eac3to - audio conversion tool

[Chumbo]

Originally Posted by Snowknight26

eac3to "G:\Encoding Tools\temp\FEATURE_1_MERGED.DD+.stream.00.mpa" "..\temp\audio\2fast.done.dts"
E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "..\temp\audio\2fast.done.L.wav"...
Creating/writing file "..\temp\audio\2fast.done.R.wav"...
Creating/writing file "..\temp\audio\2fast.done.C.wav"...
Creating/writing file "..\temp\audio\2fast.done.LFE.wav"...
Creating/writing file "..\temp\audio\2fast.done.SL.wav"...
Creating/writing file "..\temp\audio\2fast.done.SR.wav"...
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.L.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.R.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SL.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SR.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.C.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.LFE.wav.".
Surcode sais/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Main problem is the file names not being accepted, 2nd is that "sais" should be "says".
Would switching to v1.0.29.0 fix the path problems?

You shouldn't use relative paths like that since you're using an external program in this case. The surcode settings are registry-based and "..\" means nothing to SurCode and if it did, it would be relative to surcode's path and not what you intend. Stick to a fully-qualified path and you'll be fine.

If you're too lazy to type in the full path, then map a drive to your long-and-often-used location.

[Snowknight26]

eac3to is in the same folder as SurCode is, but oh well, I had a problem anyway that would have prevented it from getting the file names to begin with. Still the issue of sais.

[MichalHabart]

Hello Madshi,
i found strange thing. When i try to convert TrueHD to DTS, it correctz detects bit depth of TrueHD but it still creates all wav files twice, one for 24 bit and the second for 16. Can i somehow prevent this behaviour? Because it requires twice space on HDD and decode is then twice slower.

[Falcon4]

Originally Posted by Furiousflea

..... Just tried running EAC3To on a different computer in the house after installing the sonic audio decoder, now its just saying...



...I despair, would really appreciate some help, before it ran it did say that "msvcr71.dll" was missing? I assume the Sonic decoder needs this, I downloaded that dll and placed it in my system32 folder but then it just ran until the point with the quotes

Same problem I was having after moving to another computer to do my encoding dirty work. I knew it was because I didn't have the Sonic decoder installed, so I tried copying the DirectShow filters to the new computer and registering them. That got the filter to be identified, but it still wouldn't decode. I thought it was part of bundled software with the Vaio I ripped the Blu-Rays on, but I remembered it was a Sonic CinePlayer decoder pack that I installed which gave me the DTS functionality. So I just installed the pack (without the player) and voila, it decodes again.

I'll just point you in the general direction to get you started on how to find that decoder... lemme tell you, it comes in very handy.

Now, as for my most recent problem, which, again has no reference at all in this thread... (grr)... I've used eac3to to happily convert/extract the audio for 5 movies so far... the inconvenience of having to use Audition to downmix the tracks to stereo is nothing compared to the convenience of having it work at all. Well... until most recently. I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode. If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?

edit: It has to be a WAV formatter problem... I mean, if I calculate 48000(hz) * 2 (bytes; 16 bits per sample) * 840 (seconds; 14 minutes) * 6 (channels), I get 483,840,000, or about 461 MB. My dbPowerAmp properties tab shows that the file size should be (actually, it claims it "is") 475.5mb - where back on the "general" tab, it clearly says the file is 4.46gb. What the heck gives?

[madshi]

Originally Posted by Thunderbolt8

does TrueHD conversion of eac3to currently also support files with 96kHz ?

MLP decoding does. So I guess TrueHD decoding does, too.

[madshi]

Originally Posted by MichalHabart

Hello Madshi,
i found strange thing. When i try to convert TrueHD to DTS, it correctz detects bit depth of TrueHD but it still creates all wav files twice, one for 24 bit and the second for 16. Can i somehow prevent this behaviour? Because it requires twice space on HDD and decode is then twice slower.

Well, it doesn't make sense to even check the TrueHD bitdepth when encoding to AC3 or DTS, so I'll remove the bitdepth check in those special cases.

