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nashville,tn

So im curious, what settings and file types does everyone prefer or seem to find offers the best preservation of quality when needing to downgrade file sizes?

For example my gear set up these days is extremely spartan, i've got: a quailty set of monitors and power amp for sound amplification and my sp404sx for recording mono tracks from my dmg. Recording into the sampler of course means i cant multi-track, but the quality of the playback is spot on relative to the input. Easy enough, transfer either aiff or wav files into my computer via the sd card and pc reader.

Then its a bit of a mystery. I havent found a combination via mp3 i like(the most universal for my intents); a 46mb conversion gets squashed to a 4-6mb even at the highest/best quality. Thusly, when a track reaches the nets it's acquired a murky film that sounds a bit thin and flat. I've experimented with all the settings and must be missing something. Same scenario with either fully mastered or just sketch stuff to email or post.
??
Or is knowing best encoding just a trial by fire, trade secret kinda deal, like putting a hand kerchief over your trumpet so peeps cant steal yr fingerings? wink

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Brunswick, GA USA

It depends on the kind of music you make, so keep experimenting. From the sounds of things, I suggest trying different encoders.  When I want to encode something quickly, I usually go with 192kbps mp3s, from 48Mhz 16-bit sample renderings.

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In terms of lossless compression, the best general-purpose algorithm is usually LZMA.

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Sweeeeeeden
chunter wrote:

It depends on the kind of music you make, so keep experimenting. From the sounds of things, I suggest trying different encoders.  When I want to encode something quickly, I usually go with 192kbps mp3s, from 48Mhz 16-bit sample renderings.

*Hu-hum*
48 kHz.
And you should probably avoid this for online use, as Flash tends to prefer 44.1 kHz. (Maybe this was only true for older versions of the player, though.)

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Boston, MA
nitro2k01 wrote:
chunter wrote:

It depends on the kind of music you make, so keep experimenting. From the sounds of things, I suggest trying different encoders.  When I want to encode something quickly, I usually go with 192kbps mp3s, from 48Mhz 16-bit sample renderings.

*Hu-hum*
48 kHz.
And you should probably avoid this for online use, as Flash tends to prefer 44.1 kHz. (Maybe this was only true for older versions of the player, though.)

48 Mhz would be pretty gnar, though.

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Sweeeeeeden
campbell wrote:

48 Mhz would be pretty gnar, though.

Well, as it turns out, hz is a different unit than Hz. 1 hz is defined as 1 mHz (one millihertz). So 48 Mhz ends up being 48 kHz after all.

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Boston, MA
nitro2k01 wrote:
campbell wrote:

48 Mhz would be pretty gnar, though.

Well, as it turns out, hz is a different unit than Hz. 1 hz is defined as 1 mHz (one millihertz). So 48 Mhz ends up being 48 kHz after all.

the boy who was amused at the prospect of ridiculous samplerates in me just became a man. it's time for him to move on now.

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BOSTON

weird thing ive noticed recently is that i actually prefer chipmusic / electronic tracks to be in ~128kbp. and i dont mean just bloops and bleeps LOL8BIT, but produced tracks and everything. that grainy sound of the artifacts just evens out digital high frequency stuff really nicely to my ears.

128k definitely sucks for acoustic instruments and cymbals though, no question

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Boston, MA
BR1GHT PR1MATE wrote:

128k definitely sucks for acoustic instruments and cymbals though, no question

cymbals at 128kbps are the blurrrrrrst. snares are pretty terrible, too.

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nashville,tn
PlainFlavored wrote:

In terms of lossless compression, the best general-purpose algorithm is usually LZMA.

werds. thanks for the tip, will explore.

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.sav

</thread>

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Seattle, WA US

320kbps is considered the standard when it comes to the digital distribution of profesh mp3s.. If you ever cared to have your music dropped live in a DJ set, you would do this.

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Gosford, Australia
sugar sk*-*lls wrote:

Then its a bit of a mystery. I havent found a combination via mp3 i like(the most universal for my intents); a 46mb conversion gets squashed to a 4-6mb even at the highest/best quality. Thusly, when a track reaches the nets it's acquired a murky film that sounds a bit thin and flat.

a lot of streaming sites like soundcloud transcode your file to 128kbps for streaming purposes, so it could just be that.
personally i'd never use anything lower than 320kbps* for mp3's if possible. the lossy nature of the format means you will lose at least some information in the conversion but the audible difference between 320kbps mp3 and wav is pretty miniscule.

*although i am kinda partial to the smokey character of 96kbps! tongue

Last edited by Victory Road (Jun 7, 2012 5:23 am)

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sweden

Don't use 48Khz unless you're working directly to video.

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--

Last edited by ioflow (Jan 5, 2017 8:54 pm)

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A gray world of dread

Maybe related: I've made a comparison of low-kbps mp3 encoders here. Note that for high kbps LAME is pretty good, and what I usually use for mp3 encoding. The latest Ogg/Vorbis is superior in quality/size ratio, so I use that for encoding CDs and stuff.

I can see no reason not to use VBR unless LAME's algortihm obviously messes up. But that happened to me only once or twice, ever.

Also:

nordloef wrote:

Don't use 48Khz unless you're working directly to video.