Can someone pls post the sample rate of an MP3 which plays back @ proper speed on your Fuze?

So your procedure was to transfer the test signal file to the Fuze, play the file from the Fuze through a pair of IEMs while recording the IEMs’ output through a pair of microphones, then compare that recording with the original test signal file using your ears.  Right?

The problem with using your ears is that we have no way of knowing that your hearing is precise enough to distinguish a 0.7% difference.  That difference is only ~1/10th of one semitone.  If you recorded the Fuze’s output through the dummy head, why not upload it for us?

If you were going to test again, I would recommend ditching the dummy head and instead running the signal from the Fuze directly into your computer’s audio interface.

I would just do it myself, but I don’t have a Fuze (want to see the results of these tests before deciding whether or not to purchase).

Message Edited by maxplanck on 02-15-2009 10:12 PM

@conversionbox wrote:

Edit: I just thought of this if thedifference is 1 or 2 cents off, and as you say most people wont hear it, is it a probelm at all. Not IMO its just an inconveniance.

MOST people won’t hear it. Some people will hear it. Some people may not hear it, but are interested in training their ears by listening to music. If they train their ears using an off-pitch player, then they will be screwed.

Therefore it is a big problem. 

Message Edited by maxplanck on 02-15-2009 09:54 PM

No IEMS Direct line into the coputer which replecates the process. And I dont work at that studio anymore, so I cant get at that recording or I would gladly post it along with the results.

OK, thanks.  If your home computer’s motherboard even has a line in jack (usually the jack is red, and of the headphone type, located on the back of the computer case), you can record the output of your Fuze through this jack using free software such as Audacity if you don’t already have DAW software installed.  Would only take a few minutes  :stuck_out_tongue:  

Test tone file is conveniently located here:

http://www.mediacollege.com/audio/tone/files/1kHz\_44100Hz\_16bit\_05sec.mp3 

Has anyone tested a 48KHz file? I have an e200v2 which I have been praying for rockbox to finally get working on… However in the time I have spent waiting I have been trying to chip in and learn about porting rockbox; as far as I know the e200v2 has the same chip as the Fuze does, the Austrian Microsystems AS3525.  I looked at the AS3525 site (http://www.austriamicrosystems.com/eng/Products/Mobile-Entertainment/High-Performance-Microcontrollers/AS3525) where they state “Sampling Frequency: 8-48kHz” for the Sigma Delta (sample rate [SRC]) converters.  Since the SRC (and the rest of the audio signal path; ex. DSP or any processing) probably operate at the highest sample rate supported by the chip I’d assume that anything at 44.1kHz is getting upsampled to 48kHz;  I would try and run a test at 48kHz; I would however I am at school (not living at home) and dont have any cables or anything like that to run the test, just some headphones :slight_smile:  

Until I can get rockbox running and fine tune the eq parametricly I plan to use foobar2000 to ‘convert’ all my music for headphone listening and while converting also process a custom EQ dsp (Electri-Q), a headphone (B2SP) and run SoX (very good sample rate converter) to convert to 48kHz sample rate… Another thing; what good is the eq on the e200 (presumably all the other sansa players) when all you can see is 5 bands; you dont know the frequency or slope of the eq lol…  anyways; hope this all helps or has atleast helped the discussion.

@mp3geek wrote:


 


I found my wife’s Chromatic tuner again and you guys sparked my interest so I made a few new test files using Audacity and tested them. The results are shown below.

  1. 440 Hz sine wave at 44100 sample rate encoded with LAME MP3 - 20 cents slow
  1. 880 Hz sine wave at 44100 sample rate encoded with LAME MP3 - 20 cents slow
  1. 440 Hz sine wave at 48000 sample rate encoded with LAME MP3 - dead on
  1. 440 Hz sine wave at 22050 sample rate encoded with OGG         - dead on
  1. 440 Hz sine wave at 96000 sample rate encoded with OGG         - dead on

 

There seems to be a pattern here. Does anyone want to comment.

 

Note that the real time clock is about 5 minutes slow - but I set it about a month ago.

 

p.s. I can hear a clear difference in pitch between these files.

 

http://forums.sandisk.com/sansa/board/message?board.id=sansafuse&message.id=17849#M17849 


@donp wrote:

 

 

 

 

So what level of precision does “consumer electronics” have?

 I made 1000 Hz  and 1002 Hz wave files in cool edit, converted to flac and ogg/vorbis for the players, and burned to 2 CD’s.  Playing 2 sources at the sime time will give a beat frequency equal to the difference in the tones’ frequencies (as anyone who’s tuned 2 instruments against each other knows).  The 1002 hz file was a sanity check to make sure I could hear the beat when 2 sources are known to be off… worked in all cases.

