GUIDE: How To Properly Encode Dolby Digital Audio (AC3)

[Black Hole]

Well, you should test that the AC3 in that file is indeed 5.1, because it may perfectly be a 2.0 file. At 128 kbps you got 64 kbps per channel, which is not that bad for AC3 compression.

[Buggle]

How about the effect of sample rate on the endquality of the AC3? All DVD's I've come across are 48kHz, but SoftEncode lets me select any other resolution as well. Say my source is 44,1kHz, would it yield better results if I converted this puppy to 48 kHz before encoding?

[ursamtl]

Originally posted by Buggle
How about the effect of sample rate on the endquality of the AC3? All DVD's I've come across are 48kHz, but SoftEncode lets me select any other resolution as well. Say my source is 44,1kHz, would it yield better results if I converted this puppy to 48 kHz before encoding?

The sample rates determine the end use of the file. 44.1kHz files are normally used for creating AC3WAV files that can be written to a regular CD as if they were tracks for an audio CD. If this CD is then played back by a DVD player hooked up to a decoding receiver via a digital coax or optical connection, it fools the receiver's decoder into thinking it's an AC3 stream coming from a DVD and thus you have a surround audio CD!

48kHz is for use as DVD video sountracks.

In general upsampling will not "improve" an audio file per se. The higher the sampling rate when recording or digitizing sound, the higher the frequency range. However, once the sound is digitized or recorded, upsampling will not add to or improve the information that's already there. Besides, if you had the same sound recorded at 44.1kHz and 48kHz with all other conditions being equal, it would be extremely difficult to hear any difference. The upper limit of normal human hearing is about 20kHz. So whether a sound source is digitized at 44.1kHz with a high-frequency limit of about 22.05kHz or 48kHz with a high-frequency limit of 24 kHz, only your dog will notice a difference! :)

[oayz]

Guys, what is the right way to adjust volume of the AC3 file? I have several clips I'm going to use for DVD authoring and I need to set their relative volumes.

I assume I can do this by changing DialNorm parameter - without recompressing. Is it correct? Are there any tools? Does anybody know where in thje AC3 file DialNorm is located?

I did try to encode AC3 with different DialNorm and they play with different volume in WinDVD and standalone DVD player. PowerDVD and Media Player Classic play them with the same volume - seems like they normalize all AC3 clips and pay not attention to DialNorm.

[livius76]

i have a movie sound track with -24.6 RMS
found in SoundForge...so i have to put in AC3Machine an Attenuate Volume by -31-(-24.6)=-6.6db ~ -7, or a Gain -7 ? :confused:
thk i try Acid Pro for a strightfourd method...:(


Thx a lot & keep in touch!

[maa]

The first two links in the guide no longer exist - please delete this post after correction - thanks

maa

[KpeX]

The first two links in the guide no longer exist - please delete this post after correction - thanks

maaThanks, edited in archive.org links.

[SomeJoe]

Here are some updated links (Dolby has updated their site):

Standards and Practices for Authoring Dolby Digital and Dolby E Bitstreams (http://www.dolby.com/assets/pdf/tech_library/20_Dolby_E._Standards.P.pdf)

Dolby Digital Professional Encoding Guidelines (http://www.dolby.com/assets/pdf/tech_library/46_DDEncodingGuidelines.pdf)

And here's a new one that has a nice explanation of every metadata parameter in an AC3 stream:

A Guide to Dolby Metadata (http://www.dolby.com/assets/pdf/tech_library/18_Metadata.Guide.pdf)

[Sakuya]

The second page (http://pages.sbcglobal.net/wilsondr/ddexacid2.gif) is the Bitstream Information. Set these parameters are appropriate for your source material.

I was wondering what the "Dolby surround mode" is for? If I have a normal TV source audio, should I be setting it to NOT? Thanks for this great guide!

[desta]

Hi all...

My first post here, so please excuse me if I shouldn't be adding this as a reply rather than creating a new thread, but this seemed the most appropriate place for it to go.

I have a 5.1 AC3 file that I have separated into it's 6 mono channels via BeSweet (BeLight to be exact). I've then loaded these back into Softencode to create a new AC3 stream, but with a data rate of 384 instead of the original 448. I used softencode to get the specific information from the original AC3, which looks something like this...

