eac3to - audio conversion tool

[nautilus7]

Don't know. I use the cli. The command i wrote to you, doesn't work?

[mutha88]

Don't know. I use the cli. The command i wrote to you, doesn't work?

Well... how can i use eac3to.exe? When i opened it a MS-DOS window flashes and that's all... how to open it? If i can open it, then i will read and understand the 1st post, but i dunno how to start the exe... im so lame :stupid: :rolleyes:

[nautilus7]

Navigate to the folder that eac3to is located in the "ms-dos" window. It is called cmd (command prompt) actually. Then type eac3to bla bla bla.

[Furiousflea]

...Just tried another DTS Master Audio Lossless track and am getting same results as before (Progress bar stops part way through, but always at same point for each track)

:(

Any ideas anyone what I can do, surely 2 tracks in a row with this happening, there must be something wrong....Just don't have a clue what to try...

:(

[tebasuna51]

Possible. I don't know what to say.

What is the frame length that delaycut reports for every stream? I get 16 ms for eac3 and 32 ms for ac3. I got 5.333 ms for another eac3 track with a 1536 kbps bitrate.

The frame length is function of bitrate and samplerate.

Really the most elemental atom is the Block (256 samples or 5.333 ms if 48 KHz (256/48000)).

With ac3 a frame is always 6 blocks (1536 samples or 32 ms if 48 KHz).

The new eac3 can have 1, 2, 3 or 6 blocks per frame then each eac3 48 KHz frame can be of 5.333, 10.667, 16 or 32 ms.

If you have 5.333 ms each frame at 1536 kbps the FrameLength must be: 1536000*0.005333/8 = 1024 bytes.

[nautilus7]

Yeah, that's right, but don't understand how the 5,3333 ms delay is introduced in my case (640 kbps eac3, 640 kbps ac3).

[Thunderbolt8]

does the eac3to PAL / NTSC slowdown function slow down to 23.976fps or to EXACT 24000/1001 = 23.9760239...fps?

[nautilus7]

Look at the 1st post. It says 24000/1001.

[moshmothma]

eac3to v2.06 released

http://madshi.net/eac3to.zip

* doing FLAC -> FLAC now copies metadata from source to destination file
* MLP files are correctly decoded now (by both Nero and libav/ffmpeg)
* runtime for padded DTS files is shown correctly now

thanks. THe new version allows me to convert my dvda mlp files to wav. The channels are mapped perfectly with some manual tweaking (-0,1,4,5,2,3). Maybe a -dvda switch would be good. Thanks again

[Furiousflea]

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

C:\EAC3To>eac3to recall1.dtshd 1.wavs
DTS Master Audio, 5.1 channels, 16 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
The DirectShow audio decoder didn't accept the input stream.

...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

[mutha88]

Hello it is me again. I have encoded a *.wav DTS file into *.dts file.

The input *.wav is 988 MB. The output *.dts is 972 MB. Is this a well encoded DTS ?! :stupid:

[ACrowley]

Is there a proper way to make a 7.1 -> 6.1 downmix (for DTS-ES 6.1 Discrete encoding)?
(got the encoder, so that's not the problem)

If nobody 'complains' I'm gonna make the Cs [Center Sorround] channel by mixing BL & BR channel.

/edit
6.1 PCM: BL = BR = CS :)

I think a 7.1 dts capable decoder should autom downmix 7.1 into 6.1 by blendig the Channels ?!
Thats the way 6.1 to 5.1 works because those dts Streams are downwards compatible. The BC is blended into BL / BR for 5.1 Output

But Sonic Decoder isnt 7.1 capable and no other Decoder...Sonic maximum is 6.1

So you wont get 8 Mono Channles from Sonic/eac3to
Output from 7.1 dtshd is 5.1

3) DTS decoding fully supports 6.1, but 7.1 tracks are decoded as 5.1 only.

@mutha88
Hä ? No need to reencode a dts wave to dts :) Simply "write" the dts wave to dts lossless (besliced etc) Reencoding in this case is uselsss.

