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

[SomeJoe]

How to Properly Encode Dolby Digital Audio (AC3)


Introduction

Many people on the forum have experienced problems when encoding audio using Dolby Digital. These problems are primarily volume-related, with some dynamic range compression issues as well. This guide aims to educate about how Dolby Digital audio should be encoded, and how to make it sound best.


References

The primary references for the information contained in this guide are two guides on Dolby's web site. The first is Standards and Practices for Authoring Dolby Digital and Dolby E Bitstreams (http://web.archive.org/web/20031206104650/http://dolby.com/pro/digaudio/pa.ma.1102.Standards.S.pdf), which has the best information on Dynamic Range Compression. The other is Dolby Digital Professional Encoding Guidelines (http://web.archive.org/web/20040716131627/http://www.dolby.com/tech/L.mn.0002.DDPEG1.pdf) which gives an excellent explanation of the dialogue Normalization parameter. You will need Adobe Acrobat Reader (http://www.adobe.com/products/acrobat/readstep2.html) to view these .pdf documents.


Philosophy of Dolby Digital

Dolby Labs has been doing high-quality audio with cutting-edge techniques for a long time, using their past experience as a guide. As such, there is often confusion about their methods and philosophy to those of us who are not privy to that information. Of prime example is the current problem: Why is Dolby Digital so much quieter compared to my original sound?

Most audio destined for DVDs is audio originally recorded for use in the movie theater. The movie industry has a huge advantage when producing audio for the theater -- the theater has large speakers and amplifiers, and a quiet, near-ideal listening environment. Huge dynamic ranges are possible, where the slightest whisper of dialogue is audible, yet gunshots and explosions can be earth-shattering. Dolby's dilemma was: "How do we bring this audio, with its huge dynamic range, into the home?" This is a major problem -- most homes don't have the speakers and amplifiers necessary to shake the living room. Further, background noise in the home can easily drown out those subtleties in the soundtrack.

Dolby's answer is to allow the decoder to modify the sound to compensate for these problems. Low-volume sounds are boosted automatically so they can be heard, whereas high-volume sounds are quieted down so that speakers aren't blown and other persons in the home are not disturbed. Further, Dolby Digital allows for different program material to be equalized, so that volume does not have to be adjusted when switching between inherently quiet programs and inherently loud programs.


Decoder Specifics

The methods I'm about to present here for encoding Dolby Digital are generic and do not apply specifically to any one encoder. All Dolby-certified encoders (and some non-certified ones) will have the appropriate parameters available to follow this procedure. I have personally tested the Sonic Foundry 5.1 Plug-In Pack for ACID Pro, as well as Sonic Foundry Soft Encode. These methods should also work for BeSweet, Vegas Video + DVD, Scenarist, and other software-based encoders.


Basic Parameters

Every Dolby Digital encoder has some basic parameters that need to be set.

The first is the channel combination, presented as (Number of front channels)/(Number of rear channels), with an optional ".1" added to represent a low-frequency-effects (LFE) channel if present. i.e. 2/0 represents normal left and right stereo sound. 3/2.1 represents a standard "5.1" setup, of Front Left, Front Right, Front Center, Rear Left, Rear Right, and LFE. This parameter should obviously be set to the number of channels of program material you will be encoding.

The other major basic parameter is the bitrate. Obviously, higher bitrates allow for less compression. Typical bitrates used are 192 kbps for 2/0 program material, and 448 kbps for 5.1 program material.


Referencing Volume to a Known Level - Dialogue Normalization

To meet the Dolby Digital requirement that different programs should have approximately the same listening level (thus the consumer does not have to adjust volume level between programs), Dolby Digital incorporates a parameter called dialogue Normalization. This metadata parameter tells the decoder how far away from the reference level the average sound pressure level of the material's dialogue is.

The movie industry masters their soundtracks in a specific way. The maximum rated sound level (where all amplifiers are putting out their rated power) is 0 dB. Sounds below that level are rated in terms of how many decibels (dB) they are down from that maximum level. As such, these values are negative. The movie industry typically masters the "normal" listening level of dialogue (where people are speaking in a normal voice) at -31 dBFS. In other words, a speaking voice is at an average of -31 dB when referenced to the 0 dB maximum sound level, hence the term decibels of full scale (dBFS).

Since movie content is the largest class of programs to go on DVD, Dolby chose -31 dBFS as the reference level for audio on DVD, where 0 dB represents the maximum encodable digital sound level (full scale).

The dialogue normalization parameter needs to be set to the LAeq level of your program material's dialogue. LAeq stands for the long-term A-weighted sound pressure level. Loosely, this is the average volume level of your source material's dialogue. Us lowly consumers really don't have a tool that can measure this parameter, but we can get close. Sonic Foundry's Sound Forge has a "Normalization" feature that can measure the RMS level of a .wav file (or the portion thereof containing dialogue). CoolEdit may also have a feature like this. To use it in Sound Forge, open your .wav file containing the movie audio. Select a section containing dialogue (no sound effects or music). Go to "Process"/"Normalize". Select the "Average RMS Power (Loudness)" radio button. Then click the "Scan Levels" button. The displayed "RMS" level is very close (within 1-2 dB) to the LAeq level.

That RMS level is the number that the dialogue normalization parameter should be set to. In other words, if the RMS level in Sound Forge shows as -17.6 dB, set the dialogue normalization parameter in your Dolby Digital encoder to -18 dBFS.

The decoder will perform an attenuation of (31 + dialnorm) dB to the program material when played back. So, in this case, the decoder will attenuate by (31 + -18) = 13 dB. This will bring the average sound level of the material to (-17.6 - 13) = -30.6 dBFS. The program is now played back at approximately -31 dBFS, the reference level.

-31 dBFS is a lower average volume level than what is typical from other sources. It will be noticeable that you will have to turn the volume up on your system when playing a DVD versus playing broadcast, tape, or other non-Dolby Digital program material.


Allowing Comfortable Listening - Dynamic Range Compression

Meeting the other end of the requirement, that the consumer should be able to listen to quiet and loud sections of the program without having to adjust volume levels, requires a decrease in the dynamic range of the program. A movie, with whispers at -50 dbFS and explosions at -5 dBFS can't be comfortably listened to in the average home. The whisper is drowned out by extraneous background noise, and if the explosion is played at a tolerable level that doesn't wake up the neighbors, regular dialogue at -31 dBFS requires straining to adequately hear.

Dolby solves this problem by compressing the dynamic range of the program material. Quiet sounds are automatically boosted in volume so that they're audible, and loud sounds are automatically cut down in volume to tolerable levels.

There are several dynamic range compression profiles available that are custom tailored to the particular flavor of program material. However, all of them share the same basic features. All of the compression profiles can be thought of as an input-output "black box", where certain input volume levels are mapped to certain output volume levels. Observe this graph (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/dddrclabeled.gif), which is a graph of one of the Dolby Digital compression profiles (Film Light).

The blue line is the "unity gain" line, also referred to as the "no compression" line. This line represents that the dynamic range compressor feature of the decoder is essentially turned off, and no boost or cut of the program material is done.

The purple line is the compression profile for "Film Light". It is divided into 5 different sections, as are the other Dolby Digital compression profiles:

Unity Gain = Volume neither boosted nor cut
Variable Boost = Beginning of increasing volume of soft sounds
Constant Boost = Increase volume by a fixed amount for very soft sounds
Early Cut = Beginning of attenuating volume for loud sounds
Cut = Very loud sounds almost clamped to a maximum volume level

The application of a compression profile like this allows the soft sounds to be heard while preventing speaker overdrive and disturbances by the loud sounds.

The Dolby Digital encoder offers 5 different compression profiles that can be specified depending on the nature of the program material being encoded. This graph (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddcompprof.gif) illustrates all of the available profiles. The profiles range from no compression ("None"), to fairly light compression ("Music Light") all the way to extremely aggressive compression ("Speech").

For the exact dB numbers where each range of dynamic range compression is located on the graph, see the Dolby documents cited in the references at the beginning of this guide.

Many authors, when compressing audio to Dolby Digital, are turned off by the idea of dynamic range compression. You have this well-mixed audio with nice dynamic range and are then going to kill it by compressing that dynamic range. This is a valid concern, but should be answered by looking at what the listening environment is going to be. If you are authoring a DVD only for yourself, and you have a home theater room that can deliver theater-like sound, perhaps a compression profile of "None" is suitable for you. However, this profile may not sound good in a more mundane living room. Some experimentation may be in order to determine what compression profile will sound best for you. Most Hollywood DVDs use "Film Light" or "Film Standard".

The following point, however, cannot be stressed enough: In order for the Dynamic Range Compression to work as designed, the Dialogue Normalization parameter MUST be properly set first!

All of the dynamic range compression profiles assume that the average volume level of the program material's dialogue being fed to the dynamic range compressor is -31 dBFS. If that is not the case, boost or cut will be applied to the material when it shouldn't be!

A prime example is the situation where an average volume .wav file (with an LAeq of -16 dBFS, for example) is fed to Soft Encode using Soft Encode's default dialogue Normalization and Dynamic Range Compression settings. The default dialogue Normalization setting is -27 dB, and the default Dynamic Range Compression is set to "Film Standard". Because of the misadjusted dialnorm parameter, only (31 + -27) = 4 dB of attenuation is applied to the audio, so the average volume level is (-16 - 4) = -20 dBFS instead of the expected -31 dBFS. This places the audio on the DRC graph at the incorrect position, and now most of the audio is being played back in the Early Cut and Cut ranges. This causes the audio to sound flat and dull, with a possible audible "pumping" of the volume up and down as the decoder changes between Early Cut and Cut based on average volume level. Here is a representative graph (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddwrongdialnorm.gif). If dialnorm had been properly set to -16 dB, the audio would be centered at -31 dBFS, and would sound like it is supposed to.