File size: 195,442,353 bytes
AC-3 File type: Non-Intel byte order (0x0b)
Total frames: 109,063
Frame size: 1,792 bytes
Sample rate: 48,000 Hz
Data rate: 448 kbps
Audio coding mode: 3/2 (L, C, R, l, r) LFE
Bit stream mode: Main audio service: Complete main
Dialog normalization: -27 dB
Center mix: -3 dB
Surround mix: -3 dB
Copyright: On
Original: On
Start time: 00:00:0.00 *
End time: 00:58:10.02
Room type: Large room, X curve monitor
Mix level: 105 dB SPL


... I've used these settings to re-encode, with the obvious exception of the data rate, and then created the new file.

After finishing, I played back the new AC3 and found it was much quieter than the original. To make sure it wasn't just my ears playing tricks on me, I used BeSweet again, but this time on the new file - separating it down to new 6 mono channels. I opened them up in Sound Forge, compared them to the originals, and there was a clear drop in the peaks.

I repeated the process again, but this time turned all the filters and compression off in the preprocessing tab. Re-encoded again, and again the same results.

Thinking it might possibly be down to the change of data rate, I went through it all again, and this time kept it at 448 - the same as the source. Once again it produced the same results as before... a quieter stream.

Now I'm at a loss to think what I'm doing wrong. All I really want to do is duplicate the original AC3 file, keeping it the same volume. Other than raising the levels on the main arrange page (which I would've thought would introduce clipping), I can't think what to do.

Any thoughts or suggestion would be greatly appreciated.

Apologies again if I'm posting this in the wrong place. I did look around the forums for any answers, but couldn't find any.


Edit: Just to add, the mono waves taken from both the original and newly encoded AC3 files were created using the '32bits Mono Waves' option in BeLight... not the '16bits'... whether that would make a difference.

[tebasuna51]

@desta
Maybe...
When you decode in Beligth, for reencode purpose, don't use Dynamic compression (Azid settings). Each time you use Dynamic compression the peaks are attenuated. With Dynamic compression unchecked you have the full original Dynamic range.

[desta]

Hi tebasuna51...

Dynamic compression was turned off at all times in BeLight. Cheers for the suggestion though, and replying.

:)


edit: actually, while I think of it... two things:

- should I have ripped the wavs from the original ac3 as 16bit, rather than 32bit?

- when re-encoding in softencode, with no actual changes to the wavs themselves, should all the pre-processing filters and compression be turned off or left checked?


Thanks again.

[Hans Ohlo]

Hi tebasuna51...

Dynamic compression was turned off at all times in BeLight. Cheers for the suggestion though, and replying.

:)


edit: actually, while I think of it... two things:

- should I have ripped the wavs from the original ac3 as 16bit, rather than 32bit?

- when re-encoding in softencode, with no actual changes to the wavs themselves, should all the pre-processing filters and compression be turned off or left checked?


Thanks again.
hi,
i have the same problem. i extract 6 32bit wav files (also tried 16bit) with belight (nothing turned on) and use acid to encode them to ac3.

maybe i did not handle acid right but the result sound dimmer. i have serveral questions:
1. how do i map the wave files to the distinct channels (i use the sourround panner from acid, switched all channels off except the one of the file, i also upped the volume to 0db of each file/track. in the sourround panner there stays a -6db at the used channel...)
2. i can't switch off the center and sourround mix level in the bitstream options

how does encoding 6 discrete files exactly and with no lowering or attenuating to the channels and how do i get the same dynamic and volume as the original source (i only had to cut some minor stuff)?

[leonid_makarovsky]

My expirience with Dalog Mormalization is following. I recorded a VHS to my hard drive and wanted to make a DVD out of it. After doing research I concluded to have Dialog Normalization set to -31dB. I unchecked all other options in all tabs. My DVD player was Philips DVP-642 which was connected to my Onkyo stereo receiver using RCA (analog connection). During playback I noticed some sort of clipping on right channel during loud parts. When played back in my computer using WinDVD and PowerDVD, the clipping didn't take place. (my computer's soundcard M-audio was also conected to my stereo receiver using analog connectors). I thought for a while and re-encoded with -27dB. Clipping diappeared, but the overall volume was lower than if I had this DVD with Uncompressed PCM from the same WAV file.

