eac3to - audio conversion tool [Archive] - Page 145

survivant001

30th November 2008, 21:16

what that means ?

eac3to v2.79
command line: "D:\DVD-tools\RipBot264v1.11.5\Tools\eac3to\eac3to.exe" "F:\Batman-7.ts" 2:"C:\temp\RipBot264temp\job3\audio.1.mp2" -progressnumbers
------------------------------------------------------------------------------
TRP, 1 video track, 2 audio tracks
1: MPEG2, 704x480 60i /1.001 (4:3)
2: MP2, 2.0 channels, 160kbps, 48khz, -226ms
3: MP2, 2.0 channels, 160kbps, 48khz, 15038ms
[a02] Extracting audio track number 2...
[a02] Applying MPx delay...
[a02] Creating file "C:\temp\RipBot264temp\job3\audio.1.mp2"...
[a02] This track is not clean. Processing aborted.
[a02] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 303874048.

how can I fix that ?

Snowknight26

30th November 2008, 21:32

Sorry, mate, but this is a problem which is obviously not related in any way to eac3to. It seems that something with your Haali installation is broken. Can't help you with that, unfortunately...

How about internal matroska read/write support? ;)

madshi

30th November 2008, 23:22

what that means ?

[a02] This track is not clean. Processing aborted.

how can I fix that ?
That means that your source file is probably damaged/corrupt. eac3to currently doesn't handle such files well. You'll have to use a different tool for extracting the video/audio tracks from this TS file.

How about internal matroska read/write support? ;)
So many other important things to do first...

madshi

30th November 2008, 23:32

For those interested, the eac3to (SSRC) resampling graphs are now online here:

http://src.infinitewave.ca

As far as I can see, eac3to belongs into the top group of steep resamplers. It's intentionally not as steep as the original SSRC algorithm, but still belongs to the steepest algorithms in the comparison. However, if you are willing to sacrifice high frequency response and if you don't mind some aliasing artifacts, there are other resamplers which have noticeably less ringing. I've learned that there's not one "best" resampler. Going steeper gives you get better high frequency response, but you buy it with stronger ringing. So the "best" resampling algorithm/parameters depend on the material and also on your taste...

survivant001

1st December 2008, 00:52

@madshi

2.79 works with my file that I tried to convert, but I'm not able to extract the audio track from this file : (10 megs)

http://www.mediafire.com/?sharekey=fc577931a2d88b41d2db6fb9a8902bda

still a cartoon reordered from my FTA.

here the info from mediainfo

General
Complete name : D:\DVD-convertion\done\Batman-5.TSSplit.1-57.ts
Format : MPEG-TS
Format profile : No PAT/PMT
File size : 10.0 MiB
Duration : 1mn 18s
Overall bit rate : 1 067 Kbps

Video
ID : 6690 (0x1A22)
Format : MPEG Video
Format version : Version 2
Format profile : Main@Main
Format settings, Matrix : Default
Duration : 1mn 18s
Bit rate mode : Constant
Bit rate : 705 Kbps
Nominal bit rate : 809 Kbps
Width : 704 pixels
Height : 480 pixels
Display aspect ratio : 4/3
Frame rate : 29.970 fps
Colorimetry : 4:2:0
Bits/(Pixel*Frame) : 0.080

Audio #1
ID : 6691 (0x1A23)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Bit rate mode : Constant
Bit rate : 160 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Resolution : 16 bits
Video delay : -742ms

Audio #2
ID : 6692 (0x1A24)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Bit rate mode : Constant
Bit rate : 160 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Resolution : 16 bits
Video delay : -742ms

survivant001

1st December 2008, 00:53

That means that your source file is probably damaged/corrupt. eac3to currently doesn't handle such files well. You'll have to use a different tool for extracting the video/audio tracks from this TS file.

