EVOB De/Multiplexers

[Rectal Prolapse]

Note: SoX write WAVe, which have broken header (if it's over 4GB), only Wavewizard (maybe) can read it properly...

Could this explain why, when I mux the file made by SoX into an MKV, I lose sound halfway through a 2 hour movie? Could it be that the muxer stops at the 4 gb limit and ignores the rest of the WAV file?

Hmmm. Is there a way to fix the header? I can use a hex editor if needed.

[MichalHabart]

Could this explain why, when I mux the file made by SoX into an MKV, I lose sound halfway through a 2 hour movie? Could it be that the muxer stops at the 4 gb limit and ignores the rest of the WAV file?

Hmmm. Is there a way to fix the header? I can use a hex editor if needed.

And why do you want to mux wav file into mkv? Try to create something useful from wav, ie dts or at least ac3. you will spend lot of space

[Rectal Prolapse]

It would save an encoding step - and disc space isn't an issue as I can delete the MKV after I'm done watching. Also, audio quality would be higher. Besides, remuxing already chews up tons of disc space already - what difference would it make in the end?

And what if I just wanted to waste the disc space? Nothing wrong with that - disc space is CHEAP. I guess my attitude is at odds with most of the doom9 members - it seems everyone here wants to encode their favorite 3 hour 1080p lossless audio movie onto a single layer DVD-R: http://forum.doom9.org/showthread.php?p=964128#post964128 . ;) :p (for the humor-impaired that was a joke!)

My reasoning is this: I can't play EVO's natively and reliably right now with my favorite players - but I think in the future this won't be a problem, so deleting the MKVs is no loss as long as you have the originals...you can always recreate them later when the tools improve.

[madshi]

Hey guys,

I've written a little tool named "eac3toac3", which is able to convert E-AC3 files to AC3. **WAIT**. Before you rejoice: The tool really does nothing but what was already suggested in this thread (E-AC3 -> RAW -> WAV -> AC3). Basically it automates what you have to do with GraphEdit and running all the various tools like sox and aften etc. Anyway, the tool works fine for me and makes the conversion process a bit more comfortable.

http://madshi.net/eac3toac3.zip

eac3toac3 v1.0, freeware by madshi.net

Usage: eac3toac3 srcFilename.eac3 destFilename.ac3

This tool can convert a 2.0 or 5.1 channel E-AC3 file to AC3.

For this to work correctly you need these filters to be installed:
(1) DTS/AC3/DD+ Source
(2) Sonic Audio Decoder 4.2
(3) ffdShow Audio Processor
(4) Dump filter

Furthermore these tools must be located in the same folder as eac3toac3:
(1) sox 13
(2) aften revision 449
Please note that newer ffdShow tryout versions make problems for me. I have things working correctly with the ffdShow tryouts build from December 2006. Here are some download links for the neccesary freeware tools/filters:

http://ffdshow-tryout.sourceforge.net/
http://sourceforge.net/project/showfiles.php?group_id=10706&package_id=10619&release_id=485785
http://win32builds.sourceforge.net/aften/index.html

The "eac3toac3" tool has a few tricks up it's sleeve:

(1) It automatically detects whether the E-AC3 file is really an E-AC3 file, how many channels it has and which sampling rate etc.
(2) It automatically changes the Sonic Audio Decoder settings to deliver the needed channels.
(3) It automatically finds out whether the intermediate raw audio file has a bitdepth of 16 or 24 bits and adjusts the "sox" parameters accordingly.

If you don't want to convert to AC3, but to something else, you can use ".wav" or ".raw" as destination extensions. The eac3toac3 tool will then simply stop when the ".raw" respectively ".wav" file is done. Intermediate files are automatically deleted. Only the source and destination file are left on the harddisk after the tool has run through.

[JeffAlso]

I've written a little tool named "eac3toac3"
Excellent!! Thank you very much!!

[Rectal Prolapse]

Wow nice madshi! What problems with FFDShow were you having?

Also, do we need to do configure FFDShow Audio Processor in any way?