 Playing Cool edit against foobar on the same PC, flac, wav, or ogg, no beat (everything consistantly in tune) 

CD player vs DVD player (both fairly cheap consumer models, different brands and about 20 years apart in age) - roughly 30 seconds per beat  (1/30 hz), for an difference of 1 part in 30,000 or 0.003% 

CD player or Cool Edit vs  Sansa E200 (rockbox playing flac)- beat of ~1/3 Hz, error 1 part in 3000, or 0.03%.

Cool Edit vs Neuros player (playing ogg file) - Also about 1/3 Hz, so 0.03%

 Cool Edit vs Clip (playing flac) - beat of 7 Hz, error about 1 part in 140 or 0.7%   I checked this one by generating a new wave of 1007 Hz, which was in tune with the Clip playing the “1000 Hz” file.

So the Clip’s pitch error (and presumably play speed error) is over 20x worse than my other portables, and 200x worse than the difference between my CD player and DVD player.

So the typical standard for consumer electronics (including an older Sansa model) really is a lot better than the current lot.

http://forums.sandisk.com/sansa/board/message?board.id=sansafuse&message.id=17849#M17849

From the test results posted so far, it looks like 44.1kHz files are the ones being played back too slow.  DonP’s measurement indicated a ~0.7% slowdown. MP3Geek’s measurement (probably cruder than DonP’s due to his methodology) indicated a ~1.15% slowdown (~20 cents).

@My initial suspicion was the same as yours, that 48kHz files were being played back @ 44.1kHz, thus resulting in slowdown. However, if that were the case, the slowdown would be 8.125%, which is far larger than the amount of slowdown indicated by the test results. This suspicion is also defeated by the fact that 48kHz files apparently play back @ normal speed, the problem only occurs when 44.1kHz files are played back (I’m still awaiting more solid confirmation of this though, we’ll know when we see more people post their test results).

The fact that MP3Geek’s Fuze can play some files back at the correct speed indicates that the internal clock mechanism is probably ok, so I think that this problem can probably be corrected via firmware (don’t know for sure though, since I don’t know the device’s internals). 

DonP, the sample rate of your test file was 44.1kHz, right? 

Message Edited by maxplanck on 02-16-2009 01:46 PM

my theory is/was that its trying to preform a sample rate conversion from 44.1khz to 48khz; its not trying to play it back at 48khz, something in the sample rate converter could be ‘messed up,’ since 44.1khz to 48khz is a ‘messy’ conversion (not evenly dividable as something like 22.05khz ->44.1khz is).  be it too little processing power, or some other reason the chip is not converting the sample rate fast enough or maybe 0.7% is the difference between an ‘even’ division in the chips clock cycles…

That is why I am using my computer to convert my tunes to 48khz since it seems to be the sample rate the chip is using; sample rate conversion (accurate sample rate conversion) takes alot of processing power if you want to do a good job and maintain quality (granted, since we are dealing with mp3 the quality will probably not show up, might as well not make it worse than necessary though :P)

http://src.infinitewave.ca/ 

See this site for a comparrison of software sample rate converters (as an example of what I mean in the quality difference) 

EDIT: another example reguarding to the chips SRC; have you ever watched video on a computer that is way to slow to play it? you can either watch the video choppy or you can watch ‘smoothly’ with less frames (frame rate goes down).  I have a feeling that there may be some sort of ‘waiting delay’ that the chip is producing 0.7% of the time; like if the dac is waiting for the processor… 

Message Edited by Chesteta on 02-16-2009 02:33 PM

It sure would be weird for the Fuze to be designed to play everything @ 48kHz, thus requiring a messy sample rate conversion from 44.1kHz for almost every file, since 44.1kHz is by far the most common sample rate for audio files (someone pls correct me if I’m wrong about that). I have a hard time swallowing the idea that even a drunk engineer would design the system that way.

But if there’s anything in this world that tops human thoughtlessness, it’s human thoughtlessness. With that in mind, those theories sound plausible to me. 

I may be missing something, I have no experience with hardware design.

In any case, let’s see what peoples test results are. That will shed some more light on the problem and hopefully reveal it to be a bad batch rather than a universal problem with the Fuze, and thus we can hopefully just avoid the bad batch and leave this problem for the guys @ Sandisk who are actually getting paid for their time to figure out.

