in memory improves playback and can improve the sonics of lossless files.
All you have to do to test is install a Ram drive utility and put files you want to test onto that Ram drive and compare them to files played off the hard drive.
Files from a Ram drive play smoother and fast forward seek etc. works better. They sound smoother and more laid back too, with less grain.
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The sound will be identical regardless of whether the file is in memory or on disk. It's the same input bits either way. And MC is always bit-perfect.
You can force MC to use a bigger memory buffer, although it may not provide the results you expect. MC already intelligently adapts its buffering based on the speed of the source and the speed of the machine.
Here's the registry key you can play with, but again we recommend using the default 6000 (6 seconds) for most all uses:
HKEY_CURRENT_USER\Software\JRiver\Media Center 12\Player Core\Secondary Buffer Minimum MS
Thanks.
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thanks Matt ! i put it at 54000 just to make sure it buffers everything.. even the albums i downloaded off of tranceaddict ;D
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I played with the buffer and found that a larger buffer did improve the sound. But too large a buffer and the player began to have hiccups around cueing and playing tracks. I found that more than 30 minutes or 30000 gives it problems. The actual amount might well be system dependent.
If you look at the amount of RAM used in your system, when you add to the buffer it goes up substancially when a file is playing. It goes up by about 200 MB when I go over about 3 minutes of buffer, and stays there no matter how much larger I make the buffer, so there is some internal limit happening here.
Also. be sure you have enough ram in your system to handle MC using another 200MB without putting your system into virtual memory swapping. I would think that this would only provide a boost in sound quality to a system that has 1 gig of memory of more, and has virtual memory turned off.
If anybody else tries this and notices a difference in sound quality please post! Also if you can put some music files into a ram drive (a virtural drive created inside your extra memory) and give a listen, see if you hear a difference. I did, and it's a nice improvement. BTW, there is a Foobar plugin that does this, and it improves the SQ of Foobar too, so this is not really so far off base.
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i did full file buffering with foobar and noticed an improvement, no doubt moreso with our beloved jrmc ;D . how large a buffer did you start using ?
if i have 560000 ms of secondary buffer, when i pause what i'm playing the timer and slider just keep on going regardless.
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say.. Tuckers, what equipment are you using ? i noticed the change even with my 10$ 10 - year old jvc's (audio technica headphones broke last night).. definitely noticed it with my speaker system ;D
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Hi there.
I've been using full file buffering with foobar before. And it proved in many listening tests to be better
than playback without buffer.
I played a bit with the proposed buffer size in the registry, changed it from 6000 to 420.000.
Doesn't matter what I am entering, if you look at the allocated RAM size of the application via TaskManager
it does not show any extensive buffering. The allocatd RAM size should be in the range of 60MB, when loading a
6min .wav track.
The TaskManager just shows 17 to 20MB allocation.
Perhaps somebody can explain this behaviour.
I could imagine that the buffer size doesn't really matter, since it is just a buffer. It will not prevent from further data-streaming into the buffer. That's obviuos if you look at the accesses to the HD.
It would be nice feature to have have full file buffering available, not only for playback, its also a great feature which for writing CDs.
A nice test: Get youself a RAMDISK. www.ramdisk.tk and than give it a try!
\Klaus
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I think you have to do some tests to get the most buffer you can in your system before it starts acting up. In mine its somewhere around 30000.
I have a high-end studio quality system, so even small differences are readily apparent.
When I increase the buffer, the amount of ram does not incease until I play a song, then it jumps up about 175 MB. I am not getting the 50 - 60 MB increase, but a 175 MB increase. It could be the way MC is dynamically adjusting the bufffer to suit your system.
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define high end ;D
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http://www4.head-fi.org/forums/showthread.php?t=89382
and they didn't believe me ::)
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Hi Tuckers.
175MB that's a lot. How big is the file?
By the way. I'd call my setup also High-End. I guess we are talking about the same subject.
A bit of background: I am running an ASIO driver, with a USB DAC connected. (asio-driver from usb-audio.de!)
For all folks running USB-soundcards - try above mentioned driver! It replaces the sound deterioating MS USB Audio-Driver.
Just played a bit more with JRiver.
Even with the 6000 buffer it plays better than foobar -- the long time reference player.
Distortions are going down by large margin.
I caught hick-ups when playing tracks at buffer-values of 420000. The sound did not improve very much.
