eac3to - audio conversion tool

[madshi]

http://madshi.net/eac3to.zip

New versions here https://www.rationalqm.us/board/viewforum.php?f=18

eac3to v3.36, freeware by madshi.net

- can show information about audio, video, VOB/EVO/(M2)TS and MKV files
- can decode and encode various audio formats
- can remove dialog normalization from AC3, E-AC3, DTS and TrueHD tracks
- can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks
- can extract TrueHD stream from Blu-Ray TrueHD/AC3 tracks
- can extract DTS core from DTS-HD tracks
- can remove DTS zero padding and repair outdated DTS-ES headers
- can apply positive or negative audio delays
- can reduce bitdepth of decoded audio data by using TPDF dithering
- can resample decoded audio data (using SSRC or r8brain)
- can apply/reverse PAL speedup on decoded audio data (SSRC/r8brain)
- can demux video / audio tracks of EVO/VOB/(M2)TS and MKV sources
- can list available titles of Blu-Ray and HD DVD discs
- can extract Blu-Ray and HD DVD chapter information and subtitles
- can mux MPEG2, VC-1 and h264 video tracks to Matroska
- can remove pulldown flags from MPEG2, VC-1 and h264 video tracks

eac3to sourcefile[+sourcefile2] [trackno:] [destfile|stdout] [-options]

Examples:
eac3to source.pcm destination.flac
eac3to source.thd destination.flac destination.ac3
eac3to blurayMovieFolder movie.mkv
eac3to source.evo 1: chapters.txt 2: video.mkv 3: audio.flac 5: subtitle.sup
eac3to feature_1.evo+feature_2.evo movie.mkv

Options:
-448 use e.g. "192", "448" or "640" kbps for AC3 encoding
-768 use "768" or "1536" kbps for DTS encoding
-core extract the DTS core of a DTS-HD track
+/-100ms apply a positive or negative audio delay
+/-3dB apply a positive or negative audio gain (volume change)
-0,1,2,3,4,5 remap the channels to the specified order
-edit=0:00:00,0ms loops or removes audio data at the specified runtime
-silence/-loop forces usage of silence (or looping) for audio edits
-down6 downmix 7 or 8 channels to 6 channels
-downDpl downmix multi channel audio to Dolby Pro Logic II
-downStereo downmix multi channel audio to simple Stereo
-phaseShift shift phase (when doing stereo downmixing, see "down2")
-mixlfe mix LFE in (when doing stereo downmixing, see "down2")
-down16 downconvert decoded audio data to 14..23 bit
-slowdown convert 25.000 and 24.000 content to 23.976 fps
-speedup convert 23.976 and 24.000 content to 25.000 fps
-23.976/... define source fps to be "23.976", "24.000", "25.000", ...
-changeTo24.000 change source fps to "23.976", "24.000", "25.000", ...
-resampleTo48000 resample audio to "44100", "48000" or "96000" Hz
-r8brain use r8brain resampler instead of SSRC
-quality=0.50 Nero AAC encoding quality (0.00 = lowest; 1.00 = highest)
-8 define PCM file to be "1".."8" channels
-16 define PCM file to be "16" or "24" bit
-little define PCM file to be "little" or "big" endian
-96000 define PCM file to be 44100, 48000, 96000 or 192000 Hz
-override forcefully overrides PCM auto detection with manual values
-sonic/nero/... force the use of a specific decoder (not recommended)
-keepDialnorm disables dialog normalization removal (not recommended)
-decodeHdcd decodes HDCD source track (usually 16 -> 20 bit)
-demux demuxes 1st video track and all audio and subtitle tracks
-stripPulldown strips the pulldown from MPEG2 video tracks
-keepPulldown disable removal of pulldown for MPEG2, h264 and VC-1 tracks
-seekToIFrames make all h264/AVC "I" frames seekable
-check checks if the source EVO/(M2)TS file is clean.
-test checks if the external filters are installed & working
-lowPriority moves processing to background/idle priority
-shutdown automatically shutdown the PC after processing is done

Supported source formats:
(1) RAW, (L)PCM
(2) WAV (PCM, DTS and AC3), W64, RF64
(3) AC3, E-AC3
(4) DTS, DTS-ES, DTS-96/24, DTS-HD Hi-Res, DTS-HD Master Audio
(5) MP1, MP2, MP3 audio
(6) AAC audio
(7) MLP, TrueHD, TrueHD/AC3
(8) FLAC
(9) EVO/VOB/(M2)TS and MKV

Decoded audio data can be stored as / encoded to:
(1) RAW, (L)PCM
(2) WAV (PCM only), W64, RF64, AGM
(3) WAVs (multiple mono WAV files, PCM only)
(4) AC3
(5) DTS
(6) AAC
(7) FLAC

For best AAC decoding you need:
(1) Nero 7 (Nero 8 won't work!)

For DTS encoding you need:
(1) SurCode DVD DTS - version 1.0.21 or newer

For AAC encoding you need:
(1) Nero AAC Encoder

For video muxing you need:
(1) Haali Matroska Muxer

wiki:

Not written by me, not finished yet, but definitely worth a look:

http://en.wikibooks.org/wiki/Eac3to

bug tracker:

http://eac3to.bugs.madshi.net

GUIs:

If you prefer a graphical user interface, here are three choices (not written by me):

Eac3to and More GUI:
http://forum.doom9.org/showthread.php?t=135095

HD DVD/Blu-Ray Stream Extractor:
http://forum.doom9.org/showthread.php?t=141829

Clown BD:
https://www.videohelp.com/software/Clown-BD

eac3to µGUI:
Link don't work, GUI obsolete[www homecinema-hd com/index.php?page=eac3tougui&lang=en_US]

links:

arcsoft: http://www.arcsoft.com
dcadec: https://github.com/foo86/dcadec [Recommended replace the libdcadec.dll (https://forum.doom9.org/showthread.php?p=1953397#post1953397) from the eac3to package with the updated v2.0 in the link]
nero: http://www.nero.com
NeroAacEnc: https://web.archive.org/web/20160923100008/http://ftp6.nero.com/tools/NeroAACCodec-1.5.1.zip
sonic: http://www.sonic.com
dts: http://www.surcode.com
libav: http://www.mplayerhq.hu [The avcodec-54 and avutil-52 can't be replaced by new ones (now 59/58)]
aften: http://win32builds.sourceforge.net/aften/index.html
flac: http://sourceforge.net/project/showfiles.php?group_id=13478&package_id=12675[you can replace the libFLAC.dll with the 1.4.2 (2022-10-22) last version from here (https://www.rarewares.org/lossless.php) (rename the 32 bits libFLAC_dynamic.dll)]
r8brain: http://www.voxengo.com/product/r8brain
haali: http://haali.cs.msu.ru/mkv
mkvtoolnix: http://www.bunkus.org/videotools/mkvtoolnix

WARNING:

Nero 8 won't work for eac3to cause Nero 8 doesn't allow its DirectShow filters to be used from outside of Nero ShowTime. Nero 9 is currently not supported, either. So please use/install Nero 7 if you want to use Nero DirectShow filters in eac3to.

background information:

Please note that there are three common problems with "properly" decoding compressed audio tracks. The first problem has to do with that the purpose of a media player is different than the purpose of people like us who want to reencode audio tracks. A media player may want to adjust volume levels to specific parameters and circumstances. We reencoders absolutely want to avoid any processing on the precious audio data. This is a conflict which can not always be solved. The three known problems are:

(1) Some decoders apply DRC (Dynamic Range Compression). Stupidly Dolby's license requests that DRC must be applied, unless your decoder has an option to turn it on/off. Only then the decoder is allowed to not apply DRC. The purpose of DRC is to limit the dynamic range of an audio track, mainly to not annoy neighbours or to accomodate to noisy living room conditions. For our purposes DRC is catastrophic.
(2) Some decoders forcefully downconvert to 16bit, or raise volume for unknown reasons or do other stupid stuff.
(3) Some decoders don't make use of the full available information, but just extract and decode the "core" of the compressed audio data, which obviously results in less than optimal audio quality.

evaluation of available decoders:

The Sonic (E-)AC3 decoder forcefully applies DRC. As a result I do not recommend to use the Sonic (E-)AC3 decoder. The current version of the Sonic TrueHD decoder doesn't work properly at all. The Sonic DTS decoder is good for DTS, DTS-ES, DTS-96/24, DTS-HD Master Audio and DTS-HD High Resolution tracks. It's quite slow, though and it decodes DTS-HD 7.1 tracks only as 5.1. Most DTS-ES and DTS-HD Master Audio 6.1 tracks are fully decoded as 6.1, though. The Nero TrueHD decoder is working perfectly fine, but is limited to 5.1 channels. The Nero (E-)AC3 works fine for most audio tracks, but sometimes DRC is still enabled, so it's not really recommended. The Nero DTS decoder ignores the additional DTS-HD information and only decodes the DTS core. The Cyberlink decoders always only output 16bit and can't be used outside of PowerDVD. So they currently do not make a lot of sense for eac3to. The libav/ffmpeg (E-)AC3 decoder works really well, but don't support 7.1 decoding yet. The libav/ffmpeg DTS decoder works well, too, but currently ignores the additional DTS-HD information, so especially DTS-HD Master Audio can't be decoded in full quality. The libav/ffmpeg TrueHD decoder works perfectly fine including full 7.1 decoding. The ArcSoft DTS decoder works well for DTS and DTS-HD decoding, but decodes many 7.1 tracks either incorrectly or with forced processing/mixing. It does support every format and channel configuration, though. The ArcSoft TrueHD and ArcSoft (E-)AC3 decoders are currently not supported by eac3to. The dcadec decoder works great for all kinds of DTS formats and channel configurations, except LBR/XSA (low bitrate) content is not properly supported yet.

default decoders used by eac3to:

AAC: Nero
MP1, MP2, MP3: libav/ffmpg
(E-)AC3: libav/ffmpeg
TrueHD: libav/ffmpeg
DTS(-HD): dcaDec
DTS-LBR/XSA: ArcSoft

making the ArcSoft DTS decoder work in eac3to:

Sometimes the ArcSoft DTS decoder doesn't work even after you've installed the full retail software. This can usually be fixed by manually adding the ArcSoft "Bin" folder (e.g. "C:\Program Files\Common Files\ArcSoft\Bin") to your environment path. If you don't know how to do this, google "environment path".

Don't ask for pirated versions of ArcSoft, Nero, Sonic and Surcode. You should buy those.

EDIT (2016-02-13 and later) by tebasuna51:
At this moment only Nero 7 DirectShow are needed to decode AAC, ArcSoft and Sonic are not needed at all (libdcadec and libav can do the job).
There are troubles decoding some new implementations of EAC3, then for decode AAC and EAC3 is better use a modern version of ffmpeg.
And Surcode can be replaced by the experimental ffdcaenc encoder to dts like you can see after

In addition to eac3to audio outputs: PCM, AC3 (libAften), DTS (Surcode), AAC (NeroAacEnc M4A) and FLAC (libFLAC), you can use the eac3to stdout to pipe PCM to any encoder than support stdin.

Generic syntax:

eac3to INPUT stdout.wav [-EAC3TO_PARAMETERS] | ENCODER PARAMETERS - OUTPUT

Where:
- INPUT, ENCODER and OUTPUT must be in quotted full paths if aren't at same folder than eac3to.
- INPUT can be any supported audio file or a track from a supported container: "D\tmp\input.mkv" 2:
- PARAMETERS can be any supported encoder parameters.
- The order of: PARAMETERS - OUTPUT, can be different for some encoders. See examples below.

You can use also Aften.exe, NeroAacEnc.exe and Flac.exe encoders to override eac3to defaults.
Also you can use other soft to filter output: eac3to INPUT stdout.wav | sox ... | encoder ...
Some tested examples:

eac3to INPUT stdout.wav | Lame (http://www.rarewares.org/mp3-lame-bundle.php) -V 2 - output.mp3
eac3to INPUT stdout.wav | TwoLame -b 192 - output.mp2 (Twolame v0.3.10b)
eac3to INPUT stdout.wav | ffdcaenc (http://forum.doom9.org/showthread.php?p=1702067#post1702067) -i - -o output.dts -l -b 1509.75
eac3to INPUT stdout.wav | OggEnc2 (http://www.rarewares.org/ogg-oggenc.php) -q 3 --ignorelength -o output.ogg -
eac3to INPUT stdout.wav | opusenc (http://forum.doom9.org/showthread.php?p=1714809#post1714809) --ignorelength --bitrate 128 - output.opus
eac3to INPUT stdout.wav | qaac (http://forum.doom9.org/showthread.php?p=1395355#post1395355) -V 99 --ignorelength --adts --no-delay -o output.aac -
eac3to INPUT stdout.wav | fhgaacenc (http://forum.doom9.org/showthread.php?t=161806) --vbr 5 --ignorelength - output.m4a
eac3to INPUT stdout.wav | NeroAacEnc -br 192000 -he -ignorelength -if - -of output.m4a
eac3to INPUT stdout.wav | Flac -5 --ignore-chunk-sizes -o output.flac -
eac3to INPUT stdout.wav | Aften -b 640 -pad 0 -readtoeof 1 -exps 32 -s 1 - output.ac3
eac3to INPUT stdout.w64 | ffmpeg (http://ffmpeg.zeranoe.com/builds/) -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 output.ac3

Note than last ffmpeg example require PCM data with w64 header, the other encoders support wav data greater than 4GB with special parameter -ignorelength or equivalent.
Of course ffmpeg can be used for other supported audio output format, but is highly recommended for AC3 because libAften.dll is outdated and can cause troubles (https://forum.doom9.org/showthread.php?p=1794293#post1794293)

[twan69666]

Awesome, we've needed a separte thread for a while. Cant wait to play with some TrueHD tracks! Thanks for all the hard work Madshi

[Darth Pinous]

Thanks for the tool.

It's good to know that it's finally possible to convert TrueHD.
As I've been playing around with Nero these past days, could you tell what sort of working graph permit the decoding of TrueHD ?
Is it Nero Splitter -> Nero Audio Decoder 2 ?
Do you need to specify anything in the Nero Audio Decoder 2 options ?

[madshi]

I'm using this graph:

"Nero File Source (Async) -> Nero Splitter -> Nero Audio Decoder 2 -> Dump"

You need to rename GraphEdt.exe to Recode.exe, otherwise the filters won't connect. I didn't change anything in the audio decoder settings. Make sure you have the "Remix" mode set to "None", though, otherwise the audio decoder might mess around with the audio data.

[Kurtnoise]

Why the Nero filters are needed ? The Haali Media Splitter is not appropriate for this task ?

[madshi]

The Haali Media Splitter doesn't "export" the TrueHD track. And even if it did, we'd still need the Nero Audio Decoder for TrueHD decoding. Haali's Media Splitter isn't decoding, of course, it's only splitting.

One thing I just found out is confusing me quite a lot:

The TrueHD track I was testing (Vendetta) is supposed to be 48kHz 16bit. The Nero Audio Decoder always outputs 24bit. Ok, so I thought I could simply convert that to 16bit without any loss. However, I've just checked: If I convert the 24bit Nero output to 16bit and then back to 24bit, the result is different to the original. It seems that there's more than 16bit real information in the Nero output. But the TrueHD track is only 16bit! I don't quite understand this. My only explanation would be that the Nero audio decoder does some processing on the decoded data, e.g. volume adjustment or something like that. That's kind of bad, though, since that means that converting to 16bit may end up harming the audio quality. But keeping 24bit *significantly* increases the final FLAC file size. Arghh.......

[zgx]

Awesome work with TrueHD!

We just need to figure out the 24/16 bit thingy. As you say you don't want to harm the audio but at the same time you don't want your FLAC files to be larger then they need to be.

:thanks:

[Rectal Prolapse]

Perhaps this can be modified to work with Blu-ray m2ts files, at least for Blu-ray TrueHD tracks?

Anyways - great work!

[Neo Fagin]

Unfortunately, 64k 1.0 eac3 (Dog Day Afternoon) is still unsupported when using Sonic. Is this a limitation of Sonic's decoder?

[madshi]

Mono files are not supported by eac3to, simply because I don't have such a file for testing. If you can send me the first 5MB of an eac3 mono file, I can probably add support for that, I guess.

About m2ts: Probably Nero supports that, too. But again I need a sample file. If you can get me a m2ts file with only TrueHD in it and no video (e.g. by using the Transport Stream Remuxer), 5MB should be enough.

[Rectal Prolapse]

madshi, do you know if TrueHD is decoded in Stereo only or multichannel? Damn, I should have asked before I bought the decoder pack.

[madshi]

Definitely in multichannel. All 5.1 channels have valid information in them. Even more so than I'd like... :D (see problem I explained above with 24 -> 16 -> 24 conversion)

[Neo Fagin]

Mono files are not supported by eac3to, simply because I don't have such a file for testing. If you can send me the first 5MB of an eac3 mono file, I can probably add support for that, I guess.

http://www.sendspace.com/file/5ahu3b

a 5MB sample of a mono file

[Rectal Prolapse]

Thanks madshi! I just realized that when I look at filter properties (in GraphEdit) it always says the audio is 2 channel PCM, even when it isn't. :)

[tebasuna51]

The TrueHD track I was testing (Vendetta) is supposed to be 48kHz 16bit.
For what 16 bit?

BitDepth have no sense with encoded audio in frequency domain.
In the document 'Digital Audio Compression Standard (AC-3, E-AC-3) Revision B' say:
"8.2.1.1 Input Word Length
The AC-3 encoder accepts audio in the form of PCM words. The internal dynamic range of AC-3 allows input wordlengths of up to 24 bits to be useful."

Then if a input of 24 bits is appropriate for ac3 encoder also the output can be 24 bits.

Edit: and Aften accept 24 bits input without problem.

[madshi]

For what 16 bit?

BitDepth have no sense with encoded audio in frequency domain.
In the document 'Digital Audio Compression Standard (AC-3, E-AC-3) Revision B' say:
"8.2.1.1 Input Word Length
The AC-3 encoder accepts audio in the form of PCM words. The internal dynamic range of AC-3 allows input wordlengths of up to 24 bits to be useful."

Then if a input of 24 bits is appropriate for ac3 encoder also the output can be 24 bits.

Edit: and Aften accept 24 bits input without problem.
You are correct - but only when talking about (E-)AC3. Things are totally different with TrueHD. TrueHD is a lossless codec. That means, it takes 16bit, 20bit or 24bit input data and does *NOT* convert that into frequency domain (doing that wouldn't be lossless), but simply compresses the audio data similar to how ZIP works. So if you feed e.g. a 16bit track into the TrueHD encoder and if you then feed the encoded data into a TrueHD decoder, you should get a perfect 1:1 copy of the original data out of the decoder.

[shon3i]

Any chance to implement stdout, so we can use in BeHappy or other 3rd party app as decoder?

[madshi]

http://www.sendspace.com/file/5ahu3b

a 5MB sample of a mono file
Here's version 1.6 with mono E-AC3 support:

http://madshi.net/eac3to.zip

Please note that Sonic's audio decoder crashes with mono E-AC3 files. Nero's audio Decoder works fine, though. So eac3to automatically switches to Nero, if you feed it a mono file. I think for E-AC3 decoding you don't need the HD-DVD/Blu-Ray plugin, btw. That plugin is only needed for TrueHD decoding, if I'm not mistaken.

[madshi]

Any chance to implement stdout, so we can use in BeHappy or other 3rd party app as decoder?
Which format would I have to output to stdout to make BeHappy happy? Raw or Wav? Or something else? What you can do right now is to ask eac3to to decode to a WAV file. You can then use this WAV file in BeHappy. The main problem is that eac3to doesn't directly receive the data from the decoders. Instead eac3to asks the decoders to dump the output into a file (and there's no way for me to change that). So the raw decoded data must be written to a file first, anyway. No way around that. Of course, once that raw file is written, eac3to could open and read it and feed it into stdout. Or eac3to could first ask Sox.exe to convert the raw file into a wav and feed *that* file to stdout. But I'm wondering how much would be won by doing that.

[tebasuna51]

@madshi
Sorry by my previous post, I was thinking in ac3 of course.

@shon3i
For BeHappy we need AviSynth plugin decoders, stdout is useless.
But maybe we can open graph filter files (.grf) with DirectShowSource().

[Neo Fagin]

E-AC3, 1.0 channels, 2:04:39, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.

Nero 7.8.5.0 premium

[Rectal Prolapse]

madshi and others: According to this insider (a mixing guy), TrueHD decoders always output 24 bit:

http://www.avsforum.com/avs-vb/showthread.php?p=10386940&&#post10386940

In TrueHD, it is unnecessary for the codec to know the wordlength, as it delivers exactly what was input to the output. All TrueHD decoders support up to 24 bits. If the source is, say, 16 bits, the output is 16 MSB bits with lossless data, and 8 LSB bits of zeros.

[Thunderbolt8]

E-AC3, 1.0 channels, 2:04:39, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.

Nero 7.8.5.0 premium
already seeing 7.9.8.2 just normal version though, not premium -.-

[dchard]

Madshi:

1. is there any way to convert the trueHD track to mono wavs?

2. What nero version should I need, if I need trueHD decoding?

Thank you, very impressive work.

Dchard

[madshi]

E-AC3, 1.0 channels, 2:04:39, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.

Nero 7.8.5.0 premium
That's strange. It works for me! Please make sure you have the "Remix Mode" set to "None" in the "Nero Audio Decoder 2" filter.

Can you please try eac3to on the 5MB chunk you sent me? Maybe it works with the 5MB chunk but not with the full size file?

[madshi]

madshi and others: According to this insider (a mixing guy), TrueHD decoders always output 24 bit:

http://www.avsforum.com/avs-vb/showthread.php?p=10386940&&#post10386940
Thank you for the link! Roger Dressler actually is working at Dolby.

Hmmmm... What he sais is exactly what I was expecting to see. However, the least 8 LSB bits were not zero in my tests with the Nero audio decoder. That's exactly what is confusing me. I was expecting them to be zero, but they aren't.

[madshi]

1. is there any way to convert the trueHD track to mono wavs?
What do you need that for? DTS encoding? Anyway, there are multiple ways to get mono wavs. One way would be to ask eac3to to give you a wav file. You can then with appropriate tools convert that to mono wavs. I think there are some freeware tools that can do that. Sorry, don't know which.

2. What nero version should I need, if I need trueHD decoding?
I'm using the latest version with the latest version of the HD-DVD/Blu-Ray plugin. I don't know if earlier versions work. Or is your question not about version numbers, but about which edition I'm using? I'm using the Ultra Edition, but I guess that any edition works, as long as you have the HD-DVD/Blu-Ray plugin installed. Not sure which edition you need to have to be able to install that plugin, though. Also please note that I'm just guessing here.

[MacGyver2k]

E-AC3, 1.0 channels, 2:04:39, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.

Nero 7.8.5.0 premium

got the same problem as mentioned above, here is a complete mono eac3 file for ya to test with:
http://rapidshare.com/files/32135670/falke_eng.rar.html

[pasta03]

Where I can find Nero Audio Decoder 2

[madshi]

Where I can find Nero Audio Decoder 2
http://www.nero.com

[madshi]

got the same problem as mentioned above, here is a complete mono eac3 file for ya to test with:
http://rapidshare.com/files/32135670/falke_eng.rar.html
It works for me !!!

Which size does the raw file have when eac3to complains? Mine is 1026344448 bytes.

[mikeyakame]

to activate the nero truehd/eac3 decoding in nero you need to enter the serial number for the bluray/hddvd plugin as far as i know, this opens the features up on the dshow filters, (dont ask me how as I dont have a clue/dont care since they work perfectly when paid for) as I couldnt decode dts/eac3 with the filters until I activated them with the serial....

so dont bother asking people for these filters, as they are useless to you without a working serial.

madshi,

if you want to decode the truehd in 16bit, add the nero sound processor to the chain in graphedit, and set it to output 16bit lpcm, i always use this method to transcode eac3 to ac3, so im pretty sure it should work for truehd as well, ill give it a go later on this evening when ive got a sample track to test.

edit: ill test out that mono eac3 track too, and see what results i get.

[mikeyakame]

madshi,

i believe their problem with it not working is the filters arent fully activated, thus not outputting a readable eac3 stream, rather outputting garbage as they require activation for eac3 support.

edit:

and for whomever asked, the nero plugin supports m2ts demuxing and all that, it finds all the streams, etc, you can use a chain with nero file source (async) -> nero audio decoder -> directsound (output) and select whichever streams you want to play back, as this method works great for dumping the lpcm / eac3 from bluray to raw output, with all channels in tact.

[mikeyakame]

D:\Temp\Encoding\HDDVD Tools>eac3to falke_eng.eac3 test.ac3 -640
E-AC3, 1.0 channels, 1:58:47, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
Audio Decoder output was 24 bit this time.
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...
Starting "aften" failed. A valid *.wav file was created successfully, though.

19/05/2007 08:35 PM 1,026,344,528 test.wav

works perfectly for me too.

edit:

Macgyver,

I just noticed when you tried to process it the time stamps are wrong, you have 2:04:xx, I have 1:58:xx, seems like you need to activate the filters to be able to use eac3 support.

[madshi]

if you want to decode the truehd in 16bit, add the nero sound processor to the chain in graphedit, and set it to output 16bit lpcm
But this will not tell the decoder to output 24bit in the first place. Instead it will take the decoder's 24bit and downconvert it to 16bit. Of course that's possible to do. But it shouldn't be any better than using sox to downconvert 24bit to 16bit (which is what eac3to is currently using). The problem we need to solve is to find out why the 24bit output of the Nero filter seems to contain information in the least significant 8 bits when being fed a 16bit TrueHD track. That shouldn't be the case. And downconverting such a 24bit output may result in quality loss... :(

[madshi]

add the nero sound processor to the chain in graphedit, and set it to output 16bit lpcm, i always use this method to transcode eac3 to ac3
Btw, IMO that's a bad idea!! There's no advantage whatsoever in cutting the eac3 decoder output down to 16bit, if you plan to reencode in ac3. The size of the final ac3 file does not depend on which bit depth you feed it. And cutting 24bit down to 16bit can actually hurt sound quality. So you'll get better audio quality with the same file size if you keep the eac3's decoders at 24bit!

[zgx]

Just to confirm your findings about "V for Vendetta" Madshi. The movie is 132 min long and when I demux the TrueHD track I get a file that is "1 368 231 154" bytes. That equals ~1350 Kbps. The source that was losslessly compressed has to be 16bit/48Khz or else TrueHD would have had to compress increadibly much better then all other available lossless codecs.

[mikeyakame]

im pretty sure i had to do that to get sonic dd encoder to accept the input mono waves, im pretty sure it had some kind of problem accepting 24-bit input..its been a while so I dont really remember the reason, but its either that or wave wizard had problems with 24bit wave files.

and for the record, i haven't been able to hear any quality loss, as a matter of fact, the encoded ac3 sounds as good as the eac3 source after i encode it with sonic dd encoder. aften and the few other encoders i tried did produce a slight quality loss on the resulting ac3...the db levels and range seemed to be a little out compared to the original track

[madshi]

im pretty sure i had to do that to get sonic dd encoder to accept the input mono waves, im pretty sure it had some kind of problem accepting 24-bit input..its been a while so I dont really remember the reason, but its either that or wave wizard had problems with 24bit wave files.
Ah well, don't know, maybe the Sonic DD encoder doesn't like 24bit input, I don't know. I've no experience with tha encoder.

[madshi]

Just to confirm your findings about "V for Vendetta" Madshi. The movie is 132 min long and when I demux the TrueHD track I get a file that is "1 368 231 154" bytes. That equals ~1350 Kbps. The source that was losslessly compressed has to be 16bit/48Khz or else TrueHD would have had to compress increadibly much better then all other available lossless codecs.
Did you have a chance to check out the content of the decoding? Can you confirm that the least significant 8 bits are not zero? It's really too bad cause if true it means that we don't have a 1:1 lossless decoding yet... :(

[mikeyakame]

if the true-hd is outputting in 24bit..try changing the settings for your sound card to 16bit output depth...

im trying it now...ill let you know in a couple of minutes whether it works

edit: nope makes no difference, nero doesn't have a setting for outputs...not sure...ill play with a true-hd track later on

[zgx]

Did you have a chance to check out the content of the decoding? Can you confirm that the least significant 8 bits are not zero? It's really too bad cause if true it means that we don't have a 1:1 lossless decoding yet... :(No have not gotten around to buy the filters yet so cannot try it myself at the moment. But if you take the original 24 bit PCM file that you got from the decode and then open it in a HEX editor you should be able to figure out if one third of the bits contains zeros (8bit, 8bit, 8bit, 8bit, 8bit, 8bit, 8bit, 8bit...). In an hex editor "00" probably represents 8bit with just zeros.

[Austin Forgotten]

Hi there,

I am very curious about the TrueHD format, as I have spent a considerable amount of time looking at MLP. I do not own any HD equipment to rip HDDVD or Blueray. I have a few questions and I sure would like to get my hands on a demo/test file to have a look. For one, what is the file extension used for TrueHD format? Oh, I suppose I should have asked this 1st.. Has anyone been able to demux TrueHD to a file yet? Anyway if someone could provide a link to a test file [doesn't have to be to large :-) ]. Thanks.

-AF

[zgx]

Hi there,

I am very curious about the TrueHD format, as I have spent a considerable amount of time looking at MLP. I do not own any HD equipment to rip HDDVD or Blueray. I have a few questions and I sure would like to get my hands on a demo/test file to have a look. For one, what is the file extension used for TrueHD format? Oh, I suppose I should have asked this 1st.. Has anyone been able to demux TrueHD to a file yet? Anyway if someone could provide a link to a test file [doesn't have to be to large :-) ]. Thanks.

-AFThere are a few programs able to demux TrueHD streams out of a HD DVD. The most common is probably EVOdemux available at http://pel.hu/down

I think different programs use different extensions for TrueHD.

At the moment the major problem is that the only way to play TrueHD or decode it is from within EVO files (using PowerDVD for example or now eac3to).

[zgx]

Anyway if someone could provide a link to a test file [doesn't have to be to large :-) ].Check your mailbox in a few minutes.

[Rectal Prolapse]

madshi: I just looked at a raw PCM dump and I see the same thing - the 3rd byte of each sample is not always zero! What is going on here...?

[mikeyakame]

this is the info for The Matrix TrueHD track.

Audio attribute :
Audio application mode : surround
Audio format : MLP
Sampling frequency : 48000
Sampling quantization : 24
Number of audio channels : 6
Number of audio streams : 8
Bitrate: 2544 Kbps

I get both 24-bit outputs with this track on Nero Showtime & PowerDVD 7.3.2911

[pasta03]

Someone can give me direct link to Nero Audio Decoder 2 not to all nero recording only for decoder

[Revgen]

Nero packages the recorder and every software it makes together without any separate packages. You cannot get it any other way. At least not legally.

[Revgen]

BTW do Nero's decoders work in Windows XP.

[zgx]

this is the info for The Matrix TrueHD track.

Audio attribute :
Audio application mode : surround
Audio format : MLP
Sampling frequency : 48000
Sampling quantization : 24
Number of audio channels : 6
Number of audio streams : 8
Bitrate: 2544 Kbps

I get both 24-bit outputs with this track on Nero Showtime & PowerDVD 7.3.2911The Matrix TrueHD soundtrack is "1 682 946 322" bytes long. The movie is 136 minutes long. That gives it a bitrate of 1611 Kbps. That's about the same bitrate as a 6 channel 16/48 movie soundtrack compressed to FLAC.

Uncompressed 16 bit/48 kHz/6 channel PCM: 4500 Kbps
Uncompressed 24 bit/48 kHz/6 channel PCM: 6750 Kbps

It's hard to say but I do not believe it's possible to losslessly compress a 24/48 soundtrack from an action movie like The Matrix to 1611 Kbps and the way I see it they must have used a 16/48 source.


People here say that the least significant 8 bits are not zero as they should be. Is it possible that even if the TrueHD track is stored with the least significant 8 bits as zeros that the Nero Audio Decoder reads the 16 bit most significant bits and upsample them to 24 bit?

[madshi]

BTW do Nero's decoders work in Windows XP.
Yes, I'm running everything in XP.

[madshi]

this is the info for The Matrix TrueHD track.

Audio attribute :
Audio application mode : surround
Audio format : MLP
Sampling frequency : 48000
Sampling quantization : 24
Number of audio channels : 6
Number of audio streams : 8
Bitrate: 2544 Kbps
Where do you have this information from? Do you have a software which outputs this kind of information? 8 audio streams for MLP is extremely strange because to my best knowledge MLP only supports 6 channels!

FWIW, highdefdigest claims Matrix TrueHD to be 48/16:

http://hddvd.highdefdigest.com/matrix_umc.html

[zgx]

Where do you have this information from? Do you have a software which outputs this kind of information? 8 audio streams for MLP is extremely strange because to my best knowledge MLP only supports 6 channels!

FWIW, highdefdigest claims Matrix TrueHD to be 48/16:

http://hddvd.highdefdigest.com/matrix_umc.htmlI think the Matrix HDDVD actually has 8 different audio tracks. Look at my reply a few posts up for the information on The Matrix.

[Neo Fagin]

Another unusual error, this time with TrueHD (Matrix Reloaded)

Muxing evo's first audio track to raw. Please wait...
Converting the raw file to wav. Please wait...
sox.exe sox: Channels value '0' is not a positive integer
"sox" reported error code "1". A valid *.raw file was created sucessfully, though.

[madshi]

Another unusual error, this time with TrueHD (Matrix Reloaded)

Muxing evo's first audio track to raw. Please wait...
Converting the raw file to wav. Please wait...
sox.exe sox: Channels value '0' is not a positive integer
"sox" reported error code "1". A valid *.raw file was created sucessfully, though.
Ah, I think that's a bug I introduced when adding support for mono E-AC3 files. I'll upload a fixed build soon.

[Neo Fagin]

Just out of curiosity, what sox/aften commandlines does eac3to use during the conversion process?

[madshi]

Just out of curiosity, what sox/aften commandlines does eac3to use during the conversion process?
Depends on the audio tracks, of course... :)

Here's version 1.7:

http://madshi.net/eac3to.zip

There's a new option "-8". eac3to can currently not automatically find out how many channels the TrueHD track has, so it assumes 6 channels. If you know it to be 8 channels, use the "-8" option. I'm not sure how many channels Matrix has. Also I'm not sure if Nero's decoder is even able to output more than 5.1. Probably not. So the default (6 channels) is probably the right choice for Matrix, too.

[Neo Fagin]

Nero vision reports Matrix Reloaded to be 7.1 (and says it's unsupported by Nero at this time) but it does play audio despite the error, though I don't know if it's playing 5.1, stereo, mono or what. That may be the cause of the numerous problems the track is causing.

[madshi]

Well, the sox complaint you reported was a clear bug in eac3to. So you should retry with version 1.7. Maybe it works. But I guess decoding will probably be limited to 5.1 channels.

[Neo Fagin]

Retrying now, I'll let you know if it works or not.

[Neo Fagin]

Seems to work now.

May as well ask for a "feature" so to speak - since this tool writes raw/wav files to disk as intermediates to the process, would it be possible to make this tool capable of doing two transcodes at once if given the option - ac3 and flac on the same command line? Saves having to run it twice since the raw/wav files are already created on disk for either process individually.

[madshi]

Seems to work now.
So you have valid 5.1 now? Or were you able to get 7.1?

May as well ask for a "feature" so to speak - since this tool writes raw/wav files to disk as intermediates to the process, would it be possible to make this tool capable of doing two transcodes at once if given the option - ac3 and flac on the same command line? Saves having to run it twice since the raw/wav files are already created on disk for either process individually.
That's kind of difficult. The main problem is that for ac3 encoding eac3to keeps using 24bit (cause it's no disadvantage for final file size), while for flac recording the TrueHD track is converted to 16bit. So the raw files already differ.

[Neo Fagin]

So you have valid 5.1 now? Or were you able to get 7.1?
Not sure. All I know is before, it wrote a 3.5gb raw file and failed on the sox step, this time it wrote a 6.5gb raw file and is now in the middle of the wav transcode step. It's taking an immensely long time due to the size of the files involved. In this instance I'm doing ac3 so it'd only be 5.1 results. I'll make a flac track afterwards and see how it goes.
That's kind of difficult. The main problem is that for ac3 encoding eac3to keeps using 24bit (cause it's no disadvantage for final file size), while for flac recording the TrueHD track is converted to 16bit. So the raw files already differ.Ah. No worries, just figured it'd be a big time-saver.

[madshi]

Not sure. All I know is before, it wrote a 3.5gb raw file and failed on the sox step, this time it wrote a 6.5gb raw file and is now in the middle of the wav transcode step. It's taking an immensely long time due to the size of the files involved. In this instance I'm doing ac3 so it'd only be 5.1 results.
Well, you could feed sox any data, e.g. an "exe" file and it would still be happy. So that sox and ac3 encoding runs through is not yet a proof of that everything is fine. If the raw file contains 8 channels, but eac3to thinks it's 6 channels, the output will be "interesting". But let's just wait and see... :)

[Neo Fagin]

The output appears to be okay 5.1 AC3. I haven't loaded it into Vegas or the like to do a comparison with ac3 made from the eac3 via sonic, though, but during playback the channels all appear to be in the right spots etc.

Could it be possible to do the flac/ac3 at once transcode thing if the -keep24 switch is used, thusly both flac and ac3 would logically use the same 24-bit raw file? Flac's primarily an archival medium in this instance so filesize isn't a terrible concern. If not, no big deal, just the thought occured to me. I suspect it's maybe a half-dozen lines of code to implement at any rate.

[zgx]

Not sure. All I know is before, it wrote a 3.5gb raw file and failed on the sox step, this time it wrote a 6.5gb raw file and is now in the middle of the wav transcode step. It's taking an immensely long time due to the size of the files involved. In this instance I'm doing ac3 so it'd only be 5.1 results. I'll make a flac track afterwards and see how it goes.
Ah. No worries, just figured it'd be a big time-saver.If you make a FLAC track first then run:

flac -d yourflacfile.flac
aften -whateveroptions yourwavfile.wav

But as Madshi pointed out for FLAC files the PCM is converted to 16 bit first. But since most TrueHD tracks seams to use 16 bit source material it shouldn't really matter that much. The AC3 file will probably sound just the same.

[madshi]

Could it be possible to do the flac/ac3 at once transcode thing if the -keep24 switch is used, thusly both flac and ac3 would logically use the same 24-bit raw file?
In that case you can do a simple batch file like this:

eac3to some.evo some.wav -keep24
eac3to some.wav some.flac
eac3to some.wav some.ac3

That's as fast as you can get. If eac3to did this for you internally, it wouldn't be any faster.

since most TrueHD tracks seams to use 16 bit source material it shouldn't really matter that much. The AC3 file will probably sound just the same.
Well, if only true TrueHD decoder output was really 16bit when being fed a 16bit TrueHD track... :( I've just contacted Nero support about this problem. Hopefully they'll send me a useful reply.

[ACrowley]

mhh strange

When i want to type in the Nero HDDVD/BluRay Plugin Serial, Nero tells me i have to upgrade to newest Version first.

i use the lstest Version....?

[zgx]

I've just contacted Nero support about this problem. Hopefully they'll send me a useful reply.Yes lets hope they fix it would be awsome to get a bit perfect decode.

[madshi]

mhh strange

When i want to type in the Nero HDDVD/BluRay Plugin Serial, Nero tells me i have to upgrade to newest Version first.

i use the lstest Version....?
That's quite strange. Which version do you have? You should probably contact Nero support.

[序列人]

mhh strange

When i want to type in the Nero HDDVD/BluRay Plugin Serial, Nero tells me i have to upgrade to newest Version first.

i use the lstest Version....?

me to...
my ver. is Nero7.Premium.Reloaded.v.7.5.9.1.MULTiLANGUAGE...

[madshi]

As far as I can see the latest version is 7.8.5.0.

[Darth Pinous]

This means that the serial is wrong, nothing to do with a version problem. If you buy the plug-in from the Nero website, you will see that the serial they will give you is waaay different from the one you're trying to use...

Just tested Nero plugin to convert Matrix Reloaded Dolby TrueHD track into Wavpack lossless track. Man, what a killer audio track ! :cool:

I tried first to convert the WAV file to 16 bits with WaveWizard before converting it to Wavpack, then to convert directly the 24bit WAV file to Wavpack.

The 16bits Wavpack was 1,43 GB (which make a 1,5 Mbps average bitrate, quite dynamic movie !) while the 24bits Wavpack was 2,9 GB (I use option -h for Wavpack reencodes).

Despite the huge size difference, I couldn't hear any differences between the two audio tracks. So, not sure that there is one.


Does anyone know how to mux an elementary audio track (let's say a demuxed Dolby TrueHD track) in an EVO container or could anyone point me to a guide explaining how to do it ?

[zgx]

Does anyone know how to mux an elementary audio track (let's say a demuxed Dolby TrueHD track) in an EVO container or could anyone point me to a guide explaining how to do it ?"Sonic Scenarist Advanced Content" would be my best bet.

[orbitlee]

Guys, don't bother madshi with the trouble by pirated nero keygen.

madshi, great work. I managed to build the filter chain in graphedit for TrueHD. Looks like it's time to consider the truehd source file :-)

BTW, I got a sample of m2ts with TrueHD audio track(from Blu-ray Nine Inch Nails), unfortunately nero splitter does not split it correctly(only ac3 audio track, no truehd).

[madshi]

Guys, don't bother madshi with the trouble by pirated nero keygen.
Thanks! :)

madshi, great work. I managed to build the filter chain in graphedit for TrueHD. Looks like it's time to consider the truehd source file :-)
That would be very nice!!

[Rectal Prolapse]

orbitlee: Confirmed. I couldn't get the NIN Blu-ray working with Nero either. No TrueHD track to select. :(

[Borbus]

I'm using this with the Sonic Audio Decoder. I have the decoder set to environment quiet. I've tried converting eac3 to wav. I have the soundtrack in eac3 and ac3 from a DVD. Comparing eac3 decoded with this to ac3 decoded with foobar2000 this seems to be obviously dynamically compressed.

Has anyone else noticed this? Is it a bug with Sonic?

[madshi]

I'm using this with the Sonic Audio Decoder. I have the decoder set to environment quiet. I've tried converting eac3 to wav. I have the soundtrack in eac3 and ac3 from a DVD. Comparing eac3 decoded with this to ac3 decoded with foobar2000 this seems to be obviously dynamically compressed.

Has anyone else noticed this? Is it a bug with Sonic?
Try using "environment normal". That might do the trick. At least that's what I'm using.

[Borbus]

Changing the environment setting doesn't seem to actually do anything. The environment setting you should be using is "Quiet" because this is when no dynamic range compression is applied.

I wish they made these options straight forward, e.g. Dynamic Range Compression, Off, On, Full rather than a confusing backward system.

[madshi]

The environment setting you should be using is "Quiet" because this is when no dynamic range compression is applied.
Are you sure about that? I thought "Normal" would mean "Range Compression Off".

I wish they made these options straight forward, e.g. Dynamic Range Compression, Off, On, Full rather than a confusing backward system.
Very much agreed!!

[tebasuna51]

Are you sure about that? I thought "Normal" would mean "Range Compression Off".
I agree too about the confusion.

But, at least in PowerDVD and the Azid decoder, "Normal" mean apply the attenuation (DRC) present in the ac3 stream.

"Quiet" or "None" mean don't apply at all DRC, preserving the full dynamic range.

"Half", "Heavy", ... mean apply a part or more than the DRC calculated at encoder time and present in the ac3 stream.

[Borbus]

Well these are the options I remember PowerDVD having:


Quiet Environment: Experience the full dynamic range of Dolby Digital sound.
Normal Environment: Experience the compressed dynamic range of Dolby Digital sound.
Noisy Environment: Experience boosted sound. Strongly recommended for notebook users.

Also quoted here on Hydrogen Audio: http://www.hydrogenaudio.org/forums/index.php?showtopic=53219

[madshi]

Ah, thanks guys. But then Borbus sais: "Changing the environment setting doesn't seem to actually do anything." Hmmmm... Very confusing.

This discussion makes me think that maybe the problems with the 16/24 bit TrueHD output may also have to do with DRC? There's a setting for DRC in ShowTime, but not in the decoder settings dialog.

[zgx]

Thanks for quoting me Borbus. This is something that has been bothering me for some time now. Why can't they just make it user friendly instead of complicating things? As far as I know only DD/AC3 and TrueHD and probably also DD+/EAC3 has metadata used by DRC.

[Borbus]

Am I doing this right? I'm changing the environment setting in the Cineplayer DVD Decoder Options in the Control Panel. It does throw up some weird error when I quite that but the setting seems to apply.

[madshi]

Sounds right to me.

[XeroFightsAlone]

What does this error entail?

"Connecting pins (2) failed."

I have no video track in my evo file, only TrueHD audio.

[madshi]

What does this error entail?

I have no video track in my evo file, only TrueHD audio.
Do you have the Nero HD-DVD/Blu-Ray registered?

[Revgen]

Is there a way that you could add DTS support too?

For some reason, the libavcodec decoders don't decode the DTS audio from my Basic Instinct HD-DVD very well. There's pops and clicks throughout. The Sonic Decoder plays them fine.

[madshi]

Is there a way that you could add DTS support too?

For some reason, the libavcodec decoders don't decode the DTS audio from my Basic Instinct HD-DVD very well. There's pops and clicks throughout. The Sonic Decoder plays them fine.
Well, you could try using the same way you need to go for TrueHD, that is: Use EvoDemux to create one EVO file with nothing but one audio track in it. I think this should also work with DTS, although I haven't tested that yet.

[Revgen]

Eh, nevermind. This Basic Instinct track appears to have problems even with Sonics decoder, only in different parts of the file.

Looks like I'll have to use the AC3 track from my DVD version.

[madshi]

Eh, nevermind. This Basic Instinct track appears to have problems even with Sonics decoder, only in different parts of the file.
Well, going the EVO way I suggested, you'd be using the Nero decoders.

[Revgen]

Well, going the EVO way I suggested, you'd be using the Nero decoders.

Well, I haven't bought nero yet. It'll take me a month or so before I can afford it. Guess I'll just have to wait.

[Borbus]

Thanks for quoting me Borbus. This is something that has been bothering me for some time now. Why can't they just make it user friendly instead of complicating things? As far as I know only DD/AC3 and TrueHD and probably also DD+/EAC3 has metadata used by DRC.

Does Sonic actually output anything different when you change the settings? I've tried loud, normal and quiet and it always gives me the same output which is dynamically compressed.

[moshmothma]

I take it I cannot use Nero Directshow filters to playback TrueHD at this point? How is playback of HDDVD and Bluray with Nero Showtime? Thanks

[madshi]

I take it I cannot use Nero Directshow filters to playback TrueHD at this point?
I think you can, but you have to rename the media player to "recode.exe".

[Pumpa]

New Ahead Nero v7.9.6.0, so maybe new filters ???

[ACrowley]

Well, you could try using the same way you need to go for TrueHD, that is: Use EvoDemux to create one EVO file with nothing but one audio track in it. I think this should also work with DTS, although I haven't tested that yet.


Strange

I had Clicks ans Pops with Perfume HDDVD DTS (core)Track

First i demuxed it with evodemux and had the cracking in the dts core.
But demuxed with Graphedit and SonicHDDemuxer it plays clean
Maybe you try it

Never had an clicks etc in a HDDVD DTS HD Track with AC3Filter since i demux DTS HD with SonicHDDemuxer

[XeroFightsAlone]

Do you have the Nero HD-DVD/Blu-Ray registered?
I only have HD DVD playback capability with PowerDVD. Is there a way to dump the audio from that?

[Daodan]

When I use graph mehtod to create wavs (so not using eac3to) I create the chain (file source>nero splitter>nero audio), I can play the graph nicely, but when I try to use that graph as directshowsource in avs it sais no 8,16,24 bit.. output from grf. Tried to interpose some filters between nero and avs and still same error. Graphs with dd+ (using sonic) or anything else have no problem.
Anyone else getting this? Does it have to do with nero messing around so that it can't connect? Any ideas around this?
Thanks.

EDIT: eac3to 5.1 wav is fine, but I find it easier and safer using direct grf>avs>getchannel>wav>dts than messing with 5.1 wavs.

[madshi]

I only have HD DVD playback capability with PowerDVD. Is there a way to dump the audio from that?
Maybe it's possible, but I don't know how.

@Daodan, have you renamed the exe file you're using to "recode.exe"? If that doesn't help, I don't have any other ideas.

[Badness]

Eac3to works great for me, but I was wondering is there a way to use a filter in the avs file to allow avisynth to read the eac3 file directly without changing it over to ac3 first.

[Daodan]

@Badness, eac3 (dd+ files) you can feed directly in avs using graphedit (build chain with correct filters (see the evob demultiplexer thread)), and open the grf as dss.

@madshi, yes, graphedit is renamed to recode.exe. Also another annoying thing is that I can't seems to use nero for vc1 decoding. I can connect all including the video decoder but not any renderer, or when I try to open the grf in avs, again the annoying error (this time requiring for a proper input colorspace).
I'm quite disappointed that you have to pay for a plugin that doesn't allow you almost anything (at least for encoding purposes), most probably because of the stupid limitation nero has put over their filter's usability in other applications.

[Badness]

What if I dont want to use graphedit, is there a avisynth dll that I can load?

[HookedOnTV]

Trying to go to ac3 and I get this:

E-AC3, 5.1 channels, 1:56:28, 640kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The NCBENUM return code is: 0x0
The NCBENUM return code is: 0x0
The file size of the raw file doesn't seem to fit.

What does it mean?

[madshi]

The file size of the raw file doesn't seem to fit.

What does it mean?
It means that the Sonic Audio Decoder's output did not meet the expectation. Probably it has only decoded two channels instead of 6 or something like that. Try uninstalling and reinstalling the Sonic filters. Or play with the audio settings (control panel -> CinePlayer options).

[ideas_man69]

i've noticed that sonic's audio decoder has some options:

audio output: 2.0, 5.1 etc - explains itself
environment: quiet, normal, noisy - this is the one that interests me.

this is to do with dynamic range yeah? which option do i chose so the original sound is untouched? i've been using normal but i have a feeling that quiet might be the one that doesn't modify the volume.

[infernix]

Am I doing this right? I'm changing the environment setting in the Cineplayer DVD Decoder Options in the Control Panel. It does throw up some weird error when I quite that but the setting seems to apply.

Can you try comparing the DRC levels when you set Sonic to SP/DIF? When I choose that, the DRC selection box turns grey, which seems to me that it completely disables DRC. In theory one would expect that with SP/DIF it'll just send the stream as-is.

It seems to work for me but I can't listen to the output right now (working remotely - it might be totally corrupted or silent), nor can I compare the DRC levels.

[HookedOnTV]

It means that the Sonic Audio Decoder's output did not meet the expectation. Probably it has only decoded two channels instead of 6 or something like that. Try uninstalling and reinstalling the Sonic filters. Or play with the audio settings (control panel -> CinePlayer options).

Bingo! I thought I read something about the number of channels the decoder was configured for but couldn't find it.

Thanks.

[InorganicMatter]

Hey guys, I'm trying to convert an audio track from Jet Li's Fearless, and I keep getting this:

"The file size of the raw file doesn't seem to fit."

Hmmm....I've got the full Nero 7.6.9.0 suite, what gives?

[madshi]

Hey guys, I'm trying to convert an audio track from Jet Li's Fearless, and I keep getting this:

"The file size of the raw file doesn't seem to fit."

Hmmm....I've got the full Nero 7.6.9.0 suite, what gives?
Guys, if you have a problem with eac3to, please always post the full text eac3to text output.

[InorganicMatter]

Guys, if you have a problem with eac3to, please always post the full text eac3to text output.

Sorry, didn't think the rest important:

Can only be decoded by Nero's filters....
Muxing to raw...
File size doesn't match

[madshi]

That's still not the full text. You haven't posted whether eac3to reports this eac3 track to be mono, stereo or multichannel. It seems it's mono, right?

[InorganicMatter]

That's still not the full text. You haven't posted whether eac3to reports this eac3 track to be mono, stereo or multichannel. It seems it's mono, right?

Yeah, it's that mono sample posted earlier. I'm getting this error on anything I try to convert.

[madshi]

Did you buy and register the Nero decoder correctly? Don't know why it doesn't work for you. The sample that was posted earlier works perfectly fine for me.

[InorganicMatter]

Did you buy and register the Nero decoder correctly? Don't know why it doesn't work for you. The sample that was posted earlier works perfectly fine for me.

Buy and register? :confused: I'm converting a demuxed EAC3 file, not an audio file in an EVOB. The readme says that to do that I need:

(1) "Sonic HD Demuxer", "Nero File Source / Splitter" or "DTS/DD+/AC3 source"
(2) "Sonic Audio Decoder 4.2" or "Nero Audio Decoder 2"
(3) "Dump" filter

The ones I bolded come with Nero 7 Ultra (which I have installed), and I installed the Dump filter correctly. I thought I only needed the HD/BD plugin if I was converting a track contained within an EVOB. Right? Or is the readme incorrect?

[ideas_man69]

yeh for some reason - what ever i do, i can not get audio from nero's filters

[mikeyakame]

cause you didnt buy the plugin and enter a legit serial, thats why the nero filters dont work. you can't just install it and cause the filters are there assume they work and there is a problem with them, this is false, the problem is and has been mentioned over n over that you must get a legit serial to activate the HD components of the plugins.

[InorganicMatter]

cause you didnt buy the plugin and enter a legit serial, thats why the nero filters dont work. you can't just install it and cause the filters are there assume they work and there is a problem with them, this is false, the problem is and has been mentioned over n over that you must get a legit serial to activate the HD components of the plugins.

I thought the readme said I only need the HD plugin if I'm converting audio in an EVOB, not if it is already demultiplexed. I own Nero 7 Ultra and have it installed properly with my serial, but I though that was all I needed for converting demultiplexed audio.

[madshi]

I'm not fully sure. I think the Nero Audio decoder does work without the HD-DVD/Blu-Ray plugin. But I'm not fully sure which formats it supports and which not. It definitely does not support TrueHD without a properly registered HD-DVD/Blu-Ray plugin. Originally I thought it would do E-AC3 decoding without the plugin, but I'm not so sure, anymore. What I can say is that the mono sample that was posted earlier in this thread does work for me.

[InorganicMatter]

What I can say is that the mono sample that was posted earlier in this thread does work for me.

You have the BD/HD plugin, right? Because the only thing I'm doing different than you is I don't have that plugin. Maybe I do need it after all, even for a demultiplexed file.

[madshi]

I do have the plugin. I can't say for sure, though, if it's really what you need to E-AC3 decoding. As I said, I'm not sure about that.

[The_Keymaker]

Fellow forum members,

I have made a feeble attempt at a GUI driven interface for eac3to. EAC3toGUI.exe v0.6. You can get it here:

http://www.sendspace.com/file/jwghja
(EDIT: Above link is for the updated version v0.63)

Just place the .exe in the same same directory as the eac3to.exe program. If you want to place it in another directory, you must select the directory where eac3to is located. Just click on settings and browse to eac3to.exe.

Please try it out and provide constructive feedback and report any bugs. I will do my best to implement reasonable features and improvements.

By the way, eac3to.exe does NOT like paths or filenames with spaces. please keep this in mind when naming or storing your source or output files.
EDIT: Based upon input from Madshi, I've added support for filenames with spaces in EAC3toGUI v0.63

Special thanks to Madshi for his work on eac3to.exe

Regards,
The_Keymaker

[SvT]

Hello The_Keymaker.

I was waiting for a gui so I wanted to try this out !

My eac3to.exe is on my e: drive so I placed your GUI in the same dir. Now when I click convert the first command is C: and the program can't find eac3to.exe.

Am I missing something because even when I point out the location of to exe to your GUI it won't work.

Thanks for your help.

P.S. The location of the exe is correctly stored in the ini file.

[madshi]

@The_Keymaker, that's cool... :-) Btw, eac3to should be able to cope with spaces in filenames, as long as you put quotes around the file names. Well, at least it should. Please let me know if there's a problem with that. Thanks!

[The_Keymaker]

@SvT,

This version (v0.61) should fix the problem. Please confirm.

[See subsequent posts for latest version]

The_Keymaker

[SvT]

The_Keymaker !

Yes, this version seems to do the trick !

Just for fun I moved all the files to my D: drive and the program still does the job ! If I run into anything else I let you know !

[The_Keymaker]

@SvT,
Thank you for the feedback. Let me know if you discover anymore bugs.

@Madshi,
I will be uploading an update that has quote support for filenames soon.

[The_Keymaker]

OK, here is the latest update for EAC3toGUI (v0.63).

http://www.sendspace.com/file/jwghja

It now places quotes around the source and destination to accommodate filenames with spaces.

The_Keymaker

[SvT]

Hello again, I just needed a beer so I was out. On returning I saw this new version so I gave it a try !

Unfortunately it doesn't work for me !

You added support for " (quotes) very good but as you read my drive is the E and in my language typing " and e ends up like ë ! Also (you must think I do this on purpose) my HDDVD movie is on my i and typing i and " ends up like ï ! You can imagine this doesn't look good on the command line !

eac3to.exe Ï:\Bourne\HVDVD_TS\FEATURE_1.EVO"Ë:\Movie\test.ac3"-6

Hope this helps !

EDIT: When I move my project to my D drive (and added some additional spaces) this is the command line !

eac3to.exe "D:\TEST WITH SPACES\FEATURE 1.EVO""D:\MORE TESTING\test.ac3"

You see the missing space between in and output ! This stops the eac3to.exe !

Good thing is spaces in filenames and directories are totally accepted !

[The_Keymaker]

@SvT,

Sorry about that. Here is the link to the previous version (v0.61) in case you need it.

http://www.sendspace.com/file/7apfxk

Going forward, I may have to keep a separate version for those that do not desire support for filenames with spaces OR just have one version with an option the end user can enable or disable.

The_Keymaker

EDIT: SvT I have not been able to duplicate the missing space between the source and destination paths. Can you run the program again and verify if the space is missing in the command line preview box?

[SvT]

@The_Keymaker,

Finally I figured it out !!!!

Version 0.61 is working perfectly ! Even with my e and i drive ! It works for all keyboard layouts (no spaces allowed).

Version 0.63 is working perfectly ! With e and i drive and spaces allowed !

BUT My keyboard settings are Dutch/US(international) and this crashes 0.63 ! In this language " + e = ë and version 0.63 doesn't like this ! This also causes the missing space between the in/output (if I only select my d drive) ! If I change my keyboard settings the GUI runs fine !!!

Thanks for the GUI, it's working fine now with my new keyboard settings !

[madshi]

I don't really understand why the keyboard settings should make a difference. :confused:

[madshi]

Guys, I've just uploaded eac3to 1.8. It now also supports PCM as input format. This should make life a bit easier for the Blu-Ray guys.

http://madshi.net/eac3to.zip

eac3to v1.8, freeware by madshi.net

[...]

Options:
-192 use 192 kbit/s for AC3 encoding
-448 use 448 kbit/s for AC3 encoding
-640 use 640 kbit/s for AC3 encoding
-orbitlee force the use of Orbit Lee's DD+ source filter
-nero force the use of Nero's filters
-keep24 don't convert TrueHD output to 16 bit
-96000 TrueHD output or PCM file is 96 kHz
-8 TrueHD output or PCM file is 8 channels
-16 PCM file is 16 bit
-24 PCM file is 24 bit
-little PCM file is little endian
-big PCM file is big endian
-0,1,2,3,4,5 remap the channels to the specified order
-blu-ray similar to "-0,1,2,5,3,4 -big"
See first post of this thread for the full help text.

PCM support is kind of intelligent: If you don't specify a bitdepth, eac3to tries to find out automatically which bitdepth the PCM file has. So you can simply do: "eac3to some.pcm some.flac -blu-ray". That should automatically detect the bitdepth, remap the channels correctly and finally FLAC the audio without size limitations.

[SvT]

I don't really understand why the keyboard settings should make a difference. :confused:

It all has to do with the way you type your quote " !

On my keyboard if I want to type a quote it's " + space !!! otherwise it will affect the next character ! if I type quote + e this results in ë if I type quote space e it ends up like "e

In this way it effects the command typed out on the command line ! But as I said if I set my keyboard to quote = " without the space the GUI is working great !

Hope this helps.

[madshi]

It all has to do with the way you type your quote " !

On my keyboard if I want to type a quote it's " + space !!! otherwise it will affect the next character ! if I type quote + e this results in ë if I type quote space e it ends up like "e

In this way it effects the command typed out on the command line ! But as I said if I set my keyboard to quote = " without the space the GUI is working great !
Yeah, I understand that. But the GUI isn't typing the chars one by one. Instead it most probably calls a specific API to run eac3to, which gets a full path with " around it. Normally these language specific tricks apply to manual keyboard input, only, but not to win32 APIs. Anyway...

[Borbus]

Can you try comparing the DRC levels when you set Sonic to SP/DIF? When I choose that, the DRC selection box turns grey, which seems to me that it completely disables DRC. In theory one would expect that with SP/DIF it'll just send the stream as-is.

It seems to work for me but I can't listen to the output right now (working remotely - it might be totally corrupted or silent), nor can I compare the DRC levels.
Wouldn't that transcode it to AC3 or stereo PCM? I can't get it to work like that anyway. It works for a while but then the dump file just stops getting bigger.