So many other important things to do first...

can you suggest me a tool that can do that ?

do you think you can add where the audio track failed ? like 341megs : 23:34.112

like that I could just split my movie.. and convert before and after that time.

madshi

1st December 2008, 09:00

can you suggest me a tool that can do that ?
You can use TsRemux or tsMuxeR to do the demuxing/extracting. Then you can run the audio track(s) through delaycut to fix them. Finally, if there's any other audio processing, you can then use eac3to.

do you think you can add where the audio track failed ? like 341megs : 23:34.112
It's already there: "Aborted at file position 303874048."

So the problem is somewhere around 303874048 bytes (probably a few bytes before that).

asarian

1st December 2008, 21:37

Madshi (or anyone else knowledgeable in these matters), i have a Japanese LCPM 2.0 sound track (Macross Frontier, Blu-Ray), which I'd like to convert to DTS 1.5 Mb/s, if possible. I have SurCode 1.0.29. But when I run eac3to, like this:

eac3to 00002.m2ts 3: c:\video\mf4.dts

Then eac3to starts to create two wav files (for left and right, it seems). Not quite what I was looking for. :) Am I missing something? I've done a lot of DTS decoding, just never encoding.

Thanks

nautilus7

1st December 2008, 21:39

It's simple, your source track is 2.0 only. Thus the DTS will be stereo.

asarian

1st December 2008, 22:21

It's simple, your source track is 2.0 only. Thus the DTS will be stereo.

Thanks. I see the two wavs were just an intermediary state, prior to Surcode starting.

nautilus7

1st December 2008, 22:49

Oh, your question had to do with the existence of the wav files, not the number of them...

Yes, surcode needs to be fed with mono wav files.

asarian

1st December 2008, 23:45

Hmm, on my Vmware box I now get the following:

command line: eac3to 00002.m2ts 3: c:\video\mf2.dts -1536
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1:12:28
1: Chapters, 7 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Writing WAVs...
[a03] Creating file "c:\video\mf2.R.wav"...
[a03] Creating file "c:\video\mf2.L.wav"...
[a03] The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

The eac3to FAQ says: "Surcode doesn't like long filenames/paths. Change them and you 'll be ok." I don't see any long filenames, though. Anyone else has any idea?

Thanks

nautilus7

2nd December 2008, 00:07

It should be the vmware. Surcode is generally a strange application.

tebasuna51

2nd December 2008, 00:51

i have a Japanese LCPM 2.0 sound track (Macross Frontier, Blu-Ray), which I'd like to convert to DTS 1.5 Mb/s, if possible. I have SurCode 1.0.29. But when I run eac3to, like this:

eac3to 00002.m2ts 3: c:\video\mf4.dts

Then eac3to starts to create two wav files (for left and right, it seems). Not quite what I was looking for. :) Am I missing something? I've done a lot of DTS decoding, just never encoding.

Surcode only can make 5.0 or 5.1 encodes.

BTW, I can't understand why you need convert a lossless format:
LPCM 2.0 16 bit 48 KHz -> 48000 x 16 x 2 = 1536 Kb/s
to a lossy format:
DTS 2.0 1536 Kb/s

Use flac to less space or preserve LPCM

asarian

2nd December 2008, 01:28

Surcode only can make 5.0 or 5.1 encodes.

BTW, I can't understand why you need convert a lossless format:
LPCM 2.0 16 bit 48 KHz -> 48000 x 16 x 2 = 1536 Kb/s
to a lossy format:
DTS 2.0 1536 Kb/s

I don't have an HDMI 1.3 receiver (yet). So, since I'll be streaming this BD series to my PS3, using the optical out, I figured I wouldn't be able to do LPCM over the 'bitstream' channel. But now I'm not so sure anymore. :) it seems "LPCM 2.0 48Khz" is supported over optical. So maybe it's just 7.1 LPCM that needs to be done over HDMI per se? This warrants some tests.

tebasuna51

2nd December 2008, 02:48

I don't have an HDMI 1.3 receiver (yet). So, since I'll be streaming this BD series to my PS3, using the optical out, I figured I wouldn't be able to do LPCM over the 'bitstream' channel. But now I'm not so sure anymore. :) it seems "LPCM 2.0 48Khz" is supported over optical. So maybe it's just 7.1 LPCM that needs to be done over HDMI per se?