Recently I bought the 7.1 receiver. I connected my DVD player to the receiver digitally. When I played the DVD that I enceded with -31dB, clipping didn't take place. And the overall volume was louder than the reference DVD with Uncompressed PCM soundtrack. Playing -27dB DVD one against the reference (PCM) DVD showed the identical volume of soundtracks.

Basically I use AC3 2.0 to have the soundtrack that is maximum close to the original WAV file. I usually use Uncompressed PCM soundtracks, but when the video is longer than 75 minutes in order to fit it on DVD I prefer to use AC3 than to sacrifice video bitrate to make it lower than 8mbs.

--Leonid

[craftech]

It is hard to believe this post hasn't drawn much criticism yet. There is absolutely no way that the method of determining dialnorm at the start of this thread will produce anything but a muffled, muted, and terrible sounding audio movie track.

I am sure that any of the subsequent posters who asked the original poster about this method and never reposted their results came to the same conclusion.

The reason is simple. The standard for dialnorm was not based upon the overall soundtrack, it is based upon the dialog. The simplistic method of calculating it described at the start of the post cannot work because the original posted doesn't understand the basis for dialnorm in the first place.

Considering the level of equipment necessary to determine it properly you are better off setting the dialnorm to -31 dB to start with and working your way down by trial and error leaving the other two parameters set to "None".

[leonid_makarovsky]

Considering the level of equipment necessary to determine it properly you are better off setting the dialnorm to -31 dB to start with and working your way down by trial and error leaving the other two parameters set to "None".

And what would the error be? Would there be clipping?

--Leonid

[craftech]

And what would the error be? Would there be clipping?

========

Distortion, but moreover the "levels" would be audibly too high. Ear-piercing dialog or singing voices if they are present.

Craftech

[leonid_makarovsky]

And what would the error be? Would there be clipping?

========

Distortion, but moreover the "levels" would be audibly too high. Ear-piercing dialog or singing voices if they are present.

Craftech

Ok, as I previously said, when I use analog RCA output for audio going from my DVD player to my receiver, I do notice clipping (distortion) if I set it to -31db level. When I use SPDIF out, I don't notice any clipping or distortion at -31db. Now my in my WAV file (which I use as a source) the peak level is no more than -0.1db to 0db. And the average loudness is most likely ranging from -3db to like -7db. So you're saying there's a chance to get the distortion? Thanks.

--Leonid

[drob]

Very helpful guide, however one point is still unclear, is it preferable to use the "use equal loudness contour" when calculating the dialnorm (referring to sound forge)?

[3ngel]

I was compressing a 2 chan 48khz to ac3 using Soft Encode.
The wav had a freq peak around 23khz, but the resulting (decoded) ac3 has a hard upper limit of 20khz (a kind of mp3 freq view).
Is that normal? I'm doing something wrong?
Thanks

[tebasuna51]

Is normal.

You can see in Soft Encode -> Encode Settings -> Audio bandwidth, the max value is 20.3 KHz for a 48 KHz wav with max Data Rate.

A wav 48 KHz is really 48000 samples/sec, then a 24 KHz tone have only 2 samples for period (a triangular wave instead a sine curve). The hard upper limit 20.3 KHz is a reasonable value for this input signal.

[3ngel]

I see, thank you very much.
Dts has this same limitation?

[tebasuna51]

To preserve 23 KHz info you need use at least 96000 samples/sec.
And in DTS specs:

"DTS X96k Stream: DTS extended audio stream that enables encoding of original LPCM audio at up to 24 bits per
sample with the sampling frequency of up to 96 kHz"

But I can't help you about the soft needed. Maybe another user..

[3ngel]

But, to obtain a 23khz would not to be enough to have 48khz (Nyquist theorem) instead of 96khz?

[raquete]

first i want to thank SomeJoe for this magnific guide.
congratulations! ;)

edit: obsolent.....perfect answer here: http://forum.doom9.org/showpost.php?p=841761&postcount=90

as english is not my "mother language", i have some doubts and if someone could clarify me i will be thankfull:
1- The default Dialog Normalization setting is -27 dB, and the default Dynamic Range Compression is set to "Film Standard".
where the -27 value for Dialog Normalization came from? i can't find this in "anywhere" inside Dolby.Inc or in any .pdf file(i download lots).

