mp3 player virgin needs help

[The Lone Rider]

Where the fact of the matter was mp3 was the illegal underground format of ripping music, and didn't work on either windows media player, or any apple device at its time of conception.

Mp3 is not and has not ever been that at any point of design intention. Many musicians collaborate with others all around the world, as well as film companies needed to send audio files here and there so the ie.. editing suites can do their job while the sound man is still finishing his! Mp3 allowed for the audio files to be sent quickly all over the world, allowing people to do their part.

See http://www.artistcollaboration.com/ as an example. Composers record songs, post an Mp3 with a request for this or that added to it. Someone loads the mp3 to their gear, records the track, sends back a mixed mp3 and says "Is that what you wanted?" "Yeah it is" and then he sends what he recorded uncompressed to the original composer.



For a DSE mp3 player, it should be as following.

Convert your CDs/.wav to mp3 (i use dbPower Amp and MusicMatch Jukebox), or download some. Plug your mp3 player in via USB, it should show up as a removable disk drive. Copy mp3s to that disk drive via my computer. Done. Enjoy the music.

The amount of songs you can fit on there has to be balanced with the quality of sound you want. You can record 32kbps and listen to hiss if you want, and have 200,000 songs on there, or you can go the other extreme and have about 10 320kbps (these are exaggerations to make a point).

Mp3 players, like speaker systems, list the max they can do - not neccessarily do well. So if it can hold 120 songs - check if thats 120 songs at 128kbps and if they are stereo.

Similarly with speakers/amp systems, you see so many that are advertised as 200 watts. But read the small print and it's 200 watts PPM, which means it can do 200 watts for one second before it blows your speaker cone. Look even closer and it says it can only do 50 watts RMS, which basically is the average it can do and is the rating you should go by. Look even closer still, and you will find that its 50watts total.. and that means if its a stereo system, you really just bought yourself a 25watt speaker system.

Lesson to be learned, read the fine print, don't listen to adverts or salesmen, ask people who actually know.

Edit--
And Apple being better then everything else is their marketing, not fact. My Dell PC shits all over my mac, and they are matched spec for spec with few differences. PCs just need a little bit more TLC at times.

[RiderInBlack]

For a DSE mp3 player, it should be as following.

Convert your CDs/.wav to mp3 (i use dbPower Amp and MusicMatch Jukebox), or download some. Plug your mp3 player in via USB, it should show up as a removable disk drive. Copy mp3s to that disk drive via my computer. Done. Enjoy the music.

Windows Media Player will do this too AND I still recomend ya go to WMA files (not to be confused with WAV files, so please stop going on about them) not mp3 for ya DSE player They are better than mp3 and ya get more songs on. Please got ta http://www.microsoft.com/windows/win...serlocale=1409 if ya don't beleave me.

[The Lone Rider]

Encoded WMA files are licensed format and using MS encoders sends information to microsoft to gather for the marketing.

WMA also sounds worse then a well encoded high quality mp3.

.wav ALSO sounds better then WMA files if it is a standard 16bit 44.1khz .wav file, as it would be uncompressed and have true clarity.

WMA is MS grasp at the portable audio format, and fizzled out like the minidisc and digital compact cassette that other companies put out, and WMA is predominately used by web authors or those who don't reach for alternatives to MS products first.

Real Media audio files suck as well.

[pete376403]

Oh for trivias sake, the reason 44.1khz 16bit was decided as the red book standard has to do with one of the people on the committee listened to bach or bethoven or something like that... and anyway he commented that this one particular song is about 80 mins long and he has to turn over his tape/record in order to hear it all. So it is decided the format needs to fit about 80mins on, and by the way CDs are made, that meant audio had to be encoded at 44.1khz 16bit. The story is something along those lines anyway..

Close....
The original target storage capacity for a CD was an hour of audio content, and a disc diameter of 115 mm was sufficient for this. However, according to Philips, Sony vice-president Norio Ohga suggested extending the capacity to 74 minutes to accommodate a complete performance of Beethoven’s 9th Symphony;[3] however, Kees Immink of Philips denies this.[1] The extra playing time subsequently required the change to a 120 mm disc.