[otaku975]

hello i still have the problem when trying to convert a mono file....


E-AC3, 1.0 channels, 2:02:49, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 2,92 MB.
The expected file size for 24 bit is 4,38 MB.
The real file size is 0 Bytes.

nero is correctly installed... and i tried this one two computers with two versions of nero , and it does the same...
nero audio decoder 2 is set to none in the options...
if anyone has an idea..
thanks

[madshi]

The real file size is 0 Bytes.
Seemingly the Nero Audio Decoder didn't output anything, as the final audio file was 0 byte long (= empty). eac3to has no chance to do anything if the Nero Decoder doesn't work correctly. Now I'd really like to know why the Nero audio decoder doesn't work for you. Maybe it does only work if the HD-DVD/Blu-Ray plugin is registerd? I don't know...

[otaku975]

Seemingly the Nero Audio Decoder didn't output anything, as the final audio file was 0 byte long (= empty). eac3to has no chance to do anything if the Nero Decoder doesn't work correctly. Now I'd really like to know why the Nero audio decoder doesn't work for you. Maybe it does only work if the HD-DVD/Blu-Ray plugin is registerd? I don't know...


well i installed nero normally , how do i know or get the hddvd bluray plugin to be registered ?

[madshi]

how do i know or get the hddvd bluray plugin to be registered ?
By buying it? :) Ok, I understand that it's ugly to buy something if you're not sure it will work. But well, don't know what else to say.

[otaku975]

well i gave up and converted it via graphedit , worked fine :p
thanks for the help madshi

[madshi]

Well, that's strange. If it works via graphedit, it should also work via eac3to. Hmmmm...

[The_Keymaker]

I am travelling but hope to have a revised EAC3toGUI release featuring Madshi's newest options when I return.

The_Keymaker

[starkhouse33]

well i gave up and converted it via graphedit , worked fine :p


How did you setup your graph? I'm trying to get this to work with a mono track I have.

[madshi]

How did you setup your graph? I'm trying to get this to work with a mono track I have.
Does eac3to fail to convert that mono track for you, too? Does it also say that the Audio Decoder output seems to be 0 bytes long? Thanks...

[ACrowley]

any news about the TrueHD Output 16bit or 24bit ?

Well, for a AC3/DTS/WMA reencode is doesnt Matters when íf the Source Wave Bitrate is 16 or 24bit.
Ouptut Filesize is the same on 16 and 24Bit Source.
For AC3 Aften can take 24Bit wave Input, Steinberg Nuendo Encoder too.
So 24bit Output from TrueHD is no Problem

But for a FLAC encode it matters cause the FLAC encoded Output Filesize on 24bit Wave Source is by far bigger

I cant hear a difference between 16 and 24bit TrueHD Output encoded to FLAC
So whats correct, 16bit or 24bit TrueHD Output from Nero Decoder ?

Question about NeroAudioDecoder:

Is NeroAudioDecoder using any DRC when i set it to Remixmode none" ? I dont thing so
And Sonic ? its still not cleared if the Decoder use any DRC ?

[starkhouse33]

Does eac3to fail to convert that mono track for you, too? Does it also say that the Audio Decoder output seems to be 0 bytes long? Thanks...

when I try to convert a mono track with eac3to I get this

E-AC3, 1.0 channels, 1:58:47, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 5.36 MB.
The expected file size for 24 bit is 8.04 MB.
The real file size is 0 Bytes.

[xkodi]

@madshi

without a doubt TrueHD track found on 'V for Vendetta' HD-DVD is 24bit/48kHz.

why i'm sure about that? what is my test procedure?

TrueHD is based on MLP (aka Packed PCM), so i've used 'SurCode MLP' to create test MLP files from WAV files with different resolution and number of channels e.g. 24bit/44.1kHz/5.1, 16bit/48kHz/2.0, etc. and named them test2444151.mlp, 164820.mlp, etc.

then i've used WinDVD 8.0 B06.109 to open those test MLP files, WinDVD doesn't produce sound while playback MLP files, but every time for every test MLP file, WinDVD shows (from the menu 'Setup->Information') the right resolution and number of channels, exactly the same as the resolution and number of channels of the source WAV file from which the corresponding MLP file was created:

Stream:
Type: Packed PCM
Bitrate: 0 Kbps
Number of main channels: 6
LFE Channel: None
Sampling Frequency: 44.1khz
Bits Per Sample: 24
Output:
Sample Rate: 44 KHz
Bits per Sample: 16

so that means without a doubt, that WinDVD checks the MLP (TrueHD) headers for the real resolution of the encoded audio.

the next step was to demux the 'V for Vendetta' HD-DVD using EVOdemux and to rename the resulted TrueHD file to truehd.mlp and then WinDVD 8.0 B06.109 said that it is:

Stream:
Type: Packed PCM
Bitrate: 0 Kbps
Number of main channels: 6
LFE Channel: None
Sampling Frequency: 48khz
Bits Per Sample: 24
Output:
Sample Rate: 48 KHz
Bits per Sample: 16

so the TrueHD (MLP) track found on 'V for Vendetta' HD-DVD is 24bit/48kHz/5.1

to double check the accuracy of the information that WinDVD outputs, i've also done another test:

this time with the bonus TrueHD track from the 'Nine Inch Nails Live: Beside You in Time' HD-DVD, which is:

633 042 317 bytes in size

and WinDVD says it's 24bit/48kHz/5.1.

i've created, using eac3to, both 16bit and 24bit WAV files from that TrueHD track and then using 'SurCode MLP' compressed them to MLP, the result:

16bit WAV file size: 660 487 104 bytes
compressed to MPL size: 358 807 274 bytes

24bit WAV file size: 990 730 628 bytes
compressed to MPL size: 649 693 162 bytes

the MLP compressed size of the 16bit WAV file is 358 807 274 bytes and MLP compressed size of the 24bit WAV file is 649 693 162 bytes, which is almost the same number as the original TrueHD track - 633 042 317 bytes in size and because TrueHD is optimized MLP compression again can be concluded that the original TrueHD track is 24bit exactly as WinDVD reports.

[Rectal Prolapse]

An AVSForum Insider has mentioned that TrueHD decoders always output 24 bit.

I think you need to do your test again on a disc that is known to have a 16 bit TrueHD audio track. I have a feeling that you will think that all TrueHD releases are 24 bit, when in fact the majority (or even all) of them are 16 bit.

[xkodi]

An AVSForum Insider has mentioned that TrueHD decoders always output 24 bit.

I think you need to do your test again on a disc that is known to have a 16 bit TrueHD audio track. I have a feeling that you will think that all TrueHD releases are 24 bit, when in fact the majority (or even all) of them are 16 bit.

i do not agree, because in the first test i don't use the output of TrueHD decoders, instead i'm using series of homemade MLP tracks with known resolution to prove, that WinDVD can find and shows the real resolution of the MLP/TrueHD tracks correctly and so WinDVD uses the MLP/TrueHD headers to find that information and there is no usage of TrueHD decoders at all in that test.

in the second test, yes, but only to double check the results from the first test, i'm using Nero TrueHD decoder to decompress 600MB TrueHD track (demuxed from real HD-DVD disc) in both 16bit and 24bit WAV files and then compress them again to MLP and that test shows that the initial source for the original TrueHD track is 24bit, because if the original TrueHD track is created from 16bit WAV it will be only 340MB.

so i can't see any flaw in the tests and that is why i'm posting here, if you have something more concrete for the possible flaw, also why you are so sure, that TrueHD tracks are 16bit, WinDVD and the 'Surecode MLP' test shows exactly the opposite and in that test, i'm not using the output of TrueHD decoders at all.

[edit] and here is another prove, even if TrueHD decoders always output 24 bit, WinDVD shows that the stream is 16bit if and only if it is really 16bit (the output is from homemade 16bit MLP track):

Stream:
Type: Packed PCM
Bitrate: 0 Kbps
Number of main channels: 6
LFE Channel: None
Sampling Frequency: 44.1khz
Bits Per Sample: 16
Output:
Sample Rate: 44 KHz
Bits per Sample: 16

[madshi]

without a doubt TrueHD track found on 'V for Vendetta' HD-DVD is 24bit/48kHz.
:eek:

TrueHD is based on MLP (aka Packed PCM), so i've used 'SurCode MLP' to create test MLP files from WAV files with different resolution and number of channels e.g. 24bit/44.1kHz/5.1, 16bit/48kHz/2.0, etc. and named them test2444151.mlp, 164820.mlp, etc.

then i've used WinDVD 8.0 B06.109 to open those test MLP files, WinDVD doesn't produce sound while playback MLP files, but every time for every test MLP file, WinDVD shows (from the menu 'Setup->Information') the right resolution and number of channels, exactly the same as the resolution and number of channels of the source WAV file from which the corresponding MLP file was created:

so that means without a doubt, that WinDVD checks the MLP (TrueHD) headers for the real resolution of the encoded audio.

the next step was to demux the 'V for Vendetta' HD-DVD using EVOdemux and to rename the resulted TrueHD file to truehd.mlp and then WinDVD 8.0 B06.109 said that it is:
Well, this does not fully convince me yet. Why? Because the headers of MLP files differ from the headers of TrueHD files. eac3to is able to analyze MLP files and extract the "bitdepth" information from the MLP header. But eac3to is not able to do the same for TrueHD files. I'm not sure. Maybe WinDVD is able to do that. Maybe WinDVD is cleverer than eac3to, that's possible. But it's also possible that WinDVD is reporting the correct bitdepth for MLP files while always reporting 24bit for TrueHD files.

to double check the accuracy of the information that WinDVD outputs, i've also done another test:

this time with the bonus TrueHD track from the 'Nine Inch Nails Live: Beside You in Time' HD-DVD, which is:

633 042 317 bytes in size

and WinDVD says it's 24bit/48kHz/5.1.

i've created, using eac3to, both 16bit and 24bit WAV files from that TrueHD track and then using 'SurCode MLP' compressed them to MLP, the result:

16bit WAV file size: 660 487 104 bytes
compressed to MPL size: 358 807 274 bytes

24bit WAV file size: 990 730 628 bytes
compressed to MPL size: 649 693 162 bytes

the MLP compressed size of the 16bit WAV file is 358 807 274 bytes and MLP compressed size of the 24bit WAV file is 649 693 162 bytes, which is almost the same number as the original TrueHD track - 633 042 317 bytes in size and because TrueHD is optimized MLP compression again can be concluded that the original TrueHD track is 24bit exactly as WinDVD reports.
Ok, that quite clearly shows that that bonus track is most probably 24bit. But that's still no 100% proof that Vendetta is 24bit, too. Can you do the same test with Vendetta, too? Or could you try to find a TrueHD track which WinDVD reports at being 16bit only?

Now if only we could make eac3to decode demuxed MLP/TrueHD files then we could do another double check and decode your home made MLP files and check whether the Nero decoder produces bit identical output! You don't happen to have an authoring software available which can package your home made MLP files into an EVO container, so that eac3to can decode your MLP files?

[madshi]

when I try to convert a mono track with eac3to I get this

E-AC3, 1.0 channels, 1:58:47, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 5.36 MB.
The expected file size for 24 bit is 8.04 MB.
The real file size is 0 Bytes.
I've just tried it again and it works on my test mono chunk. Would it be possible for you to make that mono track available for me for testing?

[xkodi]

Can you do the same test with Vendetta, too? Or could you try to find a TrueHD track which WinDVD reports at being 16bit only?


in several days i will do the same test with Vendetta and post the results; i have only two HD-DVD discs with TrueHD and they both say 24bit, but i will look for more TrueHD samples.


You don't happen to have an authoring software available which can package your home made MLP files into an EVO container, so that eac3to can decode your MLP files?

actually i can pack them in AOB container and as far as i know EVOdemux can convert AOB to EVOB, but i don't know if 'Nero Audio Decoder 2' is capable to decode pure MLP, but i will try that test in several hours and post back the result.

[madshi]

Thanks, xkodi!

[xkodi]

actually i can pack them in AOB container and as far as i know EVOdemux can convert AOB to EVOB, but i don't know if 'Nero Audio Decoder 2' is capable to decode pure MLP, but i will try that test in several hours and post back the result.

24bit/48kHz/5.1ch WAV file -> 'SurCode MLP' -> 24bit/48kHz/5.1ch MLP file -> 'discWelder CHROME' -> AOB container -> 'EVOdemux 0.626' -> EVOB container

but the resulted EVOB file opened again with EVOdemux says that the stream is LPCM, not MLP (PPCM) and WinDVD also reports LPCM and nothing can playback that EVOB, so seems that is a dead end, because EVOdemux can't convert AOB with MLP to EVOB correctly, maybe MLP can't be muxed in EVOB at all, only the TrueHD modification.

[mp4p2p]

eac3toGUI 0.63 this link:http://www.sendspace.com/file/jwghja

my web can't open,may be is isp question,

please,who can send me?

my email:dhdbase@gmail.com

thanks

[madshi]

maybe MLP can't be muxed in EVOB at all, only the TrueHD modification.
Actually some HD-DVD tracks are MLP and not TrueHD. So it's definitely possible to mux MLP into EVOB. I think EvoDemux just doesn't do it properly, sadly... :(

[Sagittaire]

I have problem ... ???

D:\Mes dossiers\Codec\HDDVD\eac3to>eac3to.exe azerty.eac3 azerty.wav
E-AC3, 5.1 channels, 0:00:30, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
Getting "Dump" instance failed

[madshi]

I have problem ... ???
You need to register the dump filter which ships with GraphEdit. Use "regsvr32 dump.ax".

[xkodi]

Actually some HD-DVD tracks are MLP and not TrueHD. So it's definitely possible to mux MLP into EVOB. I think EvoDemux just doesn't do it properly, sadly... :(

very conclusive test, at least in my opinion:

1) 'Nine Inch Nails Live: Beside You in Time' bonus track, TrueHD size:

633 042 317 bytes

2) decompressed using eac3to to 16bit WAV, size: 660 487 112 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

358 807 274 bytes

3) decompressed using eac3to to 16bit WAV and then converted to 24bit using Wavewizard v0.54b,
now the least significant bits are zeroes checked with hexeditor, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

362 789 876 bytes

4) decompressed using eac3to to 24bit WAV file, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

649 693 162 bytes

so 2) and 3) shows that if the source WAV file is not real 24bit, but 16bit padded to 24bit with zeroes
for least significant bits MLP compression is enough intelligent and that doesn't significantly increase
the size of the compressed file and that plus 3) and 4) clearly shows that the source WAV file for the
'Nine Inch Nails Live: Beside You in Time' bonus TrueHD track should be real 24bit.

p.s. soon i will do the same test for Vendetta TrueHD track.

[Rectal Prolapse]

xkodi: Well, I'm going by the reported bitdepth from the studios and hidef reviewers - although they aren't always reliable.

Perhaps the only thing we can do is ask the Insiders on AVSForum.com.

Good luck with your tests!

[madshi]

very conclusive test, at least in my opinion
So you got your MLPs successfully packages into EVO now? :)

1) 'Nine Inch Nails Live: Beside You in Time' bonus track, TrueHD size:

633 042 317 bytes

2) decompressed using eac3to to 16bit WAV, size: 660 487 112 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

358 807 274 bytes

3) decompressed using eac3to to 16bit WAV and then converted to 24bit using Wavewizard v0.54b,
now the least significant bits are zeroes checked with hexeditor, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

362 789 876 bytes

4) decompressed using eac3to to 24bit WAV file, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

649 693 162 bytes
Wait a moment! If the Nero decoder works correctly, step 4 should have given you a WAV file with the least significant 8 bits being zero. And if you reencoded this, you should again end up with only a 360MB of MLP. So this clearly shows that the Nero decoder is either not working properly, or that it's doing some additional processing on the audio data after decoding. Do you agree?

2) and 3) shows that if the source WAV file is not real 24bit, but 16bit padded to 24bit with zeroes
for least significant bits MLP compression is enough intelligent and that doesn't significantly increase
the size of the compressed file and that plus 3) and 4) clearly shows that the source WAV file for the
'Nine Inch Nails Live: Beside You in Time' bonus TrueHD track should be real 24bit.
I think the main clue is that the TrueHD file is only barely smaller than the 16bit WAV file. This is a quite clear sign that the TrueHD file has more than 16bit. Otherwise it would have an extremely bad compression ratio, which is very unprobable.

I think we should be able to draw similar conclusions about V for Vendetta: If the TrueHD file is noticably smaller than the 16bit WAV then probably the TrueHD file is only 16bit. If the TrueHD file is however almost as big as the 16bit WAV, then the TrueHD track is probably more than 16bit. Agreed?

[xkodi]

So you got your MLPs successfully packages into EVO now? :)

no, i still can't get MLP stream muxed into EVO, the test file is TrueHD stream taken from real HD-DVD disc and then decompressed with eac3to to 16 and 24 bit WAVs, that are used during the test.


Wait a moment! If the Nero decoder works correctly, step 4 should have given you a WAV file with the least significant 8 bits being zero. And if you reencoded this, you should again end up with only a 360MB of MLP. So this clearly shows that the Nero decoder is either not working properly, or that it's doing some additional processing on the audio data after decoding. Do you agree?

if the PCM source is 16bit and because MLP is lossless, then least significant byte should be zero in the decompressed 24bit WAV, but at the moment i can't test that, because i can't find a way how to mux homemade MLP with known PCM source into EVO.

WinDVD always shows that TrueHD tracks are 24bit, maybe all TrueHD tracks are actually 24bit, no matter from the resolution of the PCM source from which are created, maybe when the PCM source is 16bit, before the MLP compression step, they do some processing to upsample it to 24bit and if that is the case Nero decoder works correctly, because that info is added before the compression.


I think the main clue is that the TrueHD file is only barely smaller than the 16bit WAV file. This is a quite clear sign that the TrueHD file has more than 16bit. Otherwise it would have an extremely bad compression ratio, which is very unprobable.

I think we should be able to draw similar conclusions about V for Vendetta: If the TrueHD file is noticably smaller than the 16bit WAV then probably the TrueHD file is only 16bit. If the TrueHD file is however almost as big as the 16bit WAV, then the TrueHD track is probably more than 16bit. Agreed?

yes, i completely agree with that, my Vendetta test clearly also supports that conclusion especially in combination with the previous 'Nine Inch Nails' test:

1) 'V for Vendetta' TrueHD track, file size:

1 368 231 154 bytes

2) decompressed using eac3to to 16bit WAV, size: 4 580 550 320 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

1 292 730 780 bytes

3) decompressed using eac3to to 24bit WAV file, size: 6 871 150 160 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

3 315 710 672 bytes

so i want to summarize:

- WinDVD always shows that TrueHD tracks are 24bit, but that doesn't mean that the source PCM from, which the TrueHD track is created, is really 24bit

- some TrueHD tracks seems to be created from real 24bit PCM source like the 'Nine Inch Nails Live: Beside You in Time' bonus TrueHD track, so downsampling them to 16bit is probably not so good idea

- other TrueHD tracks seems to be created from 16bit PCM source like 'V for Vendetta' TrueHD track and so downsampling them to 16bit is probably completely save

so maybe eac3to default behavior when converting to WAV/FLAC should be the following: to check the size of the TrueHD track and according to that automatically to decide if the PCM source is 16bit or 24bit and then choose the correct resolution for the output and of course to be possible manually to select 16bit or 24bit. what do you think about that?

[ABOOABOO]

when i used the gui i get

C:\eac3to>eac3to.exe "C:\eac3to\input.eac3" "C:\Documents and Settings\bLaCk MaM
bA\Desktop\TEST.ac3" -640
E-AC3, 5.1 channels, 1:48:25, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The audio decoder didn't accept the input stream.

anybody else got this

[The_Keymaker]

Here is the link to the latest version of Eac3toGUI. It is version 0.82 and features all of Madshi's latest options.

http://www.sendspace.com/file/9kjvud

Please let me know if you find any problems.

Regards,
The_Keymaker

EDIT: Version 0.80 had a bug which deleted the last two characters on the command line.

[madshi]

WinDVD always shows that TrueHD tracks are 24bit, maybe all TrueHD tracks are actually 24bit, no matter from the resolution of the PCM source from which are created, maybe when the PCM source is 16bit, before the MLP compression step, they do some processing to upsample it to 24bit and if that is the case Nero decoder works correctly, because that info is added before the compression.
No, there's nothing added before compression. I'm sure of that because of 2 reasons:

(1) Roger Dressler (Dolby) said that if the source was 16bit, the output of a TrueHD decoder should have the least significant 8 bits set to zero.

(2) If there was "real" information added to the source to make it 24bit, the "V for Vendetta" TrueHD file size would have to be much bigger than it really is (see your test 3).

yes, i completely agree with that, my Vendetta test clearly also supports that conclusion especially in combination with the previous 'Nine Inch Nails' test:

1) 'V for Vendetta' TrueHD track, file size:

1 368 231 154 bytes

2) decompressed using eac3to to 16bit WAV, size: 4 580 550 320 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

1 292 730 780 bytes

3) decompressed using eac3to to 24bit WAV file, size: 6 871 150 160 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

3 315 710 672 bytes
Thanks for the test. Good information!

so i want to summarize:

- WinDVD always shows that TrueHD tracks are 24bit, but that doesn't mean that the source PCM from, which the TrueHD track is created, is really 24bit
Agreed.

- some TrueHD tracks seems to be created from real 24bit PCM source like the 'Nine Inch Nails Live: Beside You in Time' bonus TrueHD track, so downsampling them to 16bit is probably not so good idea
Agreed.

- other TrueHD tracks seems to be created from 16bit PCM source like 'V for Vendetta' TrueHD track and so downsampling them to 16bit is probably completely save
I'd agree only if the decoder behaved correctly and set the 8 least significant bits to zero. But the Nero decoder doesn't do that. I've no idea what exactly the Nero decoder is doing. Maybe it's doing range compression? Or changing volume slightly? Anyway, the output of the Nero decoder is clearly "finer" than 16bit, either due to a bug or due to additional processing on the audio data. So downsampling the decoder output to 16bit sounds like losing information to me.

------

Here comes a polite request to all people who have bought the HD-DVD/Blu-Ray plugin: Could you please contact Nero, report about the incorrectly working TrueHD decoder and ask them to fix this? I've already done that, but got no reply at all. Maybe we'll have more success if multiple people complain... Thanks!

[madshi]

Here is the link to the latest version of Eac3toGUI. It is version 0.82 and features all of Madshi's latest options.
Thanks! Just had a look at the GUI. Looks nice to me. Some comments:

(1) I know the help doesn't say it, but theoretically the AC3 bitrate can be anything between 32 and 640. So if you want you could add further bitrates, or even a slider or up/down buttons or whatever you like. Smaller bitrates than 192 may make sense e.g. for mono tracks.

(2) I'd suggest to rename the "Select Filter" text to "Force Filter" cause there are also Sonic filters that eac3to can use, if you don't activate any of the "force" switches. Alternatively you could replace the two checkboxes with 3 radios like "use whatever filter eac3to prefers", "use Nero filters" and "use orbitLee's filter".

(3) 96khz is also a valid option for PCM. 8 channels is also a valid option for TrueHD (although currently it doesn't make much sense, since the Nero decoder currently only supports 5.1 decoding).

(4) eac3to supports any channel order you like, not just 0,1,2,3,4,5 and the Blu-Ray one. E.g. you could do "5,4,0,1,2,3" or "2,1,4,0,3,5" or for 8 channels you could do "5,7,1,2,6,4,3,0" or anything else. I just don't know if there's any use for any other order than the default one (0,1,2,3,4,5[,6,7]) and the Blu-Ray one (0,1,2,5,3,4[,6,7]). But you can tell eac3to to reorder the channels to any strange order you like. Don't know if it's worth it adding this capability to the GUI, though.

(5) The TrueHD options can all be selected at the same time, but the GUI always allows only one option at the same time.

(6) Most PCM options can also be selected at the same time. Of course you can only select either 16bit or 24bit. And you can only select "big" or "little". But you could select e.g. "24bit", "8 channels" and "big" at the same time. Btw, you may want to rename "big" and "little" to "big endian" and "little endian", so people know what the options mean.

(7) Supported input file extensions are: "pcm", "eac3", "dd+", "ddp", "ec3", "wav" and "evo".

(8) It would be nice (but not necessary) if you disabled all options which don't make sense for the input file type. E.g. the TrueHD options could be disabled if the user selects an E-AC3 input file.

These are just some comments and suggestions. You don't need to implement any of this, of course!

[otaku975]

hello
i was trying to do some conversion
but i was wondering one thing
its DTSHD to DTS
wich extension should the dtshd file be set to , and same for dts output
thanks!

[madshi]

i was trying to do some conversion
its DTSHD to DTS
eac3to currently doesn't do this conversion. I think there are other tools that do that, but you won't find them in this thread. Sorry, don't have a link at hand.

[xkodi]

eac3to currently doesn't do this conversion. I think there are other tools that do that, but you won't find them in this thread. Sorry, don't have a link at hand.

http://forum.doom9.org/showpost.php?p=967297&postcount=900

[The_Keymaker]

Thanks for the feedback Madshi.

Here is the link for the latest version of EAC3toGUI (version 0.90):

http://www.sendspace.com/file/vech6e

EAC3toGUI Version 0.90 changelog:

- Changed filter section title to "Force Filter"
- Added "endian" to big and little option labels
- Added 128kbit and 64kbit AC-3 bit rate options
- Now allows simultaneous selection of TureHD options
- Now allows simultaneous selection of PCM options
- Added logic to allow selection of common TrueHD and
PCM options (96KHz, 8 Ch) but not BOTH!

This version does not yet feature allowing specific channel orders or file type checking (i.e., PCM or TrueHD). My program can only determine filetype by extension. If this is sufficient, I will try to implement that in the next version.

Please let me if you discover any problems.

Regards,
The_Keymaker

[madshi]

Here is the link for the latest version of EAC3toGUI (version 0.90)
Looks good - thanks!!

This version does not yet feature allowing specific channel orders or file type checking (i.e., PCM or TrueHD). My program can only determine filetype by extension. If this is sufficient, I will try to implement that in the next version.
If you ask me, filetype by extension is just fine.

[Blackburst67]

I have some probleme to encode the sound of "Nine Inch Nails BESIDE YOU IN TIME".

I have the entiere HD-DVD and encode the video part, but I can't encode the sound.
The TrueHD gives me an error with eac3to

C:\Documents and Settings\Garf\Bureau\eac3>eac3to.exe "D:\55.wav" "D:\nin.ac3"
-448 -keep24
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: Signed 24-bit 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

"aften" reported error code "3221225477".

C:\Documents and Settings\Garf\Bureau\eac3>

eac3to already convert my TrueHD in EVO files and gives me an 6ch wav.

And when I demux the other track in E-AC3 and convert with eac3to, I have an eroor with Dolby who can't parsing the header of the file...

[madshi]

I have some probleme to encode the sound of "Nine Inch Nails BESIDE YOU IN TIME".
That error code in hexa is "C0000005" which means that "aften" crashed in a bad way while trying to encode the wav. Can you play that wav? Does it sound ok if you play that file? Which aften version are you using? Try a different aften version. Maybe that one you're using is buggy.

And when I demux the other track in E-AC3 and convert with eac3to, I have an eroor with Dolby who can't parsing the header of the file...
Can't say anything about that without seeing the exact error message.

[Binnugsai]

I have so some HD-DVD in my hard disk and use evodemux.

So i want to know if i have to use for the sound format when demux to "sound.evo" or sound.ddp.... which on do i use exactly !

And when i try for example to use the English sound from King Kong about 1500 k, i have some prob and a error message with Aften revision.......

So please, can someone can upload with eac3togui with inside the good tools, aften, sox etc.....

Thank you for all everybody.

Note : i try to demux one EVO, the full movie from King Kong, Movie and sound, i use VC1 and ddp for the sound and i have a file kingkong.evo, i try to use eac3togui 0.90 and failed. Nothing, no reaction about the software.

[madshi]

@Binnugsai, it's a bit difficult for me to help you without having specific erorr messages. For TrueHD tracks you should use EvoDemux to mux the TrueHD track (and only this track) into a separate EVO file. For E-AC3 tracks you should simply demux the E-AC3 tracks. The TrueHD EVO files and demuxed E-AC3 files can then be converted to AC3 or FLAC by using eac3to (or the GUI).

[Binnugsai]

Thank you Mdshi, nice to, you.

So you said me to use for TrueHD in a separate EVO in output,
feature1.evo and feature2.evo with only one sound, that's it !. and for the dd+( eac3) just demux but fir the output format, dd+ or ddp or another one.....

[foxyshadis]

very conclusive test, at least in my opinion:

1) 'Nine Inch Nails Live: Beside You in Time' bonus track, TrueHD size:

633 042 317 bytes

2) decompressed using eac3to to 16bit WAV, size: 660 487 112 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

358 807 274 bytes

3) decompressed using eac3to to 16bit WAV and then converted to 24bit using Wavewizard v0.54b,
now the least significant bits are zeroes checked with hexeditor, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

362 789 876 bytes

4) decompressed using eac3to to 24bit WAV file, size: 990 730 628 bytes
compressed back to MPL using 'SurCode MLP', MLP file size:

649 693 162 bytes

so 2) and 3) shows that if the source WAV file is not real 24bit, but 16bit padded to 24bit with zeroes
for least significant bits MLP compression is enough intelligent and that doesn't significantly increase
the size of the compressed file and that plus 3) and 4) clearly shows that the source WAV file for the
'Nine Inch Nails Live: Beside You in Time' bonus TrueHD track should be real 24bit.

p.s. soon i will do the same test for Vendetta TrueHD track.

I can't help with identification, but I do want to point out that this method is flawed: The output could be dither rather than true 24 bit, both are about equally hard to compress. (Not that I could ever tell the difference, even with my nice old amp.) Of course, it could just as easily be encoded dither as outputby the decoder, you just have to assume otherwise.

[Beastie Boy]

@Binnugsai, use EVOdemux on Feature_1.EVO with the option selected to automatically process the second EVO. This will give you a seperate file for each of the streams (video, audio, subtitles). Converting a DD+ file is the easiest, this is all I have done so far.

Using eAC3to (and the GUI if you prefer), load the relevent dd+ file as input, select and output directory and file name and click convert.
Ensure that the SOX and Aften files are all in the same directory as eAC3to.
Just a couple of things to be aware of, though they may now be fixed: eAC3toGUI loads ec3, wav & evo files, so you may want to change the file extension of your audio file. Also, I had an error message saying that my audio file couldn't be found, even though I loaded it from the GUI. I moved the file to the root of the drive and it worked OK, so I guess there is a problem with path names.

I have never done a TrueHD track, but from what I have read, you need to remux using EVOdemux and create an EVO with ONLY the audio in it. Also, you must have the Nero HD-DVD/Blu-Ray filter installed and registered.

Hope this helps. Cheers, Beastie.

[Binnugsai]

Thank you Beastie, nice. So i choose in output for the audio dd+, example kingkong.dd+ and change the extension after kingkong.eac3 or another one !.

Thanks

[Binnugsai]

In evodemux we have to choice the extension for the video, audio and subtitle. by default we have mpa for audio. That's why i ask you if i can use between .eac3 or another one to convert it after.

Thanks Bonnugsai

[Beastie Boy]

Demux, then right click the dd+ file and select Rename. Change extension to .ec3 and hit enter. Now load the file in eAC3toGUI. Type output file name (eg King_Kong.ac3), select output directory and click Convert.

Cheers, Beastie.

[Binnugsai]

Thank you so much Beastie Boy. I try and see that tonight.

See you soon. Binnugsai

[Binnugsai]

Hello, so i try to encode a dd+, Pitch.Black.HD-DVD.1080pEnglish 5.1.ec3 to test.ac3 with eac3toGui and i have that :

C:\DVD Outils HD\EAC3toGUI 0.90>eac3to.exe "C:\DVD Outils HD\EAC3toGUI 0.90\Pitc
h.Black.HD-DVD.1080pEnglish 5.1.ec3" "G:\test.ac3" -640 -nero
E-AC3, 5.1 channels, 1:51:54, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1,66 MB.
The expected file size for 24 bit is 2,49 MB.
The real file size is 0 Bytes.

C:\DVD Outils HD\EAC3toGUI 0.90>

Did i miss something, i force nero and i have Nero burning Premium in my hard disk. Did i install the bluray/Hddvd plugin in plus or not necessary !!! is there something wrong.

Thank you for your help
Binnugsai

[honai]

Does anyone have a neat command-line for converting a TruHD track (inside an EVO container) to 6 discrete mono WAVs (which will be fed into SurCode DTS) using eac3to?

[madshi]

Hello, so i try to encode a dd+, Pitch.Black.HD-DVD.1080pEnglish 5.1.ec3 to test.ac3 with eac3toGui and i have that :

C:\DVD Outils HD\EAC3toGUI 0.90>eac3to.exe "C:\DVD Outils HD\EAC3toGUI 0.90\Pitc
h.Black.HD-DVD.1080pEnglish 5.1.ec3" "G:\test.ac3" -640 -nero
E-AC3, 5.1 channels, 1:51:54, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1,66 MB.
The expected file size for 24 bit is 2,49 MB.
The real file size is 0 Bytes.

C:\DVD Outils HD\EAC3toGUI 0.90>

Did i miss something, i force nero and i have Nero burning Premium in my hard disk. Did i install the bluray/Hddvd plugin in plus or not necessary !!! is there something wrong.

Thank you for your help
Binnugsai
That seems to be the same problem other people are having with mono E-AC3 files. That's very strange. It seems that the Nero Audio Decoder sometimes doesn't output any data. I wish I could reproduce that on my PC! It works all fine for me...

[madshi]

Does anyone have a neat command-line for converting a TruHD track (inside an EVO container) to 6 discrete mono WAVs (which will be fed into SurCode DTS) using eac3to?
I guess I can add support for that to eac3to. May take a few days, though... BTW, is there a way to automate the SurCode DTS encoder?

[Beastie Boy]

@Binnugsai : I think I will have to leave it to the real experts to help with your problem. I was fortunate, it just worked.
Here's a couple of things to try:
1) Have you tried without forcing the Nero filter? Untick the Nero box.
2) Do you have the correct filters installed as per the first post in this thread?
3) Can you play the mpa file in graphEdit?

Cheers, Beastie.

[Binnugsai]

Beasty : Thanks a lot. I think if you install Nero Burning Premium you have inside Bluray and HDDVD plugin for DD+ and TruHd.
1 ) So about Untick the nero box ! where.....What do you mean.... I try yes without forcng the nero filter.
2 ) filters, i think yes. you can say me witch one!
3) I don't try to play the mpa audio with graphedit.
Perhaps that is my problem, i demux with evodemux in audio.dd+ and change as you said me in audio.ec3.

Thanks

[Beastie Boy]

1) On eAC3toGUI is a panel with check boxes labeled Nero and OrbitLee. Make sure neither are ticked.
2) The required filters are in the original post by madshi.
Nero Premium does not include the HD-DVD/Blu-ray plugin, as far as I know.

[Beastie Boy]

The filters are not freeware, so requesting them is a breech of the forum rules. Consider editing your post to avoid a strike, and I would suggest removing your e-mail address.
Sorry.

[Binnugsai]

Sorry, i will see for the filter. thank you for all.

[homerpez]

Oh, boy.. a new adventure...

I've figured out DD+ from HD-DVD (Orbitlee's filter + Sonic Audio 4.2 and it works), and I've figured out 5.1 losless PCM from Blu-Ray (Tranzcode + Wavwizard + Behappy).

Now I've finally come across a Dolby "True HD" track on Ghost Rider (Blu-Ray).

Orbitlee's filter chokes on it, and Tranzcode GUI says it's not a 6-channel track. So... I'm assuming that Eac3to GUI is the tool I might need. It looks like this movie actually uses the 7.1 channels.

I seem to be a bit confused to start with... can Eac3to GUI handle this one? I'm experimenting via trial-and-error as we speak, but hopefully someone can explain the right way (if this has been figured out).

So far, my experiments have resulted in a horrible-sounding ac3, which sounds like garbled digital static.

Aiming for the backup result to have simple 5.1 Channel AC3 audio... thanks in advance if anyone can give me a crash course in how to do this properly.

[madshi]

@homerpez, currently eac3to only supports TrueHD inside of EVO containers. But if you can send me a little chunk of your Blu-Ray M2TS file (20MB should be enough), I may be able to add support for Blu-Ray TrueHD, too. Much depends on how the Nero filters behave. For sure you'll need the Nero HD-DVD/Blu-Ray plugin. You can try right now if things work by renaming GraphEdit.exe to "Recode.exe" and then using "File Source Async -> Nero Splitter -> Nero Audio Decoder -> Dump". Does that work?

[homerpez]

@homerpez, currently eac3to only supports TrueHD inside of EVO containers. But if you can send me a little chunk of your Blu-Ray M2TS file (20MB should be enough), I may be able to add support for Blu-Ray TrueHD, too. Much depends on how the Nero filters behave. For sure you'll need the Nero HD-DVD/Blu-Ray plugin. You can try right now if things work by renaming GraphEdit.exe to "Recode.exe" and then using "File Source Async -> Nero Splitter -> Nero Audio Decoder -> Dump". Does that work?

Whoops... I'm sorry, I failed to read that part, or for that matter, realize there was any difference between HD-DVD and Blu-Ray versions of the same audio type...

I'd be happy to send you a sample, the question is, how to split up an m2ts file? (I'm quite the newbie)

I think you can somehow do this with a hex editor or something, but I might need a tutorial on how to get 20MB out of it, and then to you.

I guess I should put this backup on the back burner until this support is developed... and keep the original disc away from the kids until then :p

BTW... the method you suggested to test this from the demuxed True HD file (Filesource -> Nero Splitter) did not work... created some wild error and could not connect at step #1, to the splitter. I probably should just have you experiement with the .m2ts file, since you know what you're doing, and I don't. :thanks:

[honai]

I guess I can add support for that to eac3to. May take a few days, though... BTW, is there a way to automate the SurCode DTS encoder?

That would be great, thanks!

I don't know of any way to automate the encoder other than using generic third-party tools that script mouse events.

[ACrowley]

yeah Madshi , a Option to output 6 mono waves with the Output Order L R C LFE SL SR would be very nice

Usually is use Wavewizard to get mono channels form the wave decoded by eac3to (Nero)

@otaku975

Why do you want to reencode a DTSHD to DTS ? Every DTS HD Tracks has encoded a DTS "core" inside ,which is a "standard" DTS 1536kbps 5.1. playable withg all DTS Decoders
You can simply extract this core with dtscore.exe

No reencode

[honai]

Every DTS HD Tracks has encoded a DTS "core" inside ,which is a "standard" DTS 1536kbps 5.1.

Are you sure this is also true of HD-DVD DTS-HD tracks? I was under the impression that the compatibility core of DTS-HD is only mandatory for Blu-ray titles.

[The_Keymaker]

@honai,

I have been successful at converting TrueHD files to 6 mono waves using using eac3to and my EAC3toGUI program.

I use eac3to/EAC3toGUI to convert the TrueHD track (in an EVO) to wav file. I then use tranzcode to convert the multichannel wav file to 6 mono waves and the use surcode DTS to convert the mono waves to DTS.

Once I finish fixing bugs and adding features to EAC3toGUI, I will tackle a one click utility (using madshi's eac3to, EAC3toGUI, tranzcode and surcode DTS) to do this.

By the way on my system - Krell processor, Pass Labs amps and B&W Nautilus 802 speakers - I can easily tell the difference between high rate Dolby digital (640kbps) and DTS (1536kbps), with DTS sounding better.

since I have eschewed analog out for the time being, I can not comment on the sound quaility of TrueHD vs other similar formats.

[honai]

@The_Keymaker

Sounds really great, thanks for your efforts!

BTW nice setup there! I'm running standard consumer HW here, but over SPDIF I can also hear the difference between full-bitrate DTS and AC3@640kbps, so TruHD->DTS seems like the only way to go.

[crapz0rs]

Hello, so i try to encode a dd+, Pitch.Black.HD-DVD.1080pEnglish 5.1.ec3 to test.ac3 with eac3toGui and i have that :

C:\DVD Outils HD\EAC3toGUI 0.90>eac3to.exe "C:\DVD Outils HD\EAC3toGUI 0.90\Pitc
h.Black.HD-DVD.1080pEnglish 5.1.ec3" "G:\test.ac3" -640 -nero
E-AC3, 5.1 channels, 1:51:54, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1,66 MB.
The expected file size for 24 bit is 2,49 MB.
The real file size is 0 Bytes.

C:\DVD Outils HD\EAC3toGUI 0.90>

Did i miss something, i force nero and i have Nero burning Premium in my hard disk. Did i install the bluray/Hddvd plugin in plus or not necessary !!! is there something wrong.

Thank you for your help
BinnugsaiCorrect, the bluray/hddvd is installed by default, but you need to license it separately.
If you start "Nero ProductSetup" and goto license, you'll you probably do not have a serial installed for bluray/hddvd plugin.
It's a seperate purchase from the nero website.

[homerpez]

HEAVILY UPDATED TO REFLECT MY PROGRESS:

@homerpez, currently eac3to only supports TrueHD inside of EVO containers. But if you can send me a little chunk of your Blu-Ray M2TS file (20MB should be enough), I may be able to add support for Blu-Ray TrueHD, too. Much depends on how the Nero filters behave. For sure you'll need the Nero HD-DVD/Blu-Ray plugin. You can try right now if things work by renaming GraphEdit.exe to "Recode.exe" and then using "File Source Async -> Nero Splitter -> Nero Audio Decoder -> Dump". Does that work?

I discovered an obscure way to decode this track, and it was similar to what you'd posted here, madishi. However, I still couldn't do anything with the True HD track or LCM streams demuxed... I found that running either the TrueHD or LPCM streams through EAC3toGUI, or tranzcode, or dtscore, would result in garbled output or no data at all.

But... just for kicks, I loaded up the .m2ts stream (instead of any demuxed stream) and connected to the Nero filters that way.

Filesource(Async) "0066.m2ts" --> Nero Splitter --> Nero Audio Decoder 2 (this must be what they call it in the Vista version of Nero Ultra)

I then connected the Nero Decoder to "Directshow device" audio renderer. I could then listen, and select which audio stream I wanted to decode using the Nero Splitter (it was stream "0" which I think is the LPCM).

Once found, I disconnected the audio renderer, saved the graph file, referenced it in an AVS file, and loaded it into BeHappy.

I encoded it into 5.1 ac3 format, and it sounds great. I think even the channels are mapped correctly. :)

Unfortunately, this method didn't use EAC3to... didn't need it... but a working solution is a working solution.

Also, I didn't need to rename Graphedit to anything else... it just simply worked as described. If it's repeatable, this seems a fairly simple way of accomplishing things.

[The_Keymaker]

New version (0.96) of EAC3toGUI is now available. It can be obtained from this link:

http://www.sendspace.com/file/btfl0x

Made several changes and improvements. Change log:

0.96 - Reconfigured destination file input scheme.
- Added quote support for executable path. Should fix the
program launching problem reported by Beastie Boy.
- Implemented Source file validation by extension.
- Implemented destination file validation by extension.
- Implemented option enabling by input file type.
- Added Vertical scroll to command line output.

Please report any bugs you may find.

Regards,
The_Keymaker

[Binnugsai]

@The_Keymaker

Thank you for all, we can't download the new version of EactiGui 0.96. Wrong link !!!!!!!

[Binnugsai]

Correct, the bluray/hddvd is installed by default, but you need to license it separately.
If you start "Nero ProductSetup" and goto license, you'll you probably do not have a serial installed for bluray/hddvd plugin.
It's a seperate purchase from the nero website.


Thank you crapz0rs, very nice. everything is ok now.

Best regards.

[Beastie Boy]

The_Keymaker, many thanks for the updated GUI. I have tested it and still ge a file not found error, but it isn't caused by the pathname. By default, EVOdemux outputs the audio file with .DD+. in the file name. When this is passed to eAC3to it appears in the console as .DD>., hence the file isn't found, nor is the output file if this name also contains DD+.
I changed the text to .DDp and all worked OK.
Hope this helps.

Cheers, Beastie.

[The_Keymaker]

Is anyone besides Binnugsai having problems downloading EAC3toGUI v0.96 from the link?

I tested the link and was able to download it with no problem.

The_Keymaker

[The_Keymaker]

@BeastieBoy

Yes I believe the "+" sign is reserved by Windows/DOS for another purpose so one should avoid using this symbol in filenames. I'm not sure I can workaround this in EAC3toGUI.

I always rename the files spit out by EVOdemux to something more conventional.

Thanks for the feedback!
The_Keymaker.

[Beastie Boy]

I always rename the files spit out by EVOdemux to something more conventional.

So do I, so it's not a problem :)
The link worked fine for me.

Cheers, Beastie.

[Binnugsai]

@The_Keymaker

Ok i the link is ok and worked fine. Thanks

So, i want to know how do we use DTSCore to convert a DTS-HD audio, in command line with Dos !!!!, witch one. Don't you think that we can perhaps using Betsweet to demux into 6 chanel wav and remux with Surcode dvd.......

Thanks

[ACrowley]

Are you sure this is also true of HD-DVD DTS-HD tracks? I was under the impression that the compatibility core of DTS-HD is only mandatory for Blu-ray titles.

yes im sure on HDDVD too, DTSHD is DTSHD ,its the same

I use a lot of extracted dtscores from HDDVD ( Perfume ...)

@The_Keymaker

Ok i the link is ok and worked fine. Thanks

So, i want to know how do we use DTSCore to convert a DTS-HD audio, in command line with Dos !!!!, witch one. Don't you think that we can perhaps using Betsweet to demux into 6 chanel wav and remux with Surcode dvd.......

Thanks

Why ? Makes 100% no Sense .You will lost a lot of Quality with Surcode reencode

As i say you can extract the dtscore lossless from DTS HD ! So you have the encoded 1536kbps DTS core without Quality loss from a senseless reencode

Its simple. Start your Windows command Promt and type in

dtscore.exe dtshd.dts dtsnew.dts

Thats all to get the DTS Core from DTS HD

[Binnugsai]

yes im sure on HDDVD too, DTSHD is DTSHD ,its the same

I use a lot of extracted dtscores from HDDVD ( Perfume ...)



Why ? Makes 100% no Sense .You will lost a lot of Quality with Surcode reencode

As i say you can extract the dtscore lossless from DTS HD ! So you have the encoded 1536kbps DTS core without Quality loss from a senseless reencode

Its simple. Start your Windows command Promt and type in

dtscore.exe dtshd.dts dtsnew.dts

Thats all to get the DTS Core from DTS HD

@ACrowley

Thanks a lot. I saw at the end that is not necessary to use DTScore. I try with Rambo First Blood with DTSHD and after use Evodemux with mpa extension for the audio, i have just change the name in xxxxxx.dts and i can read it and use it to make my mkv movie. I read too with Powerdvd without problem. So

[xkodi]

@madshi

finally, i've managed to mux correctly standart MLP files (homemade, so with known format) into EVOB container.

i've made small program, that reads the EVOB blocks from the 'V for Vendetta' TrueHD EVO file and then leaves the EVOB headers, but replaces the TrueHD data with the MLP data from my homemade MLP file and it works - 'Nero Audio Decoder 2' plays the newly created EVOB with the MLP track into it.

so i made very extended test with MLP streams with known format and it shows, that:

1) 'Nero Audio Decoder 2' always uses 24bit output, even when the source MLP stream is 16bit

2) when the source MLP stream is 16bit, 'Nero Audio Decoder 2' doesn't always decode least significant byte to zero, so it doesn't decode correctly (losslessly) when the input stream is 16bit

the source code and binary of the test mux program, plus all files using during the test and detailed descriptions are in the following archive (1.8MB):

http://rapidshare.com/files/38903695/nero_mlp_test.rar.html

so everyone of you can re-done the test and review it.

i'm awaiting to hear what you guys think about the test, but seems after all, 'Nero Audio Decoder 2' has a bug.

maybe someone of you will even extend the test with more conclusive samples.

[madshi]

@xkodi, thanks very much for your test!

In the meanwhile I've finally got a reply from Nero support, asking for more details. I've send them detailed instructions on how to reproduce the problem. Hopefully they'll fix it!

[madshi]

P.S: @xkodi, are the 24bit MLP files decoded correctly? I mean are they 100% identical to the source?

[xkodi]

P.S: @xkodi, are the 24bit MLP files decoded correctly? I mean are they 100% identical to the source?

yes, the 24bit MLP files are decoded byte by byte identical to the source WAV file, but decoded 16bit MLP files are not identical to the source WAV file, they are outputed as 24bit, but not always the least significant byte is zero - exactly the same behaviour we noticed with TrueHD files.

p.s.fixed 'info.txt' from the archive is here:

http://rapidshare.com/files/38906969/info.txt.html

[madshi]

yes, the 24bit MLP files are decoded byte by byte identical to the source WAV file
That is good news! So at least the Nero decoder is perfect for 24bit MLP and most probably also for 24bit TrueHD files!

but decoded 16bit MLP files are not identical to the source WAV file, they are outputed as 24bit, but not always the least significant byte is zero - exactly the same behaviour we noticed with TrueHD files.
If you compare the original 16bit WAV source to the 24bit MLP decoder output, do you see a pattern? Are the 2 most significant bytes always 100% correct and the least significant byte contains "random" data? Or are the 2 most significant bytes not fully correct, either? If only the least significant byte is wrong and the 2 most significant bytes are correct, I could add code to eac3to to simply zero out the least significant byte and we'd be ready to go!

Thanks!

[xkodi]

If you compare the original 16bit WAV source to the 24bit MLP decoder output, do you see a pattern? Are the 2 most significant bytes always 100% correct and the least significant byte contains "random" data? Or are the 2 most significant bytes not fully correct, either? If only the least significant byte is wrong and the 2 most significant bytes are correct, I could add code to eac3to to simply zero out the least significant byte and we'd be ready to go!

Thanks!

yes, the most significant bytes of the decoded 16bit MLP are always identical to the source, so if the least significant bytes are stripped off from the decoded file, then it is identical to the source. the wrong least significant bytes usually are small numbers like 0x01, 0x03, etc.

so, yes, code to eac3to to remove the least significant byte, when the source TrueHD/MLP is 16bit, seems to be the most proper solution based on the info we now have and we can easily judge, according to the previous test with the compression ratios, which TrueHD track is 24bit and which is 16bit, but still the thing that most least significant bytes with 16bit TrueHD stream are not zero bugs me, because with 16bit MLP most least significant bytes are zero and very few of them are not zero, but my MLP test samples are small, around 100Kb, so maybe then it is normal.

[madshi]

I've just checked your files. "164851_nero_decode" and "164851_merge.wav" are 100% identical - if I strip the least significant byte from the "164851_nero_decode" file. And I've done another check: The least significant byte is *ALWAYS* zero in "164851_nero_decode".

Now I'm confused... :confused:

Where did you find a non-zero least significant byte?

[xkodi]

I've just checked your files. "164851_nero_decode" and "164851_merge.wav" are 100% identical - if I strip the least significant byte from the "164851_nero_decode" file. And I've done another check: The least significant byte is *ALWAYS* zero in "164851_nero_decode".

Now I'm confused... :confused:

Where did you find a non-zero least significant byte?

you are right, i've done so many tests and now, after i've double checked the files, i'm also confused where, when and if i really saw that or made a mistake in the hurry. that is why i made the archive, so someone else to verify my tests and conclusions, because maybe i've made mistakes, for the moment seems that the correct conclusion is that Nero always uses 24bit output and always correctly decodes MLP, so we are almost back to the very beginning with the TrueHD tracks ...

[madshi]

you are right, i've done so many tests and now, after i've double checked the files, i'm also confused where, when and if i really saw that or made a mistake in the hurry. that is why i made the archive, so someone else to verify my tests and conclusions, because maybe i've made mistakes, for the moment seems that the correct conclusion is that Nero always uses 24bit output and always correctly decodes MLP, so we are almost back to the very beginning with the TrueHD tracks ...
Agreed. Seems that MLP decoding is perfect, but TrueHD decoding probably not. Too bad that you don't have a TrueHD encoder... :)

[xkodi]

Agreed. Seems that MLP decoding is perfect, but TrueHD decoding probably not. Too bad that you don't have a TrueHD encoder... :)

maybe i should look if there are Blu-ray discs, that have both PCM track and TrueHD track, then i can put part of the TrueHD track into EVOB, the same way i put MLP into EVOB, decode and compare it with the PCM track, if i'm lucky the TrueHD should be created from the same PCM.

p.s. Ghost Rider Blu-ray disc seems to have both PCM and TrueHD tracks, so maybe in several weeks, it will be possible for me to do such test, sounds like good test.

[madshi]

maybe i should look if there are Blu-ray discs, that have both PCM track and TrueHD track, then i can put part of the TrueHD track into EVOB, the same way i put MLP into EVOB, decode and compare it with the PCM track, if i'm lucky the TrueHD should be created from the same PCM.

p.s. Ghost Rider Blu-ray disc seems to have both PCM and TrueHD tracks, so maybe in several weeks, it will be possible for me to do such test, sounds like good test.
I think those Blu-Rays with both PCM and TrueHD have 16bit PCM and 20bit TrueHD.

[Beastie Boy]

I'm currently trying to go from eAC3 to multichannel WMA whilst preserving as much quality as possible, but something isn't right.
I used eAC3to to go from my ec3 file to wav. The resulting wave file would not play using MPC, it simply crashed. Maybe I need another filter?
Anyway, undeterred I used Tranzcode to produce 6 mono wave files. These files will play but the sound is garbled, just a series of squarks. I know eAC3to works OK on my machine, because the AC3 files it produces are OK.

Any ideas where I am going wrong?

Cheers, Beastie.

PS. If this problem is not related to eAC3to I'll start a new thread so as not to go off-topic.

[The_Keymaker]

Latest version of EAC3toGUI (version 0.97)

http://www.sendspace.com/file/zdfq3x

Changelog:
- Fixed bug that would not allow *.wav files as input.

Apologize for any inconvenience this may have caused.

The_Keymaker

[ACrowley]

Thx keymaker

Great work!


Maybe its better to rename "Orbitlee" to "Sonic" ?
I mean Orbitlee DTS/AC3/DD+ Source is only the Source Filter ,the Decoder is SonicCinevisionAudioDecoder4.2

So you can select Nero=NeroAudioDecoder2 or Sonic=SonicCinevisionAudioDecoder4.2

And a .raw input extension would be good

[madshi]

I'm currently trying to go from eAC3 to multichannel WMA whilst preserving as much quality as possible, but something isn't right.
I used eAC3to to go from my ec3 file to wav. The resulting wave file would not play using MPC, it simply crashed. Maybe I need another filter?
Anyway, undeterred I used Tranzcode to produce 6 mono wave files. These files will play but the sound is garbled, just a series of squarks. I know eAC3to works OK on my machine, because the AC3 files it produces are OK.

Any ideas where I am going wrong?

Cheers, Beastie.

PS. If this problem is not related to eAC3to I'll start a new thread so as not to go off-topic.
Not sure what the cause is if the AC3 files are correct. Maybe you just need to wait until I've added mono wave output support to eac3to.

[Beastie Boy]

Maybe you just need to wait until I've added mono wave output support to eac3to.

I was hoping you would say that :p :thanks:

Cheers, Beastie.

[ACrowley]

Damn. i tried everything
I cant get anything working from a 24bit lpcm from Bluray

I tried to demux with SonicHDDemuxer, also with xport.

I use this Sox commadline and tried it as little endian or big endian, with -2 and -3
sox.exe -l -r48000 -t .raw -c 6 -3 -s e:\test.raw e:\test.wav

nothing! When i use eac3to do it autom. its the same Problem

The wave File makes only a heavy loud noise, 6 mono channels ( created with wavewizard or tranzcode) to.

The full lpcm 24bit is 7Gb big.
The encoded flac from eac3to is "6Gb" and playback fails.
Its wrong cause the flac must be around 3Gb.
And AC3/ wave output only the loud noise too.
A 24-16bit conversion create the same Result

When i try it on a 100Mb sample its the same

It happens with "all" 24bit lpcm Tracks from Bluray.
I really tried everything, checked it it again and again , with no succes

LPCM 16 bit works without Problems.

I really dont know where the Problem ? Any advice ?

I could only image its a Problem caused by demuxing.

Heres the same Problem, but no Solution so far i can see ?
http://forum.doom9.org/showthread.php?t=124080&page=2

[infernix]

I have a feature request.

Currently I'm running eac3to on a windows virtual machine in which it stores everything (source + raw + wav + ac3/flac) to a network share. For multiple reasons, I'd like to proceed on linux after the raw file has been dumped.

Seeing that most the tools used (all but the directsource stuff?) are available there, I'd like a simple parameter (like -n for make) that spits out the commands it would run after the directshow dump to raw, so that I can run sox and aften in linux. Thats basically everything after:
E-AC3, 5.1 channels, 1:59:02, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
Audio Decoder output was 24 bit this time.


Yes, I could just dump to raw myself, then do strings eac3to.exe and figure out the right commands, but i'm lazy, and it's probably really simple to add :)

[madshi]

Damn. i tried everything
I cant get anything working from a 24bit lpcm from Bluray
With the SonicHDDemuxer you'll probably get little endian instead of big endian. You may need to adjust the parameters of the tools you're using. E.g. when using eac3to, add "-little" as a test.

[madshi]

I'd like a simple parameter (like -n for make) that spits out the commands it would run after the directshow dump to raw
Ok, you're the 2nd person asking me about this. So I guess I should implement it.

[ACrowley]

With the SonicHDDemuxer you'll probably get little endian instead of big endian. You may need to adjust the parameters of the tools you're using. E.g. when using eac3to, add "-little" as a test.



EDIT :
I have it! Very strange cause i tried all theye combinations before...mhh

Problem : even the 24bit lpcm was demxued with graphedit, it works only with Big Endian Switch in Sox

1 .sox.exe -B -r48000 -t .raw -c 6 -3 -s e:\test.raw e:\test.wav
2. wavewizard 0 1 2 5 3 4
3. encode to whatever

@Madshi
Ofcourse i tried all Combinations mayn Times before.

I can say :eac3to fails (for me)on these 24bit lpcm , But it works manually .
Even with Big or Little Endian switch, nothing! No working wave, no flac,no ac3. outputs is a loud noise.

eac3to e:\test.pcm e:\lpcm.wav -blu-ray -24 ( i tried with -B / -L all combinations)

Heres the sample. Its LPCM 24Bit 5.1 from BluRay "Apocalypto"

http://rapidshare.com/files/39409043/LPCM_24Bit.rar.html

[kweek]

First thanks for this great prog :)

I don't know if this is a bug or if I'm doing something wrong but I discovered this because some one said that you can create smaller flacs with the -8 option (default is -5 see http://flac.sourceforge.net/documentation_tools_flac.html#encoding_options).

When I use the following command
"eac3to.exe input.ddp output.flac -nero"
and I want to compress the flac some more with the command
"flac output.flac -o output.reenc.flac -f -8".

I get the following error
"output.flac: ERROR: FLAC input has STREAMINFO with unknown total samples which is not supported"

With "metaflac.exe output.flac --list" I get this.
METADATA block #0
type: 0 (STREAMINFO)
is last: false
length: 34
minimum blocksize: 4096 samples
maximum blocksize: 4096 samples
minimum framesize: 0 bytes
maximum framesize: 0 bytes
sample_rate: 48000 Hz
channels: 6
bits-per-sample: 24
total samples: 0
MD5 signature: 00000000000000000000000000000000
METADATA block #1
type: 4 (VORBIS_COMMENT)
is last: false
length: 40
vendor string: reference libFLAC 1.1.4 20070213
comments: 0
METADATA block #2
type: 1 (PADDING)
is last: true
length: 8192

Notice that streaminfo is corrupt. No minimum/maximum framesize, no total samples and no MD5 signature.

Now that someone who told me about the -8 option also gave me some pointers to fix this. Remember the sample_rate, channels and bits-per-sample.

Create a raw file from the corrupt flac.
"flac -d output.flac -o temp.raw --force-raw-format --endian=little --sign=signed"

Next create the flac from the raw with the information from metaflac (with the -8 option if you want)
"flac -8 --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed -o "output.fixed.flac" - < temp.raw"

Now you can see that when you use metaflac on the new file it has the correct streaminfo, but I think eac3to should set the correct streaminfo to begin with :).
METADATA block #0
type: 0 (STREAMINFO)
is last: false
length: 34
minimum blocksize: 4096 samples
maximum blocksize: 4096 samples
minimum framesize: 417 bytes
maximum framesize: 42568 bytes
sample_rate: 48000 Hz
channels: 6
bits-per-sample: 24
total samples: 310497536
MD5 signature: c5fd63920d96db81ec780bacb00ca4eb
METADATA block #1
type: 4 (VORBIS_COMMENT)
is last: false
length: 40
vendor string: reference libFLAC 1.1.4 20070213
comments: 0
METADATA block #2
type: 1 (PADDING)
is last: true
length: 8192

[ACrowley]

First thanks for this great prog :)

I don't know if this is a bug or if I'm doing something wrong but I discovered this because some one said that you can create smaller flacs with the -8 option (default is -5 see http://flac.sourceforge.net/documentation_tools_flac.html#encoding_options).