The real way forward (for now): Try the RAMdisk I refered to. I got 2 GB RAM on my machine, got myself the Qsoft enterprise demo version to be able to configure a 1GB RAM-DISK. On my machine it takes 12s to copy a whole CD to the RAM drive.
I never ever before had such a cristal clear sound on my system. Even Foobar with full file buffering on is far off this setup.
JRiver coded a brilliant audio-engine. I never had a better player on my PC.
Perhaps a nice RAMDISK plugin could be a nice future product.
\Klaus
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It doesn't matter how big the file is, the it uses the same amount of Ram.
Yes MC is the best sounding player out there. I hope they value the listening experience of it's users and make it even better!
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It doesn't matter how big the file is, the it uses the same amount of Ram.
175MB looks to me like a 2o min.wav. I don't get it. ?
and they didn't believe me ::)
Its always the same. Try to start a discussion about sound quality at hydrogene. They'll stone you. Or your post is taken out.
And that's exactly the reason, why others like JR are taking the lead!
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Well, I think putting in a large number in the secondary buffer simply causes the buffer to allocate the maximum of memory it has available. So the memory does not change. I tried a two minute file and a 60 minute file - no difference.
I'm using FLAC, what format are you using? Could this be a factor?
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I am running wav-tracks only. Flac could be an issue. Perhaps some extra space is needed for converting the FLAC - God knows.
But it could also depend on the PC and OS itself and its RAM allocation. I learned that when setting up the RAM disk.
The OS has to let you allocating a big chunck of contingous RAM. Depending in which area of RAM you're
fishing, you could possibly face some restrictions.
Isn't there a developer around, stopping me on my wild guesses? ::)
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I tested playing the same file in wav or flac formats and they both use the same amount of RAM, about 175 MB.
How much total memory do you have and how much is used? I have a gig, but am upgrading to 2 as soon as I get the memory in.
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The sound will be identical regardless of whether the file is in memory or on disk. It's the same input bits either way. And MC is always bit-perfect.
My thoughts as well, making me wonder whether this "better" sound is actually being output or is it a feeling.
The only thing better i can see is faster response times.
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Instead of drawing an abstract conclusion, give it a listen for yourself.
It's pretty easy to set up a test.
Then you can say it's all in our heads with the authority of personal experience.
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My thoughts as well, making me wonder whether this "better" sound is actually being output or is it a feeling.
The only thing better i can see is faster response times.
I'd recommend to read this: http://www.rme-audio.com/english/techinfo/lola.htm you can find a lot of interesting stuff over there.
Further have a look at musicxp.net to optimize the PC.
Bit-Perfect does not mean Sound-Perfect!! All kind of processes, task priorities, hardware drivers and hardware can add more latency to the signal. This is causing more or less distortion at the ouput, since we are talking of a realtime audio bit-stream.
Further this audiostream sees many different clocks on its way. Perhaps you know how much effort studios put into their systems to have one master clock for all gear.
By having the file in stored in the RAM, you got rid of quite some latency affecting sources.
If you would have a real good soundcard or external DAC with a buffer and a well done precisioin reclocker, the latency impact on playback should not be that big.
I tested playing the same file in wav or flac formats and they both use the same amount of RAM, about 175 MB.
How much total memory do you have and how much is used? I have a gig, but am upgrading to 2 as soon as I get the memory in.
I got 2gig. 1gig is reserved for the RAM disk.
Did you notice a slight difference between FLAC and WAV? A very trustful source mentioned that FLAC sounds slightly worse
than WAV, even though its lossless.
I guess its again the discussed latency effect, when converting FLAC to PCM more CPU power is required. Since its done
in realtime I'd guess there is some truth to it.
One day I'll try.
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I'd like to keep this thread on track, so PM or email me to discuss the issues of Flac and Wav etc.
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Tuckers, please let me know when you get your additional 1gb if it fixed the stutter. thanks ;D
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Further have a look at musicxp.net to optimize the PC.
I did, it had one tip i had not heard of, selecting background services for processor scheduling instead of programs (http://musicxp.net/tip_detail.php?tip_number=1). Testing to see if it makes any noticeable diff.
Bit-Perfect does not mean Sound-Perfect!! All kind of processes, task priorities, hardware drivers and hardware can add more latency to the signal. This is causing more or less distortion at the ouput, since we are talking of a realtime audio bit-stream.
