NON-4/4 Old School Electro: Feedback PLZ!

[Locus M]

Hi exxxy, Emdot, Jazzy and thanks to the newcomers, loco pope and kodera for dropping by with some encouragement. Every word counts

exxxy and Emdot: I have made a new mix based on your primary feedback with just about every change suggested + a few. I have first seen your second suggestions now, exxxy, but actually we must have thought just about the same thing with the melodies because I have done some things you say.


http://www.box.net/shared/g5910dqsen


Changes are:

1) Volume issue fixed

2) A little more low-end by simply changing the low cut (sharp drop) from 60 Hz to 45 Hz. Hardly audible in headphones, but your subwoofer will know.

3) The main pad does now pan from side to side to create more 3- dimensional space.

4) So does the second pad, but in a more narrow span and faster. I think the effect is almost subliminal, though it does add to the space somehow.

5) The question-answer play between the bowed synthbell and the distorted tube is now panned with the first lead slightly to the right and the latter to the left, this adds a little to the space.

6) The lush trippy delay-lead is still centred but its delays are spread much wider than before, which adds to the space IMO.

7) On large scale mastering, the mids are widened a little and the highs are fully widened to add further to the space.

8. I have recompressed the track for a little less density but more present percussion peaks.


Hope this version is little more stereo then


All the Best and thanks once again you for your time and interest

[emdot_ambient]

[edit]The following was about the original mix...

Okay, I know you said you turned down the volume for the encoding quality, but I just had a look at the file in Adobe Audition and I see that the entire track is attenuated about -11dB from 0dB...that is, except for a few peaks, the entire track is at -12dB quieter than it could be simply by turning up the volume. (0dB being the loudest you can get without clipping.) There's also almost 20 seconds of silence at the end of the track and about 1.25 seconds at the start of the track...Do you own a wave editing program? If not, consider getting one. They will let you do tasks like normalization and trimming the dead space before and after the track, as well as encoding to mp3 trouble free.

Normalizing the track to 0dB (i.e. amplifying the entire track so that the loudest peak hits 0dB) raises the average volume to about -1dB and the track totally rocks.

It's much better than I thought it was in my headphones, because I could hear it finally.

[emdot_ambient]

I have made a new mix based on your primary feedback with just about every change suggested + a few.

And well done, too. Great track!

[aciddose]

since mp3 doesn't store any absolute transient information you can normalize to -10db rms most of the time. it's a bad idea to normalize based upon peaks since then you'll get wildly varying apparent loudness.

normalization used to matter as a dynamic range issue when we used formats that stored peak/transient values - that is no longer the case with floating point and transform-based encoding like mp3 or aac.

maintaining a flat apparent loudness between all tracks is good though!

when normalizing based upon rms values, you might need to insert a limiter in some tracks depending upon their content.

most of my tracks where i have the volume i want sit around -15db rms average with a peak of -6db.

[Locus M]

Do you own a wave editing program? If not, consider getting one. They will let you do tasks like normalization and trimming the dead space before and after the track, as well as encoding to mp3 trouble free.

Hi Emdot. Oh, I can edit audio in Sonar with regard to just about everything and my Orion as well as my Samplitude MS 14 can normalise and trim too. Usually I do not normalise but limit and softclip my wavefiles for a peak at -2 db. I do not really know why, except that it feels save, just in case some peaks went out of control. No, I think my problem is that I simply have not learned the MP3 media yet. It is first during the last two years I have messed with it at all. And if you had heard my MP3s for just a year ago, you would be shocked!

I have made a new mix based on your primary feedback with just about every change suggested + a few.

And well done, too. Great track!

Bless you Emdot!

since mp3 doesn't store any absolute transient information you can normalize to -10db rms most of the time. .......most of my tracks where i have the volume i want sit around -15db rms average with a peak of -6db.

Thanks for the advise aciddose. I have been struggling to learn proper MP3 conversion and you have slipped some advise in a totally different thread, which I actually have used (about low + high cut and experimenting with stereo settings ), so it is not the first time I'll take advise from you here. What I still do not get, is why I hear so much quantisation crunch on constant birate. Tried with joint stereo and stereo at 128, 192 kbits/s and 224 kbit. Does volume matter with regard to that? What is the reason for you to stay at -6db peak and -15 db rms?