2- The decoder will perform an attenuation of (31 + dialnorm) dB to the program material when played back. what i found is that the dialnorm is "centralized" in -31 db(as the the pictures posted) and not (31 + dialnorm).please correct me if i'm wrong and explain me.

2- I loaded it into Sound Forge and measured the RMS level (http://pages.sbcglobal.net/wilsondr/ddexsfrms.gif) of the entire file as -20 dBFS.
a long time i don't use sound forge that have lots of options in the normalization dialogue(i don't remember all),i'm using audition and for each db choosed to normalize give different result.
-16dB was used as "reference" in the dialogue normalization tab like show the picture in the link.why not use -31,-27 or other value because each value choosed give different volume in the result.

thanks for answers.

[tebasuna51]

i can't find this in "anywhere" inside Dolby.Inc or in any .pdf file(i download lots).
Try with this another .pdf:
http://www.dolby.com/assets/pdf/tech_library/18_Metadata.Guide.pdf

[raquete]

Try with this another .pdf:
http://www.dolby.com/assets/pdf/tech_library/18_Metadata.Guide.pdf

perfect tebasuna51.
this .pdf answer my 2 first questions,thank you so much :cool:

about the last question:
measured the RMS level of the entire file as -20 dBFS in audition "Group waveform normalize" :

source with 0dB( 100% volume)
http://img228.imageshack.us/my.php?image=20source0db6bs.png

same source with -3dB ( 70.79% volume)
http://img149.imageshack.us/my.php?image=20source3db9fi.png

from audition help:
Group waveform normalize
Eq-Loud
Is the final loudness value with an equal-loudness equalization curve that takes into account frequencies to which the human ear is most sensitive. If you select the Use Equal Loudness Contour option in the Normalize tab, this value determines how much to amplify the audio to normalize it.
Loud
Is the final loudness value without equal-loudness equalization. If you don't select the Use Equal Loudness Contour option in the Normalize tab, this value determines how much to amplify the audio to normalize it.
Max
Is the maximum RMS (Root-Mean-Square) amplitude present. This value is based on a full-scale sine wave being 0 dB, and it conforms to the width specified in the Advanced section of the Normalize tab.
Avg
Is the average RMS of the entire waveform. This value isn't used for normalization.
% Clip
Is the percentage of the waveform that would be clipped as a result of normalization. Clipping won't occur if limiting (in which loud passages are decreased in volume) is used; instead, the louder portions of audio are limited to prevent clipping. In general, avoid values higher than 5% to prevent audible artifacts from occurring in the louder portions of audio.

Quote (from guide):
I loaded it into Sound Forge and measured the RMS level of the entire file as -20 dBFS. .
...as each source have different volume, we have different volume in result.
how much dbs we have to use before to load the source and find the RMS level ?

thanks!
;)

edit: changing the too big screenshots (that are ugly) for the imageshack urls.

[3dsnar]

But, to obtain a 23khz would not to be enough to have 48khz (Nyquist theorem) instead of 96khz?
Exactly, 48 kHz is enough. It is not important how many samples are representing each period, such sine can be (during resampling for example) restored nearly perfectly (assuming it is below the Nyquist frequency).
More here.
http://en.wikipedia.org/wiki/Nyquist_theorem

[tebasuna51]

@3dsnar
@3ngel
I agree with you for uncompressed wav, or using flac and similar.
But I doubt the info at 23 KHz is well preserved with encoders (ac3, mp3, aac, ...) at normal bitrates. I don't know dts.

[3dsnar]

Yeah, exactly. But this is more related to the compression technology, than to the sampling theorem (i.e. people normally do not perceive sounds above 20 kHz, so there is not sens to encode such high frequencies - so it is probably cuted out to save some bitrate).
Cheers, 3d

[3ngel]

I wonder if Dts use this same (absurd in my opinion) cut frequency policy. Anyone tried it?

[3dsnar]

Hmm, not necessarily absurd. Since you can not hear such high frequencies, why to encode them?
(this is in gerenal the idea behing limiting the upper frequency band).
And this is somewhat true - most people cannot hear a sinosoid over 20 kHz, but AFAIK preserving higher frequencies is important for the harmonics of the sounds. I.e. cutting out harmonic components of the signal (even though they are over 20 kHz) affects the sound quality.