Line Mode and RF Mode

The Dolby Digital compressors have the ability to further alter the compression profile to compensate for different transport mediums. Most of the time, audio is transported between devices in "Line Mode", where a line-level is used. There is also "RF Mode", meant for broadcasting of Dolby Digital and devices that send audio via RF cables to a TV set. RF mode sound from the decoder uses a higher average volume level (-20 dBFS vice -31 dBFS) in order to correlate volume level well with other, non-Dolby broadcast audio, and also can use a more aggressive Dynamic Range Compression to prevent overmodulating the signal. There is an option in most Dolby Digital encoders to turn on that overmodulation limiter (in Soft Encode, it is labeled as "RF Overmodulation Protection"). Since we are primarily interested in authoring for DVD which will operate in Line Mode, we do NOT want to insert the additional Dynamic Range Compression that RF Overmodulation Protection will add. Therefore, for DVD authoring, the RF Overmodulation Protection option should be turned off.


Example Compression Settings

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. (Note that this audio was from a filmed seminar, and thus the entire audio file was dialogue. Because of this, I did not select a particular range to compute the RMS level, but I rather selected the whole file. In your file, you should select only a portion that is representative dialogue.) I loaded it into Sound Forge and measured the RMS level (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddexsfrms.gif) of the entire file as -20 dBFS.

Knowing that, here are some screen shots of the proper settings to encode this file (Sonic Foundry's 5.1 Plug-In Pack for ACID Pro was used as the encoder. Your encoder may not have all options available.)

The first page (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddexacid1.gif) in ACID is the Audio Service Configuration, where the coding mode (2/0), the data rate (192 kbps), and dialnorm (-20 dBFS) are set.

The second page (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddexacid2.gif) is the Bitstream Information. Set these parameters are appropriate for your source material.

The third page (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddexacid3.gif) is the Extended Bitstream Information page. Your encoder may not allow these options to be set. I typically do not set any options here.

The final page (http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/ddexacid4.gif) is the Preprocessing page. Here is where you set the Dynamic Range profile you want to use for the material. The Line Mode setting is the one that will actually be used by the DVD player or Dolby Digital receiver, since the sound will be coming from the line outs. The RF Mode generally won't be used, but I typically set it to the same profile as the Line Mode. Make sure RF Overmodulation Protection is not checked.


Conclusion

I hope that this guide has given some insight into the proper methods that should be used to get the most out of your Dolby Digital encoding.

[kxy]

Thanks SomeJoe, for your clear and detailed explanations. :)

[JensG.]

Great work, SomeJoe!

(Dammit, now I will always have to think about sub-optimal audio in my divx!)

;)

[Julio]

Hi,

I think I've understood the whole post of SomeJoe, but I've got some more questions with a WAV (of a rock concert) I'm trying to encode to AC3, using Soft Encode :

Before reading this post, I thought it was best to set Dialog Normalization to -31 dB, because when I used lower settings it gave me an AC3 that, when decompressed to WAV, had extremely low dynamics.

So I encoded it with -31 dB dialog normalization, and I got what SomeJoe calls "up and down pumping of the volume" at certain moments in my soundtrack ! Therefore I think my dialog normalization parameter is not right.

Here are the values I got when analyzing my soundtrack with Cool Edit 2000 :

Min RMS Power : -87.2 dB
Max RMS Power : -12.0 dB
Average RMS Power : -22.8 dB
Total RMS Power : -22.4 dB

So, should I set dialog normalization to something like -23 dB ?

[SomeJoe]

Originally posted by Julio
... a WAV (of a rock concert) ...

Min RMS Power : -87.2 dB
Max RMS Power : -12.0 dB
Average RMS Power : -22.8 dB
Total RMS Power : -22.4 dB


Yes, I would set Dialog Normalization to -23 dB. Since your sound is of a rock concert, and the Max RMS power only goes to -12 dB (about 10 dB above the average level), you can probably set Dynamic Range Compression to None. If you find that the loudest parts of the concert are too loud to comfortably listen to, go back and set Dynamic Range Compression to Music Light.

[nuked]

So I've heard there are some gain markers in the AC3 that tell how to do dynamic range compression when decoding. Then you can either do it or not. I asume ACID does not encode these markers, it's just compressing the raw tracks, right? Now if I don't compress, is there a filter that I can compress with on playback that will use the same -31 db standard and that do as good a job as besweet? Why compress if you can leave it optional?

nukeD

[SomeJoe]

Originally posted by nuked
So I've heard there are some gain markers in the AC3 that tell how to do dynamic range compression when decoding. Then you can either do it or not. I asume ACID does not encode these markers, it's just compressing the raw tracks, right? Now if I don't compress, is there a filter that I can compress with on playback that will use the same -31 db standard and that do as good a job as besweet? Why compress if you can leave it optional?

I am not well-versed on the particulars of the AC3 file structure, but as far as I know, both the Dialog Normalization parameter and the Dynamic Range Compression parameters are metadata in the ac3 stream. In other words, the actual audio is not altered to apply these parameters, the parameters are only there as flags to tell the decoder what modifications to apply to the audio.

As such, if the decoder is properly programmed, it could ignore some or all of these metadata parameters, and not apply those modifications to the decoded audio. However, I know of no common hardware or software decoders that will do this. (There are some Dolby Digital decoders in some high-end home receivers that give the user some control over the Dynamic Range Compression, but that's about all I've seen). I know of no software decoder or filter that can optionally implement user-defined parameters on decode.

[nuked]

Ac3filter useses the metadata and leaves the decompression optional.
It's very cool.
If you turn it on you can watch the little DRC gain bar go up
and down while it plays back. If ACID really uses the metadata,
I should be able to see that little bar working. Maybe I'll give it
a try. That option could be reason enough for me to encode in stereo
ac3 instead of mp3.

nukeD

[nuked]

oh... I mean did I say ACID... I meant something cheaper ;P
no disrespect... it's probably worth it, but not for me.
I misunderstood and thought this was available as a plugin
for besweet now... guess it's still just the ac3enc.

UPDATE:
For the record, I can't get ac3enc to use this metadat trick for encoding
dynamic range control, but actually I'd have been very surprised if it had since ac3enc isn't what's actually doing the dynamic range compression in besweet. This would be a very neat feature to have in ac3enc though(as if DSPGuru has nothing else to do :) ).

[KoVaR]

so....
is it possible to make 6.1 and 7.1 files ?
it should be set to 3/3.1 for 6.1 ?
but for 7.1 ?
i think that 3/2/2.1 should be correct

:p

[resonator]

Works well with 2.0 material, however, if I want to mess around with 5.1 encodings (i.e. combining 2 sloppy truncated ac3 files by resyncing them using VDub), how am I supposed to find the RMS level of a 5.1 mix? Should I load every mono file into SoundForge, let it detect the RMS value and then use the avarage of these 6 values? Any suggestions?

[echooff]

Severyone on this thread seems knowledgable. Maybe someone could tell me what I'm doing wrong. Everytime I try to encode to ac3 (both 2.0 and 5.1)with either Besweet or softencode all I get is a File full of squeal. It seems to be the proper size and besweet's log file is normal. I have ac3 filter installed. My stero handles surround but all it plays is squeal also. I tried the different methods on the sitcky's in the audio forum. I have found several guides on the web and tried them. I am still coming up empty. Any suggestions?

[SomeJoe]

Originally posted by resonator
Works well with 2.0 material, however, if I want to mess around with 5.1 encodings (i.e. combining 2 sloppy truncated ac3 files by resyncing them using VDub), how am I supposed to find the RMS level of a 5.1 mix? Should I load every mono file into SoundForge, let it detect the RMS value and then use the avarage of these 6 values? Any suggestions?

I wouldn't use the average of all 6, because some channels of a 5.1 mix are more important than others.

I might start with loading just the left and right channels into sound forge and making a stereo .wav, and finding the RMS of that. Then, if you feel there's a decent amount of content in the center and rear channel mixes, that would raise the overall RMS level. For example, if you make a left/right .wav, and find the RMS is -18 dBFS, and there is a pretty good amount of surround and center, assume the RMS level is 3 dB higher then that. So I'd set dialnorm in the encoder to -15 dBFS.

Another way to do it would be if the mix is a true 5.1 mix with all dialog in the center channel. You might be able to just measure RMS of the center channel by itself and use that for dialnorm.

These are just a couple ideas. You may have to modify this depending on your content.


Originally posted by echooff
Everytime I try to encode to ac3 (both 2.0 and 5.1)with either Besweet or softencode all I get is a File full of squeal. It seems to be the proper size and besweet's log file is normal. I have ac3 filter installed. My stero handles surround but all it plays is squeal also. I tried the different methods on the sitcky's in the audio forum. I have found several guides on the web and tried them. I am still coming up empty. Any suggestions?

If the encoded file you're making is a .wav, that's AC3 frames encapsulated in a .wav header. On the computer, it will try to play this as PCM audio, not AC3, and you'll get noise.

