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Old 20th June 2003, 20:56   #1  |  Link
SomeJoe
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How To Properly Encode Dolby Digital Audio (AC3)

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, which has the best information on Dynamic Range Compression. The other is Dolby Digital Professional Encoding Guidelines which gives an excellent explanation of the dialogue Normalization parameter. You will need Adobe Acrobat Reader 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, 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 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. 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 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 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 is the Bitstream Information. Set these parameters are appropriate for your source material.

The third page 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 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.
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Last edited by tebasuna51; 2nd October 2013 at 13:03. Reason: Updated links
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Old 21st June 2003, 21:45   #2  |  Link
kxy
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Thanks SomeJoe, for your clear and detailed explanations.
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Old 23rd June 2003, 08:19   #3  |  Link
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Great work, SomeJoe!

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

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Old 24th June 2003, 15:47   #4  |  Link
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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 ?
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Old 24th June 2003, 17:01   #5  |  Link
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Quote:
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.
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Old 9th July 2003, 04:21   #6  |  Link
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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?

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Old 9th July 2003, 21:44   #7  |  Link
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Quote:
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.
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Old 12th July 2003, 02:30   #8  |  Link
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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.

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Old 12th July 2003, 03:26   #9  |  Link
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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 ).

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Old 27th July 2003, 23:04   #10  |  Link
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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

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Old 4th August 2003, 00:42   #11  |  Link
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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?
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Old 4th August 2003, 13:16   #12  |  Link
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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?
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Old 4th August 2003, 15:38   #13  |  Link
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Quote:
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.


Quote:
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.
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Old 6th August 2003, 12:25   #14  |  Link
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Strange Problem with AC3 File and TMPGenc DVD Athor

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 arenīt recognized ??

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Old 8th August 2003, 03:34   #15  |  Link
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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.

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Old 8th August 2003, 09:14   #16  |  Link
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Hi,

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

@nuked I didnīt have the Intel byte order checked any time

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Old 8th September 2003, 19:07   #17  |  Link
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EXCELLENT Sticky but I don't have Sound Forge so I need some help please

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!

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Old 9th September 2003, 16:33   #18  |  Link
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Re: EXCELLENT Sticky but I don't have Sound Forge so I need some help please

Quote:
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.
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Old 9th September 2003, 23:52   #19  |  Link
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Re: EXCELLENT Sticky but I don't have Sound Forge so I need some help please

Quote:
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
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Old 14th October 2003, 03:31   #20  |  Link
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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,
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