Exact.
"LPCM 2.0 48Khz" is supported over optical.
7.1 LPCM needs to be done over HDMI.

Snowknight26

2nd December 2008, 03:33

eac3to can't detect this FLAC track but madFLAC Source accepts it and is decoded fine with ffdshow.. well, apart from time issue.
http://www.stfcc.org/misc/departed.flac

Only thing I can do is make a graph of madFLAC Source -> ffdshow audio decoder -> Dump, then use eac3to to convert that PCM track to FLAC. :P

madshi

2nd December 2008, 09:13

eac3to can't detect this FLAC track but madFLAC Source accepts it and is decoded fine with ffdshow.. well, apart from time issue.
Will be fixed in the next build.

shanghai2004

2nd December 2008, 14:03

eac3to v2.79 released

http://madshi.net/eac3to.zip

* improved m2ts file joining overlap detection (mainly for interlaced video)
* vob/evo audio delay detection now uses "vobu start presentation time"
* program streams which are neither VOB nor EVO are now reported as "MPG"
* resampling is now automatically activated for AC3/DTS encoding, if necessary
* "Mersenne Twister" random number generator is used for dithering now
* zero padded DTS tracks are now displayed as such
* fixed: 32bit PCM conversion to floating point was broken
* fixed: with some (rare) movies first subtitle began after 50 minutes runtime
* only plugins with the extension *.dll are loaded now

Madshi, seems somthing wrong now....

F:\Chicago\HVDVD_TS>c:\eac3to\eac3to hv001t01.evo+hv001t02.evo+hv001t03.evo+hv00
1t04.evo+hv001t05.evo+hv001t06.evo+hv001t07.evo+hv001t08.evo+hv001t09.evo+hv001t
10.evo+hv001t11.evo+hv001t12.evo+hv001t13.evo+hv001t14.evo+hv001t15.evo+hv001t16
.evo+hv001t17.evo+hv001t18.evo+hv001t19.evo+hv001t20.evo+hv001t21.evo+hv001t22.e
vo+hv001t23.evo+hv001t24.evo+hv001t25.evo+hv001t26.evo
EVO, 1 video track, 2 audio tracks, 1 subtitle track, 14:47:35
1: Joined EVO file
2: h264/AVC, 1080i60 /1.001 (16:9)
3: E-AC3, 5.1 channels, 894kbps, 48khz
(core: E-AC3, 5.1 channels, 3024kbps, 48khz)
4: E-AC3, 2.0 channels, 448kbps, 48khz
5: Subtitle

Wrong duration is listed and something seems to be wrong with the 5.1 track listing. When the files are processed, an endless list of 13ms audio overlaps are reported in the 5.1 track.

Tried eeac3to v2.79 and v2.78 with this result, same EVO files where processed before with older version of eac3to without problems (cannot remember what version though... if important I can check)

madshi

2nd December 2008, 14:26

Wrong duration is listed and something seems to be wrong with the 5.1 track listing. When the files are processed, an endless list of 13ms audio overlaps are reported in the 5.1 track.

Tried eeac3to v2.79 and v2.78 with this result, same EVO files where processed before with older version of eac3to without problems (cannot remember what version though... if important I can check)
Can you upload a little sample for me? Maybe 2 of those EVO parts which are rather small (if there are any such)?

Snowknight26

2nd December 2008, 14:39

Will be fixed in the next build.

Just out of curiosity, what was the issue? Bad FLAC track possibly?