[/CODE]

-8 is maximum quality with maximum compression ,but slower encoding.
-5 should output a liottle bit bigger File with less compression and lower Quality

eac3to use -8 too, i would only use 8 for "HD Audio"

[kweek]

-8 is maximum quality with maximum compression ,but slower encoding.
-5 should output a liottle bit bigger File with less compression and lower Quality

eac3to use -8 too, i would only use 8 for "HD Audio"

Hi, thanks for you response. But the post was more about that eac3to doesn't set the streaminfo right.

Do you mean that eac3to uses the -8 option for flac? I'm using eac3to mainly for highdef audio.

[ACrowley]

Hi, thanks for you response. But the post was more about that eac3to doesn't set the streaminfo right.

Do you mean that eac3to uses the -8 option for flac? I'm using eac3to mainly for highdef audio.


yes eac3to should use -8
The flac output has the same Size compared to a media coder -8 output.

################


ABout Sonic Cinevision 2 beta :

There a a few new Decoder Versions. Located in
C:\xxSonic/Cinevision/Filters

1. Video Decoder (I think based on Elecard/Moonlight) all are .ax Files and you can reg/use them all
-Sonic AVC/H264 Decoder
-Sonic VC1 Decoder
-Sonic MPEG Demuxer
-no AudioDecoder

2. The Cinemaster HD 4.3 Decoder ,all are .dll Files
-SonicHDDemxuer 4.3
-SonicCinemsterVideoDecoder 4.3
-SonicCinemsterÁudioDecoder 4.3

The .ax Filter are working very good on BluRay H264/VC1 Video.
No Artefacts on H264 PAFF etc.

More interesting for this Thread is ,that the 4.3 CinemasterAudioDecoder connects to TrueHD!
The TrueHD is playing ,but you can hear a noise in the backround. But basicly it should work now.
I Tried it on Matrix HJDDVD TrueHD 24bit

The other Thing is the Cinevision 2 has no installation fo the Cinemaster 4.2 Decoder. When i try to reg the manually it fails with a Error Message.
You can overwrite the 4.2 dlls with the new 4.3. But Avisynth wont connect anymore.
So, it would be nice if somebody can find out a workaround how to properly reg. the Cinemaster HD 4.3 Decoder

[The_Keymaker]

New version of EAC3toGUI (version 1.01). EDIT: Version 1.00 had a channel mapping bug! The following link is for the latest bug free version

http://www.sendspace.com/file/pkz6ew

There are extensive changes and and improvements for this latest release. Some that can be noticed are:

- Changed to TABBED GUI interface.
- Added custom channel order capability w/error checking.
- Added a channel order preview window.
- Added additional (32kbit) data rate.
- Renamed "orbitlee" filter to "Sonic-Orbitlee".
- Fixed a bug that disabled destination file input when source file changed.

Remember, this is just a GUI. Fot it to work, you must download and install Madshi's eac3to.exe program as well as the others listed in the first post of this thread. Just copy EAC3toGUI to the same directory as eac3to.exe and run (if you decide to place EAC3toGUI in a different directory, please set the location of eac3to.exe using the "Settings" menu item in EAC3toGUI).

Please feel free to post bugs, improvement suggestions or usage experience.

Regards,

[xkodi]

@madshi
new Sonic decoders version 4.3 are capable to decode raw MLP/TrueHD stream, so they connect to both EVOB or raw MLP/TrueHD stream.

i made the same test i made before with Nero:

24fake4851_nero_decode and 24fake4851_sonic_decode are identical

244851_nero_decode and 244851_sonic_decode are indentical

but 164851_nero_decode and 164851_sonic_decode are not, Nero output is 24bit with least significant byte zero, but Sonic properly detects, that the stream is 16bit and outputs only 16bit:

http://img72.imageshack.us/img72/4364/sonic16bitju1.png (http://imageshack.us)

and the result from the test with TrueHD:

Vendetta TrueHD stream, Sonic output:
http://img165.imageshack.us/img165/6774/sonicvfvthdgc4.png (http://imageshack.us)

Vendetta TrueHD stream, Nero output:
http://img72.imageshack.us/img72/5306/nerovfvthddl4.png (http://imageshack.us)

so Sonic decode of Vendetta TrueHD stream has least significant byte of every sample set to zero, but when listen to the decoded stream it has some kind of noise, so seems it's faulty.

Nine Inch Nails: BYIT' bonus track TrueHD, Sonic output:
http://img72.imageshack.us/img72/5986/ninsonickb4.png (http://imageshack.us)

Nine Inch Nails: BYIT' bonus track TrueHD, Nero output:
http://img72.imageshack.us/img72/4136/ninnerooo3.png (http://imageshack.us)

with NIN TrueHD track there are some different bytes, but most of the bytes are identical, from previous tests we know, that NIN TrueHD is probably 24bit, so seems that with 24 Sonic performs almost perfect, again there is some noise in the sound.

so both Nero and Sonic are perfect with MLP, Sonic is better, because has 16bit output, when MLP stream is 16bit, plus can decode raw MLP.

when it comes to TrueHD, like Nero, Sonic always reports the TrueHD stream as 24bit, but when TrueHD stream is 16bit (Vendetta) seems Sonic has faulty decode, because there is noise, when TrueHD stream is 24bit (NIN bonus track) again there is noise, but as bytes the output is almost identical to Nero.

[ACrowley]

@xkodi have you properly reg the 4.3 Decoder dll , or only overwritten the 4.2 Decoder ?
Cause i cvant reg the 4.3 dll.

When i overwrite the old 4.2 ( from Oct.2006 new dll are from march 2007) i can use teh Decoder in graphedit or for playback.
But avisyth cant use them with a incorrect Pin Error Message

IS there DRC on TrueHD from Sonic? So i still prefer Nero cause i know surely no DRC will be appilied on anything

But im not sure if theres any DRC in a MLP decoded Stream

I have a noise in backround on the EVO truehh of Matrix with Sonic

@madhsi and keymaker

A FLAC output Function would be nice. I mean output 24bit to 16Bit flac
And mono wave would be nice too

[xkodi]

@xkodi have you properly reg the 4.3 Decoder dll , or only overwritten the 4.2 Decoder ?
Cause i cvant reg the 4.3 dll.

When i overwrite the old 4.2 ( from Oct.2006 new dll are from march 2007) i can use teh Decoder in graphedit or for playback.
But avisyth cant use them with a incorrect Pin Error Message


i overwrote the old 4.2 DLLs with the new 4.3 ones, because trying to regsvr32 them doesn't work and after that - no problems using them with GraphEdit without any kind of error messages, i don't know for Avisynth, because i haven't tried that yet, but probably it won't work and give the same error message like you said.


IS there DRC on TrueHD from Sonic? So i still prefer Nero cause i know surely no DRC will be appilied on anything

But im not sure if theres any DRC in a MLP decoded Stream

I have a noise in backround on the EVO truehh of Matrix with Sonic

at least with MLP stream the output of Sonic is byte by byte identical to the original PCM source from which the stream is created, so nothing is additionally applied, with TrueHD stream, i don't know, i also have noise and i'm still not sure what is the cause, because in hex editor the output of Sonic filter seems much more proper than Nero - every least significant byte of the Vendetta TrueHD track decode is zero as it should be, because my previous test with MLP compression ratios shows that Vendetta should be made from 16bit PCM source and with the NIN TrueHD track, which should be made from 24bit PCM source, the Sonic output is almost identical to Nero, very few bytes are different, but there is still noise

[ACrowley]

Maybe ist possible to tweak the Registry Settings from Sonic Audio Decoder 4.2, to get clean TrueHD Output c

There are a lot of more Options as in System Control Panel

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\AudioDecoder

[xkodi]

Maybe ist possible to tweak the Registry Settings from Sonic Audio Decoder 4.2, to get clean TrueHD Output c

There are a lot of more Options as in System Control Panel

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\AudioDecoder

according to CineVision help:

The Audio track allows you to select an audio file for playback with the encoded video during preview. You can select the following types of audio files:

PCM (.WAV)
DTS (.dts, .cpt)
DTS-HD (all modes)
Dolby Digital (.AC3)
Dolby Digital +
Dolby Tru-HD

but i don't know how to open file for preview using CineVision and need to read more of the help, but if Tru-HD file plays fine in CineVision, there is hope, that it can be made to work correctly outside CineVision, because these filters are beta and maybe simply the TrueHD decode is broken, so first we should learn how to play TrueHD using CineVision and then if the playback is OK search for solution, how to use the filters with GraphEdit.

[ACrowley]

yes, youre right

With 4.3 and new HDDemuxer i can start/play a single trueHD (.mlp) in zoomplayer.
But also with the backround noise and crazy runtime (few thousand h)

Is there any chance to output SPDif as AC3 from DD+.TrueHD etc ?

I tried some registry switches and i got the Audiooutput Option " DolbyDigital" avaible in the Sonic Audio System Control Panel.
I cant rember which Registry switch it was.
Sonic should be able to output HD Audio as AC3 ,or not ? i m not sure..

However, i use AC3 Filter behind NeroAudioDecoder2 , works perfect

[madshi]

Hi, thanks for you response. But the post was more about that eac3to doesn't set the streaminfo right.
I'm aware of that problem - and it can't be fixed right now. Why? Because I'm intentionally calling flac.exe in a way that produces the un-initialized header info.

:eek:

I have to do that because flac.exe can't cope with >2GB output files otherwise. Try your "fix" on a flac file which is bigger than 2GB and you'll see that it won't work. Flac.exe just can't handle such big flac files (in Windows only), unless you run it in a special mode. The special mode results in the flac header not being fully initialized, unfortunately.

[madshi]

@xkodi & ACrowley, that's nice that Sonic's newest beta filters handle MLP and TrueHD! Too bad, though, that they are buggy with TrueHD decoding right now... :( Hopefully they'll fix that in the final version.

@xkodi, it seems that we finally have a final confirmation about that the Vendetta track is really only 16bit and the NIN one really more than 16bit. It's good that we have this information confirmed!! Now we just need a bug free TrueHD decoder, of course.

Have you guys tried "Sonic HD Demuxer -> Nero Audio Decoder" yet? Of course that's no full solution right now, either, but at least it might allow direct playback of TrueHD tracks without background noise right now? (Haven't had a chance yet to check out the beta Sonic filters myself yet).

[ACrowley]

@madshi
No NeroAudioDecoder cant connect with SonicHDDemuxer

[madshi]

No NeroAudioDecoder cant connect with SonicHDDemuxer
Too bad!!

[ACrowley]

edit. SRY

NeroAudioDecoder2 "can" connect to SonicHDDemuxer 4.3.
But only to the EVO. not to the single mlp.
But when i select the TrueHD the graph fails

I made my own encodes in MLP 16bit 5.1
Playback in Zoomplayer with SonicHDDemuxer-SonicCinemasterAudioDecoder4.2 (4.3) works pervect without backround noise
In othe rDShow Players too.

But SonicHDDemxuer must have Propbmes cause the Runtime is always wrong. Must be 1min39sec ,but Runtime is 18h23min .
Runtime from SupermanReturns TrueHD in Zoomplayer is thousand of hours. Must be caused by Sonic
But Playback is clean

[xkodi]

@madshi and @ACrowley

do you know how to distinct DirectShow filter from DMO? i'm asking this, because i think that Sonic decoder 4.3 is DMO and it's not DirectShow filter, because with both MLP and TrueHD there are a lot of "discontinuities" :

http://img528.imageshack.us/img528/8274/sonicdiscos3.png (http://imageshack.us)

and in text on msdn.microsoft.com i've read something like:

When a DMO, which was not designed to work in DirectShow, receives the DISCONTINUITY flag, it performs a flush and loses the data stored. To work around this issue, the best solution is probably to write a custom DMO Wrapper filter that does not set the DISCONTINUITY flag.

so maybe Sonic 4.3 is DMO and some buffer are flushed and thus from time to time random values are send on the filter output and that results in noise, it affects only the TrueHD implementation, because there are a lot of "discontinuities" when playing MLP too, but the sound is normal.

i don't know if there are any DMO wrapper DirectShow filters available, that can be tried with Sonic 4.3 filter or how to know for sure that it is DMO ...

[EDIT] using 'InterVideo Container DMO' connected to the output of Sonic audio 4.3 fixed the "discontinuities" errors, but sound still has noise ...

[madshi]

using 'InterVideo Container DMO' connected to the output of Sonic audio 4.3 fixed the "discontinuities" errors, but sound still has noise ...
Too bad. It sounded promising... :(

[madshi]

eac3to 1.9 released:

http://madshi.net/eac3to.zip

Most important new features:
(1) multi channel mono wav output added
(2) automated SurCode DTS encoding added

Full list of changes:
(1) 24bit PCM handling works now (was buggy before)
(2) "-blu-ray" option removed
(3) with PCM input files "bigendian" is default now
(4) with 5.1 PCM input blu-ray style channel remapping is default now
(5) switches "-16" and "-24" are valid for both TrueHD and PCM input now
(6) eac3to now creates the WAV files on its own instead of using sox
(7) target extension ".wavs" results in one mono wav for each channel being created
(8) SurCode DVD DTS encoding automation added
(9) new options "-768" and "-1536" for DTS encoding
(10) TrueHD output is not downconverted to 16bit by default, anymore.
(11) new option "-down16" downconverts the raw data from 24 -> 16 bit (not limited to TrueHD input)
Due to the many changes I did to the code, some new bugs may appear. Please let me know about any problems you run into.

For Blu-Ray PCM you can now get multiple mono wavs or a DTS encoding simply by doing:

"eac3to source.pcm dest.wavs"
"eac3to source.pcm dest.dts"

No additional parameters are needed, unless eac3to fails to automatically detect the bitdepth. In that case you need to specify "-16" or "-24".

My apologies to The_Keymaker, cause he has to redo the option checkboxes so often...

[madshi]

I can say :eac3to fails (for me)on these 24bit lpcm
Fixed in eac3to 1.9. Thanks for the sample!

[honai]

@madshi

Great work, thank you so much for filling that request! Your dedication is very much appreciated!

[Beastie Boy]

Many thanks madshi, the mono wave output is just what I need.
The_Keymaker, over to you... :) :thanks:

Cheers, Beastie.

[ACrowley]

Fixed in eac3to 1.9. Thanks for the sample!


nice madhsi...i will try it:)

Otherwise i found a 100% working Method for any LPCM24 bit to whatever

Mono wave output is what i need so i can feed Steinberg Nuendo without a extra Step

great work!
@xskodi i dont know a DMO wrapper...
Yes, all my tested mlp sounds normal in any dshow Player. But in TrueHD theres still noise with Sonic 4.3 Decoder
However, Nero works so far

[crapz0rs]

I have surcode 1.0.2.3 installed, but the dts creating doesn't work.
Audio Decoder output was 24 bit this time.
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Creating the DTS file failed. The multi channel wav files are valid, though.
Encoding the 6 mono wavs in surcode works fine though :-)

[madshi]

I have surcode 1.0.2.3 installed, but the dts creating doesn't work.
Audio Decoder output was 24 bit this time.
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Creating the DTS file failed. The multi channel wav files are valid, though.
Encoding the 6 mono wavs in surcode works fine though :-)
Argh, my fault. I'll upload a newer build in a few minutes.

[madshi]

New version uploaded, same link as before. Please check the version information (right click on eac3to.exe -> properties). It must be "1.9.0.2". Hopefully DTS encoding works now.

[crapz0rs]

Audio Decoder output was 24 bit this time.
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Done. :thanks:

[ACrowley]

@madshi

LPCM 24Bit to wave works now! nice...

But the flac Problem still exits for me

24Bit LPCM to 24Bit flac results in a flac which has the same Size as the raw! Strange...
On the Sample both 24Bit Files are 90MB
On a 6.7GB LPCM the FLAc from eac3to is 6.7GB too

I tried all combinations again on the same sample
So i use Audacity 1.3.3 to encode to flac 24bit

2 Questions:

1. Are you using Besiled 0.3 at the end of the dts chain to fix dts output? I think so...
Maybe you can remember, we talked about via PN

eact3to dts encode works with all 16 and 24bit Sources

2. Whne i convert a lpcm.raw to wave with -Bluray, the Message "remapping channes" will not longer appear ?
Output wave has correct channelmapping

[honai]

Besliced 0.3 at the end of the dts chain to fix dts output

What exactly is happening at this step? How is the DTS stream from Surcode erroneous?

[madshi]

But the flac Problem still exits for me

24Bit LPCM to 24Bit flac results in a flac which has the same Size as the raw! Strange...
On the Sample both 24Bit Files are 90MB
On a 6.7GB LPCM the FLAc from eac3to is 6.7GB too
I'll check that. Maybe I'll switch to the flake encoder. Maybe that won't have problems with > 2GB output. Don't know...

1. Are you using Besiled 0.3 at the end of the dts chain to fix dts output?
No, I found a better way... :) There's a registry value which tells SurCode to give out a more compatible DTS stream (named "PAD ZEROS"). Unfortunately SurCode keeps on resetting this registry value to the "bad" setting all the time, so it's very uncomfortable for manual use. Of course eac3to solves all these troubles for you.

eact3to dts encode works with all 16 and 24bit Sources
I had to replace sox to make that work. SurCode didn't like the 24bit WAV files produced by sox. So now eac3to doesn't use sox for WAV file creation, anymore, but does that work itself. The WAV files created by eac3to should be similar to what wavwizard outputs. They seem to be more compatible then what sox creates.

2. Whne i convert a lpcm.raw to wave with -Bluray, the Message "remapping channes" will not longer appear ?
Output wave has correct channelmapping
The message doesn't appear, anymore, because I'm now doing that step together with some other steps like creating mono wavs etc. In the older eac3to build the channel remapping was a separate step. In the new build eac3to is sometimes doing multiple steps at once to save time. You may have noticed (or maybe not) that the LPCM -> mono WAV conversion is rather quick. It should be 2x-3x as fast as before.

[madshi]

What exactly is happening at this step? How is the DTS stream from Surcode erroneous?
Older versions of SurCode (the latest version 1.0.29 doesn't have this problem) by default added some zero bytes after each DTS frame. A normal DTS frame for 1536kbit/s is 2013 bytes long. SurCode by default writes 2013 real bytes and then 35 zero bytes for every frame. Don't ask me why they're doing that, I've no idea. delaycut, besplit and most other software doesn't like this very much. Fortunately, I've found a way to convince SurCode to stop adding those 35 zero bytes after each DTS frame and added that to eac3to.

[ACrowley]

Madshi is right
With a Surcode encoded dts you will get these Errors in the Header and mostly wrong Runtime/Filesize and this Error in mkvmerge :

audio/video go out of sync, but this stream is damaged.
Warning: dts_packetizer: skipping 35 bytes (no valid DTS header found).
This might make audio/video go out of sync, but this stream is damaged.

To fix the dts you can also use Besliced.exe (0.3)

Besliced 0.3 exe included in :
http://dspguru.notrace.dk/BeSplitv0.9b6.zip

Simply open Besliced.exe and move your dts into the window and select fix.
You will get a lot of Errors in the Log like this :

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] | Input : G:\test.dts
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Channels Count: 5.1, Bitrate: 1536kbps
[00:00:00:000] | Output : G:\test_Fixed.dts
[00:00:00:000] +---------------------
[00:00:00:021] Stream error : Sync found after 35 bytes
[00:00:00:032] Stream error : Sync found after 35 bytes
[00:00:00:042] Stream error : Sync found after 35 bytes
[00:00:00:053] Stream error : Sync found after 35 bytes
[00:00:00:063] Stream error : Sync found after 35 bytes
[00:00:00:074] Stream error : Sync found after 35 bytes
[00:00:00:085] Stream error : Sync found after 35 bytes
[00:00:00:096] Stream error : Sync found after 35 bytes
[00:00:00:106] Stream error : Sync found after 35 bytes
[00:00:00:117] Stream error : Sync found after 35 bytes
[00:00:00:128] Stream error : Sync found after 35 bytes
[00:00:00:138] Stream error : Sync found after 35 bytes
[00:00:00:149] Stream error : Sync found after 35 bytes
[00:00:00:159] Stream error : Sync found after 35 bytes
[00:00:00:170] Stream error : Sync found after 35 bytes
[00:00:00:181] Stream error : Sync found after 35 bytes
....

After fixing the dts is clean ,with correct Runtime/Filesize and no more Problems and you have a proper nice dts

###############################################

@Madshi
1 .The Surcode Registry Key "PAD ZEROS"). it should be set to 0 ,right ?

2. When i use -down16 on the sample, the flac output still has the loud noise
.wav output is ok (in 16 and 24bit)
And -flac in 24bit ,has the Filesize Problem i told you above

3.Flake encoder would be nice. Its faster und higher -12 compression
But :
At the Moment you get only 16bit flac from flake encoder :
Signed 24-bit 48000 Hz 6-channel
WARNING! converting to 16-bit (not lossless)

http://forum.doom9.org/showthread.php?t=120871

Hi,
I wrote Flake. 24-bit encoding is on the TODO list, but right now does downsample to 16-bit. As far as Winamp crashing...that's really odd. It should not be doing that, especially with the settings you used, which do not utilize any of Flake's experimental stuff. My guess is that the issue is Winamp since Foobar decodes okay. But then again, Flake has not been through very vigorous testing at early this stage in its existance.

Anyway, it seems that your only choice for now to encode 24-bit is the libFLAC encoder...I just wanted to chime in and say that I'm working on adding it to Flake sometime in the future.
-Justin

[madshi]

1 .The Surcode Registry Key "PAD ZEROS"). it should be set to 0 ,right ?
Right.

2. When i use -down16 on the sample, the flac output still has the loud noise
.wav output is ok (in 16 and 24bit)
And -flac in 24bit ,has the Filesize Problem i told you above

3.Flake encoder would be nice. Its faster und higher -12 compression
But :
At the Moment you get only 16bit flac from flake encoder
Argh, that's bad.

[The_Keymaker]

New version (v1.15) of EAC3toGUI available here:

http://www.sendspace.com/file/ltxl6s

Changelog Version 1.15:
- Added DTS rate options
- Added Down convert option "-down16"
- Added "wavs" and "dts" filemasks
- Changed PCM and TrueHD options scheme to allow selection of common options (16bit, 24bit, 96Khz, 8ch)

Please report bugs, suggestions and overall usage experience.

Regards,
The_Keymaker

[honai]

A normal DTS frame for 1536kbit/s is 2013 bytes long. SurCode by default writes 2013 real bytes and then 35 zero bytes for every frame. Don't ask me why they're doing that, I've no idea.

Probably to get to the "magic" 2048 bytes size. I'm not sure but I think that DTS over SPDIF gets wrapped/padded in such a way, too, though transparently by the soundcard driver or AC3Filter.

[ACrowley]

@The_Keymaker

BluRay channelmapping Switch is removed cause eac3to will do it autom. now ,right ?

Little Suggestions

1. I know in your opinion (youre right) its enough to type in the dest. File extension to set Output Format.
But ,imho, a small scrolldown Outputformat Window would be nice

2. The Downconvert Option would be more obvious with "Downconvert to 16Bit"

3. Maybe a little Error ? When you select "Browse" to set the Output, the Windows Name is "Select the Source File" , shouldnt it be "Save as" ?
And Filetyp is (ac3,raw,wav), Theres no flac ( ac3,raw,wav,flac)

4. And the Input Extension .raw would be nice

Ofcourse its all marginal, your Gui works great! THX

greets

[The_Keymaker]

New version (version 1.18) of EAC3toGUI released. Download link:

http://www.sendspace.com/file/ruv4ub

Changelog:

1. Added *.raw source file extension mask
2. Added *.flac destination extension mask
3. Relabeled down convert option for clarification
4. Relabeled destination file browse dialog

Thanks to ACrowley for feedback and suggestions.

As usual, please provide feedback, bug reports and usage experience.

Regards,
The_Keymaker

[ACrowley]

THX..great work!

But to 1:

.raw apppears in the mask ,When i load a .raw i get a ErrorMessage about a not valid Extension.

And is it possible to add a Proggress Bar (%) or something like that ?

[Prophecy650]

don't know what I am doing wrong, says

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

D:\BREAKUP\HVDVD_TS>"C:\Documents and Settings\Ivan\Desktop\New Folder (3)\eac3t
o.exe" "D:\BREAKUP\HVDVD_TS\EXTENDED.EVO" "D:\BREAKUP\HVDVD_TS\EXTENDED.ac3" -
448 -24
Muxing evo's first audio track to raw. Please wait...
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: Signed 24-bit 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

"aften" reported error code "3221225477". A valid *.wav file was created sucessf
ully, though.

D:\BREAKUP\HVDVD_TS>

When I got to the wav file everything is sped up. the audio sounds like its beeing fastfoward?

[madshi]

@Prophecy650,

as far as I can see from the "The Break-Up" HD DVD specs, this HD DVD doesn't have a TrueHD track, but only some E-AC3 tracks. So you may want to try demuxing the E-AC3 track by using EvoDemux and then feed that into eac3to. Maybe that works better?

[Prophecy650]

thanks trying it right now

[Prophecy650]

i went to evodemux changed it to eac3 not Im getting this

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

C:\Documents and Settings\Ivan\Desktop\New Folder>"C:\Documents and Settings\Iva
n\Desktop\New Folder (3)\eac3to.exe" "C:\Documents and Settings\Ivan\Desktop\New
Folder\FEATURE_1_FEATURE_2.DD>stream.00.eac3" "C:\Documents and Settings\Ivan\
Desktop\New Folder\FEATURE_1_FEATURE_2.DD>stream.00.ac3"
Source file "C:\Documents and Settings\Ivan\Desktop\New Folder\FEATURE_1_FEATURE
_2.DD>stream.00.eac3" not found.

[madshi]

Please rename the file to something else. My XP doesn't even allow ">" inside of a file name. Don't know how you got such a character in your file name.

[Prophecy650]

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

C:\Documents and Settings\Ivan\Desktop\New Folder>"C:\Documents and Settings\Iva
n\Desktop\New Folder (3)\eac3to.exe" "C:\Documents and Settings\Ivan\Desktop\New
Folder\FEATURE.eac3" "C:\Documents and Settings\Ivan\Desktop\New Folder\FEATUR
E.ac3" -448 -nero
E-AC3, 5.1 channels, 1:46:31, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
Audio Decoder output was 24 bit this time.
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: Signed 24-bit 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

"aften" reported error code "3221225477". A valid *.wav file was created sucessf
ully, though.


how do I get it to ac3 now? thanks alot madshi, I can hear the audio atleast now

[madshi]

Looks like an Aften crash to me. Can you please try a different Aften build? I'm using one from here:

http://win32builds.sourceforge.net/aften/index.html

Don't remember which one. Might be revision 449.

[Prophecy650]

the file is 5gb+ and the audio only like 23 min?

[madshi]

Which file is 5GB? The raw/wav file can be that big. But don't worry, the final AC3 file will be much much smaller. The raw/wav files are just temporary intermediate files. They're so big because they're totally uncompressed. AC3 is a strong compressing algorithm.

[Prophecy650]

madshi I love you lol, got it to work with the 449 like u said, thanks a million

[starkhouse33]

whats the difference between using aften builds VERSION 0.07 and SVN REVISION 449.

[madshi]

whats the difference between using aften builds VERSION 0.07 and SVN REVISION 449.
I've no idea.

[MichalHabart]

Hello madshi.
Is it possible to add parameter for working directory? Because it is not best idea to create mono wavs in same directory where source eac3/pcm file is located.
Thanks in advance

[tebasuna51]

whats the difference between using aften builds VERSION 0.07 and SVN REVISION 449.

The rev449 is a intermediate version between v0.06 (with problems with big files) and v0.07 (rev493).

I don't know problems with Wisodev v0.07 compiles, please report if the problems related in this thread was with Aften v0.07 or a previous one.

With rev449 some of the changes in v0.07 (http://forum.doom9.org/showthread.php?p=993727#post993727) may not be present.

Edit: The rev449 is the first using the parameter -readtoeof 1 to support wav files >4GB. See this post (http://forum.doom9.org/showthread.php?p=961856#post961856) and following.

[madshi]

it is not best idea to create mono wavs in same directory where source eac3/pcm file is located.
Why is that?

[MichalHabart]

Why is that?

Because it slows encoding down :)

I have also same problem posted here several days ago:

eac3to.exe "D:\VOBs\Ghost Rider\bits0001.pcm" h:\ghost.dts -1536
RAW, 5.1 channels, 2:03:25, 16 bits, 48khz
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Creating the DTS file failed. The multi channel wav files are valid, though.

I checked i have Surcode 1.0.23 properly installed and version of eac3to tool is 1.9.0.2 :(

Can you please help me find what is wrong then?

[madshi]

Because it slows encoding down :)
Argh, so you want the temp files to be on another drive?

eac3to.exe "D:\VOBs\Ghost Rider\bits0001.pcm" h:\ghost.dts -1536
RAW, 5.1 channels, 2:03:25, 16 bits, 48khz
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Creating the DTS file failed. The multi channel wav files are valid, though.

I checked i have Surcode 1.0.23 properly installed and version of eac3to tool is 1.9.0.2 :(
Does DTS encoding work if you put the mono wav files manually into SurCode?

What does your registry have here:

HKLM\Software\Minnetonka Audio Software\SurCode DVD DTS\Home

Is there a "surcodedvd.exe" in the path listed in that registry value?

[MichalHabart]

Argh, so you want the temp files to be on another drive?

Yes, exactly


Does DTS encoding work if you put the mono wav files manually into SurCode?

What does your registry have here:

HKLM\Software\Minnetonka Audio Software\SurCode DVD DTS\Home

Is there a "surcodedvd.exe" in the path listed in that registry value?

It works manually but key contains this:
C:\Program Files\SurCode DVD DTS

[madshi]

It works manually but key contains this:
C:\Program Files\SurCode DVD DTS
Is there a "surcodedvd.exe" in that path? Which OS are we you using?

[MichalHabart]

Is there a "surcodedvd.exe" in that path? Which OS are we you using?

No, as i posted before. there is just this:
C:\Program Files\SurCode DVD DTS

nothing more.

I am using WXP Pro

[madshi]

No, as i posted before. there is just this:
C:\Program Files\SurCode DVD DTS
No. I mean open My Computer and look what you find in "C:\Program Files\SurCode DVD DTS". Is there a "surcodedvd.exe" file in that folder?

[MichalHabart]

No. I mean open My Computer and look what you find in "C:\Program Files\SurCode DVD DTS". Is there a "surcodedvd.exe" file in that folder?

Yes, surcodedvd.exe is in this folder

[Binnugsai]

Hello,

I use eac3toGui 1.18 and try to convert an TrueHD with extension Vendeta.evo only with one audio witch i i demux with Evodemux and i have this kind of error !!!

Microsoft Windows XP [version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

O:\V pour Vendetta HD-DVD.1080p>"C:\DVD Outils HD\EAC3toGUI 0.90\eac3to.exe" "O:
\V pour Vendetta HD-DVD.1080p\Vendetta.EVO" "O:\V pour Vendetta HD-DVD.1080p\Ve
ndettaEnglish.ac3" -24
Muxing evo's first audio track to raw. Please wait...
Getting demuxer audio output pin failed.

O:\V pour Vendetta HD-DVD.1080p>

Can someone can help me !!!!

[madshi]

Yes, surcodedvd.exe is in this folder
When you start SurCode manually, what does the title bar say? Does it say "DTS Encoder" or something else? Does the encode button say "Encode" or something else? eac3to depends on these texts.

You don't have accidently deleted the "HookSurcode.dll" file which comes with eac3to? Cause that file is needed for SurCode automation.

[madshi]

I use eac3toGui 1.18 and try to convert an TrueHD with extension Vendeta.evo only with one audio witch i i demux with Evodemux and i have this kind of error !!!

O:\V pour Vendetta HD-DVD.1080p>"C:\DVD Outils HD\EAC3toGUI 0.90\eac3to.exe" "O:
\V pour Vendetta HD-DVD.1080p\Vendetta.EVO" "O:\V pour Vendetta HD-DVD.1080p\Ve
ndettaEnglish.ac3" -24
Muxing evo's first audio track to raw. Please wait...
Getting demuxer audio output pin failed.
That means that the Nero splitter doesn't seem to like this EVO file. Do you have the latest Nero version installed?

[MichalHabart]

When you start SurCode manually, what does the title bar say? Does it say "DTS Encoder" or something else? Does the encode button say "Encode" or something else? eac3to depends on these texts.

You don't have accidently deleted the "HookSurcode.dll" file which comes with eac3to? Cause that file is needed for SurCode automation.


Title says "DTS Encoder" and button is named "Encode".
HookSurcode.dll file is in same directory as eac3to.exe

[madshi]

Convert the mono wav files to dts. Please wait...
Creating the DTS file failed. The multi channel wav files are valid, though.
How much time delay is between those 2 lines? Does the "Creating the DTS file failed" line appear without delay? Or does it appear after some seconds/minutes? If there's a delay, can you please search your harddisk. Maybe the "ghost.dts" file was stored in a folder where it wasn't supposed to be stored?

[Binnugsai]

@ Madshi,

Yes i have Nero 7.9.6.0, so !!!

[Wilbert]

@xkodi,

but i don't know if 'Nero Audio Decoder 2' is capable to decode pure MLP, but i will try that test in several hours and post back the result.
Is this true? I'm unable to connect a mlp stream to this decoder, but i haven't bought this HD plugin yet.

Does it work when you remux MLP (from AOB) in EOV? Or is this remuxing not possible yet?

[MichalHabart]

How much time delay is between those 2 lines? Does the "Creating the DTS file failed" line appear without delay? Or does it appear after some seconds/minutes? If there's a delay, can you please search your harddisk. Maybe the "ghost.dts" file was stored in a folder where it wasn't supposed to be stored?

Delay between these 2 lines is less then 1 second and ghost.dts is nowhere on my system.

[madshi]

Yes i have Nero 7.9.6.0, so !!!
Don't know why it doesn't work then. Try using GraphEdit (Nero File Source Async -> Nero Splitter -> Nero Audio Decoder 2 -> Dump). You need to rename GraphEdit.exe to Recode.exe, though. Otherwise the filters won't work.

[madshi]

Delay between these 2 lines is less then 1 second and ghost.dts is nowhere on my system.
Ok, I guess I'll have to add some more error messages to track down the cause of the problem.

[Binnugsai]

Don't know why it doesn't work then. Try using GraphEdit (Nero File Source Async -> Nero Splitter -> Nero Audio Decoder 2 -> Dump). You need to rename GraphEdit.exe to Recode.exe, though. Otherwise the filters won't work.


Thank you i try and tell you about.

[The_Keymaker]

New version (v1.19) of EAC3toGUI available here:

http://www.sendspace.com/file/jz17up

Changelog:

1. Fixed *.raw source file extension bug.

Regards,
The_Keymaker

[Roscoe62]

I'm really sorry to ask a question that may have already been answered but...

Is this tool able to successfully Translate a Dolby TrueHD soundtrack into an LPCM soundtrack?

I know a lot of people are going all the way and producing a FLAC/DD/DTS track, but all I really want is LPCM.

Thank you :)

[madshi]

I'm really sorry to ask a question that may have already been answered but...

Is this tool able to successfully Translate a Dolby TrueHD soundtrack into an LPCM soundtrack?

I know a lot of people are going all the way and producing a FLAC/DD/DTS track, but all I really want is LPCM.

Thank you :)
Currently we're still having some problems with the TrueHD decoders. The Sonic one adds a bit of noise. The Nero one seems to do some processing on the audio data, so that a 16bit TrueHD track comes out as 24bit.

Anyway, yes, basically eac3to lets you convert TrueHD to LPCM. However, you may need to change the channel mapping (eac3to should be able to do that for you, if you tell it what you need) because Blu-Ray LPCM tracks have a somewhat "strange" channel order. Furthermore eac3to gives out little endian while you probably need big endian to get a Blu-Ray compatible LPCM track. You can change the endian by using sox.

[MichalHabart]

Ok, I guess I'll have to add some more error messages to track down the cause of the problem.

Will wait for newer version :)

[Roscoe62]

Currently we're still having some problems with the TrueHD decoders. The Sonic one adds a bit of noise. The Nero one seems to do some processing on the audio data, so that a 16bit TrueHD track comes out as 24bit.

Anyway, yes, basically eac3to lets you convert TrueHD to LPCM. However, you may need to change the channel mapping (eac3to should be able to do that for you, if you tell it what you need) because Blu-Ray LPCM tracks have a somewhat "strange" channel order. Furthermore eac3to gives out little endian while you probably need big endian to get a Blu-Ray compatible LPCM track. You can change the endian by using sox.

Thank you Madshi :)

Don't need it to be Blu-Ray compatible, just as long as I can mux it into a .ts stream (via TSRemux) and it will play back correctly.

[jdbuckeye2009]

hey a while back a few people were having trouble with DD+ mono files, and were getting errors like this:

E-AC3, 1.0 channels, 2:04:39, 64kbit/s, 48khz
Mono E-AC3 tracks can only be decoded by Nero's filters, switching to Nero.
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.

I saw that some people got it to work, but it didn't really say what they did differently. I am trying to get the audio for Casablanca, and nothing is working.

For the record, I have nero 7.9.6.0 Premium, the newest version of eac3to, and vista x64. Any help would be appreciated.

[crapz0rs]

jdbuckeye2009: Buy the hddvd/bluray plugin for nero.
It is separately licensed.

After you enter the serial, the already installed filters will unlock.

[Roscoe62]

Currently we're still having some problems with the TrueHD decoders. The Sonic one adds a bit of noise. The Nero one seems to do some processing on the audio data, so that a 16bit TrueHD track comes out as 24bit.

Madshi,

As things stand, if you were to choose one of those TrueHD decoders over the other, which would you choose? i.e. which currently does the better job?

[madshi]

I saw that some people got it to work, but it didn't really say what they did differently. I am trying to get the audio for Casablanca, and nothing is working.

For the record, I have nero 7.9.6.0 Premium, the newest version of eac3to, and vista x64. Any help would be appreciated.
Some people say that you need the Blu-Ray/HD-DVD plugin for that to work. I'm not fully sure myself. I have the plugin and it works for me. But I don't know if it would work without the plugin for me, too.

[madshi]

As things stand, if you were to choose one of those TrueHD decoders over the other, which would you choose? i.e. which currently does the better job?
I've not tried the Sonic decoder myself yet. I guess if you HAD to use a decoder right now, I'd probably choose Nero. But if you ask me, wait until there's a corrected revision of either the Nero or Sonic decoder available. In the meanwhile Nero support has told me that they've forwarded my "complaint" to the developers. So maybe we'll get a fixed Nero TrueHD decoder sooner or later...

[madshi]

Argh, just got a reply from Nero. The reason why the 16bit TrueHD decoder output is not 16bit is because the TrueHD decoder applies Dynamic Range Control. Nero has to do that to fulfill the Dolby license. They say Dynamic Range Control can be turned off in Nero ShowTime. But unfortunately that doesn't have any effect on the decoder when using it through GraphEdit... :(

It gets worse: The Nero support told me that in future Nero versions the Nero license will not allow, anymore, to use the Nero filters outside of the Nero software.

So I'm sad to say that at this point I cannot recommend to buy the Nero HD DVD/Blu-Ray plugin, anymore.

If you don't like this, please complain to Nero. The more people complain the better chances we have to achieve a change. Thanks.

[Wilbert]

So I'm sad to say that at this point I cannot recommend to buy the Nero HD DVD/Blu-Ray plugin, anymore.
It seems that we can't buy it in the Netherlands. Can you buy it at all outside America?

[madshi]

It seems that we can't buy it in the Netherlands. Can you buy it at all outside America?
I was able to buy it from Germany.

[ACrowley]

Argh, just got a reply from Nero. The reason why the 16bit TrueHD decoder output is not 16bit is because the TrueHD decoder applies Dynamic Range Control. Nero has to do that to fulfill the Dolby license. They say Dynamic Range Control can be turned off in Nero ShowTime. But unfortunately that doesn't have any effect on the decoder when using it through GraphEdit... :(

It gets worse: The Nero support told me that in future Nero versions the Nero license will not allow, anymore, to use the Nero filters outside of the Nero software.

So I'm sad to say that at this point I cannot recommend to buy the Nero HD DVD/Blu-Ray plugin, anymore.

If you don't like this, please complain to Nero. The more people complain the better chances we have to achieve a change. Thanks.

really bad, hopefully Sonic comes up with new allround Decoder (without TrueHD backround noise)
The current 4.3 Audio Decoder from Cinevision 2.0.2 Final isnt working without Noise

So, we have to use older Versions in the Future to decode TrueHD.
Maybe it works again with Graphedit.exe to recode.exe renaming.
Who knows. Offic. you never cant use Nero Decoders out of Showtime

I have no big Problem with the Nero TruHD 24Bit Output.
Would be bad when output when the Output is always to low with 16bit instead of 24bit.
I mean you can encode to 16bit flac or to dts fom 24bit wave without Problems
I think Sonic will apply Dynamic Range Control on TrueHD too ?

But EAC3 decoding would be bad without Nero ,because i dont want get back the DynmicRangeCompression from Sonic

[xkodi]

really bad, hopefully Sonic comes up with new allround Decoder (without TrueHD backround noise)
The current 4.3 Audio Decoder from Cinevision 2.0.2 Final isnt working without Noise

File Source (Async.) -> Sonic HD Demuxer -> Sonic Audio Decoder 4.3 -> Nero Sound Processor -> Default DirectSound Device

when 'Nero Sound Processor' is set to stereo there is no noise and that is very interesting, so most probably the Sonic decoder works and as i mentioned before the noise is caused by incompatibility with DirectShow filters or something like that, plus both new versions - 4.2.0.84 and 4.3.0.151, that can play TrueHD have the same noise problem, so if it is 4.2.0.84 bug and is so obvious, why it's not fixed in 4.3.0.151...

[jdbuckeye2009]

ugh, it sounds kinda iffy as to whether the nero HD plugin will work, so I am a bit reluctant to shell out the $30+ for it. Maybe I'll just rent the Casablanca DVD, grab the audio, and mux it in or something. I kind of doubt that I am going to encounter a whole lot of HD-DVDs with mono DD+ audio anyways...

Thanks for the responses though.

[ACrowley]

File Source (Async.) -> Sonic HD Demuxer -> Sonic Audio Decoder 4.3 -> Nero Sound Processor -> Default DirectSound Device

when 'Nero Sound Processor' is set to stereo there is no noise and that is very interesting, so most probably the Sonic decoder works and as i mentioned before the noise is caused by incompatibility with DirectShow filters or something like that, plus both new versions - 4.2.0.84 and 4.3.0.151, that can play TrueHD have the same noise problem, so if it is 4.2.0.84 bug and is so obvious, why it's not fixed in 4.3.0.151...


Can be that Sonic 4.3 Decoder will work without Noise when we can reg the 4.3 Decoders correctly in System.
At the Moment we can only overwrite them with 4.2 Decoder cause when you try to re them you will ge the Error Message

Maybe theers no Noise cause Sonic has Problems with Mutichannel Decoding on TrueHD

File Source (Async.) -> Sonic HD Demuxer -> Sonic Audio Decoder 4.3 -> Nero Sound Processor -> Default DirectSound Device

[xkodi]

Maybe theers no Noise cause Sonic has Problems with Mutichannel Decoding on TrueHD

but Sonic output to 'Nero Sound Processor' is in multichannel and then 'Nero Sound Processor' converts that multichannel output to stereo:

http://img183.imageshack.us/img183/8800/nspfs8.png

"Channel Mask: 0x3f" means 5.1 channels and "Channel Mask: 0x03" means stereo.

when i use the following graph, i.e. dump the multichannel output from Sonic to file:

File Source (Async.) -> Sonic HD Demuxer -> Sonic Audio Decoder 4.3 -> Dump

and then convert the dump file to WAV file with sox and split that file to 6 mono WAVs with Wavewizard there IS noise in all 6 channels, so 'Nero Sound Processor' do something very interesting, because there are two possibilities:

1) somehow 'Nero Sound Processor' converts multichannel WAV file with noise in all channels to stereo WAV file without noise, which seems impossible to me, because multichannel to stereo conversion basically should be mixing those 6 channels to 2 channels and thus the noise will be mixed too.

2) somehow 'Nero Sound Processor', when set to stereo, prevents Sonic to do errors and thus the multichannel output that Sonic produces in this case is different (error free) from the multichannel output that it produces, when is connected to Dump.ax filter and this option seems the right one, at least for me.

or maybe someone have other ideas how to solve that puzzle...

[Wilbert]

@xkodi, please respond:

http://forum.doom9.org/showthread.php?p=1021485#post1021485

[DeepBeepMeep]

eac3to looks to be a very interesting tool. Is there a guide somewhere that explains how to use it to convert a Blu-ray LPCM track to DTS for instance. There seems to be lots of ways to extract the LPCM from the Blu-ray and I end up with garbage noise wav tracks.

Thanks.

[madshi]

eac3to looks to be a very interesting tool. Is there a guide somewhere that explains how to use it to convert a Blu-ray LPCM track to DTS for instance. There seems to be lots of ways to extract the LPCM from the Blu-ray and I end up with garbage noise wav tracks.
Try xport to demux the LPCM and then "eac3to some.pcm some.dts". You may need to specify the bitdepth ("-16" or "-24") when calling eac3to, if it can't figure that out on itself. If you don't get this to work, let me know which movie you've problems with.

[madshi]

@xkodi,


Is this true? I'm unable to connect a mlp stream to this decoder, but i haven't bought this HD plugin yet.

Does it work when you remux MLP (from AOB) in EOV? Or is this remuxing not possible yet?
IIRC the latest Sonic filters (Sonic HD Demuxer -> Sonic Audio Decoder) work good for demuxed MLP. Just demux the MLP and name the file "*.MLP". I can drop such a file in GraphEdit and it works at once. I wouldn't recommend the Nero filters at this time due to the processing (probably Dynamic Range Control) they're applying to the decoded data.

[Roscoe62]

Madshi,

on the previous page you said that eac3to could be used to convert a TrueHD track into LPCM. Could I trouble you to explain how this can be done? I'm guessing using the gui is NOT the way to go here.

Also, I'm trying to do the same thing with some of my DD+ audio tracks (convert to LPCM) - and I felt sure that eac3to could also do that.

Please let me know if I'm wrong, and thank you for your patience. :)

FWIW...the only reason I want LPCM is to preserve as much quality as possible (I see AC3 as a reduction in quality) and also to not have to use Sonic or Nero filters at playback time.

Thanks again!

[madshi]

on the previous page you said that eac3to could be used to convert a TrueHD track into LPCM. Could I trouble you to explain how this can be done? I'm guessing using the gui is NOT the way to go here.

Also, I'm trying to do the same thing with some of my DD+ audio tracks (convert to LPCM) - and I felt sure that eac3to could also do that.
Basically you'd do "eac3to source.eac3 dest.raw" or "eac3to source.evo dest.raw". The raw file is then basically LPCM. However, eac3to outputs little endian in the raw file while a Blu-Ray LPCM track is usually big endian. So you may need to use "sox" to change endian from little to big. Furthermore you may have to remap the channels cause the Blu-Ray LPCM channel mapping is kind of strange. Eac3to uses "-0,1,2,5,3,4" to convert Blu-Ray LPCM files to the usual channel mapping. You need to do the reverse, so try calling eac3to with "-0,1,2,4,5,3". So in short again:

(1) "eac3to source.whatever dest.raw -0,1,2,4,5,3"
(2) call "sox" to convert little endian to big endian

Don't have the sox parameters available right now, sorry.

[DeepBeepMeep]

Thanks Madshi for your answer. When converting Blu-ray to DTS or AC3 do I have to remap channels even I have already used the -Blu-Ray option ? Many thanks again for your great tool and your help.

[madshi]

Thanks Madshi for your answer. When converting Blu-ray to DTS or AC3 do I have to remap channels even I have already used the -Blu-Ray option ? Many thanks again for your great tool and your help.
The "-blu-ray" option is no more in the current version. eac3to now defaults to the options necessary for Blu-Ray handling if you feed it a PCM file. The only thing you may have to specify is the bitdepth ("-16" or "-24"). Channels are automatically remapped correctly.

[eas4uk]

Hi Madshi

Great tool which I have now been using for a few weeks - big thanks to you mate :)

Is there anyway that this tool can be used to convert DTS-HD to .ac3 from the command line?

Cheers

eas4uk

[madshi]

Is there anyway that this tool can be used to convert DTS-HD to .ac3 from the command line?
Can't you can already do that with e.g. BeHappy? DTS-HD contains a standard DTS core, I believe, so basically BeHappy should be able to do the conversion. Of course you'd lose the additional information of DTS-HD this way.

[DeepBeepMeep]

Thanks Madshi.

I may need the sonic audio decoder. I would like to buy them but can't find them on the sonic website. Are they part of some package? Thanks

[xkodi]

@xkodi,


Is this true? I'm unable to connect a mlp stream to this decoder, but i haven't bought this HD plugin yet.

Does it work when you remux MLP (from AOB) in EOV? Or is this remuxing not possible yet?

yes, it's true, however to use Nero decoder you need first to mux the MLP into EVOB and its not easy at all to do that and EVODemux can't correctly mux AOB into EVOB, so the good choice for MLP is the Sonic decoder, it can playback raw MLP stream, so you need only to extract MLP tracks from AOB using DVDAExplorer, both Nero and Sonic do not apply DRC when playing MLP.

[eas4uk]

Can't you can already do that with e.g. BeHappy? DTS-HD contains a standard DTS core, I believe, so basically BeHappy should be able to do the conversion. Of course you'd lose the additional information of DTS-HD this way.

I never had any joy with BeHappy, however, eac3to has never let me down yet, no worries I'll persevere with BeHappy until I get it to work...

Thanks again for eac3to though :thanks:

[madshi]

I never had any joy with BeHappy, however, eac3to has never let me down yet, no worries I'll persevere with BeHappy until I get it to work...

Thanks again for eac3to though :thanks:
I might add DTS(-HD) support to eac3to sooner or later. But it's not a priority right now.

[eas4uk]

I might add DTS(-HD) support to eac3to sooner or later. But it's not a priority right now.

That would be great if you can at some point :)

[ACrowley]

Can't you can already do that with e.g. BeHappy? DTS-HD contains a standard DTS core, I believe, so basically BeHappy should be able to do the conversion. Of course you'd lose the additional information of DTS-HD this way.


Maybe Tranzcode.exe (0.4) would be the Better implemention for eac3to

Its a small exe and Tranzcode 0.4 can directly decode the
dts hd to wave. It decodes the included dts core ofcourse

simple command :
tranzcode.exe inputdts.dts output.wav

switches :
/ -mch = multichannel wave output
/-16 or 24 or 32

and you can extract single channles only

[madshi]

Maybe Tranzcode.exe (0.4) would be the Better implemention for eac3to

Its a small exe and Tranzcode 0.4 can directly decode the
dts hd to wave. It decodes the included dts core ofcourse

simple command :
tranzcode.exe inputdts.dts output.wav

switches :
/ -mch = multichannel wave output
/-16 or 24 or 32
Sounds good. But maybe Nero and/or Sonic will make use of the added DTS-HD information? That would be preferred, of course, over only using the core.

[MichalHabart]

Hi Madshi,
i tried different PCM track but still the same error. Creation of DTS failed. When i do manually, everything is ok, surcode will create proper DTS. Is there something i could be doing wrong?

And the second question, mono wav file are correct as it says. Does it mean that they are also mapped to be feeded to surcode in order 0-1-2-3-4-5? (In this case, is C channel really C?)

[ACrowley]

@MichaelHarbart
eac3 mono wave (.wavs) output is :
1 Left
2 Right
3 Center
4 LFE
5 SL
6 SR

Surcode input order is the same

Have you Surcode DTS Pro 1.0.23 installed ?
You need minimum v1.0.23

@madshi

No Nero only decodes the Core. You can see it when you craete a graph with NeroAudioDecoder2 on a dtshd.
Decoding info is only 1536kbps (core depended)
And i never had succes to connect dts with sonic. When its possible ,i dont believe Sonic Decoder can decode full dtshd
Theres currently no Decoder which can decode the full dtshd
Imho the simple core extraction makes most Sense, unless you want encode standard dtscore to ac3

Maybe you can implement both Options for dts hd.
Decoding dtshd(core) to wave with Tranzcode and extract dtscore from dtshd

[MichalHabart]

@MichalHarbart
eac3 mono wave (.wavs) output is :
1 Left
2 Right
3 Center
4 LFE
5 SL
6 SR

Surcode input order is the same

Have you Surcode DTS Pro 1.0.23 installed ?
You need minimum v1.0.23


Yes, i have it installed in standard directory C:\Program Files\Surcode DVD DTS
Creation of mono wav files is always OK but then "Creating the DTS file failed". This message appears immediately after mono wav files are created.

[madshi]

i tried different PCM track but still the same error. Creation of DTS failed. When i do manually, everything is ok, surcode will create proper DTS. Is there something i could be doing wrong?
Currently I'm not sure. Surcode automation seems to work for some people and not for others. I'm planning to add more detailed error information to the next eac3to build to find out why Surcode automation fails when it fails.

And the second question, mono wav file are correct as it says. Does it mean that they are also mapped to be feeded to surcode in order 0-1-2-3-4-5? (In this case, is C channel really C?)
eac3to should name the mono files correctly. C should really be C etc.

[madshi]

No Nero only decodes the Core. You can see it when you craete a graph with NeroAudioDecoder2 on a dtshd.
Decoding info is only 1536kbps (core depended)
And i never had succes to connect dts with sonic. When its possible ,i dont believe Sonic Decoder can decode full dtshd
Theres currently no Decoder which can decode the full dtshd
Imho the simple core extraction makes most Sense, unless you want encode standard dtscore to ac3

Maybe you can implement both Options for dts hd.
Decoding dtshd(core) to wave with Tranzcode and extract dtscore from dtshd
Ok, thanks for the information. Not sure yet what I'll do.

[Roscoe62]

Madshi,

Thanks for the method to transcode DD+ to LPCM. When using sox to change to Big endian, because the input file is raw I have to tell sox a few parameters of the input file so it can process it - i.e. sample rate, sample size, data encoding & channels.

Is there any way to tell what the output file eac3to produces will be when the input file is DD+ so I can pass sox the correct data?

Again, thank you for your time!

[madshi]

Madshi,

Thanks for the method to transcode DD+ to LPCM. When using sox to change to Big endian, because the input file is raw I have to tell sox a few parameters of the input file so it can process it - i.e. sample rate, sample size, data encoding & channels.

Is there any way to tell what the output file eac3to produces will be when the input file is DD+ so I can pass sox the correct data?

Again, thank you for your time!
Should be 48khz, 24bit, usually 6 channels, little endian.

[Roscoe62]

One more quick question. If nothing else is used in the eac3to command line, which DD+ decoder is used by default? I have both Sonic 4.2 and Nero (blu-ray & hd-dvd plugin) but I think my Sonic decoders have gone a bit "screwy", even after reinstalling them. If I wish to use the Nero decoder do I have to rename eac3to.exe to recode.exe?

I'm very sorry for so many questions!

[cstrikedish]

A great tool! thanks for share!
:thanks:

[madshi]

If nothing else is used in the eac3to command line, which DD+ decoder is used by default?
Sonic.

I have both Sonic 4.2 and Nero (blu-ray & hd-dvd plugin) but I think my Sonic decoders have gone a bit "screwy", even after reinstalling them. If I wish to use the Nero decoder do I have to rename eac3to.exe to recode.exe?
No, that's not necessary. Just use the "-nero" switch.

[Roscoe62]

Well, I finally decided to give up on LPCM. I was finding it too difficult to mux it into anything, so I decided to use flac. Gotta say the procedure is MUCH easier!

Anyway, when it comes to setting up a directshow graph to play back my completed mkv file, for those encoding to flac, what filter is the best to use? I'm currently using ffdshow (although an older version) and it seems to work OK, but I wanted to know if there's anything better out there?

Also, is there any perceived quality difference between keeping 24 bit or downconverting to 16 bit?

As an alternative, is transcoding to a multi-channel wav file a better way to go, quality-wise?

Thanks to Madshi for a truly break-through app! :)

[madshi]

for those encoding to flac, what filter is the best to use? I'm currently using ffdshow (although an older version) and it seems to work OK
I had tried some other decoders, but they didn't even work properly for me at all. I'm using ffdshow, too.

Also, is there any perceived quality difference between keeping 24 bit or downconverting to 16 bit?
I've not really tested that yet. Of course that will also depend on whether the source was 24 bit or not.

As an alternative, is transcoding to a multi-channel wav file a better way to go, quality-wise?
FLAC is lossless, so it's 100% identical to LPCM and to WAV. Basically WAV is the same as LPCM, just with a little header in front of the LPCM data. FLAC is like a zipped WAV.

[musky5789]

Can anyone tell me where i can get "Dump" filter from???

[honai]

Speaking of FLAC, what's the easiest way to transcode FLAC audio streams into separate mono WAVs suitable for feeding into Surcode DTS?

[xkodi]

new CinemasterAudio.dll v4.3.0.169 + SonicHDDemuxer.dll v4.3.0.89 are out (as part of CineVision 2.0.2final).

there are no "discontinuities" errors with the new versions:

http://forum.doom9.org/showthread.php?p=1019802#post1019802

and as a result there is no noise when decoding TrueHD, but the output is very different from the old CinemasterAudio.dll v4.3.0.151 + SonicHDDemuxer.dll v4.3.0.73 (part of CineVision 2.0.2beta):

http://forum.doom9.org/showthread.php?p=1019241#post1019241

which is very confusing, because the old version has almost the same output as Nero for "Nine Inch Nails: BYIT bonus TrueHD track" and now it is entirely different:

http://img49.imageshack.us/img49/371/ninsonicnewwo9.png (http://imageshack.us)

and the "Vendetta TrueHD stream" output is no more 16bit, but seems more like 8bit:

http://img526.imageshack.us/img526/6397/sonicvendnewhq8.png (http://imageshack.us)

the big surprise is that after overwriting of the new DLLs with the old ones, there is again no noise and the output is byte by byte identical, so maybe some registry settings are affecting the behavior of the Sonic codec, but i haven't time to dig deeper and find out what is really going on.

[madshi]

Can anyone tell me where i can get "Dump" filter from???
It comes with GraphEdit.

[madshi]

new CinemasterAudio.dll v4.3.0.169 + SonicHDDemuxer.dll v4.3.0.89 are out (as part of CineVision 2.0.2final).

there are no "discontinuities" errors with the new versions:

http://forum.doom9.org/showthread.php?p=1019802#post1019802

and as a result there is no noise when decoding TrueHD
That's extremely strange cause I still have noise with the same dlls!!

and the "Vendetta TrueHD stream" output is no more 16bit, but seems more like 8bit
How does it sound? Does the WAV look and sound alright?

[xkodi]

That's extremely strange cause I still have noise with the same dlls!!How does it sound? Does the WAV look and sound alright?

madshi, both TrueHD tracks sound completely normal and the fact, that you are experiencing noise with the new Sonic DLLs and that here even after deletion of the new DLLs and going back to the old ones i don't have noise is more than extremely strange, i don't know and i am really running out of ideas what is going on...

P.S. i'm comparing part of Nine Inch Nails track with a lot of drums and Sonic even sounds much better than Nero, at least in my opinion.

[edit] and with Vendetta track: Nero output is very bad - the voices are low and unclear, with Sonic the voices are very clear and the volume is much pleasant than the Nero.

[ACrowley]

That's extremely strange cause I still have noise with the same dlls!!


How does it sound? Does the WAV look and sound alright?

I have still noise to ! on demuxed .mlp and evo

[madshi]

@xkodi, what magic did you do? :) Did you do anything else besides running the installer and copying the new dlls over the old ones in the "common" folder?

[ACrowley]

I think hes only overwritten the 4.3 over 4.2 after installation

Its not possible to reg the 4.3 Decoders manually, you will get a Error

I dont know, ive the Feeling the noise has to do with LFE.
When the TrueHD Track starts ,without LFE Effects its clean until LFE Effects come in

Maybe its a simple Registry Settting

However, i have still noise with 4.3 and TrueHD. Encoded mlp from Surcode works perfect
But now i have succes to connect to dts and dts hd ,but Sonic decodes only the core:
mpc player properties :Audio: PCM (DTS) 48000Hz 6ch 1509Kbps [Output])

[Nikos]

I convert DD+ to AC3 with eac3to. I want to know more about -down16 downconvert raw data to 16 bit.
When i must to use it?

[madshi]

I convert DD+ to AC3 with eac3to. I want to know more about -down16 downconvert raw data to 16 bit.
When i must to use it?
You are not forced to use it - ever. You *can* use it if you want to save space when doing FLAC encoding. The size difference between 16bit and 24bit with FLAC encoding is quite big. Of course the audio quality suffers a bit. But only if you have good ears and a good receiver and good boxes etc...

Do *NOT* use "-down16" for AC3 and DTS encoding because there's no benefit in doing that. There's neither an advantage in size nor in sound quality.

[Nikos]

Thank you madshi for the precious explain.

[xkodi]

@xkodi, what magic did you do? :) Did you do anything else besides running the installer and copying the new dlls over the old ones in the "common" folder?

it's not a magic i found the reason: i don't know why but "Control Panel -> Sounds and Audio Devices -> Volume -> Speaker Settings -> Speakers -> Speaker Setup" was changed from "5.1 surround sound speakers" to "Desktop stereo speakers":

http://img262.imageshack.us/img262/3969/spkyh4.png (http://imageshack.us)

when i set it back to "5.1 surround sound speakers" the noise is back, so at least Sonic can decode first 2 channels without noise and now we know how to make it to do that.