Further this audiostream sees many different clocks on its way. Perhaps you know how much effort studios put into their systems to have one master clock for all gear.
By having the file in stored in the RAM, you got rid of quite some latency affecting sources.
I can appreciate a studio doing this if they have many sources and want things to be in sync, but for a single user using one app !!!
Either the stream starts a little early or a little late, but when it starts, its coming at a constant rate. One stream.
Reading this (http://www.rme-audio.com/english/techinfo/lola_latec.htm) leads me to think its more an issue if you have multiple sources generating sound at random times rather than one.
"The phenomenon of varying latency comes up with the use of software synthesizers and samplers, i. e. sound generation triggered from outside the system."
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i think i got rid of the stutter, playing around in system of control panel led me to select background services instead of programs in processor scheduling. but this is with the digital out..
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Either the stream starts a little early or a little late, but when it starts, its coming at a constant rate. One stream.
My stream starts from RAM for the time being. That's closer as starting it from HD. I can't get closer to the DAC for
now.
Another thing regarding latencies:
Did you see at RME the measured latency added by W98 compared to W2K. I don't want to look it up again.
But I think it was 23ms compared to 3ms. Something in that range.
Did you see what latency was added by WS_FTP running in the background also mentioned somewhere on the RME page.
http://www.rme-audio.com/english/techinfo/hdsp_notetune.htm it was 12ms and generating nice clicks from time to time.
I think users who intend to use the PC as high-grade audio source must look out for all these little tweaks to squezze the
most out of it.
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I can't completely follow this thread, but try a google search for:
via chipset latency setting
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I can't completely follow this thread
::) ;D
They must have far, far, far better ears than you or I, Jim.
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dual cpu's help a lot, I've had dual systems exclusively since the P3 came out and hate working on single cpu systems - they suck big time
and now there is the benefit of dual core cpu's the cost has come down a heck of a lot!!
each card/device in your system works on a system of interrupts - kind of like a doorbell ringing while your cooking dinner, the interruption may make something burn...
with a dual system, one can answer the door while the other continues to cook
same with pc hardware - a single cpu can ony deal with one thing at a time, if it's busy doing somethinge else that can't be interrupted when the sound card needs more data you'll get a variance in the sound output
creative soundcards are notorious for this problem
slightly older pc's that use pci and not pci-express can also hit a problem with the 133MB barrior
on these pc's the total amount of data being moved about between devices is limited to 133MB per sec and the latency can be pretty bad when doing simple things like copying files etc
since upgrading to a new motherboard/cpu etc I've been in wonderland - there's nothing I can throw at it that causes a problem
simultanously copying 6 files from one place to another on the same hard drive whilst playing videos/surfing/copying files - no problem, the only limiting factor is the speed of the hard drives to keep up
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::) ;D
They must have far, far, far better ears than you or I, Jim.
I can't completely follow this thread, but try a google search for:
via chipset latency setting
Could it be that your audio-engine sounds that great by coincidence?
And you didn't even know about it, because you never heard it performing on a High-End audio setup? ;D
Try the RAMdisk I mentioned earlier. It doesn't cost anything and it'll take just 30min. I hope your audio rig is good enough that you can hear what we are talking about.
By the way did anybody play with the thread-priorities in the task-manager for MC, e.g. setting it to Realtime while playing?
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soundcheck, how do i do that ? is that in task manager ?
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hmm.. while we're on the subject of tweaks, go to sounds (that's what it is in vista), make the soundcard you are using as default then check the use only default devices checkbox. let's see if you notice that. (btw, did that on a laptop with stock speakers, there was an effect).
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looking at task manager to see the processes, media center seems to have a dynamic buffer. it goes up all the way to 175-200mb then goes back down to 8mb.
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soundcheck, how do i do that ? is that in task manager ?
Go to Task-Manager Processes. Look for the media-center process under processes. Right-Click on the the media-center process opens a menu. There you'll find the priorities menu item with 6 priorities to choose from.
Switching to Realtime gives you a warning. Just ignore it.
I also realized on some application that choosing the lowest priority work. I heard that would give the background tasks e.g ASIO driver higher priorities then e.g. the front-end application graphics. For pure audio you do not need graphics accelaration.
Just give it a try and listen.
hmm.. while we're on the subject of tweaks, go to sounds (that's what it is in vista), make the soundcard you are using as default then check the use only default devices checkbox. let's see if you notice that. (btw, did that on a laptop with stock speakers, there was an effect).