[3ngel]

Hmm, not necessarily absurd. Since you can not hear such high frequencies, why to encode them?
That's not the point. If i have a certain signal, i want that signal intact with all its frequencies, unless some freq limitation it's clearly declared (and as far as i know there is no freq limitation declared in the ac3 specifications).

but AFAIK preserving higher frequencies is important for the harmonics of the sounds. I.e. cutting out harmonic components of the signal (even though they are over 20 kHz) affects the sound quality.
That's the point :)

[raquete]

cutting out harmonic components of the signal (even though they are over 20 kHz) affects the sound quality.
it's right guys but low frequences have more audible harmonics....think in 60Hz and sum...now with 20K or more,just a few or "nothing"!
your amplifier/receiver maybe can answer more than 20K but the speakers...i can't trust.
we can listen 20Khz(when someone can) is "alone",not in music because have strong diference in low frequences(and middles) for very high frequences for our ears.20Khz is impossible to be listen in musics.
i want that signal intact
maybe you have that signal intact but you can't listen diferences with more than 20Khz(or a little less).
don't "bore" with too high(and inaldibles) frequences,pay double atention in the basses that have lots of harmonics and more volume in musics.basses are the "soul" of the quality,when it's good...sounds better! ;)

ps:can someone take a look in the screenshots in my post on page 4 of this thread ( http://forum.doom9.org/showthread.php?p=840734#post840734 ) to remove my doubts about "normalizations" using sources with differents volumes?
( tebasuna51 (thanks again) send one on cool link that answer 99% of my doubts and only one more answer is needed)

thank you all :)

edit: typos

[tebasuna51]

ps:can someone take a look in the screenshots in my post on page 4 of this thread ( http://forum.doom9.org/showthread.php?p=840734#post840734 ) to remove my doubts about "normalizations" using sources with differents volumes?
( tebasuna51 send one on cool link that answer 99% of my doubts and only one more answer is needed)
I don't answer before because I'm not sure about this. Take my comments only like a opinion. You have two chances with this kind of source (seems have a narrow Dynamic Range):

1) Make a Dolby compliant ac3.
The sources don't need to normalized before encode and use
source with 0dB DialNorm=-12 dB (Avg), DRC=Music Light
same source with -3dB DialNorm=-15, DRC=Music Light
Then the ac3 decoder can play your ac3 at same global volume than others Dolby ac3 contents.

2) Make a ac3 not Dolby compliant but to be played at same volume than others contents like mp3, CD Audio, TV, ...
Normalize to desired level and encode with:
DialNorm=-31 dB, DRC=None
Then the ac3 must be played at same volume than wav source.

[raquete]

edit: obsolent post,correct answer here: http://forum.doom9.org/showpost.php?p=841761&postcount=90

source with 0dB DialNorm=-12 dB ...same source with -3dB DialNorm=-15...Normalize to desired level and encode with
i'm sure you're right tebasuna51,but you show me the parameters of each audio to encode(as quoted)...i will do the tests.

what i really think is why SomeJoe use the preset http://pages.sbcglobal.net/wilsondr/ddexsfrms.gif
[Sys] Normalize RMS to -16 dB (music) using "scan levels"...?? see that he don't posted how much dBs had his source too,then, i still have doubts in the guide,not in the test that you propose.

if the result of your test isn't right or if is right(that i trust), the doubt remains the same...(why the -16 preset? because using any other preset give different result ...not -20) :(

thank you so much!
;)

[tebasuna51]

@raquete
I'm not sure if I understand your question. Maybe...

1) When SomeJoe use Sound Forge "Normalization" feature is only to measure the RMS of wav source. The normalization is not applied (Cancel button) then the preset is indifferent.

2) Using Audition "Group waveform Normalize" you don't need go to step 3 to fix any normalization. Just in step 2 press the button "Scan for Statistical Information" and see the Avg value, this is your Dialog Normalization value for this wav source. Now you can "Close" the window.
The "*Percent over ... to -20 dB" (in red in your picture) is useless, you can put any value at step 3 "Normalize" and always the Avg value is the same ("Reset" and "Scan..." in step 2).

3) Your two pictures are coherent. First wav RMS Avg value -12 dB, second wave (3 dB below the first) RMS Avg value -14.76 dB. Not exact but coherent, different wav different RMS Avg value.