[madshi]

when i set it back to "5.1 surround sound speakers" the noise is back
Too bad.

But it may explain why Sonic didn't notice the bug: Most people only have stereo speakers on their desktop!

so at least Sonic can decode first 2 channels without noise and now we know how to make it to do that.
Right. When will version 4.4 come? :)

[xbox360]

Did you know that Dolby TrueHD is Dolby MLP Lossless, so if you have the SurCode MLP Lossless encoder you have Dolby TrueHD encoder !.

[SBeaver]

I'm getting some type of error when trying to convert eac3 with sonic decoder.
Apparently the dump filter doesn't work, or sonic doesn't work.
This was on Vista x64 (sonic worked ok with xp x64)
Sonic and dump filter show up in dshow filter manager so that's not the problem.
I can't test Nero right now because I havn't bought the plugin.

[madshi]

Did you know that Dolby TrueHD is Dolby MLP Lossless, so if you have the SurCode MLP Lossless encoder you have Dolby TrueHD encoder !.
TrueHD is based on MLP, but has some extensions, also the header is slightly different. So no, the SurCode MLP encoder is not a TrueHD encoder. And the decoders behave differently for MLP and TrueHD. E.g. Sonic's latest decoder adds ugly noise when decoding TrueHD but it perfect with MLP.

[madshi]

I'm getting some type of error when trying to convert eac3 with sonic decoder.
Apparently the dump filter doesn't work, or sonic doesn't work.
This was on Vista x64 (sonic worked ok with xp x64)
Sonic and dump filter show up in dshow filter manager so that's not the problem.
I can't test Nero right now because I havn't bought the plugin.
Which Sonic version are you using? What do you mean with "some type of error"? Some more details about the error would be helpful. Do you think it's a problem with eac3to? Or does it also occur in GraphEdit?

[SBeaver]

Which Sonic version are you using? What do you mean with "some type of error"? Some more details about the error would be helpful. Do you think it's a problem with eac3to? Or does it also occur in GraphEdit?

I'd say the problem must be with the sonic filter.
The error message was something about dump filter unable to... something. It created an empty file that is.
I can't put graphedit together at all because I don't have the right splitters apparently. (sonic audio decoder is available there though)
Maybe this problem went off topic now, what I meant to ask was if there are any compatbillity problems with vista x64 and dump and sonic or maybe eac3to.

I have version 4.2 btw, worked well in xp but not in vista x64.

[eas4uk]

Has anyone had the "demuxer didnt accept the source file" error message before? When its trying to mux the eac3 file to raw?

I'm getting this now - I've used evodemux to join the two feature.evo s together and then demuxed the audio, renamed to filename.eac3 and then used the eac3to gui as I have loads of times before and I'm now suddenly getting this error...

Any ideas? Its definitely a DD+ track 1.5mbps bitrate btw

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

D:\Audio>"D:\Audio\eac3to.exe" "D:\Audio\Black.eac3" "D:\Audio\Black.ac3" -640

E-AC3, 5.1 channels, 1:55:33, 1536kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The demuxer didn't accept the source file.

D:\Audio>

[madshi]

That's a bit strange since eac3to seems to recognize the file just fine. Please try to rename the file to black.ddp. Does that help? Also please try whether the Sonic HD demuxer accepts this file in GraphEdit, if you know how to use GraphEdit. Maybe uninstalling and reinstalling the Sonic filters work. That has fixed some problems in the past.

[eas4uk]

Renaming got the same result

I'm a bit of a noob with graphedit but i have tried

file source (async) >> Sonic HD Demuxer
&
file source (async) >> Sonic audio decoder

and both time graph edit states that a connection cant be created

I'll try uninstalling sonic and reinstalling and post back shortly

Edit:

Nope that hasn't solved it either :(

[madshi]

file source (async) >> Sonic HD Demuxer
That's the correct thing to do. If the Sonic HD Demuxer refuses to accept the eac3 file, there's probably something wrong with the Sonic installation or something like that.

[eas4uk]

I've uninstalled and reinstalled sonic and still the problem remains

Any other suggestions Madshi?

[The_Keymaker]

@eas4uk

Instead of demuxing the audio track from the joined feature evos, try using EVOdemux to just demux the audio tracks by themselves.

That is, in EVOdemux, unselect everything except the audio track you desire, and then demux. As long as you have the "continue with second evo" box checked under options, EVOdemux will demux the desired audio track for the entire movie.

The_Keymaker

[Wilbert]

Renaming got the same result

I'm a bit of a noob with graphedit but i have tried

file source (async) >> Sonic HD Demuxer
&
file source (async) >> Sonic audio decoder

and both time graph edit states that a connection cant be created
I've got the same problem (with input *.mlp). I'm using Sonic CinePlayer HD DVD Decoder v4.2, but i've no idea where to get/buy v4.3. I can't find it on http://www.sonic.com/products/Consumer/ :confused:

[eas4uk]

@The Keymaker

Just tried that - same problem I'm afraid

[eas4uk]

I've got the same problem (with input *.mlp). I'm using Sonic CinePlayer HD DVD Decoder v4.2, but i've no idea where to get/buy v4.3. I can't find it on http://www.sonic.com/products/Consumer/ :confused:

I've just tried 4.3 but that wont work either.

[SBeaver]

[QUOTE=eas4uk;1024784]I've just tried 4.3 but that wont work either

There is a link for 4.3 below - but just a warning: I didnt post that so I cant verify the source (i just googled it and came across it)

*link removed*
This version seems to have helped my problem, but I think it's against the rules to post links like that.

graphedit was very happy with this version. the only problem now is I have to use haali instead of the sonic hd demuxer to play in MPC because sonic demuxer breaks video and gives me green blocks all over.

edit:sorry removed

[eas4uk]

Apologies for the link - mods can you please remove the link from the above quote

many thanks

eas4uk

[madshi]

I've uninstalled and reinstalled sonic and still the problem remains

Any other suggestions Madshi?
You may want to try orbitlee's filter. It can replace the Sonic HD Demuxer. You can tell eac3to to use orbitlee's filter instead of the Sonic HD Demuxer. However, I rather guess that if the Sonic HD Demuxer won't work, the Sonic Audio Decoder probably won't work, either.

I've got the same problem (with input *.mlp). I'm using Sonic CinePlayer HD DVD Decoder v4.2, but i've no idea where to get/buy v4.3. I can't find it on http://www.sonic.com/products/Consumer/
Currently the only way to get the Sonic v4.3 filters is through Sonic Cinevision 2.0.2. It's rather expensive, though! Hopefully they'll soon update their much cheaper Decoder set which you can buy separately. I believe only v4.3 supports MLP files. With v4.2 you're out of luck, IIRC.

[eas4uk]

Ok so im sure its my Sonic set up now as installed everything onto my laptop last night and tried it and it worked fine on there...

So why wont it work on the PC after I uninstall and reinstall - is there some registry settings that I need to delete after the uninstall to completely wipe it from the system???

[ACrowley]

try to install 4.1 Package, then 4.2 over it

I had same Problems...

[xkodi]

I believe only v4.3 supports MLP files. With v4.2 you're out of luck, IIRC.

4.2.0.840 also works with MLP, it's the first version, that i know to work with MLP, the previous 4.2.0.102 version doesn't work with MLP.

[otaku975]

whats the right extension for TRUEHD track ?
.dts ?

thanks

[madshi]

whats the right extension for TRUEHD track ?
.dts ?
Definitely not .dts. Maybe using .mlp would make sense. Or maybe .thd. Don't know if there is an "official" extension for TrueHD tracks yet. Personally, I'm using .thd.

[eas4uk]

try to install 4.1 Package, then 4.2 over it

I had same Problems...

Right then - against ACrowley's advice, I have just done a fresh install of XP and after installing all progs, filters, codecs etc...

The problem is still here!

So, after some head scratching, I deregistered the dtsac3source.ax file I'd previously registered when installing everything, and presto! Sonic is working again!

Edit: spoke to soon - D'oh forgot I need that filter to convert eac3 to ac3 - Anybody know why this filter is giving me grief then - Sonic wont work after I reinstall so its definitely this

[topace1]

Hi Madshi,

I'm glad that I have found this tool from here. I tried it by using command line "eac3to source.eac3 target.ac3", but got error message "Parsing of Dolby Digital headers failed". I'm a newbie, I don't know what the problem is and how to solve it.

The eac3 file was extracted from a mkv video using mkvextract.exe ("mkvextract tracks xxxx.mkv --raw 2:xxxx.eac3").

I'm using Windows XP Sp2, and have "Sonic HD Demuxer", "Sonic Audio Decoder 4.2" and "Dump" filter installed.

Thank you!

[madshi]

I tried it by using command line "eac3to source.eac3 target.ac3", but got error message "Parsing of Dolby Digital headers failed".
Are you sure that this file is really a eac3 track? Maybe it's something else? What happens if you drop the mkv file into the "mkvtoolnix" tool? What audio track type does that tool report?

[eas4uk]

Right then - against ACrowley's advice, I have just done a fresh install of XP and after installing all progs, filters, codecs etc...

The problem is still here!

So, after some head scratching, I deregistered the dtsac3source.ax file I'd previously registered when installing everything, and presto! Sonic is working again!

Edit: spoke to soon - D'oh forgot I need that filter to convert eac3 to ac3 - Anybody know why this filter is giving me grief then - Sonic wont work after I reinstall so its definitely this

is there a different filter I can try instead of the dtsac3source.ax ?

Madshi - How would I go about using Orbitlee opposed to Sonic btw?

[madshi]

Madshi - How would I go about using Orbitlee opposed to Sonic btw?
See first post in this thread.

[ACrowley]

Hi Madshi,

I'm glad that I have found this tool from here. I tried it by using command line "eac3to source.eac3 target.ac3", but got error message "Parsing of Dolby Digital headers failed". I'm a newbie, I don't know what the problem is and how to solve it.

The eac3 file was extracted from a mkv video using mkvextract.exe ("mkvextract tracks xxxx.mkv --raw 2:xxxx.eac3").

I'm using Windows XP Sp2, and have "Sonic HD Demuxer", "Sonic Audio Decoder 4.2" and "Dump" filter installed.

Thank you!

Remember there a few Formats which ar not working anymore after demuxing from mkv. To bad.
WVC1 Video, FLAC Audio...Perhaps EAC3 cant be used anymore too, after demuxing

I will try it

EDIT
I found a working Way how to demux EAC3 from mkv. Use Avimux 1.17.7
Select "extarct Binary" and you will get a full working EAC3

[topace1]

Are you sure that this file is really a eac3 track? Maybe it's something else? What happens if you drop the mkv file into the "mkvtoolnix" tool? What audio track type does that tool report?

mkvtoolnix shows it's e-ac3. maybe something is wrong with this file, because i tried another file, it worked! so everything is fine. Thanks for your kind reply!

Remember there a few Formats which ar not working anymore after demuxing from mkv. To bad....

I found a working Way how to demux EAC3 from mkv. Use Avimux 1.17.7
Select "extarct Binary" and you will get a full working EAC3


Thank you for the information. I'm newbie in this area and learning. I will try Avimux.

[eas4uk]

See first post in this thread.

Where do I download the orbitlee filter - ive googled it but not coming up with much :confused:

[arty]

can somebody just summarize all the above?

like:
5.1 TrueHD track -> nero filters(all version?) -> DRC applied, not recommended
5.1 TrueHD track -> sonic filters(version?) -> noise, only 2.0, not recommended
5.1 TrueHD track -> no possible conversion without DRC

5.1 DD+ track -> nero filters(all version?) -> DRC applied, not recommended
5.1 DD+ track -> sonic filters(version?) -> perfect output
5.1 DD+ track -> sonic filters(version?) are highly recommended

?

or at least can somebody pls correct/finish my example?

Thank you!

[xkodi]

can somebody just summarize all the above?

like:
5.1 TrueHD track -> nero filters(all version?) -> DRC applied, not recommended
5.1 TrueHD track -> sonic filters(version?) -> noise, only 2.0, not recommended
5.1 TrueHD track -> no possible conversion without DRC

5.1 DD+ track -> nero filters(all version?) -> DRC applied, not recommended
5.1 DD+ track -> sonic filters(version?) -> perfect output
5.1 DD+ track -> sonic filters(version?) are highly recommended

?

or at least can somebody pls correct/finish my example?

Thank you!

Nero filters:

- Nero Audio Decoder 2 (NeAudio2.ax)
- Nero File Source Async. (NeFileSourceAsync.ax)
- Nero Splitter Filter (NeSplitter.ax)

all 3 files from the same Nero version has same versions:

1) 4.9.4.1 (Nero version 7.8.5.0)
2) 4.10.5.0 (Nero version 7.9.6.0)
3) 4.11.3.1 (Nero version 7.10.1.0)

all versions of Nero filters, behave exactly the same with MLP and TrueHD tracks

Sonic filters:

- Sonic HD Audio Decoder (CinemasterAudio.dll)
- Sonic HD Demuxer (SonicHDDemuxer.dll)

0) CinemasterAudio.dll v4.2.0.102, SonicHDDemuxer.dll v4.2.0.61
1) CinemasterAudio.dll v4.2.0.840, SonicHDDemuxer.dll v4.2.0.59
2) CinemasterAudio.dll v4.3.0.151, SonicHDDemuxer.dll v4.3.0.73
3) CinemasterAudio.dll v4.3.0.169, SonicHDDemuxer.dll v4.3.0.89

0) is from "Sonic CinePlayer HD-DVD Decoder v4.2" and as far as i know is the first version capable of DD+ decoding, but can't decode MLP and TrueHD tracks
1) is from "Sonic CineVision v1.2"
2) is from "Sonic CineVision 2.0.2beta"
3) is from "Sonic CineVision 2.0.2final"
all 1)-3) can decode MLP and TrueHD tracks

5.1 TrueHD track -> Nero filters, all versions -> DRC applied, not recommended
MLP track in EVOB -> Nero filters, all versions -> perfect output
raw MLP track -> Nero filters, all versions -> doesn't work

5.1 TrueHD track -> Sonic filters, versions 1)-3) -> noise, only 2.0, not recommended
MLP track in EVOB -> Sonic filters, versions 1)-3) -> perfect output
raw MLP track -> Sonic filters, versions 1)-3) -> perfect output

5.1 TrueHD track -> no possible conversion without DRC
MLP track in EVOB and raw MLP track -> perfect output is possible

i don't have experience with DD+, so someone else should tell.

[Wilbert]

@xkodi and others,

I'm using this version:
3) CinemasterAudio.dll v4.3.0.169, SonicHDDemuxer.dll v4.3.0.89

When using a 5.1_96kHz_24bit MLP stream (i'm sure it's 96kHz by looking at the MLP header), the stream is "converted" to (passed through as ?) 88.2kHz:

http://www.geocities.com/wilbertdijkhof/sonic_hddemuxer.jpg

MLP header info:

http://www.geocities.com/wilbertdijkhof/mlp_header.jpg

Do you also have this problem? (When playing the graph the audio seems fine, so it's not mangled up ...) When loading the graph in AviSynth with directshowsource.dll the samplerate is also reported as 88.2kHz.

[ACrowley]

i don't have experience with DD+, so someone else should tell.

SonicCinemasterAudioDecoder applies DRC on EAC3 Decoding.
NeroAudioDecoder2 not
So its recommend to use only Nero Decoder to decode EAC3

[honai]

Which is the default decoder on the eac3to command-line for DD+, Sonic or Nero?

[madshi]

When using a 5.1_96kHz_24bit MLP stream (i'm sure it's 96kHz by looking at the MLP header), the stream is "converted" to (passed through as ?) 88.2kHz
That's a bit strange, but it's not completely without sense, as 48khz and 44.1khz are the usual frequences and 88.2khz happens to be exacty 2 * 44.1khz. Personally, I don't have any MLP files with 96khz, so I can't reproduce the problem. My only MLP sources are movie tracks and they seem to be 48khz only at this point in time.

[madshi]

Which is the default decoder on the eac3to command-line for DD+, Sonic or Nero?
Sonic is currently the default decoder. You can force nero with a switch (see first post of this thread).

[xkodi]

That's a bit strange, but it's not completely without sense, as 48khz and 44.1khz are the usual frequences and 88.2khz happens to be exacty 2 * 44.1khz. Personally, I don't have any MLP files with 96khz, so I can't reproduce the problem. My only MLP sources are movie tracks and they seem to be 48khz only at this point in time.

i will test that in a few hours and post back the results ...

[hristoff2]

Nero filters seem to apply DRC though...need to compare Sonic Decoder and Nero output first but I'm kind of disappointed after comparing DVD and Nero output. (dvd center wav vs. nero center wav)

[arty]

thank you xkodi, ACrowley, hristoff2!

this means we should NOT convert/playback DD+ or TrueHD with any codec because of the DRC :( bad news ... really bad.

even if we'll have a hdmi audio output we can not enjoy the HD discs with our PC :( the whole industy s*cks :( we are forced to use expensive set-top-hddvd/bd/players 'coz maybe intervideo/cybelink decoders also produce bad output we just don't know ... :(

An old dvd will sound better with its low bitrate ac3 track than a DRC'ed TrueHD track :(

[hristoff2]

Actually my version current version of Nero (and it's filters) could be corrupt. (7.9.6.0)

Trying 7.8.5.0 now...latest test I did (30mins before) I encountered that Sonic @ "quiet environment" was even more dynamic than the output of my Nero 7.9.6.0 and it's filters. :mad:
Nero's DRC can be turned off in Nero Showtime though...but until now I thought this would be directed to TrueHD decoding only...I wonder if anyone actually made tests like this:

Sonic vs. Nero
Nero vs. DVD
Sonic vs. DVD

http://nwgat.net/woot/files/51/comparison_thumb.png (http://nwgat.net/woot/files/51/comparison.png)

I could provide all this stuff (the ac3 files) and the eac3 file with the correct timecodes if someone can explain how to cut an eac3 file.

[honai]

I encountered that Sonic @ "quiet environment" was even more dynamic

Do you mean it was "more dynamic" or "more compressed"? (compression as in dynamic range compression) EDIT: Never mind, just took a look at the screenshot and saw what you mean.

@arty, hristoff2

Thanks for your investigation. I've made a post in the other forum summarizing the situation.

[madshi]

Wow, that looks really bad. It's about time we get a useful ffmpeg E-AC3 decoder. Hopefully Google Summer of Code will help!

[ACrowley]

Mh.... i thought too Nero will not apply DRC on DD+ ?
But Sonic does surely

Sonic Decoder Output sounds always louder...on AC3 too compared to AC3 Filter

Mhh i dont think we can 100% compare a DVD AC3 with EAC3 .Shouldnt EAC3 sound mor dynamic so the Waveform is different...
But Sonic shows no differenc between quite and normal..it was reported in few Threads

But no doubt....the comaprisons Screens show Nero is more Flat

However, encoded AC3 from DD+ sounds good imho. Also i usually mux the untouched EAC3 into my reencodes, so....

This is a sample from EAC3 ( Serenity HDDVD 1536kbps)
Both Center waves are loaded into Audacity
1st in Nero 2nd Sonic


http://nwgat.net/woot/files/57/Comparison_thumb.png
(http://nwgat.net/woot/files/57/Comparison.png)

[hristoff2]

Wow, that looks really bad. It's about time we get a useful ffmpeg E-AC3 decoder. Hopefully Google Summer of Code will help!

It gets worse o_O
...
Just saw it and thought it's relatively comparable...take a look at this huge shit:

http://nwgat.net/woot/files/51/comparison_2_thumb.png (http://nwgat.net/woot/files/51/comparison_2.PNG)

...so we've got 2x shit, but nero is even shittier because it's 1/2 loud. (compare overall volume from Nero to DVD and Sonic to DVD...Sonic has at least the correct "overall" volume)

[honai]

As long as the range remains the same as in the original it's no big deal, you just turn up the volume knob. From the screenshot it looks as if Nero's output is really just quieter but not less dynamic than Sonic's.

[hristoff2]

As long as the range remains the same as in the original it's no big deal, you just turn up the volume knob. From the screenshot it looks as if Nero's output is really just quieter but not less dynamic than Sonic's.

Sure, but that means it's 1:0 for Sonic because it has one issue less than Nero. :D

[ACrowley]

Sure, but that means it's 1:0 for Sonic because it has one issue less than Nero. :D

yeah..but what is the Reference ? As i say i dont know i f we can compare EAC3 and AC3 so easy..dont think so

However..most reencodes from EAC sound better as DVD AC3

[honai]

most reencodes from EAC sound better as DVD AC3

... in your own humble opinion. ;)

Maybe it's time for another double-blind listening test: DRCed EAC3->AC3 vs AC3 from DVD. :)

[ACrowley]

... in your own humble opinion. ;)

Maybe it's time for another double-blind listening test: DRCed EAC3->AC3 vs AC3 from DVD. :)


Hey Kollega..."humble opinion" ? Cool down!
Sometimes a reencode from a 1536Kbps EAC3 sounds better as a 384Kbps 5.1 from DVD

[honai]

Sorry if it came across as offensive, it wasn't meant to be. I was just trying to point out that the general claim "most reencodes from EAC sound better as DVD AC3" shouldn't be made without more empirical data.

[ACrowley]

Sorry if it came across as offensive, it wasn't meant to be. I was just trying to point out that the general claim "most reencodes from EAC sound better as DVD AC3" shouldn't be made without more empirical data.

No Problem....
Ofcourse you can say it Generally..
But for example a reencode from 1.5Mbps EAC3 to 640Kbps AC3 or to 768kbps 24Bit WMA 5.1 (when i encode WVC1) sounds better ,compared with a standard 384kbps DVD AC3..not always, but often.
Thats what i mean

And about the AC3-EAC3 Comparison .. How was the Wave decoded from AC3 ?
I mean there can be a huge difference between the Methods how the wave was decoded from AC3
Example :
Source is AC3 (untouched) 448kbps.
Decoded with :ffdshow raw dump(no DRC)/ SteinbergNuendo(no DRC)/ Tranzcode 0.4(no DRC)/ Besweet(no DRC in Command)

Example :

http://nwgat.net/woot/files/57/other/Audacity%20AC3_thumb.png
(http://nwgat.net/woot/files/57/other/Audacity%20AC3.png)

You can see there the Waveform is different. All without DRC.
I would say Steinberg Nuendos AC3 Decoder is Refernce because its a off. Pro. Dolby Decoder/Encoder ( same Result with Digigram,Softencode Decoder)

[arty]

interesting ... I made my comparsions with azid too ! (besweet uses azid)

maybe it decodes wrong? I haven't applied normalisation either but the waveform looked similar to your example...

[ACrowley]

interesting ... I made my comparsions with azid too ! (besweet uses azid)

maybe it decodes wrong? I haven't applied normalisation either but the waveform looked similar to your example...
Maybe it has something to do with the Downmix Overflows or DRC is active even its not present in the commandline

A made a RMS Calculation on all Center Wave samples.
Source is AC3 448Kbps 5.1 has -27db DialogNormalization

- Nuendo AC3 Decoder,Soft Encode Decoder, ffdshow Dump : -30,4 db
- Besweet : -20db
- Cyberlink AudioDecoder :-24db
- SonicCinemasterAudioDecoder 4.2 : - 22,08db
- NeroAudioDecoder2 : -30,4db

Heres the Wavestructure in Audacity from Reference Nuendo compared with Nero and Besweet and Sonic4.3


http://nwgat.net/woot/files/57/ok_thumb.png
(http://nwgat.net/woot/files/57/ok.png)

I cant see anything wrong with NeroAudioDecoder2, its has the same Values and Structure as the Dolby Decoder from Nuendo and SoftEncode



PS, Sry my English is terrible

[ACrowley]

Now the Tests with AC3 and EAC3 Samples

1. AC3 with Nuendo (its the same Result with ffdshow dump and Softencode Dolby Decoder)
2. EAC3 with Nero
3. EAC3 with Sonic


http://nwgat.net/woot/files/57/other/eac3_thumb.png
(http://nwgat.net/woot/files/57/other/eac3.png)

The AC3 was decoded with SteinbergNuendo 3.2
EAC3 with Nero and Sonic

RMS Values :
- AC3 with Nuendo : -30,9db ( same with ffdshow dump and Softencode decoder)
- EAC3 with Nero : -30,5db
- EAC3 with Sonic : -22,4db

[arty]

thanks for the explanation, seems really professional :)

don't you have the same track in TrueHD ? So you could add a decoded trueHD wav into the comparsion ... (1 x ac3 + 2 x eac3 + 1 x truehd)

[ACrowley]

thanks for the explanation, seems really professional :)

don't you have the same track in TrueHD ? So you could add a decoded trueHD wav into the comparsion ... (1 x ac3 + 2 x eac3 + 1 x truehd)


i will see...I cant say anthing sure for TrueHD at the Moment

[tebasuna51]

interesting ... I made my comparsions with azid too ! (besweet uses azid)

maybe it decodes wrong? I haven't applied normalisation either but the waveform looked similar to your example...
To clarify the behavior of ac3 decoders I make a test with:

- NicAudio.dll (AviSynth plugin)
- Azid (standalone or with BeSweet)
- Cyberlink PowerDvd Audio decoder (Dolby Digital certified ?)
- ffdshow

Decoder test over a 5.1 regular ac3 with Dialog Norm. = -27 dB
All decoders without Dynamic Range Compression.

(dB values) NicAudio Azid() Azid(-n1) Cyberlink ffdshow
-------------- -------- ------- --------- --------- -------
RMS power ch0: -29.31 -29.31 -33.30 -33.31 -33.29
RMS power ch1: -27.90 -27.90 -31.91 -31.92 -31.91
RMS power ch2: -20.21 -20.21 -24.20 -24.22 -24.20
RMS power ch3: -47.67 -47.68 -51.68 -51.68 -51.67
RMS power ch4: -31.46 -31.46 -35.47 -35.48 -35.46
RMS power ch5: -33.61 -33.62 -37.62 -37.63 -37.62

Like you can see NicAudio and Azid without parameters are 4 dB loud than Cyberlink, ffdshow and Azid(-n1).

When we send the -n1 parameter to Azid is: apply the Dialog Normalization (31 - 27) = 4 dB attenuated.

Then Cyberlink and ffdshow apply this attenuation by default (correct because are players).

But if we want recode the signal, we don't need this attenuation, then the adequate decoders are Azid() or NicAudio (or apply +4 dB to others decoders)

Edit: SoftEncode Decoder have four decode modes with results between NicAudio and more attenuated (drc applied).

[ACrowley]

My RMs Values was taken from Adobe Audition..
On the Center Channel only..

With certified i mean the Decoder/Encode from Nuendo is a prof. Dolby Software, not a Dshow Decoder like Cyberlink with a Playback license from Dolby
I think Nuendo is a Reference Decoder.
The Values from Nuendo (or Digigran,Softencode) are always around -30db, Azid always around -22-25db

Softencode Decoder with DRC "Line Out" should be (?) "no DRC" it has ~ -30db in my Tests
Line Out means for me it goes out as it comes in, without the touch from DRC...i think so

[tebasuna51]

With certified i mean the Decoder/Encode from Nuendo is a prof. Dolby Software

If you are a owner of a certified encoder/decoder you can make a definitive test:

Use a 5.1 wav and encode to ac3, after decode to wav, the rms values from source and decoded wav must be the same.

[ACrowley]

If you are a owner of a certified encoder/decoder you can make a definitive test:

Use a 5.1 wav and encode to ac3, after decode to wav, the rms values from source and decoded wav must be the same.

Yep ,its the same..

I decode AC3 to wave in Steinberg Nuendo with the Dolby Decoder.
Then the reencode with Steinberg Nuendos Dolby encoder.

Always with -31db DialogNormalization and no DRC.
So i get the same Volume Level compared to the wave source

And as i say, Nero is always right with ~- 30db

[tebasuna51]

I decode AC3 to wave in Steinberg Nuendo with the Dolby Decoder.
Then the reencode with Steinberg Nuendos Dolby encoder.

Always with -31db DialogNormalization and no DRC.
So i get the same Volume Level compared to the wave source

And as i say, Nero is always right with ~- 30db
Calling a0.ac3 the ac3 source, decoded to a0.wav.

Encode to a1.ac3 with -27db (or other value distinct of -31) DialogNormalization and no DRC.

Decode to a1.wav with Steinberg Nuendo Dolby decoder, to a2.wav with Azid and to a3.wav with Azid -n1.

Please put:
Rms a0_center (source)
Rms a1_center (Nuendo decoded)
Rms a2_center (Azid decoded)
Rms a3_center (Azid -n1 decoded)

[ACrowley]

Your requested Test on AC3 from DVD 448kbpos Center Ch, (90% without) Dialog


http://nwgat.net/woot/files/57/other/test_thumb.png
(http://nwgat.net/woot/files/57/other/test.png)

RMS a0_Center : -25.23db
RMS a1_Center : -25.24db
RMS a2_Center : -25.24db
RMS a3_Center : -25.25db

No difference in Waveform. No difference in loud Environmet Sounds.

##################################

Other Tests on the untouched AC3 Source (this new Sample works better as my Test above)
RMS Values :

a0_Center Nuendo : -25.24db
a0_Center azid no DRC : - 12.38db
a0_Center azid with DRC : -21.22db
a0_Center Tranzcode, no Option : -21.23db


The disabled DRC from azid works well, cause some Parts in the Wave with loud Environmet have Full Dynamics, no Compression
With DRC the whole wave is small.
Nuendo and Tranzcode wave looks flat all time long...I dont know if its only a Volume difference, because i cant imagine Nuendo applies DRC ?


http://nwgat.net/woot/files/57/other/nuendo_thumb.png
(http://nwgat.net/woot/files/57/other/nuendo.png)

When you compare it now with the Results above...why are all louder Parts from the Source wave are not present as like as from Besweet/ffdshow ?
I cant Image Nuendo applies DRC..but it looks like DRC, no dynamic Parts anymore. When i compare it with Nero its the same

[tebasuna51]

With your last test seems:

- Nuendo apply DRC and Dialog Norm.
- Azid-DRC only DRC
- Azid nothing (full dynamic)
- ffdshow only Dialog Norm.
- Tranzcode only DRC

[madshi]

You guys are confusing me. Now I'm totally unsure which E-AC3 and AC3 decoder should be used...

:confused:

Anyway, just keep on researching. Hopefully we'll find out exactly which decoder is the way to go. Personally, I've just done some TrueHD checks. Here are my results:

(1) In Nero ShowTime the DRC setting shows an effect. Basically with DRC turned on the peaks are not as loud as without DRC turned off, but the rest of the audio track is more or less unchanged.

(2) With eac3to or GraphEdit, the Nero output is identical to Nero ShowTime with DRC turned *OFF*. It doesn't matter if DRC is turned on/off in ShowTime. In eac3to and GraphEdit DRC seems to be always turned off.

(3) PowerDVD knows 3 different settings: Quiet, Normal and Noisy. In my tests Quiet and Normal are identical and the WAV looks similar to Nero's output with DRC turned off. When using PowerDVD with "Noisy" the output is even more compressed than with Nero DRC turned on.

(4) Sonic is totally useless. It's much louder than anything else. And there's noise in every channel. Strangely, if you compare the Sonic WAV to Nero/PowerDVD, the center information seems to talk over into Left/Right channels in the Sonic decoder. Something fishy going on there.

(5) Unfortunately Nero's output for 16bit TrueHD tracks contains full 24bit information. That means, Nero must be doing some processing on the 16bit decoder output. It doesn't seem to be DRC to me. Must be something else. This is bad for reencoding, obviously.

(6) Unfortunately PowerDVD always only output 16bit, even if the TrueHD track is 24bit!! As a result I'm not totally sure if PowerDVD leaves the 16bit decoder output untouched. Maybe it does, maybe not.

It seems that there is no really good TrueHD available yet. Nero's and PowerDVD's WAVs look quite similar in Audacity, but the bits are different. PowerDVD is always 16bit, Nero is always 24bit.

All I wrote above is strictly only about TrueHD decoding.

[madshi]

Nero DRC off / Nero DRC on
http://madshi.net/TrueHD/Nero.noDRC_.png (http://madshi.net/TrueHD/Nero.noDRC.png)http://madshi.net/TrueHD/Nero.DRC_.png (http://madshi.net/TrueHD/Nero.DRC.png)

PowerDVD DRC off / PowerDVD DRC "noisy"
http://madshi.net/TrueHD/PowerDVD.noDRC_.png (http://madshi.net/TrueHD/PowerDVD.noDRC.png)http://madshi.net/TrueHD/PowerDVD.Noisy_.png (http://madshi.net/TrueHD/PowerDVD.Noisy.png)

Sonic
http://madshi.net/TrueHD/Sonic_.png (http://madshi.net/TrueHD/Sonic.png)

[d0ORk]

I tried to convert the audio stream that EVODemux made.
But I'm getting an error:
Muxing evo's first audio track to raw. Please wait...
The demuxer didn't accept the source file.

Is it an issue with EVODemux?

[madshi]

Is it an issue with EVODemux?
That's very hard to say. It could be an issue with EvoDemux. But more likely the Sonic filters refuse to work on your PC. Try playing the Evo with PowerDVD (if you have that installed). With the latest update patch PowerDVD player EVO files with only a TrueHD track just fine for me. If PowerDVD plays your EVO fine, then the EVO is probably ok.

[d0ORk]

mhh. the video works, but not the audio file that EVODemux made. At least, PowerDVD breaks down if I want to play it.
Any other idea how to demux the audio? Good is, that EVODemux merges the 2 evo files into one (video)

[madshi]

@d0ORk, which audio format is that?

[d0ORk]

it's DD+ on the Test HDDVD

[arty]

I think I completly lost :D

... of course thank you for your efforts!

[ACrowley]

With your last test seems:

- Nuendo apply DRC and Dialog Norm.
- Azid-DRC only DRC
- Azid nothing (full dynamic)
- ffdshow only Dialog Norm.
- Tranzcode only DRC

yep, youre right...theres obviously no other possibility. So it makes my 1st Test useless!!

Nuendo certified Dolby Decoder seems to apply both, DRC and Dialog Normalization.
Can be true ,because you can see in the Options from Soft Encode Dolby Decoder and from Digigram, (both older certified Dolby Dec-Enc Versions) that youve few Options to change DRC and Dialog Norm for Ac3 to pcm decoding
But You cant change Decoder Options in Nuendo.
So Nuendos/Cubase default seems to be DRC/Dialog Norm Output, not nice..all my last AC3 reencodes from last the Years are all from Nuendo 3.2 :)

Anybody know which Decoder Option is 100% "no" DRC AND Dialog Normalization in SoftEncode ? There are following DRC Decoder Options :
-Custom Analog /Dialog Norm
-Custom Mode / Digital Dialog Norm
-Line Out
-RF Remod Mode

The Softencode wave and RMS Values looks as like as Besweet (no DRC enabled) when RF Remod Mode is active. All other Modes are as like as Nuendo or Tranzcode.

I found this :
"Normal is similiar to 'Line out mode' and Heavy is similiar to 'RF-remod mode' of Sonic Foundary Soft Encode. It seems the WinDVD directshow filter (iviaudio.ax) invokes Heavy DRC also. However, Heavy is NOT recommended
because it usually results in obvious volume transitions, especially for music titles. "-Hiroko Hori (author of DVD2AVI) "

So ,why is the Wave with RF Mod from SoftEncode Output (DRC according to the Posting) the same compared with Besweet and "no" DRC ? Its all reversed ?!
Soft Encode Setting at "Line Out" is as like as Nuendo

All i want is to find a reference to decode AC3 to Wave "without DRC and Dialog Normalization.
Then i can compare TrueHD/EAC3 with AC3

[madshi]

it's DD+ on the Test HDDVD
Well, demuxing + decoding via eac3to should work fine. There's no need to keep DD+ inside the EVO.

[d0ORk]

It cant decode it

Q:\test>"D:\_eac3to\eac3to.exe" "Q:\test\PEVOB_1_MERGED.rebuilt.
EVO" "Q:\test\PEVOB_1_MERGED.DD.6ch.s03.German 5.1.ac3" -448 -24
Muxing evo's first audio track to raw. Please wait...
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.07
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid or empty chunk in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, tho
ugh.

Q:\test>

[madshi]

Try another Aften build. Maybe revision 449 or 433.

[d0ORk]

I just watched the .raw file while demuxing. It's just the name like 'example.raw' but it's always 0byte :(

[tebasuna51]

All i want is to find a reference to decode AC3 to Wave "without DRC and Dialog Normalization.
Then i can compare TrueHD/EAC3 with AC3

Like I say you, use Azid-BeSweet, NicAudio-AviSynth or Foobar also.

[ACrowley]

Like I say you, use Azid-BeSweet, NicAudio-AviSynth or Foobar also.


Yeah..sure.
To bad that Nuendo decodes with DRC/DN without Options to change it ( 1000$ Software!)

But still 2 Questions :

1. Which Softencode DRC Mode is without DRC/DialogN. ?
2. Besweet Downmix Overflows
Its no Problem so far i know..only a warning and is relevant when downmixing to 2Ch right ?

greets

[d0ORk]

Somehow I got it to work. :)
But theres a problem left...

Q:\test>"D:\_eac3to\eac3to.exe" "Q:\test\PEVOB_1_MERGED.DD.6ch.s
03.German 5.1.ddp" "Q:\corpse_demux\PEVOB_1_MERGED.DD.6ch.s03.German 5.1.ac3"
-448
E-AC3, 5.1 channels, 1:17:12, 640kbit/s, 48khz
Muxing eac3 file to raw. Please wait...
The NCBENUM return code is: 0x0
The NCBENUM return code is: 0x0
Audio Decoder output was 24 bit this time.
Unfortunately the Audio Decoder has output only 2 channels.
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: Signed 24-bit 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

progress: 100% | q: 233.2 | bw: 34.0 | bitrate: 448.0 kbps

Done.

Q:\test>

How can I get it to be 5.1?

Thanks for your help

[arty]

Yeah..sure.
To bad that Nuendo decodes with DRC/DN without Options to change it ( 1000% Software!)

But still 2 Questions :

1. Which Softencode DRC Mode is without DRC/DialogN. ?
2. Besweet Downmix Overflows
Its no Problem so far i know..only a warning and is relevant when downmixing to 2Ch right ?

greets

I got overflows even when set -6ch in besweet ... (decode to 6 mono waves). So the resulted wavs' waveform seems a bit "flat" where they reached the "maximum" ... :(

[madshi]

Unfortunately the Audio Decoder has output only 2 channels.
Try changing the Sonic filter settings (see Control Panel -> CinePlayer).

[d0ORk]

it works.
Thanks for all your help.
Now I'm trying to get the thing to .mkv with megui.
Wish me some luck. :rolleyes:

[ACrowley]

I got overflows even when set -6ch in besweet ... (decode to 6 mono waves). So the resulted wavs' waveform seems a bit "flat" where they reached the "maximum" ... :(

So far i know it has no Effect. Its only a Warning.
But maybe it can cause Problems when you donwmix 5.1 to stereo.
But when you decode all channles it will not have any Effect

Anway, The Optimum would be to decode AC3 to wave with certified Dolby Decoder.
To Bad Nunedo outputs DRC without any Decoder Options ( 1000$!)
Soft Encode/Digigram has a Dolby Decoder too ( Dolby Engine too)..but as i say above i dont know wich Option means 100% "No" DRC/D.N

Besweet/Foobar is fine so far

[tebasuna51]

So far i know it has no Effect. Its only a Warning.
But maybe it can cause Problems when you donwmix 5.1 to stereo.
But when you decode all channles it will not have any Effect
The typical azid warning:
W7: Downmix overflow (1: +0dB)
is not related with channels downmix and is produced when the decoder, translating from frequency domain to time domain, reach a value upper to 0 dB and can't be translated exactly.

Is not a problem and must be assumed like others imperfections in the encoder/decoder process with lossy formats.

Soft Encode/Digigram has a Dolby Decoder too ( Dolby Engine too)..but as i say above i dont know wich Option means 100% "No" DRC/D.N
But when RF Remod is similar to heavy(=loud) it should be the Full Dynamic Range, it makes Sense. Lineout= DRC+DN
- Custom analog: DRC only
- Custom digital: DRC + DN
- Line Out: also DRC + DN
- RF Remod Mode: DRC + Gain (+8 dB in my sample but I don't know the reason)

Then RF Remod Mode is not Full Dynamic Range.

[ACrowley]

The typical azid warning:
W7: Downmix overflow (1: +0dB)
is not related with channels downmix and is produced when the decoder, translating from frequency domain to time domain, reach a value upper to 0 dB and can't be translated exactly.

Is not a problem and must be assumed like others imperfections in the encoder/decoder process with lossy formats.


- Custom analog: DRC only
- Custom digital: DRC + DN
- Line Out: also DRC + DN
- RF Remod Mode: DRC + Gain (+8 dB in my sample but I don't know the reason)

Then RF Remod Mode is not Full Dynamic Range.

Yes, youre Right, thats waht it looks like :

http://nwgat.net/woot/files/57/other/soft_thumb.png
(http://nwgat.net/woot/files/57/other/soft.png)

I made AC3 reencodes from all waves . These Settings should keep alwys Source Volume Level:
http://forum.doom9.org/showthread.php?t=118759&highlight=Sony+Vegas

The AC3 Reencode from Azid Waves with disabled DRC/DN sounds perfect :) Same Dynamics/Wavestructure/RMS Value ,compared with the AC3 Source.

Reencodes from SoftEncode "RF Mode Wave" plays at lower Volume in loud Dynamic Parts but a little bit gained...so youre Right
I cant find any Soft Encode decoding Mode with all disabled and full Dynamic Range..strange

At least you can see in the Screenshot NeroAudioDecoder decodes AC3 with full DRC+DialogNormalization.
No Effect from Showtime Options. Sonic applies DRC and or DialogN. too.

However , the AC3 decoding Thing is cleared for me. THX :)

[madshi]

@ACrowley, could you make that test ac3 file available, please?

[ACrowley]

Once again with TrueHD ,This is from the 300 HDDVD :


http://nwgat.net/woot/files/57/other/300_thumb.png (http://nwgat.net/woot/files/57/other/300.png)

The Decoded Wave has less dynamics as like as the DVD AC 448kbps

I tried to change the NeroShowtime Options, but no Effect!

@ACrowley, could you make that test ac3 file available, please?
I used this AC3 file for my last Test.
This AC3 Sample is perfect for a Testrun because it has a lot of dynamic Sounds and a Speech Part

http://rapidshare.com/files/45393728/untouched_AC3_Reference_Sample.ac3.html

EDIT:
...can be true maybe its only DalogNormalization
I will compare it with Azid and DialogNormalization

[madshi]

Nero applies DRC on TrueHD Decoding. The Decoded Wave has less dynamics as like as the DVD AC 448kbps
Are you sure that it's not just lower volume (dialnorm?)? Looking at the screenshot it seems to me that everything is just not as loud as in the AC3 WAV.

In your previous tests with the Serenity AC3, the first half of the AC3 was significantly compressed with the DRC decoders, but the 2nd half was more or less identical to the non DRC decoders. So there it was clearly visible that the DRC compresses some parts of the track but not everything. In your 300 test I don't see this behaviour. It seems to me that the whole TrueHD decode is just of lower volume.

Or am I wrong?

I used this AC3 file for my last Test.
This AC3 Sample is perfect for a Testrun because it has a lot of dynamic Sounds and a Speech Part
Yeah, that's a very good test AC3. Thanks!

[madshi]

Here are my test results with Nero. I've tried the Nero filters through GraphEdit. And I've also checked what Nero outputs if you use Nero "ShowTime". It seems that the DRC setting in ShowTime does affect AC3 decoding, but only inside of ShowTime. When using the Nero filter in GraphEdit, AC3 decoding seems to have very strong DRC applied. Here are the results:

http://madshi.net/TrueHD/neroac3_.png (http://madshi.net/TrueHD/neroac3.png)

First wav is "Nero filter in GraphEdit". Second wav is "ShowTime with DRC turned on to 50%". Third wav is "ShowTime with DRC turned off". The third wav looks fine to me. It seems to have dialnorm applied, but no DRC.

[Wilbert]

Try changing the Sonic filter settings (see Control Panel -> CinePlayer).
Any idea what the environment setting is doing in the audio tab?

[madshi]

Any idea what the environment setting is doing in the audio tab?
Nope.

[ACrowley]

Are you sure that it's not just lower volume (dialnorm?)? Looking at the screenshot it seems to me that everything is just not as loud as in the AC3 WAV.
Or am I wrong?

Yeah..youre Right ! Nero applies "only" DilaogNormalization on theTrueHD ! No DRC
However, the Difference is big compared with azid Full dynmics without DCR/DialogN

http://nwgat.net/woot/files/57/azid2_thumb.png
(http://nwgat.net/woot/files/57/azid2.png)

I will make more Test if i can get a difference when i change Showtime Options. But in my last Tests i cant

We could get a final Result when somebody compare 300 HDDVD True HD it with the 300 Bluray LPCM
---
@Madhsi
I ask you again.. ? How do you dump the wave from my AC3 Test sample in your Test without DRC ? Your Graph etc ?
I cant get any effect when changing Showtime Options.
I mean its a essetial Information...

[madshi]

Ok, it seems that Nero does do DialNorm in the TrueHD decoder. That explains nicely why Nero outputs a 16bit TrueHD track with full 24bit of information!

Two questions for Dolby experts:

(1) Can DialNorm be properly undone on the final WAV?

(2) Does the (E-)AC3 / TrueHD data contain information about whether (and how much) DialNorm shall be executed? If so, maybe we can manipulate the (E-)AC3 / TrueHD data before sending them to the decoder?

[ACrowley]

Ok, it seems that Nero does do DialNorm in the TrueHD decoder. That explains nicely why Nero outputs a 16bit TrueHD track with full 24bit of information!

Two questions for Dolby experts:

(1) Can DialNorm be properly undone on the final WAV?

(2) Does the (E-)AC3 / TrueHD data contain information about whether (and how much) DialNorm shall be executed? If so, maybe we can manipulate the (E-)AC3 / TrueHD data before sending them to the decoder?

Mh, Im not sure.
But i think not after decoding to wave.
When the wave is compressed at lower Volume its bad. What will you do ?
The best is to avoid any DRC/DialogN...

But i think this is a good Conclusion :
Heres a Comparions with 300 HDDBD/Bluray (AC3,TrueHD,EAC3) Audio and it looks like :

1. "AC3" BluRay 640kbps untouched / to Compare it with : No DRC ,No DialogNorm
2. NERO "TrueHD" / applies DialogNorm, No DRC
3. NERO "EAC3" / applies DRC and DialogNormalization, even with changed Showtime Settings
4. SONICinemaster4.3 "EAC3" /looks like applies DRC and DialoN. + Gain
5. "AC3" Bluray 640kbps untouched / to compare it with : only DialogNorm.


http://nwgat.net/woot/files/57/other/sshot-3_thumb.png
(http://nwgat.net/woot/files/57/other/sshot-3.png)

Conclusion is

NeroAudioDecoder2 :
- not good for EAC3 cause it applies full DRC and DIalogNormalization
- Better then nothing for TrueHD cause its "only" DialogNormalization
- I find no Way to change Nero Settings via Showtime or Registry to change it

SonicCinemasterAudioDecoder4.2
- not so low Volume as Nero, but looks like DRC and or DialogNormalization with "DolbyAuidoGain" buuhh
- TrueHD decoding still not work without Backroundnoise with 4.3
- cant see any Changes with different Settings in Sonic System Panel

About the AC3 to wave decoding :
-Azid/Foobar/NiscAudio can decode it fully without DRC/DialogN, so i use it to keep Full Dynamics
-Steinberg Nuendos cert. Dolby Decoder applies DRC and DialogN. No Way to change Dolby Decoder Options
-SoftEncode cert Dolby Decoder works very strange. Cant find any untouched Output.

So the Target is to find a Way to change Decoding Options for Nero and Sonic. to get EAC3 /TrueHD without DRC/DialogN.
And ofcourse hoping for a fast ffmpeg EAC3 Decoder/Ac3Filter Release

[madshi]

Conclusion is

NeroAudioDecoder2 :
- not good for EAC3 cause it applies full DRC and DIalogNormalization
- Better then nothing for TrueHD cause its "only" DialogNormalization
- I find no Way to change Nero Settings via Showtime or Registry to change it

SonicCinemasterAudioDecoder4.2
- not so low Volume as Nero, but looks like DRC and or DialogNormalization with "DolbyAuidoGain" buuhh
- TrueHD decoding still not work without Backroundnoise with 4.3
- cant see any Changes with different Settings in Sonic System Panel

About the AC3 to wave decoding :
-Azid/Foobar/NiscAudio can decode it fully without DRC/DialogN, so i use it to keep Full Dynamics
-Steinberg Nuendos cert. Dolby Decoder applies DRC and DialogN. No Way to change Dolby Decoder Options
-SoftEncode cert Dolby Decoder works very strange. Cant find any untouched Output.
Yes, I think you've nailed it. However, there's one more option: Namely playing AC3 and E-AC3 inside of ShowTime and capturing the output (possible with a little hacking). This should probably help getting rid of DRC for AC3 and E-AC3 with Nero. However, I don't know yet how to get rid of DialNorm.

-----

I'm still wondering whether DialNorm can't be undone afterwards. I mean it's "only" a volume change. Can't we turn up volume again in the PCM/WAV file to undo that change? I'm not sure how much harm would be done by this "volume down + volume up"?

[ACrowley]

Yes, I think you've nailed it. However, there's one more option: Namely playing AC3 and E-AC3 inside of ShowTime and capturing the output (possible with a little hacking). This should probably help getting rid of DRC for AC3 and E-AC3 with Nero. However, I don't know yet how to get rid of DialNorm.

-----
I'm still wondering whether DialNorm can't be undone afterwards. I mean it's "only" a volume change. Can't we turn up volume again in the PCM/WAV file to undo that change? I'm not sure how much harm would be done by this "volume down + volume up"?

yep....

However. AC3 decoding is not necesarry because azid/Foobar works optimal.
It was only own Target to clear it because i encode a lot of
AC3s.


Hmm...The DialogNormalization is not only a simple overall Levelchange ,isnt it ? Its a dynamic Thing too?
So a simple Volume + would not help..dont know

heres a good EAC3 Sample for Testing
http://rapidshare.com/files/45531448/300_EAC3_640kbps.rar.html

Can you try Showtime Output Setting "Passtrough" with your Tweak?
"Passes Audio to your Soundcard without further processing"

And
1. What is your Showtime Capture/Dump Tweak ?
2.in your other Test you use PowerDVD for decoding TruehD ? how ??

[tebasuna51]

Hmm...The DialogNormalization is not only a simple overall Levelchange ,isnt it ? Its a dynamic Thing too?
So a simple Volume + would not help..dont know

Apply the Dialog Normalization is always a fix attenuation for all the stream. Each ac3 frame (32 ms. for 48 KHz.) have a Dialog Normalization value but must be the same for all frames in the stream.

The DRC attenuation/gain value can vary for each block (5.33 ms., 6 blocks per frame).

A gain of (31 - DialogNorm) dB can restore the original volume and must be used to reencode to ac3 using the same DialogNorm and DRC settings.

[madshi]

Apply the Dialog Normalization is always a fix attenuation for all the stream. Each ac3 frame (32 ms. for 48 KHz.) have a Dialog Normalization value but must be the same for all frames in the stream.
Do you happen to know which bits in the AC3 header tell the decoder how much Dialog Normalization is needed? Do you think we can just modify those bits to get rid of DialNorm?

A gain of (31 - DialogNorm) dB can restore the original volume and must be used to reencode to ac3 using the same DialogNorm and DRC settings.
So DialNorm 31 means no change at all, right?

[ACrowley]

Do you happen to know which bits in the AC3 header tell the decoder how much Dialog Normalization is needed? Do you think we can just modify those bits to get rid of DialNorm?
So DialNorm 31 means no change at all, right?

@tebasuna51

When i "reencode" a AC3 this should be correct : (?)

1. AC3 to wave with AZID , no DRC/DialogN.
BeSweet.exe -core( -input "E:\test.ac3" -output "E:\test.wav" -6ch )

2. Reencode with Sony Vegas Prof Dolby Encoder v7 ( or a other cert. Dolby Encoder) and these Settings :
http://forum.doom9.org/showthread.php?t=118759&highlight=Sony+Vegas

-31db DialogNormalization, no DynamicRangeComp ,no other Settings
It will guarantee you the volume level of the output AC3 will be the same as the original Wave?

So i dont have to care about Source AC3 Values because i get the Source Wave Levels.
Must be correct because i get the same Wavestructure and RMS Value as like as the AC3 Source (based on azid decoding)

The other,full Dolby Complaint, Method (when i want produce AC3) is to use a DRC Profile in Combi with a proper DialogNom. Value.
To get this correct DialogNormalization Value, i have to run a RMS Calculation and set it for the DialogN...right ?

[tebasuna51]

Do you happen to know which bits in the AC3 header tell the decoder how much Dialog Normalization is needed?
Bits 55-59 for 2/0 (stereo) ac3
Bits 57-61 for 3/2 (surround) ac3
Five bits make a value between 0 and 31
Do you think we can just modify those bits to get rid of DialNorm?
Yes, but you need recalculate also the CRC values (bytes 3-4 at the begin of the frame, I don't remember now if we need recalculate the ending CRC frame also) if you want the decoder accept the frame.
So DialNorm 31 means no change at all, right?
Ok, all these bits to 1.

Maybe you can read this post (http://forum.doom9.org/showthread.php?p=964460#post964460) and related.

[madshi]

Bits 55-59 for 2/0 (stereo) ac3
Bits 57-61 for 3/2 (surround) ac3
Five bits make a value between 0 and 31

Yes, but you need recalculate also the CRC values (bytes 3-4 at the begin of the frame, I don't remember now if we need recalculate the ending CRC frame also) if you want the decoder accept the frame.

Ok, all these bits to 1.

Maybe you can read this post (http://forum.doom9.org/showthread.php?p=964460#post964460) and related.
Thank you!! :)

How about mono AC3? And do you know the bits for E-AC3 and TrueHD, too?

[tebasuna51]

When i "reencode" a AC3 this should be correct : (?)

1. AC3 to wave with AZID , no DRC/DialogN.
BeSweet.exe -core( -input "E:\test.ac3" -output "E:\test.wav" -6ch )

Correct, now you have the full dynamic range of original source.

2. Reencode with Sony Vegas Prof Dolby Encoder v7 ( or a other cert. Dolby Encoder) and these Settings :
http://forum.doom9.org/showthread.php?t=118759&highlight=Sony+Vegas

Like Skelsgard say in the post:
"This project will guarantee you the volume level of the output AC3 will be the same as the original WAV"

But is not Dolby compliant, with the settings DialNorm=31, DRC=None you obtain a kind of encode similar to ogg, mp3, ... without the 'advantages' of Dolby normalization (the dialogs must be at -31 dB, great dynamic range but low RMS ...).

The problem is listen an ac3 and after a CDA or mp3 with modern music (low dynamic range with high RMS). This is the origin of the 'low volume' problem of ac3 streams.

-31db DialogNormalization, no DynamicRangeComp ,no other Settings
It will guarantee you the volume level of the output AC3 will be the same as the original Wave?

There are decoders that can be instructed to ignore the BSI values but we can't do anything about that. For instance the Ac3Filter have a DRC function than ignore the values in the ac3 stream and analyze the signal to make a 'dynamic' DRC.

The other,full Dolby Complaint, Method (when i want produce AC3) is to use a DRC Profile in Combi with a proper DialogNom. Value.
To get this correct DialogNormalization Value, i have to run a RMS Calculation and set it for the DialogN...right ?
Ok. You can read the Sticky: GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (http://forum.doom9.org/showthread.php?t=56020)
and remember select only a fragment with representative dialog volume (maybe only in central channel) to get the rms value for DialNorm

[tebasuna51]

How about mono AC3?
For mono AC3 the DialNorm bits are 51-55
And do you know the bits for E-AC3 and TrueHD, too?
About E-AC3 you can read the Anex E of Digital Audio Compression Standard (AC-3, E-AC-3) Revision B (http://www.atsc.org/standards/a_52b.pdf) (still in my 'todo' list)

About TrueHD, I don't know documentation.

Edit: For E-AC3 DialNorm is always bits 46-50, don't have CRC at the beginning but yes at the end.

[madshi]

@tebasuna51, thanks very much again for the helpful information! Do you happen to have a link or maybe even some code about how to recalculate the (E-)AC3 CRC?

[tebasuna51]

@madshi
Sorry but today is my first approach to eac3.

[honai]

@madshi

Could you please update your first post with the findings concerning DRC/Dialog Normalization in Nero Audio Decoder? Thanks.

[ACrowley]

Correct, now you have the full dynamic range of original source.
Like Skelsgard say in the post:
"This project will guarantee you the volume level of the output AC3 will be the same as the original WAV"

But is not Dolby compliant, with the settings DialNorm=31, DRC=None you obtain a kind of encode similar to ogg, mp3, ... without the 'advantages' of Dolby normalization (the dialogs must be at -31 dB, great dynamic range but low RMS ...).

The problem is listen an ac3 and after a CDA or mp3 with modern music (low dynamic range with high RMS). This is the origin of the 'low volume' problem of ac3 streams.

There are decoders that can be instructed to ignore the BSI values but we can't do anything about that. For instance the Ac3Filter have a DRC function than ignore the values in the ac3 stream and analyze the signal to make a 'dynamic' DRC.

Ok. You can read the Sticky: GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (http://forum.doom9.org/showthread.php?t=56020)
and remember select only a fragment with representative dialog volume (maybe only in central channel) to get the rms value for DialNorm
yep.
I use AdobeAudition 2 for RMS calculation to get DialogNormalization Value .
I use Adobe Audition.

Wich Method would "you" pers recommend ?
- Measure RMS ,set it for DialogNormalization with DCR Profile "Film".
-Or all disabled , -31db always for DialogNorm

1 AC3/EAC3 to AC3 "re"encoding
2. TrueHD/LPCM/DTS encoding to AC3 ?

I mean most guys i know use -31db Dialog Norm with no DRC.
I think eac3to/aften use it too.
I can remember there where some heavy discussions in other Forums.
Ofcourse, as i say the -31db/no DRC Method isnt DolbyCompliant, but it will guarantee Source Wave Volume Level.

Comparison from Source/Dolby Conform/ -31db not Conform
http://img509.imageshack.us/img509/6803/dolbykj2.th.png (http://img509.imageshack.us/my.php?image=dolbykj2.png)
they all have the same RMS Values and Wavestructure

I pers use -31db with no DRC when i reencode from AC3/EAC3.
When i compare the wave from Source AC3 with the reencode. it has the same RMS Values and Wavestructure
And i play AC3 on PC via AC3Filter (no DRC enabled) or SPDIF to my Surround Receiver/Speaker


For channel upmixes( more a "producing" as reencodig) RMS for DialogN+DRC
But i think maybe its better to encode Dolby Compliant from lossless HD Audio ?
So, what is your Advice ?

THX

[tebasuna51]

@ACrowley
There are 3 options:

1) Encode to ac3 with DialNorm 31, DRC None.
Recommended for users with good audio equipment and without problems with childs or neighbors.

2) Encode with the proper DialNorm and DRC.
Recommended for users with control over the decoder (PC, good receivers).

3) Apply the DRC, Maximize (Peak method) and encode to ac3 with DialNorm 31, DRC None.
Users with low cost audio equipments and with the "low volume ac3 problem"

Obviously the second method is more flexible but...

- The Dialog Normalization concept from Dolby is useless if you listen TV audio, CDA, mp3 with the same equipment. Actually the -31 dB ac3 volume is far off the typical volume of others sources.

- The Dynamic Range Compression concept is very interesting with sources with wide dynamic range, is useless for modern music with high volume but short dynamic range.

- If the encoder to use is Aften, remember: the actual DRC routines are in beta stage with differences (http://forum.doom9.org/showthread.php?p=893111#post893111) with the behavior of commercial encoders.

[ACrowley]

Thank you

- No, i encode not with Aften, i use SonyVegas Dolby Digital Pro Encoder v7 or STeinberg Nuendos DD encoder.
Both are certified Dolby Encoder

The most AC3 Tracks i use are from movies has a wide Dynamic (HDDVD,Bluray)
Currently and in the Past i use Method 1.

Actually the -31 dB ac3 volume is far off the typical volume of others sources.


When i compare the Source (always azid decoded no DRC;DialogN) its looks exactly as like as encodes from both Methods and with the same RMS Values
http://img509.imageshack.us/img509/6803/dolbykj2.th.png (http://img509.imageshack.us/my.php?image=dolbykj2.png)

So far i can say , for my Receiver via SPDIF ( AC3Filter)

Source :sounds great, high Dynamics, big Differences between DRC Modes
Method 1 :sounds as good as the Source with DRC off. No big Differences between DRC Modes, a bit louder as the Source
Method 2 :sounds by far lower Volume with or without DRC, mmmhhh
-On PC via AC3Filter i cant hear any real differences

One last Question : Can Method 1 cause clipping ?
THX for all your help!

[madshi]

Good news!!

http://madshi.net/eac3to.zip

The latest eac3to version 1.10 now detects if E-AC3 tracks have Dialog Normalization activated (most tracks have) and removes it automatically! As a result by using eac3to Dialog Normalization with E-AC3 decoding is no longer a problem. (Thanks to tebasuna51 for the helpful tips!)