[Chumbo]

@madshi,
Just curious, have you tried using AC3Filter instead of ffdshow in the chain? I use AC3Filter when using a graph and it's perfect.

[madshi]

What problems with FFDShow were you having?
See here:

http://forum.doom9.org/showpost.php?p=967601&postcount=626

Also, do we need to do configure FFDShow Audio Processor in any way?
I don't remember to having done any configuration. Just try without. If you run into problems, let me know.

Just curious, have you tried using AC3Filter instead of ffdshow in the chain? I use AC3Filter when using a graph and it's perfect.
Do you mean "DTS/AC3/DD+ Source -> Sonic Audio Decoder 4.2 -> AC3Filter -> Dump"? No, I've not tried that yet. Is there a specific reason why you prefer AC3Filter over ffdshow?

[Deckard2019]

I don't remember to having done any configuration. Just try without. If you run into problems, let me know.
You don't use ffdshow to directly output AC3 ? So why do you use it ?
DTS/AC3/DD+ Source -> Sonic Cinemaster Audio Decoder 4.2 -> WAV dest -> FileWriter will create RAW file.

[MichalHabart]

You don't use ffdshow to directly output AC3 ? So why do you use it ?
DTS/AC3/DD+ Source -> Sonic Cinemaster Audio Decoder 4.2 -> WAV dest -> FileWriter will create RAW file.

No, this will never work!!!

You must use DTS/AC3/DD+ Source -> Sonic Cinemaster Audio Decoder 4.2 -> Dump

FIleWriter or WAV Dest has bug inside which will create unusable wav file if it is bigger then 4GB

[Deckard2019]

No, this will never work!!!
Ok. Wrong copy/paste (I don't use this method as I use BeHappy to bypass RAW/WAV file creation).
But what about ffdshow ? I still don't understand why/when it's used in eac3toac3 ...

[madshi]

But what about ffdshow ? I still don't understand why/when it's used in eac3toac3 ...
The reason is that the Sonic Audio Decoder is very unreliable without adding ffdshow to the processing stages. I found that the Sonic Audio Decoder sometimes outputs 16 bit, sometimes 24 bit. Furthermore more often then not only 2 channels contained valid information while the remaining 4 channels were garbage. This all doesn't happen with ffdshow in the chain. Don't ask me why. I don't really know.

[Deckard2019]

The reason is that the Sonic Audio Decoder is very unreliable without adding ffdshow to the processing stages. I found that the Sonic Audio Decoder sometimes outputs 16 bit, sometimes 24 bit. Furthermore more often then not only 2 channels contained valid information while the remaining 4 channels were garbage. This all doesn't happen with ffdshow in the chain. Don't ask me why. I don't really know.
Ok. Thank you. The 4 crappy channels thing happened to a friend of mine.
With same source, same tools, it doesn't happen to me ...

Outputting directly AC3 from ffdshow is still considered as a wrong method ?
(For now, I use BeHappy/Aften with .grf embedded in avs script. grf only contains orbitlee filter and sonic one).

[MichalHabart]

Ok. Thank you. The 4 crappy channels thing happened to a friend of mine.
With same source, same tools, it doesn't happen to me ...

Outputting directly AC3 from ffdshow is still considered as a wrong method ?
(For now, I use BeHappy/Aften with .grf embedded in avs script. grf only contains orbitlee filter and sonic one).

Yes, ffdshow can be used only for changing bit depth of output. AC3 created directly by ffdshow has wrong channel order

[Jack-Bauer]

AC3 created directly by ffdshow has wrong channel order

Matrix filter can be inserted in beetween sonic and ffdshow, allowing to set its matrix as desired.

Jack

[zgx]

Well, LPCM does not seem to work at the moment. With xport or graphedit i get raw file with same size but when i use sox to create wav, i receive 77 minutes instead of 100 and the sound inside is horrible (sounds like when you call to fax machine :) )

Does anyone manage to get it work?I did get it to work with Graphedit. I selected the m2ts file with File Source (Async) and selected the LPCM track in Sonic HD Demuxer.