Thunderbolt8

2nd December 2008, 18:48

Don't know how to do that. Do you have a few 2.0 mono samples?
this ac3 track here is 2.0 and at least supposed to be mono:

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

madshi

2nd December 2008, 19:15

Just out of curiosity, what was the issue? Bad FLAC track possibly?
No runtime information in the FLAC track. That threw eac3to off.

this ac3 track here is 2.0 and at least supposed to be mono
Unfortunately this behaves just like any true stereo track does. The header says stereo, there are 2 full channels in there and they are *not* bit perfect identical. So I don't see any reasonable way to find out that this track is mono instead of stereo. Ok, technically I could probably decode the whole track and check whether there are any "big" differences in the waveform anywhere. But I don't think it's worth it...

alc0re

2nd December 2008, 23:23

Why is it that whenever I check any dts file extracted by eac3to with MediaInfo, the length is always slightly shorter than the video or any ac3 file extracted?

Examples:

1) Skinwalkers Bluray

eac3to v2.79
command line: eac3to c:\BDRip\Skinwalkers 2) 1:"C:\BDRip\Skinwalkers\Demuxed\Chapters.txt" 2:"C:\BDRip\Skinwalkers\Demuxed\Skinwalkers.mkv" 4:"C:\BDRip\Skinwalkers\Demuxed\Skinwalkers.dts" -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:31:50
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 (16:9)
3: DTS Master Audio, French, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)

Media Info on Video MKV : 1h 31mn
Media Info on dts : 1h 30mn

2) The Untouchables Bluray

eac3to v2.78
command line: eac3to c:\BDRip\TheUntouchables 1) 1:"C:\BDRip\TheUntouchables\Demuxed\Chapters.txt" 2:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.mkv" 3:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.ac3" 4:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.dts"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 4 subtitle tracks, 1:59:27
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3 EX, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB

Media Info on Video MKV : 1h 59mn
Media Info on Audio DTS : 1h 57mn
Media Info on Audio AC3 : 1h 59mn

Am I doing something wrong? Am I mistaken to think this is going to cause an audio sync issue as the movie gets further along? Could this be a bug with MediaInfo?

Its also listing the skinwalkers core dts file as 1536Kbps when eac3to said its 1509. I open the mkv and the ac3 files I get in Zoom player and they both are the correct length for the movie, but I can't get .dts files to open in Zoom player to check the lenght of the DTS file...and I do have ac3filter installed...

Snowknight26

3rd December 2008, 00:54

Probably a calculation bug. 1:59:27 * (1509/1536) = 1:57:21.

alc0re

3rd December 2008, 07:10

Yah, I think you're right about a calculation error, although it doesnt do it with .ac3 files. Anyways, I kinda figured out its fine.

I just muxed the .dts files in question by themselves to .m2ts files with TSMuxer, and checked the resulting .m2ts file. They are the correct length.

madshi

3rd December 2008, 09:18

Why is it that whenever I check any dts file extracted by eac3to with MediaInfo, the length is always slightly shorter than the video or any ac3 file extracted?
What length is displayed if you do "eac3to Skinwalkers.dts"?

asarian

3rd December 2008, 10:28

Hello,

I'm not sure whether this is a tsMuxeR issue or an eac3to one (but I assume the latter, sorry), but when I use eac3to to extract a RAW/PCM stream from a BD of Macross Frontier, tsMuxeR doesn't recognize the resultant .pcm file any more ("Can't detect stream type"), whereas it is recognized when I select it directly from the m2ts. I uploaded a sample:

3: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
http://rapidshare.com/files/169776965/mf2-test.pcm.html

I extracted it with the .pcm extension.

Thanks

G_M_C

3rd December 2008, 11:47

tebasuna51

3rd December 2008, 11:52

I'm not sure whether this is a tsMuxeR issue or an eac3to one (but I assume the latter, sorry), but when I use eac3to to extract a RAW/PCM stream from a BD of Macross Frontier, tsMuxeR doesn't recognize the resultant .pcm file any more ("Can't detect stream type"), whereas it is recognized when I select it directly from the m2ts.