Furthermore I've found a way to talk the Nero E-AC3 decoder into not applying DRC, even though we're using it outside of ShowTime. That means: With the help of the Nero decoder we now have near perfect E-AC3 decoding. Enjoy!!

Finally, the Surcode automation was slightly improved. It now outputs some more messages. If it still fails for some people, please post the full eac3to output here. Thanks.

Because of the latest information about DRC and Dialog Normalization and the latest improvements in eac3to for the Nero decoder, I've now made Nero the default E-AC3 and TrueHD decoder. So the flag "/nero" is no more. There is a flag "/sonic" now to force the use of the Sonic filters. I do not recommend the Sonic filters, though, cause they still apply DRC and I don't know how to disable that.

@ACrowley, I hope you'll find some time to test the new version to make sure that E-AC3 decoding is really without faults now. Thanks!

[InorganicMatter]

SWEET!!

I think I shall reconvert the audio of the HDDVD I just finished converting to 720p AVC.

[honai]

Wow, that's amazing! Thank you so much!

[Rectal Prolapse]

Nice work madshi and the rest of you!

[ACrowley]

Good news!!

http://madshi.net/eac3to.zip

The latest eac3to version 1.10 now detects if E-AC3 tracks have Dialog Normalization activated (most tracks have) and removes it automatically! As a result by using eac3to Dialog Normalization with E-AC3 decoding is no longer a problem. (Thanks to tebasuna51 for the helpful tips!)

Furthermore I've found a way to talk the Nero E-AC3 decoder into not applying DRC, even though we're using it outside of ShowTime. That means: With the help of the Nero decoder we now have near perfect E-AC3 decoding. Enjoy!!

Finally, the Surcode automation was slightly improved. It now outputs some more messages. If it still fails for some people, please post the full eac3to output here. Thanks.

Because of the latest information about DRC and Dialog Normalization and the latest improvements in eac3to for the Nero decoder, I've now made Nero the default E-AC3 and TrueHD decoder. So the flag "/nero" is no more. There is a flag "/sonic" now to force the use of the Sonic filters. I do not recommend the Sonic filters, though, cause they still apply DRC and I don't know how to disable that.

@ACrowley, I hope you'll find some time to test the new version to make sure that E-AC3 decoding is really without faults now. Thanks!

Test Sample EAC3 640kbps 5.1 (300HDDVD)
http://img528.imageshack.us/img528/4598/sshot3he5.th.png (http://img528.imageshack.us/my.php?image=sshot3he5.png)

you can see ec3to 1.10 works now without DRC AND without DialogNorm (for EAC3). TrueHD is still with DialogNorm.
Madshi....nice work!:thanks:

SO the new Conclusion is :

TrueHD ,always, with NeroDecoder : DRC OFF ,Dialog ON
EAC3 "only" via "eac3to" with NeroDecoder : DRC OFF DialogN OFF

Later Today when im have Time i will show you that the Dynamics are as like as the AC3 640kbps from BluRay ! Nice

[madshi]

Test Sample EAC3 640kbps 5.1 (300HDDVD)
http://img528.imageshack.us/img528/4598/sshot3he5.th.png (http://img528.imageshack.us/my.php?image=sshot3he5.png)

you can see ec3to 1.10 works now without DRC AND without DialogNorm (for EAC3).
Thanks for confirming!

[ACrowley]

Heres a Comparison from AC3 640kbps Bluray and decoded EAC3 640kbps HDDVD (Nero eac3to 1.10) from 300

http://img257.imageshack.us/img257/8132/sshot3ih8.th.png (http://img257.imageshack.us/my.php?image=sshot3ih8.png)

Its obvious.
Thew 1.10 Version works perfect . No DRC ,No DialogNorm. anymore with NeroAudioDecoder

[honai]

Excellent!

I have observed an interesting effect.

I just encoded the English DD+ track from Batman Begins to DTS (w/ Surcode) and muxed the track alongside the DD track from the R1 DVD. With the DTS track dialog is considerably "louder" than the DD version. I route audio to my receiver over SPDIF (bit-perfect), and the receiver displays "DIAL. NORM +4" for the DTS track (Pioneer 1014). So it seems that DD+ -> DTS yields dialog loudness that is stronger than the "reference" from the DVD. Not that it sounds unpleasant. I'm just wondering if we aren't running into unintended consequences by ignoring the Dialog Normalization in the source DD+ track ...

[madshi]

I just encoded the English DD+ track from Batman Begins to DTS (w/ Surcode) and muxed the track alongside the DD track from the R1 DVD. With the DTS track dialog is considerably "louder" than the DD version. I route audio to my receiver over SPDIF (bit-perfect), and the receiver displays "DIAL. NORM +4" for the DTS track (Pioneer 1014). So it seems that DD+ -> DTS yields dialog loudness that is stronger than the "reference" from the DVD. Not that it sounds unpleasant. I'm just wondering if we aren't running into unintended consequences by ignoring the Dialog Normalization in the source DD+ track ...
As far as I understand it, the DD+ dialnorm stuff changes the volume of the whole track (all 6 channels). It doesn't just change the center channel. So I don't think that ignoring the DD+ dialnorm should have any specific effect on the center channel. But that's just AFAIK. Are you sure that it's not the whole track which is louder? That could be easily explained, because the original DD track will have dialnorm in it, but DTS has not. The the original DD track volume will be lowered by your receiver, while the DTS track should not be lowered. Maybe the receiver even raises volume for DTS tracks. That would result in even more loudness. But all this should affect all channels and not just the dialog (center).

[honai]

You're right, picked a few more listening spots, all channels are louder. The receiver doesn't do anything other than adhere to the standard. Maybe Surcode embeds some additional data?

In comparison I can say that the transcoded DTS is indeed louder than reference DTS tracks from retail DVDs.

EDIT:

On my receiver DTS tracks are typically less loud than DD tracks from the same DVD. I suspect that's because due to the algorithmic nature of DTS vs DD the DTS version allows for a greater dynamic range and thus needs more headroom, resulting in an overall lower volume. However, with Batman Begins the transcoded DTS is much louder than the original DD track.

Maybe the receiver even raises volume for DTS tracks.

Yes, that's what seems to be happening as indicated by the "+4" in "DIAL. NORM +4". For DD tracks the receiver doesn't indicate any volume adjustment.

[madshi]

So, should I add a switch to allow dialnorm to happen instead of always defeating it? Probably it makes sense to defeat dialnorm if you reencode in AC3, because then you can tell the AC3 encoder to reuse the original dialnorm value (eac3to 1.10 output the original dialnorm value for your information). But maybe when reencoding to DTS it would be better to let dialnorm happen during decoding? However, I think when decoding 16bit TrueHD tracks it would be better to defeat dialnorm to make lossless reencoding easier without quality loss and with small file sizes.

[honai]

Yes, make it an option, that'd be perfect!

EDIT:

Hm, come to think of it - maybe there is indeed something in the DTS header (from the Surcode encoder) that tells the DTS encoder in the receiver if Dialog Normalization should be applied. Perhaps we should ask someone who's familiar with the DTS standard. Using Google I found conflicting information, i.e. that DTS has/has no facilities for Dial. Norm. information in the stream.

EDIT2:

http://www.laaudiofile.com/images/dtsencoder.jpg

Seems to follow the DD standard of 27..31dB Dialog Normalization. Surcode doesn't expose that parameter, but the resulting DTS stream should definitely receive the Dial. Norm. value obtained from the original EAC3 stream.

[ACrowley]

I havent any Problems with reencoded AC3 when the DialogNorm was ignored through Source decoding

Always reencoded with -31db ,no DRC Profile

Should be no Problem

[madshi]

Here's a thread about DialNorm which I find worth reading:

http://www.hometheaterforum.com/htf/showthread.php?t=230926

[tebasuna51]

Yes, make it an option, that'd be perfect!

EDIT:

Hm, come to think of it - maybe there is indeed something in the DTS header (from the Surcode encoder) that tells the DTS encoder in the receiver if Dialog Normalization should be applied. Perhaps we should ask someone who's familiar with the DTS standard. Using Google I found conflicting information, i.e. that DTS has/has no facilities for Dial. Norm. information in the stream.

EDIT2:

http://www.laaudiofile.com/images/dtsencoder.jpg

Seems to follow the DD standard of 27..31dB Dialog Normalization. Surcode doesn't expose that parameter, but the resulting DTS stream should definitely receive the Dial. Norm. value obtained from the original EAC3 stream.

Of course DTS format have also a DialNorm field in the header (also have DRC support) but the solution isn't suply the original attenuated to the DTS encoder, the original must be at full dynamic and you can set the DialNorm in your DTS encoder if you want.

@madshi
Instead a option to apply or not the patch to override the DialNorm attenuation I think honai need a way to pass the original ac3 DialNorm value to the DTS encoder like parameter.

[madshi]

@madshi
Instead a option to apply or not the patch to override the DialNorm attenuation I think honai need a way to pass the original ac3 DialNorm value to the DTS encoder like parameter.
Unfortunately the Surcode DTS encoder doesn't seem to have an option to specify the DialNorm value.

Maybe I should manipulate the final DTS file to fill in the right DialNorm value? :)

[ACrowley]

Unfortunately the Surcode DTS encoder doesn't seem to have an option to specify the DialNorm value.

Maybe I should manipulate the final DTS file to fill in the right DialNorm value? :)


HKEY_CURRENT_USER\Software\Minnetonka Audio Software\Surcode DTS\OPTIONS

Theres Value for DialogNorm and DRC

Looks like default applies DialogN, maybe you should leave its on default...i never had Problems with Surcode encoded dts and DialogNorm and Dynamic Compress etc. Not with full Dynamic Sources and not with pre Dynam. compressed Sources

.mhh

@honai
I think DTS should sound louder as like as DD on most Receivers..not lower Volume.
On my Pioneer dts is always louder

[hristoff2]

Thanks once again, awesome. :)


Source: Shooter eac3 5.1 (1536kbps), English

top: eac3to 1.9 / Nero / Center CH
bottom: eac3to 1.10 / Nero / Center CH


http://www.imagebanana.com/img/ajezqffy/thumb/drc_no_drc_comp.png (http://www.imagebanana.com/view/ajezqffy/drc_no_drc_comp.png)

[madshi]

HKEY_CURRENT_USER\Software\Minnetonka Audio Software\Surcode DTS\OPTIONS

Theres Value for DialogNorm and DRC
I don't have these values there? How are they named exactly? Thanks...

Looks like default applies DialogN, maybe you should leave its on default...
I agree with tebasuna51 that I should forward the original E-AC3's DialNorm value to the DTS encoder. That makes more sense to me than letting the encoder use a default DialNorm value.

Why are you saying "applies"? There's nothing about DialNorm that the encoder "applies". The encoder does not do any DialNorm processing. All it does is writing a specific DialNorm number to the header of every encoded frame. DialNorm is "applied" by the decoder, not by the encoder!

[ACrowley]

I don't have these values there? How are they named exactly? Thanks...


I agree with tebasuna51 that I should forward the original E-AC3's DialNorm value to the DTS encoder. That makes more sense to me than letting the encoder use a default DialNorm value.

Why are you saying "applies"? There's nothing about DialNorm that the encoder "applies". The encoder does not do any DialNorm processing. All it does is writing a specific DialNorm number to the header of every encoded frame. DialNorm is "applied" by the decoder, not by the encoder!


Thats what i mean.. i told you my English is bad. I should better say the encoder applies the DialogNorm Info into the encode..better ?:)

Do you know how a DTS encoder handles Dialog Norm value ?
I dont if the dts encoder handles the encoding/decoding off DialogN as like as AC3,when -27db is in EAC3/AC3 if -27db is correct for a reencoded dts too? I really dont know the DTS tech Specs

But when you plan to set the DialogNorm from EAC3 into dts,it should be done for a TrueHD Source DialogNorm too ?
Mhhh i never thought about..all my dts reencodes are at default without any special DialogNorm Setting because theres simply no Option in Surcode/Nuendo

HKEY_CURRENT_USER\Software\Minnetonka Audio Software\Surcode DTS\OPTIONS
http://img252.imageshack.us/img252/4904/sshot3ge8.th.png (http://img252.imageshack.us/my.php?image=sshot3ge8.png)

[madshi]

Ok...Do you knw how a "Studio" DTS encoder handles Dialog Norm value ? Surcode is a certified DTS Encoder too..however, but more semi Pro
Maybe the Manual from DTS (HD) Master Suite shows how to do. I dont if the dts encoder handles the encoding/decoding with DialogN s like as AC3,when -27db is in EAC3 if -27db in dts is correct too? I really dont know
I don't know, either, but I would guess so... :)

P.S: You can test it out. Just set different DialNorm values in the Surcode registry keys. Then test encode in DTS. Then decode again and check the WAV.

But when you plan to set the DialogNorm from EAC3 ,it should be done for a TrueHD Source DialogNorm too ?
For TrueHD sources I would set DialNorm to -31dB in the DTS encoder because we currently can't turn DialNorm off during decoding.

So, how is the exact names of these registry values? :)

[madshi]

@ACrowley: Wow, you have many more values in your registry than I have! Which Surcode version are you using? The Dialnorm value in your registry is "-27". I wonder what effect (if any) the "3db Attenuation" registry value has? Do you have time to test that? Would be interested if it has any effect! Unfortunately I'm currently very short on time, so I can't test that myself right now...

[honai]

Same here.

The "3dB Attenuation" is also an option in the encoder GUI, AFAIK it's for rear-channel attenuation.

I'm using 1.0.21, so ACrowley's version is probably newer. Or it's a different product than "Surcode DVD Pro".

[ACrowley]

Same here.

The "3dB Attenuation" is also an option in the encoder GUI, AFAIK it's for rear-channel attenuation.

I'm using 1.0.21, so ACrowley's version is probably newer. Or it's a different product than "Surcode DVD Pro".


I use Surcode DVD DTS Pro 1.0.23

I will Test it. But what i know surely is that changing the DynamicCompress Reg.Setting has no Effect!
The wave has same Peaks compared with DRC off encoded

@Madshi
You say in the Screenshot its -27db ? I dont know how to set the DialogN. Values with Registry (Hexa)Dezimal Values ?
Then i can Test it

[madshi]

You say in the Screenshot its -27db ? I dont know how to set the DialogN. Values with Registry (Hexa)Dezimal Values ?
Then i can Test it
Use Windows calc. Enter "-27", then switch to "Hex". You'll get 16 chars. Cut off the first 8. The last 8 chars is what you need to put into the registry.

[d0ORk]

When I updated to the newest eac3to.exe I'm getting this:

Q:\test>"D:\_eac3to\eac3to.exe" "Q:\test\sample-long.DD.stream.00.ea
c3" "Q:\test\sample-long.DD.stream.00.ac3" -448
E-AC3, 5.1 channels, 0:01:08, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 37,50 MB.
The expected file size for 24 bit is 56,25 MB.
The real file size is 0 Bytes.

I evodemuxed the sample-long.evo and two others now.

[d0ORk]

Is there a download source for the previous version of eac3to.exe ? That one worked for me

[ACrowley]

I made a Test with Surcode dts pro 1.0.23 and the DialogNorm Registry Settings.

http://img66.imageshack.us/img66/6021/sshot3mi9.th.png (http://img66.imageshack.us/my.php?image=sshot3mi9.png)

All dts are decoded to wave with Tranzcode 0.4b

Theres absolutly no difference when i change the RegistryValues fro Dialog Normalization
And not whne changing Dynamci Compress

So it has no Effect so far i can see

[madshi]

Maybe Tranzcode ignores these values? Does Tranzcode ignore DialNorm for AC3 decoding? I think ffdshow would be a good test decoder cause it executes DialNorm for AC3 decoding, IIRC.

Another useful test would be to check if those DTS files are bit wise identical or whether there are any bytes different. If the DTS files are 100% identical then no decoder on earth can output a different result for the 3 DTS files.

[madshi]

When I updated to the newest eac3to.exe I'm getting this:

Q:\test>"D:\_eac3to\eac3to.exe" "Q:\test\sample-long.DD.stream.00.ea
c3" "Q:\test\sample-long.DD.stream.00.ac3" -448
E-AC3, 5.1 channels, 0:01:08, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 37,50 MB.
The expected file size for 24 bit is 56,25 MB.
The real file size is 0 Bytes.

I evodemuxed the sample-long.evo and two others now.
Don't know why that happens. It seems to work for everybody else? Do you have the Blu-Ray/HD DVD plugin correctly licensed/registered? And are you using the latest Nero version?

[tebasuna51]

Maybe Tranzcode ignores these values? Does Tranzcode ignore DialNorm for AC3 decoding? I think ffdshow would be a good test decoder cause it executes DialNorm for AC3 decoding, IIRC.

Another useful test would be to check if those DTS files are bit wise identical or whether there are any bytes different. If the DTS files are 100% identical then no decoder on earth can output a different result for the 3 DTS files.

I test with SurCode DVD Pro DTS v1.0.23.

1) Initially i don't have "DIALOG NORMALIZATION" at register

2) With "DIALOG NORMALIZATION"=dword:ffffffe5 (-27)

3) With "DIALOG NORMALIZATION"=dword:ffffffe1 (-31)

the three dts are bit identical.

BTW, in encoder options I can only choose:

-Data Rate:
1536 -> "DTS BITRATE INDEX"=dword:00000001
768 -> "DTS BITRATE INDEX"=dword:00000002

-Sample Rate:
48000 -> "SAMPLING RATE"=dword:00000000
44100 -> "SAMPLING RATE"=dword:00000001

-Attenuate rear channels 3 dB:
Yes -> "SURROUND DOWNMIX"=dword:00000001
Not -> "SURROUND DOWNMIX"=dword:00000000

[Exiton]

Don't know why that happens. It seems to work for everybody else? Do you have the Blu-Ray/HD DVD plugin correctly licensed/registered? And are you using the latest Nero version?

Err... I updated eac3to and I'm getting the same error messages.

[ACrowley]

@madshi

Tranzcode applies no DRC..thats sure
But i think DialogNormalization ,as like as ffdhow

But with AC3 Filer the decoded dts are always at bit higher Peaks (a bit), SO ithink AC3 Filter applies no DRC, no Dialog Norm
However, these 3 Test Samples are identical with all decoders..no Effect when changing the Values in Registry

[madshi]

the three dts are bit identical.
Argh. Ok, thanks for testing!

Are you absolutely sure that DTS really knows Dialnorm at all? I rather doubt it right now. After a google search it seems to be that DTS doesn't have a Dialnorm like feature.

[madshi]

Err... I updated eac3to and I'm getting the same error messages.
Hmmm... Does GraphEdit.exe (renamed to Recode.exe) give out more than 0 bytes when doing "Nero Audio Decoder 2 -> Dump"?

[ACrowley]

Argh. Ok, thanks for testing!

Are you absolutely sure that DTS really knows Dialnorm at all? I rather doubt it right now. After a google search it seems to be that DTS doesn't have a Dialnorm like feature.


Madshi , im not sure
Look at here, its a Comparion from original dts Source decoded with all Decoders :
http://img123.imageshack.us/img123/42/46819065qo5.th.png (http://img123.imageshack.us/my.php?image=46819065qo5.png)
AC3 Filter has higher Levels..

This is a Comparions from AC3Filter Source with a Surcode reencode with -10db DialogNorm RegistryValue
http://img66.imageshack.us/img66/7116/84926638sa7.th.png (http://img66.imageshack.us/my.php?image=84926638sa7.png)

No difference.
When it would be possible to change DialogNorm via Registry for Surcode ,there must be a visible Difference

Read here (from Hardware Sound Processor)
http://www.hometheaterhifi.com/volume_4_3/meridiandts.html

DTS also does not have flags that support late evening listening (compression) or dialog normalization, while DD does.

Then it makes sense why Surcode output is identical with DRC/DialogN Reg.Values On or Off, and AC3 Filter is only a bit louder

Also i was wondering why honai told us his AC3 Tracks are louder as dts..it should be reversed.

[honai]

Also i was wondering why honai told us his AC3 Tracks are louder as dts

What I meant was that typically my receiver plays DD louder than DTS, but with the new eac3to the transcoded DTS tracks were louder than the same DD tracks from retail DVDs.

[ACrowley]

What I meant was that typically my receiver plays DD louder than DTS, but with the new eac3to the transcoded DTS tracks were louder than the same DD tracks from retail DVDs.

yes..i know
And i was only wondering why typically dts is always louder as ac3 :)
Ofcourse the decoding from eac3to 1.10 is "louder" now , because it preserves the full dynamics instead of older Version with full DRC/Dialog.
So your dts reencode has more dynamics/is louder too. I think a AC3 reencode from EAC3 is closer to the Source. Surcode isnt the best Encoder ....

[d0ORk]

It's working again.
I had to unpack the files all again and made sure that the old ones are overwriiten. Stupid error of me.

Err... I updated eac3to and I'm getting the same error messages.

[tebasuna51]

Argh. Ok, thanks for testing!

Are you absolutely sure that DTS really knows Dialnorm at all? I rather doubt it right now. After a google search it seems to be that DTS doesn't have a Dialnorm like feature.
Of course DTS have DialNorm and DRC like you can see in DTS Coherent Acoustics; Core and Extensions (http://webapp.etsi.org/action/PU/20021224/ts_102114v010201p.pdf) (pag. 14 and 17)

"Embedded Dynamic Range Flag V DYNF 1 bit
DYNF indicates if embedded dynamic range coefficients are included at the start of each sub-frame. Dynamic range correction may be implemented on all channels using these coefficients for the duration of the sub-frame."

"Dialog Normalisation Parameter/Unspecified V DIALNORM/UNSPEC 4 bits
...
The dialog normalization gain (DNG), in dB, is specified by the encoder operator and is used to directly scale the decoder outputs samples. In the DTS stream the information about the DNG value is transmitted by means of combined data in the VERNUM and DIALNORM fields (see table 5.22)."

Other question is if a encoder use these features.

[madshi]

Hmmmm... Thanks for the PDF, it's helpful, although a bit confusing. I'll do some more tests later when I have more time.

[ACrowley]

I Made more Tests.
The Surcode DTS Pro 1.0.23 Registry Settings for DialogNorm and DRC have no Effect

[madshi]

Ok, the next test would be to check which dialnorm value Surcode writes into its DTS encodes and to manipulate this value and see if it shows any effect with any decoders.

[d0ORk]

Is it possible to see a delay somewhere?
When I demux the Departed Trailer, Evodemux shows no delay. But when I encode and mux it to .mkv with MeGui the video/audio is async.

[madshi]

Is it possible to see a delay somewhere?
When I demux the Departed Trailer, Evodemux shows no delay. But when I encode and mux it to .mkv with MeGui the video/audio is async.
EvoDemux shows the timestamps (PTS) of the first video and audio packets. If they don't match, you can calculate the delay from that.

[d0ORk]

I still got a strange thing.
The source evo is 1min 8seconds. After evodemux the vc1 (I use that ending for the demuxed video) is 3min 10sec...

[Rectal Prolapse]

madshi, is it possible to have eac3to output a new DD+ track with dialnorm (and maybe DRC?) removed? It would be a great option for realtime playback of DD+ tracks without the step of conversion.

[madshi]

madshi, is it possible to have eac3to output a new DD+ track with dialnorm (and maybe DRC?) removed? It would be a great option for realtime playback of DD+ tracks without the step of conversion.
Removing the DRC information from the DD+ track would be theoretically possible, I think, but much harder than removing DialNorm. Removing DRC would mean parsing through the full DD+ structure, which is relatively complicated. And if there's a parsing bug, the DRC removing would destroy the DD+ track. Because of that I'd rather not do that. Removing Dialnorm is oh so much easier, because it's right at the beginning of each audio frame, so not much parsing is needed and there's no big danger for bugs. I could easily add an option to give out a modified DD+ track without DialNorm. Actually you can do that yourself already: Just ask eac3to to convert the DD+ track to e.g. "raw". Then during the decoding phase you can copy the "something.tmp.ddp" file. This temporary file is a full copy of the original DD+ track, just with DialNorm turned off.

But I have to say that defeating DialNorm alone doesn't make much sense, as long as DRC is still active, because DRC is the much worse problem.

[Rectal Prolapse]

Ah, thanks madshi! :)

For realtime playback, is Nero Audio Decoder the best solution right now?

[Daodan]

Sorry to interrupt this lenghty DRC/normalize talk, but does anyone know a method for decoding truhd from mt2s files? No splitter I have sees that (sonic, nero, cyberlink, etc). Only xport can demux but that doesn't help much as I don't think there is a filter source for that.

[madshi]

For realtime playback, is Nero Audio Decoder the best solution right now?
I think for realtime playback I'd use the Sonic one right now. It seems to be a bit less bad than the Nero one in terms of DRC. But maybe you should better use ShowTime or PowerDVD. That way you could get around DRC.

does anyone know a method for decoding truhd from mt2s files?
Haven't tried that yet cause I haven't run over a TrueHD Blu-Ray yet. But I know they are beginning to appear.

[Daodan]

Here's a piece: http://d01.megashares.com/?d01=2209572
Track 1 is 5.1 truhd. I found smth intersting with that, will post after a few more tests (I managed to get some sound out of it, but has some crackling in it).
Also Sonic demux sees the demuxed truhd with xport...only problem is that nero audio can't connect to that...

[Rectal Prolapse]

I read that Blu-ray TrueHD tracks have a DD5.1 640 kbps track interleaved with the TrueHD stream. It is NOT a core track, but a separate track.

madshi: Unfortunately, I have major PQ issues with PowerDVD, and showtime crashes on a lot of blu-ray titles. Oh well. I guess I'll stick with Sonic for now, or actually use eac3to for what you designed it for and mux my own. :)

[Daodan]

Yeah, that's probably why I manegd to get some sound. Having it interleaved sounds like a lol idea, why would they do that? Just because the crappy players don't support truhd?

[ACrowley]

Yeah, that's probably why I manegd to get some sound. Having it interleaved sounds like a lol idea, why would they do that? Just because the crappy players don't support truhd?

yeh...can be true

All Decoders decodes the 640kbps 2.0 AC3 so far i can see. Nero cant connect to it

[MichalHabart]

Hello Madshi,
i still have the same problem converting pcm file to dts. But i finally find the problem, look:

D:\VOBs\!EAC3toAC3>eac3to.exe "D:\VOBs\Air Force One\bits0001.pcm" E:\English.dts -1536 -16
RAW, 5.1 channels, 2:04:34, 16 bits, 48khz
Convert the multi channel raw file to mono wav files. Please wait...
Convert the mono wav files to dts. Please wait...
Surcode sais/asks: "Invalid Wave File E:\English.L.wav.".
Surcode sais/asks: "Invalid Wave File E:\English.R.wav.".
Surcode sais/asks: "Invalid Wave File E:\English.SL.wav.".
Surcode sais/asks: "Invalid Wave File E:\English.SR.wav.".
Surcode sais/asks: "Invalid Wave File E:\English.C.wav.".
Surcode sais/asks: "Invalid Wave File E:\English.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...
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...
Creating the DTS file failed. The multi channel wav files are valid, though.

It seems to me that he creates mono wav files in different location (in fact in same where source pcm file is) and then he is trying to find them somewhere else (where destination file should be). Can you please change it in eac3to?

[madshi]

It seems to me that he creates mono wav files in different location (in fact in same where source pcm file is) and then he is trying to find them somewhere else (where destination file should be). Can you please change it in eac3to?
Yey, I'm glad we found the cause of the problem! :)

[madshi]

eac3to 1.11 released

http://madshi.net/eac3to.zip

(1) bugfix: (L)PCM -> DTS encoding automation failed when source and destination folders differed
(2) added: new "-allowDnr" switch allows Nero's audio decoder to apply DNR
(3) added: new "-keepDialnorm" switch disables removal of E-AC3 dialnorm information

[MichalHabart]

Yey, I'm glad we found the cause of the problem! :)

Me too and thanks for fast fix. Will try tomorrow :)

Edit: Now everything is working properly. Once again, thank you madshi for excellent work.

[ACrowley]

eac3to 1.11 released

http://madshi.net/eac3to.zip

(1) bugfix: (L)PCM -> DTS encoding automation failed when source and destination folders differed
(2) added: new "-allowDnr" switch allows Nero's audio decoder to apply DNR
(3) added: new "-keepDialnorm" switch disables removal of E-AC3 dialnorm information


(1) is DynamicRangeCompression ,right ?
Maybe short : DRC is more obvious ?

@Keymaker
No more Gui Versions ?

[madshi]

Ouch, why did I write "DNR"? I guess I was sleeping. Of course I meant DRC!! Will fix that in the next build.

[madshi]

eac3to 1.12 released

http://madshi.net/eac3to.zip

(1) The tools "flac.exe", "aften.exe" and "sox.exe" are now distributed in the eac3to zip to make your life a bit easier.
(2) Correct channel mapping for 7.1 LPCM tracks is default now.
(3) The new option "-down6" allows downconverting of 7.1 tracks to 5.1 (this option is activated by default for AC3 and DTS encoding).
(4) I'm shipping a modded "flac.exe" with eac3to now, which has no problems with 2GB file output, anymore!! As a result the flac encoding by eac3to should now be more or less perfect.
(5) Fixed a number of bugs.

Btw, those Blu-Ray soundtracks which say 6.1 LPCM are actually 7.1. Ok, two channels are identical, that's probably why the cover sais 6.1. But the LPCM track contains full audio data for 8 channels. So use the "-8" switch for 6.1 LPCM tracks.

[SvT]

I can read 5 more reasons to download this program !!!!!!!!!!!!!!

Thank you very much !!!!!!

[ACrowley]

@madshi

So is 1.12 a "full Vesrion" of eac3to ?
Or do we need all the older Files like the .pas Files etc ?

Nice work on flac.exe..i will test it

greets

[madshi]

So is 1.12 a "full Vesrion" of eac3to ?
Or do we need all the older Files like the .pas Files etc ?
It's a "full version". The pas etc files are the source code of eac3to, which is now in the sub folder "sources". You can delete this "sources" folder if you don't need/want to have the source code of eac3to. You should keep the other subfolders and files. Ok, you can delete sox, if you don't plan to downconvert anything from 24bit to 16bit. And you can delete the aften/flac sub folders, if you don't ever plan to use eac3to to encode ac3/flac.

[ACrowley]

ok, thx

On more suggestion, maybe you should note in eac3to.exe that Sonic Decoder will apply DRC and DialogNormalization, that you can only avoid it with NeroDecoders.
Sure, we here know it, but for other Users its more obvious

And, for bluray lpmc to wave , do i need still the -bluray parameter ?
Or works the Auto CHmapping for lpcm too ,as like as for eac3/TrueHD(when i want standard L=0 R=1 C=2 LFE=3 SL=4 SR=5 ) ?

EDIT:
Nice work!! lpcm24bit to flac 24bit works perfect! With Timeline and no 2gb Restriction!
Older eac3to Version outputs always a 24bit flac which was corrput8 to big, only noise) but now it works !

Ich wüsste nicht wo die encoder ohne deine Arbeit wären im HDDVD/Bluray Audio Sektor...weiter so!

[madshi]

On more suggestion, maybe you should note in eac3to.exe that Sonic Decoder will apply DRC and DialogNormalization, that you can only avoid it with NeroDecoders.
Sure, we here know it, but for other Users its more obvious
E-AC3 DialNorm is fully defeated by eac3to, doesn't matter which decoder you're using. But I should probably add a warning about DRC with Sonic's decoder, I agree.

And, for bluray lpmc to wave , do i need still the -bluray parameter ?
Or works the Auto CHmapping for lpcm too ,as like as for eac3/TrueHD(when i want standard L=0 R=1 C=2 LFE=3 SL=4 SR=5 ) ?
Channels should automatically be mapped correctly for every kind of input, unless you have a strange input. For Blu-Ray PCM I expect the data to be demuxed by xport or TsRemux, though. If you use the GraphEdit dump method to demux a Blu-Ray PCM track, you'll have to set different parameters in eac3to, e.g. endian will be "little" instead of big etc...

[ACrowley]

E-AC3 DialNorm is fully defeated by eac3to, doesn't matter which decoder you're using. But I should probably add a warning about DRC with Sonic's decoder, I agree.

Ah, Ok....sure ea3to "reverse" DialogN itselft...
Sonic Decoder outsinde eac3to applies starnge Mix of DRC and or DialogNorm+gain


Channels should automatically be mapped correctly for every kind of input, unless you have a strange input. For Blu-Ray PCM I expect the data to be demuxed by xport or TsRemux, though. If you use the GraphEdit dump method to demux a Blu-Ray PCM track, you'll have to set different parameters in eac3to, e.g. endian will be "little" instead of big etc...

Yes..works perfect

But thats not my Experience....All LPCM Tracks i have demuxed with SonicHDDemxuer/graphedit are working with BigEndian.
Shouldt eac3to detects autom wich Endian it is ?
When i set no Endian Switch the Output is ok

And your tweaked flac.exe...is it working only eac3to intern or can i use the 2gb workarounds outside eac3to ?

[madshi]

Sonic Decoder outsinde eac3to applies starnge Mix of DRC and or DialogNorm+gain
I'm quite sure it's no DialNorm with the later eac3to versions. But probably DRC + gain.

But thats not my Experience....All LPCM Tracks i have demuxed with SonicHDDemxuer/graphedit are working with BigEndian.
Strange. Well, that's just fine.

Shouldt eac3to detects autom wich Endian it is ?
I can try to add auto Endian detection.

And your tweaked flac.exe...is it working only eac3to intern or can i use the 2gb workarounds outside eac3to ?
Of course that tweaked flac.exe works just fine on its own, but I will not tell you that, because I want eac3to to gain world domination... :)

P.S: That said, I've fixed the *output* 2GB limitation. The *input* limitations are still valid. You can work around them by feeding flac.exe though stdin (that's what eac3to is doing), but it's not very comfortable to do that manually.

[ACrowley]

Ok thx
Yes ,youre Right. Sonic Output looks more like DRC+Gain, very bad

Ah, Ok...It works everytime for me because Big Endian is default and all my LCPM are Big Endian as i say even demxued with Graphedit
Autodetection would be nice.
I know that SonicHDDemxuer output should be little Endian..But its reversed, with little Endian i get garbage on 24bit lpcm
Maybe the older 4.2 Demxuer outputs little E.

And i was talking about flac "output limitation"

[yonta]

-down6 downconvert raw data from 8 to 6 channels
-down16 downconvert raw data to 16 bit
-96000 TrueHD output or PCM file is 96 kHz
-8 TrueHD output or PCM file is 8 channels
-16 TrueHD output or PCM file is 16 bit
-24 TrueHD output or PCM file is 24 bit (default)
-little PCM file is little endian
-big PCM file is big endian (default)


I have a 24bit-5.1ch-48kHz-big.endian lpcm file and I tried converting to a flac.
The info about the lpcm file is from the output of xport

LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 24
LPCM Audio Sample Rate = 48000

So, I figured I don't have to use any option above and used this simple command line.

eac3to "the.wild.LPCM24.pcm" "The.Wild.flac"

This is what eac3to says;
Tried to find out the bitdepth automatically, but the algorithm failed.
Please use either "-16" or "-24" to tell eac3to which bitdepth the source has.

probably auto-detection is default instead of "-16" or "-24"?

[madshi]

probably auto-detection is default instead of "-16" or "-24"?
True, auto detection is default. You'll have to specify "-24".

[pyrates]

Is it possible to convert truehd 8 channel from either inside an evo or demuxed to a pcm file and then I can use each3to to convert that to flac 7.1 audio file? I did try renaming the demuxed mlp file to .pcm but it only picks up the first 21-22 minutes of the movie instead of the full 2 hours worth which all the dolby digital plus tracks do. If I can somehow convert that truehd 8 channel track to pcm 7.1, then I can use each3to to convert it to a flac 7.1 audio file.

[madshi]

The Nero TrueHD decoder can only do 5.1. And the Sonic TrueHD decoder is totally unusable. So currently I don't see how you can decode all 8 channels of a 7.1 TrueHD track.

[pyrates]

I checked with Powerdvd Ultra and it can only decode Dolby TrueHD 5.1. So that can't be used either.

[madshi]

I checked with Powerdvd Ultra and it can only decode Dolby TrueHD 5.1. So that can't be used either.
Thanks for testing. Wasn't sure about that myself. However, PowerDVD only outputs 16bits, so it wouldn't have been the optimal solution, anyway.

[ACrowley]

I checked with Powerdvd Ultra and it can only decode Dolby TrueHD 5.1. So that can't be used either.


And PowerDVD can play eac3 ofcourse , but only in evo, as like as TrueHD only in evo

Nero/WindDVD8 can open/play single eac3 tracks, but truehd again only in evo, with nero.
With winddvd8 no Truehd possible (only fileinfo)

[The_Keymaker]

For Acrowley and others using EAC3toGUI, I will try and implement changes and release an updated GUI shortly.

I've been working on a WiFi remote for my HTPC and have not followed eac3to much lately.

It would help if Madshi could PM me OR briefly post a list of changes since the last revision of EAC3toGUI.

Regards,
The_Keymaker

[pyrates]

When using Nero audio decoders 7.10.1.0, I get this output:

eac3to.exe L:\Encoding\PEVOB_1_MERGED.DD+.stream.00.ddp L:\Encoding\movie.ac3 -640 -keepDialnorm -24
E-AC3, 5.1 channels, 1:48:32, 640kbit/s, 48khz, dialnorm: -27dB
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
Exception EIntOverflow in Modul eac3to.exe bei 0001F6E6.
Integerüberlauf.

And eac3to just crashes with this output when I use sonic audio decoder 4.2.0:

eac3to.exe L:\Encoding\PEVOB_1_MERGED.DD+.stream.00.ddp L:\Encoding\movie.ac3 -640 -sonic -keepDialnorm -24
E-AC3, 5.1 channels, 1:48:32, 640kbit/s, 48khz, dialnorm: -27dB
Muxing eac3 file to raw. Please wait...

Faulting application eac3to.exe, version 1.12.0.0, time stamp 0x2a425e19, faulting module CinemasterAudio.dll, version 4.3.0.169, time stamp 0x466494c3, exception code 0xc0000005, fault offset 0x000efb73, process id 0x450, application start time 0x01c7da2b199021d5.

[hanti]

D:\dvdrip\eac3to\eac3to.exe D:\dvdrip\work\300\eac3.eac3 D:\dvdrip\work\300\output.ac3 -640
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
pls help ;)

[pyrates]

.
pls help ;)

Use sonic audio decoder 4.2.0 instead and add -sonic to the command arguements when encoding, then it should work.

[ACrowley]

Use sonic audio decoder 4.2.0 instead and add -sonic to the command arguements when encoding, then it should work.

not recommned. With SonicDecoder you get full DRC+gain.

[pyrates]

not recommned. With SonicDecoder you get full DRC+gain.

Well that's how I got it to work. By default it uses the nero audio decoder.

[ACrowley]

Well that's how I got it to work. By default it uses the nero audio decoder.

simply the best is to use only NeroDecoder. No RDc and no DialogN= good :)

[hanti]

so i want to use nero but still getting this error :/

The file size of the raw file doesn't seem to fit.
i have even nero hd/bluray plugin ;)

[madshi]

Please wait for the next eac3to version. I think I will have the EIntOverflow problem solved there. However, I doubt that it will fix the problem with the unexpected raw file size. It seems there's something wrong with your Nero installation, I'd guess...

[bullet_ballet]

I must have the same problem with my install because I get the same error consistently. I don't have the HD plug-in, but I understand that's only necessary for TrueHD.

I might just have to call Ahead and try and get my money back.

[ACrowley]

I must have the same problem with my install because I get the same error consistently. I don't have the HD plug-in, but I understand that's only necessary for TrueHD.

I might just have to call Ahead and try and get my money back.


I would say Nero is necessary for EAC3 to, cause ist the only Way to keep the full Dynamcis

@hanti
Try to rebuild the Evo with only your desired Audio and run eac3to onto it . I think its only your Audio File

[calinb]

Thanks for the great tool, madshi! Would it be possible to add a -dtswav option? I get mkvmerge errors when I attempt to mux a .dts file but a dts .wav file works okay.

dts2wav.exe doesn't appear to handle 48k / 24-bit .dts files so adding it to my batch file won't help.

-Cal

Edit:
or perhaps the syntax

eac3to.exe <inputfile.xxx> <outputfile.wav> -1536

and

eac3to.exe <inputfile.xxx> <outputfile.wav> -768

could be supported to produce a dtswav output file. Curently, it results in an error.

"Bitrate "1536" is only allowed for DTS encoding."

[squal]

Don't know if this has been resolved somewhere in the thread but I had this problem and resolved it so I'm posting how I fixed it.

When trying to decode anything to wav (ddp or mlp) I got this:

E-AC3, 5.1 channels, 2:12:32, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.

Finally I discovered the actual problem was the dtsac3source-full.ax file which was corrupted or something. I had the same exact version on another computer which I had downloaded a couple of weeks before. That one was working, I knew, so I unregistered the old one (don't know if this is necessary)...

regsvr32 /u C:\WINDOWS\system32\dtsac3source-full.ax

...got the new one, and it worked.

If your raw file is 0 bytes and then eac3to crashes, chances are this is your problem.

Get some new dtsac3source from some different mirror or download site, register it and try again. That worked for me.

The version I have is 1.0.0.1, BTW.

Hope this helps.

[madshi]

Would it be possible to add a -dtswav option? I get mkvmerge errors when I attempt to mux a .dts file but a dts .wav file works okay.
Ehm, now I'm confused. What is a "dtswav" file? Is it a wav file or a dts file or what? :confused:

[madshi]

Finally I discovered the actual problem was the dtsac3source-full.ax file which was corrupted or something.
That's interesting - thanks! I'm wondering why it happens in the first place, though, because eac3to by default doesn't even use the dtsac3source filter! The filter is only used if you use the "-orbitlee" switch or if the "Nero File Splitter" fails to work. Usually the Nero File Splitter should work just fine.

[bullet_ballet]

Thanks for the tip, squal. Unfortunately, I still can't get eac3to to work. I tried -orbitlee switch and reinstalling, but nada. Here's hoping something good comes along.

Oh, and thanks for the tool, madshi. Hopefully I get to try it sometime.

[tebasuna51]

Ehm, now I'm confused. What is a "dtswav" file? Is it a wav file or a dts file or what? :confused:
Really WAV is a container and can contain PCM-uncompressed audio but also compressed audio mp3, dts, ...

Then a dtswav is a .wav file with a wav header continued by the dts stream.

In SurCode DTS v1.0.23 when you select Destination... -> Type you can choose betwen:

DTS Compact (.cpt)
DTS Padded (.dts)
DTS WAV (.wav)

[calinb]

Thanks for answering madshi's question, tebasuna51. Yes--Surcode allows the GUI user to specify the target file as a .wav file but the .wav contains DTS code rather than the more common PCM or, as you point out, other code.

The only other time I've used dtswav was when I played around with authoring a DTS audio CD. Using dtswav, it was possible to use conventional audio CD burning software--just burn the DTS .wav files like PCM .wav audio files. Now I'm interested in the dtswav format to cleanly mux DTS without errors or audio stuttering using mkvmerge.

[The_Keymaker]

New Version (v1.25) of EAC3toGUI. Download at:

http://www.sendspace.com/file/yagije

Change log for version 1.25:
- Added -allowDrc option
- Added -keepDialnorm option

As usual, place EAC3toGUI in the same directory as eac4to.exe and run. Alternatively, you can place it anywhere as long as you set the location of eac3to.exe using the "Settings" menu option. Please report any errors or suggestions.

@madshi:
if you like, you may want to consider establishing a permanent link for EAC3toGUI and putting it in the first post or distributing it with eac3to.

Regards,
The_Keymaker

[Darth Pinous]

Does anyone know a method for decoding truhd from mt2s files? No splitter I have sees that (sonic, nero, cyberlink, etc). Only xport can demux but that doesn't help much as I don't think there is a filter source for that.

I tried some things with a blu-ray trueHD track, but no luck so far.
The file demuxed by xport can't be imported in Scenarist (Scenarist sees it as an AC3 track and refused it because the bitrate is above 448Kb).

However, the file demuxed by xport is seen as a dolby trueHD track by Sonic HD Demuxer in graphedit. So I tried a dump with the graph FileSource(async) -> Sonic HD Demuxer-> Dump. The resulting file is a couple of Kb smaller, and still can't be imported in Scenarist, but the error is different : "Maximum frame size of MLP Audio file exceeds HD DVD limit.", so Scenarist sees it as a TrueHD track, but the file header is certainly wrong...

[yonta]

filename recognition problem:

command line and eac3to output is;
eac3to.exe f:\innocence.LPCM16.pcm f:\innocence.ac3 -640 -16
RAW, 5.1 channels, 0:01:20, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN - r552
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

error opening input file: f:\innocence.wav
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.

Aften is looking for "innocence.wav" but this file doesn't exist.
the wav file eac3to created is in fact "innocence.LPCM16.wav".

[dade49]

filename recognition problem:

I see this problem as well. There is an error in eac3to that is sending aften.exe the filename of the output.wav. eac3to is creating the filename as source.wav, so the file doesn't exist.

eac3to.exe LPCM.pcm audio_640k.ac3 -640 -16
RAW, 5.1 channels, 1:37:08, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

error opening input file: audio_640k.wav
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.

The wav file created is named LPCM.wav in this case. Aften is being fed the wrong filename from eac3to.

As always Madshi, thanks for all your work. I hope to see ver 1.13 soon with this bugfix.

-Dade

[big-guy]

I am ripping a Blu-Ray movie and am encountering a problem.
There are 3 tracks included: Japanese TrueHD 5.1 (DD5.1 Core), LPCM 2.0 and English DD+ 2.0.
As you can see only the first track is usefull...
Is there any way I can transcode a Dolby TrueHD track from a Blu-Ray disc? I've read that with Blu Ray streams the TrueHD stream has a normal AC3 stream at probably 640kbps weaved in (Like with DTS-HD). Does anyone here know, how I can do that with a TrueHD track?

Thanks.

[madshi]

Really WAV is a container and can contain PCM-uncompressed audio but also compressed audio mp3, dts, ...

Then a dtswav is a .wav file with a wav header continued by the dts stream.

In SurCode DTS v1.0.23 when you select Destination... -> Type you can choose betwen:

DTS Compact (.cpt)
DTS Padded (.dts)
DTS WAV (.wav)
Ah, thanks!

Would it be possible to add a -dtswav option? I get mkvmerge errors when I attempt to mux a .dts file but a dts .wav file works okay.
I'd prefer if Mosu could fix mkvmerge. However, I've just tried muxing a DTS file encoded by eac3to/Surcode with mkvmerge and it works perfectly fine for me. If all else fails, I can of course add dtswav support.

[madshi]

@madshi:
if you like, you may want to consider establishing a permanent link for EAC3toGUI and putting it in the first post
Done.

[madshi]

filename recognition problem

I see this problem as well. There is an error in eac3to that is sending aften.exe the filename of the output.wav. eac3to is creating the filename as source.wav, so the file doesn't exist.

Fixed in latest version (see below).

[madshi]

Is there any way I can transcode a Dolby TrueHD track from a Blu-Ray disc? I've read that with Blu Ray streams the TrueHD stream has a normal AC3 stream at probably 640kbps weaved in (Like with DTS-HD). Does anyone here know, how I can do that with a TrueHD track?
I've just added this functionality to the latest eac3to version.

[madshi]

eac3to v1.13 released

http://madshi.net/eac3to.zip

(1) "eac3to src.ac3 dst.ac3" removes Dialog Normalization from AC3 files.
(2) "eac3to src.eac3 dst.eac3" removes Dialog Normalization from E-AC3 files.
(3) "eac3to src.thd dst.ac3" extracts the AC3 frames from a Blu-Ray TrueHD track and removes Dialog Normalization.
(4) some bugfixes

[ACrowley]

eac3to v1.13 released

http://madshi.net/eac3to.zip

(1) "eac3to src.ac3 dst.ac3" removes Dialog Normalization from AC3 files.
(2) "eac3to src.eac3 dst.eac3" removes Dialog Normalization from E-AC3 files.
(3) "eac3to src.thd dst.ac3" extracts the AC3 frames from a Blu-Ray TrueHD track and removes Dialog Normalization.
(4) some bugfixes

nice :)

Would be great when there would be a solution to remove DialogN on TrueHD..i know youve not enough Infos for TrueHD at the Moment

Is it now possible to decode AC3 with eac3to ?
Sure, its not so important because there are enough stable Methods to decode AC3 without DRC/DialogN

[honai]

@madshi

I just converted a TrueHD track from an .evo to mono WAVs. eac3to did not print any information about which decoder was used, and if DRC was removed. Is this behavior intentional? Am I to assume that Nero was used, and that all went fine?

[madshi]

Would be great when there would be a solution to remove DialogN on TrueHD..
Agreed!

i know youve not enough Infos for TrueHD at the Moment
That's true(HD).

Is it now possible to decode AC3 with eac3to ?
No.

Sure, its not so important because there are enough stable Methods to decode AC3 without DRC/DialogN
Right. And since eac3to can now remove dialnorm information from AC3 tracks, too, there should be even more stable methods. E.g. ffdshow should now work, too, I think.

[madshi]

I just converted a TrueHD track from an .evo to mono WAVs. eac3to did not print any information about which decoder was used, and if DRC was removed. Is this behavior intentional? Am I to assume that Nero was used, and that all went fine?
If you used eac3to v1.12 or v1.13, Nero is the default TrueHD decoder. DRC is not applied by Nero's current TrueHD decoder at all (as far as I can say). So there's nothing eac3to had to remove. You can assume that Nero was used and that all went fine.

[Nikos]

Usually i convert the .eac3 audio to 6 mono .wav and then i convert them to .ac3 with aften or vegas. I know that the Dialog Normalization value in a .ac3 file is essential.

My question:
Is there any method to find out the Dialog Normalization value of a .eac3 file and then put the same value in the final .ac3 file?

[dade49]

Removes Dialog Normalization from AC3 tracks.

Madshi,
This worked awesome on most of my AC3 tracks. I do have 2 examples where eac3to detected DialNorm, but it dumps out. EX:

C:\>eac3to.exe old.ac3 new.ac3
AC3, 5.1 channels, 1:48:07, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...

C:\>

No new file is created. Is there a way to run eac3to in debug mode? If I send you a sample file, is there a certain size you want, and/or is there a specific frame size I should cut to? Do you want the beginning or the middle somewhere?

-Dade

[calinb]

I'd prefer if Mosu could fix mkvmerge. However, I've just tried muxing a DTS file encoded by eac3to/Surcode with mkvmerge and it works perfectly fine for me. If all else fails, I can of course add dtswav support.

Hmm--With .dts, I get a screenful of these warnings and stuttering playback. No problem with dtswav. I can certainly see what Mosu thinks about it.

Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged.
Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged.

[madshi]

Usually i convert the .eac3 audio to 6 mono .wav and then i convert them to .ac3 with aften or vegas. I know that the Dialog Normalization value in a .ac3 file is essential.

My question:
Is there any method to find out the Dialog Normalization value of a .eac3 file and then put the same value in the final .ac3 file?
Sure. When using eac3to to convert the eac3 audio to 6 mono wavs, eac3to will output the Dialog Normalization value of the eac3to track (see end of first line of output). However, are you sure that you want to put it back in to the AC3 file? Personally, I don't like dialnorm.

[madshi]

This worked awesome on most of my AC3 tracks. I do have 2 examples where eac3to detected DialNorm, but it dumps out. [...] No new file is created. Is there a way to run eac3to in debug mode? If I send you a sample file, is there a certain size you want, and/or is there a specific frame size I should cut to? Do you want the beginning or the middle somewhere?
If you want to do frame correct cutting, you can drop the track into delaycut. It will tell you how many bytes per frame that AC3 track is using. I'm not sure what I need. I'd suggest that you try copying the first 2MB into a new file. Then simply check whether eac3to fails on that. If yes, those 2MB will be plenty enough. Heck, even 500KB would be enough, as long as the little chunk makes eac3to fail.

[madshi]

Hmm--With .dts, I get a screenful of these warnings and stuttering playback. No problem with dtswav. I can certainly see what Mosu thinks about it.

Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged.
Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged.
Strange, I'm not getting this problem on my PC. Are you encoding with 768 or 1536? I'm always using 1536. Maybe that's the difference? Anyway, I think this is a job for Mosu to fix - unless the DTS file is really buggy.

[ACrowley]

Usually i convert the .eac3 audio to 6 mono .wav and then i convert them to .ac3 with aften or vegas. I know that the Dialog Normalization value in a .ac3 file is essential.

My question:
Is there any method to find out the Dialog Normalization value of a .eac3 file and then put the same value in the final .ac3 file?

"Dont" set it to the same Value for the reencoded AC3

Heres a little Discussiuon on this Thread a few Sites before
Conclusion about AC3 encoding :

Youve 2 Options

1. Encode with -31db DialogNorm. DRC Profiles "off"
http://forum.doom9.org/showthread.php?t=118759&highlight=sony+vegas
Keeps the VolumeLevels from Wavesource. You get the Full Dynamcis back", but not fully Dolby Compliant, nevermind

2. Measure the RMS Value from the Center Channel and set it for DialogNormalization "with" a DRC preset
http://forum.doom9.org/showthread.php?t=56020

I suggest Method 1

[Zelos]

MAdshi great job !

Something strange , i tried to extract english channel from bourne identity movie.
I get the wav which is about 1h58 if i believe eac3to.
When i tried to play it with a player i get a 35mins sound channel , even when i try to convert it to ac3 !

For people who had problemns with The file size of the raw file doesn't seem to fit.
It's a registration pb , buy a real serial , i just bought one on Ebay ( i precise a REAL one from an original box ) , and cost me 15€ ;)

[madshi]

Something strange , i tried to extract english channel from bourne identity movie.
I get the wav which is about 1h58 if i believe eac3to.
When i tried to play it with a player i get a 35mins sound channel, even when i try to convert it to ac3 !
The "wav" format can't really properly handle files > 2GB (or was it 4GB?). I guess that's the problem you're having. eac3to usually works around this problem. Some other software packages do that, too. But some do not. I recommend to directly recode the E-AC3 track to either AC3, DTS or FLAC with eac3to. This way you should get a proper result. If you want to use a different AC3 encoder, you'll have to find out how to make that AC3 encoder properly handle big "wav" files. Or maybe you can use 6 mono "wav" files instead of one big 5.1 file. That should help, too.

[tebasuna51]

The "wav" format can't really properly handle files > 2GB (or was it 4GB?).

The wav format can handle until 4 GB.

The limit is two fields with 4 bytes (to store filelength and datalength) only can support values until 2^32.

The 2 GB limit is a wrong soft than read/write these 4 bytes with signed int then the limit is 2^31 -> 2 GB, if the fields are read/write with unsigned int the soft can manage until 4 GB.

The workaround to support wav > 4 GB is ignore these two fields in wavheader and read until end of file. The only problem is if the wav file have extrachunks (some extra info) after the audio data, then are converted also to audio with final clicks.

[Zelos]

thanks for the help.
So the best way is to separate the channels then to mux them ?

[Nikos]

"Dont" set it to the same Value for the reencoded AC3

Heres a little Discussiuon on this Thread a few Sites before
Conclusion about AC3 encoding :

Youve 2 Options

1. Encode with -31db DialogNorm. DRC Profiles "off"
http://forum.doom9.org/showthread.ph...ght=sony+vegas
Keeps the VolumeLevels from Wavesource. You get the Full Dynamcis back", but not fully Dolby Compliant, nevermind

2. Measure the RMS Value from the Center Channel and set it for DialogNormalization "with" a DRC preset
http://forum.doom9.org/showthread.php?t=56020

I suggest Method 1

Thanks ACrowley for the reply. I had read the above threads. The method 1 is ok but in method 2 there is not an accurate way to measure the right Dialog Norm. value.

I wonter why you say "Dont set it to the same Value for the reencoded AC3", i suppose that the .eac3 file had the correct Dialog Norm value. Correct me if i am wrong.

[madshi]

I wonter why you say "Dont set it to the same Value for the reencoded AC3", i suppose that the .eac3 file had the correct Dialog Norm value. Correct me if i am wrong.
The problem is that many studios just leave the DialNorm value at their default setting. Because of that it is more or less useless IMHO.

[madshi]

So the best way is to separate the channels then to mux them ?
The best way to do what? You haven't really said what your final aim is.

[madshi]

The wav format can handle until 4 GB.

The limit is two fields with 4 bytes (to store filelength and datalength) only can support values until 2^32.

The 2 GB limit is a wrong soft than read/write these 4 bytes with signed int then the limit is 2^31 -> 2 GB, is the fields are read/write with unsigned int the soft can manage until 4 GB.
Yeah, that makes sense, thanks.

[tebasuna51]

thanks for the help.
So the best way is to separate the channels then to mux them ?

The best way depend of final format/encoder, if you use ac3/Aften or aac/NeroAacEnc you need use a wav multichannel > 4 GB and inform the encoder with -readtoeof 1 (for Aften) or -ignorelength (for NeroAacEnc)

Other format/encoders must need separated channels in monowav's.

[kidbeach]

looks like an awesome program, I'm going to check it out today...

a side note, graphedit doesn't come with dump.ax anymore I found it here:

http://www.apecity.com/tivo/dump.zip

[Zelos]

yes sorry.
The final aim is to obtain an AC3 file , the channel is in 5.1

[Nikos]

The problem is that many studios just leave the DialNorm value at their default setting.
Because of that it is more or less useless IMHO.

Thanks madshi for the explanation.

[Zelos]

The best way depend of final format/encoder, if you use ac3/Aften or aac/NeroAacEnc you need use a wav multichannel > 4 GB and inform the encoder with -readtoeof 1 (for Aften) or -ignorelength (for NeroAacEnc)

Other format/encoders must need separated channels in monowav's.

it works with the Aften command.
Thanks !
I checked on the website but this command doesn't exist.
Is there any way to choose a bitrate more than 448 kbps in Aften ?

[madshi]

it works with the Aften command.
Thanks !
I checked on the website but this command doesn't exist.
Is there any way to choose a bitrate more than 448 kbps in Aften ?
I don't understand. Why don't you simply use "eac3to source.eac3 dest.ac3"? This will reencode with Aften with 640kbps.

[Zelos]

I don't understand. Why don't you simply use "eac3to source.eac3 dest.ac3"? This will reencode with Aften with 640kbps.

Right madshi , i tried but i have an error at the end :Starting "aften" failed. A valid *.wav file was created successfully, though.

i tried with the last eac3to but still the same error.
I use the eac3 gui .

[madshi]

Right madshi , i tried but i have an error at the end :Starting "aften" failed. A valid *.wav file was created successfully, though.
I always need the full eac3to output.

Have you kept all subfolders as they are? Is there a subfolder named "aften" in your eac3to folder? And is there an "aften.exe" in that folder? Did you do anything to this file? E.g. replacing it with a different build or anything else?

[Zelos]

I always need the full eac3to output.

Have you kept all subfolders as they are? Is there a subfolder named "aften" in your eac3to folder? And is there an "aften.exe" in that folder? Did you do anything to this file? E.g. replacing it with a different build or anything else?
everything is inside and i have replace nothing

[Zelos]

ok , i erased all , and download again to decompress all the files in the right place.
That works fine !

Thanks for the help ;)

[dade49]

If you want to do frame correct cutting, you can drop the track into delaycut. It will tell you how many bytes per frame that AC3 track is using. [...]

Madshi,

I PM'd you 2 examples of AC3 files that crash eac3to when removing DialNorm. I still vote for a verbose switch so that more info is output on issues like this. Thanks for all the help, and the great utility.

-Dade

[ACrowley]

Thanks ACrowley for the reply. I had read the above threads. The method 1 is ok but in method 2 there is not an accurate way to measure the right Dialog Norm. value.

I wonter why you say "Dont set it to the same Value for the reencoded AC3", i suppose that the .eac3 file had the correct Dialog Norm value. Correct me if i am wrong.

Maybe you have mistaken something..
You, should not use the DialogNormValue from the AC3/EAC3 Source (if AC3 or EAC3), right.
Reencoding with -31db (dont care about Source Value) and "no" DynamicRange Profile will guarrante the Input Wave Volume Level with uncompresssed Dynamics.

But it has nothing to do with wrong EAC3 Input Values (by the Way,they are not wrong) :), its only a guideline for reencoding and nothing else :)

Ofcourse you can measure the RMS accurat enough. Just use the Center Wave with a Dialog Part . Use Adobe Audition and you will get a Value which is accurate enough to use it for DialogeNorm in your AC3 reencode .In this Case you have to use a DynamicRange Profile. Otherwise it makes no Sense.
As i say , its the full DolbyCompliant Method to encode AC3..

But i ,and most guys i know ( eac3to too) are using the -31db with no DRC Method. Its sounds best on my Receiver/Setup

@Madshi
I have a few eac3 Tracks ( FullMetalJacket HDDVD) with Problems when removing DialoNorm with eac3to. eac3 starts with removing DialNorm...and after a few seconds the whole Process stopps without Error Message. It works with DialNor enabled

[madshi]

In this Case you have to use a DynamicRange Profile. Otherwise it makes no Sense.
Why ?

[ACrowley]

Why ?
http://forum.doom9.org/showthread.php?t=56020

The following point, however, cannot be stressed enough: In order for the Dynamic Range Compression to work as designed, the Dialogue Normalization parameter MUST be properly set first!

