michi95

[quote author=”Orio”]BAE is set to bypass in my machine, and i only APPRECIATE the FM mode: "that only worth the $29.99 price"[/quote]Not everyone has the courage to speak that frankly.
So BA (Live) is set to bypass in my machine, too ! 😆
I use it for hosting other plug-ins only.
You see sursprisingly we have more in common as you might have thought.

[quote author=”Orio”]From the point of audio quality you have to listen to the Track the way the Artist want you to listen to it : From a Vinyl Records, not on your computer my friend[/quote]I do not agree, because IMO most artists (musicians) have only a very small influence to the recording/mixing/mastering process.
Every kind of recording (stored on vinyl, CDs or whatever analog or digital system) is only a limited reflection of music and always somehow artificially altered (damaged !).
The only way to listen to (traditional) music is to hear it while it is live performed, not on your turntable with a needle that tortures vinyl (music frozen in plastic !) ➡
http://www.whatisvinyl.com
my friend 😈 😆 8)

It happens again and again to me that I hear fantastic music at a concert and then I am disappointed when I afterwards listen to the CD (or vinyl !), because it sounds deadly boring.
And this is not only a problem based on too much compression and limiting.
For people like me that are not perfect musicians the principle of multitrack recording is a great help.
But very often multitrack recording leads to sterility, because it is somehow a perverted technique to record something in small pieces that was meant to be played in one go by the whole band.
We all know most guitarists are batshit crazy and this was true for Les Paul (R.I.P.) the inventor of multitrack recording, too.

E.g. Australian singer Sarah Blasko’s studio recordings are only a very weak interpretation of the musical energy that erupts while her live performances with her musicians.

[quote author=”leecovuk”]One of my problems is I find it hard to determine when video is slightly lagging behind audio or vice versa. Randomly trying slightly different delay values drives me towards insanity.[/quote]Don’t drive yourself crazy !
[quote author=”leecovuk”]I wish there was some kind of magical ‘automatic V/A sync checker and adjuster’ in DVBViewer or a Directshow filter. Or maybe a slider control, which I could adjust back and forth in real time to adjust sync. That could be in DVBViewer, Directshow or Breakaway[/quote]It is not necessary at all (especially it is not possible inside Breakaway or any other audio part of the chain, because it has to be done with the video – it is a question of logic) !
Not necessary at all, because:
It is a static delay !
If it is not, there must be something else wrong on your system (a hidden CPU eater that leads to performance instability while HD decoding ?).
Using the same directshow filter chain (decoders) there should be no change !
Only when you switch from standard TV to a HD channel (and vice versa) you change the directshow filter chain (rebuild the decoding graph inside DVBViewer with other decoders needed for HD).
[quote author=”leecovuk”]I wish there was some kind of magical ‘automatic V/A sync checker and adjuster’ in DVBViewer[/quote]There are thousands and thousands people that use DVBViewer (or other DVB apps) for standard and HD TV without any need for such thing.
If there is noticeable delay, than it is your special problem – a problem caused by your system (a decoder like DivX H.264 or whatever else ?), because you have written that even without BAE you have a noticeable delay.

Have you ever seen a standalone TV with a manually audio/video sync adjustment option ?
It is not necessary (or it should not be) !
There are only some very rare exceptions (cheap, low budget TV shops, etc.).
But on regular TV stations it is very seldom that the broadcasted content is already out of sync (it is more or less three or four times a year that I manually adjust the video delay, but most of the time I use none !).

[quote author=”michi95″]If you (in theory) have audio/video in perfect sync it needs (IMO) adjustments of +/- 150 ms (or more) to notice a delay.[/quote]
➡ Use steps of +/- 50 ms to determine the static (!) delay !
Even for AVI muxing (in the past) I have never used smaller steps than +/- 25 ms, because everything else (smaller steps) is beyond the human reception capabilities (because the human brain has its own built-in sync compensation adjustment processing).
So, I am not surprised about your problems you have described:
[quote author=”leecovuk”]Randomly trying slightly different delay values drives me towards insanity.[/quote]You are a human being and not a machine !