I configured a specific hardware profile for audio. (I am using an external USB-DAC). I switched off all devices, which are not needed in my audio setup such as on-board audio, network-adapter, wireless-lan even CDROM,......
At the same time it saves a lot of battery-power. And frees up processing time.
By running ASIO or Kernelstreaming you are passing the Windows Kernel Mixer by. Your above mentioned setting should not have
an impact on these kind of setups. Getting rid of the Windows kernel mixer is the very first thing to be done anyhow!
The tweaking discussion is mainly of topic - I do agree. Still - perhaps it helps to understand that the PC needs quite some adjustments to sound best.
The buffer/RAMDisk issue I still regard as a major tweak for high-end audio playback.
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oh.. you never know what goes on in windows ;D
added: say.. there might be something wrong with media center cause i am using an old version (.1.191) and it's not stuttering... not only that, the memory usage is not 175mb. it's 17mb.
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Please try re-downloading and re-installing the latest.
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oh.. you never know what goes on in windows ;D
added: say.. there might be something wrong with media center cause i am using an old version (.1.191) and it's not stuttering... not only that, the memory usage is not 175mb. it's 17mb.
See. This is what I mentioned before. I havn't seen a larger value than 17-20MB even with 420000 in the registry. Tuckers came up with the 175MB. I need to check it out again.
Please try re-downloading and re-installing the latest.
My download is 3days old. Is there a newer release than 11.1.196?
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See. This is what I mentioned before. I havn't seen a larger value than 17-20MB even with 420000 in the registry. Tuckers came up with the 175MB. I need to check it out again.
My download is 3days old. Is there a newer release than 11.1.196?
199 is the first post on this board.
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The newest builds can always be found in a sticky post at the top of this board. The "auto-update" system in MC typically lags behind the newest build by a few builds (so that they can make sure it is really, really stable before they widely distribute it).
I've really never had a serious problem with a new build that couldn't be cured by reverting to an older one. If there ever is a serious problem, they're really good about pulling them down before they can cause any havok.
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199 is the first post on this board.
The newest builds can always be found in a sticky post at the top of this board. The "auto-update" system in MC typically lags behind the newest build by a few builds (so that they can make sure it is really, really stable before they widely distribute it).
I've really never had a serious problem with a new build that couldn't be cured by reverting to an older one. If there ever is a serious problem, they're really good about pulling them down before they can cause any havok.
Just installed it. I am learning fast. ;) THX.
A bit Off topic again: Here is an interesting link on the latency jitter subject:
http://www.rme-audio.com/english/techinfo/lola_latec.htm
I doubt that MS drivers, like the usb-audio driver are programmed accordingly.
That's why it sounds pretty poor. This is what the guys at www.usb-audio.de with their USB-ASIO driver improved I guess.
I hope above article helps not to put latency-jitter into the "Voodoo-corner".
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So - from buffer perspective nothing changed on 11.1.199 doesn't matter what I am entering in the registry RAM allocation never exceeds approx.
20MB.
One more Off topic -- Audio Tweak Finding (Perhaps we should open an Audio Playback Tweak post instead!-- That can be done later!)
Anyhow.
Finding:
The DSP Studio shows on option for "output format". If I tag it the sound is cristal clear. If I untag it the sound gets distorted and very close to foobar playback quality.
I am running 16bit 2channels 44,1 and ASIO.
Does that mean that the original data are somehow treated with the DSP?
Any ideas? Any confirmations?
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16 bit just sounds better than 24 bit. even with foobar ;D . of course foobar's playback at 16bit won't beat jrmc's at 24...
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16 bit just sounds better than 24 bit. even with foobar ;D . of course foobar's playback at 16bit won't beat jrmc's at 24...
I was wondering if the orignal data are somehow manipulated with the DSP. If this is the case, somebody must have coded
a real nice algorithm. At least the way it looks/sounds on the first glance.
On the other hand - I am always suspicous about PC based DSPs for high-end audio. I got to do some more listening tests tonight.
Perhaps somebody (JimH!?) can explain what's happening when turning the DSP output format option on. Why does the perceived sound quality improve? Do you start a sample rate conversion routine, even on 44,1khz?
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The sound will be identical regardless of whether the file is in memory or on disk. It's the same input bits either way. And MC is always bit-perfect.