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

madshi

3rd December 2008, 17:27

Another idea:
DIRAC time stretching (http://www.dspdimension.com/technology-licensing/dirac/). The LE version is free to implement.
I don't know any free tool that does high-quality timestretching, so that would definitely be another killer feature and very useful for 23.976 --> 25 audio conversions.
I've looked into this. The DIRAC documentation contains this text:

2.3 Phase locked multi-channel processing vs. multiple channel processing

The STUDIO version of DIRAC supports stereo while DIRAC PRO supports an infinite number of
channels (memory permitting) that it can process in a phase-locked (synced) manner at the same time. All of these
simultaneous channels are being processed using a phase-locked processing algorithm that ensures that the stereo
(or surround/multi-channel) phase relationship is preserved.

It is important to understand how this works and what this means exactly.

In a stereo recording, important localization cues are provided to the listener through the relative timing of a sound
source between the left and the right ear (channel). If a time stretching process changes the relative timing of the
two channels by even a minimal amount, the stereo image will be perceived as “distorted”. Also, mono
compatibility will no longer be guaranteed, which means that if you mix down the two stereo channels to a mono
channel (as is the case in some TV and radio equipment) you will end up with very audible artifacts perceived as
phasing or even cancellations.

If you have the situation that the relative phase between channels matters, it is imperative to use the multi-channel
processing mode of DIRAC STUDIO and PRO (all channels are being processed at the same time). As a rule of
thumb, phase is always important with stereo recordings, or recordings of the same sound source that were made
simultaneously through different microphones. It is almost always the case with the channels in a surround mix. In
these cases, you should use DIRAC in multi-channel mode, by setting up a single DIRAC object for multiple
channels.
So I contacted the DIRAC company and asked about whether it would make any sense at all to use the free DIRAC version for movie tracks. Here's the reply I received:

"If you are planning on time stretching and pitch shifting 5.1 and 7.1 recordings relative phase is essential. You would need to use the PRO version of DIRAC in order to do this."

In other words: The free DIRAC version is useless for our needs, sadly.

asarian

3rd December 2008, 22:04

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

Okay, thanks. Hadn't dealt with LPCM before, and didn't realize they were that raw. :)

rack04

4th December 2008, 02:32

Any help with the following error? The source is Hellboy 2 Blu-ray.

eac3to v2.79
command line: eac3to "F:\Blu-ray\HELLBOY2_D1" 1) 1: "F:\Blu-ray\Hellboy 2.txt" 2: "F:\Blu-ray\Hellboy 2.h264" 4: "F:\Blu-ray\Hellboy 2.ac3" 10: "F:\Blu-ray\Hellboy 2.sup"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 5 subtitle tracks, 1:59:49
1: Chapters, 21 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
6: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), French
Creating file "F:\Blu-ray\Hellboy 2.txt"...
[a04] AC3 encoding doesn't support back channels. Will mix them into the surround.
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[s10] Extracting subtitle track number 10...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Mixing surround channels...
[a04] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "F:\Blu-ray\Hellboy 2.h264"...
[a04] Creating file "F:\Blu-ray\Hellboy 2.ac3"...
[s10] Creating file "F:\Blu-ray\Hellboy 2.sup"...
[a04] This TS/M2TS file seems to be damaged (sync byte missing).
[v02] This TS/M2TS file seems to be damaged (sync byte missing).
[s10] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.

asarian

4th December 2008, 02:37

Any help with the following error? The source is Hellboy 2 Blu-ray.

eac3to v2.79

[a04] Creating file "F:\Blu-ray\Hellboy 2.ac3"...
[s10] Creating file "F:\Blu-ray\Hellboy 2.sup"...
[a04] This TS/M2TS file seems to be damaged (sync byte missing).
[v02] This TS/M2TS file seems to be damaged (sync byte missing).
[s10] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.

