Print

Please login or register.
1 Hour 1 Day 1 Week 1 Month Forever
Login with username, password and session length

[Von]

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...

[Fred1]

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

[Mr ChriZ]

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).

[LesC]

so how about that file buffering ? sure would save some battery life off of a laptop

[JimH]

We'll consider it in a future version.  Under normal circumstances, it shouldn't make any difference.

[Tuckers]

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.

[Matt]

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.

[LesC]

thanks Matt ! i put it at 54000 just to make sure it buffers everything.. even the albums i downloaded off of tranceaddict

[Tuckers]

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.

[LesC]

i did full file buffering with foobar and noticed an improvement, no doubt moreso with our beloved jrmc . 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.

[LesC]

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

[soundcheck]

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

[Tuckers]

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. 

[LesC]

define high end

[LesC]

http://www4.head-fi.org/forums/showthread.php?t=89382

and they didn't believe me 

[soundcheck]

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

[Tuckers]

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!

[soundcheck]

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!

[Tuckers]

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?

[soundcheck]

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?

[Tuckers]

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.

[hit_ny]

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.

[Tuckers]

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.

[soundcheck]

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.

[Tuckers]

I'd like to keep this thread on track, so PM or email me to discuss the issues of Flac and Wav etc.

[LesC]

Tuckers, please let me know when you get your additional 1gb if it fixed the stutter. thanks

[hit_ny]

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. 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 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."

[LesC]

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..

[soundcheck]

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.

[JimH]

I can't completely follow this thread, but try a google search for:

via chipset latency setting

[glynor]

I can't completely follow this thread

    

They must have far, far, far better ears than you or I, Jim.

[dlone]

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

[soundcheck]

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? 


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?

[LesC]

soundcheck, how do i do that ? is that in task manager ?

[LesC]

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).

[LesC]

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.

[soundcheck]

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.

[LesC]

oh.. you never know what goes on in windows

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.

[JimH]

Please try re-downloading and re-installing the latest.

[soundcheck]

oh.. you never know what goes on in windows

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?

[JimH]

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.

[glynor]

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.

[soundcheck]

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".

[soundcheck]

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?


 

[LesC]

16 bit just sounds better than 24 bit. even with foobar . of course foobar's playback at 16bit won't beat jrmc's at 24...

[soundcheck]

16 bit just sounds better than 24 bit. even with foobar . 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?

[Listener]

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

[MerlinWerks]

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...
http://en.wikipedia.org/wiki/Loudness_war

[LesC]

yeah that would be great but i don't listen to pop..

say soundcheck, can you CHECK if .200 sounds better than .199 ? i'm getting paranoid here. 

[soundcheck]

say soundcheck, can you CHECK if .200 sounds better than .199 ? i'm getting paranoid here.

The paranoid stage I entered apparently by the time I started reviewing SW-Players a while ago! So - You are not alone.