You can force MC to use a bigger memory buffer, although it may not provide the results you expect. MC already intelligently adapts its buffering based on the speed of the source and the speed of the machine.
Here's the registry key you can play with, but again we recommend using the default 6000 (6 seconds) for most all uses:
HKEY_CURRENT_USER\Software\JRiver\Media Center 12\Player Core\Secondary Buffer Minimum MS
Matt and Jim,
I'm not a believer in Ram Disk tweaks and such but I did try out setting the the Secondary Buffer MS value to 12000. It had a useful result for me.
I've been using MC 11 on a P4 2.8GHZ laptop with integrated graphics sharing the 512 MB of RAM. When I played music with MC 11 while running Thunderbird and Firefox 1.5x, I sometimes heard small glitches when Firefox is loading a new page, scrolling the existing page or reloading a page. I could produce the behavior pretty reliably. When I hooked up a 19" LCD running at 1280 by 1024 (instead of the laptop's 1024 by 768 pixels), the glitches were louder and more obvious.
After changing the Secondary Buffer size, I haven't heard any more glitches. At this point, my observation is just an observation but it gives me another tool for eliminating glitches.
Win XP isn't a friendly environment for a soft real-time application like music playback. I'm trying to get to high quality computer based playback that I can use instead of CD playback. I appreciate this help and I would appreciate more tips.
Bill
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The DSP Studio shows on option for "output format". If I tag it the sound is cristal clear. If I untag it the sound gets distorted and very close to foobar playback quality.
I am running 16bit 2channels 44,1 and ASIO.
Does that mean that the original data are somehow treated with the DSP?
Any ideas? Any confirmations?
When you enable the Output Format processing, check the value of the "Overflow Handling" dropdown box near the bottom right of the DSP Studio. I believe the default value is "Clip Protection" and I believe this is the minimum amount of processing introduced as long as you've left the upper choices at the defaults of "Source XXXXX...".
Far too many modern CDs are mastered to "Sound Loud" which along with reducing the dynamic range of the music, any peaks that do remain often end up "clipped" resulting in distortion. Try switching the Overflow Handling on a variety of material and see if you can hear any differences.
A somewhat classic article on the subject:
http://www.webculture.net/prorec/articles.nsf/articles/8A133F52D0FD71AB86256C2E005DAF1C
More info and additional links than you'll probably ever want... ;D
http://en.wikipedia.org/wiki/Loudness_war
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yeah that would be great but i don't listen to pop.. ;D
say soundcheck, can you CHECK if .200 sounds better than .199 ? i'm getting paranoid here. ;D
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say soundcheck, can you CHECK if .200 sounds better than .199 ? i'm getting paranoid here. ;D
The paranoid stage I entered apparently by the time I started reviewing SW-Players a while ago! So - You are not alone. ;D
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Hi soundcheck,
I'm afraid I may have been mistaken in my reply to you on the effect of enabling the "Output Format" DSP. For some reason I was always under the impression that the "Overflow Handling" was not in use until you enabled one of the DSP functions. Upon closer inspection it appears that "Overflow Handling" is on full time, so it is unlikely that it is the reason for less distortion when you enable "Output Format" in DSP Studio. ::)
I downloaded the ramdisk you recommended and will give it a try this weekend. I also intend to ABX it, ramdisk vs. HDD. ABX software is free and easy to setup if you care to try it yourself.
Available here:
http://www.pcabx.com/training/getting_started.htm
Have Fun!
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I'm afraid I may have been mistaken in my reply to you on the effect of enabling the "Output Format" DSP. For some reason I was always under the impression that the "Overflow Handling" was not in use until you enabled one of the DSP functions. Upon closer inspection it appears that "Overflow Handling" is on full time, so it is unlikely that it is the reason for less distortion when you enable "Output Format" in DSP Studio. ::)
I downloaded the ramdisk you recommended and will give it a try this weekend. I also intend to ABX it, ramdisk vs. HDD. ABX software is free and easy to setup if you care to try it yourself.
Available here:
http://www.pcabx.com/training/getting_started.htm
The clipping protection you just need if you play with the preamp, equalizer and automatic replay gain.
If you play the tracks pure as they are, you can forget the clipping protection. That's the way how I see and use it.
So just turn it to "NONE" if you're not playing with above mentioned features.
By the way did you notice the sound difference when, tagging the "output format" box? It can't be just me. ?