Though human individuals are very different in their subjective recognition capabilities.
I know some people that do not notice even a delay of 250 ms.
These people need a 400 ms delay to feel that something is wrong.
We are talking about a/v sync for playback (and broadcasts are playback).

For real human to machine interaction ➡ live processing while recording: singing, playing (virtual) instruments you cannot live with delays.
There you need ultra low latency (less than 5 ms using ASIO).
But 25 ms more or less delay watching TV (videos) is IMO irrelevant (as long as the general present a/v delay on your system is ok).

Of course, if you want (with ambitions to win the Nerd Of The Year award) you can calculate the additional delay introduced by BAE, the driver of your soundcard and every decoder, etc. and add this as standard delay inside ffdshow video processing options (include it in every ffdshow preset you use).

And keep in mind that with every change of your position in the room relative to your loudspeakers the audio delay changes also.
One meter makes a difference of 3,3 ms for an audio signal (the sound through the air) !
While the light needs only 0,000000003333…. ms for a meter !
This is not exact, but this way it looks more elegant !

For some people mathematics is their hobby. 8)
For other mathematics is a sick perversion. 😥

[quote author=”JesseG”]and I’m taking most of my good personal secrets to the grave. 😆 [/quote]And then the worms will find out every little secret and will build the best (and final) audio processing unit ever and will live in fabulous wealth ! 😆

[quote author=”leecovuk”]Regarding any video processing I do in ffdshow, I routinely use Picture Properties to raise brightness and lower saturation for all pc video playback that can use ffdshow as well as live TV.[/quote]On XP I use Overlay Mixer as video renderer (and do brightness, contrast, saturation and hue adjustments with DVBViewer’s built-in control GUI) – no need for ffdshow.
So, if none of the on Windows 7 present alternative video renderers can access hardware colour control, you should try to adjust these (fixed) correction settings inside the catalyst driver control.

[quote author=”leecovuk”]ffdshow audio is set to use ‘Mixer’ to force 2 channel stereo output and usually I use no other processing in there.[/quote]For sure there are differences (but relative small) using different codecs, but not that massive in relation to the final delay of video/audio sync.

But ffdshow is special:
There are some (nightly) builds that do not work for me -> results in slowdowns and other unwanted effects.
That’s why I still use an older version:
ffdshow_rev2968_20090525_clsid

-> try different ffdshow builds !

[quote author=”leecovuk”]In the DivX H.264 decoder, I again have brightness up and saturation down.[/quote]Does this mean that you do not use ffdshow (processor) in conjunction with HD (DivX H.264 decoder) ?
DVBViewer ➡ Plugins ➡ Video Postprocessor ➡ Exclude for Current Decoder !
With additional ffdshow processor you loose important hardware acceleration (GPU) functions.
Have you tried to use the ATI MPEG Video Decoder (for H.264) instead ?
(The decoder is not included in Catalyst download versions ! You have to install first the old driver from the original installation CD that came with your ATI HD 4350 card and than update it with the latest Catalyst download version !).

[quote author=”leecovuk”]As you know, the problem with setting delays in ffdshow is finding a magical ‘suits all’ setting.[/quote]If you (in theory) have audio/video in perfect sync it needs (IMO) adjustments of +/- 150 ms (or more) to notice a delay.
You can try use a standard video delay of 150 ms inside ffdshow processor (decoder) to suit standard and HD TV.

Or use the "Preset autoload conditions" in ffdshow:

➡ on decoder match ➡ DivX H.264
This way your preset with additional delay (try 300 ms) is only used for HD broadcasts but not for standard TV.

[quote author=”JesseG”]Actually tho… BAE automatically changes what the primary soundcard is, depending on if it’s running or hard bypassed or closed. 🙂 so you shouldn’t have to change any apps from primary…[/quote]Yes, as long as the "Manage" box is checked (default).
I had forgotten to mention, that hostmax has to uncheck this box.
In this special case (having the odd wish to use SRS post Breakaway) it is necessary that Breakaway Pipeline is not the default audio playback device (because SRS Labs Audio Sandbox must be) !