[madshi]

Originally Posted by Falcon4

I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode. If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?

This has been discussed multiple times before. The WAV file header can't really handle files over 4GB. There's no proper way to store a file size of 4GB (or more) in the WAV file header. As a result if the WAV reading program/filter doesn't add special handling for such big files it will only read a part of the whole file. Some programs have special handling for such big WAV files. E.g. eac3to can read WAV files of more than 4GB. Many programs can't do that.

Originally Posted by Falcon4

I mean, if I calculate 48000(hz) * 2 (bytes; 16 bits per sample) * 840 (seconds; 14 minutes) * 6 (channels), I get 483,840,000, or about 461 MB. My dbPowerAmp properties tab shows that the file size should be (actually, it claims it "is") 475.5mb - where back on the "general" tab, it clearly says the file is 4.46gb. What the heck gives?

There's no way a 1.5GB DTS file can be 14 minutes. So your calculation doesn't make any sense to me.

[Falcon4]

Since eac3to is the only program to understand DTS files, dbPowerAmp couldn't possibly have read the file, and hence there's no way I could have been referring to the DTS file in those calculations... while you came to the correct conclusion that a 1.5gb DTS file couldn't possibly be 14 minutes, you oh so happily pinned it on my calculation's fault. I was calculating the size of the raw WAV data based on the returned length (14 minutes) of the WAV file. And it turned out to be 475 MB. Beh...

Anyway, thanks for the info about the WAV files. That explains why I was able to do 5 other movies - they were all under 4gb for the WAV output. I didn't think WAV files would store their length info like that. But hey, Microsoft is Microsoft, no?

And, no, I did searches in this thread on words like "incomplete" or "partial", and both words returned no results. So I can't see how it's been discussed before unless you were speaking a different language. It's sure nice to have such quick replies though... even if a little rude.

I guess I'll try finding a way around this problem since Audition can't import FLACs and I can't split the file into chunks to process in parts... =\

edit: Is it possible to force eac3to to output individual tracks as WAVs like I've seen other outputs in this thread do? I think I'm going to try "-0", "-1", etc to get it to output single tracks while I wait for a reply though... the "-0,1,2,3,4,5" command line option may help here if it allows me to specify less than 6 channels...
editedit: Nope, it doesn't. Ugh. Okay, I'm stuck here... maybe I'll try to find a WAV splitter in the meantime.

[madshi]

Originally Posted by Falcon4

And, no, I did searches in this thread on words like "incomplete" or "partial", and both words returned no results. So I can't see how it's been discussed before unless you were speaking a different language.

Search for "4GB" in this thread and you'll find plenty of posts.

Originally Posted by Falcon4

even if a little rude.

Says the most polite poster in this thread...

Originally Posted by Falcon4

edit: Is it possible to force eac3to to output individual tracks as WAVs like I've seen other outputs in this thread do? I think I'm going to try "-0", "-1", etc to get it to output single tracks while I wait for a reply though... the "-0,1,2,3,4,5" command line option may help here if it allows me to specify less than 6 channels...
editedit: Nope, it doesn't. Ugh. Okay, I'm stuck here... maybe I'll try to find a WAV splitter in the meantime.

eac3to source.dts dest.wavs

[Falcon4]

Originally Posted by madshi

Search for "4GB" in this thread and you'll find plenty of posts.

Couldn't have known to search for that if I didn't even know that was a limitation to begin with
(Maybe it would be a good idea to put a warning in the program itself that almost no programs will be able to read a 4+gb WAV file...?)

Says the most polite poster in this thread...

Yeah, sorry, I'm just a little ticked that the only program in existence to be able to decode a much-in-demand format is so hard to use properly... I mean, seriously, the new generation of high-definition formats are nothing but consumer-unfriendly and there are thousands of people probably dying to do this. Eac3to (and xport.exe) is(/are) the only thing that can manage the audio portion of it. And I'm even having an impossible time using it. But don't get me wrong - not being a desktop programmer, I am thankful that it exists at all.

eac3to source.dts dest.wavs

Ahh -- to the documentation-editor with you then!
edit: That worked great!