According to a Sunday Tribune interview [1] the story is slightly more involved. At that time (1979) Philips owned Polygram, one of the world’s largest distributors of music. Polygram had set up a large experimental CD plant in Hanover, Germany, which could produce huge amounts of CDs having, of course, a diameter of 11.5 cm. Sony did not yet have such a facility. If Sony had agreed on the 11.5 cm disc, Philips would have had a significant competitive edge in the market. Sony was aware of that, did not like it, and something had to be done. The long-playing time of Beethoven's Ninth imposed by Ohga was used to push Philips to accept 12 cm, so that Philips’ Polygram lost its edge on disc fabrication.
http://www.answers.com/topic/compact-disc-2

[The Lone Rider]

Close....
The original target storage capacity for a CD was an hour of audio content, and a disc diameter of 115 mm was sufficient for this. However, according to Philips, Sony vice-president Norio Ohga suggested extending the capacity to 74 minutes to accommodate a complete performance of Beethoven’s 9th Symphony;[3] however, Kees Immink of Philips denies this.[1] The extra playing time subsequently required the change to a 120 mm disc.

According to a Sunday Tribune interview [1] the story is slightly more involved. At that time (1979) Philips owned Polygram, one of the world’s largest distributors of music. Polygram had set up a large experimental CD plant in Hanover, Germany, which could produce huge amounts of CDs having, of course, a diameter of 11.5 cm. Sony did not yet have such a facility. If Sony had agreed on the 11.5 cm disc, Philips would have had a significant competitive edge in the market. Sony was aware of that, did not like it, and something had to be done. The long-playing time of Beethoven's Ninth imposed by Ohga was used to push Philips to accept 12 cm, so that Philips’ Polygram lost its edge on disc fabrication.
http://www.answers.com/topic/compact-disc-2

Yeah, I knew it was something like that.

Audio engineer for years, historian never

[RiderInBlack]

WMA also sounds worse then a well encoded high quality mp3.

But hardly worth a red rep "WMA sucks, and is for horse shoe fitting posturing tools such as thyself. Quote Dyers" . If ya have a problem be a man and post it.

[Street Gerbil]

Ahem ahem...
Guys, any digital recording, even in raw format, is by definition lossy. Instead of recording an analog waveform, as it once was with vynil records, you are taking samples. On a CD you sample the input waveform 44k times a second (i.e. very very often) but it still means that when you play it back, you play back an approximation rather than the original waveform (quantification loss). Secondly, your storage for each sample is not infinite. At best you can use 16bits i.e. values from 0 to 65535 to record frequencies between 20Hz to 20.000Hz. Obviously you have to introduce some approximation here as well (PCM compression loss).
I am not familiar with full specs for SACD but given its sampling rate of almost 2.5MHz (vs. 44kHz on CDDA) it is probably as close to the real deal as it gets.
Unfortunately SACD cannot be read/ripped/recorded on the PC so I am going to stick with my CD player for a while.
Sorry for being such a smartass. Just my $.02
P.S. Use EAC for ripping. It may take a whole night per disk (vs. ten minutes) but the resulting mp3s will sound soooo much better!


EDITED -----

The only thing music related at the time was remixed and looped midi files on an SB16 (if you were lucky) - because your mate upgraded his flash DX40 to play doom with actual music.

Dude, does the name Gravis Ultrasound Max ring a bell? The decision of Gravis to get out of soundcard business and leave the market to the likes of Creative is nothing short of a crime against humanity.