You should use SoftEncode or BeSweet to make a standard .ac3 file instead, and try to play that.

If that doesn't work, there may be something wrong with your filter setup. You should probably start a new thread to get help with that since the problem is really not an encoding parameters issue.

[bitbrain2101]

Hi,

I have come over a strange problem encoding AC3-files with SoftEncode or Digigram Multichannel Encoder. I have encoded AC3 5.1 and 2CH Stereo files from mono wav-files with 48k/16bit.I can play and hear these files with PowerDVD and the Dolby Logo is shown in PowerDVD.But if I try to import these AC3-files in TMPGenc DVD Author it tells me "illegal File".If I import AC3-Files ripped from DVD everything is ok.I examined these files with GSpot,my self-encoded AC3-files are unknown,but the DVD-ripped AC3-files are shown as AC3-files with their bitrate.Is there perhaps some kind of header missing that my self-encoded files aren´t recognized ??

bitbrain2101;)

[nuked]

hmmm... I just found the same issue, where windvd works but matrix mixer doesn't. I re-encoded with intel byte order unchecked and it worked. I think that's the only thing I did diferently but I'm not sure. This could certainly make sense though.

correction: ac3filter, not matrix mixer of course.. anyway.. I guess it's mute.

[bitbrain2101]

Hi,

I have found a solution for the problem.I re-encoded the AC3-file with AC3Machine and now it works fine.

@nuked I didn´t have the Intel byte order checked any time

bitbrain2101;)

[FulciLives]

Hello :)

I don't have Sound Forge but I do have CoolEdit Pro 2.0 and here is what I get when I analyzed a WAV file:

---------------------------------------------------

Left Right
Min Sample Value: -29437 -29493
Max Sample Value: 29848 29591
Peak Amplitude: -.81 dB -.89 dB
Possibly Clipped: 0 0
DC Offset: -.001 -.001
Minimum RMS Power: -52.88 dB -53.46 dB
Maximum RMS Power: -10.38 dB -10.2 dB
Average RMS Power: -18.14 dB -17.95 dB
Total RMS Power: -17.54 dB -17.36 dB
Actual Bit Depth: 16 Bits 16 Bits

Using RMS Window of 50 ms

---------------------------------------------------

So what value do I use here for setting the DIALOG NORMALIZATION?

BTW I'm using the AC-3 encoder that comes with Scenarist in case that makes a difference (I doubt it does as the available settings seem to be the same as in your examples).

So basically I'm just confused by the output above that CoolEdit Pro 2.0 provides in relation to your guide which uses Sound Forge.

Thank you! :)

- John "FulciLives" Coleman

[SomeJoe]

Originally posted by FulciLives
Average RMS Power: -18.14 dB -17.95 dB


Average RMS power is what you're interested in.

-18 dBFS is the appropriate dialog normalization value for this material.

[FulciLives]

Originally posted by FulciLives
Hello :)

I don't have Sound Forge but I do have CoolEdit Pro 2.0 and here is what I get when I analyzed a WAV file:

---------------------------------------------------

Left Right
Min Sample Value: -29437 -29493
Max Sample Value: 29848 29591
Peak Amplitude: -.81 dB -.89 dB
Possibly Clipped: 0 0
DC Offset: -.001 -.001
Minimum RMS Power: -52.88 dB -53.46 dB
Maximum RMS Power: -10.38 dB -10.2 dB
Average RMS Power: -18.14 dB -17.95 dB
Total RMS Power: -17.54 dB -17.36 dB
Actual Bit Depth: 16 Bits 16 Bits

Using RMS Window of 50 ms

---------------------------------------------------

So what value do I use here for setting the DIALOG NORMALIZATION?

BTW I'm using the AC-3 encoder that comes with Scenarist in case that makes a difference (I doubt it does as the available settings seem to be the same as in your examples).

So basically I'm just confused by the output above that CoolEdit Pro 2.0 provides in relation to your guide which uses Sound Forge.

Thank you! :)

- John "FulciLives" Coleman

*** EDIT ***
Just wanted to say thank you for answering my question SomeJoe :)

[Sunix]

Hi,

I just read your guide and found it very usefull.

However I am a little confused about what value to use for normalisation.

In Sound forge, I get these values:

RMS: -18.4 db WITHOUT use equal loudness contour option checked
RMS: -21.1 db WITH use equal loudness contour option checked

And if I run Cool Edit on the same file, I get these values:

In FS Square mode:

Minimum RMS Power: -57.74 dB -54.83 dB
Maximum RMS Power: -5.01 dB -6.13 dB
Average RMS Power: -31.11 dB -31.37 dB
Total RMS Power: -22.34 dB -22.91 dB


In FS Sine mode:

Minimum RMS Power: -54.73 dB -51.82 dB
Maximum RMS Power: -1.99 dB -3.12 dB
Average RMS Power: -28.1 dB -28.36 dB
Total RMS Power: -19.33 dB -19.9 dB


I read in a previous message that Average RMS Power from cool edit should be used when you just have cool edit as program, but these value are far away from those in Sound Forge. Even in Sound forge value change depending of selected options.

So what is the best value to use for normalisation?

Thanks in advance,
Sunix

[SomeJoe]

Originally posted by Sunix
In Sound forge, I get these values:

RMS: -18.4 db WITHOUT use equal loudness contour option checked
RMS: -21.1 db WITH use equal loudness contour option checked

And if I run Cool Edit on the same file, I get these values:

In FS Square mode:

Minimum RMS Power: -57.74 dB -54.83 dB
Maximum RMS Power: -5.01 dB -6.13 dB
Average RMS Power: -31.11 dB -31.37 dB
Total RMS Power: -22.34 dB -22.91 dB


In FS Sine mode:

Minimum RMS Power: -54.73 dB -51.82 dB
Maximum RMS Power: -1.99 dB -3.12 dB
Average RMS Power: -28.1 dB -28.36 dB
Total RMS Power: -19.33 dB -19.9 dB


I read in a previous message that Average RMS Power from cool edit should be used when you just have cool edit as program, but these value are far away from those in Sound Forge. Even in Sound forge value change depending of selected options.


Interesting. :p I don't have Cool Edit, so I haven't been able to read the documentation to see what the difference is between "Average RMS Power" and "Total RMS Power". However, if you look back in this thread, previous posters who have asked me this question about CoolEdit, their Average RMS and Total RMS values were very close to each other. Yours are very different.

Also I haven't seen any documentation on what the difference is between "FS Square" mode and "FS Sine" mode, but if I had to guess I would pick FS Sine, because that seems to say that the RMS value is being computed from a Fourier series based on sine wave computations, which is how it should be done.

As far as Sound Forge goes, I'm reasonably confident that you should use the RMS value you get without the equal loudness contour.

Since the RMS value is a definite, fixed value for the file, the differences we see in values between all methods must come from algorithm differences in the way each is computed. I would say that from your file, it appears as if Cool Edit's FS Sine, Total RMS Power is the closest algorithm to Sound Forge's RMS without equal loudness contour.

Because of that, I would say your appropriate DialNorm setting is -19 dBFS.

If you or anyone else has appropriate documentation from CoolEdit on an explaination of their reported RMS parameters, I'd be able to make a more educated recommendation as to which value to use. Also keep in mind that the true setting of dialnorm is the LAeq level -- using RMS is our low-budget, attainable, close approximation. :D

---------------------------------------------------------------------

Edit:

I went and looked at the help file for Sound Forge and found this regarding the equal loudness contour:

Checking the Use equal loudness contour option causes the scan to take into effect the Fletcher-Munson Equal Loudness Contours. Essentially, very low and high frequency material is less audible than mid-range audio. This option forces the scan to weigh this factor into the RMS calculation.

In view of that, I would recommend that you use the equal loudness contour setting when performing a scan for normalization. This is contrary to my above recommendation. :eek:

Also in view of that, I would then change my recommendation for your DialNorm level. Since Total RMS Power, FS Sine from CoolEdit is -19 dB, while RMS, equal loudness contour enabled level from Sound Forge is -21 dB, you could use either with confidence or use their average of -20 dB. A +/- 1 dB error will not make a very (if any) audible difference in the resulting AC3, and indeed the error imposed by using RMS instead of true LAeq could be more than this anyway.

[macik-pacik]

Hi!! I am just a new here, at this forum and DVD burning as well. I do not have problems with a quality and processing AC3 files, I use Maestro to authoring, but sometimes -e.g. Matrix reloaded, I got large pieces of audio streams - 2x370MB /english and czech/. that means I am to reduce video quality through CCE. Are there possibilities to reduce AC3 to a smaller size??
Thanks a lot. j.:)

[clapper]

quote:
--------------------------------------------------------------------------------
Checking the Use equal loudness contour option causes the scan to take into effect the Fletcher-Munson Equal Loudness Contours. Essentially, very low and high frequency material is less audible than mid-range audio. This option forces the scan to weigh this factor into the RMS calculation.
--------------------------------------------------------------------------------

Are you sure that the curve should be included? It sounds as thou the program would, in essence, apply an EQ curve to the program (non-destructive to the file) before analysis, and thereby raise or lower the result?

[SomeJoe]

Originally posted by clapper
Are you sure that the curve should be included? It sounds as thou the program would, in essence, apply an EQ curve to the program (non-destructive to the file) before analysis, and thereby raise or lower the result?

