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General Audio / Re: Clipping audible when lossy encoding?
Last post by halb27 -
That's why I said if one is paranoid and wants to absolutely avoid clipping regardless of audibility, it is necessary to measure the analog output of individual files and playback devices.
That's the point. I don't worry about the clipping per se, but I'm worried about situations where clipping might be audible.

Thank you for the test, anyway. It shows that when worrying about clipping one should be worried not just about peaks above fullscale generated by the digital process chain but just as well about intersample peaks which occur even with no processing at all.

It also shows a way to avoid clipping to a high degree of probability by giving a headroom of 6db before going analogue.

But if I understand you correctly you think I don't have to worry about audibility of clipping. Unfortunately you didn't go into detail. But it's in line with j7n's post which gave me confidence already.
It is just you daydreaming.

I get the impression that you think that "psychoacoustic" is a dirty word; something to to be avoided. And that you have no idea just how clever lossy compression such as MP3 is. It is actually quite genius... and whether you want to use lossy-compressed audio or not, it is worth taker a deeper look into the whole world of digital music and lossy compression.

Being a maths dunce, I fall at the first formulae. But there are people like JJ Johnston who have a knack of explaining some of the concepts of digital music and compression. Seek this stuff out: you will benefit greatly. or... well, I can only really say that I did!

You seriously need to do some groundwork before letting your imagination loose on this.
Just noticed this in my components menu, should I just run the installer again? What data should I back up if so?
in the same or similar fashion as lossy encoders do, with the exception that there wouldn't be any psychoacoustic evaluation.
That sounds like you'd simply be making the compression less intelligent.  

Does the concept have any merit or it's just me daydreaming?
What are you trying to accomplish?    It seems to me that there are already CODECs that address every conceivable compromise/trade-off...   
General Audio / Re: Clipping audible when lossy encoding?
Last post by bennetng -
Just did some experiments. My Realtek ALC892 has -5dB analog I/O differences with my X-Fi Titanium HD, so it means my X-Fi has 5dB headroom when recording the output of ALC892, which is good enough to see how ALC892 handle 0dBFS+ signals.

The attachment has two test files at 44 and 48k peaked at -1dBFS (Beware of loud volume!), I encoded them to different formats and examined their peak values. Take into account that these are mono files so their bitrates are actually pretty high. As shown in the attachment, opus has the highest peaks in both 44 and 48k, but exceptionally high in 44k, which I guess is caused by resampling rather than the lossy encoding process.

However, when I played the lossless version of both test files and captured ALC892's output, both 44 and 48k files needed to reduce 6dB to completely avoid clipping. So it means the lossless files themselves have DAC intersample peaks from +5 to +6dB, which means Track Peak from 1.778 to 2, not too far from the lossy peaks isn't it? Vorbis peaks are low because it encoded the whole spectrum, unlike other codecs with obvious lowpass.



That's why I said if one is paranoid and wants to absolutely avoid clipping regardless of audibility, it is necessary to measure the analog output of individual files and playback devices.

My ALC892's RMAA results are pretty good.,111980.msg922878.html#msg922878
My post said ALC887, it was a mistake. It was ALC892 on my ASRock B85M Pro4 motherboard.
The "wouldn't be any psychoacoustic evaluation" and further coding gain from frequency domain coding are at odds.
This also may be interesting Oldfield watermarked at qobuz?

I'm looking at two speakers models while i try to decide which to set up for engineering and playback.  I want a near flat response of course.  One model has a frequency response variance of +-0.75dB and one has a frequency response variance of +-6dB.

Can one make up the difference with DSP and room treatments?  This article: leans towards "no", but that was 5 years ago and things are always advancing in the DSP realm.

What are your current thoughts on this?

Yes, I use Math Audio Room EQ. It'a s freeware for foobar. I use foo_record and VB Cable "driver" to stream all my Windows sound thru foobar.
I'm with studio monitors. They're really one of the best. I though they're amazing... All people think though that...
Today if I stop the DSP, I'm wondering how is that possible:
1. Stereo effect is immediately reduced
2. High become strange
3. Sound is no more crystal clear, I loose details
4. Most instruments sound like a mix... more like a single distorted sound... unclear... instead each string as perfect distinguishable detail.

I don't like the fact I lost so many years with my audiophile friends with passive corrections, when the true is one: only digital correction could be perfect. Of course only EQ without phase correction always makes sound worse.
Start reading at,89818.0.html and the links therein.

UMG owns Decca and DG and EMI ... I have not seen information about watermarking physical CDs, so UMG are doing their very best to rip off their paying customers and encourage 3.14159rates.
Let's see ... how many EMI releases have I bought in any format since they started doing the pesky copy protection some fifteen years ago? I think I got Deep Purple: Bananas, Kraftwerk: Tour de France soundtrack, and Iron Maiden: AMOLD.  IIRC, the latter was a promo, and the others I didn't know were EMI (DP was on Sanctuary in the US). I own some seven thousand physical units (most are full-lengths), but I don't need to buy from Universal.
Scientific Discussion / Windows Audio Calibration
Last post by emo_hp -
Hi All!

I'm new to the forum. I like foobar and all the plugins. But I faced one problem which I cannot solve as a developer.

So in short:
Is there a way to use foobar DSP plugins on Windows audio level like Audio Processing Object (APO) first introduced in WIndows Vista.

My setup:
Foo_record plugin with "record://" link added in the playlist, played with 50ms (minimal latency) and some DSPs added. Installed VB Cable (In/Out virtual audio driver) and it is selected as default windows sound device, foobar is set to use the physical sound card (in my case audio DAC).

All this works great with Mathaudio Room_EQ, my calibration microphones, and sometimes dynamic compressors. But the problem is I got some increasing delay over time (especially after hibernate) where I need to restart playback periodically. Also on my atom tablets the delay is bigger.

I also managed to "copy" calibration to cheap Panasonic Condenser Mics (2-3 EUR) without any additional electronics and the result is quite good, difference is only hard-noticeable when a quality Audio DAC is used (where I believe the problem is that I calibrate a particular Mic to a particular laptop input, which is standard lo-quality sound card).

So the real problem is: as a C# developer I don't know how (and if it deserves the efforts if possible) to create an "foobar" source application directly controlling the plugins and output setting (which is based on C++). But generally even automated, it would be external component, not on APO level almost like "driver". Of course it could be converted to be APO compatible/installable somehow. But generally if something already exists, something like "APO foobar Plugin processor" would be great. There is a ton of useful plugins that can make whole audio to sound really hi-end...

If someone can help me... thanks in advice.
I'm open to questions for non-experienced users who want to know more about audio calibration, setup, mics and so.

EQs are 1, room calibrators - absolutely different story. Most audiophiles don't like EQs. And it's normal - they make linear response but phase is totally out of sync - not like sound but more like a noise with a particular frequency. On other hand additional phase correction (+ resonance canceling as a bonus) always make better sound even on a hi-end system (as no system is absolutely in sync thru the whole response). Removing that "correction" (a bad word for audiophiles) makes the sound ugly, unclear and so... I tested tenths of systems including hi-end ones... almost no exception... only one - it was an auto-correcting car amplifier where nothing can help consistently.

Thanks in advice,