File Source (Async) -> Sonic HD Demuxer -> Sonic Cinemaster Audio Decoder 4.2 -> Dump

Then I converted the raw file to wav and got a working file:
sox -r48000 -t .raw -c 6 -3 -s file.raw file.wav


I'm now trying out the new version of xport (since I don't trust Sonic Audio Decoder). It finds a LPCM track and dumps it:

LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000

But don't know what to do with the dump. Have tried all sorts of options in Sox and BeSweet but I can't convert it to a working wav.

[MichalHabart]

I did get it to work with Graphedit. I selected the m2ts file with File Source (Async) and selected the LPCM track in Sonic HD Demuxer.

File Source (Async) -> Sonic HD Demuxer -> Sonic Cinemaster Audio Decoder 4.2 -> Dump

Then I converted the raw file to wav and got a working file:
sox -r48000 -t .raw -c 6 -3 -s file.raw file.wav


I'm now trying out the new version of xport (since I don't trust Sonic Audio Decoder). It finds a LPCM track and dumps it:

LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000

But don't know what to do with the dump. Have tried all sorts of options in Sox and BeSweet but I can't convert it to a working wav.

Well, that is exactly the problem i have. but it seems that raw dump is not working, i should get 5GB file but i get only 3,5GB. So thatswhy my soxed wav is not going to work. have you tried to load it to wavewizard? it should be able to tell you length and channels in that wav file.

[drmpeg]

The LPCM output of xport is big endian (just like it's stored in the .m2ts file).

This sequence works:

xport -h2 movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 2 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode

This sequence fails:

xport -h movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode ( -6ch )

Error 58: Error : Unknown Input-File Format : "bits.wav".

Seems to me that sox can't create a 6 channel .wav file that besweet understands.

Ron

[madshi]

Outputting directly AC3 from ffdshow is still considered as a wrong method ?
I don't know. I've read in this thread that it gives inferior results. And I read that Aften is considered to be a good AC3 encoder, so that's why I did it this way.

[madshi]

Seems to me that sox can't create a 6 channel .wav file that besweet understands.
Aften revision 449 is happy with sox created 6 channel wav files, as long as you use the Aften "-readtoeof 1" parameter.

[MichalHabart]

The LPCM output of xport is big endian (just like it's stored in the .m2ts file).

This sequence works:

xport -h2 movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 2 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode

This sequence fails:

xport -h movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode ( -6ch )

Error 58: Error : Unknown Input-File Format : "bits.wav".

Seems to me that sox can't create a 6 channel .wav file that besweet understands.

Ron

And have you tried to open that multiwav in something else, for example in wavewizard?
I was succesfull in this. probably just besweet has these problems. I was able to demux 6 channel lpcm to raw from Terminator 1 Bluray and converted it to wav and now i made DTS track from it. It sounds wonderful :)

[The1n]

Do i have to change the channelmapping if im just going to convert eac3 to aac in megui, im using graphedit and loads the graph with avisynth script.
When i play the track in graphedit all channels sound ok, but when i encode it to aac multichannel suddenly the center is gone.

The1n

[zgx]

Perfect lossless LPCM 5.1 to FLAC conversion
It's really easy but has taken me quite a few hours to get it all right.

Programs needed:
xport by drmpeg @doom9
http://www.w6rz.net/xport.zip

sox v13.0.0
http://sourceforge.net/projects/sox

Wavewizard 0.45b
http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip

MediaCoder 0.5.1
http://mediacoder.sourceforge.net/download.htm

flac 1.1.4
http://flac.sourceforge.net/download.html


Step 1 - Extract the LPCM stream from Blu-ray
xport -h movie.m2ts 1 3 2

Explaination:
"1 3 2" selects program 1, video stream 3 and audio stream 2.
Since video stream 3 does not exist you will only demux audio (change to 1 if you want video as well).
The LPCM stream should most often be number 2.


Step 2 - Map channels
sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa file.wav

Start Wavewizard and enter channel mapping (F2).
- Create a new mapping "0 1 2 5 3 4".
- Select your new mapping.
- Click "Enable channelmapping"
- Click "Convert" and create a new file.