All the Best

[exxxy]

i use joint-stereo variable bit rate at full quality, or with a max of 192 or 256 to keep the file size a little lower.
i dont bother with any high or low cuts etc, just using the joint stereo and vbr gives a perfect result.


easy to use bonc encoder (has the full range of lame settings) or for ease of use AIMP. with the 'convert with' context menu option enabled is super convenient and you can set the max bitrate for the variable bit rate.

http://www.bonkenc.org/
http://www.filehippo.com/download_aimp/

[Locus M]

http://www.bonkenc.org/
http://www.filehippo.com/download_aimp/

Great! Thanks. At least I have a little to experiment with here.

[aciddose]

"why I hear so much quantisation crunch on constant birate?"

using a constant bitrate you allow for variable quality encoding.

when the audio stream contains more audible information, you limit the encoded bitrate to a fixed value so the encoder must "throw away" the extra audible information.

you should use variable bitrate, which translates to a constant-quality. using lame's vbr -V 3 in "high quality" -h mode should work fine on most tracks.

this way, when the audio stream contains more audible information the bitrate is increased to account for that so that the encoder does not "throw away" any data. when the audio stream contains less audible information the bitrate is decreased so that the encoder does not need to put "empty" or "redundant" values into the data stream.

using a fixed bitrate these days is just plain stupid.

[dacaumodo]

Liking this track very much.

[Locus M]

using a constant bitrate you allow for variable quality encoding.

Thanks for the explanation, that makes sense could you enlighten me on the volume issue also....plz?

Liking this track very much.

Thanks Mate

[Locus M]

Hi again
For you who still hang on, I have made a "final mix" (hopefully) in which some things that disturbed my new gained self-confidence were corrected:

1. Volume set to -6 peak pr. suggestion from Aciddose (do not exactly know why I am doing it for other reasons that AD's own mp3s sounds very clean, so if he says so, it stays so).

2. The first pad was a little out of control in pan and volume fluctuations. There are still fluctuations due to compression but they should integrate better with the music now.

3. First and second pad are better separated now and the second is more present.

4. The distorted tube lead was too much out of control in its random panning. Corrected by limiting the stereo width about 80%.

5. High conga drum is now more present.

5. The highs in the track were a little too "sharp" IMO. EQed for a
more "soft" sound. Should still be bright and clear, though.

6. I have added yet a little more subs to subtly enhance the bass drums.


Aladdin's Quest

Final mix (?):

http://www.box.net/shared/1r1o4i25nf


All the Best

[aciddose]

i like my digital output to be lower amplitude because cheaper devices tend to have a little bit of distortion above -6db - a lot of devices actually consider it 0db with 6db headroom.

-10db rms will generally get your peaks anywhere from -3db to +3db, so you need a limiter to prevent clipping off the stronger peaks. i prefer not to apply a limiter over everything so i use 6db of headroom instead - you generally get this by using -15db rms normalization.

[Locus M]

i like my digital output to be lower amplitude because cheaper devices tend to have a little bit of distortion above -6db - a lot of devices actually consider it 0db with 6db headroom.

-10db rms will generally get your peaks anywhere from -3db to +3db, so you need a limiter to prevent clipping off the stronger peaks. i prefer not to apply a limiter over everything so i use 6db of headroom instead - you generally get this by using -15db rms normalization.

OK, I get it. Will stick to those -6 db at -15 rms then. Well maybe higher rms dependent of how much limiting and softclipping I apply. Thx for info

[aciddose]

when you apply any type of dynamic range compression, limiting, soft clipping, to hard clipping what you're doing is making the rms amplitude move closer to match the peak amplitude.

having -15rms and -6peak is generally a fairly dynamic sounding "open" mix. it's exactly the same as -10rms and -3peak/+3peak, but the variability in the peaks can cause some clipping to occur. some people get pissed off when you create tracks with -15rms because they expect -10rms and don't care if clipping is occurring or not.

a mix that is completely compressed (like gabber, acdc or others) might have something more like -6rms 0peak.

[Aroused by JarJar]

Groovy man!

It's really a great orchestration for all kinds of rhythms.