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[katywu]

I don't know where to start with this one, it seems very strange. First thing to say is that until recently everything worked fine but now I'm experiencing some very strange problems when I select tracks to play.

For MP3s no problem, I select them and they play almost instantly. For FLAC files ripped from my CDs then there is a more noticeable delay whilst they buffer but they normally start OK although I can get a second or two of music then an interrupt whilst the buffering continues for about six seconds then the track plays.

The main problem appears to be with 24/96 files.....they buffer for a long time >30seconds and then appear to start playing but no sound whatsoever. If I then select an MP3 or FLAC to play then the same - no sound. I have to close MC and start again to recover.

I've played around with the buffer settings but I don't seem to be able to get a stable solution....this is really annoying as everything was working fine and I'm not aware of making any changes at all.

Last night I upgraded to the latest version of MC but exactly the same!

I'm using XP with firewire to a Focusrite PRO24 audio interface and then SPDIF out to my DAC.

I'm using the Focusrite ASIO drivers with memory playback. Whilst switching off memory playback does improve the responsiveness of playback time the same issues exist with highres files and subsequent attempts to play std files.

I also have Foobar on my machine which works fine for all files??

I'm sat here scratching my head, any ideas? ?

Thanks

Mike

[katywu]

Anyone got any ideas?

I guess I could do a complete re-install

Thx

[JimH]

There's no need to reinstall.  Please see this thread:
http://yabb.jriver.com/interact/index.php?topic=60735.msg409948#msg409948

[katywu]

Jim

Thanks but I don't want to up sample all my music collection, I just want to play at the original resolution

Doesn't the proposed solution involve up sampling?

Thanks

Mike

[nilbor]

Hmm, this rings a bell. This may very well be the same problem that I have been troubled with. See:
http://yabb.jriver.com/interact/index.php?topic=60471.0

The problem I saw started sometime after I started playing/ converting my flac files to 24/96 in DSP Studio/Output Settings and it caused similar problems as described here. Occasionally buffering takes a long time, occasionally playback stops for periods, MC may even freeze for long times or even restart. I tried going back to not using DSP studio to convert to 24/96 and it seems to be stable (only tested it for 2 hours).

[JimH]

There is no harm in upsampling.  You don't lose anything.

Some devices don't switch very gracefully.

Make sure you have a recent version of MC.  15.0.159 is on the download page.

[pcstockton]

You might not lose anything by upsampling but you might get some things you do not like.

I cant see why ASIO, KS, or WASAPI output in MC wont send along exactly what the file is regardless of your DAC.

Maybe your soundcard does not handle auto sample rate switching..... many dont.

MC has NO clue what DAC you are using, so the DAC is irrelevant.

I personally would NEVER use ANY kind of EQ or upsampling.  I want to listen to it the way it was intended by the artist, engineer, masterer etc....

[DarkPenguin]

You might not lose anything by upsampling but you might get some things you do not like.

Like what?

[pcstockton]

only your ears can tell you

[DarkPenguin]

only your ears can tell you

So no first hand experience with any issues from doing this?  Or you just don't?

Edit: This isn't me being hostile, btw.  And I'm isolating upsampling from any other file modifications.  I'm curious how a simple upsample can change anything outside of you now have to push more data.

[pcstockton]

I personally do not upsample, oversample or EQ etc, in the media player.  I dont like the idea of it.  If my DAC wants to employ it that it is a different issue.  I am not a NOS kind of guy by any means.

Since I do not upsample (and cant see a reason why I would), I cannot comment on its relative sound quality.  But I can say I know more than a few who think they can hear a difference and none feel it is a positive change.  I have learned enough in last 6 years of computer audio that bits are NOT always bits.

My comment was meant earnestly.  Listen for yourself to see if it is better, worse or equal.  Your mileage may vary et al...

I would recommend "auto sample rate switching" as a necessary feature of whatever you use to output the audio (whether digital or analog) from your computer.   For me that is of utmost importance.  I prefer to have the media player send the files unfettered to the soundcard.  I use ASIO so the OS doesn't change anything along the way.  Then I want the soundcard to detect the sample rate and automatically switch to match.

From there I rely on the very intelligent minds and ears of the designers of my Naim DAC to do any sampling, and reclocking etc.

It is just my preference to keep it simple and clean.

I would not be surprised if you found upsampling to be harmless.  But I also would not be surprised to hear that you do not prefer it.