I should better say "makes more Sense" .
When you want a Dolby Compliant AC3 ,use DRC .
But DRC needs a proper DialogNormlization Value = RMS calculated

When you use no DRC ,-31db is the Best Value to keep Wave Source Dynamic/Volume ( lange Rede lurzer Sinn, thats all what counts for me)
So why should i touch the -31db Value , "without" a DRC Profile. -31db makes "more Sense" because i dont change anything
There are these 2 Methods to reencode to AC3. Taker the one "or" the other Method...thats what i mean :)

PS
you know, my English is very bad an i cant say everything i want in english

[madshi]

The following point, however, cannot be stressed enough: In order for the Dynamic Range Compression to work as designed, the Dialogue Normalization parameter MUST be properly set first!
Yeah. So DRC needs DialNorm to be set correctly. But where is it written that DialNorm needs DRC to be set?

So why should i touch the -31db Value , "without" a DRC Profile. -31db makes "more Sense" because i dont change anything
Either you want DialNorm or you do not want it. If you want it, you tell the encoder to set it to the desired value. If you do not want it, you tell the encoder to set it to -31db. I don't see what DRC has to do with this.

There are these 2 Methods to reencode to AC3. Taker the one "or" the other Method...thats what i mean :)
Or take the third method which is to set a proper DialNorm value but not use any DRC. I don't see any reason why this shouldn't be used.

Well, personally I don't like neither DRC nor DialNorm. But I can understand if other people like DialNorm. So why forcing them to set DRC, if they only want DialNorm?

[madshi]

I PM'd you 2 examples of AC3 files that crash eac3to when removing DialNorm.
eac3to expects clean AC3 files. It aborts if there are errors in the AC3 file. The 2 examples you sent me were not clean. Please run them through besweet or delaycut to fix all the CRC errors etc. Afterwards eac3to will handle these files just fine.

[Nikos]

Thanks madshi and ACrowley for the explanations.

@madshi when i set a value of Dialog Norm. -27 db without DRC then the decoder simply attenuate the whole stream by [31 + (-27)] = 4 db.

I made a test with the audio from the "RIO Bravo" movie (.eac3 mono, -27 db Dial. Norm). I convert to mono .wav with eac3to.

eac3to without any parameter:
peek value --> -1.1 db, RMS value --> -28.2 db

eac3to with -keepDialnorm:
peek value --> -5.1 db, RMS value --> -31.4 db

eac3to with -keepDialnorm and -allowDRC:
peek value --> -9.4 db, RMS value --> -32.4 db

The above values taken from whole stream, not from a part. I use Sound Forge with "ignore sound below -45 db" and "use equal loudness contour".


@ACrowley "Just use the Center Wave with a Dialog Part" it's not enough. We must use several parts from Center Wave with Dialog only and calculate the average RMS because there are high volume Dialog parts and low volume Dialog parts.

[madshi]

@madshi when i set a value of Dialog Norm. -27 db without DRC then the decoder simply attenuate the whole stream by [31 + (-27)] = 4 db.
Correct. That's how DialNorm works... :)

[tebasuna51]

Just my opinion:

- DialNorm: is a tool to equalize volume from different sources. If you only listen ac3 sources is a valid option to maintain the volume control at same position. If you listen also modern CDA music, commercial TV, mp3, ... use DialNorm -31 dB (and maximize peak to 0 dB).

- DRC: useful in bad listen environment (neighbors, night, bad audio equipment, ...). Recommended, if your audio equipment is bad you need them, if is good you can disable/enable at your choice. Remember, the audio is equal encoded, only attenuation/gain values are added and the decoder can use or not.

[yonta]

eac3to generates a wrong wav file

source is 16bit 48khz 7.1ch Blu-ray lpcm and tried converting to ac3.

command line and output is:
F:\eac3to>eac3to f:\Innocence.pcm f:\Innocence.ac3 -8 -16 -down6
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.07
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.

this wav file eac3to created crashes mpc.

mediainfo says;
Codec : PCM
Codec/Family : PCM
Codec/Info : Microsoft PCM
Bit rate : 6144 Kbps (should be 4608 Kbps?)
Channel(s) : 6 channels
Sampling rate : 48 KHz
Resolution : 16 bits

[madshi]

eac3to generates a wrong wav file

source is 16bit 48khz 7.1ch Blu-ray lpcm and tried converting to ac3.

command line and output is:
F:\eac3to>eac3to f:\Innocence.pcm f:\Innocence.ac3 -8 -16 -down6
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.07
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.

this wav file eac3to created crashes mpc.

mediainfo says;
Codec : PCM
Codec/Family : PCM
Codec/Info : Microsoft PCM
Bit rate : 6144 Kbps (should be 4608 Kbps?)
Channel(s) : 6 channels
Sampling rate : 48 KHz
Resolution : 16 bits
Thanks for the information. I've checked and the "-down6" option results in some fields being set incorrectly in the wav header. I'll fix that in the next build. In the meanwhile you could try using the Aften build that ships with eac3to instead of 0.07. I'm not sure if the older Aften build ignores the incorrect wav header but there have been cases before where the older Aften build worked fine where Aften 0.07 failed.

[yonta]

In the meanwhile you could try using the Aften build that ships with eac3to instead of 0.07. I'm not sure if the older Aften build ignores the incorrect wav header but there have been cases before where the older Aften build worked fine where Aften 0.07 failed.

same thing happens with the Aften build in eac3to.zip.
rev.561 of Aften does create a file which is just stereo@640k and produces only noise.

[madshi]

Ok, thanks, will upload a fixed build later.

[ACrowley]

yes Madshi, ofcourse youre right...its reversed. DRC makes only Sense with proper Dialnorm. As written in the Guide
My Post was very vague and i dont force anybody to anything

@Nikos..i know :) When its a full lengt Track you cant use only a small Part.

However.. i dont use DRC and no DialNorm.. i encode with -31db ( -31db +31db = attenuate 0db) and no DRC Profile

[Nikos]

However.. i dont use DRC and no DialNorm.. i encode with -31db ( -31db +31db = attenuate 0db) and no DRC Profile
I agree the optimal solution.

One more question:
If one ac3 file has Dialog Norm. and DRC, the software ac3Filter bypass them or not?

[honai]

DRC can be enabled/disabled in AC3Filter. Don't know about Dial.Norm.

[Nikos]

DRC can be enabled/disabled in AC3Filter. Don't know about Dial.Norm.
This is another thing.
I mean if the ac3Filter can bypass the Dialog Norm. and DRC written as metadata inside the ac3 stream.

[honai]

This is another thing.
I mean if the ac3Filter can bypass the Dialog Norm. and DRC written as metadata inside the ac3 stream.

No, it's not, it's precisely the answer to your question. ;)

The DRC option in AC3Filter allows for respecting or ignoring the DRC data in the AC3 stream.

If you set AC3Filter to passthrough AC3 to an external receiver of SPDIF then the receiver (or your hand at the remote control) decides whether DRC should be applied.

[tebasuna51]

IOne more question:
If one ac3 file has Dialog Norm. and DRC, the software ac3Filter bypass them or not?

I don't know DialNorm (is not a problem a fix attenuation), but, in accordance with my test, the DRC values in ac3 stream are bypassed and ac3filter apply a owner function to make the DRC.

This kind of function can be applied for any source, not only ac3, and is a better solution than ac3 DRC but require more resources, is not habitual in receivers or standalone players.

[tebasuna51]

yes Madshi, ofcourse youre right...its reversed. DRC makes only Sense with proper Dialnorm. As written in the Guide
My Post was very vague and i dont force anybody to anything
Of course anybody can do anything, but "DRC makes only Sense with proper Dialnorm" ???

The values of DRC are modified by DialNorm but the concept is absolutely different:

DialNorm try to normalize all the sources at same level (dialogs at -31 bB), ... but is a lose battle, the others formats go 10 dBs upper.

But DRC is a help for some situations. The ideal solution is, like I say in precedent post, the player make the function at player time, but waiting for this solution we can use ac3 DRC, not necessarily with DialNorm.

[ACrowley]

When i have have mistaken something ,ok no Problem.
I can only read the Guide ,where you can find for "AC3 encoding"

The following point, however, cannot be stressed enough: In order for the Dynamic Range Compression to work as designed, the Dialogue Normalization parameter MUST be properly set first

When its wrong....nevermind. THX for the Infos. But No need to correct me with fat black letters
I dont care...as i say i dont use DRC in encoding/decoding

[big-guy]

I've just added this functionality to the latest eac3to version.

Thank you very much :D.
Keep up the great work :p.

[Nikos]

Thanks all for the helpfull informations.
You are very illustrative.
I feel happy!!!

[tebasuna51]

But No need to correct me with fat black letters

Sorry, really I want support your previous comments about don't care the Dolby specs.

My option is use DRC but not DialNorm, the guide, of course, try to be Dolby compliant.

Sorry.

[big-guy]

Somehow the audio I ripped isn't timed correctly... The movie itself is 01:59:55. While the extracted AC3 track is 01:49:55 in length ><. So if you have any way to fix that? Or is my BD disc just crap >_>... Otherwise I'll just use the AC3 448kbps stream from the DVD...

I used this command: eac3to bits0001.thd WoH.ac3
That should be correct right?
I demuxed the thd stream with xport. With the -a option.

[madshi]

That's strange. Which movie is that?

[ACrowley]

Sorry, really I want support your previous comments about don't care the Dolby specs.

My option is use DRC but not DialNorm, the guide, of course, try to be Dolby compliant.

Sorry.

everything Ok!
You helped me a lot with all these previous AC3 encoding/decoding DRC/Dialnorm Thing .
No i see Things clear. THX :)

[big-guy]

That's strange. Which movie is that?

It's The Wings of Honnêamise BD Rip. (I got the rip from usenet, everything was ok with it, no faulty rars and such).

If you could find a solution for such a weird error ? otherwise I'll just use the 448kbps audio from the DVD.

Also the subs are troublesome :mad:, need to enter all text manually...

[onlym3]

Has the "File size of the raw file doesn't fit" problem been solved yet? I get that problem on 3 different pcs and other than running on Windows XP they dont have anything in common. One has Nero 7.10 installed, the other 7.8.5 so i doubt they both suffer from a bad Nero install. And one doesnt have the mention dtsac3source filter installed so that cant be it either.

[madshi]

It's The Wings of Honnêamise BD Rip.
I don't have that BD. I have two other BDs with TrueHD, though. I'll double check whether the length of the extracted AC3 file is ok with those tracks. Maybe there's a bug in eac3to.

[madshi]

Has the "File size of the raw file doesn't fit" problem been solved yet? I get that problem on 3 different pcs and other than running on Windows XP they dont have anything in common. One has Nero 7.10 installed, the other 7.8.5 so i doubt they both suffer from a bad Nero install. And one doesnt have the mention dtsac3source filter installed so that cant be it either.
Do you have that problem with every E-AC3 track you try to convert? Or only with mono E-AC3 tracks?

[onlym3]

Do you have that problem with every E-AC3 track you try to convert? Or only with mono E-AC3 tracks?I tried 2 EAC3 tracks from 2 different HDDVDs, same result. And they arent mono.

[madshi]

I tried 2 EAC3 tracks from 2 different HDDVDs, same result. And they arent mono.
Have you tried GraphEdit? You need to rename it to Recode.exe. Then try "Nero File Source /Splitter -> Nero Audio Decoder 2 -> Default DirectSound Device". Does sound play for you that way?

I'm not sure, but probably you need the Blu-Ray/HD DVD plugin? Originally I thought you don't need it for E-AC3 decoding. But right now I guess you do.

[onlym3]

I'm not sure, but probably you need the Blu-Ray/HD DVD plugin? Originally I thought you don't need it for E-AC3 decoding. But right now I guess you do. 24.99$ later in can confirm that, works after i activated the Bluray/HDDVD Plugin.

[orbitlee]

I don't have that BD. I have two other BDs with TrueHD, though. I'll double check whether the length of the extracted AC3 file is ok with those tracks. Maybe there's a bug in eac3to.


@madshi

From what I know, TrueHD in Blu-ray contains DD 5.1 640K core. And, just like DTS-HD, you have to handle sync word emulation very carefully to avoid the incorrect length of extracted AC3. I wrote a tool to extract the DD 5.1 core from Blu-ray TrueHD, and tested on 3 BDs without any problem. I'll send the source code to you so you can merge into eac3to.

[madshi]

24.99$ later in can confirm that, works after i activated the Bluray/HDDVD Plugin.
Thanks, that's good to know.

[madshi]

@madshi

From what I know, TrueHD in Blu-ray contains DD 5.1 640K core. And, just like DTS-HD, you have to handle sync word emulation very carefully to avoid the incorrect length of extracted AC3. I wrote a tool to extract the DD 5.1 core from Blu-ray TrueHD, and tested on 3 BDs without any problem. I'll send the source code to you so you can merge into eac3to.
Thanks, the source code would be welcome and I'd love to integrate it into eac3to! :)

I think I was careful about the sync word stuff etc. I'm checking for the signature and I'm only accepting frames with a valid CRC. Every AC3 frame should have a valid CRC. Because of that valid AC3 frames should be easy to identify. I think it's more difficult with DTS. There CRC is optional and rarely used, as far as I know. Anyway, I'll compare your source code to mine and check if I can find significant differences... :)

[madshi]

@madshi

From what I know, TrueHD in Blu-ray contains DD 5.1 640K core. And, just like DTS-HD, you have to handle sync word emulation very carefully to avoid the incorrect length of extracted AC3. I wrote a tool to extract the DD 5.1 core from Blu-ray TrueHD, and tested on 3 BDs without any problem. I'll send the source code to you so you can merge into eac3to.
Could you make your DDCore tool available as an exe file? That way "big-guy" could test whether it works better for him than eac3to's solution. I've checked your code and it looks alright to me. But if your code doesn't work better than mine I'd prefer to keep mine.

Thanks!

[orbitlee]

Here is my DDCore executable.
http://www.sendspace.com/file/h1thbq

Usage: DDCore source.thd dest.ac3

It ONLY works on TrueHD from Blu-ray!

[pyrates]

Here is my DDCore executable.
http://www.sendspace.com/file/h1thbq

Usage: DDCore source.thd dest.ac3

It ONLY works on TrueHD from Blu-ray!

You can test it using the 300 hd dvd, that has a truehd 8 channel track. And how is it just for truehd from blu-ray? Isn't truehd the same on both kinds of discs?

[madshi]

Here is my DDCore executable.
http://www.sendspace.com/file/h1thbq

Usage: DDCore source.thd dest.ac3

It ONLY works on TrueHD from Blu-ray!
Thanks! So big-guy, does DDCore work better for you than eac3to with "Wings of Honnêamise"?

[madshi]

how is it just for truehd from blu-ray? Isn't truehd the same on both kinds of discs?
No, it's different. HD DVD is straight TrueHD with no AC3 in it. In Blu-Ray TrueHD consists of an AC3 track with added TrueHD blocks. The end result is more or less the same. HD DVD needs a bit less space.

[woah!]

i just tried ddcore on a thd track out of Vacancy BR disc and it works very well and fast :)

seemed to got the same result from eac3to aswell, if not faster heh, and correct times on both.

[madshi]

i just tried ddcore on a thd track out of Vacancy BR disc and it works very well and fast :)

seemed to got the same result from eac3to aswell, if not faster heh, and correct times on both.
Thanks for testing! :)

[woah!]

opps sorry that should be Ghostrider THD track not Vacancy .... still both give me correct files at about the same speed.

[big-guy]

Thanks! So big-guy, does DDCore work better for you than eac3to with "Wings of Honnêamise"?

Sorry, I've already deleted the source file. And just used the DVD track ><. Will test it the next time I rip a BD...

:thanks: for the hard work everybody :D.

[andybno1]

Don't know if this has been resolved somewhere in the thread but I had this problem and resolved it so I'm posting how I fixed it.

When trying to decode anything to wav (ddp or mlp) I got this:

E-AC3, 5.1 channels, 2:12:32, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.

Finally I discovered the actual problem was the dtsac3source-full.ax file which was corrupted or something. I had the same exact version on another computer which I had downloaded a couple of weeks before. That one was working, I knew, so I unregistered the old one (don't know if this is necessary)...

regsvr32 /u C:\WINDOWS\system32\dtsac3source-full.ax

...got the new one, and it worked.

If your raw file is 0 bytes and then eac3to crashes, chances are this is your problem.

Get some new dtsac3source from some different mirror or download site, register it and try again. That worked for me.

The version I have is 1.0.0.1, BTW.

Hope this helps.

where can I get the 1.0.0.1 version from? I am getting the above problem, dont know if it helps but I get this after getting the above message: http://img508.imageshack.us/img508/2788/crashay9.jpg

[madshi]

eac3to v1.14 released

http://madshi.net/eac3to.zip

(1) DTS dialog normalization can be removed now
(2) DTS core can be extracted from DTS-HD track now

Usage:
eac3to source.dts dest.dts
eac3to can detect if a DTS-HD stream is clean or not. If it's clean, the DTS core extraction should be perfect with no audio sync issues and no crackling sound etc. If the DTS-HD stream is not clean, eac3to will complain, but it will still do the extraction. In this case you need to check if audio is in sync throughout the whole movie.

I've tested the new DTS core extraction on all my HD DVD and Blu-Ray movies with such tracks. The tracks were demuxed by EvoDemux (for HD DVDs) and xport (for Blu-Rays). eac3to reported all tracks to be clean - except the "Perfume" tracks. This problem has been reported before. For whatever reason the "Perfume" DTS-HD tracks demuxed by EvoDemux are corrupt. Well, eac3to detects that, which is good. So, if eac3to doesn't complain you can feel safe that everything is fine.

One important thing to note: The current version of the "DTS/AC3/DD+" filter plays all DTS(-HD) streams from Blu-Ray and HD DVD incorrectly. Basically as soon as you seek, audio sync is lost. I've fixed the problem in the filter and uploaded a new version here:

http://madshi.net/dtsac3source.zip

This is not a unicode build, though, so it might not work with DTS/AC3/DD+ files in folders with "strange" file names. I hope we'll get a unicode build from orbitlee or Casimir later.

Another way to work around the DTS/AC3/DD+ filter problem is to mux the HD DVD and Blu-Ray DTS (core) tracks into a container, e.g. MKV or TS. That way the sync problems with seeking should also disappear.

[andybno1]

Thanks for the new eac3to and the upload of dtsacesource but could anyone tell me why I am getting this:

C:\audio>eac3to audio-in.eac3 audio-out.ac3 -640
E-AC3, 5.1 channels, 1:00:10, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1.93 GB.
The expected file size for 24 bit is 2.90 GB.
The real file size is 0 Bytes.

[madshi]

Thanks for the new eac3to and the upload of dtsacesource but could anyone tell me why I am getting this:
Probably you need the Nero HD DVD / Blu-Ray plugin.

[ACrowley]

THX Madshi ! Yure the Man :)

I had the same Problems with Perfume HDDVD DTS HD..
But its clean when the DTS HD is demuxed with SonicHDDemuxer in Graphedit :)

Your Source Filter Fix works good on dtscore Tracks , nice !

greets

PS ,maybe The Keymaker can update his gui with new Tab for TrueHD/DTS HD extraction etc

[honai]

Thanks for the new version, madshi.

One suggestion: Could you implement the padding fix that BeSliced performs on DTS tracks produced by some versions of Surcode? That way eac3to would be your one-stop solution for converting audio from HD discs. ;)

[madshi]

One suggestion: Could you implement the padding fix that BeSliced performs on DTS tracks produced by some versions of Surcode? That way eac3to would be your one-stop solution for converting audio from HD discs. ;)
I think eac3to v1.14 should already do that just fine. It will complain that the track is not clean, but you can ignore that warning in this case. To be sure I'd suggest that you test run the same Surcode DTS track through both BeSliced and eac3to. The end result should be 100% bit identical.

Btw, when using eac3to to automate Surcode the DTS tracks are already padding free.

[madshi]

P.S: Wait, I'm wrong. eac3to v1.14 will not do that. I'll add that to the next build.

[andybno1]

Probably you need the Nero HD DVD / Blu-Ray plugin.

is this a seperate download to the actual installer of nero ultra? cause I chose typical install with ultra serial. Oh yeh I have version 7.1.0

[Nikos]

is this a seperate download to the actual installer of nero ultra? cause I chose typical install with ultra serial. Oh yeh I have version 7.1.0
You need the separate HD DVD/Blu-ray serial number!!!

[andybno1]

ah right lol

[madshi]

Correct, it's not a separate install. You don't have to install anything. But you need to register a separate license key. When I had bought the plugin I was waiting for a download link for the plugin installer. Only after a while I understood that I don't need to install anything, but that registering the license key was enough to make it work... :)

[Xorcist]

Thanks for this great software!

I have used it to convert E-AC3 track on Heroes HD-DVD to AC3, but for one of the episodes (Episode 5) eac3to reports that the E-AC3 is not clean and tells me to use delaycut to fix it. However delaycut doesn't seem to support E-AC3.. Or am I mistaken?

EVOs has been demuxed using EVOdemux 0.627 b7.

[The_Keymaker]

@Acrowley,

I don't think madshi added any new OPTIONS in the latest version of eac3to. Madshi's updates in v.1.14 are automatically invoked based upon the filename extension(s).

If there are new options I will try and update EAC3toGUI. Are there any new command line options I am not aware of?

Regards,
The_Keymaker

[andybno1]

Correct, it's not a separate install. You don't have to install anything. But you need to register a separate license key. When I had bought the plugin I was waiting for a download link for the plugin installer. Only after a while I understood that I don't need to install anything, but that registering the license key was enough to make it work... :)

where do you put the serial for the blu-ray/hd dvd plugin?

[SvT]

where do you put the serial for the blu-ray/hd dvd plugin?

Once you've installed Nero you open Nero ProductSetup and select Licences panel. Click add and insert your serial to get the plugin to work.

Goodluck !

[madshi]

Thanks for this great software!

I have used it to convert E-AC3 track on Heroes HD-DVD to AC3, but for one of the episodes (Episode 5) eac3to reports that the E-AC3 is not clean and tells me to use delaycut to fix it. However delaycut doesn't seem to support E-AC3.. Or am I mistaken?

EVOs has been demuxed using EVOdemux 0.627 b7.
Ehm yes, you're right. delaycut (sadly) doesn't support E-AC3 yet. What I can offer to you is a switch to either ignore dirty E-AC3 tracks (the E-AC3 audio decoder might eventually stumble then - but probably not) or to remove any dirty audio frames (which may affect audio sync). What do you think?

[Xorcist]

Ehm yes, you're right. delaycut (sadly) doesn't support E-AC3 yet. What I can offer to you is a switch to either ignore dirty E-AC3 tracks (the E-AC3 audio decoder might eventually stumble then - but probably not) or to remove any dirty audio frames (which may affect audio sync). What do you think?

That would be really cool. It would give the user 3 options with dirty E-AC3 tracks; plain old error like today, possibly a good AC3 track if the decoder is ok with errors or an AC3 track that is out of sync depending on the number of dirty audio frames. If eac3to could print out the number of frames (and possibly timecode) removed in the last case it would help as well, but the most important thing is to have the switch you talk about.

I take it you are talking about possible improvements in new versions of eac3to?

[madshi]

I take it you are talking about possible improvements in new versions of eac3to?
Yes...

[Xorcist]

Ok, just wanted to make sure you weren't talking of any current options that I somehow hadn't noticed :)

[madshi]

I don't think madshi added any new OPTIONS in the latest version of eac3to. Madshi's updates in v.1.14 are automatically invoked based upon the filename extension(s).
That's correct. It's just that now "dts", "dtshd" and "ac3" are accepted as input formats, if appropriate output formats are used. Also "eac3" etc is accepted as output format, if "eac3" (or similar) is also the input extension.

[sl1pkn07]

exist eac3to tool for linux (native)?

[madshi]

exist eac3to tool for linux (native)?
Nope.

[sl1pkn07]

is possible port your program to linux?

[andybno1]

Once you've installed Nero you open Nero ProductSetup and select Licences panel. Click add and insert your serial to get the plugin to work.

Goodluck !

cheers for the hint ;)

EDIT: small problem I entered my serial and in product setup I get ? where the serial should be and eacto still not processing the file grrrr I dont wanna try sonic cause I could never get the delay synced in the muxed file

[madshi]

is possible port your program to linux?
Not really because big parts of it are relying on Windows DirectShow filters.

[madshi]

EDIT: small problem I entered my serial and in product setup I get ? where the serial should be and eacto still not processing the file grrrr I dont wanna try sonic cause I could never get the delay synced in the muxed file
If you have properly bought the plugin you can contact Nero customer support about why the serial number doesn't work in your case... ;)

[andybno1]

If you have properly bought the plugin you can contact Nero customer support about why the serial number doesn't work in your case... ;)

all I have to say is BUGGER lol

[dvd-freak]

If you have properly bought the plugin you can contact Nero customer support about why the serial number doesn't work in your case... ;)

I think, I would REALLY LIKE TO BE THERE when this call is made.

lol :rolleyes:




@mashi:
Any news on the Dolby TrueHD dialnorm issue? DRC can be disabled, right?

[madshi]

Any news on the Dolby TrueHD dialnorm issue? DRC can be disabled, right?
DRC is luckily disabled by default in the current Nero TrueHD decoder version. I've not found a way to disable Dialnorm yet. If only I had some kind of TrueHD specification/documentation!

[andybno1]

ok I took the plundge and purchased the plugin all registered now I am at a new fence with the followin error

C:\audio>eac3to audio-in.eac3 audio-out.ac3 -640
E-AC3, 5.1 channels, 1:00:10, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
Audio Decoder output was 24 bit this time.
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...
Starting "aften" failed. A valid *.wav file was created successfully, though.

[madshi]

ok I took the plundge and purchased the plugin all registered now I am at a new fence with the followin error
Well, you're nearly there. The Nero E-AC3 decoder evidently works fine now for you. But for whatever reason aften didn't get started correctly. Do you have a subfolder named "Aften" in the eac3to folder? And is there an "aften.exe" in that "Aften" subfolder?

[andybno1]

I downloaded the last zip file and put all exe files into the main folder......... is this wrong?

[sportcenter]

so not possible to encode anything above 640 ac3?...

[ACrowley]

so not possible to encode anything above 640 ac3?...

No
maximum is 640kbps

[madshi]

I downloaded the last zip file and put all exe files into the main folder......... is this wrong?
Yes, that is wrong. Just unzip and leave everything as it is.

[Xorcist]

I earlier mentioned having problems with dirty EAC3 tracks on some Heroes episodes. Turns out my harddrive were having some problems and the rip actually had some messed up files. I re-ripped and eac3to worked perfectly :)

[madshi]

I earlier mentioned having problems with dirty EAC3 tracks on some Heroes episodes. Turns out my harddrive were having some problems and the rip actually had some messed up files. I re-ripped and eac3to worked perfectly :)
That's good to know. So eac3to told you the files were messed up... :)

[sportcenter]

No
maximum is 640kbps

so for doing a lossless 1.5mbps .wma audio track i would need to use .wavs, then vegas 7 or equivalent audio encoder to give me the .wma?

[Exiton]

Question for anyone who knows. I've rebuilt an .EVO containing a single TrueHD track. When I use EAC3TO, the raw file it generates is 0kb. I can do conversions of eac3 -> ac3 all day long no problem. Anyone know the possible problem leading to the raw output failing?

EDIT: I've got the following in my system because I can see the last three in graphedit. The nero filter works because the standard EAC3 -> AC3 conversions work.

(1) Nero HD-DVD/Blu-Ray plugin
(2) "Nero Splitter"
(3) "Nero Audio Decoder 2"
(4) "Dump" filter

EDIT #2: I think it may be a bad source file. I'm doing some testing and will post back. I could duplicate the error on another machine so I think it's the source.

[sportcenter]

Question for anyone who knows. I've rebuilt an .EVO containing a single TrueHD track. When I use EAC3TO, the raw file it generates is 0kb. I can do conversions of eac3 -> ac3 all day long no problem. Anyone know the possible problem leading to the raw output failing?

EDIT: I've got the following in my system because I can see the last three in graphedit. The nero filter works because the standard EAC3 -> AC3 conversions work.

(1) Nero HD-DVD/Blu-Ray plugin
(2) "Nero Splitter"
(3) "Nero Audio Decoder 2"
(4) "Dump" filter

what is the error message say?....are you putting

eac3to.exe audio.eac3 audio.wav -nero

thats the only way you will get a RAW file that stays...otherwise it creates the RAW filein the begining and then converts to .ac3 or whatever...

[ACrowley]

what is the error message say?....are you putting

eac3to.exe audio.eac3 audio.wav -nero

thats the only way you will get a RAW file that stays...otherwise it creates the RAW filein the begining and then converts to .ac3 or whatever...


You can set the output Extension to .raw when you want a .raw or to wav (5.1 wave) or .wavs (mono waves).

Or to ac3 /dts
So youve a lot of options :)

so for doing a lossless 1.5mbps .wma audio track i would need to use .wavs, then vegas 7 or equivalent audio encoder to give me the .wma?
When you need 6 mono wave use .wavs output extension.
Then feed your prefered encoder to encode to whatever, for example to WMA10pro or WMA 9.2 lossless.
But note that WMA lossless wont play smooth with WVC1 VBR Peak Video on your Xbox360.

[Nasgul]

I get the same error than Yonta trying to convert a bluray lpcm to ac3... Any ideas?


>eac3to.exe ..\..\Remux\rock_3.pcm ..\..\Rock3_448.ac3 -16 -448
RAW, 5.1 channels, 0:23:30, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.07
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.


I tried both Aften 0.07 and the version that ships with eac3to, same error.

Regards,
Nas

[liquidator87]

I see that eac3to encode to flac directly from raw samples. Let's say I want to use the same "strategy" for encoding, but I've 6 mono wave files, how can I obtain the big raw multichannel file? Can wavewizard do this?

Edit: yes it can

[sportcenter]

But note that WMA lossless wont play smooth with WVC1 VBR Peak Video on your Xbox360.

yeah i know, but it sounds excellent on the pc...

thanks

[madshi]

I get the same error than Yonta trying to convert a bluray lpcm to ac3... Any ideas?

>eac3to.exe ..\..\Remux\rock_3.pcm ..\..\Rock3_448.ac3 -16 -448
RAW, 5.1 channels, 0:23:30, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.07
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.

I tried both Aften 0.07 and the version that ships with eac3to, same error.
Which eac3to version is that? And can I please see the output of the Aften version that ships with eac3to?

[Nasgul]

Hi madshi,


Which eac3to version is that? And can I please see the output of the Aften version that ships with eac3to?


I tried both Eac3to versions 1.14 and 1.13.
I´m not at my home pc right now, but I´m pretty sure the error message looked exactly the same with the aften version that shipped with eac3to.
I will check again this evening.

Thanks,
Nas

[Creator1]

Hello to all,

First I want to send many thanks to madshi for this great program! :thanks:

I used eac3to with success but I still have a small problem that maybe some of you guys can give me a tip on?

What I am trying to do is import LPCM to Scenarist HDMV (Blu-Ray). What I do is this:

1) I take an eac3 source (DD+) and use eac3to to transform it into ac3 (DD).

2) eac3to goes into it's different stage of treatment and at some point, it generates a big .WAV file (from the .raw file) which contains the 5.1 channels. It is about 6GB.

3) I then open MUI Generator in Scenarist and import the 6GB WAV file into it. It auto detects the file as a LPCM with 6 channels L-R-SL-SR-C + Lfe. So all is super so far.

4) Then I click the "Create files" button so that the MUI Generator c reates the importation stats files but it gives an error saying that the "ChannelAssign" specified does not correspond to the data in the file?

So, in other words, the tool correctly auto-detects the content of the wav files but still gives me an error on the channels assignment. I tried all the other possibilities (8 channel, mono, 1 ch, 4ch, etc.) and it always says this?

What I can think of is maybe there are "names" for each channels embedded in a wav file and the names created by eac3to (probably more by Aften I think) are not compliant so the program does not know which channel is the LFE, which channel is the center, etc.?

Anyone had that problem or can give me a tip that I can try out? Are the channels in a WAV file "named" like I said and if so can I use a program like Audacity to change the names?

Basically, I am trying to make a blu-ray disc with a LPCM soundtrack coming from a DD+ source.

Thanks in advance for any help! :)

[harlis]

Hi

First of all Tanks for this application! (seems to be a cool tool)

I tried to use it to convert eac3 traks to ac3.
used v1.13 and v.1.14
I have Nero installed and the bluray pug registered i have also the sonic filters installed but i always get the same result - nothing :(

this is the commandline
The dump filter (from the graphedit pack) is installed (i've registered it again manualy to be sure)

It creates a 2. file named fdolby.ddp (dialognorm removed i guess) and then aborts.

H:\d\a>eac3to fdolby.eac3 fdolby.ac3 -640
E-AC3, 5.1 channels, 1:58:29, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Getting "Dump" instance failed.

so any idea what i'm doing wrong or what i'm possibly missing?

[Creator1]

Hi

First of all Tanks for this application! (seems to be a cool tool)

I tried to use it to convert eac3 traks to ac3.
used v1.13 and v.1.14
I have Nero installed and the bluray pug registered i have also the sonic filters installed but i always get the same result - nothing :(

this is the commandline
The dump filter (from the graphedit pack) is installed (i've registered it again manualy to be sure)

It creates a 2. file named fdolby.ddp (dialognorm removed i guess) and then aborts.

H:\d\a>eac3to fdolby.eac3 fdolby.ac3 -640
E-AC3, 5.1 channels, 1:58:29, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Getting "Dump" instance failed.

so any idea what i'm doing wrong or what i'm possibly missing?

This is a problem with the Dump filter. It is not correctly registered. I had the same problem before I registered it. I know it sounds weird because you said that you registered it but the problem is still that clearly from the log you just typed. Did you move the Dump.ax file after registering it. I think you need to not move it but I am not sure.

[sportcenter]

Hi

First of all Tanks for this application! (seems to be a cool tool)

I tried to use it to convert eac3 traks to ac3.
used v1.13 and v.1.14
I have Nero installed and the bluray pug registered i have also the sonic filters installed but i always get the same result - nothing :(

this is the commandline
The dump filter (from the graphedit pack) is installed (i've registered it again manualy to be sure)

It creates a 2. file named fdolby.ddp (dialognorm removed i guess) and then aborts.

H:\d\a>eac3to fdolby.eac3 fdolby.ac3 -640
E-AC3, 5.1 channels, 1:58:29, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Getting "Dump" instance failed.

so any idea what i'm doing wrong or what i'm possibly missing?

you are missing the dumpfilter

also make sure your command line is:

eac3to fdolby.eac3 fdolby.ac3 -640 -nero

so you avoid DRC and normalization

[harlis]

Thanks to you all

I removed and reinstalled everything now the dump stuff worked (seems like i moved the dir) but got a message like 0byte something....
but that's caused by my nero stuff is to old or faulty :(

so i used the sonic switch and now it WORKED :)

Thanks again for all the help and for this Great Tool :)

[Creator1]

Thanks to you all

I removed and reinstalled everything now the dump stuff worked (seems like i moved the dir) but got a message like 0byte something....
but that's caused by my nero stuff is to old or faulty :(

so i used the sonic switch and now it WORKED :)

Thanks again for all the help and for this Great Tool :)

It is highly recommenced to use the Nero pluging though. It does the best job overall for the sound quality. But it is good that it works now for you.

[sportcenter]

so i used the sonic switch and now it WORKED :)


but remember you will have DRC normalization and compression...translation, the dvd audio will have better peaks than your encode....

you need to use nero to avoid that

[harlis]

Yes I will have to check my nero and maybe i need to update it so i will get better results.

[Creator1]

but remember you will have DRC normalization and compression...translation, the dvd audio will have better peaks than your encode....

you need to use nero to avoid that

From what my friend told me, DRC is evil and pretty much destroy a surround soundtrack. I think he said it makes everything flat. You definitely want to avoid that.

Can someone tell me what dialog normalization is though? From what I could see so far, it seems to be the thing they use on DVDs and now HD-DVDs and Blu-Ray so that the sound is way lower than, for example, a video game sound. I did a test with dialNorm removed and the movie was playing loud on -50dB on my amp and I normally put any movie at -36dB or something like that to have the same volume level. I did a encode with "keepNorm" but I have yet have the time to compare both tracks but on the PC Speakers it seemed like the new encode is lower in volument like I suspected.

Am I right in my understanding of dialNorm? If so, I guess I will keepNorm always because I am used to put the volume high on movies and I may get "surprised" everytime if the volume is too loud on my BD-R projects.

Is there a negative in keeping dialog normalization on? I know DRC is bad but is dialNorm bad for the sound quality or is it just a volume thing?

Thanks for any help.

[madshi]

What I am trying to do is import LPCM to Scenarist HDMV (Blu-Ray). What I do is this:

1) I take an eac3 source (DD+) and use eac3to to transform it into ac3 (DD).

2) eac3to goes into it's different stage of treatment and at some point, it generates a big .WAV file (from the .raw file) which contains the 5.1 channels. It is about 6GB.

3) I then open MUI Generator in Scenarist and import the 6GB WAV file into it. It auto detects the file as a LPCM with 6 channels L-R-SL-SR-C + Lfe. So all is super so far.

4) Then I click the "Create files" button so that the MUI Generator c reates the importation stats files but it gives an error saying that the "ChannelAssign" specified does not correspond to the data in the file?
The WAV file eac3to creates doesn't have any channel assignment information in it. Maybe the MUI Generator doesn't like that. But it's difficult to create wav files that everybody likes, especially once the wav files get bigger than 2GB or even 4GB.

Can the MUI Generator accept raw PCM data instead of a wav file?

[madshi]

eac3to fdolby.eac3 fdolby.ac3 -640 -nero
"-640 -nero" isn't really needed because both switches are default in the newer eac3to versions.

[Creator1]

The WAV file eac3to creates doesn't have any channel assignment information in it. Maybe the MUI Generator doesn't like that. But it's difficult to create wav files that everybody likes, especially once the wav files get bigger than 2GB or even 4GB.

Can the MUI Generator accept raw PCM data instead of a wav file?

Thank you for the reply.

I tried the .raw file also in the MUI generator and it did create the stats file but when I tried to import the .ves file in scenarist it said it could not read the file. I think it was "mixed up" and that it does not accept raw. In the doc it says it accepts ".lpcm" and ".wav". I don't think I can import 6 channels separately either from what I saw. It needs to be a single .wav file with correct channel assignment.

How does channel assigment works? Can I edit that in a WAV editor or in a HEX editor?

Thank you.

[madshi]

In the doc it says it accepts ".lpcm" and ".wav".
"lpcm" is raw data. However, it's usually big endian while eac3to's raw output is usually little endian, IIRC. You could try using sox to convert eac3to's raw output to little endian. Maybe that helps.

How does channel assigment works? Can I edit that in a WAV editor or in a HEX editor?
It needs a slightly different wav header than the one eac3to is writing. You could try rewriting the file with some wav compatible tools, e.g. wavwizard. Maybe those make eac3to's wav file acceptable for Scenarist.

[ACrowley]

@creator1

Whn you talking about Scenarist Ac3 Encoder, you need 16bit waves.
Simply use .wavs Output with 16bit donwconvert in eac3to.
These waves are compatible with the Scenarist Audio Transcoder to load each channel

[Creator1]

"lpcm" is raw data. However, it's usually big endian while eac3to's raw output is usually little endian, IIRC. You could try using sox to convert eac3to's raw output to little endian. Maybe that helps.


It needs a slightly different wav header than the one eac3to is writing. You could try rewriting the file with some wav compatible tools, e.g. wavwizard. Maybe those make eac3to's wav file acceptable for Scenarist.

Excellent tips. I will try the sox thing first and then the other one if it fails. Thanks!

I have another question for you. It seems like eac3to's default bits are 24 bits for the sound right? The first line that appears when eac3to starts, is it the stats of the source file? Because on my eac3 (DD+) it said 24 bits but I am wondering if it is the default value that was said there or if it really is that the source file is 24 bits? I think the source file is 24 bits because it describes the length of the file (2:00:34 for example), the format (DD+), etc.

@creator1

Whn you talking about Scenarist Ac3 Encoder, you need 16bit waves.
Simply use .wavs Output with 16bit donwconvert in eac3to.
These waves are compatible with the Scenarist Audio Transcoder to load each channel

Yes but it's Scenarist HDMV, not Scenarist SD DVD and when I loaded up my 24 bits WAV file it auto-detected as 24 bits and it seems to be acceptable to Scenarist since it is in the possible pop-up option for LPCM.

Do you know of a way to import all the 6 channels separately to create a 5.1 track in Scenarist HDMV? I will need to check that out then if it is a possibility.

[madshi]

I have another question for you. It seems like eac3to's default bits are 24 bits for the sound right? The first line that appears when eac3to starts, is it the stats of the source file? Because on my eac3 (DD+) it said 24 bits but I am wondering if it is the default value that was said there or if it really is that the source file is 24 bits? I think the source file is 24 bits because it describes the length of the file (2:00:34 for example), the format (DD+), etc.
There's no thing like a specific bitdepth for AC3 and E-AC3 tracks. The (E-)AC3 headers don't contain any such information. In the end it's the AC3 decoder which decides which bitdepth it outputs. The Nero audio decoder usually outputs 24bits. eac3to just takes what the E-AC3 decoder outputs.

[Creator1]

There's no thing like a specific bitdepth for AC3 and E-AC3 tracks. The (E-)AC3 headers don't contain any such information. In the end it's the AC3 decoder which decides which bitdepth it outputs. The Nero audio decoder usually outputs 24bits. eac3to just takes what the E-AC3 decoder outputs.

Oh ok, so the eac3to utility just records what Nero's E-AC3 plugin outputs when playback? That is pretty cool.

I suppose that for .LPCM or .WAV file there is a bit depth though inside the file?

[ACrowley]

Oh ok, so the eac3to utility just records what Nero's E-AC3 plugin outputs when playback? That is pretty cool.

I suppose that for .LPCM or .WAV file there is a bit depth though inside the file?


BluRay LPCM is 16Bit or 32Bit
Wave can be from 8-32Bit

Its a huge difference in Filesize.
But, imho, when you convert LPCM 24bit to 16bit Flac its enough. Its not lossless anymore. You wont hear any big differrence
24Bit Filesite is not worth it, imho

[Creator1]

BluRay LPCM is 16Bit or 32Bit
Wave can be from 8-32Bit

Its a huge difference in Filesize.
But, imho, when you convert LPCM 24bit to 16bit Flac its enough. Its not lossless anymore. You wont hear any big differrence
24Bit Filesite is not worth it, imho

Ok thanks.

I am just trying to make the LPCM work for the sake of testing. I may end up using the DTS 1.5Mbps re-encode since I am not an audiophile much but more of a videophile. It's always good to have lossless on audio too though.

[maya]

Ok, I am new at using the eac3to program. I have used evo demux to strip everything out of the first feature evo file for "A Perfect Storm" and now I am left with the video and the single DD+ audio track. I open up Eac3toGui and set up the source file, it automatically sets the output file, and I set the rate to 640kbits, Nero as force filter, and leave everything else at defualt. When I hit convert the black command prompt screen comes up and states the following:

I: Evodemux ARea>"C:\Documents and Settings\Ryan\Desktop\eac3to.exe" "I:\Evodemux ARea\Perfect Storm 1.EVO"
"I:\Evodemux ARea\Perfect Storm1.ac3"-640-nero-24
"C:\Documents and Settings\Ryan\Desktop\eac3to.exe"is not recognized as an internal or external command, iperable program or batch file.
I:\Evodemux ARea>

I have no idea what this means. Does it have something to do with certain filters not being installed? I have the newest version of nero, and have selected "Nero" as the force filter. I don't really understand what the following are, that are needed for the program to work:

Nero File Source/Splitter
Nero Audio Decoder 2
Dump Filter

I did a search of my c: drive and saw that these things are there as .ax extensions (whatever that means). I do not have the Nero HD-DVD/Blu Ray plugin-assumed I would never need it due to the fact that I never watch VC-1 on my PC. I put these VC-1 files on my Tvix 4100 and play them from there.

I guessed that since I own the newest version of Nero, that all the Nero stuff above was in there somewhere. Any help would be greatly appreciated.

[madshi]

Oh ok, so the eac3to utility just records what Nero's E-AC3 plugin outputs when playback?
Yep.

I suppose that for .LPCM or .WAV file there is a bit depth though inside the file?
For WAV yes. For LPCM no. LPCM contains nothing but the sound samples. There's no "header" or any other information in a LPCM file about how many channels there are or which bitdepth is being used. However, eac3to tries to find out the LPCM bitdepth by analyzing the audio data.

BluRay LPCM is 16Bit or 32Bit
It's 16bit or 24bit.

[madshi]

I open up Eac3toGui and set up the source file, it automatically sets the output file, and I set the rate to 640kbits, Nero as force filter, and leave everything else at defualt. When I hit convert the black command prompt screen comes up and states the following:

I: Evodemux ARea>"C:\Documents and Settings\Ryan\Desktop\eac3to.exe" "I:\Evodemux ARea\Perfect Storm 1.EVO"
"I:\Evodemux ARea\Perfect Storm1.ac3"-640-nero-24
"C:\Documents and Settings\Ryan\Desktop\eac3to.exe"is not recognized as an internal or external command, iperable program or batch file.
Eac3toGui is just a wrapper around eac3to. So you need to download eac3to, too, and put both eac3to and eac3toGui into one folder.

I do not have the Nero HD-DVD/Blu Ray plugin-assumed I would never need it due to the fact that I never watch VC-1 on my PC. I put these VC-1 files on my Tvix 4100 and play them from there.
I'm not 100% sure but it seems that for E-AC3 decoding you do need the Nero HD-DVD/Blu-Ray plugin.

[TheSof]

Please forgive me if this has been asked before.

What happens if the eac3, lpcm or dts has more than 5.1 channels?

Does surcode support dts es? In the surcode log file, at the start of an encode it reports "ES=NO".

Does aften and eac3 support more than 5.1?

Cheers, and many thanks to madshi for his work on eac3to.

[madshi]

Please forgive me if this has been asked before.

What happens if the eac3, lpcm or dts has more than 5.1 channels?

Does surcode support dts es? In the surcode log file, at the start of an encode it reports "ES=NO".

Does aften and eac3 support more than 5.1?
LPCM with 7.1 is supported by eac3to. E.g. you can do LPCM -> FLAC with full 7.1. You need to tell eac3to that the LPCM track has 8 channels, though, by activating the "-8" option.

I've not seen any E-AC3 tracks with more than 5.1 yet. As a result I don't know what Nero's audio decoder does with such tracks. Maybe it puts out full 7.1, maybe it only puts out 5.1, maybe it doesn't put out anything. I've no idea. Let's revisit this topic when the first 7.1 E-AC3 track shows up.

[ACrowley]

Does surcode support dts es? In the surcode log file, at the start of an encode it reports "ES=NO".

No. Surcode can only encode to 16/48 dts 5.1.
DTS Pro Series or DTS HD Master Suite can encode 6.1 ( 24/96 too)

Tranzcode/Azidts/NiscDTSSource cant decode to 7 mono waves from DTS ES 6(+)1 They all decode to 5.1 only. In this Case the BC is blends into BL BR because 6.1 DTS ES is downwards Compatible. So you shouldnt loose any BC Sounds in theory.
Its the the same with DD EX

Possible Methods to decode/record all Channels from DTS ES and DD EX:

To get 7CH mono Waves from DTS ES 6.1 you can decode/record it via ASIO in Sony Vegas
Play the DTS ES 6.1 Discrete in PowerDVD (6.1 Output) and record it through ASIO with Sony Vegas.
It works via graphedit and CyberlinkAudioDecoder too with dts es 6.1 discrete

- Sony Vegas 6.1 Project ,recording all (already)mapped 7 Channels through Creative ASIO Input
http://img251.imageshack.us/img251/4187/sshot4as3.th.png (http://img251.imageshack.us/my.php?image=sshot4as3.png)

Now youve 7 mono Waves. Ofcourse you cant encode Discrete 6.1 AC3.
But you can encode the BC in the Matrix to Dolby Digital EX in Sony Vegas.

Other Method for me is my Audigy2ZS Card. It has a build in Decoder which is capable to decode DD EX and DTS ES 6.1.
Works fine with Nero Showtime (HDDVDPlugin) Playback and Output Setting to SPDIF.
Recording again with Vegas through ASIO like above. So i am able to get 7 mono Waves from 6.1 Sources (DD EX,DTS ES)
I succesfully made some DD EX 5(6).1 reencodes from DTS ES 6.1 and DD EX with this Methods.

Alternative to encode more than 5.1 Discrete Channels is FLAC. I made some 6.1 Discrete encodes with FLAC, works perfect
But these are only Experiments for me. I have 1 5.1 Setup and im still satisfied with 5.1.
6.1 (ore more) Formats are downwards compatible...so its no Problem
Does aften and eac3 support more than 5.1?
No 6.1 Discrete AC3 isnt possible. Only Dolby Digital EX delivers Matrix encoded BC Channel.

[TheSof]

I thought I had everything working but now I have a new problem.

Running a 1.5mbps dts track via spdif to my processor and all is well.

But a 1.5mbps surcode track plays with stuttering, crackling popping sounds. But decoding this and spitting out analog to my processor and all is OK.

Does anyone know why this is?

[Creator1]

I: Evodemux ARea>"C:\Documents and Settings\Ryan\Desktop\eac3to.exe" "I:\Evodemux ARea\Perfect Storm 1.EVO"
"I:\Evodemux ARea\Perfect Storm1.ac3"-640-nero-24
"C:\Documents and Settings\Ryan\Desktop\eac3to.exe"is not recognized as an internal or external command, iperable program or batch file.
I:\Evodemux ARea>

I did a search of my c: drive and saw that these things are there as .ax extensions (whatever that means). I do not have the Nero HD-DVD/Blu Ray plugin-assumed I would never need it due to the fact that I never watch VC-1 on my PC. I put these VC-1 files on my Tvix 4100 and play them from there.

The error is because the eac3toGUI program cannot find the eac3to.exe program on your system.

Like madshi said, eac3toGUI is only the user interface to facilitate the usage of eac3to. The real eac3to program is named eac3to.exe and needs to be downloaded separately. After you have downloaded eac3to.exe, you can start eac3toGUI and then go to the menu where you can specify where the eac3to.exe application is and point to it so that eac3toGUI knows where to find the application.

Also, you will need Nero HD-DVD/Blu-Ray plugin. eac3to didn't work for me until I got that. Even if you have the full NERO latest version installed, if you did not pay the extra 25$ for the HD-DVD/Blu-Ray plugin it is not "activated" and thus not functional. You need to go to NeroProductSetup on the start menu and then go to "licenses" or something like that and add a serial number to unlock the hd-dvd/blu-ray plugin.

For WAV yes. For LPCM no. LPCM contains nothing but the sound samples. There's no "header" or any other information in a LPCM file about how many channels there are or which bitdepth is being used. However, eac3to tries to find out the LPCM bitdepth by analyzing the audio data.

Ok that's pretty neat. I didn't have time to try the big endian versus little endian thing on the .raw file but I'll try it out when I can. I doubt it will work but who knows until I try. :thanks:

[madshi]

I thought I had everything working but now I have a new problem.

Running a 1.5mbps dts track via spdif to my processor and all is well.

But a 1.5mbps surcode track plays with stuttering, crackling popping sounds. But decoding this and spitting out analog to my processor and all is OK.

Does anyone know why this is?
Was that Surcode file created by eac3to or did you use Surcode yourself? When not being automated by eac3to, Surcode adds some padding to the DTS files. I've no idea if this may be responsible for the problems you're having or not. Maybe it's something else. Could be that your receiver isn't fully DTS compatible. The eac3to created Surcode files all play perfectly fine with my receiver at least.

[ACrowley]

I thought I had everything working but now I have a new problem.

Running a 1.5mbps dts track via spdif to my processor and all is well.

But a 1.5mbps surcode track plays with stuttering, crackling popping sounds. But decoding this and spitting out analog to my processor and all is OK.

Does anyone know why this is?


Run it through Besclied 0.3
Simply start Besliced and Dra&Drop your dts into the Windows and select fix.
The fixed dts should be clean.
Its a "must be" to fix surcode dts !
eac3to should use a own internal fix to fix surcode dts

I never had any Problmes with Surcode dts

[TheSof]

Was that Surcode file created by eac3to or did you use Surcode yourself? When not being automated by eac3to, Surcode adds some padding to the DTS files. I've no idea if this may be responsible for the problems you're having or not. Maybe it's something else. Could be that your receiver isn't fully DTS compatible. The eac3to created Surcode files all play perfectly fine with my receiver at least.

Yeah I was using ac3to. I can't understand it. I'm using a Denon 5803 (A1SR) amp/processor which should be fine. It could be my soundcard (RME), but it passes non surcode 1.5mbps dts fine.

[TheSof]

Run it through Besclied 0.3
Simply start Besliced and Dra&Drop your dts into the Windows and select fix.
The fixed dts should be clean.
Its a "must be" to fix surcode dts !
eac3to should use a own internal fix to fix surcode dts

I never had any Problmes with Surcode dts

I think eac3to does this, but I ran it through besliced just in case, and it didn't help.

Must just be my processor at fault if the computer can decode the surcode dts stream OK. These top of the range thx ultra 2 processors are not all they're cracked up to be.

[maya]

The error is because the eac3toGUI program cannot find the eac3to.exe program on your system.

Like madshi said, eac3toGUI is only the user interface to facilitate the usage of eac3to. The real eac3to program is named eac3to.exe and needs to be downloaded separately. After you have downloaded eac3to.exe, you can start eac3toGUI and then go to the menu where you can specify where the eac3to.exe application is and point to it so that eac3toGUI knows where to find the application.

Also, you will need Nero HD-DVD/Blu-Ray plugin. eac3to didn't work for me until I got that. Even if you have the full NERO latest version installed, if you did not pay the extra 25$ for the HD-DVD/Blu-Ray plugin it is not "activated" and thus not functional. You need to go to NeroProductSetup on the start menu and then go to "licenses" or something like that and add a serial number to unlock the hd-dvd/blu-ray plugin.



Ok that's pretty neat. I didn't have time to try the big endian versus little endian thing on the .raw file but I'll try it out when I can. I doubt it will work but who knows until I try. :thanks:

thanks to both of you. I am at work now but will report back on how things go after I get home.

[TheSof]

More trouble for me, this time with FLAC. As soon as I mkvmerge some avc with FLAC, explorer.exe stops responding whenever I go anywhere near the mkv with FLAC.
Can anyone help me (again:)?

[honai]

That's likely a problem with Haali Matroska splitter. You should ask in the respective threads.

[maya]

The error is because the eac3toGUI program cannot find the eac3to.exe program on your system.

Like madshi said, eac3toGUI is only the user interface to facilitate the usage of eac3to. The real eac3to program is named eac3to.exe and needs to be downloaded separately. After you have downloaded eac3to.exe, you can start eac3toGUI and then go to the menu where you can specify where the eac3to.exe application is and point to it so that eac3toGUI knows where to find the application.

Also, you will need Nero HD-DVD/Blu-Ray plugin. eac3to didn't work for me until I got that. Even if you have the full NERO latest version installed, if you did not pay the extra 25$ for the HD-DVD/Blu-Ray plugin it is not "activated" and thus not functional. You need to go to NeroProductSetup on the start menu and then go to "licenses" or something like that and add a serial number to unlock the hd-dvd/blu-ray plugin.



Ok that's pretty neat. I didn't have time to try the big endian versus little endian thing on the .raw file but I'll try it out when I can. I doubt it will work but who knows until I try. :thanks:

Ok, I purchased the Nero HD DVD/Blu Ray Plugin and installed it. And under settings in the eac3togui I click on settings and point to where eac3to.exe is located and get the following message in the command prompt screen:

C:\Program Files\EAc3 Stuff>"C: Program Files\Eac3 Stuff\eac3to.exe" I:\Evodemux Area\Perfect Storm1.EVO" I:\Evodemux Area\Perfect Storm1.ac3-640-nero-24

Muxing evo's first audio track to raw. Please wait...
Getting "Dump" Instance failed

C:\Program Files\EAc3 Stuff>

So, I don't know what to do now. It appears that the program started to do something, but immediately quit-I don't know if I have the "dump" filter installed-I don't even know what it is, let alone how to install it. Again, any help would be greatly appreciated. Thank you.

[TheSof]

Ok, I purchased the Nero HD DVD/Blu Ray Plugin and installed it. And under settings in the eac3togui I click on settings and point to where eac3to.exe is located and get the following message in the command prompt screen:

C:\Program Files\EAc3 Stuff>"C: Program Files\Eac3 Stuff\eac3to.exe" I:\Evodemux Area\Perfect Storm1.EVO" I:\Evodemux Area\Perfect Storm1.ac3-640-nero-24

Muxing evo's first audio track to raw. Please wait...
Getting "Dump" Instance failed

C:\Program Files\EAc3 Stuff>

So, I don't know what to do now. It appears that the program started to do something, but immediately quit-I don't know if I have the "dump" filter installed-I don't even know what it is, let alone how to install it. Again, any help would be greatly appreciated. Thank you.

I think you need to download graphedit, and then register the dump.ax.

[tvjunky]

Maybe there is someone out there who can help me with that:

I have done some backups of my hd-dvds last week and found the following problem.

I converted the ddp-stream to ac3 (or wav) with eac3to. After that i converted the resulting ac3 or wav to aac 5.1 with besweet. All files are perfect as long as the movie is shorter than 124 minutes. If the length of the movie is longer than 124 minutes, besweet converts always only the first 124 minutes and stopped without any error. so i think this could be a problem of eac3to. so far i tried "van helsing", "troja", "riddick", "underworld" and "king kong" (all european versions).

Is there anyone having the same problem? I have read the whole thread here but didn't found any solution, so any help will be greatly appreciated!

p.s.: sorry for my bad german school english ;-)

[maya]

I think you need to download graphedit, and then register the dump.ax.

Thank you very much, that was the final piece of the puzzle it appears. It produced the AC3 audio file. Now, my next quesion, and hopefully my last, how do I put that AC3 audio file back into my Perfect STorm .EVO file?

[madshi]

I converted the ddp-stream to ac3 (or wav) with eac3to. After that i converted the resulting ac3 or wav to aac 5.1 with besweet. All files are perfect as long as the movie is shorter than 124 minutes. If the length of the movie is longer than 124 minutes, besweet converts always only the first 124 minutes and stopped without any error. so i think this could be a problem of eac3to.
Does the AC3 file you get from eac3to play with the correct length? If so, it's very unprobable that eac3to is at guilt here. Maybe the aac 5.1 encoder generally doesn't like tracks longer than 124 minutes? I've no idea. Btw, what you're doing is not really optimal. You're decoding E-AC3 which is fine. Then you're reencoding to a lossy codec (AC3). Which is fine if AC3 is your final aim. But if you want to end up with another format you should skip the AC3 step to keep audio quality as high as possible. You should probably ask eac3to to give you a wav file and then convert that wav file to aac by using BeSweet.

[madshi]

Thank you very much, that was the final piece of the puzzle it appears. It produced the AC3 audio file. Now, my next quesion, and hopefully my last, how do I put that AC3 audio file back into my Perfect STorm .EVO file?
Currently I don't know any way to put an external track into an EVO file. What you could do is to mux the EVO + AC3 into either MKV or TS. Muxing into MKV is rather easy, as long as the movie is not H264. Muxing into TS is a bit more complicated but also possible from what I've heard. Personally, I'm currently keeping all audio tracks in separate files. MPC can play these just fine together with the movie EVO file.

[maya]

Currently I don't know any way to put an external track into an EVO file. What you could do is to mux the EVO + AC3 into either MKV or TS. Muxing into MKV is rather easy, as long as the movie is not H264. Muxing into TS is a bit more complicated but also possible from what I've heard. Personally, I'm currently keeping all audio tracks in separate files. MPC can play these just fine together with the movie EVO file.


How do you do it and with what program? Also, could I use TSremux to put the video file into .ts or .m2ts and then merge the AC3 audio?

[madshi]

MPC (Media Player Classic) automatically loads the external audio files, if you name them just like the movie file. E.g. if the movie file is named "someMovie.evo", name the audio file "someMovie.ac3". You can also add something to the name, e.g. "someMovie - English.ac3". MPC will automatically load such audio files and play them.

[maya]

MPC (Media Player Classic) automatically loads the external audio files, if you name them just like the movie file. E.g. if the movie file is named "someMovie.evo", name the audio file "someMovie.ac3". You can also add something to the name, e.g. "someMovie - English.ac3". MPC will automatically load such audio files and play them.

What I want to do is play these on my Tvix 4100 media player-I doubt that it would recognize the audio and video (it plays .ts, .m2ts, .mkv, mpeg2, x264 containers). It will not play .evo files or DD+ or Truehd. I can use TSremux to convert the video, but somehow I need to get that audio to play along with the video. Any ideas?

[madshi]

What I want to do is play these on my Tvix 4100 media player-I doubt that it would recognize the audio and video (it plays .ts, .m2ts, .mkv, mpeg2, x264 containers). It will not play .evo files or DD+ or Truehd. I can use TSremux to convert the video, but somehow I need to get that audio to play along with the video. Any ideas?
The Dvico people are just now working on adding MKV support. So it might be the easiest way for you to mux your movie + AC3 into MKV. But as I said, remuxing h264 is a bit problematic. As long as your movie is VC-1 there's no problem.