I'll have a look at ABX. THX for the hint.
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The DSP Studio shows on option for "output format". If I tag it the sound is cristal clear. If I untag it the sound gets distorted and very close to foobar playback quality.
I am running 16bit 2channels 44,1 and ASIO.
Does that mean that the original data are somehow treated with the DSP?
Any ideas? Any confirmations?
The clipping protection you just need if you play with the preamp, equalizer and automatic replay gain.
If you play the tracks pure as they are, you can forget the clipping protection. That's the way how I see and use it.
So just turn it to "NONE" if you're not playing with above mentioned features.
By the way did you notice the sound difference when, tagging the "output format" box? It can't be just me. ?
If the source file is 2-channel/16/44.1 and only the 2-channel/16/44.1 DSP options are selected the output is not altered. That can be verified by bit comparing Disk Writer output files.
Also, if all other DSP options are disabled the plain clip protection does not change the output. It kicks in only if the used DSP settings would make the file clip.
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soundcheck,
I will give a closer listen when en/disabling output format.
You really owe it to yourself to setup the ABX test, it can be very revealing ;)
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I will give a closer listen when en/disabling output format.
It's needless to ABX that. If the source format and the selected output format are the same (e.g. 2-ch/16-bit/44.1 kHz) the actual output is exactly bit to bit identical in both cases. I have tested this with a bit compare tool.
MC's Disk Writer is a nice tool for making test files with other DSP settings. The resulting wave files can be blind tested for example with ABC/HR (http://ff123.net/abchr/abchr.html) or Java ABC/HR (http://www.rarewares.org/files/others/abchr-java-0.5b.zip). These are more comprehensive than the plain PCABX.
Here is tutorial for ABC/HR: http://ff123.net/64test/practice.html
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Hi Alex,
Thanks for the info and the links. Since the ramdisk is thought to sound better by some, my main intent is to simply encourage an ABX test between playback with the ramdisk vs. hdd as an alternative to relying on (organic) memory alone.
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It's needless to ABX that. If the source format and the selected output format are the same (e.g. 2-ch/16-bit/44.1 kHz) the actual output is exactly bit to bit identical in both cases. I have tested this with a bit compare tool.
Again " Bit Identical" does not necessarily mean "Sound Identical". As soon as you are catching latency-jitter, you got an issue in the time domain for realtime applications, even if the bits are correct. And this is audible!
Refering back to the ABXing. One should go for complex, well recorded material like a nice orchestral track (e.g. from Chesky records, OPUS3 records in XRCD quality to name my personal references) for ABXing. Otherwise it is rather difficult to hear the latency impact easily.
If you own a real good stereo you'll hear easliy reverberations of the room, where the recording was taken. Yes - You shouldn't need a DSP
faking Church or Jazz Club reverberation. It's usually already on the life recording. ;) If you can hear it well it is a good sign that your stereo is able
to reproduce micro-details of the recording. Again- Don't forget to switch of the Church DSP, you'd just fool yourself. ;D
The good thing though. If you don't know what's on the recording, it won't bother you that you can't hear it. ;D
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Again " Bit Identical" does not necessarily mean "Sound Identical". As soon as you are catching latency-jitter, you got an issue in the time domain for realtime applications, even if the bits are correct. And this is audible!
so true.. yet hard to prove :(
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Again " Bit Identical" does not necessarily mean "Sound Identical". As soon as you are catching latency-jitter, you got an issue in the time domain for realtime applications, even if the bits are correct. And this is audible!
Perhaps, but that does not apply in this case. AFAIK, the DSP part is simply bypassed when the selected options do not differ from the source format so all processing inside MC is identical (= non-existent). The next audio component after MC gets exactly the same PCM stream in both cases.
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oh ok. thought you were talking about file buffering..
anyways, before, when i'd play flac files (they're 16/44.1) with or without output format checked i'd still get the same output in my stereo system.
now.. since i'm on ogg (aotuv), since lossy has no bitdepth, i have to select 16bit or i'd get grainy highs.. (same as output to 32bit)
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MC's Vorbis decoder outputs PCM stream in 16-bit integer. So you may be experiencing the effect of DSP processing from 16-bit to a higher bit depth.
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On second thought, I'd guess you are merely experiencing the effect of Windows Kernel or the sound card driver reducing the bit depth from a higher value to 16-bit.