That is true, and that's essentially what Sound Forge would be doing. I would still recommend it be included because the objective is to make use of a quantifiable scale (dB of RMS power) that is as closely as possible related to the perceived volume level of the material. (Which should be as close as possible to the actual LAeq level of the material). Since the equal loudness contour applies a psychoacoustic model to the sound when computing the perceived loudness, you can view the application of this curve as an updated algorithm for computing the RMS power (if you're trying to associate RMS power with perceived loudness).

At any rate, if you don't believe the equal loudness contour should be used or is not applicable to the material, it can be turned off in Sound Forge's normalization dialog box.

Furthermore, I believe the difference in final measurement with and without equal loudness contour may be well within the tolerance/error that results from using an RMS power measurement rather than LAeq. I would state that if the error associated with using RMS is too high for your application, that you should probably be using a higher end tool to measure LAeq directly instead of RMS power.

I would like to actually test and see how close the various RMS measurements from Sound Forge and CoolEdit/Audition are getting to LAeq, but I unfortunately don't have any tool that can measure LAeq directly. Thus my "poor man's" approach to the entire problem. ;)

[Kilyan]

Hi!

I'm making dolby ac3 soundtracks in my language for dvds like saving private ryan and a lot more, taking it from vhs source. My problem is that if I use your method (For example I get -14 dB rms in Sounforge) and encode in softencode I get a much quieter audio track (2.0 dolby encoded )than the english 5.1 . If I use the -27dB I get better results but the audio is pumping (companding and expanding). So what's the problem here? How could I get a resonably loud audio without pumping?

BTW why all ac3 tracks Ive seen so far on dvds use the -27dB level?

Thanks

[SomeJoe]

Originally posted by Kilyan
How could I get a resonably loud audio without pumping?

If you want it louder than the reference -31 dBFS level but don't want the pumping, you will have to turn the dynamic range compression off.

[tbrunner]

Hi

I have also a lot of problems with bad AC3 sound resulting in to low volume dialogue (very difficult to understand) and to loud sound (explosions, music and so on).

My scenario is the following: I take a divx/xvid avi file and encode it to mpeg2 with AC3 sound to burn it on a dvd and watch it on a stand alone dvd player. I prefer AC3 sound as it is supported both for NTSC and PAL dvd's. As I would like to have the highest possible bitrate for the video, I encode the sound to two channel stero with 192 KB.

The sound in this divx/xvid files is either
a) MPEG Layer 1/MPEG Layer 2/MP3 or
b) AC3 5.1 or
c) AC3 2/0

My way of processing is:

1) convert the sound to WAV: For this I use the MPEG2 Encoder (Mainconcept in my case) to produce elementary video and audio streams. For the audio stream I select wav format. As the Mainconcept encoder uses direct show to read the file, AC3 Filter can be used to decode the AC3 sound and downmix it to 2 channel. I assume this works with every encoder who supports either direct show directly or can read avisynth files.

2) convert the WAV to AC3 using AC3 machine (I do not use AC3 Machine to downmix AC3 5.1 sound directly to two channels as proposed in the DOOM9 guide, because as result, I still get a 5.1 AC3. This surprises me not as the resulting command line is the same regardless if channels mode 5.1 or stero is selected).

Now my question is, where and what do I set to archive the proper 5.1 downmix to 2 channels, Dialog Normalization and Dynamic Range Compression. I would like to use as much as possible freeware tools.

I assume that the solution depends on the sound type in the avi.

a) MPEG Layer 1/MPEG Layer 2/MP3 (I assume that the low volume problem also occurs with this sound types as the people are not properly encoding the input AC3).
How and with what tool do I have to reencode the WAV to a WAV so that if feeding the WAV to AC3 Machine the sound is correct in the resulting AC3?

b) AC3 5.1
As the AC3 Filter provides so many settings, I assume that it is feasible to create a WAV file that can just be feed to AC3 Machine and the resulting AC3 sound is correct. The most obvious setting is to choose 2/0 stereo as output. To what do I set the other settings.

c) AC3 2/0
Some as AC3 5.1 but with different settings in AC3 filter.

Can someone give advice on the correct settings?

[tbrunner]

Hi

I have encoded a wav to ac3 exactly following the instructions in the first post of this thread. The result sounds very satisfying. I had to set the dialog normalization to -24 db. For the dynamic range compression I used "Film: Standard".

After encoding to ac3 I analyzed the resulting ac3 file. Opening it again with Soft Encode it showed -24 db for dialog normalization in the stream settings. I played it back with Radlight 3.03 over AC3 Filter. In the AC3 filter settings (right click on radlight, Advanced/Filters/AC3Filter) the DRC level slider was moving.

This seams to prove that setting the dialoge normalization and the dynamic range compression "only" affects the meta data of the ac3 stream.

I have encode the same wav with AC3 Machine. Afterwards Soft Encoded showed -31 db for dialog normalization and the DRC slider was not moving in AC3Filter.

For me this means that AC3 Machine expects as input a wav file which is "properly" (dynamic range compressed, loudness attenuated) mixed.

Does somone know how to mix the wav file (drc, loudness attenuation)?

[tbrunner]

Hi

If the source is a rip from a cd or the sound of a DV-AVI file from a camcoder, is it then also necessary to apply dynamic range compression? I would expect not, as the input has a normal "dynamic range" and needs not to be changed. Is this assumption correct?

Is there a way to measure the dynamic range of a given file to find out if and what dynamic range compression should be applied?

Thank you for any hints.

[Mug Funky]

@kilyan:

if you're getting your soundtracks from VHS (especially if it's taped from live broadcast), it will already have a compressed dynamic range. compressing further will inevitably result in pumping.

easiest way to check if you need to compress at all is to take a look at the waveform overall. if everything is at a similar volume then it's already been limited and further compression will just reduce quality.

so i'd turn DRC off altogether.

as far as why most DVDs use -27dB dialog normalisation? probably because it's the default. the commercial DVD author i know had no idea what dialog normalisation is all about, and apparently they generally leave it at -27. If the sound is bad (very rarely on the whole) they just turn it off and re-encode (after reading this thread i set him straight on this. hehehe).

@tbrunner:

the situation is similar with DV tapes... most DV camcorders will have an internal limiter that prevents clipping. they do, however have a much wider range than VHS (especially recorded from broadcast through RF), and so could benefit from compression. again, look at the waveform for loud transients and quiet speech, etc. (the DV cams i've used seem to only do significant limiting when you plug an external mic into them. the internal mic i'd say is also limited, but far less).

[Soapm]

Great faq Somejoe! Thanks. I now know how bad I've been destroying the audio.

Can you give us more input on how to fix an AC3 5.1 file?

Ex. After reading your faq, when I demux a file that has 5.1 AC3 audio I use the file verify option in soft encode to check it. If the diagnorm is -27 and it sounds ok I don't touch it. However, I have one that sound encode will not open. Any ideas how to fix it without destroying the quality?

I also have one that the diagnorm says reserved. Any idea what that means? Is this a good thing?

What if I just want to add light compression to a 5.1 AC3 file? Ieas on that?

[SomeJoe]

Originally posted by Soapm
Can you give us more input on how to fix an AC3 5.1 file?

Ex. After reading your faq, when I demux a file that has 5.1 AC3 audio I use the file verify option in soft encode to check it. If the diagnorm is -27 and it sounds ok I don't touch it. However, I have one that sound encode will not open. Any ideas how to fix it without destroying the quality?

I also have one that the diagnorm says reserved. Any idea what that means? Is this a good thing?

What if I just want to add light compression to a 5.1 AC3 file? Ieas on that?

If Soft Encode won't open the AC3 then the file is probably corrupted. You may be able to fix the file with one of the tools that fixes bad AC3 frames (not sure of the names of them ... maybe one is called AC3fix?)

I have no idea what the "reversed" message means. Never have seen that before, but I only use Soft Encode rarely.

Changing the metadata parameters of an existing AC3 file (i.e. changing the dialog normalization parameter or the DRC mode) should be theoretically possible by altering the metadata in the AC3 stream. But I know of no software which can do this. Writing it wouldn't be too tough for some of the guys around here IF we could find some detailed specs on the AC3 file structure -- but I doubt Dolby is going to post that. ;)

[magilvia]

Thanks SomeJoe for this really instructive post.
I've only question: how to set in Soft Encode "Audio production information", which are "Mix level" (default to 105db SPL) and "room type" ?
Is it best to leave "Info exist" checked?
Thanks again

[magilvia]

One other question too :p :D
Has anybody verified if the average RMS of CoolEdit and SoundForge is compatibile with that measured with Goldwave ?
I like Goldwave MUCH more than CoolEdit plus it is SHAREWARE!

[SomeJoe]

To get further information on the audio production information and the room type, I think you'll have to go to those .pdf files that are on Dolby's web site. I'm not positive what those settings do, nor do I know if typical AC3 decoders/receivers will use that metadata to alter the playback sound. I typically leave those settings at the default, with the "Info Exist" unchecked.

I have never used Goldwave, so I'm not positive if it's measured RMS readings are on par with Sound Forge of CoolEdit. But if the reading says that it's an RMS measurement, it's probably very close.

[oldiexyz]

Hi all,

emmm... may I come back to an older reply by bitbrain2101?

On August 6, 2003, he asked about a problem with Digigrams MC en- / decoder. I do have exactly the same problems: AC3s produced by this program cannot be used in other apps like TMPGEng DVD Author and cannot be "reconverted" into 6 waves by BeSweet. On the other hand, the Digigram DEcoder cannot produce WAVs from AC3s made by e.g. the AC3Machine which uses the ac3enc.dll.