[mutha88]

I think a 7.1 dts capable decoder should autom downmix 7.1 into 6.1 by blendig the Channels ?!

Thats the way 6.1 to 5.1 works because those dts Streams are downwards compatible. The BC is blended into BL / BR for 5.1 Output

But Sonic Decoder isnt 7.1 capable and no other Decoder...Sonic maximum is 6.1

So you wont get 8 Mono Channles from Sonic/eac3to
Output from 7.1 dtshd is 5.1



@mutha88
Hä ? No need to reencode a dts wave to dts :) Simply "write" the dts wave to dts lossless (besliced etc) Reencoding in this case is uselsss.

I have a question. I've extracted the *.wav file from a movie... but this extracted *.wav file doesn't have the Center audio.... just left and right... i can't heard the voices in the *.wav file... how can i fix it? With which program should i exctract the *.wav file from the movie ?! :stupid:

[madshi]

The libav decoder output an unexpected bitdepth.
Fixed in eac3to v2.07.

BTW, the message for ID 20:
"5.1 wrong order channels"
I think must be replaced by a correct remapping
Implemented in eac3to v2.07.

To see if the remap is make when output ac3 I try:
eac3to "God Save The Queen.mlp" god.ac3 -libav -resampleTo48000
MLP, 5.1 wrong order channels, 24 bits, 96khz
Resampling to 48khz...
Encoding AC3...
invalid sample rate
without success.
This worked for me with v2.07. So I guess one of the other fixes also fixed this. Could you please retry, just to be sure?

[madshi]

With the new Version of eac3to i get following error when I want encode one DD+ track into Dts or Ac3

The Format of the Source File Could not be detected.
That means that your E-AC3 file is probably not clean. Probably there are some garbage bytes in front of the real E-AC3 data. Please try fixing the stream with delaycut.

[madshi]

...Just tried another DTS Master Audio Lossless track and am getting same results as before (Progress bar stops part way through, but always at same point for each track)

:(

Any ideas anyone what I can do, surely 2 tracks in a row with this happening, there must be something wrong....Just don't have a clue what to try...
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.

.....:( Just tried running EAC3To on a different computer in the house after installing the sonic audio decoder, now its just saying...
The Sonic decoder doesn't seem to be installed correctly on that different computer.

[madshi]

thanks. THe new version allows me to convert my dvda mlp files to wav. The channels are mapped perfectly with some manual tweaking (-0,1,4,5,2,3). Maybe a -dvda switch would be good. Thanks again
Channel mapping should be automatically corrected by v2.07. If you find that it's not corrected correctly, please let me know.

[madshi]

Please excuse my stupidity, but how can i encode to DTS? I have buyed and installed SurCode, but when I launch "eac3to" i can't see the DTS rate, filters and so one. I can't select them... where can i read a FAQ/guide for this wonderfull tool?
Try the latest eac3to version v2.07. The older version didn't recognize the latest Surcode version while v2.07 should do.

Hello it is me again. I have encoded a *.wav DTS file into *.dts file.
Hmmmmm... I don't think eac3to supports "wav dts" files... :confused:

I have a question. I've extracted the *.wav file from a movie... but this extracted *.wav file doesn't have the Center audio.... just left and right... i can't heard the voices in the *.wav file... how can i fix it? With which program should i exctract the *.wav file from the movie ?!
Depends on the movie format. Is it DVD? Or HD DVD? Or Blu-Ray? Or something else?

[madshi]

i've got a truehd track that's 2.13gb and using nero filters, the flac i encoded from it turned out to be 3.26gb. using -libav, the flac is 1.88gb

i guess it's a 20bit track.. here's a 5mb sample http://rapidshare.com/files/72692697/001.thd.html
Ouch - that's a good sample! Compression ratio of the TrueHD file seems is about 2.87:1. The Nero FLAC has 1.83:1 while the libAV FLAC ends up with 3.17:1. This is the beginning of the movie so compression ratio is probably slightly better here than later in the movie. Overall I agree with you that this is likely to be neither 16bit nor 24bit. I'm not sure if it's 20bit. Maybe. Or maybe 18bit. Anyway, from the FLAC compression ratios it seems to me that libAV decodes it correctly while Nero seems to apply some processing. Maybe dialnorm removal doesn't work for this track for whatever strange reason? I'll have to invest some more time into checking this sample out. But it will probably take some days cause I'm short on time right now...