Explaination:
The channels in the LPCM file are mapped "L,R,C,BL,BR,LFE" as opposed to "L,R,C,LFE,BL,BR". This is corrected with Wavewizard but in order for it to accept the file you need to use sox first.


Step 3 - Encode your FLAC
- Install MediaCoder 0.5.1. Make sure you replace "flac.exe" under the "codecs" directory so you get version 1.1.4.
- Drag and drop your file(1).wav to the MediaCoder window.
- Under Audio select "FLAC" as Encoder and press F5 to start the encode.

Explanation:
FLAC doesn't like 4+ GB wave files but somehow it works with MediaCoder.


Step 4 - Enjoy your FLAC
Enjoy your new FLAC!

May I sugest using mkvmerge (http://www.bunkus.org/videotools/mkvtoolnix/) to mux your FLAC with your video stream to a Matroska container.

[moshmothma]

Hmm, I get to the wavewizard part and get the following error when I try to convert:

Error on: "file.wav". This file is not recognized by wavewizard.

[zgx]

Hmm, I get to the wavewizard part and get the following error when I try to convert:

Error on: "file.wav". This file is not recognized by wavewizard.Sorry made a typo. Should of course be "sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa file.wav" and not "-3".

[LRN]

I can't beleive they're using LPCM...They must be completely dazzed by BluRay disc sizes! There's NO reason to but more than 500 kilobits on one channel, and if there IS, they could have used FLAC, WAVPack, damn MLP or something! But not the LPCM...it's so DULL.

P.S. How about fullscale "X.1 E-AC3 -> 5.1 AAC" guide?

P.P.S. Where i could get Sonic 4.2 decoder? 4.1 doesn't works in shown graphs :(

[Rectal Prolapse]

Wow zgx - just what I was looking for! Hopefully there won't be any issues with FLAC audio decoding past the 4GB mark - when I use WAV I get silence once that barrier is surpassed.

[madshi]

Perfect lossless LPCM 5.1 to FLAC conversion
Just for my interest: How big is the FLAC compressed audio track file compared to the original LPCM one? Thanks!

[LRN]

20-30%, iirc

[zgx]

Just for my interest: How big is the FLAC compressed audio track file compared to the original LPCM one? Thanks!About 25% of the original. For "16 bit/48 kHz/6 channel" movie soundtracks I have gotten files from 1100 to 1900 Kbps so it doesn't take more space then a 1.5 Mbps DTS track does. The only downside is that you need 6 analog RCA cables or HDMI in order to get lossless sound to your transciever. You can of course let AC3filter transcode to AC3 for S/PDIF ouput. Then it won't be lossless but will still sound really good.

[drmpeg]

Perfect lossless LPCM 5.1 to FLAC conversion

Excellent! Although 6-channel (3/2+lfe) is the most common format on current discs, here's the channel mapping for all the LPCM formats:

mono M1 X
stereo L R
3/0 L R C X
2/1 L R S X
3/1 L R C S
2/2 L R LS RS
3/2 L R C LS RS X
3/2+lfe L R C LS RS lfe
3/4 L R C LS Rls Rrs RS X
3/4+lfe L R C LS Rls Rrs RS lfe

X = zero samples, Rls = Rear left surround, Rrs= Rear right surround

Ron

[Rectal Prolapse]

Strange - when I play back a WAV in MediaCoder, the center speaker track is sent to the Rear Left speaker! I hope this doesn't screw up encoded audio either.

Anyways - I took a DD+ track from an HD-DVD, passed it to Sonic Audio Decoder 4.2 and Dump. Then I took the resulting WAV file and ran it through Sox, then I ran WaveWizard with the channels remapped as suggested. I hope this is the right procedure for HD-DVD DD+ files?

Anyways, I will be doing some tests tonight...

EDIT: Remapping the channels appears to be a BAD idea! There is no need to remap with WaveWizard it appears, in my testing.