Originally posted by bitbrain2101
Hi,

I have come over a strange problem encoding AC3-files with SoftEncode or Digigram Multichannel Encoder. I have encoded AC3 5.1 and 2CH Stereo files from mono wav-files with 48k/16bit.I can play and hear these files with PowerDVD and the Dolby Logo is shown in PowerDVD.But if I try to import these AC3-files in TMPGenc DVD Author it tells me "illegal File".If I import AC3-Files ripped from DVD everything is ok.I examined these files with GSpot,my self-encoded AC3-files are unknown,but the DVD-ripped AC3-files are shown as AC3-files with their bitrate.Is there perhaps some kind of header missing that my self-encoded files aren´t recognized ??

bitbrain2101;)

In the following reply (by "nuked") there was an remark about the "intel byte order" (vs. "motorola"?). I guess, this could be a reason, but...


...sorry, where ist the flag set? What do I have to change to make Digigram's en- / decoder work compatible???

Trying to fix the output file using BeSplit / BeSliced leads to an empty (0 kb) file. Log:

BeSplit v0.9b6 by DSPguru.
--------------------------

Logging start : 02/16/04 , 15:51:45.

d:\tools\besweet\BeSplit.exe -core( -input E:\DVDs\6channel-dg.ac3 -fix -logfile d:\temp\BeSliced.txt -type ac3 -output E:\DVDs\6channel-dg_Fixed.ac3 ) -profile( BeSliced v0.3 )

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] | Input : E:\DVDs\6channel-dg.ac3
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Channels Count: 5, Bitrate: 448kbps
[00:00:00:000] | Output : E:\DVDs\6channel-dg_Fixed.ac3
[00:00:00:000] +---------------------
[00:00:00:000] | Writing E:\DVDs\6channel-dg_Fixed.ac3
[00:00:00:000] +---------------------
[00:00:00:000] Operation Completed !
[00:00:01:000] <-- Process Duration
Logging ends : 02/16/04 , 15:51:46.


The "demux" log using BeSweet is:

BeSweet v1.5b25 by DSPguru.
--------------------------
Error 59: Failed to sync to payload's start position : "e:\DVDs\6channel-dg.ac3"Using azid.dll v1.9 (b922) by Midas (midas@egon.gyaloglo.hu).

Logging start : 02/16/04 , 15:56:15.

D:\Tools\beSweet\BeSweet.exe -core( -input e:\DVDs\6channel-dg.ac3 -output e:\DVDs\6channel-dg- -6ch -logfile D:\Tools\beSweet\BeSweet.log ) -azid( -c normal -g 0.95 -L -3db )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : e:\DVDs\6channel-dg.ac3
[00:00:00:000] | Output: FL, FR, SL, SR, C, LFE
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Total Gain: 99.554dB, Compression: Normal
[00:00:00:000] | LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +---------------------
[00:00:00:000] <-- Transcoding Duration

Logging ends : 02/16/04 , 15:56:15.


What does "error 59" mean, and how can it be fixed?

Thanks in anticipation!!


I.

[pixita]

can someone point me some links to download the necessary software to do this?
Thanks

[ursamtl]

Originally posted by pixita
can someone point me some links to download the necessary software to do this?
Thanks

The BeSweet homepage at http://dspguru.doom9.org/ should be a good starting point.

You can also find some of the freeware at http://www.needfulthings.webhop.org/ in the audio/tools folder.

In addition, you can check Doom9 in the Downloads section. And, if in doubt, you usually find just about anything legal at www.google.com.

Happy encoding!

[skobipe]

Hi
I have 6 mono wav. files to make an .ac3 from.
I got those results in sounf forge:

* Merging the front left/right channels into streo file an
checking the rms gives: -29.8 DB.
* Merging the surround left/right channels into streo file an
checking the rms gives: -34.5 DB.
* Checking the rms of the center channel gives: -34.4 DB.

How could I know the right value to put in the dialog normalization
according to these results?

Thank You

[ursamtl]

Originally posted by skobipe
Hi
I have 6 mono wav. files to make an .ac3 from.
I got those results in sounf forge:

* Merging the front left/right channels into streo file an
checking the rms gives: -29.8 DB.
* Merging the surround left/right channels into streo file an
checking the rms gives: -34.5 DB.
* Checking the rms of the center channel gives: -34.4 DB.

How could I know the right value to put in the dialog normalization
according to these results?

Thank You

First look for your loudest result of these three, which is the fronts. Then take 31 + -29.8, to get an attenuation of 1.2dB..

Here are a couple of good references:
http://www.macprovideo.com/aPack/dialNorm.html
http://www.hometheaterhifi.com/volume_7_2/feature-article-dialog-normalization-6-2000.html

[skobipe]

@ursamtl

Thank you!
The guides were very helpful.
If I may, I have another minor question:
I use soft encode for encoding and in the "preprocessing" tab
I can choose "90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.
I wanted to know which of the two, if any, will produce better results.

Thank You

[ursamtl]

Originally posted by skobipe
@ursamtl

Thank you!
The guides were very helpful.
If I may, I have another minor question:
I use soft encode for encoding and in the "preprocessing" tab
I can choose "90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.
I wanted to know which of the two, if any, will produce better results.

Thank You

You're very welcome skobipe. The more we share our collective knowledge, the more progress we all make.:)

The 90° phase shift is not essential for decoding through a Dolby Digital 5.1 system, but it is if your mixed might be played back on a Dolby Pro Logic or surround system. These expect the surrounds to be phase shifted by 90°

The 3dB attenuation is usually necessary for files that will end up being played back on consumer home theater equipment.

If you want details just Google them and you'll find there's tons of stuff around on the net.

Hope this helps. Happy mixing!
Steve.

[ursamtl]

One other good reference for AC3 of course is the www.dolby.com site in their Information menu.

Steve.

[ursamtl]

In another thread, I read that SurCode Dolby Digital v2 and the AC3 Encoder for Acid use newer Dolby libraries than SoftEncode. Has anybody been able to compare the results from encodes done with both libraries?

Steve.

[keithmac]

"90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.

Are these 2 options actually altering the sound itself or just metadata tags that are read by the decoder?

The 90 degree phase shift option, if it actually alters the sound data itself, and you apply it to waves derived from a dvd ac3 source (that should already have been phase shifted) surely it will be shifted by 180 degrees on the 2nd ac3?.

Same with the 3db attenuation, is it just a metadata value or is the sound itself altered while encoding?

Would be interesting to have a list of what options are just metadata for the decoder to read and what options actually modify the sound during encoding.

[ursamtl]

Originally posted by keithmac
"90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.

Are these 2 options actually altering the sound itself or just metadata tags that are read by the decoder?

The 90 degree phase shift option, if it actually alters the sound data itself, and you apply it to waves derived from a dvd ac3 source (that should already have been phase shifted) surely it will be shifted by 180 degrees on the 2nd ac3?.

Same with the 3db attenuation, is it just a metadata value or is the sound itself altered while encoding?

Would be interesting to have a list of what options are just metadata for the decoder to read and what options actually modify the sound during encoding.

Hi keith,

There's a good explanation of this at YOU ARE SURROUNDED ([URL=http://www.soundonsound.com/sos/dec01/articles/surround5.asp).

Regards,
Steve.

[keithmac]

Thanks for the link, some good reading there!

[Capturebat]

In The AC3 machine it has dialog normalization reduction, but you can't input a value. Does it automatically scan and adjust for you?

[Paulcat]

It seems i'm in the right thread for this (I hope!)

I have a video file which has AAC 5.1 audio. I can use FAAD to convert that to either a 5.1 WAV file or a stereo WAV file. I would like to take the audio from AAC 5.1 to AC3 5.1 in order to put this video onto a dvd and still have surround. I use TMPGEnc DVD Author for mastering DVD's which I assume will remux my video with AC3 5.1, but for converting video files I use TMPGEnc Plus which downmixes everything to stereo.

What would be the SIMPLEST way (eg the LEAST amount of different software packages!) for me to convert a 5.1 WAV file into a 5.1 AC3 file?

I attempted this with BeSweet 1.4 and AC3 Machine (and BeSweet GUI 0.6) but it stated the AC3Enc.dll was missing (which I have since found) only to read that this DLL is next to useless for creating proper AC3 audio streams (or at least that's what people appear to be saying).

I'm new at this and the terminology is still way over my head, HELP!

[00diabolic]

Hi

I have been encoding my moviez with mp3 audio for as long as I can remember. I know all the tricks when it comes to video. Yet audio has always been mp3.

Recently I got ahold of a movie and was shocked that the audio was ac3 and file size was still normal. I figured ac3 had to be huge cuz thats whats on dvd's. So I started testing and found ac3machine, ac3enc.dll, and besweet (which I have used with gknot before) and discovered I could make an ac3 that was 128kps and was the exact same size as my audio in mp3 format. That blew my mind. 5.1 Ch 128kps audio same size as mp3 2 ch.

In the ac3machine guide it says "Set whatever bitrate you see fit. Going below 224kbit/s for a 5.1ch AC3 doesn't make much sense and going above the input bitrate doesn't make any sense either."

Then I read the above. Yet At 224 that audio was twice as big 220megs, no good. 128 same size as Mp3 @ 128 around 125megs.

So my question is this. Is ac3 5.1 128kps audio better then mp3 2ch 128kps audio that I have been using for years?