-Patrick

[Alex B]

"Upsampling" is quite off-topic in this thread, but here's my comment anyway.

When a digital audio signal is resampled the new samples in the timeline are calculated. The signal doesn't stay unaltered. However, when a high quality resampler is used there is no audible difference. For instance, you can up and down sample an audio signal several times and you will still fail to hear any difference in a blind test when the resulting signal is compared with the original. Our ears are not very precise measuring instruments and in any case the faint distortion that is caused by such procedure will be masked by the much bigger distortions that are caused by the analog amplification and especially the final transducers (speakers or headphones).

However, resampling is a waste of CPU cycles and I'd recommend it only when it can fix problems that are caused by inadequate hardware.

I have no problems with playing various sample rates with my old Terratec DMX6 fire sound card. It switches sample rates automatically and perfectly fine using WASAPI Exclusive, ASIO, Kernel Streaming, Direct Sound or Wave Out (the last two on XP*). I have not experienced any delays or freezes when it switches between sample rates, e.g. from 44.1 KHz to 96 KHz and back again.

The above mentioned Terratec model is not available anymore, but perhaps a card like M-Audio Audiophile 192 (both are based on the Via Envy24 chip) would be a good digital interface for connecting an external high end DAC through SPDIF.

* The audio engine in Vista and Windows 7 is always set to a constant sample rate and it resamples when needed. ASIO, WASAPI Exclusive and Kernel Streaming can bypass it.

EDIT

Increasing the bit depth value is different from upsampling. If no DSP is applied it is fully lossless. When the signal is modified by any DSP the processing in MC is done in the 64-bit float mode and it is preferable to set the output to the highest bit depth that is supported by the hardware.

[pcstockton]

The M-Audio Transit does not have auto sample rate switching... just FYI.  I dont know about their PCI cards though.

The ESI Juli@ is a great card for $150.

RME makes a bunch.

Also check out the M2tech hiface.

[katywu]

Upsampling hasn't fixed the problem although at least some music played.....I set the upsampling to 96Khz and the bit depth to 24 bit both of which are supported by my firewire audio interface (Focusrite PRO/24) and my DAC (Naim).

When I select a 96/24 file to play the J River spends about 30 seconds of buffering and then commences to play but stops intermittently to re-buffer. A standard 44/16 FLAC file (upsampled) seems now to play OK even after I have tried to plat a 96/24 file which is an improvement.

As said to start with this all worked fine for a while ? ?

I currently have version 15.0.142 installed....so will download the later version suggested and try again, if that fails I will re-install and report back.

I can't see how this problem has anything to do with my audio interface or DAC as it did work OK originally with J River and works perfectly well with Foobar.

Thanks for the suggestions so for

Mike

[katywu]

I installed 15.0.159 and it made no difference.

I completely removed J River and re-installed. I just imported a small number of songs (approx. 30 and all 24/96) and in standard playback mode everything works fine, quick response and no stuttering. If I set J River to play from memory then I experience delays with buffering and the music stops/starts intermittently. OK I can live without memory playback for now so I then imported a large portion of my CD collection (about 6000 songs) all 44/16 FLAC files. From this point the problem returned....J River won't play 24/96 files (stays in buffering or when the song starts to play there is no sound!). The problem then manifests itself with standard 44/16 files i.e. they appear to start playing but no sound.....restarting J River sorts the problem until I play another 24/96 song!

Is there any other information that would be useful in trying to resolve this problem?

Thanks

Mike

[Vincent Kars]

There are some firewire issues in XP
Tried the hotfixes?
http://www.focusrite.com/answerbase/en/article.php?id=264

[katywu]

Vincent

Thanks for the suggestions....with regards the tuning then I have previously completed these tweaks. I'm already on SP3 and didn't hotfix SP2 so it would appear that I don't need any further update. It would seem strange if this does turn out to be a firewire issue given all my other audio programs work perfectly well.

My machine is dedicated to audio playback so has been tweaked during build to maximise SQ. I would be interested to hear how others have configured the advanced performance features of XP when using J River. I notice some of these settings do impact DPC latency although again no impact to my other programs. My DPC latency with performance set as background is approx 60microseconds and about 6 microseconds when not set as background.

I currently have the advanced perf set as background/programs/no page file - I have played around with the settings but still experiencing problems. Ideally I want to use J River with memory playback with std and hi-res files - this isn't looking very promising at present.

Cheers

Mike