AFAIK, you should not be able to hear the effect of MC's DSP when going from 16 to a higher bit depth if no other DSP options are selected and the sound card supports the selected bit depth. I cannot, not even with high end headphones.
Also, if your sound card does not natively support the 44.1 kHz sample rate the output will be resampled to 48 kHz after MC. In that case the best sample rate option in DSP Studio would probably be 48 kHz. MC has a high quality SW resampler. It produces better quality than Windows Kernel resampling.
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thanks but i'm using a DMX 6fire (based on the Envy24 chipset) that does not resample. and then i'm running windows vista rc1 with ASIO... ;D
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So we have the same soundcard. I have a Terratec DMX 6fire 24/96 on my HTPC.
Vista's audio handling is a completely new factor. I have read about it, but I have not seen any independent test reviews or tried it personally. Can an old ASIO driver still work properly with it? Can a driver bypass all the new processing that Vista supposedly does?
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Alex B, i'm sorry but you're right, i tried it and it sounds exactly the same.. and come to think of it.. how could it output a 32 bit signal to my dac since the dip is only 24/44.1. i really don't know what it was that i heard before... apologies ;D
i'm using xp drivers for the 6fire in rc1's x86 version so i guess that explains why it doesn't do the processing you mentioned.. my onboard on the other hand with vista drivers.. all this weird stuff about being able to set output format (dolby ?) and sampling rate..
could you post a link about all that extra processing vista does ? i'd like to read it..
thanks ;D
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Here ya go... 8)
http://arstechnica.com/news.ars/post/20060907-7682.html
http://www.avsforum.com/avs-vb/showthread.php?t=713073
http://blogs.technet.com/windowsvista/articles/450038.aspx
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many thanks ! interesting read ;D
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Here ya go... 8)
http://arstechnica.com/news.ars/post/20060907-7682.html
http://www.avsforum.com/avs-vb/showthread.php?t=713073
http://blogs.technet.com/windowsvista/articles/450038.aspx
Hi MerlinWerks.
Great AVSforum thread, thx.
As far as I understood MS still gives MC space for soundimprovement on Vista.
Will MC use in the future , I think they called it, "Exclusive Mode" to replace e.g. ASIO?
The MS audio tech specs of Vista are quite limited and far away from being audiophile.
Will MC do something about it?
I was a bit disappointed that the MS guy did not jump on the Jitter issue somebody mentioned.
They were just hammering on the "Bit Perfection" issue all the time.
As I mentioned more than once, Latency-Jitter is to me the well hidden enemy of audiophile audio from PC.
I don't have a clue, why this issue is not really being discussed.
Further, a comment was made that streaming from HD, while audio-playback is going on, would not
be the best setting for audiophile PC-audio. Pretty much confirming that what I figured when introducing my RAM-disc. 8)
Just to mention it- i mean the ramdisc-thing- , since this thread, if I recall it right, is about file buffering. ;D
I am wondering if Vista at all addresses file buffering.
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fwiw, after reading thru this thread (and checking my skepticism at the door), I modified the "PlayerCore Min MS" setting to 12000 and I noticed a couple of things:
- I too could get the occasional "click" when browsing with FF and playing music. I no longer get this
- Task Manager would always cause the same "click", now it does not anymore
- I notice no difference in audio "quality", it's still as great as ever.
Two other things I was able to do as well. I set the latency switch on the RME Digi96 to it's lowest setting and I was able to set the MC setting on playback to 1 sec.
Computer setup:
XPSP2, Celeron D, overclocked to 2.86GHz, 512MB RAM, RME Digi96/8 PST using their lastest driver, 2.11 (ASIO)
I run a low priority Folding At Home session, SpeedFan and CachemanXP. MC runs without visualizations and I do not use the DSP or any "PN" plugins. MC usually peaks about about 5% and hovers at 2-3% in task manager. My CPU is always pegged at 100% and runs about 45C.
Audio Setup:
(Transcendent Sound Balanced Power Supply) => RME => Denon DVD5000 DAC => TS GG Pre => TS T8-LN monoblocks => Hammer Dynamics Super12s
Also, I've tried Vista a few times on this machine and the RME drivers appear to work fine.
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Also, I've tried Vista a few times on this machine and the RME drivers appear to work fine.