[raquete]

edit: obsolent post.correct answer here: http://forum.doom9.org/showpost.php?p=841761&postcount=90

@raquete
I'm not sure if I understand your question. Maybe...
sorry,i know...is my bad english,my fault.:mad:

1) When SomeJoe use Sound Forge "Normalization" feature is only to measure the RMS of wav source. The normalization is not applied (Cancel button) then the preset is indifferent.
....but he applied:
For this example, I will use an audio file that was captured from analog material. It is a plain stereo .wav file, 48 kHz, 16-bit. I loaded it into Sound Forge and measured the RMS level of the entire file as -20 dBFS.

Knowing that, here are some screen shots of the proper settings to encode this file...

The first page in ACID is the Audio Service Configuration, where the coding mode (2/0), the data rate (192 kbps), and dialnorm (-20 dBFS) are set.

here the screenshot od the "first page in ACID" http://pages.sbcglobal.net/wilsondr/ddexacid1.gif

this is what i'm talking about.
using the "[Sys] Normalize RMS to -16 dB (music)" to find the normalization he got -20dB,using another preset(-10,-31dB or other) we get a completely different result
what i really don't understand is why -16dB preset was used to find the normalization (-20dB in this case) that was applied in the encoder (Acid) as show this screenshot from the guide.

do you know what i mean now tebasuna51?
thank you so much,(great guy). :thanks:
;)

edit: later i post the results using -16,-20,-27 and -31db in waves group normalize using the same source with same volume.
(all results are differents)

[raquete]

@
tebasuna51 and all

see what happens using differents values.

source: off the ground-P.McCartney
extracted from cd have -.23dB (97.39%) no norm or amplify.

group waveform normalize
adjusted in "normalize to a level of" "x" using "Equal Loudness Contour" (normalize tab)
and "scan for statistical information" (ananlyze loudness tab)
( Eq-Loud=-9.48 / Loud=-10.79 / Max=-4.94 / Avg=-12 / %Clip=0% )
of course always give the same result in the "analyse loudness" tab because is the same source (no matter what "x" value is choosed to scan)

loading the source 4 times,adjusting "x" values for each and results after run normalize :

-16dB= -6.75dB (45,97%)

-20dB= -10.75dB (29,01%)

-27dB= -17.75dB (12,96%)

-31dB= -21.75dB ( 8.18%)

as i "told you",for each value chosed give different result with the same source.
now think when you have differents sources/differents volumes to normalize. :scared:

:thanks: :thanks: so much.

[tebasuna51]

@raquete
Don't mistake:

1) Normalize a wav, with SounForge/Audition, is modify the amplitude (volume).
source: off the ground-P.McCartney
extracted from cd have -.23dB (97.39%) no norm or amplify.

group waveform normalize
adjusted in "normalize to a level of" "x" using "Equal Loudness Contour" (normalize tab)
and "scan for statistical information" (ananlyze loudness tab)
( Eq-Loud=-9.48 / Loud=-10.79 / Max=-4.94 / Avg=-12 / %Clip=0% )
of course always give the same result in the "analyse loudness" tab because is the same source (no matter what "x" value is choosed to scan)
Ok.
loading the source 4 times,adjusting "x" values for each and results after run normalize :

-16dB= -6.75dB (45,97%)

-20dB= -10.75dB (29,01%)

-27dB= -17.75dB (12,96%)

-31dB= -21.75dB ( 8.18%)

as i "told you",for each value chosed give different result with the same source.
"after run normalize", for what?
You have different wav with different volume then you have different RMS Avg value.

2) Calculate the DialNorm to ac3 Dolby compliant encode. You need only know the RMS Avg of your wav source. The amplitude is not modified, only the parameter DialNorm is added to the ac3 stream.
Then "...but he applied:" don't mean the volume is modified (like Normalize) only the parameter DialNorm is set to -20 dB

[raquete]

"after run normalize", for what?....
You need only know the RMS Avg of your wav source. The amplitude is not modified, only the parameter DialNorm is added to the ac3 stream.
:scared: :stupid:
oh boy,only now i understand,...you're completely right and the guide too of course.
now i'm very embarassed: what i do with my obsolents posts here?!? :confused: (maybe one advice is needed in each showing that they are obsolents with a link to your last post)

:thanks: to SomeJoe and triple :thanks: for you tebasuna51 that really (with big patience) help me to understand it all and remove all my doubts. :cool:

beers for you and SomeJoe, you are very cool!