And in this (strange) context (to enable the use of Breakaway Processing pre SRS) it is necessary to select in every application (winamp, foobar2000 or other multimedia/audio players) "Breakaway Pipeline 1" manually as output instead of "Primary Sound Driver" (because SRS is the primary sound driver or default system device).

Though I highly recommend to not use SRS at all:
1. The quality of it is processing is really a joke (or a shame for a company like SRS Labs – pro ? 😆 😆 😆 )
2. All versions above 1.9.04 (1.10.1 and 1.10.2) do not work on my XP system.
Even better: Some elements of its driver crashes my DVB application when I try to use a BDA driver based DVB-S card.
I had to delete the SRS driver manually.

➡ Forget this useless SRS Sandbox
and use Volumouse instead.
➡ "The cursor is on screen edges" option to enable the volume control using the mouse wheel !

[quote author=”JesseG”]As far as with analog style band-based processing, sometimes less (bands) is more[/quote]Yes.
[quote author=”RodeoJack”]it’s only logical that a 64 bander should be right around the corner, eh?[/quote]More bands mean a higher complexity of the whole system.
The question is how good you can control a very complex system ?
And to emulate virtual the principle of analog band processing is of course not a way to reach what I have described as spectral intelligence.
We need better digital realtime analysis and systems with a wider range of adaptive reaction.
[quote author=”JesseG”]We’re still not very far from the stone age, technologically speaking. I have no doubt that someone will figure out new awesome ways of processing audio that don’t sound worse than doing nothing.[/quote]So, do you think that the sources pre Breakaway (of this old tracks) sound better than what we hear processed with Passive Aggressor ?
The question is:
Does Passive Aggressor enhance the music besides levelling ?
Yes or no ?
I think it does enhance it (the spectral balance).
But I believe that in the future (in a few years or a few days !) it will be possible to reach a higher quality (and noise will be more or less irrelevant).
We will come to the point when it will be impossible for someone without the historic (musical) and technical background knowledge to hear if it is a recording from 1967 or from 2003.
You may call this a fake (processing), but as long as it sounds clean and natural (in contrary to the fact that you hear today very often the processing itself) I don’t care.

And in this definition Passive Aggressor is also a fake processing, because while listening you forget that it is processed (many good mastered tracks very close to the source). 🙂

[quote author=”jameskuzman”]Very consistent spectrally source-to-source.[/quote]Ok, it is better than with other presets.
But honestly, especially listen to this Moody Blues track it is obvious that there is still room for further development.
The grade of spectral intelligence is limited to the structure of the Breakaway core.
So, IMO Passive Aggressor is not the ticket to audio processing’s nirvana.

For my ears it (NIWS) still sounds as if it is a song of Muddy Blues.
I am sure it is better than the source version (Jesse had used here), but still far away from the quality when you use your imagination.
Yes, I know it is a very old (low quality) recording, but more spectral enhancement is not (should not be) a mission impossible today (even with plug and play processing).
The same is true for Pink Floyd’s "Lucifer Sam" (interesting Jesse took this and not "Interstellar Overdrive" or "Astronomy Domine").
Maybe we do not need 32 bands, but 5 PA uses (or 7 as possible maximum of the BA core) IMO is not enough to reach a higher grade of spectral consistency (without annoying artifects ➡ de-essing for bad masters ?).
And in the end spectral and dynamic consistency are two sides of the same medal.
To date Passive Aggressor is only the best compromise in comparison to all the other BA presets.
[quote author=”JesseG”]There will never be a "winner" in audio processing. And for that, we should all be very thankful.[/quote]
So, ASAP is close enough.
Leif is cooking other things now.
And we all will be surprised what tomorrow brings. 🙂

Maybe a naive question (sorry, but I am not into FM):
Are those demos generally:
A. without pre-emphasis/de-emphasis
B. with pre-emphasis (without de-emphasis)
or
C. with pre-emphasis and de-emphasis
❓

Or is the processing itself different for 75µs and 50µs emphasis (besides the pre-emphasis difference) ?