P.S: Probably the libAV FLAC is alright. So I'd suggest to use that for now. But better keep the TrueHD track until I've found out why Nero fails to work correctly here.

[madshi]

eac3to v2.07 released

http://madshi.net/eac3to.zip

* fixed libAV MLP decoding support
* added automatic MLP ID20 channel remapping
* Surcode 1.0.29 (or newer) home directory detection added

[mutha88]

Try the latest eac3to version v2.07. The older version didn't recognize the latest Surcode version while v2.07 should do.


Hmmmmm... I don't think eac3to supports "wav dts" files... :confused:


Depends on the movie format. Is it DVD? Or HD DVD? Or Blu-Ray? Or something else?

It is a HD-DVD DTS movie... KMPlayer shows me that the encoded *.dts WAV file is DTS-HD. :stupid:

[nautilus7]

Can you check the ac3 delay issues i have please? I have uploaded a sample.

Thanks for the update!

[madshi]

Can you check the ac3 delay issues i have please? I have uploaded a sample.
As tebasuna51 has already explained, every Aften encoding adds a 5.333ms delay - unless you switch Aften to a special (non recommended) mode to avoid this delay. This special mode is currently not supported by eac3to. I think the Dolby encoders behave similarly to Aften, at least I've been told so.

Are you really worried about 5.333ms of delay? Some people don't notice a delay of 100ms. Personally, I believe that I notice incorrect delays as low as 40ms, but I'm not totally sure. I'm very sure that I wouldn't notice a delay of 5.333ms, even if my life depended on it.

[tebasuna51]

Fixed in eac3to v2.07.
...
Implemented in eac3to v2.07.
...
This worked for me with v2.07. So I guess one of the other fixes also fixed this. Could you please retry, just to be sure?

Yes, all work fine now with v2.07.

Thanks.

[nautilus7]

As tebasuna51 has already explained, every Aften encoding adds a 5.333ms delay - unless you switch Aften to a special (non recommended) mode to avoid this delay. This special mode is currently not supported by eac3to. I think the Dolby encoders behave similarly to Aften, at least I've been told so.

Are you really worried about 5.333ms of delay? Some people don't notice a delay of 100ms. Personally, I believe that I notice incorrect delays as low as 40ms, but I'm not totally sure. I'm very sure that I wouldn't notice a delay of 5.333ms, even if my life depended on it.
OK, but this didn't happen to another eac3 to ac3 encode i did a few days ago.

[madshi]

Yes, all work fine now with v2.07.
Thanks for confirming.

[madshi]

OK, but this didn't happen to another eac3 to ac3 encode i did a few days ago.
It must have happened to that other eac3 to ac3 encode, too... :D

[nautilus7]

Are you trying to drive me nuts? :eek:
It didn't happen. And it's not 5,33 ms delay. It's 10,5 ms.

Oh, man... I 'll have to check the tracks again. :mad:

[Thunderbolt8]

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

[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?

[Chumbo]

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. :p

[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]

.....:( 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 (http://hostfile.org/viewalbum.php?id=209) 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]

does TrueHD conversion of eac3to currently also support files with 96kHz ?
MLP decoding does. So I guess TrueHD decoding does, too.

[madshi]

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]

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.

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]

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.

even if a little rude.
Says the most polite poster in this thread... :p

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]

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... :p
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]

... 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 (http://avisynth2.sourceforge.net/NicAudio_20070821.zip) (ac3, dts, mp2, mp3, lpcm), or a plugin to manage wav >4GB RaWav.dll (http://www.mytempdir.com/2070490), and SoundOut (http://forum.doom9.org/showthread.php?t=120025), 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]

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]

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]

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]

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?