I have a 5.1 ch system and from what I can hear the 5.1 sounds good in 128kps except for a lil hiss. So what am I missing. Is it better quality then mp3 or not?

Would that ac3 encoded to 128kps be better with a different encoder?

Ac3machine uses the ac3enc.dll and I know that its not as good as other encoders. Can I use the sonic foundry soft encode dll with ac3machine, and how would I do that? or do I have to use soft encode to do it all? Soft encode is slow, is there a fast way to encode with ac3machine and not use the ac3enc.dll?

Any suggestions would be helpful. I know that both AAC and mp3 both have 5.1 surround options would using one of them result in a file around 125megs.

THANK YOU

[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.

[raquete]

no,it's not extracted from original ac3,it's using stereo cda.

ok,thanks tebasuna51,i got the "feeling"!
;)

thank you too for the "update" in the guide SomeJoe.

[Sakuya]

I have an official DVD with a 5.1 surround track that seems to be lacking in the music area. The soft music just can barely be heard for some reason. The action packed music for action scenes can be heard somewhat but is often drowned out by the explosion sounds.

Is there a way to fix this, maybe by re-encoding in Soft Encode? :scared:

[maa]

maybe by re-encoding in Soft Encode? Or Nero7 !

[Sakuya]

Or Nero7 !

I only have Nero 6. :scared: Any other methods?

[raquete]

dialog normalization doubts remains...

same source at 100% and at 70%,in audition "group waveform normalize" was found the averages -16.11 from the 100% source and -19.21 from 70% source to be used in "dialogue normalizations".
http://img105.imageshack.us/img105/6571/10070jx3.png


here are the results in Ciler's AC3 tools" from the waves encoded as AC3 :
http://img105.imageshack.us/img105/7396/10070ac3fd5.png


.waves: 100.wav = 0dB and 70.wav = -3.098dB, then the sources have 3dBs differences.
.ac3: 100.ac3 using dialnorm 16 and 70.ac3 using dialnorm 19, then the results have 3dBs differences.

what i mean and ask:
why the 100.ac3 have too much more volume than 70.ac3 if the sources are 3dBs differents and was encoded with the same 3dBs differences where using -16 in dialnorm sounds louder than -19?

it all means that sources with differents volumes result in differents dialnorms adjusts.
sources louder remains louder and sources lower remains lower? in the end,what dialnorm is doing? ...a mess? :scared:

(why) the volumes are not equals or seamless in the HT or receiver/decoder(or in pc with MPClassic,powerdvd,etc) (?)

the parameters are right(i'm sure they are following the guide) or have something wrong with the parameters used?

we don't have to "treat" the volumes of the sources and later encode the AC3?


i (we?) need one "reference",something like a "pre-dialnorm treatment" or "know level for reference"...or something like this.


thanks. :thanks:

corrections are very welcome! ;)
(you can do your own fast test or i can host the ac3 samples if needed)

[tebasuna51]

- The encoder put DialNorm like a parameter in the header of ac3 frames, is also used for calculate the DRC info (at begining of each audio block, if present), but is not used to modify the audio samples. Then audio samples from same wav are the same with DialNorm -16 or -19. Audio samples from 100.ac3 are loud than 70.ac3 because DialNorm don't change the audio samples.

- The decoder, ideally instructed by the user, can:
1) Apply DialNorm, then
100.ac3 (0 db) - (31-16) = sound at -15 dB
70.ac3 (-3 db) - (31-19) = sound at -15 dB

2) Don't apply DialNorm
100.ac3 (0 dB) = sound at 0 dB
70.ac3 (-3 db) = sound at -3 dB

Now answers to:
why the 100.ac3 have too much more volume than 70.ac3 if the sources are 3dBs differents and was encoded with the same 3dBs differences where using -16 in dialnorm sounds louder than -19?
This decoder don't use DialNorm value.
it all means that sources with differents volumes result in differents dialnorms adjusts.
sources louder remains louder and sources lower remains lower? in the end,what dialnorm is doing? ...a mess?
Sources louder remains louder, and DialNorm is intended to offer same volume (-31 dB in average) with decoders than uses DialNorm.
(why) the volumes are not equals or seamless in the HT or receiver/decoder(or in pc with MPClassic,powerdvd,etc) (?)
Check the defaults/settings of your decoders, seems you have different adjusts.
the parameters are right(i'm sure they are following the guide) or have something wrong with the parameters used?
Seems DialNorm is well calculated.
we don't have to "treat" the volumes of the sources and later encode the AC3?

i (we?) need one "reference",something like a "pre-dialnorm treatment" or "know level for reference"...or something like this.
The same question for me. Something to do with the volume before encode?: Nothing.

Now my question is:
1) Do you want make a Dolby compliant ac3 to be played with Dolby compliant decoders, together with another Dolby compliant material, to be listened all at same average volume (-31 dB)?. If you answer yes, calculate the DialNorm and apply the appropriate DRC.

2) Do you want an ac3 to be listened together with other material (CDAudio, Mp3, Commercial TV, ...) at same volume?. If you answer yes then maximize the wav source and encode with DianNorm = 31 and DRC = None. This method (not Dolby compliant) use ac3 like other lossy encoders (mp3, ogg, aac, ...), and the ac3 volume must be the same than the wav source with all decoders.

[raquete]

first and most important:thank you for answers.

Now my question is:
1) Do you want make a Dolby compliant ac3 ... yes!

If you answer yes, calculate the DialNorm and apply the appropriate DRC.was done.

Seems DialNorm is well calculated.
yes,i just :readguid: and follow.

Check the defaults/settings of your decoders, seems you have different adjusts.was done too and the adjust is right.

Something to do with the volume before encode?: Nothing.ok.then...(back to the beginning)

- The decoder, ideally instructed by the user, can:
1) Apply DialNorm, then
100.ac3 (0 db) - (31-16) = sound at -15 dB
70.ac3 (-3 db) - (31-19) = sound at -15 dB
you don't want to mean here: 70.ac3 (-3 db) - (31-19) = sound at -12 dB ?
...or i'm really lost and :stupid:

if yes,(i'm sure following the metadata parameters)then the 100.ac3 remains louder!

from Dolby Metadata guide.pdf:
http://img239.imageshack.us/img239/9200/metadataca0.png

Sources louder remains louder, and DialNorm is intended to offer same volume (-31 dB in average) with decoders than uses DialNorm.
how can the volume level be consistent after follow this "rules" from metadata guide ?

...played with Dolby compliant decoders, together with another Dolby compliant material, to be listened all at same average volume (-31 dB)?.
how we get same volume if 100.ac3 sound at -15 dB and 70.ac3 sound at -12 dB ?


corrections (please) are welcome tebasuna51.

thanks so much. ;)

[tebasuna51]

"1) Apply DialNorm, then
100.ac3 (0 db) - (31-16) = sound at -15 dB
70.ac3 (-3 db) - (31-19) = sound at -15 dB"
you don't want to mean here: 70.ac3 (-3 db) - (31-19) = sound at -12 dB ?
...or i'm really lost
Maybe :rolleyes: , because -3 - 31 + 19 = -15.

In others words:
100.wav: max peak 0 dB, average -16 dB
70.wav: max peak -3 dB, average -19 dB

And encoded and decoded with DialNorm:
100.ac3: max peak -15 dB, average -31 dB
70.ac3: max peak -15 dB, average -31 dB

For that the volume level is consistent after follow this "rules" from metadata guide.

[raquete]

because -3 - 31 + 19 = -15.
but tebasuna, -3 is the dB level of the wav source with 70% and is not to be used in the calculations.
as 100% is 0dB, 70% is -3db(-3.092).

[tebasuna51]

but tebasuna, -3 is the dB level of the wav source with 70% and is not to be used in the calculations.
as 100% is 0dB, 70% is -3db(-3.092).
Wrong.
For what is 70% DialNorm -19 dB, just -3 dB than 100% DialNorm (-16 dB) ?
Because 100% need be attenuated 3 dB more than 70% to obtain the same output level.

For that is useless attenuate or maximize the sources before encode (for Dolby compliants ac3):
100% -> DialNorm = -16
70% -> DialNorm = -19
Output volume -> the same, max peak -15 dB, average -31 dB

[raquete]

ok.the sources and results are 3db differents form each other,we are talikng the same thing...
viewing from another angle...

the sources have to match the -31 standard where "31+(dialogue level value)=shift applied"

-16 was the average from 100%.wav:
x = 31 - 16
x = 15

and -19 from 70%.wav:
x = 31 - 19
x = 12

where "x" represent "shift applied" in the decoder to match -31.
(15dB and 12dB in this examples and they have(the same)3dBs of differences as sources)

the mess was: the parameter "shift applied"(x) is for the decoder and not "to encode".

see? i'm feeling stup but have remedy,i have hope. (lol)
(lol,i'm very complicated)

seems "logical" now tebasuna.
right? (or wrong?) :thanks:
:)

[raquete]

source: Led Zeppelin-Presence 00:44:20.866 (all 7 tracks as one single track)

from a.audition1.5 -17.21, from s.forge8a -14.4

:p what you choose?....i'm lost tebasuna.

regards. ;)

edit: screenshots later if needed

[tebasuna51]

from a.audition1.5 -17.21, from s.forge8a -14.4

:p what you choose?....i'm lost tebasuna.
Of course -16 (always minimize the possible errors:D , and take decisions without doubts :cool: )

[raquete]

Of course -16 ok.

and take decisions without doubts) no doubts.
...was only one "fast indecision" with more than 1 option. (lol)

(always minimize the possible errors..) i did it :p ...i minimize asking my :helpful:

:thanks: so much!