C:\Users\Falcon\Desktop\eac3to>eac3to.exe f:\audio.wav c:\audio1.wavs
WAV, 5.1 channels, 2:18:42, 16 bits, 48khz
Reading WAV...
Writing WAVs...
Creating/writing file "c:\audio1.L.wav"...
Creating/writing file "c:\audio1.R.wav"...
Creating/writing file "c:\audio1.C.wav"...
Creating/writing file "c:\audio1.LFE.wav"...
Creating/writing file "c:\audio1.SL.wav"...
Creating/writing file "c:\audio1.SR.wav"...
-----

(and no, that's not Vista.)

edit edit: I really only wanted the L, R, C, and LFE channels (don't need, and won't be encoding/reading, the SL and SR), so maybe an option to simply not decode/write them is possible? At any rate, I seem to now have the problem solved, so thanks!!

[menlvd]

how can it work? using surcode 1.0.29

D:\minep\me_gui\tools\eac3_to>eac3to.exe 12.ac3 test.dts -1536
AC3, 5.1 channels, 0:01:51, 640kbit/s, 48khz
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "test.L.wav"...
Creating/writing file "test.R.wav"...
Creating/writing file "test.C.wav"...
Creating/writing file "test.LFE.wav"...
Creating/writing file "test.SR.wav"...
Creating/writing file "test.SL.wav"...
Surcode sais/asks: "At least one valid source file must be specified to encode."
.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

[tebasuna51]

Originally Posted by Falcon4

... the inconvenience of having to use Audition to downmix the tracks to stereo is nothing compared to the convenience of having it work at all. Well... until most recently. I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode.

There are a limit of 4GB for wav files because a field, in the header, can't support a value greater than 2^32. You have all the data in your wav file but only a few soft can accept them.

If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?

If you are an AviSynth-VirtualDub user I recommend you use some audio dedicated plugins instead the DirectShow-VirtualDub method.

You need an audio decoder like NicAudio.dll (ac3, dts, mp2, mp3, lpcm), or a plugin to manage wav >4GB RaWav.dll, and SoundOut, GUI to encode to flac, ape, mp2, mp3, ac3, ogg, ...

Then you can use an avs script like:

#a = RaWavSource("G:\YourPath\big.wav", 2)
a = NicDtsSource("G:\YourPath\input.dts")
# A 5 -> 2 downmix Dolby ProLogic II compatible
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3254, 0.2301)
bl  = GetChannel(a, 4)
br  = GetChannel(a, 5)
sl  = MixAudio(bl, br, 0.2818, 0.1627)
sr  = MixAudio(bl, br, -0.1627, -0.2818)
blr = MergeChannels(sl, sr)
MixAudio(lrc, blr, 1.0, 1.0)
# if you want (recommended after a downmix):
Normalize(0.95)
# And now the GUI to encode
SoundOut()

You can use NicAudio decoder for dts or let Sonic decoder and use RaWav for the big wav. You can use also other downmix method.

And open the avs in VirtualDub.

[Falcon4]

Holy Jesus, that's impressive. You didn't just come up with that on-the-fly, did you? That's cool.

I don't think I could really do a single-step process like that though, because I'm encoding with a dual-core system that I'm trying to split the audio and video up into. At the moment I process the video in two simultaneous VirtualDubMod instances for two passes (with XviD), process the audio with eac3to and Adobe Audition (which the new "wavs" option will greatly assist since Audition has to re-split and scan the multichannel WAV on opening), manually apply DRC to the center channel to my ears' content (cue "You Bastard!" remarks), save as 192k CBR MP3 (again with the "You Bastard!"), then merge the two parts and mux the single audio file in one single pass (yay). It all actually works rather nicely, albeit taking a long time for the VDubMod MPEG2 parsing of 20+gb video files and whatnot. Although if I time it just right and only have a single video file to work with, I can open (parse) the same file in two VDubMod instances at the same time and they'll just fight over Windows' disk cache to read the file only once. But the result is a pristine quality 4.37gb 1080p AVI file that fits perfectly on a DVD-R.

