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Old 18th October 2003, 18:10   #21  |  Link
SomeJoe
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Quote:
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. 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.

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

Edit:

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

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.
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.

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.
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Last edited by SomeJoe; 21st October 2003 at 22:22.
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Old 30th October 2003, 10:01   #22  |  Link
macik-pacik
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AC3 and Stereo

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.
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Old 3rd November 2003, 23:29   #23  |  Link
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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?
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Old 4th November 2003, 17:57   #24  |  Link
SomeJoe
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Quote:
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.
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Old 13th November 2003, 13:08   #25  |  Link
Kilyan
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Audio too quiet

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
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Old 14th November 2003, 15:46   #26  |  Link
SomeJoe
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Re: Audio too quiet

Quote:
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.
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Old 14th November 2003, 22:29   #27  |  Link
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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?
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Old 22nd November 2003, 18:54   #28  |  Link
tbrunner
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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)?

Last edited by tbrunner; 22nd November 2003 at 19:43.
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Old 22nd November 2003, 18:58   #29  |  Link
tbrunner
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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.

Last edited by tbrunner; 22nd November 2003 at 19:21.
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Old 23rd November 2003, 18:56   #30  |  Link
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@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).
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Last edited by Mug Funky; 23rd November 2003 at 19:08.
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Old 21st January 2004, 17:48   #31  |  Link
Soapm
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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?
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Old 22nd January 2004, 15:55   #32  |  Link
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Quote:
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.
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Old 25th January 2004, 01:13   #33  |  Link
magilvia
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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
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Old 25th January 2004, 16:25   #34  |  Link
magilvia
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One other question too
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!
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Old 25th January 2004, 16:52   #35  |  Link
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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.
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Old 14th February 2004, 23:11   #36  |  Link
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Re: Strange Problem with AC3 File and TMPGenc DVD Athor

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.

Quote:
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 arenīt 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.



Last edited by oldiexyz; 16th February 2004 at 16:00.
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Old 22nd June 2004, 18:54   #37  |  Link
pixita
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can someone point me some links to download the necessary software to do this?
Thanks
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Old 22nd June 2004, 19:22   #38  |  Link
ursamtl
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Quote:
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!
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Old 20th August 2004, 03:45   #39  |  Link
skobipe
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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
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Old 20th August 2004, 13:06   #40  |  Link
ursamtl
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Quote:
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/volum...on-6-2000.html
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