Oggmachine - What is dialog normalization reduction?

[kilg0r3]

Could somebody shortly explain the function of dialog normalization reduction in Oggmachine? Thanks in advance.

[DSPguru]

-n BOOL
-------

Default: false

This selects if the decoder should use dialog normalization reduction. The
normal dialogue level in a program is defined a reference of loudness, 0db.
The BSI info variable "dialogue level" informs how much this dialogue level
is under 0db full-scale (FS) - or how much headroom there is above the
dialogue level before clip.

One of Dolby's intentions with this variable is to ensure that all dialogue
levels are played back with the same volume, regardless of the program's
amount of headroom. It is good to have when the movie you're looking at is
interrupted by a commercial break, where the headroom varies enormously.
(It prevents blowing your ears off when the break comes.)

This feature is implemented by attenuate everything such that all programs
have 31 db headroom, regardless of its original headroom. For a typical
-27db headroom program, this will case a -4db gain.

[PeterTheMaster]

i read this but dont understand it.
does it mean if a scene in the movie is really loud it will get a lower volume, but the other scenes keep their normal volume?

i thought gain max takes this loudest scene to find the maximum gain and then all the other scenes still have a lower volume than this loudest scene.

i dont know if this is clear, so i ask:
when i use gain max and dialog normalization reduction or only gain max, what exactly will be the difference between the two results?
(i always use drc heavy if this info is necessary)

and another question: besweetgui deals with ogg too, right? does this make the oggmachine obsolete? and i see the besweet gui was updated to support ogg post gain. what about the ogg machine? is it still under development or is it dead?

and another question: i am using ogg now since i was told it is better than mp3 and i read one shouldnt go below 70kbps. so i use quality 0.050 and always get between 71kbps and 74kbps.
is this quality comparable to the mp3 abr 128? becaue 128kbps i felt was kind of a de-facto standard for mp3. is it the same with quality 0.05 with ogg?

and another one: when my movie got undersized in the past i used to reencode the mp3 with a new bitrate calculated as
new_bitrate=floor(old_bitrate/achieved_filesize*desired_filesize)
how would this work with ogg? these quality values seem not linear, rather exponential to the bitrate, right? but how exactly?

and since im on it: sometimes in my ogg files some scenes seem to have a too low volume. i use drc heavy, gain max and dialog normalization reduction. and that lfe to lr -3db (what is lfe again?)
could anyone of them be the cause for this? i would suspect the last one since maybe the sound was louder on the lfe channel? because it happens in the same scenes in the german and english track. or could it be the quality is too low? but this doesnt seem logical too me, quality doesnt have to do with loudness, right?

thanks in advance for reading and answering this long post, dspguru :-)

[DSPguru]

Originally posted by PeterTheMaster
i dont know if this is clear, so i ask:
when i use gain max and dialog normalization reduction or only gain max, what exactly will be the difference between the two results?
(i always use drc heavy if this info is necessary)no difference when normalizing.
but if you wanted to have two streams with same volume, you shuold use it. more info :7.6.1 Overview
When audio from different sources is reproduced, the apparent loudness often varies from source to source. The different sources of audio might be different program segments during a broadcast (i.e. the movie vs. a commercial message); different broadcast channels; or different media (disc vs. tape). The AC-3 coding technology solves this problem by explicitly coding an indication of loudness into the AC-3 bit stream.
The subjective level of normal spoken dialogue is used as a reference. The 5-bit dialogue normalization word which is contained in BSI, dialnorm, is an indication of the subjective loudness of normal spoken dialogue compared to digital 100%. The 5-bit value is interpreted as an unsigned integer (most significant bit transmitted first) with a range of possible values from 1 to 31. The unsigned integer indicates the headroom in dB above the subjective dialogue level. This value can also be interpreted as an indication of how many dB the subjective dialogue level is below digital 100%.

and another question: besweetgui deals with ogg too, right? does this make the oggmachine obsolete? and i see the besweet gui was updated to support ogg post gain. what about the ogg machine? is it still under development or is it dead?OggMachine is a simplified version of BeSweetGUI, only for Ogg encodings.
i use BeSweetGUI, but there are users who find OggMachine easier.
as for adding post-gain fo OggMachine - you're right, need to be done!

and another question: i am using ogg now since i was told it is better than mp3 and i read one shouldnt go below 70kbps. so i use quality 0.050 and always get between 71kbps and 74kbps.
is this quality comparable to the mp3 abr 128? becaue 128kbps i felt was kind of a de-facto standard for mp3. is it the same with quality 0.05 with ogg?sorry.. subjective questions.

btw, lfe=low frequency effect. (the subwoofer channel).

[ChristianHJW]

Originally posted by PeterTheMaster
and another question: i am using ogg now since i was told it is better than mp3 and i read one shouldnt go below 70kbps. so i use quality 0.050 and always get between 71kbps and 74kbps.
is this quality comparable to the mp3 abr 128? becaue 128kbps i felt was kind of a de-facto standard for mp3. is it the same with quality 0.05 with ogg?

I personally compare a 128 kbps CBR MP3 ( Radium, not Fraunhofer Pro ) to a 103 kbps Lame ABR MP3 and a 80 kbps Ogg Vorbis file ... but this is only my opinion. Why 103 kbps ? Lame doesnt do downsampling to 32 KHz for bitrates > 103 kbps ( of course you can always set --44100 or --48000 if you dont want downsampling for smaller bitrates ).
Note that things turn to the advantage of Lame if it comes to higher bitrates ( like > 150 kbps ), as Vorbis is not tuned for higher bitrates yet. That doesnt mean Lame gets better than Vorbis, but the sound difference for the same bitrate gets much smaller or on some samples they even sound comparable .... but for lower bitrates Vorbis is unbeatable IMHO !! ( maybe by AAC, havent tested much yet )

[JohnMK]

A lot of this goes over my head. I do a lot of ripping/encoding of DVDs. Do you suggest I use dialog normalization reduction?

[DSPguru]

Originally posted by JohnMK
A lot of this goes over my head. I do a lot of ripping/encoding of DVDs. Do you suggest I use dialog normalization reduction?
only in PostGain mode.

more info on Dialog Normalization :
http://forum.doom9.org/showthread.php?s=&threadid=33661
more info on PostGain :
http://forum.doom9.org/showthread.php?s=&threadid=32814

let me know if you need mOre info :)

[JohnMK]

What I'm looking for is a ~generic~ one-size fits all answer. :)

Do you think the majority of DVD-rippers/encoders should be ticking the option? Should I use it for Monster's, INC.? Matrix? Braveheart? Star Trek The Next Generation?

[DSPguru]

Originally posted by JohnMK
What I'm looking for is a ~generic~ one-size fits all answer. :)
i believe in my proposed hybrid-gain. can't add more.