[Boulder]

from a.audition1.5 -17.21, from s.forge8a -14.4

Out of curiosity, do you have the exact same settings in both programs? In Audition there's two settings for calculating the RMS values, "0dB=FS Sine Wave" and "0dB=FS Square Wave". I just did a quick test and the first one shows -10.4dB and the latter one -13.41dB for average RMS power.

[raquete]

In Audition there's two settings for calculating the RMS values...
in "group waveform normalize" ? :confused:

[Boulder]

in "group waveform normalize" ? :confused:
Nope, in Window->Amplitude Statistics.

EDIT: if I analyze the WAV file via Group waveform normalize, I get -9.61dB as the average. When I get the stats via Window->Amplitude Statistics, it shows that the Total RMS Power is ~-9.61dB (left channel -9.64 and the right -9.58dB) but the average RMS power is ~-10.4dB with FS Sine Wave selected. I've always used the latter method for getting the avg RMS power.

[raquete]

in Window->Amplitude Statistics

square wave:
Left Right
Min Sample Value: -32768 -29392
Max Sample Value: 30988 29073
Peak Amplitude: 0 dB -.94 dB
Possibly Clipped: 1 0
DC Offset: -.695 -.382
Minimum RMS Power: -inf dB -inf dB
Maximum RMS Power: -10.69 dB -11.38 dB
Average RMS Power: -20.74 dB -21.46 dB
Total RMS Power: -19.87 dB -20.59 dB
Actual Bit Depth: 16 Bits 16 Bits

Using RMS Window of 50 ms


Sine wave:
Left Right
Min Sample Value: -32768 -29392
Max Sample Value: 30988 29073
Peak Amplitude: 0 dB -.94 dB
Possibly Clipped: 1 0
DC Offset: -.695 -.382
Minimum RMS Power: -inf dB -inf dB
Maximum RMS Power: -7.68 dB -8.37 dB
Average RMS Power: -17.73 dB -18.45 dB
Total RMS Power: -16.86 dB -17.58 dB
Actual Bit Depth: 16 Bits 16 Bits

Using RMS Window of 50 ms

regards.

[Boulder]

Humm, it doesn't explain the huge difference between Sound Forge and Audition.. and it can't be any rounding errors because the difference is too big for that :confused:

I don't have SF so I can't check what it would say about my sample WAV.

[raquete]

don't worry Boulder,i have both and i'm double confused...in what editor we can really trust? :confused:
(this is one of lots reasons that i posted "Dialogue normalizations for music (test)" thread and(maybe) i am the first to find this ... http://forum.doom9.org/showthread.php?t=114390 ...seems nobody care :( )

:thanks:

[raquete]

Boulder are you around?
did you found one final conclusion about the difference between Sound Forge and Audition?
what i saw is that checking "use equal loudness control" give one short rms and without check give high rms.the rms between this values from sound forge is seamless the rms from audition...but this is not one conclusion,only a single observation.
another detail from sound forge help:
"Ignore below
Drag the fader to determine the level of material you want to include in the RMS calculation. Any sound material below the threshold will be ignored in the calculation. This is useful to eliminate any silent sections from the RMS calculation. You should set this parameter a few dB above what you consider to be silence.
If you set this value to minus infinity, all sound data will be used. If the value is set too high (above -10 dB), there is a good chance that the RMS value is always below the threshold. In this case, no normalization will occur. Therefore, it is good to test the threshold by using the Scan Levels button."
the screenshot from the guide is using the default "[Sys] Normalize RMS to -16 dB (music)" where the Scan settings is in -45,0 dB(0,56%) where all sound with less level is ignored as silence when use Scan Levels to find the RMS.

your comments please.
best regards. :)

[Boulder]

I never got into any real conclusion - I just decided to keep on using the value Audition gives me. It apparently doesn't have the choice to ignore "silent" areas. Besides, that would just give me another setting headache :p

[maa]

Are you guys forgetting that some studio programs have "Pan Law" activated and other don't ?

Just a thought,

maa

[raquete]

@ Boulder
Audition is my choice too. ;)
that would just give me another setting headache lol.
but wait,maa give us another headache.(joke) :p

@ maa
Are you guys forgetting that some studio programs have "Pan Law" activated and other don't ?
no,i don't forgot because i don't know about "Pan Law".
tell us more.

[maa]

Right, I'll try.

When a mono signal is panned from one speaker to the other it doubles its volume when in the middle position because its now using two amplifiers and two speakers set the same.
All good hardware mixers and most software compensate for this to allow equal loadness by dropping the center volume by 6db, or 4.5 or 3db depending on mono or stereo and the way the software mixes the channels.
This is a major reason why some people think one software sounds totally different to another - if this is not adjusted the same or switched off then dissaster will follow.

Hope this helps a bit....


Cheers

M

[raquete]

When a mono signal is panned from one speaker to the other it doubles its volume when in the middle position...
you're right.
this is why i use hard center and sides(surrounds) without center.

:D

[w0rd™]

I've got some questions, just to make it abit clearer in my head, so sorry if its been explained already, but so far its not so clear for me.

1. If i'm making a 5.1 ac3 mix and i have six wavs-
w1 : -10db rms
w2: -15db rms
w3: -14db rms
w4: -17db rms
w5: -5db rms
w6: -10db rms

how do i find the dialnorm for the final ac3 file? do i have to normalize the 6 wavs first?


2. If i get my final dialnorm figure from above for the 5.1 ac3, and on the dvd i also have music for the menu which is not ac3, what would i have to normalize the menu audio to, so that its the same level as the final ac3 soundtrack?

3. In adobe audition, which value is the best to use for dialnorm?
It gives for example this output in group waveform normalize:

Eq-Loud: -13.15
Loud: - 14.26
Max: -10.29
Avg: -15.91



thanks for any help.

[raquete]

1- ...how do i find the dialnorm for the final ac3 file?
maybe it can be done usisng the Center channel alone but i'm not sure.
as is not clear what track means w1,w2 and remainders...
do a new file and paste L in the left and R in the right and save.
follow the guide in "Example Compression Settings",is very clever...
http://forum.doom9.org/showthread.php?t=56020

2- ...what would i have to normalize the menu audio to.. do the same as posted for the first question.

3- In adobe audition, which value is the best to use for dialnorm?
Avg:

regards!

[w0rd™]

yes, well i guess it has to be the centre channel, or the dialogue part of the wav file, since its for the dialnorm value! even if there's dialogue in the rear channels, the most practical way is to choose main dialogue....

thanks.

[w0rd™]

just one more thing, is there any option in audition that will allow me to highlight a certain part of the audio and check the rms of just the part i want?
wave group normalize scans the whole clip, i want to check rms value of just highlighted part.

thanks.

[Boulder]

Choose the area and then select Window->Amplitude statistics.

[raquete]

Choose the area and then select Window->Amplitude statistics.
yes,cool option for only one file.
as he have more than one track(i don't know how many in the menu)better is not better load all in edit view and use group waveform normalize than one by one?

i think that is faster but i want to know your opinions/comments Boulder!
( audition have lots of "secrets features",maybe you know something that i ignore,i can bet)
;)

off topic:
i'm sending pm to you(is not about this topic)
:)

[Boulder]

Maybe he should use only the center channel, or center + the two front channels for determining the dialnorm value. Just choose the same part in each file, the selection area timecodes can be seen at the bottom right corner.

[raquete]

yes,seems one good option!

thank you.

ot: doh :-/ ...big typos in my last post,i need to edit :o

[Rockaria]

Right, I'll try.

When a mono signal is panned from one speaker to the other it doubles its volume when in the middle position because its now using two amplifiers and two speakers set the same.
All good hardware mixers and most software compensate for this to allow equal loadness by dropping the center volume by 6db, or 4.5 or 3db depending on mono or stereo and the way the software mixes the channels.
This is a major reason why some people think one software sounds totally different to another - if this is not adjusted the same or switched off then dissaster will follow.

Hope this helps a bit....


Cheers

M
Thanks for the good info.

[raquete]

When a mono signal is panned from one speaker to the other it doubles its volume...not always!
samples? mine(i have lots) or yours?
regards

[dvdboy]

Ok, I think I follow the Dialogue Normalisation part, but I couldn't seem to see an 'idiots guide' to DRC.

I've got a piece of video with interviews and light background music, so scan the average RMS with soundforge at several places where there is speech and very little music I get:

-17.9
-20.5
-21.1
-19.5

Divided by 4 gives me an average of -19.75, so I've set the Dialogue Normalisation within Sonic's encoder to -20dB.

But how do I calculate which DRC preset to use?

Is it safe just to set this to none?

[raquete]

hi dvdboy.
-20dB seems good!
reading another thread with seamless question,the average value is the logical and safe answer.
take a look: http://forum.doom9.org/showthread.php?t=122212

[foxyshadis]

DRC is meant to keep the levels fairly constant, so if they already are that way, there's no need for it. If the music is much louder than the dialogue, you would want it (though for a talk show, you'd probably just want to remaster it instead). The more variation between soft dialogue and loud noise, the heavier the preset.