[maya]

The Dvico people are just now working on adding MKV support. So it might be the easiest way for you to mux your movie + AC3 into MKV. But as I said, remuxing h264 is a bit problematic. As long as your movie is VC-1 there's no problem.

All these files will be VC-1 since they are my HD DVD rips. But what tools do I need to get this to work, and how do I use them if not self-explanatory? I have no experience with command line prompts by the way, but if I knew what the prompts were supposed to be then I could certainly input them. Thanks in advance.

[ACrowley]

Bad news about Nero 8

Nero 8 Filters/Decoder wont work anymore outside Showtime!

For a few weeks Madshi had Contact with a Nero Programer. The Guy told him that newer Nero Versions will not allow using any Decoder/Filter Outside Showtime!
Obviously ist Fact now with Nero 8

My HDDVD/BR Plugin works correct in Nero 8 Showtime. TrueHD,DD+ Plaback works. But not outside Showtime/Graphedit(recode.ex) or eac3to anymore .What a Crap!

So when when you want decode your Eac3/TrueHD Tracks ,dont install Nero 8 and stay with Nero7 until we have a Solution

[maya]

Bad news about Nero

Nero Filters/Decoder wont work anymore outside Showtime!

For a few weeks Madshi had Contact with a Nero Programer. The Guy told him that newer Nero Versions will not allow using any Decoder/Filter Outside Showtime!
Obviously ist Fact now with Nero 8

My HDDVD/BR Plugin works correct in Showtime. TrueHD,DD+ Plaback works. But not outside Showtime/Graphedit(recode.ex) or eac3to anymore .What a Crap!

So when when you want decode your Eac3/TrueHD Tracks ,dont install Nero 8 and stay with Nero7 until we have a Solution

Thanks for letting me know, I just updated my Nero 7 Ulra edition, but I will make sure I stay away from Nero 8.

[madshi]

All these files will be VC-1 since they are my HD DVD rips. But what tools do I need to get this to work, and how do I use them if not self-explanatory? I have no experience with command line prompts by the way, but if I knew what the prompts were supposed to be then I could certainly input them. Thanks in advance.
Install the Haali Media Splitter package. The tool "gdsmux" (shipping with that package) can directly remux EVO into MKV. Once you have your MKV, you can use the tool "mkvtoolnix" to add audio tracks to the MKV. Maybe you can already do that with gdsmux, I've not used that yet. mkvtoolnix has a nice GUI, though. However, muxing the EVO file to MKV should be done with gdsmux and not with mkvtoolnix.

[maya]

Install the Haali Media Splitter package. The tool "gdsmux" (shipping with that package) can directly remux EVO into MKV. Once you have your MKV, you can use the tool "mkvtoolnix" to add audio tracks to the MKV. Maybe you can already do that with gdsmux, I've not used that yet. mkvtoolnix has a nice GUI, though. However, muxing the EVO file to MKV should be done with gdsmux and not with mkvtoolnix.

Thanks, I will give this a shot and report back

[ACrowley]

Install the Haali Media Splitter package. The tool "gdsmux" (shipping with that package) can directly remux EVO into MKV. Once you have your MKV, you can use the tool "mkvtoolnix" to add audio tracks to the MKV. Maybe you can already do that with gdsmux, I've not used that yet. mkvtoolnix has a nice GUI, though. However, muxing the EVO file to MKV should be done with gdsmux and not with mkvtoolnix.

yes, but its important to remux the mkv from gdsmux with mkvmerge and 23.976 Timecode
Because remuxed VC1/Evo output mkv from gdsmux has 29.97 FPS (as like as HaaliMediaSplitter on VC1 what is wrong)!
You must remux VC1/Evo in any case with mkvmerge and 23.976 Timecode
So a VC1/Evo Remux "must" be done with gdsmux and mkvmerge.

[madshi]

yes, but its important to remux the mkv from gdsmux with mkvmerge and 23.976 Timecode
Because remuxed VC1/Evo output mkv from gdsmux has 29.97 FPS (as like as HaaliMediaSplitter on VC1 what is wrong)!
You must remux VC1/Evo in any case with mkvmerge and 23.976 Timecode
So a VC1/Evo Remux "must" be done with gdsmux and mkvmerge.
Yep, that's true, forgot about that one.

[TheSof]

yes, but its important to remux the mkv from gdsmux with mkvmerge and 23.976 Timecode
Because remuxed VC1/Evo output mkv from gdsmux has 29.97 FPS (as like as HaaliMediaSplitter on VC1 what is wrong)!
You must remux VC1/Evo in any case with mkvmerge and 23.976 Timecode
So a VC1/Evo Remux "must" be done with gdsmux and mkvmerge.

That's worked fine for VC1 (reclock reports 23.976) but with AVC I just can't get reclock to get the FPS. There's no stutter so I suppose it matters not.

[madshi]

That's worked fine for VC1 (reclock reports 23.976) but with AVC I just can't get reclock to get the FPS. There's no stutter so I suppose it matters not.
I can't even get AVC properly remuxed into MKV. gdsmux stalls for me. Demuxing and remuxing works but audio sync is totally lost that way for me. (All tested with Equilibrium). So currently I have my h264 HD DVD movies still in EVO files. But I think we're a bit out of topic now...

[TheSof]

I can't even get AVC properly remuxed into MKV. gdsmux stalls for me. Demuxing and remuxing works but audio sync is totally lost that way for me. (All tested with Equilibrium). So currently I have my h264 HD DVD movies still in EVO files. But I think we're a bit out of topic now...

Also sorry for going OT, but I had same problems with Equilibrium, and gave up. A real nightmare disc. However, had no problems with Blue Ray AVC "The Prestige", as long as I used TSRemux to extract the avc1 and then use mkvextract. All fine except no framerate in reclock.

It's a good job most HDDVDs are vc1 (shivers at the thought of Equilibrium).

Back on topic, can someone please confirm that using eac3to is the only way to bypass DRC with the nero filter? And that by using the nero filter in a directshow player (mpc, zoomplayer, theatretek etc.) will use DRC?

[madshi]

Back on topic, can someone please confirm that using eac3to is the only way to bypass DRC with the nero filter? And that by using the nero filter in a directshow player (mpc, zoomplayer, theatretek etc.) will use DRC?
Confirmed. Well, if there's enough interest, I could probably add a tweak to switch off DRC for use in 3rd party media players. This tweak would only work for E-AC3, though. It would *NOT* work for conventional AC3.

[TheSof]

Confirmed. Well, if there's enough interest, I could probably add a tweak to switch off DRC for use in 3rd party media players. This tweak would only work for E-AC3, though. It would *NOT* work for conventional AC3.

I would be very interested in such a tweak.

I'm now going down the analog out route and using nero in TT or MPC would be ideal.

[ACrowley]

I would be very interested in such a tweak.
I'm now going down the analog out route and using nero in TT or MPC would be ideal.

this DRC Tweak is only eac3to internal and not possible for Realtime Playback. You can only Remove Dialnorm from a DD+ File

For DShow Playback of DD+:
-only in Nero Showtime it self you can switch of DRC
-Nero Decoder in a Dshow Player like mpc/WMp/Zooomplayer applies DRC+DialNorm
-SonicAudioDecoder applies DRC+DolbyGain,Sonic System Settings have no Effect. in eac3to and Dshow Players

Conclusion:
-DD+ Playback with a Dshow Player like mpc isnt possible without DRC/Dialnorm
-To decode DD+ to wave/etc eac3to+ Nero is the one and only Way to avoid DRC/Dialonorm
-TrueHD Decoding is still with DialNorm

But ffmpeg EAC3 is a chance for realtime Playback without DRC/Dialnorm. Hopefully a proper ffdshow Build is coming soon

[tvjunky]

Does the AC3 file you get from eac3to play with the correct length? If so, it's very unprobable that eac3to is at guilt here. Maybe the aac 5.1 encoder generally doesn't like tracks longer than 124 minutes? I've no idea. Btw, what you're doing is not really optimal. You're decoding E-AC3 which is fine. Then you're reencoding to a lossy codec (AC3). Which is fine if AC3 is your final aim. But if you want to end up with another format you should skip the AC3 step to keep audio quality as high as possible. You should probably ask eac3to to give you a wav file and then convert that wav file to aac by using BeSweet.

Thanks for your reply, madshi (and for this great tool!)

I have tested it again on underworld yesterday. The wav-file and the ac3-file eac3to created worked fine in all players. But converting these files to aac still doesn't work with besweet. Yesterday i tried it with neroaacenc.exe and neroaacenc_sse2. Always the same result -> audio length is 124 minutes. Still confused about this problem ...

[madshi]

this DRC Tweak is only eac3to internal and not possible for Realtime Playback.
Please reread the part where TheSof quoted me! ;) In short: I could make the DRC tweak work for realtime playback, if there's enough interest.

[madshi]

I have tested it again on underworld yesterday. The wav-file and the ac3-file eac3to created worked fine in all players. But converting these files to aac still doesn't work with besweet. Yesterday i tried it with neroaacenc.exe and neroaacenc_sse2. Always the same result -> audio length is 124 minutes. Still confused about this problem ...
This sounds like a problem with the aac encoder or BeHappy to me. I don't think eac3to is at guilt. So that means there's nothing I can do to help you. Please ask for help in the BeHappy or aac thread.

[tebasuna51]

This sounds like a problem with the aac decoder or BeHappy to me. I don't think eac3to is at guilt. So that means there's nothing I can do to help you. Please ask for help in the BeHappy or aac thread.

I think tvjunky use BeSweet, not BeHappy.

With BeHappy you can open the ac3 with NicAc3Source() or the big_wav with RaWavSource() (http://forum.doom9.org/showthread.php?p=1037550#post1037550). And after encode with NeroAacEnc.

If you have the big_wav (>4GB) don't need BeHappy because can use a direct command line.

But, always remember use the parameter -ignorelength with NeroAacEnc (or -readtoeof 1 with Aften) when the wav file is > 4GB.

[madshi]

I think tvjunky use BeSweet, not BeHappy.
Ah yes, sorry.

[tvjunky]

THANKS, tebasuna51! "-ignorelength" was the missing link!!

[Nikos]

In short: I could make the DRC tweak work for realtime playback, if there's enough interest.

Yes, i want realtime playback without DRC and Dial. norm.

[ACrowley]

Please reread the part where TheSof quoted me! ;) In short: I could make the DRC tweak work for realtime playback, if there's enough interest.

Ah, sry....nice!

[VempX]

how to encode .pcm file to flac?
I use TsRemux demux a audio track from m2ts.I name it test.pcm.but when I convert the .pcm file to flac by eac3to I get a error flac.
I demux the pcm by dsmux,then I get a correct pcm.I could play it by foobar2000.But eac3to can't convert it.
who can help me?

[madshi]

how to encode .pcm file to flac?
I use TsRemux demux a audio track from m2ts.I name it test.pcm.but when I convert the .pcm file to flac by eac3to I get a error flac.
What do you mean with "error flac"? I need more details. Please give me the full eac3to + flac output text and a description of what is wrong. TsRemux + eac3to should work fine.

[VempX]

I uploaded the .pcm file and encoded .flac file.

you can download it in here

http://www.yousendit.com/transfer.php?action=download&ufid=470C8DE25BDEDC4E

this is log
=================
E:\>"F:\eac3to\eac3to.exe" "E:\test.pcm" "E:\test.flac" -24 -big
RAW, 5.1 channels, 0:00:06, 24 bits, 48khz
Remapping channels. Please wait...
Converting the raw file to flac. Please wait...

Done.
=============================

[yonta]

F:\>eac3to\eac3to.exe Innocence.LPCM16.8ch.pcm innocence.eac3to.ac3 -8 -16 -down6 -640
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.0.8
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: RAW Signed 16-bit little-endian 48000 Hz stereo
output format: 48000 Hz stereo (2/0)

SIMD usage: MMX SSE SSE2 SSE3
Threads: 2

progress: 100% | q: 306.6 | bw: 60.0 | bitrate: 640.0 kbps

Done.

innocence.eac3to.ac3 is just noise and 4 times bigger than the correct 5.1ch file.


tried converting the eac3to-generated wav file to ac3 using aften.exe.
F:\>eac3to\aften\aften.exe Innocence.LPCM16.8ch.wav innocence.aften.ac3 -b 640

Aften: A/52 audio encoder
Version 0.0.8
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: WAVE Signed 16-bit little-endian 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

SIMD usage: MMX SSE SSE2 SSE3
Threads: 2

progress: 100% | q: 255.5 | bw: 57.9 | bitrate: 640.0 kbps

innocence.aften.ac3 seems OK

converting to the other formats than ac3 works correctly.

[madshi]

I uploaded the .pcm file and encoded .flac file.
That's one strange PCM file. Where is it from? From a Blu-Ray? Can you please demux it with xport and post here what xport reports over the LPCM attributes? It would be helpful if you could upload a sample of the m2ts file. From a quick check it could a 32bit mono PCM file. Never seen such a Blu-Ray PCM track yet!!

[madshi]

F:\>eac3to\eac3to.exe Innocence.LPCM16.8ch.pcm innocence.eac3to.ac3 -8 -16 -down6 -640
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.0.8
Could you please try the Aften build that is shipping with eac3to? Does that work? Also, which eac3to version are you using?

input format: RAW Signed 16-bit little-endian 48000 Hz stereo
Yeah, the "stereo" is the problem here.

Can you please rerun the same process and then press "Ctrl-C" while Aften is running? That should leave you with the wav file that eac3to was feeding to Aften. Can you please check the properties of the wav file? Is it properly flagged as a 6 channel wav file? Or is it incorrectly flagged as stereo?

[VempX]

er........I demux the pcm from a m2ts file by TsRemux.
This is the mpa(5.1 LPCM) which demuxed by xport.
http://www.yousendit.com/transfer.php?action=download&ufid=5E1BDD3C0DAFE50F

[yonta]

Could you please try the Aften build that is shipping with eac3to? Does that work? Also, which eac3to version are you using?

I'm using v1.14 and using the aften.exe that comes with eac3to:
F:\>eac3to\eac3to.exe f:\Innocence.LPCM16.8ch.pcm e:\innocence.eac3to.aften.from.eac3to.ac3 -16 -down6 -640 -8
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.


Can you please rerun the same process and then press "Ctrl-C" while Aften is running? That should leave you with the wav file that eac3to was feeding to Aften. Can you please check the properties of the wav file? Is it properly flagged as a 6 channel wav file? Or is it incorrectly flagged as stereo?

mediainfo says;
Audio #0
Codec : PCM
Codec/Family : PCM
Codec/Info : Microsoft PCM
Bit rate : 4608 Kbps
Channel(s) : 6 channels
Sampling rate : 48 KHz
Resolution : 16 bits

This wav file plays OK.
I tried converting this wav file to ac3 directly using aften.exe and it worked.(second command line in the first post)
That's why this problem is weird.

[Rectal Prolapse]

In short: I could make the DRC tweak work for realtime playback, if there's enough interest.

madshi, that would be wonderful!

[Creator1]

Bad news about Nero 8

Nero 8 Filters/Decoder wont work anymore outside Showtime!

For a few weeks Madshi had Contact with a Nero Programer. The Guy told him that newer Nero Versions will not allow using any Decoder/Filter Outside Showtime!
Obviously ist Fact now with Nero 8

My HDDVD/BR Plugin works correct in Nero 8 Showtime. TrueHD,DD+ Plaback works. But not outside Showtime/Graphedit(recode.ex) or eac3to anymore .What a Crap!

So when when you want decode your Eac3/TrueHD Tracks ,dont install Nero 8 and stay with Nero7 until we have a Solution

Thanks for the tip!

A trick for people who really want Nero 8... you can install Nero 7 in a virtual PC image and upgrade to Nero 8 on your real Windows :)

[madshi]

er........I demux the pcm from a m2ts file by TsRemux.
This is the mpa(5.1 LPCM) which demuxed by xport.
http://www.yousendit.com/transfer.php?action=download&ufid=5E1BDD3C0DAFE50F
Sorry, I wasn't clear. When you run "xport" it will tell you how many channels and which bitdepth etc the LPCM file has. Can you please post here what xport is posting to the screen?

[madshi]

invalid RIFF id in wav header
invalid wav file: -b
"aften" reported error code "1". A valid *.wav file was created sucessfully, though.
Can you please redownload eac3to just to make sure that the files are not corrupt? This sounds extremely strange to me. I mean Aften doesn't complain when you start it manually - but it does complain when being started by eac3to? That's weird.

Also, would it be possible for me to get a small sample of that problematic source file? Please before sending the sample check if the problem can be reproduced with the sample. Thanks!

[madshi]

I've a goody for you guys. Check out the last post of the delaycut thread:

http://forum.doom9.org/showthread.php?t=71545

I've updated delaycut to v1.3.0.0 with full E-AC3 support!!!

[yonta]

Can you please redownload eac3to just to make sure that the files are not corrupt? This sounds extremely strange to me. I mean Aften doesn't complain when you start it manually - but it does complain when being started by eac3to? That's weird.

Also, would it be possible for me to get a small sample of that problematic source file? Please before sending the sample check if the problem can be reproduced with the sample. Thanks!

redownloaded and binary-compared and the files are bit-identical.
I tested several times and this problem is reproduced everytime.
sample here (http://www.sendspace.com/file/pnfj69).

[VempX]

Sorry, I wasn't clear. When you run "xport" it will tell you how many channels and which bitdepth etc the LPCM file has. Can you please post here what xport is posting to the screen?

This is report of xport
=================================
D:\[DVDISO] AIR BDBOX VOL.1\BDMV\STREAM>xport -h 00003.m2ts 1 3 4
xport Transport Stream Demuxer 0.98
program = 1, video channel = 3, audio channel = 4
Program Number = 0 (0x0000), Program Map PID = 31 (0x001f)
Program Number = 1 (0x0001), Program Map PID = 256 (0x0100)
program descriptor = 0x05, 0x04, 0x48, 0x44, 0x4d, 0x56
program descriptor = 0x88, 0x04, 0x0f, 0xff, 0xff, 0xfe
ES descriptor for stream type 0x02 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x02, 0x44, 0x3f
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x08, 0x3c, 0x0e, 0x00
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x80, 0x31, 0x7f
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x08, 0x3e, 0x04, 0x00
Audio PID = 4355 <0x1103>, type = 0x80
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x80, 0x61, 0x7f
LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000
ts rate = unspecified
============================

[ACrowley]

I've a goody for you guys. Check out the last post of the delaycut thread:

http://forum.doom9.org/showthread.php?t=71545

I've updated delaycut to v1.3.0.0 with full E-AC3 support!!!

WOW THX..your my man :)

1. I can only load a EAC3 via drag&drop. The Open Browser recognize no ec3/eac3/ddp extension
2. i run 2 unclean EAC3 Track trough delaycut with "fix". One fixed EAC3 wirks perfect after fixing.
But the other fixed EAC3 cant be used aynmore, Nero and Sonic Decoder aborts Plaback

[madshi]

But the other fixed EAC3 cant be used aynmore, Nero and Sonic Decoder aborts Plaback
Try the different fix methods (silence, skip, ...). Do they all fail? How does the delaycut log look like? But this problem might be better handles in the delaycut thread?

[ACrowley]

Try the different fix methods (silence, skip, ...). Do they all fail? How does the delaycut log look like? But this problem might be better handles in the delaycut thread?

yes...will going into delaycut thread for it

As i say via PN.

After fix the problematic eac3 works fine in eac3to ,but Playback with Nero or SonicDecoder not..
I will test another fix Method
But can skip/silence cause async Problems when a few frames are filled with silence for exmaple ?

[madshi]

After fix the problematic eac3 works fine in eac3to ,but Playback with Nero or SonicDecoder not..
I will test another fix Method
But can skip/silence cause async Problems when a few frames are filled with silence for exmaple ?
"fix" is bad IMHO. "skip" has the biggest danger of asnyc problems. "silence" should be similar to "fix" in terms of audio sync. Even "fix" can result in audio sync problems.

[ACrowley]

"fix" is bad IMHO. "skip" has the biggest danger of asnyc problems. "silence" should be similar to "fix" in terms of audio sync. Even "fix" can result in audio sync problems.

Ok THX

The problematic EAC3 5.1 640Kbps is from Troy HDDVD.
It works with "Silence" :)

The Reencoded ac3 from "fix" Output has a loud cracking Noise, while the rencode from "silence" Output is clean.
I will better use "Silence" from now on for eac3 and ac3

[Jedi_Vader20]

Hiya guys.

I recently imported the HDDVD for Nine Inch Nails' Beside You In Time.

I ripped the audio from the disc, intending to convert it for use on my PSP. However, when I use eac3to to convert the EVO to a WAV, I get a file that looks the right size, but is only 9m35s long. Converting to FLAC/AC3/DTS gets me the right file size.

Am I doing something wrong?

[madshi]

Hiya guys.

I recently imported the HDDVD for Nine Inch Nails' Beside You In Time.

I ripped the audio from the disc, intending to convert it for use on my PSP. However, when I use eac3to to convert the EVO to a WAV, I get a file that looks the right size, but is only 9m35s long. Converting to FLAC/AC3/DTS gets me the right file size.

Am I doing something wrong?
WAV files cannot really handle files > 4GB. Some programs even stumble with WAV files over 2GB. Other programs can be configured to try to work around the WAV problem and they work alright with such big files.

[ACrowley]

Hiya guys.

I recently imported the HDDVD for Nine Inch Nails' Beside You In Time.

I ripped the audio from the disc, intending to convert it for use on my PSP. However, when I use eac3to to convert the EVO to a WAV, I get a file that looks the right size, but is only 9m35s long. Converting to FLAC/AC3/DTS gets me the right file size.

Am I doing something wrong?

On huge interleaved waves (+4gb )the Runtime is often wrong.
But the mono waves and encoded Output from it is fine, no Problem

Do you need AC3 for your PSPS ?

[tebasuna51]

The Reencoded ac3 from "fix" Output has a loud cracking Noise, while the rencode from "silence" Output is clean.
I will better use "Silence" from now on for eac3 and ac3

The "Silence" method is always the most sure (works always, maintain the sync) but can also have clicks when play.

The "Fix" method work fixing the CRC and leaving errors in data, most the times work better than "Silence" with inaudible effects, if errors are not important, but sometimes the errors are important and can crash the decoders or produce big cracking noise like you say.

[madshi]

The "Silence" method is always the most sure (works always, maintain the sync)
I think there are different kinds of corruption. The usual kind is probably that the length of the audio file is correct but that some of the data is corrupt. In that case "Silence" should maintain the sync. However, if the source container is corrupt (or if the demuxing routine has a bug) there could be additional invalid data in the source file. As a result the final repaired audio file could be too long. In that case "Skip" might maintain sync while "Silence" could lose sync.

[Jedi_Vader20]

On huge interleaved waves (+4gb )the Runtime is often wrong.
But the mono waves and encoded Output from it is fine, no Problem

Do you need AC3 for your PSPS ?

No, wanted to downmix it to two channel and encode to AAC.

When I go to try and convert, all programs only handle the first 9 minutes and 35 seconds.

How do I split into six mono wavs?

[honai]

Does DTS-core extraction from DTS-HD implicitly remove Dial.Norm. from the stream?

[madshi]

No, wanted to downmix it to two channel and encode to AAC.

When I go to try and convert, all programs only handle the first 9 minutes and 35 seconds.
Check the previous posts from tebasuna51. He explained how to call the AAC decoder to workaround this problem. There's a special parameter to make big WAV files work.

[madshi]

Does DTS-core extraction from DTS-HD implicitly remove Dial.Norm. from the stream?
Yes...

[Schotenhüter]

Getting "Dump" Instance failed

whats that:stupid:

[madshi]

This is report of xport
=================================
D:\[DVDISO] AIR BDBOX VOL.1\BDMV\STREAM>xport -h 00003.m2ts 1 3 4
xport Transport Stream Demuxer 0.98
program = 1, video channel = 3, audio channel = 4
Program Number = 0 (0x0000), Program Map PID = 31 (0x001f)
Program Number = 1 (0x0001), Program Map PID = 256 (0x0100)
program descriptor = 0x05, 0x04, 0x48, 0x44, 0x4d, 0x56
program descriptor = 0x88, 0x04, 0x0f, 0xff, 0xff, 0xfe
ES descriptor for stream type 0x02 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x02, 0x44, 0x3f
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x08, 0x3c, 0x0e, 0x00
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x80, 0x31, 0x7f
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x08, 0x3e, 0x04, 0x00
Audio PID = 4355 <0x1103>, type = 0x80
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0
x80, 0x61, 0x7f
LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000
ts rate = unspecified
============================
With the xport demuxed audio file everything seems to work for me. Just use "eac3to xportdemuxed.pcm xportdemuxed.flac". I think the audio track you demuxed with TsRemux was corrupt.

[madshi]

F:\>eac3to\eac3to.exe Innocence.LPCM16.8ch.pcm innocence.eac3to.ac3 -8 -16 -down6 -640
RAW, 7.1 channels, 0:01:00, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.0.8
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: RAW Signed 16-bit little-endian 48000 Hz stereo
output format: 48000 Hz stereo (2/0)

SIMD usage: MMX SSE SSE2 SSE3
Threads: 2

progress: 100% | q: 306.6 | bw: 60.0 | bitrate: 640.0 kbps

Done.

innocence.eac3to.ac3 is just noise and 4 times bigger than the correct 5.1ch file.


tried converting the eac3to-generated wav file to ac3 using aften.exe.
F:\>eac3to\aften\aften.exe Innocence.LPCM16.8ch.wav innocence.aften.ac3 -b 640

Aften: A/52 audio encoder
Version 0.0.8
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: WAVE Signed 16-bit little-endian 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

SIMD usage: MMX SSE SSE2 SSE3
Threads: 2

progress: 100% | q: 255.5 | bw: 57.9 | bitrate: 640.0 kbps

innocence.aften.ac3 seems OK

converting to the other formats than ac3 works correctly.
That was a bug in eac3to. Thanks for reporting it. I'll upload a new build in a moment. Btw, "-down6" and "-640" are default when converting to ac3.

[madshi]

whats that:stupid:
You need to download GraphEdit. The file "dump.ax" is shipping with GraphEdit. Then run "regsvr32 dump.ax" in the command line to register the "dump" filter.

[madshi]

eac3to v1.15 released

http://madshi.net/eac3to.zip

This time there are only a few bugfixes.

[tebasuna51]

I think there are different kinds of corruption. The usual kind is probably that the length of the audio file is correct but that some of the data is corrupt. In that case "Silence" should maintain the sync. However, if the source container is corrupt (or if the demuxing routine has a bug) there could be additional invalid data in the source file. As a result the final repaired audio file could be too long. In that case "Skip" might maintain sync while "Silence" could lose sync.

Well, you are the author of last version I can't discuss with you :rolleyes: (I'm busy with ffmpeg to analyze your version)

But with previous DelayCut versions, see the post (http://forum.doom9.org/showthread.php?p=810264#post810264) all short frames are skipped and only full erroneus frames are replaced by silence then never the output can be long than input. But with 'Skip' option always the output is short than input because short and full frames are ignored.

[madshi]

Well, you are the author of last version I can't discuss with you :rolleyes:
Heh! No, actually I think you know more about delaycut than I do. I just copied every AC3 function and changed it to E-AC3 without really diving too deep into how it all really works. I definitely didn't change anything about how "skip", "silence" and "fix" work.

I'm busy with ffmpeg to analyze your version
That's good!

But with previous DelayCut versions, see the post (http://forum.doom9.org/showthread.php?p=810264#post810264) all short frames are skipped and only full erroneus frames are replaced by silence then never the output can be long than input.
The output can't be longer than the input. But it can be longer than the correct result would be (and that is what I meant). Let me give you an example:

Blu-Ray TrueHD tracks have AC3 frames + TrueHD blocks. delaycut in "Silence" mode will extract all correct frames just fine. But it might find the AC3 sync word in some TrueHD blocks. The CRC will not fit, but "silence" mode will then add silent frames instead. As a result you'll get an audio track which is too long. Agreed? So there are situations where "skip" is the better choice compared to "silence".

[tebasuna51]

Blu-Ray TrueHD tracks have AC3 frames + TrueHD blocks.

Then you are right one more time. :)

In standard ac3 streams only ac3 frames (goods or corrupted) can be present.

[Schotenhüter]

You need to download GraphEdit. The file "dump.ax" is shipping with GraphEdit. Then run "regsvr32 dump.ax" in the command line to register the "dump" filter.

i`am test it:thanks:

[madshi]

In standard ac3 streams only ac3 frames (goods or corrupted) can be present.
One thing I'm not sure about: Can a recorded TS stream be so corrupt that a video packet is mistakingly interpreted as an audio packet? If that's possible and if you then demux the AC3 stream, it might also contain too much data, I think.

[tebasuna51]

One thing I'm not sure about: Can a recorded TS stream be so corrupt that a video packet is mistakingly interpreted as an audio packet?

Well, is possible but improbable, not only the header must match but also the CRC frame.

[madshi]

Well, is possible but improbable, not only the header must match but also the CRC frame.
Ah, ok. Don't know much about TS yet. One of the things I still need to look into... :)

[Schotenhüter]

next prob :o

the audio decoder didn`t accept the input stream

:confused:

[madshi]

next prob :o
I don't have a telepathical connection to your computer. So if you don't give me any more details I can not help you. I need the full eac3to output and information about the source audio file you fed to eac3to.

[Icemaan]

Hallo

By me your great Tool works fine. But only with 1536kps Dts.When I want downsample to 768kps it not work.The Sound looks horrible with 1536 all ok
Any Idea because I am new may be I made somethinh not correct
Thanks for help

Icemaan

[VempX]

Sorry,I have to go to school Orz
I demux the audio track from m2ts by tsremux then I get a .pcm file.
I try to use eac3to to convert it to flac,but I get a strange .flac file which I gave you~
Does eac3to can't convert .wav file?
I demux the audio by gdsmux so I get a .wav file.But eac3to can't convert it =A=

[VempX]

Sorry,I have to go to school Orz
I demux the audio track from m2ts by tsremux then I get a .pcm file.
I try to use eac3to to convert it to flac,but I get a strange .flac file which I gave you~
Does eac3to can't convert .wav file?
I demux the audio by gdsmux so I get a .wav file.But eac3to can't convert it =A=

[VempX]

sorry
net hava a wrong
please delete this post

[markrb]

Strange problem and sorry if it's been discussed before. I did try and find it.

After converting an EVO to ac3 the file is small (165mb) and plays way too fast. I checked the wav file and it is 1.74gb, but it also plays way too fast.

There is no indication of anything wrong. The screen looks exactly like the successful screen shots I have seen here.
It even says 100% done at the end when aften is finished.

I did check the EVO with the single dd+ stream and that plays fine. I have tried it both command line and using the Gui with the same results.

2 Discs with the same issue. So I don't think it's the discs.

EDIT: I think my Nero was hosed. I reinstalled and I am testing it now.

Yep that was it. Un-install Nero. Use clean tool. Re-install. Seems to work fine.

Mark

[nautilus7]

You need to download GraphEdit. The file "dump.ax" is shipping with GraphEdit. Then run "regsvr32 dump.ax" in the command line to register the "dump" filter.

Where can i download graphedit?

[markrb]

Where can i download graphedit?

Try Google. Then look for the register.bat file and run that.

Mark

[Icemaan]

Hi all together

I have following Problem
When i merge the Video and Audio with MkvMerge i get this Message

audio/video go out of sync, but this stream is damaged.
Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged

This only come bei Dts not by Ac3. Can anywhere help me.The Video File is 264 the Audio dts

Thanks a lot
Icemaan

[TheSof]

If I have just the TrueHD extracted via evodemux (.mpa), is there anyway I can get eac3to to accept it?

[honai]

@madshi

I've found a parsing problem with Pan's Labyrinth HD-DVD (EU release). For the English DTS-HD track EVOdemux reports:


DTS HD (DTS) audio stream 1 found!
First PTS = 0000D6D8
Substream id = 89
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 1023
Channel arrangement = LT + RT (left and right total)
Sampling frequency = 48 kHz
Transmission bitrate = 768 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 1023
Channel arrangement = LT + RT (left and right total)
Sampling frequency = 48 kHz
Transmission bitrate = 768 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits


It's actually DTS stereo (2.0). eac3to complains:

The source file format is unknown.

[Roscoe62]

I've followed this thread as best as I can, but I still haven't figured out whether it will do this or not. I have a couple of questions and any help is appreciated!

1. Is it possible to demux a Dolby TruHD audio track from a BR disc? From what I've seen earlier in the thread it might be possible with xport...but where can I find xport?

2. Will EAC3to convert a Dolby TruHD audio track demuxed from a BR Disc into a multi-channel wav file? (or flac). If so, how?

3. If all the above IS possible how can I mux the newly created 5.1 wav file into a .ts container with the AVC video, while still staying in sync? If .ts isn't the way to go, could it be done with mkv? (I thought mkv had problems with AVC video)

I know there are a lot of questions here, but even getting answers for the first 1 or 2 would be great.

Spiderman 1 & 2 are being released very soon and they both have TruHD soundtrack which I'd like to backup without losing any quality. This is all pretty standard from HD-DVD, but I've not done anything like this with BR.

Again, any help is appreciated!

[madshi]

By me your great Tool works fine. But only with 1536kps Dts.When I want downsample to 768kps it not work.The Sound looks horrible with 1536 all ok
Any Idea because I am new may be I made somethinh not correct
Will need to check if I can reproduce the problem. Can you please try to use "eac3to source.whatever dest.wavs"? This will give you 6 wav files. Then please convert these 6 wavs to dts with Surcode manually. Is the resulting file ok?

[madshi]

Sorry,I have to go to school Orz
I demux the audio track from m2ts by tsremux then I get a .pcm file.
I try to use eac3to to convert it to flac,but I get a strange .flac file which I gave you~
Yeah, but you also gave me the PCM track demuxed by xport - and that works fine! So it seems that the PCM track taken from tsremux is corrupt. Use xport and it should work fine.

[madshi]

I have following Problem
When i merge the Video and Audio with MkvMerge i get this Message

audio/video go out of sync, but this stream is damaged.
Warning: dts_packetizer: skipping 3 bytes (no valid DTS header found). This might make audio/video go out of sync, but this stream is damaged

This only come bei Dts not by Ac3. Can anywhere help me.The Video File is 264 the Audio dts
Where is that dts file coming from? Was it created by eac3to?

[madshi]

If I have just the TrueHD extracted via evodemux (.mpa), is there anyway I can get eac3to to accept it?
Please read the first post of this thread. Short answer: No.

[madshi]

I've found a parsing problem with Pan's Labyrinth HD-DVD (EU release). For the English DTS-HD track EVOdemux reports:

It's actually DTS stereo (2.0). eac3to complains:
Can you please rebuild the DTS-HD track into a separate EVO file (with only that DTS-HD track in and nothing else) and then send me the first 5MB of that rebuilt EVO? Thanks!

[madshi]

1. Is it possible to demux a Dolby TruHD audio track from a BR disc? From what I've seen earlier in the thread it might be possible with xport...but where can I find xport?
Yes, it's possible with xport. You can download it here:

http://www.w6rz.net/

TsRemux should also be able to demux the TrueHD track.

2. Will EAC3to convert a Dolby TruHD audio track demuxed from a BR Disc into a multi-channel wav file? (or flac). If so, how?
Blu-Ray TrueHD tracks consist of an AC3 core with some additional TrueHD packets on top of that. eac3to can extract the AC3 core. Do "eac3to source.thd dest.ac3". This will give you the AC3 core of that TrueHD track. Obviously you'll only get 640kbps AC3 quality this way and not the full TrueHD quality. Currently to my best knowledge only PowerDVD can decode Blu-Ray TrueHD tracks and PowerDVD's Audio Decoder cannot be used outside of PowerDVD.

Spiderman 1 & 2 are being released very soon and they both have TruHD soundtrack which I'd like to backup without losing any quality. This is all pretty standard from HD-DVD, but I've not done anything like this with BR.
I'd suggest that you use TsRemux to rebuild the TrueHD tracks into a separate m2ts file. I think this way you should be able to back them up without wasting much space and with good hopes that we might find a way later to fully decode them. Of course you could also backup the whole movie m2ts. That would be the easiest way, but consume more space. Ideally, why backing them up right now? You should buy the Blu-Ray discs. So you have the data on your shelf, anyway... :)

[Roscoe62]

Ideally, why backing them up right now? You should buy the Blu-Ray discs. So you have the data on your shelf, anyway... :)

Thanks for the info Madshi. I will be placing my order for the discs tomorrow but I will not use PowerDVD for playback. I only playback via .ts or .mkv files so I can use my fav player - Theatertek. So I guess I will stick with the AC3 core for the time being, until someone smarter than me can figure out how to fully decode Blu-Ray TruHD tracks.

Here's hoping anyway! :)

[TheSof]

I only playback via .ts or .mkv files so I can use my fav player - Theatertek.

TT will happily play m2ts files.

[Icemaan]

Where is that dts file coming from? Was it created by eac3to?

The Dts File was created with eac3to. The File comes from the rebuild evo with only one audio stream inside

I have split the Evo with Evosplit may be this is the problem what you think

icemaan

[madshi]

The Dts File was created with eac3to. The File comes from the rebuild evo with only one audio stream inside

I have split the Evo with Evosplit may be this is the problem what you think

icemaan
I'm not sure. Does that warning come very often or just once? Is there a timecode for that warning? Could it be the timecode where you split the evo?

[Roscoe62]

TT will happily play m2ts files.

Perhaps,

but because of the TruHD soundtrack Theatertek won't play it unless I demux the BR M2TS, re-code the audio and then re-mux into a new container. And if I'm going to all that effort I'd rather re-mux to a container I'm familiar with.

[ACrowley]

Perhaps,

but because of the TruHD soundtrack Theatertek won't play it unless I demux the BR M2TS, re-code the audio and then re-mux into a new container. And if I'm going to all that effort I'd rather re-mux to a container I'm familiar with.


recode the Audio !? Just extract the DD core, no need to renencode it. Also maybe TT decodes only the DDcore from, Bluray TrueHD ?! i

[pimpMyHD]

I am interesting in using your software, but i have a question :

do i need nero hddvd/bluray plugin, if i already have sonic scenarist filters ?

I currently do not own nero 's plugin, but i own the sonic hd filters.

[TheSof]

I am interesting in using your software, but i have a question :

do i need nero hddvd/bluray plugin, if i already have sonic scenarist filters ?

I currently do not own nero 's plugin, but i own the sonic hd filters.

You need them if you are after the best sound quality. The sonic filters add nasty DRC.

[pimpMyHD]

You need them if you are after the best sound quality. The sonic filters add nasty DRC.
So, i should be able to use eac3to without the nero filters ?

Because it does not work on my system.

[TheSof]

So, i should be able to use eac3to without the nero filters ?

Because it does not work on my system.

If you just have the sonic decoders yes it should still work. What error message are you getting?

[Thunderbolt8]

I have a disc with 5.1 DD+ audio codec, what should I do now when I want to create a remux and this should include the best possible audio I can make out of it, which is currently possible to playback with a normal setup? I also guess that keeping the track as it is is also not a good solution (apart from the question if it was playable at all then?), because theres lot of additional junk in it like with dts-hs & dtscore ?
I think the solution would be something like converting to normal ac3 or DTS. but im not sure if dts conversion is possible then from a DD+ source, or if it does make sense at all?
another question I have is the removal of dialog normalisation, is such dialog normalisation included in every audio track? is the dialog normalisation somehow intended by the film makers, otherwise one wouldnt understand what people say in certain scenes? does it sound more natural & better without dialog normalisation?

[madshi]

I have a disc with 5.1 DD+ audio codec, what should I do now when I want to create a remux and this should include the best possible audio I can make out of it, which is currently possible to playback with a normal setup?
Define "normal setup".

If you have analog connection to your receiver or if you can transport multi channel PCM audio over HDMI to your receiver then the best way to handle DD+ audio is to keep it as it is. However, if you have SPDIF/coax connection to your receiver, you can only transport AC3 or DTS. So in that case you'd have to reencode the DD+ track to either AC3 or DTS.

I also guess that keeping the track as it is is also not a good solution (apart from the question if it was playable at all then?), because theres lot of additional junk in it like with dts-hs & dtscore ?
There's no junk in DD+ at all. And it's a good idea to keep it as it is. But see above.

another question I have is the removal of dialog normalisation, is such dialog normalisation included in every audio track? is the dialog normalisation somehow intended by the film makers, otherwise one wouldnt understand what people say in certain scenes? does it sound more natural & better without dialog normalisation?
The one and only purpose of dialog normalization is that it tries to save you from having to readjust volume during advertising breaks and when you switch from one movie to another. Basically dialog normalization turns the volume knob on your receiver so that the volume of speaking voices is the same in advertising and in every movie you're watching. Without dialog normalization advertising can sometimes be much louder than the movie audio track. Or it could happen that one movie is much louder mixed than another movie. So without dialog normalization you'll have to adjust the volume everytime there's an advertising break or everytime you switch to another channel/movie to keep volume at the same level.

Unfortunately, in real life most movies leave the "dialog normalization" information in their audio tracks at the default value. That means dialog normalization often simply doesn't work correctly. Furthermore many people argue that doing dialog normalization harms audio quality. Because practically it's not really the physical volume knob of your receiver that is changed by dialog normalization, but instead the digital audio track is processed to make audio volume lower. This effectively reduces bitdepth of the audio track (a very little bit).

Sony has decided to not use dialog normalization, anymore. This is based on feedback they received from home cinema consumers. But most other studios still leave dialog normalization activated (and most of the time on the default value).

[Thunderbolt8]

well, standart setup (should have said basic setup) means I just use my pc via ac3filter (or maybe ffdshow audio) to playback the movie/audio and currently just have 2 hi-fi boxes (non computer boxes), which are addressed via the rear green output. so basically nothing speacial, no external receiver or such :D
would the original, non converted track be able to played back via ac3filter? would I be able to listen to any improvements compared to a converted ac3/dts track? I guess the removal of the dialog normalization of the DD+ track then wouldnt change the fact that its still (un)playable with ac3filter?
in case I decide to convert to dts or ac3 and also would like to use dialog normalization, is there a fixed order I should apply to, e.g. first dialog normalization removal and then converting (or vice versa)?
when remuxing with mkvmerge, should the ending for the DD+ track kept .mpa or should/must it be changed into something else ?

[madshi]

well, standart setup (should have said basic setup) means I just use my pc via ac3filter (or maybe ffdshow audio) to playback the movie/audio and currently just have 2 hi-fi boxes (non computer boxes), which are addressed via the rear green output. so basically nothing speacial, no external receiver or such :D
Not even multi channel! :(

would the original, non converted track be able to played back via ac3filter?
Nope, at least not until ac3filter adds E-AC3 decoding support.

in case I decide to convert to dts or ac3 and also would like to use dialog normalization, is there a fixed order I should apply to, e.g. first dialog normalization removal and then converting (or vice versa)?
As long as you use eac3to for the conversion, you don't have to care.

when remuxing with mkvmerge, should the ending for the DD+ track kept .mpa or should/must it be changed into something else ?
Don't know how mkvmerge behaves. But I'd suggest renaming it to "ddp" or "eac3". Just try it out. If mkvmerge detects that it's E-AC3, you're done.

[Thunderbolt8]

well, since the track should be quite small compared to the video size of the remux I guess I'll do both, convert it, but also try to mux the original track into the .mkv to be on the save side for the future. when time comes, I only need to remux without the converted track then, instead of working with the HDDVD source again :S

edit: no changing of the ending of the demuxed audio at all needed for DD+, mkvmerge detects is as eac3 at once :)

[Thunderbolt8]

hm just read the first line of the OT again, wouldnt it make more sense for me just to convert the DD+ to flac? would it be completely lossless then, compared as when taking flac from a LPCM track? I guess that might be the best option then, because I can use it already now and im sure with better hardware it should still be possible to use it still in the future (wasnt there something about analog plug that allows flac use with external receivers?). or am I mistake here?

btw. what the difference between that flac121a, flac121b and the win.zip file from sourceforge, which one should I download?

[Taktaal]

Has anyone kept an old version of eac3to? It used to work very well but the versions in the last 2 months added only bloat and more dependancies. Unfortunately I overwrote my download from earlier this year...

[juzzy999]

I cant get eac3to encode a DTS-HD track.

I wont recognise the file.. How can i re-encode it to an AC3??????

[madshi]

hm just read the first line of the OT again, wouldnt it make more sense for me just to convert the DD+ to flac? would it be completely lossless then, compared as when taking flac from a LPCM track? I guess that might be the best option then, because I can use it already now and im sure with better hardware it should still be possible to use it still in the future (wasnt there something about analog plug that allows flac use with external receivers?). or am I mistake here?
Sure you can do that. But DD+ decodes to 24bit. And reencoding that to FLAC will consume a lot of space. Lossless compression works best for 16bit. The higher the bitdepth of the audio track is the lower is the compression efficiency of lossless compression.

btw. what the difference between that flac121a, flac121b and the win.zip file from sourceforge, which one should I download?
The FLAC which comes with eac3to is the only one which can handle big files without any problems.

[madshi]

Has anyone kept an old version of eac3to? It used to work very well but the versions in the last 2 months added only bloat and more dependancies. Unfortunately I overwrote my download from earlier this year...
Which dependancies do you mean? If you have a suggestion how to improve eac3to, just let me know.

[madshi]

I cant get eac3to encode a DTS-HD track.

I wont recognise the file.. How can i re-encode it to an AC3??????
I can't say anything without more details. Where did you get the DTS-HD track from? Which file extension does it have? Try renaming it to "something.dtshd". And please post the full eac3to text output here.

[onlym3]

How do i convert a TrueHD track from a Bluray? The info on the first page says use evodemux to create an evo but that only works for HD DVDs since you first need to load an evo to be able to rebuild it.

[Thunderbolt8]

regarding the blue ray thing, how do I manage to use something similar like offsetpts for evo files for m2ts files?
afaik a common way is, when having multile m2ts files to add them together like with copy /b etc. but what if there are also slightly audio delay problems then, like in some hddvd cases? for that reason madshi created offsetpts, but how can I treat the blue ray files to prevent desyncing?

[madshi]

How do i convert a TrueHD track from a Bluray? The info on the first page says use evodemux to create an evo but that only works for HD DVDs since you first need to load an evo to be able to rebuild it.
Blu-Ray TrueHD tracks are not fully supported yet. You can demux them with "TsRemux" or "xport" and then eac3to can extract the AC3 track which is stored inside the TrueHD track. But currently decoding of the full TrueHD information is not possible yet. However, I hope to be able to make that work soon.

[madshi]

regarding the blue ray thing, how do I manage to use something similar like offsetpts for evo files for m2ts files?
afaik a common way is, when having multile m2ts files to add them together like with copy /b etc. but what if there are also slightly audio delay problems then, like in some hddvd cases? for that reason madshi created offsetpts, but how can I treat the blue ray files to prevent desyncing?
Currently I don't know any way to "delay" m2ts files. Maybe I'll enhance offsetpts sooner or later to also handle m2ts files.

[Thunderbolt8]

i wonder about one thing, I converted the 5.1 eac3 from mad max: the road warrior to ac3 and the size remained the same for the ac3 track. I did the same with the 2.0 eac3 commentary track, but the ac3 size doubled in that case. is this a normal procedure? afaik the eac3 track had 192kbps, but the ac3 file now has 384kbps :S (made with eac3to)

[madshi]

i wonder about one thing, I converted the 5.1 eac3 from mad max: the road warrior to ac3 and the size remained the same for the ac3 track. I did the same with the 2.0 eac3 commentary track, but the ac3 size doubled in that case. is this a normal procedure? afaik the eac3 track had 192kbps, but the ac3 file now has 384kbps :S (made with eac3to)
File size depends only on bitrate and movie duration. You can tell eac3to which bitrates to use. If you don't do that eac3to decides for its own.

[Thunderbolt8]

hm since the program made it right with the 640kbps for the 5.1 track I assume it could somehow autodetect the right bitrate. guess I have to convert again then and enter 192kbps manually

[madshi]

hm since the program made it right with the 640kbps for the 5.1 track I assume it could somehow autodetect the right bitrate. guess I have to convert again then and enter 192kbps manually
Well, what is "right" and what is "wrong"? Reencoding with a lossy encoder always hurts audio quality. eac3to has *intentionally* chosen a higher bitrate than the original bitrate in order to keep the audio quality loss as small as possible. For 5.1 tracks eac3to is generally using 640kbps. For stereo tracks eac3to is generally using 384kbps, IIRC. These are default values I've chosen. If you don't like them, you have the freedom to tell eac3to to use other values. But I'm not going to lower the defaults. I don't agree that matching the original bitrates is "right".

[Taktaal]

The problem is that both Nero and Sonic are really bugged. Nero is a 1.4GB program just to burn some CDs and play movies, which would be funny by itself if it wasn't so pathetic, is just about impossible to configure properly for HD-DVD playback and hoses your Windows install after a few tries of getting it to work.
And Sonic just dumps a few uncaught exceptions right onto the command line in a stunning display of programmer incompetence if you want to use it to decode E-AC3.

[honai]

The problem is that both Nero and Sonic are really bugged.

If you have found bugs you can always report them to Nero or Sonic.

Nero is a 1.4GB program

What? My installation folder weighs less than 200 MB.

which would be funny by itself if it wasn't so pathetic

What is so pathetic about it?

is just about impossible to configure properly for HD-DVD playback

Impossible as in not possible? How come that it works for others?

and hoses your Windows install after a few tries of getting it to work.

No, it doesn't. The problem most likely exists between keyboard and chair.

And Sonic just dumps a few uncaught exceptions right onto the command line in a stunning display of programmer incompetence if you want to use it to decode E-AC3.

The Nero filters are working fine.

And looking back at your post from 24th September 2007, 23:56:

Has anyone kept an old version of eac3to? It used to work very well but the versions in the last 2 months added only bloat and more dependancies.

and considering that you don't offer any specific bug reports at all it seems that your major intention in this thread is to badmouth software that - gasp! - seems to work just fine for many other users.

I find it especially rich that you claim that the latest versions of madshi's work "added only bloat". He's working on this for free, and you get even the full source code. I don't think you're in a position to discard his work in such a way.

[Taktaal]

It was perhaps not said in a very nice way, I'm sorry. But I did specify that the issues aren't with the eac3to code itself, but with the dependencies.

[honai]

So what are the issues exactly? Perhaps readers can help you out.

[madshi]

I did specify that the issues aren't with the eac3to code itself, but with the dependencies.
I'm still not sure which dependencies you're talking about. There have been no dependencies added at all in recent versions. Actually I've been *reducing* dependencies over time.

[madshi]

The problem is that both Nero and Sonic are really bugged. Nero is a 1.4GB program just to burn some CDs and play movies, which would be funny by itself if it wasn't so pathetic, is just about impossible to configure properly for HD-DVD playback
Fortunately all we need from the Nero package in this thread is the Nero Audio Decoder and that one is relatively small and works quite well.

[Thunderbolt8]

I read somewhere that with sonic audio decoder can I can actually use the DD+ track, it is able to decode it. is this right and can I benefit from this track with this decoder already fully, meaning hes able to decode & output 'everything there is' additionally compared to an ac3 only track?

[madshi]

I read somewhere that with sonic audio decoder can I can actually use the DD+ track, it is able to decode it. is this right and can I benefit from this track with this decoder already fully, meaning hes able to decode & output 'everything there is' additionally compared to an ac3 only track?
Read the first post of this thread to learn about disadvantages of the Sonic decoder.

[madshi]

eac3to v1.16 released

http://madshi.net/eac3to.zip

This version can now fully decode demuxed TrueHD files from both HD DVD and Blu-Ray!!

That means: Sony TrueHD tracks can now be decoded bit perfectly because Sony doesn't use Dialog Normalization. Sadly, for other studios TrueHD decoding still isn't bit perfect because the Nero Audio Decoder does post processing (Dialog Normalization).

Here are test results with "The Fifth Element (Remastered)":

(1) Original 20bit TrueHD track in m2ts container: 4.12GB.
(2) Demuxed TrueHD track: 3.36GB.
(3) TrueHD after removal of interweaved AC3 frames: 2.81GB.
(4) 20bit TrueHD reencoded to FLAC: 2.54GB.
(5) 16bit LPCM track reencoded to FLAC: 1.55GB.

From the numbers it seems that TrueHD decoding is perfect for Sony Blu-Ray tracks. Now if only I knew how to remove Dialog Normalization for Blu-Ray tracks of other studios... :(

[HD-DVD±RW]

@madshi just wanted to thank you greatly for providing a super useful application. I've used it on many occasions with great success. Keep up the good work.

[Thunderbolt8]

^^completely agree. well done.

[nmeli15]

When trying to convert ddp to ac3 with eac3to with Sonic-orbitlee error message.............gettin dts/ac3/dd+ source instances failed............what does that mean........

[madshi]

@madshi just wanted to thank you greatly for providing a super useful application. I've used it on many occasions with great success. Keep up the good work.
Thanks for the feedback!

[madshi]

When trying to convert ddp to ac3 with eac3to with Sonic-orbitlee error message.............gettin dts/ac3/dd+ source instances failed............what does that mean........
It probably means that the orbitlee filter is not properly registered.

[Thunderbolt8]

whats the difference between orbitlees's DD+ source filters and the ones that are applied otherwise, when those are not applied?

[homerpez]

I've hit a serious snag while trying to back up the Cantonese audio from "Kung Fu Hustle" on Blu-Ray. The goal is a standard, ordinary 5.1 AC3 file, since it's just for backup.

The "Lossless PCM" track is 5.1 audio, and Power DVD info tells me it's LPCM.

I Demux this track with TSremux, and call it "Kung Fu Hustle.pcm".

Using Eac3to (the latest as of this date), this is what it gives me:

C:\temp\TEST>"C:\VIDEO\Rippak\EAC3TO\eac3to.exe" "C:\temp\TEST\Kung Fu Hustle.pc
m" "C:\temp\TEST\Kung Fu Hustle.ac3" -448 -orbitlee -16 -big
RAW, 5.1 channels, 1:39:04, 16 bits, 48khz
Converting the raw file to wav. Please wait...
Converting the wav file to ac3. Please wait...

Aften: A/52 audio encoder
Version 0.0.8
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

Warning, unsupported file size.
input format: WAVE Signed 16-bit little-endian 48000 Hz 5.1-channel
output format: 48000 Hz 3/2 + LFE

SIMD usage: MMX SSE SSE2 SSE3
Threads: 2

progress: 0% | q: 188.4 | bw: 34.0 | bitrate: 448.0 kbps

Done.

I know the LPCM stream works, when I play the movie off the hard drive in PowerDVD,it sounds fine.

When I play back the RAW PCM file, and it's garbled digital sound (I think it's supposed to at this stage). I play back the resulting AC3 file from running Eac3to, and it's still garbled, though I can hear the sound some in the background (but it is still digital noise). On the AC3, it turns into an audible BUZZZZZZ sound.

I am fairly sure that it is 16-bit instead of 24 bit, because when I select that, the timecode becomes correct in Eac3to's status, and the resulting AC3 matches the movie (on 24-bit, it doesn't). This does no good, when the sound is digital mish-mash. :P

I have tried both 24-bit and 16-bit, big endian, little endian, as well as forcing either the nero filter or the orbitlee filter. The result is the same: the AC3 has an audible BUZZZZZZ and is digitally garbled.

Is there something I'm doing wrong? Hopefully the copied info above may be able to help? Thanks in advance...

[madshi]

@homerpez, try using xport instead of TsRemux. I've had trouble with TsRemuxed LPCM tracks before while xport has never failed me yet.

[Sephiroth0000]

Can anyone help at all??
I know most of you are far ahead with regards to HD and making it WMV and what not with the 5.1 audio but im finding myself having serious problems here. I am following a persons tutorial on converting HD-DVDs to WMV HD with 5.1 audio channels http://forum.doom9.org/showthread.php?t=123010&page=5 thread no. 87. Now I have got to the very last stage of the tutorial but when I attempt to transcode I get this error...

C:\HD>eac3to.exe "C:\hd\PEVOB_1_MERGED.DD+.stream.00.ddp" "C:\hd\output.wav"
E-AC3, 5.1 channels, 2:16:18, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Getting "Nero Audio Decoder 2" instance failed.

C:\HD>tranzcode.exe "C:\hd\output.wav"
Error: Unable to open file: C:\hd\output.wav

C:\HD>wavavimux -o audio.avi -iwav 6 output-FL.wav output-FR.wav output-C.wav ou
tput-LFE.wav output-SL.wav output-SR.wav -mask 63
'wavavimux' is not recognized as an internal or external command,
operable program or batch file.

C:\HD>cscript.exe "C:\Program Files\Windows Media Components\Encoder\wmcmd.vbs"
-input "C:\hd\Video_Only.avs" -a_input "C:\hd\audio.avi" -output "C:\hd\output.w
mv" -v_bframedist 1 -v_bdeltaqp 1 -v_mslevel 1 -v_codec wvc1 -v_keydist 3 -v_mod
e 2 -v_quality 96 -v_mmatch 0 -v_msrange 0 -v_performance 20 -v_loopfilter 1 -a_
codec wmapro -a_setting 384_48_6_16

I have tried looking up these errors but with no success. If anyone could give me any input with this I would really appreciate it because to be honest im all out of idea's.

[Taktaal]

Your first command already failed so there's no use in continuing with the others.
The error code seems to indicate that you don't have Nero installed on your system. Once you install it you should get to a point where it says something like "expected 4GB file, but got 0 byte". And that's about as far as you're gonna get!

[Sephiroth0000]

Your first command already failed so there's no use in continuing with the others.
The error code seems to indicate that you don't have Nero installed on your system. Once you install it you should get to a point where it says something like "expected 4GB file, but got 0 byte". And that's about as far as you're gonna get!

But I do have Nero installed and I also have the Blu-Ray HD DVD plugin also. You seem to know what you are talking about so do you have any suggestions at all please? Where is that I have gone wrong? I understand that it is trying to load up Nero to continue the process of the audio but what I do not understand is why the error?

[TheSof]

I just used eac3to on a demuxed truehd .mpa (superman returns) and it worked fine. However, it produced a 24bit flac. If the thd is 16 bit, should I be using the -16 switch? Or do the nero decoders only output 24bit?

[Thunderbolt8]

good question, I was wondering in a similar way, whether I should use 16 or 24 bit for FLAC for a 24-bit LPCM lossless track. madshi told somewhere above that the compression works better for 16-bit, but I cant judge how much the loss would be compared to 'real lossless' then with 24-bit or something like that. at least most people I know restrict flac to 16-bit.

[tebasuna51]

I've hit a serious snag while trying to back up the Cantonese audio from "Kung Fu Hustle" on Blu-Ray. The goal is a standard, ordinary 5.1 AC3 file, since it's just for backup.

The "Lossless PCM" track is 5.1 audio, and Power DVD info tells me it's LPCM.
...

If your file is really LPCM and you are a BeHappy user maybe you can try open this avs file in BeHappy:
NicLPCMSource("C:\temp\TEST\Kung Fu Hustle.pcm", 48000, 16, 6)

Maybe you need the last NicAudio_20070821 (http://avisynth2.sourceforge.net/NicAudio_20070821.zip)

[homerpez]

@homerpez, try using xport instead of TsRemux. I've had trouble with TsRemuxed LPCM tracks before while xport has never failed me yet.

Thanks! That worked!

It was a little effort getting xport to do what I wanted (I hate command line programs...) but it did eventually work for me. You saved the day.

[madshi]

But I do have Nero installed and I also have the Blu-Ray HD DVD plugin also. You seem to know what you are talking about so do you have any suggestions at all please? Where is that I have gone wrong? I understand that it is trying to load up Nero to continue the process of the audio but what I do not understand is why the error?
Do you have Nero 7 or Nero 8 installed? Try the latest Nero 7 version. The error eac3to is reporting simply means that the Nero Audio Decoder could not be loaded successfully. I don't know why that happened. You could try (just as a check) renaming GraphEdit.exe to Recode.exe and check if GraphEdit can successfully load the Nero Audio Decoder.

[madshi]

I just used eac3to on a demuxed truehd .mpa (superman returns) and it worked fine. However, it produced a 24bit flac. If the thd is 16 bit, should I be using the -16 switch? Or do the nero decoders only output 24bit?
The Nero Audio Decoder always outputs 24bit for TrueHD regardless of which bitdepth the TrueHD file really has. And it gets even worse: Because of dialog normalization post processing all 24bits do have valid audio data in them. The "-16" switch will not help. This switch is only there for situations where the raw audio data is really 16bit. But in your case the audio data is output by the Nero Audio Decoder is 24bit. So using the "-16" switch would result in garbled audio. You could use the "-down16" switch to downconvert the 24bit data to 16bit, but you'd lose a bit of audio quality this way. What we really need is a way to turn dialog normalization post processing off. If we do that, the Nero Audio Decoder will still output 24bit, but the lower 8 bits will be all zero. FLAC will then encode that as a 24bit FLAC file, but the final FLAC file size will be similar to a 16bit FLAC.

[Sephiroth0000]

Does anyone have any suggestions as to what I should do with regards to the problems im having? (problem on page 44). I know that the command prompt has failed because it cannot find Nero Audio Decoder 2 but I do have Nero installed and I even installed the BluRay HD DVD plugin but still with no success. I have tried looking for this Nero Audio Decoder 2 on my system but cannot find it but my Nero install is complete and it is uptodate aswell.