[Rectal Prolapse]

Hmmm I think I've run into a snag - if the WAVs are 24 bit and encoded into FLAC, the filters I have seem to choke on it. Maybe this only works if you downconvert to 16 bit first?

[Chumbo]

Do you mean "DTS/AC3/DD+ Source -> Sonic Audio Decoder 4.2 -> AC3Filter -> Dump"? No, I've not tried that yet. Is there a specific reason why you prefer AC3Filter over ffdshow?
Yep, exactly. I've always just used it because it's been solid and I've had no issues. It's also real easy to configure on the fly. :) I mentioned it because you said you had a problem with a specific version of ffdshow right? I figured AC3Filter may be a good alternative.

[Rectal Prolapse]

I can play back FLAC files in GraphEdit now. I had to install the oggcodecs. More details here:

http://forum.inmatrix.com/index.php?showtopic=4959

[Rectal Prolapse]

Okay, I think I got FLAC working with HD-DVDs' DD+ soundtracks. Again, many thanks zgx for the guide. :)

1) Use EVOdemux.exe to demux the desired DD+ audio track.
2) Use EVOdemux.exe to rebuild an EVO that ONLY contains the desired video stream.
3) Use the previously posted instructions to create an MKV file, using GraphEdit, Haali Media Splitter, and Haali Muxer.
4) Use Sonic Audio Decoder 4.2 in GraphEdit to Dump a (huge) .raw file onto your hard drive (look at previous posts on how to do this).
5) Use Sox to convert this into a .WAV file (again, look at previous posts - just make sure to use the -3 option). With DD+ this should result in a 24 bit multichannel WAV file.
6) As per zgx's instructions, convert this to flac with MediaCoder (so far as I can tell, you do NOT need WaveWizard. When I used WaveWizard to remap, it seemed that the center channel got remapped to the left rear speaker! I hope someone can confirm this for me.)
7) Use MKVMerge GUI to mux in the flac stream into a new MKV.

Now, surprisingly, I did not need to figure out the audio delay. I had to do this before, when I was encoding to AC3, but not this time! I don't know why - maybe the encoder I used ignores silence? Anyways, this is a good thing - makes things simpler!

If you use FFDShow Audio Decoder, make sure that FLAC is enabled in the codecs settings.

Enjoy!

For reference, I used the Harry Potter Goblet of Fire HD-DVD. The re-encoded 640kbps DD5.1 soundtrack was ~730MB, which is exactly the same as the DD+ soundtrack. The FLAC lossless soundtrack takes up 1300 MB of space. Pretty good!

Naturally, the FLAC/DD+ soundtrack sounds far better than the AC3 one - because I used FFDShow to create the AC3 one, which isn't very good - the volume was too low!

[Rectal Prolapse]

Hmm there appears to be a problem with massive stuttering and audio dropouts. I'll blame MKVMerge for this one. :(

[Rectal Prolapse]

More caveats: It appears that the delay setting in mkvmerge (2.0.2) does not work for FLAC audio tracks.

[zgx]

@Rectal Prolapse:
Just as you have notcied you don't need to remap the channels for a Sonic Audio Decoder dump. That is just something needed with a LPCM "xport file".

You can actually decode Blu-ray LPCM with the Sonic Audio Decoder but I don't trust that filter completely so I rather go with xport even if it takes a little longer.

Sorry to hear about stuttering and audio dropouts. I did some tries with FLAC about a week ago when I muxed the MPEG2 streams from Blu-ray together with FLAC audio and I did not experience any stuttering or dropouts. Not 100% sure if I used mkvmerge or gdsmux when I created the files.

About the delay setting i mkvmerge please let me know what you find out. Could you perhaps also try if it works better with Wavpack instead of FLAC?


When it comes to DD+ there are some movies that feature very high bitrate soundtracks. I played around with such a movie yesterday. It featured a ~2 Mbps DD+ soundtrack. The output I got from Sonic Audio Decoder was 24 bit/48 kHz. I put AC3filter in the chain to downsample to 16 bit and then made a FLAC. Out of the result I got a nice sounding 1100 Kbps FLAC. The question is if the DD+ soundtrack was 24 bit (and I just got a downsampled FLAC) or if that was just something Sonic Audio Decoder decided to output? What's the easiest way to check the bit depth of an EAC3 file?