Cool 8)
Good to know, I have the DIGI96/8 PAD. I may be giving Vista a look (on a spare box) earlier than I thought I would ;D
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I know Arnies ABX comparator tool. We found it very useful when evaluating codecs and rip quality. It's a very good sounding player too.
I did give a listen to some files played from RAM and from Hard drive with this it, but I could not hear a difference in a non-blind series of tests, so I also obviously could not get anything beyond random on the ABX either.
It's a completely different piece of software and handles files completely differently too. So I don't think it can be used to make a valid listening test for this issue.
I will try the ABX in Foobar, but fro some reason I can't find the required dll foo_abx.dll for 0.83, which is my preferred version of the program. I might yield a more significant result. Anybody know where I can find this dll? Most of the 0.83 stuff is gone from the web.
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Bit-Perfect does not mean Sound-Perfect!! All kind of processes, task priorities, hardware drivers and hardware can add more latency to the signal. This is causing more or less distortion at the ouput, since we are talking of a realtime audio bit-stream.
I'm not sure where you take that info that latency causes distortion... When you play audio files there is way enough buffer for the system to counter balance any kind of task switching. The page on the RME website refers to real time playback, on pro systems where you can't afford to have large buffers (1 ms buffer at best). MC has about 6000 times this buffer size by default so I'm not sure if any of this info does really apply.
At worst, if you had some buffering/task switching issues, you would hear some crunching not distortion. Btw, 6 second buffer is *not* real time.
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I'm not sure where you take that info that latency causes distortion
I am not saying latency is causing distortion. Latency is just another word for delay and that's what's all about it. For playback it is not a real issue as long as the latency is kind of linear.
What I am saying is that non-linear latency-jitter or jitter is causing distortion, if it is induced to the data-stream which goes (realtime) right into the DAC-chip.
The RME side explains very well the differences.
If your soundcard and its drivers are able to cope with this non-linear-latency-jitter-in-the-datastream fact, you shouldn't have a big problem, as long as your soundcard does a real great job.
I know that some high-end soundcards/DACs do some buffering and reclocking before the stream leaves towards the DAC-chip to avoid these kind of Latency-Jitter or Jitter problems.
The major source for latency jitter is the interrupt handling. And you can't get around it.
This is by the way the reason why UNIX systems are the prefered systems when talking about real-time applications.
Example: Let say you got an USB soundcard. Playback and datastream is running towards your soundcard. Suddenly "Interrupt" pops-up saying - "Hold on - I have to read the data from HD first" to fill the 6000ms buffer. This one has higher priortity than USB. Next One. Graphic Card is saying: Stop if have to refresh the GUI.
And there are at least 40 other processes running from which a lot of processes have higher prority than the USB-bus in charge for your Latency-free realtime datastream.
My theory: If the buffer is 6000 or 12000 that's not the key issue. The key issue is to avoid long interrupts from e.g. your Harddisk during playback, which potentially influence negatively the USB realtime stream at a point in time where there is no chance to get the Latency-Jitter effects cleaned up anymore.
Anyhow. I think this is well known fact - at least to MS. I hope that the "exclusicve mode" in Vista will limit these kind of impacts.
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I know that some high-end soundcards/DACs do some buffering and reclocking before the stream leaves towards the DAC-chip to avoid these kind of Latency-Jitter or Jitter problems.
Pro sound cards have 1ms buffer, as more regular soundcards do usually have more, about (or more) than 5ms buffer. Also with WDM drivers, there is another buffer in the windows kernel in the mixer. Since this buffer is running in the kernel, it would be less prone to interruptions.
I really don't believe in this interruption theory. Unless you have a really slow computer where interrupts are taking lots of time, with a modern computer, I'd be more than surprised that they are visible/audible. To affect the audio stream - given that there is 1ms buffer in the sound cards - it would require that interruptions take around 1 ms?!
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I've tried this but have come upon a couple of problems
On short tracks (about 3 minutes) if I'm on the playing now screen with a track info template, the display changes to the next track about 2 thirds of the way through the track and shows the wrong lyrics etc
Also whilst trying to correct lyrics using 'doofs lyric editor' showed the same problem
When you get towards the end of the track and then move back a bit to re-check the words MC plays the next track instead (but shows the origional track name etc in the top section of MC)
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I have been running a RAM disk for several years - it is currently 1Gb.
I shall try loading the tracks (APE format) onto the RAM disk and run MC from there to see if ther in an improvement in SQ.