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    Question withdrawn...
    Last edited by ziggy1971; 17th May 2014 at 17:05.
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    I need to convert many ac3 files back to original wav files for editing;
    You can make them wav's again but you will never get the original wav if that is what they were originally.
    AC3 is a LOSSY compression, just like MP3, once the information is gone, it is gone.

    Now if you had them in FLAC, APE, etc. you could get back the "original" wav files as those are LOSSLESS forms of compression.
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    If you would explain why on earth you feel it's necessary to edit AC3 files (I can think of one good reason, but I want you to say why you are doing it) and what your final goal is, we might be able to judge if you're even doing this in an ideal way or not.

    Noahtuck's post is not very clear in my opinion. He is trying to make the point that as AC3 is lossy, converting to WAV does not make them better in any way. But I think you were implying that your reason for the conversion was that you don't have anything that can edit AC3, not that you think that AC3 -> WAV makes things better. Well, at least I think that's why you are implying.
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    Well, my point was, if he thinks he is going to get the original WAV back he is mistaken,
    I need to convert many ac3 files back to original wav files for editing;
    Not sure how that is unclear ?

    Now when he say's this,
    I'm looking for a good tool to convert ac3 files to wav files, preserving the original dynamic range
    The original "dynamic range" of what ?
    The original WAV file he seems to want to get back or of the AC3 files he has ?

    But I guess the "OP" needs to be more clear.........

    Convert it to WAV if you need to edit it that way.
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  5. Audacity with the ffmpeg source plugin can open AC3 files (or AC3 audio in most A/V containers) and save as WAV files.
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    again...
    Last edited by ziggy1971; 17th May 2014 at 17:06.
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  7. Member hech54's Avatar
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    Originally Posted by ziggy1971 View Post
    I understand that I'll never get back to 100% "original" wav files, but even 98% is better than the DRC version
    I don't understand where you are getting these numbers from. Do you also think that there are certain processes/software that allow you to get 98% of the original quality of a 50MB wav to a 5MB MP3 conversion? If not....then please explain 98% of WHAT exactly.

    Originally Posted by ziggy1971 View Post
    I was under the impression that this was a place to ask for help.
    This is the no-nonsense message board. Your original quality is gone. The best you can hope for is your present AC3 quality. Everything you have been told in "this place", in this thread/discussion, is 100% accurate.

    Originally Posted by ziggy1971 View Post
    I guess I'm the first person to ask a question like this; yes I'm a little perterbed by the inclination...

    Then perhaps someone could explain the popularity of this post:
    http://forum.doom9.org/showthread.php?t=125966

    600+ pages and running...

    If I had a nickel for every time I heard this tactic (I guess I'm the first person to ask a question like this) I'd be a millionaire.
    And how to explain the doom thread? - Easy - a couple of geeks spouting percentages of something that cannot be accurately measured in the first place.
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    Originally Posted by Noahtuck View Post
    The original "dynamic range" of what ? The original WAV file he seems to want to get back or of the AC3 files he has?
    Professional encoders duly sanctioned and authorized by dolby come with a plenty of options. The person who edited and encoded the original soundtrack can choose to implement these options or not, and in what way (that fall within DD restrictions). Among those many options are taking/creating cues on levels in each discrete channel at chosen time points along the file and include that info in DRC metadata with the AC-3 stream (along with other data) that decoders can use to restrict dynamic range on playback, if it's so chosen on playback device.
    OP's success therefore depends on whether or not this DRC metadata is present in the original DD stream, and how his chosen s/w decoder recognizes and implements it. If it's present and not set or enabled in decoder, the resulting six decoded *.wav files should have a wider dynamic range (bigger difference between softest and loudest portions), compared with otherwise.
    I edit and encode original DD 5.1 streams with the events I shoot. Choosing what to put in each channel, what its relation to other channels should be at that point, whether or not to apply specific metadata or just encoder defaults, are all decisions that make creating the soundtrack just as complicated, if not more, compared with multi-cam NLE. Setting and implementing DRC in Hollywood action films is helpful in managing their (presumably) bombastic dynamic range, considering that these films will be played in equipment ranging from fancy home theaters to TVs with built-in DVD players. In contrast with the soundtracks I create, which consist mostly of speech, singing, and audience reaction, dynamic range rarely exceeds 10 to 15dB so I choose not to set DRC.
    For the nth time, with the possible exception of certain Intel processors, I don't have/ever owned anything whose name starts with "i".
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    again...
    Last edited by ziggy1971; 17th May 2014 at 17:07.
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  10. Member turk690's Avatar
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    Originally Posted by ziggy1971 View Post
    Encode any wav file to AC3, the decode it back to a wav file with something like BeLight. First decode the AC3 file with the Dynamic Compression set to Normal, then decode the same file without any Dynamic Compression. Open both files in a wav editor and you can clearly see the difference. Then try listening to the two different wav files, you can clearly hear the difference.
    The first assumption here is that the original DD stream was created with a legitimate DD encoder and decoded with a likewise decoder.
    Belight and besweet are nifty but they are not dolby authorized. Their authors admirably noted and mimicd dolby digital behaviour and wrote an implementation that is useful to a lot of NLeditors who don't have the means to fork dough for minnetonka surcode, for example. But it's hard to tell if and how it's implemented. Take DRC metadata. If set and used, it's supposed to tell how much gain a channel has at that instant, and how this behaviour is implemented: is the gain for that channel increased or reduced, is hard limiting applied and at what level, and after it is, should the volume at that instant be normalized, etc. No guesswork here; playback uses metadata. If this metadata is not recognized or is not used as intended, results will be different. For example, I can take any DD stream, transcode it as much as I can to its original *.wav files, then take these files, apply hard-limiting, and normalized at a constant level throughout. The results would give the impression of a reduced dynamic range, but is not identical to the case where DRC metadata was present, recognized, and implemented according to dolby specs with a legit authorized DD s/w or h/w decoder.
    For the nth time, with the possible exception of certain Intel processors, I don't have/ever owned anything whose name starts with "i".
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    I surrender...

    I know that most people can't fork over the dough for expensive "legitimate DD encoder and decoded with a likewise decoder", that's why I eluded to a simple FREE program, which does work well enough to experiment with and learn from even if it's not 100% accurate.

    Any yes, I do know about limiters, they've had a significant impact on the "Loudness Wars" in music and in theaters...

    Clearly this post is ruffling too many feathers; I'll delete whatever I can.
    Then I'll message Sony to request my money back because the AC3 encoders they use in their Vegas Pro & Sound Forge product line aren't "legit"...
    Last edited by ziggy1971; 17th May 2014 at 17:02.
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    ziggy1971, maybe you can extract the audio to waves files with eac3to and encode to waves to ac3 with aften. both of those are free.
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