[MichalHabart]

Strange - when I play back a WAV in MediaCoder, the center speaker track is sent to the Rear Left speaker! I hope this doesn't screw up encoded audio either.

Anyways - I took a DD+ track from an HD-DVD, passed it to Sonic Audio Decoder 4.2 and Dump. Then I took the resulting WAV file and ran it through Sox, then I ran WaveWizard with the channels remapped as suggested. I hope this is the right procedure for HD-DVD DD+ files?

Anyways, I will be doing some tests tonight...

EDIT: Remapping the channels appears to be a BAD idea! There is no need to remap with WaveWizard it appears, in my testing.

Of course, remapping should be used only for LPC? track, definitely not for DD+

[Rectal Prolapse]

Thanks guys - I just realized LPCM is different. :)

gdsmux? I might give that a try.

I haven't confirmed yet if it is a problem with FFDShow's FLAC decoding. I put a FLAC file into a .mka container and it had the same audio glitch issues as when it was in a .mkv with a VC-1 video stream. After more testing I realized that there was no video stutter - only the audio glitches.

I am using FFDShow-tryouts revision 964 (February 2007). What are you using for FLAC playback, zgx (and anyone else who got this working)?

Thanks again!

(For the time being I used MediaCoder to make me an AAC track using the FAAC encoder, just for fun! AAC Lossless doesn't work - I get a 4K file everytime! And WMA Lossless in MediaCoder crashes out on me.)

[foxyshadis]

How does flac compare to MLP, if you convert directly without changing the bit depth? Just curious whether it's a worthwhile savings. (The ease of playability might outweigh that anyway.)

[KoD]

Rectal Prolapse, does your soundcard support 24 bits input in hardware ? Because if it's only accepting 16 bits, then there's a downsampling being performed by the drivers or by windows without you knowing. And soundcards can choke on high bitrate, even more when it's multichanneled. I'm afraid you might be asking more from your hardware than it can do and you're blaming software for it.

[Darth Pinous]

More caveats: It appears that the delay setting in mkvmerge (2.0.2) does not work for FLAC audio tracks.

You should try WavPack instead of FLAC. You can delay a WavPack soundtrack in mkvmerge.

[MichalHabart]

Hey guys,

I've written a little tool named "eac3toac3", which is able to convert E-AC3 files to AC3. **WAIT**. Before you rejoice: The tool really does nothing but what was already suggested in this thread (E-AC3 -> RAW -> WAV -> AC3). Basically it automates what you have to do with GraphEdit and running all the various tools like sox and aften etc. Anyway, the tool works fine for me and makes the conversion process a bit more comfortable.

http://madshi.net/eac3toac3.zip

eac3toac3 v1.0, freeware by madshi.net

Usage: eac3toac3 srcFilename.eac3 destFilename.ac3

This tool can convert a 2.0 or 5.1 channel E-AC3 file to AC3.

For this to work correctly you need these filters to be installed:
(1) DTS/AC3/DD+ Source
(2) Sonic Audio Decoder 4.2
(3) ffdShow Audio Processor
(4) Dump filter

Furthermore these tools must be located in the same folder as eac3toac3:
(1) sox 13
(2) aften revision 449
Please note that newer ffdShow tryout versions make problems for me. I have things working correctly with the ffdShow tryouts build from December 2006. Here are some download links for the neccesary freeware tools/filters:

http://ffdshow-tryout.sourceforge.net/
http://sourceforge.net/project/showfiles.php?group_id=10706&package_id=10619&release_id=485785
http://win32builds.sourceforge.net/aften/index.html

The "eac3toac3" tool has a few tricks up it's sleeve:

(1) It automatically detects whether the E-AC3 file is really an E-AC3 file, how many channels it has and which sampling rate etc.
(2) It automatically changes the Sonic Audio Decoder settings to deliver the needed channels.
(3) It automatically finds out whether the intermediate raw audio file has a bitdepth of 16 or 24 bits and adjusts the "sox" parameters accordingly.