;)

[smok3]

quote from http://etvcookbook.org/audio/dialnorm.html :

"The value of the dialnorm parameter in the AC-3 elementary bit stream shall indicate the level of average spoken dialogue within the encoded audio program."

spoken dialogue, mkay? (as someboy noted before that rms calculus for the entire file is not valid...)

also, afaik rms is not a really good value for subjective loudness, there are better algos out there, like replaygain.

http://www.replaygain.org/

edit: guessing, lets say you have a 2 channel mix that you want to encode into 2ch ac3:
cut out the dialogoues, merge them and then calculate the rms (or whatever value, again RG should be better if we could match the values.) and set the dialnorm based on that.

edit2: another thing, dynamic compression: iam pretty sure this is used to prevent clipping from lossy source (like ac3) as well, and not only for subjective purposes (dolby does suggest to actually mix the audio throught the encoder, that may be one of the reasons why, right?)

[raquete]

@ tebasuna51 or anyone that can help me more if possible:

For this example, I will use an audio file that was captured from analog material. It is a plain stereo .wav file, 48 kHz, 16-bit. I loaded it into Sound Forge and measured the RMS level of the entire file as -20 dBFS.
all right.SomeJoe found -20dBFS but don't tell us how much volume had his source.
what i want to mean is that each source (differents volumes) will give differents RMS levels.
my first doubt:
how much dBs have to have the source? (before measure the RMS level)
second and deep doubt:
SomeJoe used one stereo source,then,what can i do to encode 5.1 sources (meaning L,R,C,LFE,LS and RS tracks),how much volume have to have each track?

i'm sorry for my complicated (maybe unclear) questions :o

thanks.

[ashp8]

i use wavelab5, nero wave editor(nero 7), Nero sound trax, maved3d and goldwave to edit my source material to encode i use sonic foundry soft encode 1.0.:goodpost:

I understand all licensed dolby digital encoders are the same in output, they have configuration and each configuration can be applied like in one encoder axactly same in the other another.;)

I've messed around with centre channels and resolved my issue of quiet or unnoticable centre channels by altering the audio in wave editor by volume and applying a compressor in goldwave to limit loud audio. this allows me to get a loud clear vocals channel in the centre.:) :) :)

Now my issue is using different programs they work at different bit depths and my audio sounds choppy as i boost the volume using the controls inside sonic foundry soft encode. if i lower the volume then it becomes less bright and dull and not loud enough.:angry: :angry: :angry:

i hate dolby digital. it is not for Music it lowers the centre and surround channels despite the dialog norm. volume management and it is very picky as the audio must be edited once and in 32-bit finely dithered to 16bit. DTS is much better i can pre-master the audio to make sure it is not too loud or quiet i keep my audio at -1.5db max and all centre, rear and front channels are balanced why isn't dolby digital music friendly.:angry: :angry: :angry: :angry: :angry: :angry:

how do i make sure that if i decrease volume there are absolutely no choppiness in audio i am confused.:( :confused: :( :confused:

[tebasuna51]

my first doubt:
how much dBs have to have the source? (before measure the RMS level)
second and deep doubt:
SomeJoe used one stereo source,then,what can i do to encode 5.1 sources (meaning L,R,C,LFE,LS and RS tracks),how much volume have to have each track?
1) The source volume is not a requirement to encode to ac3. Use the volume you want (maybe normalized to 95%).

2) Keep the relative volume between original channels (the word 'track' is used to the whole ac3: english audio track, spanish audio track, ...). Don't normalize each channel, use the same gain for all channels.

[raquete]

the word 'track' is used to the whole ac3..
ok,clear....but what about dialnorm?(later we talk about this)

Don't normalize each channel, use the same gain for all channels.
but what about if i'm using one source with ~95% and extracting all channels?
what i want to mean is that each channel will have:
LR ~95%(source)
C(center only) ~95% (depend of the source)
LFE ~25%(i use discrete amplifier only for lfe)
SLSR(surrounds less center) ~95%(depend of the source too)
total volume mixing the channels = "hundreds" % :scared:

the big question is: how much have to have each channel

thanks tebasuna51.