Message Edited by maxplanck on 02-16-2009 03:26 PM

jmr wrote: 

Here you go:
MP3 VBR - 119KB: http://www.sendspace.com/file/yloir3
WAV 24/96 - 2MB: http://www.sendspace.com/file/x3wy3l

 

Fuze version 01.01.22A
Max volume setting, Normal EQ setting.  Recorded using top quality settings on a Zoom H2 (my laptop doesn’t have a line in jack. D: ) 

 

My observation: if anything, my recording is a hair faster than it should be.  Adobe Audition says the original tone is centered at 1000.1Hz, whereas my recording is at 1007Hz.

 

Personally, I’ve never really noticed any difference between my Fuze and my other music devices, but this little test has me wondering a little.

Message Edited by jmr on 02-16-2009 09:45 PM

 http://forums.sandisk.com/sansa/board/message?board.id=sansafuse&thread.id=17953

jmr’s Fuze speeds up playback, not slows it down!  Perhaps our other testers got their files mixed up, perhaps their Fuzes actually speed up playback?  At least the magnitude of deviation is about the same between all tests we’ve run. 

EDIT: DonP’s Fuze speeds up playback, not slows it down.  Thanks jmr!

Message Edited by maxplanck on 02-16-2009 06:20 PM

@maxplanck wrote:

 DonP’s measurement indicated a ~0.7% slowdown.

Re-read his post.  His test had the same result as mine: a 1000Hz tone played back at 1007Hz.  That’s a speedup, not a slowdown!  :stuck_out_tongue:

doh, you’re right!  Thanks!  So that’s two players with identical behavior.  If we can get DonP to post his Revision and Firmware # that would be helpful.

Also if we can get the people who think that their Fuze plays back fine to run the test, and post their Revision and Firmware #, that would be very helpful too. 

MP3Geek, just to make sure we clearly understand your test results, can you pls tell us explicitly whether your Fuze sped up or slowed down the file that it played back?

Message Edited by maxplanck on 02-16-2009 06:23 PM

maxplank: its not a matter of “being designed to play everything @ 48kHz”, its the fact that *most* computer/digital audio hardware (besides cd players) run at 48kHz; take for example the entire creative audigy1/audigy2 line of sound cards (I used to be very active over in the kx project (www.driverheaven.net/kx-project-audio-driver-support-forum/) forums, the kx project is an independent driver for creative cards with a 10k1/10k2 processor; it allows much more configurability than the creative drivers.)  I consider it to be pretty normal to have an audio device running at 48kHz, it just so happens that the sample rate conversion in the AS3525 has an issue.  

Thanks Chesteta, I know that 48kHz is standard for computer audio interfaces.  But is it also the common playback rate for handheld digital audio players?  

I would guess not, since most MP3’s are 44.1kHz (please correct me if I’m wrong, check your MP3 library and tell me if you see mostly 44.1kHz or 48kHz files).  44.1kHz is the standard sample rate used when music is released for end listeners to purchase, since music is still published on CD (please correct me if I’m wrong, check your MP3 library and tell me if you see mostly 44.1kHz or 48kHz files).  When I check my music MP3 library, I see almost exclusively 44.1kHz.

Because of this, I would assume that most handheld digital audio players are designed to play back @ 44.1kHz, so as to minimize the number of files that they will have to perform sample rate conversion for.  I would imagine that performing sample rate converstion depletes the battery faster, so a smart engineer would want to minimize the likelihood that it would need to be performed?

yes, most of my music is at 44.1 kHz; and I do agree with you that it would make *sense* to run things at 44.1 kHz; its just that the AS3525 chip does not because it is designed to support up to 48kHz, I believe most mp3 players on the market are designed to support up to 48kHz; perhaps higher

EDIT/ADDITION: also, preforming the sample rate conversion is just a function of the chip; depending on if its a high quality sample rate conversion (lots more processing) or a simple one will be what determines the battery usage.  It seems from what other people are reporting and also what would make sense for a battery operated device, they (Austrian micro systems) probably used a more simple sample rate converter.

Message Edited by Chesteta on 02-16-2009 07:31 PM

Hmm, just because it supports up to 48kHz doesn’t necessarily mean that it physically plays back at 48kHz, or that its default playback rate is 48kHz.

Thinking a bit more about it, I don’t know which is easier to implement on a cheap handheld device: sample rate conversion, or the ability to physically play back at different sample rates.  I wish I knew more about hardware design, never interested me much before (still doesn’t, just would like to solve this problem :stuck_out_tongue:).  

