Stereo Tool

Since I'm using Breakaway, the output is sent to breakaway line3. I'm using BreakAway RTA to compare the input and the output, when I change that option the output doesn't match the input stays the same as compressed. (I'll re check everything in a few hours and post if it was my fault or not xD ) I have no use for this option, I just thought it might have been bug or something.

works fine, my fault D:
No sir, I'm using direct soundcard cause BreakAway sends the output from ST through another larger buffer that seems to fuck up when ever my CPU goes above 50% usage. Lots of popping, clicking and annoying shit when using 4096 in ST. But with direct to sound card that path is skipped and it uses I guess Stereo tool's buffer, which seems to be smaller and less prone to fuck ups due to my intense computing (photoshop and 60 tabs in firefox). I like pre-emphasis since it boosts the high freqs a lot! (that's what its supposed to do right? Make shitty mp3s sound better?) Right now I just have the the last 4 bands (high bands) turned up so it sounds kinda like what pre-emphasis (50 US) on sounds like. I have no need for the send back to winamp option, don't worry about that. x)
How do you know what the CPU usage is? Winamp's output to disk writes a wav file not a text file that shows CPU usage. Unless I'm completely mis-understanding you. xD
I figured since you wrote the program you'd be able to track its performance through what ever IDE you use.
Using Netbeans (i do java, aren't I cool?! lol) I can easily check what each thread is doing, along with the exact CPU usage and the exact resource usage. You don't do something like that?

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Some measurements, Latency vs CPU Usage
- ST v6.00 (512samples) vs. Breakaway ASIO (LowLatency/no-phase-linear mode), soundcard MAYA44 USB ASIO
48kHz, L/R In => L/R Out

- BBP ASIO: Latency about 18ms, avarage CPU Usage 13%
- ST 6.00 Standalone: Latency 20ms (buffer size at normal output is at 0.011), avarage CPU usage 11%

This look nice. But Difference in sound is large. BA Asio sounds much better.

I'll test latest ST beta...

The answer is: No.

http://en.wikipedia.org/wiki/FM_broadca ... e-emphasis
To make shitty mp3s sound better (most of them have high frequency distortion artifacts) you should consider to spend more time with Stereo Tool's Multiband settings (especially EQ before MB option in conjunction with compression/limiting/clipping with a focus on high frequency range).

So the pre-emphasis option in the FM settings is to reduce noise by boosting the high freqs?
Is the pre-emphasis option in loudness the same one that is in the FM settings?
Lets clarify that I am not using the pre-emphasis option in FM, I just like the way it boosts the high freqs.

In the end it still does make low bit-rate Mp3s sound "better" since its boosting the highs that MP3 is so nice to reduce for us. xD

Off topic question, why does america always have to use standards that don't comply with the rest of the world? Why make things hard?! >.< Metric FTW! lol
(For the sci-nerds, SI ftw tooo!!)

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Yes, but there is never pre-emphasis (FM broadcasting) without de-emphasis (FM receiver).
Yes and No !
This option in Loudness section applies pre-emphasis and de-emphasis !
http://www.claessonedwards.com/forum/vi ... ?f=5&t=172

Even if you use low CBR 128 (old standard) then we are talking about filtering frequencies above 15.5 KHz.
But with higher VBR LAME this is absolute irrelevant in practice.
So this is not the problem of MP3 encoding.
The questions is:
What happens from 1KHz to 14 KHz ?
( Compare newer LAME versions @ CBR 128 with LAME 3.93.1 !!! - so newer is not automatically better !)

Ok, it is not wrong to have higher frequencies (up to 20 KHz or more).
But this is less important, because most people older then 35 cannot hear frequencies above 17 KHz (and many younger people today have already serious ear damages).
So theory (or internet propaganda in certain audiophile circles) is one thing.
But reality is another cup of tea.

In most cases:
Lossy encoding is not the problem.
The problem today is:
Mastering !
As long as you have so much sources on CD with massive distortion and clipping or other problems of unbalanced frequencies, it does not help to have frequencies above 20 KHz.
Sometimes a CBR 128 sounds even better (less stressing).

I'm about to turn 40, and I'm on the edge of being able to hear 17K anymore, so I can confirm what you said... I put on a lowpass of 18K because there's just no point in keeping the rest of the signal for me anymore...