[The Lone Rider]

Ahem ahem...
Guys, any digital recording, even in raw format, is by definition lossy. Instead of recording an analog waveform, as it once was with vynil records, you are taking samples. On a CD you sample the input waveform 44k times a second (i.e. very very often) but it still means that when you play it back, you play back an approximation rather than the original waveform (quantification loss). Secondly, your storage for each sample is not infinite. At best you can use 16bits i.e. values from 0 to 65535 to record frequencies between 20Hz to 20.000Hz. Obviously you have to introduce some approximation here as well (PCM compression loss).
I am not familiar with full specs for SACD but given its sampling rate of almost 2.5MHz (vs. 44kHz on CDDA) it is probably as close to the real deal as it gets.
Unfortunately SACD cannot be read/ripped/recorded on the PC so I am going to stick with my CD player for a while.
Sorry for being such a smartass. Just my $.02
P.S. Use EAC for ripping. It may take a whole night per disk (vs. ten minutes) but the resulting mp3s will sound soooo much better

You forgot to mention aliasing and the Nyquist point

[Street Gerbil]

You forgot to mention aliasing and the Nyquist point

I have no idea what they are. Would you care to elaborate?

[Holy Roller]

One of the guys here at work uses FLAC format recons its the bees knees. Claims to a lossless system, sounds heaps better than Mp3 but at $600 for the player I'll live with my legend Jazz and SD cards for now. Money is better spent on the bike

[xwhatsit]

One of the guys here at work uses FLAC format recons its the bees knees. Claims to a lossless system, sounds heaps better than Mp3 but at $600 for the player I'll live with my legend Jazz and SD cards for now. Money is better spent on the bike

FLAC is nice, I use it on the computer (for free), but convert it to MP3 for the iPod. Also bigger file sizes, but still far better than WAV for exactly the same quality (like you said, it's lossless -- at least in terms of digital audio, Street Gerbil is correct about sampling).

There's quite a few cheaper audio players these days that do FLAC. I think also the Rockbox firmware you can install on existing MP3 players (iPod and iRiver are two that support it) does FLAC as well. But even with really nice canalphones like Etymotic's ER-6 line, you still get enough outside interference and microphonics when walking around with a portable audio player that FLAC is often not worth the effort, at least in my opinion. My ears aren't that good, they've been hammered a bit from sitting just in front of the percussion section for many years

[imdying]

I have no idea what they are. Would you care to elaborate?

He's saying you're full of shit cause your ears ain't that great anyway

The format choice isn't going to make a difference to the OP, she'll have crap ear phones and be on a motorbike or walking down the street anyway

[The Lone Rider]

I have no idea what they are. Would you care to elaborate?

Well this is hard to explain without drawing charts.

But it has to do with sample rates.

lets use 44khz for simple maths sake and lets say we are using a simple sine wave

In simple terms you are getting two samples per milisecond (might be microsecond or something smaller - not sure) for a 22khz frequency tone. So you have captured the 22khz, but because its only two samples you end up with something very triangle wave looking.

So in going with that, the reason 44.1khz was chosen was because it was decided that the absolute frequency you want to record should have at least two samples. So because humans have a range, when their hearing is perfect and that generally is only in the 1st year of life, you can hear 20hz to 20khz. So 44.1khz was chosen to give a bit of lee way so that 20khz in our CDs dont sound like triangle waves. This whole thing is Nyquist - that your sample rate should be double the maximum frequency you want to record.

Most digital recording studios, including my own, record at much high sample rates. This gives you more samples in the higher ranges, making your cymbals etc sounding like fine crystal! But if your intended final medium is a CD, you need that sample rate lowered to match. For myself, I am bringing songs down from 96khz to 44.1khz.

Now, theres a lot of stuff on how the conversion works that I dont know, and probably wouldnt understand. But in the process of it all you are obviously losing information - dropping samples! Bye bye my nice sounding hi hat. This process creates something called aliasing, which creates harmonics (this is getting into psycho acoustics btw, another thing I don't understand too much of) and then makes other frequencies stand out! So suddenly your guitars might sound particularly screechy when you bring the rate down.

There is a fix, called anti-aliasing which I believe uses sub harmonic noise during the conversion to combat that effect. However I don't understand it either - my software does it automatically for me.

Different programs also anti-alias differently, so some can be considered good or bad at doing it.

---Disclaimer---
The above is my best attempt at explaining it, and also remembering facts that generally I take for granted as computer audio systems are very automated and integrated now.

Thank you very much.