Maybe I'll write a guide to Blu-Ray to XviD conversion. I imagine that would help a good number of people...

edit: To clean up a little double-talk, I'm saying here that I like using Audition as opposed to a single-step process, where above I said Audition was a pain in the rear - yes, that was true, before a certain few minutes revealed that I could split the DTS decoding into individual WAVs (removing Audition's tedious split-and-rewrite time AND the scan time if I hit "cancel" and zoom in with multitrack view), as well as have Audition apply DRC to make voices more audible with just a few clicks. Yay for extra steps ensuring quality

[tebasuna51]

Originally Posted by Falcon4

edit edit: I really only wanted the L, R, C, and LFE channels (don't need, and won't be encoding/reading, the SL and SR), so maybe an option to simply not decode/write them is possible?

You are free to delete the files but I think is a minority request.

And warning, mix the LFE channel with front channels can destroy info, Dolby don't recommend use the LFE channel in downmix, only if you are sure the info in LFE is not present in front channels.

You still can use the AviSynth method with monowavs to do the downmix more easy than Audition method:

fl = WavSource("c:\audio1.L.wav")
fr = WavSource("c:\audio1.R.wav")
fc = WavSource("c:\audio1.C.wav")
...
and mix and merge like you want.

[shambles]

Originally Posted by shambles

madshi, earlier in the thread you posted that you converted pirates of the caribbean 24bit pcm to flac and that the filesize was bigger than 2gb.. was that dead man's chest or curse of the black pearl?

the reason i'm asking is because i just converted the curse of the black pearl pcm track and the bitrate is 1395 kbps, filesize 1.39gb despite both the input pcm and the output flac track being 24bit. i sort of hope something has gone wrong at my end because having a 16 bit track in a 24 bit file would be amazingly stupid when it's pcm...

sample http://www.sendspace.com/file/4730s7

any comment on this? maybe you could have bit depth checking for pcm tracks also..

[Furiousflea]

Originally Posted by madshi

Please try what I already suggested in my previous reply to you, namely encoding to ac3 to make sure that the problem is the decoding and not the WAVs file writing.


The Sonic decoder doesn't seem to be installed correctly on that different computer.

Hi madshi, sorry I didn't let you know if it was working with AC3 encoding. It doesn't, I get exactly the same results with that (progress bar stops half way through)....hmmm thanks for any help

[The_Keymaker]

Hello Forum,

Based on suggestions from madshi and ACrowley, I will try and update EAC3toGUI to accommodate the newest version of eac3to.

I hope to have something ready early next week.

Regards,
The_Keymaker

[Chumbo]

Originally Posted by The_Keymaker

Hello Forum,

Based on suggestions from madshi and ACrowley, I will try and update EAC3toGUI to accommodate the newest version of eac3to.

I hope to have something ready early next week.

Regards,
The_Keymaker

Thanks for the continued effort.

[DreckSoft]

Fist let me say thanks for the great tool you're providing.

Unfortunately the downconvert option doesn't seem to work properly. I tried to convert a PCM Track (5.1 24bit) to FLAC (20bit - 16 doesn't work either).

eac3to.exe in.pcm out_20.pcm -down20
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Reducing depth from 24 to 20 bits...
Swapping endian...
Remapping channels...
Loading white noise (needed for dithering)...
Creating/writing file "in.pcm"...
The endian swapper crashed.

Btw: Which compression level do you use when encoding flac? I assume it's 5 but I would prefer 8. Could you offer a command line switch for this?

It's strange that
Swapping endian...
Remapping channels...
seems to be done twice.

[nautilus7]

The dts encoding (with surcode) doesn't work. I 've used both 1.0.21 and 1.0.29 but i get the same error message:

Surcode sais/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

I don't type relative paths (like someone above).

Is there anyone that can actually use surcode to encode a dts file?