[kOoL tHuG]

I have converted a Pal Video to Ntsc video. But now when i remux the ac3 file to the video i get audio synce issues. I guess it is coz my video was 25 Fps before and now its 29.97 Fps. Now can plz somebody help out with how to reencode the ac3 file to properly get the audio in sync with the video?

P.S - I used Besweet and Belight. I basically didn't like the quality i get after i reencoded the original ac3 file.

[loekverhees]

Hello,

I've analyzed a 2 channel wavefile in Adobe Audition. These are the Amplitude Analysis data (FS Sine Wave used):


Left Right
Min Sample Value: -2791 -2792
Max Sample Value: 3635 3633
Peak Amplitude: -19.1 dB -19.1 dB
Possibly Clipped: 0 0
DC Offset: 0 0
Minimum RMS Power: -73.89 dB -73.41 dB
Maximum RMS Power: -27.9 dB -27.89 dB
Average RMS Power: -43.84 dB -43.82 dB
Total RMS Power: -40.8 dB -40.79 dB
Actual Bit Depth: 16 Bits 16 Bits

Using RMS Window of 50 ms


Using RMS Window of 50 ms

So, I would choose -41 dB for the dialog normalization in Sonic Foundry Soft Encode. However, the lowest value I can choose from the pulldown menu in Soft Encode is -31 dB. So, what should I choose?

[tebasuna51]

I've analyzed a 2 channel wavefile in Adobe Audition.
...
So, I would choose -41 dB for the dialog normalization in Sonic Foundry Soft Encode. However, the lowest value I can choose from the pulldown menu in Soft Encode is -31 dB. So, what should I choose?

The use of Dialog Normalization is obsolete, now each stream try to offer the max possible volume.

You can amplify your wave +19dB to reach the max peak volume at 0dB (Peak Amplitude: -19.1 dB), and you can put the -31 dB value for DialNorm to avoid any attenuation.

[loekverhees]

OK, thanks for the answer. But just wondering why the Dialog Normalization is obsolete now, and why was it useful some time ago?

[b66pak]

offtopic...do you know some freeware audio analyzers?
_

[tebasuna51]

OK, thanks for the answer. But just wondering why the Dialog Normalization is obsolete now, and why was it useful some time ago?

If all your audio sources are ac3 the DialNorm can be a useful method to have the same volume levels.

The problem is when you listen CD Audio, mp3, commercial tv and other sources without this control. All other sources try to output the bigger possible volume, and the ac3 sounds with DialNorm have a big handicap.

[loekverhees]

If all your audio sources are ac3 the DialNorm can be a useful method to have the same volume levels.

The problem is when you listen CD Audio, mp3, commercial tv and other sources without this control. All other sources try to output the bigger possible volume, and the ac3 sounds with DialNorm have a big handicap.

Ok, I get it. Thanks again for the answers :).

[b66pak]

offtopic...do you know some freeware audio analyzers?
_

nobody know?
_

[tebasuna51]

Free audio editors are Audacity and Wavosaur and have some functions to analyze the sounds.

[krabapple]

I am starting from six-channel music files recorded from SACDs directly off the 6ch analog output of an SACD player. I used Audition to record (soundcard is an M-Audio 1010lt). Recording level is such that the highest peak is just below 0dBFS (i.e., no clipping in any channel). Each recording is saved as a single interleaved 6ch .wav (PCM) file (I can also save as six mono .wav files if I want to).

I want to convert the 6ch .wav (PCM) file to AC3 so I can play it via S/PDIF connection from my laptop and let my AVR decode it; I *don't* want to change any levels at playback and I *don't* want to add any compression. So is there a way to bypass Dialnorm and DRC? Btw, for encoding I am using the FFmpg library with Audacity (beta) to do wav-->AC3 conversion, and I don't see any options other than setting bitrate and channel mapping....

[tebasuna51]

You can use Aften like encoder, based in ffmpeg library but with more settings.

There are a GUI to set all the parameters: WavToAc3Encoder (http://wavtoac3encoder.googlecode.com/files/EncWAVtoAC3-4.5.zip)

Or you can also encode with Audacity using Aften, like external program, and this command line:
d:\path\Aften -b 640 -readtoeof 1 -exps 32 -s 1 - "%f".ac3
see this post: http://forum.audacityteam.org/viewtopic.php?p=72939#p72939

The default is don't use DialNorm/DRC

[nikitenko.p]

i have never tried that, i imagine that it is not a type for the begginer.. especially one who is as uncorodinated as myself. Are the positions similar to normal, or are they specialised to be done without sight?

[nike22]

I also want to know the answer to this question.

[tebasuna51]

Please, clarify the question.

[krosswindz]

Is there any guidelines as to what should be the minimum bitrate that should be chosen for AC3 files. I read some where that for 5.1ch AC3 the "legal" minimum bitrate would be 384 kbps. I looked into the encoding guidelines (link available in the first post) it doesnt seem to mention any such limit though it has recommended settings.

[krosswindz]

^ Is there some documentation that one can refer to decide on the minimum bitrate that one can use for different channel setup.

[tebasuna51]

^ Is there some documentation that one can refer to decide on the minimum bitrate that one can use for different channel setup.
I don't know docs, the limits using Soft Encode are:

Low bitrate minimum [recommended]:
(1.0) 56, 64, 80, [96]
(2.0) 96, 112, [192]
(3.x) 128, 160, [256]
(4.x) 192, [384]
(5.x) 224, 256, 320, 384, [448], 512, 576, 640

[krosswindz]

I don't know docs, the limits using Soft Encode are:

Low bitrate minimum [recommended]:
(1.0) 56, 64, 80, [96]
(2.0) 96, 112, [192]
(3.x) 128, 160, [256]
(4.x) 192, [384]
(5.x) 224, 256, 320, 384, [448], 512, 576, 640

These are the very same settings from Dolby Documentation that I specified in my earlier post. What you say from this is that technically it is legal to gave 224kbps 5.1 AC3 soundtrack.

[tebasuna51]

Audio bandwidth (filtered signal) for AC3 5.1:
224 Kb/s 9 KHz
256 Kb/s 12 KHz
320 Kb/s 16 KHz
384 Kb/s 18 KHz
448 Kb/s 20 KHz

[jruggle]

Audio bandwidth (filtered signal) for AC3 5.1:
224 Kb/s 9 KHz
256 Kb/s 12 KHz
320 Kb/s 16 KHz
384 Kb/s 18 KHz
448 Kb/s 20 KHz
From a purely technical standpoint, you can go as low as 32kbps at up to around 20kHz bandwidth as long as the encoded data will fit in the frame. An encoder like Aften will spit out an error if the settings need to be changed (bitrate raised or bandwidth lowered). Of course, quality will generally not be very good at bitrates lower than what is recommended, but Aften leaves that decision up to the user.

[tebasuna51]

Yes, with Aften you can select anything.
I put the values (bitrate limit and bandwidth) used by the old Sonic Foundry Sof Encode like reference.

[geminigod]

Great guide, Somejoe, for a topic that is still relevant and confusing to many today! As a supplement to your guide, I wrote my own 10 cents on the subject in a different forum that some confused souls might find helpful.

http://www.fanedit.org/forums/showthread.php?4989-5.1-surround-sound-dialog-normalization

[mike20021969]

I know post #1 is almost 10 years old, but some, if not all, of the links are now dead.

[filler56789]

I know post #1 is almost 10 years old, but some, if not all, of the links are now dead.

The images with broken links are archived at:

http://web.archive.org/web/20050208121358/http://pages.sbcglobal.net/wilsondr/

[Guest]

It looks like our mod tebasuna has updated the links for us. Please let us know if there are any other things that need to be tidied up.

[szabi]

Hi

Are there any way to encode DTS-HD, DTS-MA or TrueHD audio source to Dolby Digital Plus (EAC3)?
I tried TAudioConverter (http://forum.doom9.org/showthread.php?t=165577), UsEac3to (http://forum.doom9.org/showthread.php?t=145574), PopCorn MKV AudioConverter (http://www.networkedmediatank.com/showthread.php?tid=20887), Eac3to and More GUI (http://forum.doom9.org/showthread.php?t=135095) and Clown_BD (https://forum.slysoft.com/showthread.php?25818-Movie-Only-Copies-eac3to-tsMuxer-amp-ImgBurn-Made-Easy) but none of them could do.
Any suggestion would welcome.

bye
szabi

[tebasuna51]

The only free encoder to eac3 I now is ffmpeg: http://trac.ffmpeg.org/wiki/GuidelinesHighQualityAudio , but I never tested it.

[Richard1485]

@szabi

I am curious as to why you would want to convert gold to lead. In my experience, the three formats that you listed are all better supported than E-AC-3. If space is a consideration, they all have easily extracted cores.

[szabi]

Thnx the link trac.ffmpeg.org (http://trac.ffmpeg.org/wiki/GuidelinesHighQualityAudio). Do you know about any GUI for this?
I use Aften for AC3 encoding, unfortunately the developer of aften seems finish the work to update it for E-AC3. Free E-AC3 encoder available....... (http://forum.doom9.org/showthread.php?t=161383)

@Jeff B
Yes, space saving is the key point.
There was TrueHD source but no AC3 core.
Additionally I wanted to keep the plus channels (7.1 vs 5.1), which is not possible in AC3 but can be in E-AC3.

[Richard1485]

Additionally I wanted to keep the plus channels (7.1 vs 5.1), which is not possible in AC3 but can be in E-AC3.

That makes sense, I suppose. :)