I was going to say that! lol
So sending an FM pre-emphasised signal to my receiver would be a bad idea. Also the loudness only goes up to 30, while FM gives 50 & 75. higher values = more emphasis?
Cause the replies I've been getting, though very interesting and helping me understand these things are assuming that I'm using the FM pre-emphasis.(at least thats what I feel like) I tried it, and i liked how "clear" the audio was but I'm not using it. I just pushed the EQ for the last 4 bands higher instead, gives me the same result.
If all the music files on the internet were VBR v0 I'd be a lot happier. If they were all flacs I'll be content with life. xD
If you're talking about audio quality newer might not always be better, but there should be an improvement in size. Aren't Mp3s about size? Audio quality wise to me its okay and preservation wise it sucks.
Just hit 18 and I wouldn't say i have perfect hearing but I can still register 20khz.
Would you say that the higher frequencies still give the overall music some type of "life" even if you can't hear the higher frequencies.
Looking at speakers, they vibrate (I like to use technical words to sound smarter. really. xD) differently with different signals so if you cut off a whole section say from 18Khz and up those signals are not sent to the speaker, which then doesn't vibrate at those frequencies. To me that sounds like a broken instrument.
The difference between theory and reality is what makes any type of audio work an art, its not something you can just set to 11 and expect the perfect sound.
Funny thing I was drinking a cup of tea when I read this.

Hey michi if you want to PM me with a reply or start a new thread that'll be cool. This is taking up a lot of room here when it really has nothing to do with Stereo tool news xD

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BETA601-015:
- 'Red output bar' issue fully solved for sample rates upto 48 kHz
- Fixed issue with higher sampling rates in previous BETA (there were 'gaps' and clicks in the audio, when very high frequencies were present!)
- Multiband Steepness CPU usage reduced by over a factor 4 - so now there's little difference with the Multiband in version 6.00 anymore. Behavior is nearly identical to that of the previous beta. Total CPU load for this BETA (FM Loud Bass Europe preset) is 4% higher than 6.00.

I have added a new issue to the todo list: "Fix Highpass filter for higher input frequencies.". Currently at higher input sampling rates the maximum highpass frequency is too low.

Winamp DSP plugin: http://www.stereotool.com/download/dsp_ ... 01-015.exe
Stand alone version: http://www.stereotool.com/download/ster ... 01-015.exe
VST version: http://www.stereotool.com/download/vst_ ... 01-015.dll
VST version (No SSE2): http://www.stereotool.com/download/vst_ ... 01-015.dll
Command line version: http://www.stereotool.com/download/ster ... 01-015.exe
Linux command line version: http://www.stereotool.com/download/ster ... ETA601-015 [not available]
Linux GUI version: http://www.stereotool.com/download/ster ... ETA601-015 [not available]


What remains for 6.01:
- Multiband: Adjust band 3 (and others?) 'soft limit' behavior at very low latencies. Maybe: Also check clipping (also for band 3, might be re-enabled!). 1 hour
- Multiband: Adjust extreme EQ settings before processing to avoid issues caused by steepness. 1 hour
- Multiband: Add 'Enable' button for steepness. 1 hour
- Performance: Multiband steepness: Moving UP can be done inside current loop, no separate loop needed. Maybe down too. Currently Multiband is FAR more expensive than before. - DONE, solved in a different way. 1 day
- Performance: Multiband: Remove sqrt(sqrt(cos())), pow(x, .75) etc. - too expensive, replace by lookup table. 2 hours
- Reduce downsampling frequency because currently very high frequencies (21-22 kHz @ 176.4 kHz input sampling rate) in the input can cause spikes even if Hard Limit is used. 1 hour
- And I need to check how much the performance is impacted by the latency improvements, in the case where no upsampling and downsampling is needed. This seems to be impacted way more than I expected (could also be Multiband steepness). ???
- MAYBE: Make Steepness smarter. That would much better preserve the audio, especially at very low latencies! How: Instead of setting all the levels at AT MOST the level of neighboring bands + a bit, combine adjacent bands, determine total output level, and then fix it such that this combined output level is approached more. (So one very low, one very high --> one a bit less low, one a bit less high instead of both low).
- Fix NOISE GATE behavior in VST plugin
- Fix 'red output bar' issue. DONE 1 day?
- Attempt to set GAUSS back to 0 - gives MUCH better processing of most filters (no high frequency noise). BUT: Loudness effect in Bjork - It's Oh So Quiet - can that be resolved in another way? 1 day?
- Fix Highpass filter for higher input frequencies.

Questions:
* Multiband: Question: Is Steepness behavior ok?
* Loudness: QUESTION: The changed Punch behavior, is that good or bad?

Check:
Check difference in behavior between 44.1 and 48 kHz input for multiband! This could potentially result in really big differences.

Ah, sorry, forgot to respond to that one. I've added a 'TODO' to switch Steepness processing off. 250% is currently the closest you can get to 6.00 sound (at high latencies anyway, at low latencies there was 'something like' Steepness but it was far less effective and not configurable; between band 1/2 there was also a Steepness at latency 4096).