[SomeJoe]

It is hard to believe this post hasn't drawn much criticism yet. There is absolutely no way that the method of determining dialnorm at the start of this thread will produce anything but a muffled, muted, and terrible sounding audio movie track.

The reason is simple. The standard for dialnorm was not based upon the overall soundtrack, it is based upon the dialogue. The simplistic method of calculating it described at the start of the post cannot work because the original posted doesn't understand the basis for dialnorm in the first place.

My apologies for not answering this post sooner, I have been busy in the past months and don't login and read here as often as I used to.

Craftech is somewhat correct in his statement, that dialnorm is supposed to be based on the level of the dialogue, not the entire soundtrack. I did not make this clear enough in my original guide, and for that I apologize.

However, I take exception to the statement that I don't understand the basis for dialnorm and the statement that says that this method can't produce good audio. Allow me to explain.

First, my apologies for coloring the original guide with my own experience. My company produces primarily educational DVDs that are predominantly seminars and classroom-type courses. As such, the entire soundtrack of these DVDs is dialogue, with virtually no music or sound effects. Measuring the RMS level of the entire soundtrack in this case results in a value that closely correlates with a proper dialnorm setting.

In many other instances, the soundtrack will contain a mixture of dialogue, music, sound effects, and other content. In these cases, you should indeed measure only a portion of representative dialogue when attempting to determine the RMS level of the audio to be used for setting dialnorm. This is accomplished easily enough in Sound Forge (or other software) by selecting only the section of audio with dialogue before measuring the audio with the RMS/Normalization tool.

As to the other concern recently posted in this thread, that RMS is not a good measure to use because it doesn't correlate well with LAeq, I already addressed this in the original guide. I have previously stated that RMS measurement tools, because they are available in easy-to-obtain software, provide a "poor man's" method of obtaining a value close to the real LAeq level that dialnorm is supposed to be set with. I am well aware that RMS is not perfectly correlated with LAeq, but as I (nor anyone I know) owns software or hardware to measure LAeq directly, RMS will have to serve as a substitute.

There have been some suggestions in this thread that other methods other than RMS may correlate more closely with LAeq. I have not investigated that, and have no ability to do so. Of course, should someone have some evidence that another method would work better than RMS, then by all means use it, and post your results here. (In fact, if you have software or hardware to measure LAeq of your dialogue, do that).

As further evidence that reinforces my belief that RMS measurements will suffice, I offer this: I applied a few years ago with Dolby for use of the Dolby Digital logo on my company's DVDs. The Trademark and Standardization agreement required that I submit samples of my DVDs to Dolby for approval of the method. The first samples I sent were rejected because dialnorm, DRC, and a few other parameters were not set correctly in my encoded AC3. After research in the previously mentioned/linked Dolby documents, I came up with the method I posted in the guide, and I resubmitted my DVDs to Dolby for approval. They came back approved this time. Since Dolby's approval processes are considered to be somewhat rigorous, I can only conclude that my method, even though RMS sometimes does not correlate well with LAeq, arrives close enough to correct parameters to pass Dolby's approval processes.

In the interest of being correct, I will modify the guide to emphasize that the RMS level should be computed from a dialogue portion of the soundtrack, not the entire soundtrack.

[tebasuna51]

...
total volume mixing the channels = "hundreds" %

the big question is: how much have to have each channel
What mix? Each channel is encoded separately (more or less), then all channels can be at 100% of volume (not habitual but possible).
If an ac3 5.1 is played by a stereo equipment must do a downmix with a normalized matrix to avoid saturation problems.

[raquete]

What mix? i mean "mix" as encoding AC3 :o
If an ac3 5.1 is played by a stereo equipment must do a downmix with a normalized matrix to avoid saturation problems. :helpful: ....of course,understood.
Each channel is encoded separately (more or less), then all channels can be at 100% of volume (not habitual but possible). all right.
what do you think if i use each extracted channel(C,SLSL and LFE(i mean sub-woofer)) without amplification? (for example,if the extracted center channel have 70% of volume from source(LR))
i'm asking because using all channel ~95% give me too loud volume using 6 discrete amplifiers(not receiver/HT)

thank you tebasuna51. :)

[tebasuna51]

what do you think if i use each extracted channel(C,SLSL and LFE(i mean sub-woofer)) without amplification?
If you extract the channels from a original ac3 is recommended don't modify the volume levels. But you are free to make any personnal touch.