If handheld devices operate via sample rate conversion (when necessary), then the physical playback rate is surely 44.1kHz, and sample rate conversion is only performed when playing back a file of a sample rate other than 44.1kHz. 

Is the AS3525 chip used in the Fuze?

yes; the as3525 is used in the fuze (see here: http://www.rockbox.org/twiki/bin/view/Main/SansaV2 at the top of the page where it says “Models:”)

I believe I posted this link before: (http://www.austriamicrosystems.com/eng/Products/Mobile-Entertainment/High-Performance-Microcontrollers/AS3525) this lists the specs of the AS3525; as far as ‘solving the problem’ as far as I can tell it is probably running at 48khz on the DAC’s (sigma delta converters). there MAY be a way to set it to 44.1 khz in firmware however I would not get my hopes up… This is why I have been converting my mp3’s to 48khz (just the ones on my mp3 player) through foobar2000; I set things up to output to a special ‘48khz folder’ so that it does not change all the music on my computer.  Also I can apply my more complex EQ filters that I have never found a portable audio player capable of running and do a much cleaner sample rate conversion than most if any consumer grade electronics will do (using the SoX v14.2 DSP plugin for foobar2000).  It is quite easy to do once it is set up and all of the mp3 tag information is transfered in the conversion so basically I take an album, select it, specify a folder to put it, and click convert.  Then I upload it to my sansa :slight_smile:

AHA!  Thanks for sticking with me until I got it!

Are there any other brand players which use this chip? And if so, do they suffer from the slow/fast playback problem? 

I did a search for AS3525 on the rockbox wiki and it only turned up under Sansa players.

If the chip is fully programmable to play back @ any sample rate with a decent level of accuracy (< +/- 0.03% off tune), then the problem is in Sansa’s firmware and not in the chip.  IF.

I just don’t want to have to convert my files to 48kHz, because I want to be able to swap music on/off of my player fast and easily. To have to convert every time would be a pain, and I don’t want to waste the space on my HD to store every file that I’ve converted just in case I want to delete it off of my Fuze then put it back on the Fuze some other day.

As per the AS3525 feature sheet, there are two independant programmable PLLs. So the problem most likely lies in the firmware. Sandisk may or may not ever address the problem depending on the amount of complaints which I would guess is minimal. Most people aren’t going to hear that a problem does exist. The majority of users are listening to 128k MP3 (or comparable WMA), which to me sound terrible and they pay for it without complaining about it. Ask most of them if they hear artifacts in their MP3/WMA files and they will not even know what an artifact is!

I agree this problem should not exist, as the hardware is fully capable (and should be from a quality standpoint) of being correct on playback.

If users have special needs as they have pointed out here, they are better off looking into a different player instead of hoping for a correction to the Fuze that Sandisk may never fix. It is good that the topic is brought up for Sandisk to see that it is a concern of potential buyers and owners.

I don’t think a bunch of testing by users is going to help. Sandisk has been made aware of the problem, let them do the testing.

Rockbox is another option; the port is becoming closer and closer to being finished for the as3525 models; I have been watching its development for this player for about 4 months now :slight_smile: id give it another coupple of weeks to a month and it should be definitely working at a usable state (with how things seem to be progressing as of late.)  

EDIT: I understand what you mean about not wanting to convert/reconvert music… I have an e200v2 (8gb) and an 8gb microsd card so at least for me this is not a huge issue… I generally listen to the ‘top 20’ cds that im interested in at the time; fyi, the conversion process on my computer (a core 2 duo laptop @ 1.6 ghz) takes approx. 5 mins per cd; it could go around twice as fast however I am running the DSP stuff also besides the sample rate conversion which takes up alot of CPU power too.

Message Edited by Chesteta on 02-16-2009 10:28 PM

At least by testing and sharing results, less people who care about correct playback will be suckered into purchasing this thing.

That would be sweet if Rockbox can make the Fuze play back correctly. Amazing that it takes an independent third party, open source, unpaid developer to get this done. Should be no surprise I guess, since it takes a group of unpaid users to figure out that there is a problem in the first place  :stuck_out_tongue:.  

God I hate the way large corporations operate. Someday far in the future ordinary people who are fed up with this shlt will organize and democratize the economy, and the world will be a better place for it. That is, if our species even survives the period of terrible mismanagement between now and then.  

Did you guys hear about the British and French nuclear armed and powered submarines that crashed into one another in the Atlantic a couple weeks ago? My jaw just drops in disbelief at the terrible mismanagement of our societies.

Message Edited by maxplanck on 02-17-2009 08:07 AM