If you don't want to convert to AC3, but to something else, you can use ".wav" or ".raw" as destination extensions. The eac3toac3 tool will then simply stop when the ".raw" respectively ".wav" file is done. Intermediate files are automatically deleted. Only the source and destination file are left on the harddisk after the tool has run through.

Hm, it is not working. I tried with Casino DD+ track this is what is got:

eac3toac3.exe h:\casino.dd+ g:\test.ac3
The file extension of the source file must be ".eac3", ".dd+" or ".wav".

When i rename dd+ track to eac3, it is recognised by that program :(
Can you tell me what i am doing wrong?

[orbitlee]

Hm, it is not working. I tried with Casino DD+ track this is what is got:

eac3toac3.exe h:\casino.dd+ g:\test.ac3
The file extension of the source file must be ".eac3", ".dd+" or ".wav".

When i rename dd+ track to eac3, it is recognised by that program :(
Can you tell me what i am doing wrong?


Under DOS or Windows, you can't name file with '+' inside. It is reserved by command processor(cmd.exe).
BTW, I noticed that in DD+ audio files have .ddp extension name in Scenarist SCA, .ec3 in Scenarist ACA.

[MichalHabart]

Under DOS or Windows, you can't name file with '+' inside. It is reserved by command processor(cmd.exe).
BTW, I noticed that in DD+ audio files have .ddp extension name in Scenarist SCA, .ec3 in Scenarist ACA.

So, madshi, is it possible to change aec3toAC3 program so it will take ddp instead of dd+ files?
And something more, what bitrate will your final ac3 have? Is there some possibility to add there parameter for final bitrate of ac3 (i mean 448 or 640)?

[madshi]

So, madshi, is it possible to change aec3toAC3 program so it will take ddp instead of dd+ files?
Yes, of course, I'll do that. I'll also try to use Chumbo's suggestion to use the AC3Filter instead of ffdShow.

And something more, what bitrate will your final ac3 have? Is there some possibility to add there parameter for final bitrate of ac3 (i mean 448 or 640)?
I'm always using 640 for 5.1 audio tracks. For 2.0 audio tracks I'm using 384, unless the E-AC3 track has a higher rate than that. In that case I'm using 640, too. I guess I can add a parameter, but is there anybody who would want to use 448, anyway? I mean we want best possible quality, don't we?

[madshi]

About 25% of the original. For "16 bit/48 kHz/6 channel" movie soundtracks I have gotten files from 1100 to 1900 Kbps so it doesn't take more space then a 1.5 Mbps DTS track does. The only downside is that you need 6 analog RCA cables or HDMI in order to get lossless sound to your transciever. You can of course let AC3filter transcode to AC3 for S/PDIF ouput. Then it won't be lossless but will still sound really good.
That's mighty cool! How does that compare to TrueHD or DTS Master HD Audio? Does FLAC compare well with those two codecs? I mean in that case FLAC would really be a great alternative, as many players already support it, even some hardware media streamers, while TrueHD and DTS Master HD Audio support is still very rare both in software and hardware...

[zgx]

That's mighty cool! How does that compare to TrueHD or DTS Master HD Audio? Does FLAC compare well with those two codecs? I mean in that case FLAC would really be a great alternative, as many players already support it, even some hardware media streamers, while TrueHD and DTS Master HD Audio support is still very rare both in software and hardware...FLAC, Wavpack, LPCM, MLP, TrueHD and DTS-HD Master Audio should all sound exactly the same since they are all lossless.

Any difference in "quality" has to do with the source that was used for the encode or because of differences in software/hardware used in playback.

DTS-HD Master Audio has a small plus beeing able to output a 1.5 Mbps DTS core stream over S/PDIF while all the other formats needs to re-encode to example AC3 in realtime for S/PDIF usage.

But I think that on a HTPC there are many advantages in using FLAC or Wavpack.