I believe I had the exact same error with this disc. If I recall correctly, what solved it for me was to demux the audio track first with tsMuxeR. Then eac3to would convert the demuxed stream properly.

Chumbo

4th December 2008, 04:33

@madshi,
A request please as you have time. When going from the same format to a target of the same format, i.e., using -slowdown for example, can you use the same bitrate as the source by default please? Below is an example of one track I was slowing down and noticed it was being reencoded at 640Kbps rather than just defaulting to what the source is which, in this case, is 448Kbps. All I had to do was use the -448 to make sure the target is at least the same, but it would be nice to have the feature. Thank you.
eac3to v2.79
command line: eac3to movie.1.ts 3: audio.slow.ac3 -slowdown -log=aud-slow.txt
------------------------------------------------------------------------------
TS, 1 video track, 3 audio tracks, 0:10:40
1: h264/AVC, 1080i50 (16:9)
2: AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -280ms
3: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -280ms
4: AC3 Surround, Achinese, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -280ms
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] Changing FPS from 25.000 to 23.976...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "audio.slow.ac3"...
Video track 1 contains 1075 frames.
eac3to processing took 15 seconds.
Done.

madshi

4th December 2008, 08:38

Any help with the following error? The source is Hellboy 2 Blu-ray.

[a04] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.
This looks like a damaged source file. Either it's a bad rip, or a bug in AnyDVD HD, or an authoring fault. eac3to only accepts clean sources at this time...

When going from the same format to a target of the same format, i.e., using -slowdown for example, can you use the same bitrate as the source by default please?
No, I won't do that. When reencoding audio, the source bitrate is totally independent of the target bitrate. Doing 448kbps -> slowdown -> 448kbps results in worse audio quality compared to 448kbps -> slowdown -> 640kbps.

Beastie Boy

4th December 2008, 09:04

I have become a bit confused about encoding AC3 from a THD file containing a AC3 core. If I do
eac3to input.thd output.ac3
will this extract the core AC3 track or will it encode a new track from the lossless data. What I am trying to do is encode from the lossless portion since I have read that most of the audible difference between the two tracks can be due to a different mix being used, or different masters.

It is quite clear regarding DTS with the -core option, but I need some advice regarding Dolbly. If necessary, I can convert to FLAC first.

Cheers, Beastie.

sehgal.v7

4th December 2008, 09:15

@Beastie
eac3to input.thd output.ac3
It will convert TrueHD to ac3

eac3to input.thd+ac3 output.ac3
It will extract embedded ac3 track.

madshi

4th December 2008, 09:43

The file extension of the source file doesn't really matter at all. But basically sehgal.v7 is right: If the source file contains a TrueHD/AC3 interweaved stream, asking eac3to for the AC3 file will result in a simple extract of the studio provided AC3 track. If the source file contains a straight TrueHD track, only (as is the case with HD DVDs), eac3to will encode a new AC3 track. Currently there's no way to directly force the encoding of a new AC3 track, if there's already an existing one, unless you choose one of the modification options (e.g. a different bitrate, or a volume change or something similar). But you can work around this by first converting to a TrueHD only track (name the target file "*.thd") and then in a separate step transcoding that to AC3. And yes, doing an intermediate FLAC step would have the same effect.

Beastie Boy

4th December 2008, 11:17

Thanks for the replies. I'll use an intermediate THD only track since this doesn't require any encoding and I can save it for future use (FLAC support on the NMTs maybe :) )

Cheers, Beastie.

Chumbo

4th December 2008, 17:22

No, I won't do that. When reencoding audio, the source bitrate is totally independent of the target bitrate. Doing 448kbps -> slowdown -> 448kbps results in worse audio quality compared to 448kbps -> slowdown -> 640kbps.
Wow, I didn't know that. Thanks for the explanation.