[SvT]

Does anyone have any suggestions as to what I should do with regards to the problems im having? (problem on page 44). I know that the command prompt has failed because it cannot find Nero Audio Decoder 2 but I do have Nero installed and I even installed the BluRay HD DVD plugin but still with no success. I have tried looking for this Nero Audio Decoder 2 on my system but cannot find it but my Nero install is complete and it is uptodate aswell.

Check your product setup in NERO. The serial for the plugin should show in this list ! If not ... your plugin is not fully registered. (there must be at least 2 serials 1 for Nero and 1 for the plugin)
If the serial is there, I can see no reason why eac3to can't find it.
Did you try Madshi's suggestion ? (renaming GraphEdit.exe......)

Greets.

[Sephiroth0000]

Check your product setup in NERO. The serial for the plugin should show in this list ! If not ... your plugin is not fully registered. (there must be at least 2 serials 1 for Nero and 1 for the plugin)
If the serial is there, I can see no reason why eac3to can't find it.
Did you try Madshi's suggestion ? (renaming GraphEdit.exe......)

Greets.

At last :) Thankyou! I do have the serials for both Nero and the BluRay HD plugin. Renaming Graphedit!? I know its a pain but could you go more into detail please?

[SvT]

@Sephiroth0000 See Madshi's reply to your question (post #884)

[Sephiroth0000]

Do you have Nero 7 or Nero 8 installed? Try the latest Nero 7 version. The error eac3to is reporting simply means that the Nero Audio Decoder could not be loaded successfully. I don't know why that happened. You could try (just as a check) renaming GraphEdit.exe to Recode.exe and check if GraphEdit can successfully load the Nero Audio Decoder.

I have Nero 8 and the Blu-Ray HD plugin.

[Sephiroth0000]

SVT Graphedit can load up Nero Audio Decoder 2. Im sorry everyone but this is very new to me.

[Rectal Prolapse]

Nero 8 will not work - if I read what a Nero deveoper said to someone correctly, Nero 8 blocks access to their filters outside of Nero Showtime and other Nero software.

You are out of luck - you will have to downgrade to Nero 7 in the above case.

[Rectal Prolapse]

Despite the bad news - it could be possible to defeat Nero's protection by monitoring the calls - perhaps the techniques used to make a codecave could be used...

[Sephiroth0000]

I thankyou everyone for the input they have given me and im very thankful! I have downgraded to Nero 7 and now the audio decoder is picked up and it moves on to the next step which is where I encounter another error... (you were right by the way Taktaal)

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

C:\HD>"C:\HD\eac3to.exe" "C:\HD\PEVOB_1_MERGED.DD.stream.00.ddp" "C:\HD\output.
wav" -nero
E-AC3, 5.1 channels, 2:16:18, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Muxing eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 4.38 GB.
The expected file size for 24 bit is 6.58 GB.
The real file size is 0 Bytes.

Any suggestions at all? I do appreciate this everyone and at the end I will be putting all my notes together to make my own tutorial on converting HD DVD to EVO to WMV HD with 5.1

[madshi]

The real file size is 0 Bytes.
This usually happens when the audio decoder refuses to work. I'd suggest that you try uninstalling Nero and reinstalling it. Also make sure that you have really bought the HD DVD plugin. Pirated serial numbers are known to make problems with the HD DVD plugin.

[madshi]

eac3to v1.17 released

http://madshi.net/eac3to.zip

BIG NEWS.

The TrueHD decoding is now perfect for all HD DVD and Blu-Ray 5.1 TrueHD tracks when using the latest eac3to version and Nero 7 !!

Here's a quick test:

(1) TrueHD test file: 22.8 MB.
(2) Converted to FLAC with old eac3to version: 52.6 MB.
(3) Converted to FLAC with new eac3to version: 20.3 MB.

The size problem in (2) was caused by dialog normalization post processing which I've finally gotten rid of now. Please note that TrueHD files are always 24bit, even if the source was 16bit. But as you can see in the quick test above, this doesn't harm the file size, as long as no post processing is applied by the audio decoder.

TrueHD tracks with 2.0 and 7.1 channels are not supported yet. If you have such tracks, please send me a small sample (5MB should be enough) of the demuxed TrueHD audio track.

[Taktaal]

By the way make sure you take an Acronis image before trying to get Nero to work. After the second or third try I had to reinstall Windows from scratch.

[Thunderbolt8]

Great Job, Madshi, Well Done :)


edit: I understand it right, that the less space consumed by the flac track now is only restricted to 'real' 16-bit trueHD tracks? (because I made a test with a 24-bit file (at least evodemux says its 24-bit) and the size stayed the same with the new version, without using any additional option). so what basically happened to that track, size still the same, but dialog normalization removed now?

[Sephiroth0000]

I really appreciate the answers and suggestions I am getting from people.Thankyou :) and I will let you know what happens when I do as you have all suggested.

[hristoff2]

Nice one! :)

[madshi]

edit: I understand it right, that the less space consumed by the flac track now is only restricted to 'real' 16-bit trueHD tracks?
The size advantage is only there for anything < 24bit. TrueHD tracks can be 16bit or 24bit or anything in between. Sony uses 20bit for some Blu-Rays. In theory a TrueHD track can even temporarily switch to different bitdepths throughout the movie. But for a full 24bit TrueHD track the new eac3to version doesn't bring any size advantages. However, having dialog normalization turned off is nevertheless an advantage IMHO cause combining "lossless" with "post processing" doesn't really sound like a good idea to me.

[madshi]

A note to all people interested in TrueHD -> FLAC reencoding:

The new eac3to version is brand new and not well tested yet. So I'd suggest that you do some checks to make sure that the results are as expected before you delete the TrueHD file. E.g. a good test would be to check if audio is in sync both in the beginning and end of the movie.

If you've done some testing I'd be thankful for your feedback. A simple note like "worked fine for [x] movies" would be great. Thank you!

[Sephiroth0000]

Madshi I downloaded and used that new EAC3TO you offered for download and now it has worked the file properly and created the WAV. file and then even took it to the 5.1 multi channel WAV. outputs files but it crashes yet again afterwards saying that WavAviMu is not reconised as an eternal command but that's what im supposed to use next. If you look about 2 pages back you should see the error within my bat. command file when it saids it cannot reconise it. Any suggestions please!? (Im so close :))

[Thunderbolt8]

so far I'd say it works fine for fear & loathing HDDVD. although when observing correctly the cpu usage has increased, compared to the -down16 flac file I had used with 1.16. but I guess this is normal :P I havent observed a sync difference between the 1.16 -down16 flac and the 1.17 one.

[madshi]

Madshi I downloaded and used that new EAC3TO you offered for download and now it has worked the file properly and created the WAV. file and then even took it to the 5.1 multi channel WAV. outputs files but it crashes yet again afterwards saying that WavAviMu is not reconised as an eternal command but that's what im supposed to use next. If you look about 2 pages back you should see the error within my bat. command file when it saids it cannot reconise it. Any suggestions please!? (Im so close :))
Can I please see the full output? Is it eac3to which is crashing or is it a program in your batch after eac3to?

[madshi]

so far I'd say it works fine for fear & loathing HDDVD. although when observing correctly the cpu usage has increased, compared to the -down16 flac file I had used with 1.16. but I guess this is normal :P I havent observed a sync difference between the 1.16 -down16 flac and the 1.17 one.
Thanks for the feedback. I've no idea why the CPU usage should increase, but as long as the FLAC sounds correct (and the CPU usage isn't *too* high) I don't think we need to be worried.

P.S: When using "-down16" the resulting FLAC file is really a 16bit FLAC file. The FLAC file created by 1.17 is a 24bit FLAC file. This might explain the difference in CPU usage. But the file sizes should be roughly similar. How big is your 1.16 -down16 FLAC file in comparison to the 1.17 one?

[Sephiroth0000]

C:\HD>wavavimux -o audio.avi -iwav 6 output-FL.wav output-FR.wav output-C.wav ou
tput-LFE.wav output-SL.wav output-SR.wav -mask 63
'wavavimux' is not recognized as an internal or external command,
operable program or batch file.


That is the error I get after the audio process has completed. I did check that the WavAviMux is on my system and it is(c:\programs folder\windows media encoder\components\WAVAVIMUX\WavAviMux.exe) but for some reason it saids that it is not a reconised command or something. Im so close to sealing this. Thankyou for the input I really do appreciate it.

[madshi]

C:\HD>wavavimux -o audio.avi -iwav 6 output-FL.wav output-FR.wav output-C.wav ou
tput-LFE.wav output-SL.wav output-SR.wav -mask 63
'wavavimux' is not recognized as an internal or external command,
operable program or batch file.

That is the error I get after the audio process has completed. I did check that the WavAviMux is on my system and it is(c:\programs folder\windows media encoder\components\WAVAVIMUX\WavAviMux.exe) but for some reason it saids that it is not a reconised command or something. Im so close to sealing this. Thankyou for the input I really do appreciate it.
Well, it seems that you've now successfully passed the eac3to step. That's really all I know about. I've never used WavAviMux myself yet. But the reason why it fails for you is probably that WavAviMux.exe is in a folder the batch file doesn't know about. In your batch file try adding the full path to WavAviMux.exe. You might have to put the whole path+exe in "".

[Thunderbolt8]

How big is your 1.16 -down16 FLAC file in comparison to the 1.17 one?
0.98GB: 1.16 -down16 file
2.8GB: normal 1.16
2.8GB: normal 1.17 file (both exact same size)
3.29GB: rebuilt .evo (contains only the 5.1 trueHD track)

[madshi]

both have exactly the same size
Ah, ok, that's nice.

[Thunderbolt8]

sorry, had to edit, mixed things up :P
seems like fear and loathing is a "true" 24-bit file, so the filesize didnt change, but only dialog normalization removed?

[madshi]

sorry, had to edit, mixed things up :P
seems like fear and loathing is a "true" 24-bit file, so the filesize didnt change, but only dialog normalization removed?
Yep...

[Rectal Prolapse]

madshi - wow, amazing stuff. I think you did the impossible by defeating dialnorm. Congratulations and I can't wait to try it out (when I find more titles with TrueHD!)

Thanks for such a great tool!

[madshi]

I think you did the impossible by defeating dialnorm.
Well, it was not impossible, but really difficult. Just finding and changing the dialnorm field in the TrueHD headers was not even the problem. The main problem was that the TrueHD headers are protected by a checksum (CRC). So I had to find out how this CRC is calculated. It's quite hard to figure such stuff out without having an official documentation...

[TheSof]

I'm having some trouble with the latest version. With the previous version, I went .thd => flac, no problems except for the dial norm.

Now, the same .thd file throws "The source file format is unknown".

[markrb]

Is there any benefit to using the TrueHD source over the DDP if my final mux source will be as an ac3 file?

Thanks,
Mark

[madshi]

I'm having some trouble with the latest version. With the previous version, I went .thd => flac, no problems except for the dial norm.

Now, the same .thd file throws "The source file format is unknown".
Yeah, I noticed this problem, too. Sometimes demuxed TrueHD files have some "garbage" bytes in front of the real data. Don't know why. eac3to doesn't like this. The next version will fix that.

[madshi]

Is there any benefit to using the TrueHD source over the DDP if my final mux source will be as an ac3 file?
When encoding and especially when reencoding it's the best solution to always use the highest quality source you can get. So the question is: Do TrueHD files have a better audio quality compared to DDP files? Well, that question may be harder than you think. Most people would say yes. However, DDP files are 24bit while most HD DVD TrueHD tracks are only 16bit. It's not totally clear whether DDP files with 1.5Mbps sound better or worse than 16bit TrueHD tracks. The insiders don't really agree there, either.

What is clear is that if the TrueHD track has more than 16bit, it's definitely the way to go. And if the DDP track has less than 1.5Mbps then the TrueHD track is also the way to go. If the TrueHD track is 16bit and the DDP track is 1.5Mbps then I don't know what to recommend...

[madshi]

eac3to v1.18 released

http://madshi.net/eac3to.zip

* bugfix: some TrueHD files were not accepted ("The source file format is unknown")
* change: EVO files are not accepted as source files, anymore
* added: detection and repacking of 16 bit TrueHD tracks
* added: proper detection of "DTS-HD Master Audio" and "DTS-HD High Resolution" tracks
* added: runtime information for "DTS-HD High Resolution" tracks
* bugfix: bitrate information for "DTS-HD High Resolution" tracks
* added: decoding of "DTS-HD Master Audio" tracks (Sonic)
* added: decoding of "DTS-HD High Resolution" tracks (Sonic)
* added: decoding of conventional DTS tracks (Sonic/Nero)
*** full DTS-HD decoding support ***

I think this is another very big release because of full "DTS-HD Master Audio" and "DTS-HD High Resolution" support! But actually I've not done much to make that work. It's just that I found out that the Sonic Audio Decoder can already decode DTS-HD just fine! So I just added automation for that.

Please reread the first post of this thread. I've added/corrected a lot of information there.

Basically a quick summary would be: "DTS-HD Master Audio" decoding should work bit perfectly (I cannot check if it's really bit perfect, though, I'm just guessing from what I can see). So reencoding to FLAC would make a lot of sense. "DTS-HD High Resolution" decoding works mostly fine, too (details see first post). Of course the big question with "DTS-HD High Resolution" is: Into which format should we reencode this to (especially with 24bit tracks)? Reencoding to FLAC would greatly increase file size. Reencoding to anything else doesn't make much sense. Instead we could just extract the DTS core and be done with it. Fortunately most DTS-HD tracks from HD DVD and all DTS-HD tracks from Blu-Ray are lossless (DTS-HD Master Audio). And for those FLAC reencoding should be perfect.

There's one change for TrueHD decoding: Basically eac3to now checks (after decoding) whether the decoded data contains only 16 bits of information or more than that. If there are only 16 bits of information in the raw 24 bit file, eac3to strips the zero bytes and reduces the raw file to 16 bit. This was not really all that necessary. I've compared and the space saving when encoding in FLAC is only about 0.1 percent. However, I just found it cleaner to have 16bit TrueHD tracks encoded in 16bit FLAC instead of 24bit FLAC. Anyway... Interestingly, I also compared DTS and AC3 encoding and the resulting DTS and AC3 files were bit identical, regardless of whether I stripped the zero bytes or not! Huh, I didn't expect that... :) There's a big saving if you reencode a 16bit TrueHD file to WAV, of course. But who does that? :)

Here are some quick checks with DTS-HD decoding/reencoding:

Eragon Blu-Ray, DTS-HD Master Audio, 24 bit:
- runtime 00:09:15
- 1.5Mbps dts core: 100.0 MB
- dtshd file: 267.5 MB
- reencoded to 24bit FLAC: 230.2 MB

Pan's Labyrinth HD DVD, DTS-HD Master Audio, 16 bit:
- runtime 00:09:15
- 1.5Mbps dts core: 100.0 MB
- dtshd file: 128.5 MB
- reencoded to 16bit FLAC: 77.5 MB

Perfume HD DVD, DTS-HD High Resolution, 16 bit:
- runtime 00:09:15
- 1.5Mbps dts core: 100.0 MB
- dtshd file: 134.9 MB
- reencoded to 16bit FLAC: 110.2 MB

I think these are very interesting results! It's clear to see that 24bit lossless compression (Eragon) comes at a big price. DTS-HD Master Audio compression compares "ok" to FLAC here. With 16bit lossless compression FLAC even beats the 1.5Mbps DTS core in file size! DTS-HD Master Audio compression again works "ok" compared to the DTS core. But it's quite clear that for 16bit movie tracks lossless compression is actually a great idea! It's just too bad that lossless compression works noticably worse as soon as you go above 16bit.

It should be added that lossless compression efficiency depends a lot on the audio data. If there's not much happening in the audio track, lossless compression can work very well. You can compare that to how zip behaves: With text files zip reaches extreme compression ratios. But try to compress an EXE file and the compression ratio goes noticably down. The same thing is true with lossless audio compression (FLAC, TrueHD, DTS-HD Master Audio). In comparison the lossy codecs (AC3, E-AC3, DTS and DTS-HD High Resolution) always consume the same space. The audio track file size only depends on the constant bitrate and on the movie runtime. So depending on the movie lossless codecs can sometimes beat lossy codecs in file size. That's especially true for 16bit tracks. But as soon as you go above 16bit, the situation changes dramatically because lossless compression is just much less efficient for more than 16bit (because there's more noise in the additional bit and noise is difficult to compress losslessly).

[hristoff2]

Oh snap! :devil: :)

btw. do we need to use Sonic 4.2 Audio Decoder or 4.3 for DTS-HD decoding? (I remember sth about 4.3 decoding TrueHD badly - but at least it can decode it, so they updated sth)

[madshi]

I'm not sure, personally I'm using 4.3.

[Thunderbolt8]

so I guess apart from that dts-hd high resolution, which is inferior to dts-hd master anyway and also only rarely used, eac3to can convert anything else from blue-rays/hddvds bit per bit perfectly now (to flac)?

:thanks:

[Taktaal]

Madshi I downloaded and used that new EAC3TO you offered for download and now it has worked the file properly and created the WAV. file and then even took it to the 5.1 multi channel WAV. outputs files but it crashes yet again afterwards saying that WavAviMu is not reconised as an eternal command but that's what im supposed to use next. If you look about 2 pages back you should see the error within my bat. command file when it saids it cannot reconise it. Any suggestions please!? (Im so close :))

Which codec did you use to convert the DDP? Nero or Sonic?
How did you fix the 0 byte expected error?

[madshi]

so I guess apart from that dts-hd high resolution, which is inferior to dts-hd master anyway and also only rarely used, eac3to can convert anything else from blue-rays/hddvds bit per bit perfectly now (to flac)?
Yes, I think so... :)

As I mentioned before, I've no way to check if the Sonic DTS-HD Master Audio decoding is bit perfect, but it seems like that. At least there's definitely no dialnorm and no DRC applied. And the Sonic decoder definitely gives out a different result for the DTS-HD Master Audio track compared to if you only feed it the DTS core of the same track. So it looks like being bit perfect.

[Thunderbolt8]

tested the trueHD decoding once again with fear & loathing.

3.10 GB = demuxed trueHD track
5.71 GB = raw file size
2.80 GB = FLAC file size (cant tell though if size was byte
identical with the one produced yesterday with v1.17)

"this track contains more than 16-bit of information" so we have a true 24-bit (or at least 20-bit or something like that) track here on the fear & loathing HDDVD :)

[madshi]

tested the trueHD decoding once again with fear & loathing.

3.10 GB = demuxed trueHD track
5.71 GB = raw file size
2.80 GB = FLAC file size (cant tell though if size was byte
identical with the one produced yesterday with v1.17)

"this track contains more than 16-bit of information" so we have a true 24-bit (or at least 20-bit or something like that) track here on the fear & loathing HDDVD :)
Good. Looks like things are working as expected... :)

[Thunderbolt8]

hm is it possible that the 24-bit FLAC from truehd 5.1 is very slightly out of sync? the one I created for fear & loathing now seems to be a tiny bit, I also believe it was a little better with the former versions. I muxed the flac into .mka and then muxed this together with the video with a 23.9760239 fps timecode .txt file into .mkv
I tested it with ffdshow and also the WMVideo DMO decoder (which uses less cpu performance, because its distributed to both cores; cpu usage for each core is <50 % according to task manager), same result for both (using coreflac btw.)

[madshi]

I've also noticed that I needed to add 200ms delay to my main test FLAC. I'm not sure why that is the case. I'm very sure that this is not a bug in eac3to. I guess that demuxing the TrueHD track loses timecode information which results in this slight audio delay. I don't think it's a big issue, though. Of course we'll need to check and eventually correct audio sync for all TrueHD -> FLAC files and probably also all DTS-HD -> FLAC files now. Oh well, probably that means that I need to write a tool which can apply delays to FLAC tracks! Shouldn't be difficult, fortunately. Maybe sooner or later we'll find a way to calculate the necessary delay? That would be more comfortable, of course.

So everybody who's using the new TrueHD/DTS-HD -> FLAC reencoding: Please check which delay you need to apply and try to find out whether this delay seems to relate somehow to the EVO timestamps (which you can read out with EvoDemux). If we can find out how to calculate the delay, that would be great!

[Thunderbolt8]

im not that familiar with all that technical stuff, how can I manage to find out the evo timestamps with evodemux?
and since you only mentioned truehd and dts-hd now, could this delay thing also happen when remuxing eac3 tracks to flac?

as it seems the delay needed for fear & loathing also needs to be between 175 and 225 ms

[madshi]

Check out the "PTM" of the first video frame and compare that to the "First PTS" of the TrueHD/DTS-HD/E-AC3 audio track in EvoDemux. If these are identical, there should be no audio delay necessary. If they differ, we might need an audio delay. This should be valid for any demuxed audio track, I think. EvoDemux should show the necessary delay. However, for negative delays EvoDemux doesn't show the correct value but a very big positive number instead.

P.S: Or maybe the "First PTS" of the audio track is the delay we need - independently of what the "PTM" of the first video frame sais?

These PTM and PTS values are hexadecimal values. That means: Start calc.exe, switch it to hexa mode, then enter that hex number. Then convert to decimal. Now divide the decimal number by 90 and you get milliseconds. E.g. hex "000034C8" is decimal "13512", divided by 90 is about "+150ms".

[Thunderbolt8]

values are identical for the video stream and all audio tracks on that disc. I ran offsetpts before rebuilding the video evo, but it told me that offset was fine and didnt need to be processed

edit: wait, lol, this does only apply for the 1st evo file. for the 2nd evo file theres a difference, mom


Opening file FEATURE_1.EVO

PTM of first video frame = 00000D8E
Dolby TrueHD audio stream 0 found!
First PTS = 00000D8E



Opening file FEATURE_2.EVO

PTM of first video frame = 125C7D0A
Dolby TrueHD audio stream 0 found!
First PTS = 125C438F (+3422589ms)

(3 of the 4 audio tracks of the 2nd evo have a different delay btw, only the delay of the 2 DD+ 2.0 tracks is the same)

125C7D0A - 125C438F = 397B = 14715 : 90 = 163,5 <-- the delay needed ? (if yes, for the 2nd .evo file only though)


btw. I still have a remux with the v1.17 created flac file from that trueHD track and the sync issue there in the 2nd half of the movie is NOT present. so I guess it must be some little mistake which got in when you made the latest changes maybe.

watched the same scene over and over again, still, the 1.17 version is either complete identical with the evo or has a delay of <20ms or such, which I cant really spot any more. at least I believe they are identical and other thoughts might as well just come from having this watched already too often now. the 1.18 version however, although I set the delay to the above calculated 163ms in mkvmerge now, is STILL A LITTLE out of sync. it has gotten better compared to a remux without any delay before or compared to my attempts to sync it manually, but its still a bit noticeable, compared to the original evo and the 1.17 remux.

[Roscoe62]

This is probably a tiny bit off topic, but I haven't been able to figure out how to find the "right" audio track to extract when I use xport. I know in the command line you have to specify which program number, video track & audio track you wish to extract, but how do I determine which ones are correct?

I'm guessing there's a method to it, but after searching around the forum I still haven't managed to find the answers.

Sorry for the noob question, but with eac3to doing such a fantastic job of the audio (thanks again Madshi! - eac3to is turning into a regular swiss army knife for audio!) my lack of understanding of how to correctly use xport is the only thing slowing me down.

Can anyone point me to a guide, or help me to find what I'm looking for?

Thanks!

[Thunderbolt8]

well yes, it would be nice to have a counterpart to evodemux just for blue ray :P

[Sephiroth0000]

Which codec did you use to convert the DDP? Nero or Sonic?
How did you fix the 0 byte expected error?


Taktall all I done was loaded the new EAC320 up to the folder that contained all the video, audio, grf and avi files and then loaded up the EAC320GUI (must have EAC320 file there to use properly) It actually does the convert for you. I have downgraded my Nero 8 to Nero 7.10.0.1 with the HD BluRay plugin aswell so I do not know if that makes a difference. Oh and also EAC320 GUI lets you force the encoder use so you can use Nero directly which is good to do if you ask me. Oh and WAVAVIMUX you must install it to the folder you are working from NOT WINDOWS MEDIA ENCODER.

[Sephiroth0000]

Im really coming close to getting this done properly everyone (EVO to WMV HD with 5.1) And when I do I shall be adding my own tutorial on how to do it and it will be dummy proof :)
(YES even I got stuck on doing this ha, ha, ha

[Thunderbolt8]

another thing, I tried to install the illiminable_flac decoder to see if hes better/faster than the coreflac decoder, using this download link here:
http://rapidshare.com/files/9974707/illiminable_flac_0.73.1936.exe.html

but after the installation & restart of windows I still dont seem to be able to use it, at least I cant find it as external filter in mpc. did I do something wrong or am I just blind?

[madshi]

edit: wait, lol, this does only apply for the 1st evo file. for the 2nd evo file theres a difference, mom
That shouldn't matter.

btw. I still have a remux with the v1.17 created flac file from that trueHD track and the sync issue there in the 2nd half of the movie is NOT present.
Wait a moment! So your sync problem is only in one half of the movie? I think you didn't say that before, or did I miss that? For me, after using a delay the sync works throughout the full movie.

watched the same scene over and over again, still, the 1.17 version is either complete identical with the evo or has a delay of <20ms or such, which I cant really spot any more.
That's really very strange... I can't see why 1.17 and 1.18 should behave differently!! You did use 1.17 with the *demuxed* TrueHD track, didn't you? Or did you feed the TrueHD evo file to 1.17? That would explain the difference...

the 1.18 version however, although I set the delay to the above calculated 163ms in mkvmerge now, is STILL A LITTLE out of sync. it has gotten better compared to a remux without any delay before or compared to my attempts to sync it manually, but its still a bit noticeable, compared to the original evo and the 1.17 remux.
The calculated delay is all wrong. The timecode values of the 2nd EVO file have no meaning.

[madshi]

This is probably a tiny bit off topic, but I haven't been able to figure out how to find the "right" audio track to extract when I use xport. I know in the command line you have to specify which program number, video track & audio track you wish to extract, but how do I determine which ones are correct?
Personally, I'm using TsRemux to see which audio tracks are in the m2ts files in which order. The first audio track is usually number 1, the 2nd audio track number 2 etc. The I use xport for demuxing, using the numbers I've taken from TsRemux.

[Thunderbolt8]

after watching again, I'd say the delay problem with the 1.18 file also happens in the 1st half of the movie. before I just didnt bother to check it properly enough, because I though it would only apply to the 2nd (evo) half, but as it seems there is also a slight delay present in the 1st half with no delay entered at the muxing stage.

and yes, I used a rebuilt evo (that audio track only) for the 1.17 muxing, because since remuxing and rebuilding both takes ~same time and when demuxing he would have to remux it temporarely into an evo file anyway again for the flac conversion, I thought I just save myself that time. just not possible any more in 1.18 :P

dont have further time to test things throughout the day today as it seems. maybe a bit more time in the evening but cant say that for sure.

[ACrowley]

another thing, I tried to install the illiminable_flac decoder to see if hes better/faster than the coreflac decoder, using this download link here:
http://rapidshare.com/files/9974707/illiminable_flac_0.73.1936.exe.html

but after the installation & restart of windows I still dont seem to be able to use it, at least I cant find it as external filter in mpc. did I do something wrong or am I just blind?

Thats correct..the Decoder is not listed in Directshow. Only the native FLAC Source Filter.
But any Dshow Player should select the Decoder without Problmes.

Note :
illiminable FLAC Decoder cat handle FLAC in mkv container!
So use ffdshow for FLAC in Ciontainer and illiminable for single FLAC Files

[Thunderbolt8]

ffdshow is too slow as I have noticed. though I can still play any 1080p, with ffdshow as it seems theres a little delay sometimes, which is not present when using coreflac. and because of that conversion problem here with the flac delay I thought maybe coreflac is too slow either and tried to find a faster decoder for .mkv files with mpc.

[ACrowley]

ffdshow is too slow as I have noticed. though I can still play any 1080p, with ffdshow as it seems theres a little delay sometimes, which is not present when using coreflac. and because of that conversion problem here with the flac delay I thought maybe coreflac is too slow either and tried to find a faster decoder for .mkv files with mpc.


You should use only the ffdshow AudioDecoder for FLAC in mkv...i wasnt talking about the VideoDecoder.
Thats another Thing...

[Thunderbolt8]

thats what I mean, the audiodecoder :P
im using ffdshow video anyway, but when also using ffdshow audio to decode flac sometimes a little audio delay can occur, which is not present with coreflac.

[ACrowley]

thats what I mean, the audiodecoder :P
im using ffdshow video anyway, but when also using ffdshow audio to decode flac sometimes a little audio delay can occur, which is not present with coreflac.


I never had any Delay with ffdshow Audio decoder.
Try to use 16bit output...dont think it use lot of CPU usage and a DualCore can handle it
I dont like Coreflac cause it wont connect to AC3 Filter. I use AC3Filter in the decoding chain fro flac to get AC3 640 Output ( which sound s still great)

@Madshi
Great Work!!
TrueHD decoding on demuxed Files and without DialogNorm !! Wow!
And full dts hd decoding...wonderfull!

But ive Problems with the dtshd Tracks Perfume HDDVD. eac3to shows the Information and stops...the dtshd must be little bit corrupted .
I demxued both dtshd via Graphedit beacause there where Problmes with evodemux. The extracted core was cracking, you know...
So eac3to 1.18 stops decoding bad dtshd Tracks too ?

[Sephiroth0000]

Do not know if anyone can lend a helping hand to this but I am getting the following error with my Avisynth video only file when I try to play it

DIRECTSHOWSOURCE: GRF file does not have a compatiable open video pin. Graph must have 1 output pin that will bid RGB24, RGB32, ARGB, YUY2 or YV12

Now apart from this error I have mastered everything with regards to making a HD DVD evo image into WMV HD with 5.1 (Yes the audio has been mastered) Someone help !

[Taktaal]

Has anyone actually been able to get the Sonic Decoder to work with an eac3to version after 1.12? After reading this thread and the older EVOdemux one, I think that's when the support broke.

[Sephiroth0000]

Has anyone actually been able to get the Sonic Decoder to work with an eac3to version after 1.12? After reading this thread and the older EVOdemux one, I think that's when the support broke.


Taktall what is it you are trying to do? I assume it is with Audio maybe I can help.

[ACrowley]

Do not know if anyone can lend a helping hand to this but I am getting the following error with my Avisynth video only file when I try to play it

DIRECTSHOWSOURCE: GRF file does not have a compatiable open video pin. Graph must have 1 output pin that will bid RGB24, RGB32, ARGB, YUY2 or YV12

Now apart from this error I have mastered everything with regards to making a HD DVD evo image into WMV HD with 5.1 (Yes the audio has been mastered) Someone help !

Set :

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\VideoDecoder

AllowAllRenderers to 1

Then you can use the VideoDecoder 4.3 as usual via Directshowsource in Avisynth :)

[madshi]

after watching again, I'd say the delay problem with the 1.18 file also happens in the 1st half of the movie. before I just didnt bother to check it properly enough, because I though it would only apply to the 2nd (evo) half, but as it seems there is also a slight delay present in the 1st half with no delay entered at the muxing stage.
Ok, that's actually good because that means that probably the audio is in sync throughout the whole movie, as soon as you apply the correct delay to the FLAC file. It's the same with some E-AC3 files. Not all of them are automatically perfectly in sync, either. If audio sync was correct in one half of the movie, but out of sync in the other half, that would be a much tougher problem to crack!

and yes, I used a rebuilt evo (that audio track only) for the 1.17 muxing, because since remuxing and rebuilding both takes ~same time and when demuxing he would have to remux it temporarely into an evo file anyway again for the flac conversion, I thought I just save myself that time. just not possible any more in 1.18 :P
The main reason why I removed EVO input support in 1.18 was because the dialnorm removal only works with demuxed TrueHD files. So the TrueHD track you converted with 1.17 still has dialnorm applied. When creating v1.18 I noticed that I didn't clearly say that EVO input should no longer be used. And even if I documented that in 1.18 I feared that some people wouldn't read the documentation properly. So I just removed EVO input to force all people to use the proper way to have dialnorm defeated.

Sorry, my fault, should have clearly said that EVO input is not recommended, anymore (when releasing v1.17).

[madshi]

But ive Problems with the dtshd Tracks Perfume HDDVD. eac3to shows the Information and stops...the dtshd must be little bit corrupted .
I demxued both dtshd via Graphedit beacause there where Problmes with evodemux. The extracted core was cracking, you know...
So eac3to 1.18 stops decoding bad dtshd Tracks too ?
I'm not sure. I think 1.18 should just complain about the dirty track, but try to continue. Maybe the audio decoder stopped? How does the full text output look like?

Anyway, you can avoid to have a corrupted audio file in the first place. EvoDemux has a bug with Perfume. I fear if you have rebuilt the DTS-HD tracks into a separate EVO file you might already have corrupted the audio files. Try demuxing the original EVO file by using drmpeg's EVO demuxer. Here's the download link:

http://www.w6rz.net/evob_demux.zip

Using that instead of EvoDemux for Perfume results in perfectly clean DTS-HD tracks. Unfortunately Perfume only has DTS-HD High Resolution tracks encoded from a 16bit master. So the Sonic audio decoder forcefully downconverts to 16bit. I'm not sure which is better: DTS-HD High Resolution downconverted to 16bit (Sonic) or just the core, but with full 24bit (Nero).

[Thunderbolt8]

The main reason why I removed EVO input support in 1.18 was because the dialnorm removal only works with demuxed TrueHD files. So the TrueHD track you converted with 1.17 still has dialnorm applied. When creating v1.18 I noticed that I didn't clearly say that EVO input should no longer be used. And even if I documented that in 1.18 I feared that some people wouldn't read the documentation properly. So I just removed EVO input to force all people to use the proper way to have dialnorm defeated.

Sorry, my fault, should have clearly said that EVO input is not recommended, anymore (when releasing v1.17).
just to sum up the status quo: this means we can only use demuxed trueHD tracks, because otherwise that normalization removal cant work. but then we have that flac delay issue. so when seeing it right, the way to get it all right now is to wait with any conversion until you figure out how it will be right? ;)

[Sephiroth0000]

Set :

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\VideoDecoder

AllowAllRenderers to 1

Then you can use the VideoDecoder 4.3 as usual via Directshowsource in Avisynth :)

ACrowley I think I may have my AVIsynth wrong. This is what the tutorial told me to put in my Video Only.avs file

Directshowsource ("c:\HD\Video_Only.grf", fps=23.976, audio=false, seekzero=false, seek=true, framecount=196142)

I take it I am supposed to have something there in place of DIRECTSHOWSOURCE thing right? What do I put there mate or is it right? I am using Sonic 4.3

[Sephiroth0000]

Oh and also how do I set it please? Like I said im quite new to this. What do I do withh this HKEY thing? Oh and im using Sonic 4.3 not Sonic 4.2

[Thunderbolt8]

go to the command line and type 'regedit'
Oh and also how do I set it please? Like I said im quite new to this. What do I do withh this HKEY thing? Oh and im using Sonic 4.3 not Sonic 4.2
doenst matter

[madshi]

just to sum up the status quo: this means we can only use demuxed trueHD tracks, because otherwise that normalization removal cant work. but then we have that flac delay issue.
Correct.

so when seeing it right, the way to get it all right now is to wait with any conversion until you figure out how it will be right? ;)
NO.

I don't know why you're so obsessed with the "delay issue"? Having to add a small positive or negative delay to an audio track is "everyday business" for any serious reencoder. It's a very usual thing and not really an issue. The same problem sometimes (not always) also occurs with E-AC3 tracks. And you'll have the same problem everytime you try to mux a DVD audio track to a HDTV broadcast.

The only real issue right now is that delaycut doesn't support FLAC and it probably never will. However, the very next eac3to version will be able to "delay" FLAC files. You'll still have to figure the correct delay value out yourself manually, though.

Maybe sometime in the future we'll find a way to automatically determine which delay value is needed (for TrueHD, DTS-HD and also for E-AC3 tracks). But right now I don't know how to do that. But as explained above, I don't consider this as an important issue. We can handle the situation just fine today. It just needs a very little bit of additional manual work.

[madshi]

eac3to v1.19 released

http://madshi.net/eac3to.zip

* bugfix: still some TrueHD files were not accepted ("The source file format is unknown")
* added: FLAC supported as source/input file format now
* added: full delay functionality
If you want to delay a FLAC audio track by 200ms, you can now do this:

"eac3to source.flac dest.flac 200ms"

The FLAC track will then be decoded, the delay will be applied on the raw decoded audio data, and then the final raw audio data will be reencoded with FLAC again. Since FLAC is a lossless decoder this is like unzipping, changing a text file and zipping it again. There's no loss in audio quality doing it this way because FLAC is lossless. The one little disadvantages of this delay technique is that a full decode and reencode is necessary which of course costs time.

Delay also works for any other audio format, as long as decoding or encoding is involved. E.g. you can apply a delay when converting TrueHD to FLAC. However, eac3to's new delay functionality doesn't work without reencoding. If you want to delay AC3, E-AC3 or DTS files, your obvious choice is still the "delaycut" tool, of course.

[Thunderbolt8]

I don't know why you're so obsessed with the "delay issue"? Having to add a small positive or negative delay to an audio track is "everyday business" for any serious reencoder. It's a very usual thing and not really an issue. The same problem sometimes (not always) also occurs with E-AC3 tracks. And you'll have the same problem everytime you try to mux a DVD audio track to a HDTV broadcast.
its not that im obsessed with delay, I just want to be able to have it as perfectly as possible. the problem with delay is just once you know its not 100% accurate you'll notice it throughout the film and this is really anyoing. and especially with movies where mimic when speaking and quickly numbled stuff occurs throughout the whole movie in extreme ways syncing is a real nightmare, because one scene it looks fine and in the other one it doesnt any more. manually applying a delay when muxing is not a problem, but knowing a delay for such movies is almost impossible. I already synced a king kong broadcast with something like 23.9755 fps or something like that, because a static delay didnt help me,so I had to find out how to alter the video fps to come closer to the right delay and it all took hours to find out. its quite good now, but still not perfect towards the end when watching real closely.
therefore, if a way to calculate the exact delay exist, it would be an enormous help.

[madshi]

its not that im obsessed with delay, I just want to be able to have it as perfectly as possible. the problem with delay is just once you know its not 100% accurate you'll notice it throughout the film and this is really anyoing.
I see no reason why it shouldn't be possible to get it 100% accurate. Ok, maybe only 99%. But it should be possible to get the sync so near that it looks perfect to our eyes/brain.

I already synced a king kong broadcast with something like 23.9755 fps or something like that, because a static delay didnt help me,so I had to find out how to alter the video fps to come closer to the right delay and it all took hours to find out. its quite good now, but still not perfect towards the end when watching real closely.
Well, yes, trying to sync an audio track which doesn't want to fit even after a lot of work can be a royal pain in the a**. But this should happen with TrueHD/DTS-HD/E-AC3. These tracks should only need one specific static delay. And with the right delay sync should be perfect throughout the whole movie. So I do not consider this as a problem.

therefore, if a way to calculate the exact delay exist, it would be an enormous help.
It would be more comfortable. But IMO it's not that important. As I've already mentioned several times, with some movies E-AC3 tracks are not in sync, either, and need manual fixing. And nobody has even complained about that yet.

[Thunderbolt8]

well its REALLY more comfortable, when that attempt takes up several hours already. depending on how big the delay is and how much you think that video&audio is trying to fool you right now you need to remux quite often, because the mpc audio delay tool only works up to a certain limit and if thats not enough you need to remux the whole file with a delay again and try again with that mpc setting. and when doing this with disc remuxes that take up >15gb of space it begins to take a lot of time. I wouldnt say it when it only took like 15 mins altogether
I just muxed the stupid fear & loathing thing for the Xth time now, from like 50 to 500 ms and delay is almost the same, when watching (cpu usage of both cores <50%) and it seems complete oddly :(

I just added -200ms delay right now just to test and the delay hasnt really changed a bit compared to the 500ms. theres something wrong :S
added 1000ms. same delay as with 500 and -200 :/ whats wrong? could there be a problem, because I muxed the flac into .mka before muxing it all to .mkv?

[madshi]

well its REALLY more comfortable, when that attempt takes up several hours already.
It usually takes me only some minutes to figure the correct delay value out. I think you need to improve your workflow... :)

If you call MPC with the "dub" parameter, you can check delay with the external FLAC file!!! No need to do *any* muxing for delay checking.

I just muxed the stupid fear & loathing thing for the Xth time now, from like 50 to 500 ms and delay is almost the same, when watching (cpu usage of both cores <50%) and it seems complete oddly :(

I just added -200ms delay right now just to test and the delay hasnt really changed a bit compared to the 500ms. theres something wrong :S
added 1000ms. same delay as with 500 and -200 :/
Sounds strange. Does eac3to claim to apply the delay? Try adding 10000ms and check if the FLAC runtime really gets 10 seconds longer.

[Thunderbolt8]

If you call MPC with the "dub" parameter, you can check delay with the external FLAC file!!! No need to do *any* muxing for delay checking.
I know, but as I said that works only for sure for quite small delay values. can get tricky already >500 or sometimes >200 ms

[madshi]

eac3to v1.20 released

http://madshi.net/eac3to.zip

* bugfix: some Blu-Ray TrueHD tracks were not accepted
* change: eac3to output text slightly improved

[Thunderbolt8]

eac3to v1.20 released
*canceling current remuxing*

trying to do a "fresh" demuxing and conversion to flac again. maybe it helps

[madshi]

I know, but as I said that works only for sure for quite small delay values. can get tricky already >500 or sometimes >200 ms
I've no trouble with even e.g. 20 seconds of delay. With bigger delays it helps to write the delay into the "Audio Switcher" -> "Audio Time Shift" edit box and then to restart MPC. From there you can still do smaller delay changes with the keypad "+" and "-" keys. I never need more than a few minutes to find the right delay value - as long as only one static delay value is needed. Of course it takes much MUCH longer if sync keeps drifting away in the middle of the movie.

[Thunderbolt8]

hm Ive always only done it via the audio switcher -> audio time shift box and it only works for shorter delays for me. otherwise everything just get sloooooouuwww. thats why I need remuxes sometimes.
but you mean you can actually change the delay with + and - keys WHILE watching?

edit: LMAO didnt know that! omg all the hours I spent with remuxing and manually tpying etc.:S

hm regarding the external flac, I have the flac file given the same filename as the .mkv file has (apart from the .flac ending), but I cant choose it as external audio tracks. does this only work when I put the flac into .mka container?

[madshi]

hm Ive always only done it via the audio switcher -> audio time shift box and it only works for shorter delays for me. otherwise everything just get sloooooouuwww. thats why I need remuxes sometimes.
but you mean you can actually change the delay with + and - keys WHILE watching?

edit: LMAO didnt know that! omg all the hours I spent with remuxing and manually tpying etc.:S

hm regarding the external flac, I have the flac file given the same filename as the .mkv file has (apart from the .flac ending), but I cant choose it as external audio tracks. does this only work when I put the flac into .mka container?
You need to call MPC with the "dub" command line parameter. Otherwise MPC doesn't pick the "flac" file extension up as an external audio track. Using that "dub" parameter you can even feed MPC TrueHD and DTS-HD EVO and M2TS files as external audio tracks!

[ACrowley]

I'm not sure. I think 1.18 should just complain about the dirty track, but try to continue. Maybe the audio decoder stopped? How does the full text output look like?

Anyway, you can avoid to have a corrupted audio file in the first place. EvoDemux has a bug with Perfume. I fear if you have rebuilt the DTS-HD tracks into a separate EVO file you might already have corrupted the audio files. Try demuxing the original EVO file by using drmpeg's EVO demuxer. Here's the download link:

http://www.w6rz.net/evob_demux.zip

Using that instead of EvoDemux for Perfume results in perfectly clean DTS-HD tracks. Unfortunately Perfume only has DTS-HD High Resolution tracks encoded from a 16bit master. So the Sonic audio decoder forcefully downconverts to 16bit. I'm not sure which is better: DTS-HD High Resolution downconverted to 16bit (Sonic) or just the core, but with full 24bit (Nero).


Ok....Problem is i dont have the evos anymore from Perfume
Both DTSHD was demuxed via SonicHDDemuxer in graphedit.

eac3to Text is simple :
DTS Hi-Res, 5.1 channels, 2:28:00, 16 bits, 2082kbit/s, 48khz, dialnorm: -4dB
g:\

It stops without processing...all other dtshd tracks are working perfect

DTS HD 6.1 Discrete isnt supported at the Moment right ? On Xmen3 BluRay DTSHD 6.1 is get a unsupported Message from eac3to

Ah, and the dtscore from Perfum is 16 bit , not 24bit....

[Zelos]

Hi all,

i have something strange.
i tried to encode dtshd source ( riddick ) and get this message.


J:\Test Riddick\eac3to119>eac3to feature.dtshd test.dts -768
DTS, 5.1 channels, 2:14:31, 24 bits, 1536kbit/s, 48khz
This is already a normal DTS file.

[madshi]

Ok....Problem is i dont have the evos anymore from Perfume
Both DTSHD was demuxed via SonicHDDemuxer in graphedit.

eac3to Text is simple :
DTS Hi-Res, 5.1 channels, 2:28:00, 16 bits, 2082kbit/s, 48khz, dialnorm: -4dB
g:\

It stops without processing...all other dtshd tracks are working perfect
v1.21 will at least try to decode. Well, it does on my PC at least. However, the Sonic Audio Decoder crashes due to the corrupt file. So it doesn't really help. Most DTS tracks have no CRC, so the decoder can't check if a frame is valid or not.

DTS HD 6.1 Discrete isnt supported at the Moment right ? On Xmen3 BluRay DTSHD 6.1 is get a unsupported Message from eac3to
Try v1.21.

[madshi]

i have something strange.
i tried to encode dtshd source ( riddick ) and get this message.

J:\Test Riddick\eac3to119>eac3to feature.dtshd test.dts -768
DTS, 5.1 channels, 2:14:31, 24 bits, 1536kbit/s, 48khz
This is already a normal DTS file.
That's not really all that strange. Some DTS tracks taken from HD DVD and Blu-Ray are simple conventional DTS tracks and not DTS-HD tracks. So there's nothing you need to do. The track you have is already a normal DTS track. No conversion necessary for this one.

[Zelos]

ok thanks madshi i understand now.
but how to encode to 768k ?

[madshi]

This must be something like the twentiest release in the last few days. I will slow down soon, though. Just doing the necessary bugfixes and then I'll take a little break.

eac3to v1.21 released

http://madshi.net/eac3to.zip

* bugfix: 2 channel DTS files were not accepted
* added: DTS-ES 6.1 support
* added: DTS-HD High Resolution Matrix 5.1 support
* added: DTS-HD Master Audio 6.1 support
The discrete 6.1 DTS formats only have one additional channel, opposed to LPCM tracks who have two channels (which are identical, though). To keep everything more or less similar, eac3to is doubling the 6th channel, so that 6.1 DTS tracks end up being 7.1. Of course you can use the "-down6" parameter to limit output to 5.1.

If you find any further DTS or DTS-HD tracks which eac3to is still not accepting, please send me a small sample.

[madshi]

ok thanks madshi i understand now.
but how to encode to 768k ?
Oh, I see! Currently eac3to doesn't understand what you want. You want to reencode DTS to DTS. eac3to didn't expect that because it always things that you want the max audio quality. However, you can work around that by doing "eac3to source.dts dst.wavs". This will give you 6 mono channels. You can then manually feed them to Surcode for encoding with 768kbps. It's a bit more work to do it this way, but the end result should be fine. You should run the final DTS track through delaycut, though, to remove the zero byte padding Surcode usually applies to DTS encodes.

[Zelos]

Perfect madshi !
thanks for the help.

[Thunderbolt8]

You need to call MPC with the "dub" command line parameter. Otherwise MPC doesn't pick the "flac" file extension up as an external audio track. Using that "dub" parameter you can even feed MPC TrueHD and DTS-HD EVO and M2TS files as external audio tracks!
hm have a lot of trouble with that. tried it now with additional commands "pathname:\filename.mkv /dub pathname:\filename.flac"
but coreflac kept crashing all the time right at the start of mpc. I set it to block, it still kept crashing. I removed it as external filter and set ffdshow audio decoder to active instead, but coreflacdecoder still was active and kept crashing. only when I unregistred the coreflacdecoder.ax file via regdrop it would accept ffdshow as audio renderer. but then it made those funny clicking noises all the time, no normal sound came at all and eventually also crashed.
guess I have to stick to the remuxed .mka file when I want to find out the delay :S

[nautilus7]

Hi, i tried to convert a 5.1 dts track to mono wavs using tranzcode and eac3to, but each program gave me different size files:

1. eac3to wav (center channel) 1.116.851.756 bytes
2. tranzcode wav (center channel) 1.489.135.660 bytes

Eac3to reports no DiagNorm in the dts and i used the disable DRC option in tranzcode (don't really know if there's any DRC though). Both wavs look identical in Audacity.

[tebasuna51]

Hi, i tried to convert a 5.1 dts track to mono wavs using tranzcode and eac3to, but each program gave me different size files:

1. eac3to wav (center channel) 1.116.851.756 bytes
2. tranzcode wav (center channel) 1.489.135.660 bytes

Eac3to reports no DiagNorm in the dts and i used the disable DRC option in tranzcode (don't really know if there's any DRC though). Both wavs look identical in Audacity.

The default output for Tranzcode is 32 bits float per sample, but eac3to seems output 24 bit int. This is the difference in size.

With Tranzcode you can use the parameter /24 to obtain a equivalent output.

[madshi]

hm have a lot of trouble with that. tried it now with additional commands "pathname:\filename.mkv /dub pathname:\filename.flac"
but coreflac kept crashing all the time right at the start of mpc. I set it to block, it still kept crashing. I removed it as external filter and set ffdshow audio decoder to active instead, but coreflacdecoder still was active and kept crashing. only when I unregistred the coreflacdecoder.ax file via regdrop it would accept ffdshow as audio renderer. but then it made those funny clicking noises all the time, no normal sound came at all and eventually also crashed.
guess I have to stick to the remuxed .mka file when I want to find out the delay :S
The CoreFlac filter never worked well for me. Try this one:

http://www.free-codecs.com/download/DC-Bass_Source_Filter.htm

It works very well. However, it has two problems:

(1) Output is always only 16bit.
(2) Seeking only works in the first 2GB of the FLAC file.

It's still good enough to do syncing, though. When you're done with syncing, you can still mux the final FLAC file into some container to work around the 2 bugs of this filter.

[nautilus7]

The default output for Tranzcode is 32 bits float per sample, but eac3to seems output 24 bit int. This is the difference in size.

With Tranzcode you can use the parameter /24 to obtain a equivalent output.
:thanks:

[Thunderbolt8]

The CoreFlac filter never worked well for me. Try this one:

http://www.free-codecs.com/download/DC-Bass_Source_Filter.htm

It works very well. However, it has two problems:

(1) Output is always only 16bit.
(2) Seeking only works in the first 2GB of the FLAC file.

It's still good enough to do syncing, though. When you're done with syncing, you can still mux the final FLAC file into some container to work around the 2 bugs of this filter.
hm I need some of the last sections for fear & loathing for syncing, so it wont work for me at least for this movie. But I could just mux it into .mka additionally and then sync it from there and then apply the delay to the .flac file and delete the .mka. flac muxed in mka has still exactly the same delay as before, has it? or could there be some differences because of that putting into the container?

[madshi]

hm I need some of the last sections for fear & loathing for syncing, so it wont work for me at least for this movie. But I could just mux it into .mka additionally and then sync it from there and then apply the delay to the .flac file and delete the .mka. flac muxed in mka has still exactly the same delay as before, has it? or could there be some differences because of that putting into the container?
Yeah, that should work. I think the delay should be the same inside mka as it is outside. Well, if not, you'll find out soon enough... :)

[ACrowley]

This must be something like the twentiest release in the last few days. I will slow down soon, though. Just doing the necessary bugfixes and then I'll take a little break.

eac3to v1.21 released

http://madshi.net/eac3to.zip

* bugfix: 2 channel DTS files were not accepted
* added: DTS-ES 6.1 support
* added: DTS-HD High Resolution Matrix 5.1 support
* added: DTS-HD Master Audio 6.1 support
The discrete 6.1 DTS formats only have one additional channel, opposed to LPCM tracks who have two channels (which are identical, though). To keep everything more or less similar, eac3to is doubling the 6th channel, so that 6.1 DTS tracks end up being 7.1. Of course you can use the "-down6" parameter to limit output to 5.1.

If you find any further DTS or DTS-HD tracks which eac3to is still not accepting, please send me a small sample.

WOW ! Perfect

Until eac3to there was no To0l which decodes 6.1 dts to 7 mono waves :)

Madhsi ,i think Sonic Decoder is Refertnce Decoder ,right ?
So ,for the max Quality its perfect for all DTS decodes i think so ?
I mean it should be similar or better in Quality compared with Tranzcode or NicDTSSource or Foobar ?

Mh..i will take it for all my dts in the future :)

@zelos
Riddick HDDVD has a standard dts File...you can take without any change

@nautilus
Tranzcode use no DRC on dts decoding by default.
But for AC3 tranzcode applies DRC ,but no DialNorm.

[madshi]

i think Sonic Decoder is Refertnce Decoder ,right ?
Yes, I think so.

So ,for the max Quality its perfect for all DTS decodes i think so ?
As far as I can say: Yes.

I mean it should be similar or better in Quality compared with Tranzcode or NicDTSSource or Foobar ?
Yep. FWIW, I've compared DTS decoding with Nero, Sonic and Ac3Filter. Nero and Sonic decodes were identical to the last bit! Probably both are using the DTS reference code. Ac3Filter was too loud. I reencoded the Ac3Filter with Surcode and decoded it again with Ac3Filter. The peaks got bigger and bigger! So definitely too loud. With Sonic the volume stayed the same even with an additional Sonic -> Surcode -> Sonic step.

One thing to note: For DTS Discrete 6.1 decoding you may need to manually set OS settings to 7.1 speakers. Otherwise Sonic might only output 5.1. I'll work around that with the next eac3to build. The OS settings will then not matter, anymore.

[Thunderbolt8]

have a bit of a problem with an eac3 -> flac commentary track in media player classic. the track is DD+ 2.0 192kbps 24-bit and plays fine in an outside .mka file along the movie .mkv but when muxing it into the .mkv file and selecting it via filters -> pathnamefilename -> commentary track, the playback of both, video and audio is suddenly accelerated, the video is running with ~40fps and sound has mickey mouse voices (muxed video with timecodes to 23.9760239). again everything is fine with the normal 5.1 eac3 -> flac track (wasnt even delay needed), but as soon as I switch to the commentary track inside the .mkv this happens. tried both, coreflac and also ffdshow audio decoder, but same result.

[madshi]

have a bit of a problem with an eac3 -> flac commentary track in media player classic. the track is DD+ 2.0 192kbps 24-bit and plays fine in an outside .mka file along the movie .mkv but when muxing it into the .mkv file and selecting it via filters -> pathnamefilename -> commentary track, the playback of both, video and audio is suddenly accelerated, the video is running with ~40fps and sound has mickey mouse voices (muxed video with timecodes to 23.9760239). again everything is fine with the normal 5.1 eac3 -> flac track (wasnt even delay needed), but as soon as I switch to the commentary track inside the .mkv this happens. tried both, coreflac and also ffdshow audio decoder, but same result.
That's weird. I mean if it works inside the .mka file why doesn't it work in the .mkv file? mka and mkv are really the same. Can't explain it... :(

[Thunderbolt8]

found a little workaround for my truehd syncing problem with fear & loathing. I also demuxed the english 5.1dd+ stream and compared the length of both tracks. I guess the difference is the delay I need then for my trueHD track. at least I guess its now the same as to be observed with the original evo. still, even that seems to bit off sometimes there are lot of scenes in that movie where the delay could be +100ms more as well. if it wasnt for that eac3 track I would have had to guess forever.

[nautilus7]

@ madshi

May i suggest to add .ac3 decoding to .wav(s)? I think this would be good as eac3to can all other conversions, except this one. Then, eac3to will become the most complete audio tool.

[ACrowley]

@ madshi

May i suggest to add .ac3 decoding to .wav(s)? I think this would be good as eac3to can all other conversions, except this one. Then, eac3to will become the most complete audio tool.

yeah, but imho there are enough 100% perfect working Methods/Tools

Behappy-NicAsC3Source / Azid etc..

@Madhsi
Yep. i noticed too that AC3Filter DTS decodes are to loud! Peaks are louder compared with any other decoder.
This is not DialNorm related ..its simply to loud

[madshi]

found a little workaround for my truehd syncing problem with fear & loathing. I also demuxed the english 5.1dd+ stream and compared the length of both tracks. I guess the difference is the delay I need then for my trueHD track. at least I guess its now the same as to be observed with the original evo. still, even that seems to bit off sometimes there are lot of scenes in that movie where the delay could be +100ms more as well. if it wasnt for that eac3 track I would have had to guess forever.
With "length" you mean the runtime, I guess? I'm not sure if that is a reliable way to find out delay. Especially if the E-AC3 track is not in sync, either (which happens on some HD DVDs). But it might be a good starting point, so that afterwords only fine tuning is needed?

[nautilus7]

yeah, but imho there are enough 100% perfect working Methods/Tools

Behappy-NicAsC3Source / Azid etc..
Yes, but it would be perfect if there is an all-in-one tool. I 'm not complaining though, eac3to is very good.

[madshi]

May i suggest to add .ac3 decoding to .wav(s)? I think this would be good as eac3to can all other conversions, except this one. Then, eac3to will become the most complete audio tool.
The main problem with that is that all reference AC3 decoders are usually applying DRC when being used outside of their native player software. So if I added AC3 decoding support through Sonic's or Nero's AC3 decoder, we'd end up with DRC. Of course I could use AC3Filter or a similar open source decoder. That way I could probably get around DRC, but then that wouldn't be the reference decoder and there are lots of other tools which do it that way.

[Thunderbolt8]

With "length" you mean the runtime, I guess? I'm not sure if that is a reliable way to find out delay. Especially if the E-AC3 track is not in sync, either (which happens on some HD DVDs). But it might be a good starting point, so that afterwords only fine tuning is needed?
at least the ac3 track seemed to be in sync, cant say if he is 100%, its just too difficult for that movie.
another question, when just opening one of the 2 single .evo files and checking the sound from there, could it also be possible that theres a little delay then, so that its basically only 100% accurate when starting the complete movie 'normally'. or are the delays, when just opening the evo files seperately, also always accurate?

[madshi]

at least the ac3 track seemed to be in sync, cant say if he is 100%, its just too difficult for that movie.
another question, when just opening one of the 2 single .evo files and checking the sound from there, could it also be possible that theres a little delay then, so that its basically only 100% accurate when starting the complete movie 'normally'. or are the delays, when just opening the evo files seperately, also always accurate?
I'm not sure about that. I'm always first joining the movie and muxing it to MKV before I sync the audio tracks.

[TheSof]

Using the latest eac3to, to go from supermanreturns.thd to flac, it gave an flac with a different runtime:

thd 2:34
flac 2:47

Same with V for Vendetta, an extra 10mins.

With v1.17 (i think thats the version, the one where it didn't to 16bit, had dial norm) the runtimes were the same. I can't test it at the moment, but why would this be?

[Thunderbolt8]

I'm not sure about that. I'm always first joining the movie and muxing it to MKV before I sync the audio tracks.
I just asked because I always used that single original .evo file as syncing reference. but in case this could out of sync too, I would always try to sync my remuxes wrong then -.-

[honai]

@madshi

Well, if you added "native" AC3 decoding I might be tempted to create a GUI that incorporates all the latest features of your tool.

[madshi]

Using the latest eac3to, to go from supermanreturns.thd to flac, it gave an flac with a different runtime:

thd 2:34
flac 2:47

Same with V for Vendetta, an extra 10mins.

With v1.17 (i think thats the version, the one where it didn't to 16bit, had dial norm) the runtimes were the same. I can't test it at the moment, but why would this be?
Can't imagine that there'd be a difference between v1.17 and v1.18. Except if you fed the EVO file into v1.17 instead of the demuxed file? Please check if the 2:47 FLAC stays in sync throughout the movie (after you applied the eventually necessary static delay). Maybe the 2:47 has some extra seconds of silence at the end or beginning of the movie? Don't know...

[madshi]

I just asked because I always used that single original .evo file as syncing reference. but in case this could out of sync too, I would always try to sync my remuxes wrong then -.-
The first EVO part should be in sync. But I'm not sure about the 2nd part.

[madshi]

Well, if you added "native" AC3 decoding I might be tempted to create a GUI that incorporates all the latest features of your tool.
Are there important features missing in The_Keymaker's GUI? I've no idea, I'm always using the command line, only...

Maybe I'll add AC3 decoding if I find a way to disable DRC in a reference decoder. That would be worthwhile. But it will be a while before I invest time into that. I've spent too much time on eac3to lately.

[honai]

Are there important features missing in The_Keymaker's GUI? I've no idea, I'm always using the command line, only...

His latest version is missing some command-line parameters.

Maybe I'll add AC3 decoding if I find a way to disable DRC in a reference decoder.

Yes, that's the idea.

By the way, do you have plans to implement STDIN streaming for the source?