Atak_Snajpera

4th December 2008, 20:39

eac3to detects incorrect audio delay

eac3to v2.79
command line: "C:\Users\Dawidos\Documents\Delphi_Projects\RipBot264\Tools\eac3to\eac3to.exe" "D:\_Video_Samples\ts\premiere-paff.ts"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:29
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 2.0 channels, 448kbps, 48khz, dialnorm: -27dB, -3404ms

DGAVCDec reports correct value -719 ms.

sample:http://www.mediafire.com/?onyymjatwjq

eac3to v2.79
command line: "C:\Users\Dawidos\Documents\Delphi_Projects\RipBot264\Tools\eac3to\eac3to.exe" "D:\_Video_Samples\ts\Digiturk.HD.Sirius.2.4.8E.10.jul.2007.ts"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:24
1: h264/AVC, 1080i50 (16:9)
2: MP2, English, 2.0 channels, 320kbps, 48khz, -2532ms

DGAVCDec reports correct value -980 ms.

http://x264.nl/h.264.samples/force.php?file=./jul.2007/Digiturk.HD.Sirius.2.4.8E.10.jul.2007.ts

madshi

5th December 2008, 00:02

eac3to detects incorrect audio delay

DGAVCDec reports correct value -719 ms.
You can't directly compare the audio delay values reported by eac3to and DGAVCDec because eac3to removes all video frames before the first sequence header, while DGAVCDec doesn't do that (I think). Because of that eac3to's audio delay values can be higher.

However, the audio delay with your two test streams was not really correct with the current eac3to version. The next build will use a different delay calculation for such streams with video frames before the first sequence header. The delay numbers will be higher compared to DGAVCDec, but audio should be in sync.

Ryu77

5th December 2008, 02:58

Madshi,

Previously I was using a trial version of ArcSoft TotalMedia Theatre and eac3to recognised the DTS decoder without a problem.

I found a 40% discount coupon for this software so I decided to purchase it. I uninstalled the trial version, then installed the retail version, now eac3to informs me that the DTS decoder isn't installed. Any idea why?

Jeff Flowerday

5th December 2008, 03:09

Ryu77

5th December 2008, 03:10

Is the TMT bin directory in your path? If not give that a try.

I am not sure exactly what you are asking... Would you be able to clarify exactly what I should check?

Thank you. :-)

Also, I was wondering if it must be the TotalMedia Extreme suite or is it ok to only install TotalMedia Theatre?

I ask this because I previously had TotalMedia Extreme trial installed and now I only purchased TotalMedia Theatre retail.

odin24

5th December 2008, 11:20

I am not sure exactly what you are asking... Would you be able to clarify exactly what I should check?

Thank you. :-)

Also, I was wondering if it must be the TotalMedia Extreme suite or is it ok to only install TotalMedia Theatre?

I ask this because I previously had TotalMedia Extreme trial installed and now I only purchased TotalMedia Theatre retail.

TM Theatre should be fine, it's all I have installed and the decoder works fine.

Could the old install directory still have the trial information. Maybe uninstall everything Arcsoft, delete the old install directories then re-install your full version.

Who know, it might work.

madshi

5th December 2008, 11:23

Search for "environment" in this thread. Adding the ArcSoft DLL path to that variable has helped some people in the past...

Ryu77

5th December 2008, 13:13

Search for "environment" in this thread. Adding the ArcSoft DLL path to that variable has helped some people in the past...

Still not working...

I added the path as suggested. My full path environment variable is as follows... %SystemRoot%\system32;%SystemRoot%;%SystemRoot%\System32\Wbem;C:\Program Files\Common Files\ArcSoft\Bin\

The strange thing is I have tried every way imaginable. I have installed it over a trial version where the DTS decoder was working, the full version then removes the ability for eac3to to access it. I have also tried installing the new TotalMedia Theatre on a fresh install of Windows... Nothing seems to be working.

I installed TotalMedia Theatre v2.1.6.126 if that helps.