eac3to - audio conversion tool - Page 534

Sharc · 23rd October 2010, 09:17

Quote:

Originally Posted by dansrfe

Can someone PLEASE address the issue of eac3to NOT solving delay/overlap issues when extracting wavs or the audio files from a set of stringed-together/joined m2ts files from an mpls? It gives all the warnings which is great but it doesn't actually "do" anything about it.

IIRC joining the multiple .m2ts from the .mpls into one big .mkv (as intermediate file for further processing) using MakeMKV solved the problem as well.

Added:
If you start from a BD structure you should also be successful with jdobb's BD Rebuilder.

Thunderbolt8 · 23rd October 2010, 11:07

Quote:

Originally Posted by xxx666yyy777

Try this link:

http://forum.doom9.org/showthread.ph...85#post1443085

I saw that link, but since both setups are so similar, how can you really be sure that its the one and not the other?

xkodi · 23rd October 2010, 11:25

Quote:

Originally Posted by Thunderbolt8

in dts master audio suite you can tick the option to display all 7.1 channel layouts. doing so, theres another setup similar to that of "strange setup" 7.1 tracks which has the 2 additional speakers at the front and not at the back (Lw, Rw instead of Lsr and Rsr). how can we actually be sure that these layouts named as "strange setup" by eac3to are that one with the additional 2 channels at the back and not at the front?

eac3to "-logdts" switch displays detailed information about the DTS file. BTW, how to trick DTS software and get more layouts?

Thunderbolt8 · 23rd October 2010, 12:32

just tick "display all 7.1 channel layouts" under preferences

Thunderbolt8 · 23rd October 2010, 17:43

when having a 32-bit float wavepack file and I want to convert it losslessly to .flac, how to do that best?

when converting what 32-bit float file to .wav and then to flac (because wavepack input is not supported by eac3to), then I get the message "reducing bithdepth from 64 to 24-bits" and the final flac file will have 24-bits which is not lossless any more, right?

same when I convert the 32-bit float file to 32-bit non floating .wav (is this step still lossless btw?) and then to flac, then I get "reducing bitdepth from 32 to 24 bits" and the final flac file isnt lossless any more either, correct?

if so, why is the creation of 32-bit flac files not supported? according to wikipedia, flac supports up to 32-bit: "It can handle any PCM bit resolution from 4 to 32 bits per sample"

flyingernst · 25th October 2010, 09:23

Hi guys. I have a problem
System Win 7 x64 + newest eac3to + Arcsoft tmt3 3.0.1.180

I cant encode a *.dtsma to flac because eac3to didn´t find the arcsoft decoder.
What did I wrong?

Thank, greetings, flyingernst

Edit: I tryed this: http://forum.doom9.org/showpost.php?...postcount=9076 but I get an error

edit edit: i installed Version 2, now it works

yaous · 27th October 2010, 11:22

Sorry for the newbie questions.

Quote:

1: h264/AVC 1080p24 /1.001 (16:9)
2: AC3, German, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)

'eac3to 1) 3: d:\audio.ac3'

this command line extracts embedded ac3 track?
or re-encode TrueHD to ac3?

mrr19121970 · 27th October 2010, 11:30

It will extract the core, to be 100% sure of this add -core.

Code:

eac3to 1) 3: d:\audio.ac3 -core

xxx666yyy777 · 1st November 2010, 14:39

Quote:

Originally Posted by Sharc

IIRC joining the multiple .m2ts from the .mpls into one big .mkv (as intermediate file for further processing) using MakeMKV solved the problem as well.

Added:
If you start from a BD structure you should also be successful with jdobb's BD Rebuilder.

So, does anyone have a definite description of this problem and when/how it occurs? Are these audio delays occuring, when converting to .wav or also, when simply demuxing (to .dts/.dtsma)?

Thanks.

xxx666yyy777 · 1st November 2010, 19:38

Quote:

Originally Posted by xxx666yyy777

So, does anyone have a definite description of this problem and when/how it occurs? Are these audio delays occuring, when converting to .wav or also, when simply demuxing (to .dts/.dtsma)?

Thanks.

I did some small tests and its getting really interesting. First, here is the "make-up" of the BD:

C:\Windows\System32>eac3to y:\ 3)
M2TS, 1 video track, 5 audio tracks, 4 subtitle tracks, 1:43:01, 100.758p
1: Chapters, 34 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 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-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 EX, French, 5.1 channels, 640kbps, 48kHz
7: AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish

Now, I performed several "actions" to obtain the 4th stream - the DTS-HD with the embedded DTS-ES and I compared the durations of the different streams. The audiogaps, as reported by eac3to are:

Code:

a04 Audio overlaps for 5ms at playtime 0:00:44.
a04 Audio overlaps for 12ms at playtime 0:03:12.
a04 Audio overlaps for 9ms at playtime 0:04:16.
a04 Audio overlaps for 11ms at playtime 0:11:46.
a04 Audio overlaps for 6ms at playtime 0:13:37.
a04 Audio overlaps for 7ms at playtime 0:15:39.
a04 Audio overlaps for 11ms at playtime 0:16:00.
a04 Audio overlaps for 6ms at playtime 0:16:43.
a04 Audio overlaps for 10ms at playtime 0:19:24.
a04 Audio overlaps for 9ms at playtime 0:21:04.
a04 Audio overlaps for 8ms at playtime 0:43:06.
a04 Audio overlaps for 12ms at playtime 0:48:20.
a04 Audio overlaps for 10ms at playtime 0:48:53.
a04 Audio overlaps for 10ms at playtime 1:05:27.
a04 Audio overlaps for 10ms at playtime 1:05:56.
a04 Audio overlaps for 11ms at playtime 1:08:40.
a04 Audio overlaps for 10ms at playtime 1:10:41.
a04 Audio overlaps for 10ms at playtime 1:12:10.
a04 Audio overlaps for 6ms at playtime 1:15:25.
a04 Audio overlaps for 11ms at playtime 1:17:36.
a04 Audio overlaps for 11ms at playtime 1:20:47.
a04 Audio overlaps for 10ms at playtime 1:25:00.
a04 Audio overlaps for 7ms at playtime 1:35:54.
a04 Audio overlaps for 6ms at playtime 1:36:42.
a04 Audio overlaps for 10ms at playtime 1:38:39.

The different actions I used to extract the audio are listed below. The streams have the same length in hours/mins/seconds, except for the miliseconds. I have listed the miliseconds and the ectraction method used in the list below:

Code:

363 - eac3to straight to w64
363 - eac3to straight to flac
363 - eac3to straight to wavs
363 - eac3to straight to flac then eac3to to wavs
355 - tsmux the playlist, then eac3to to w64
364 - MakeMKV to mkv, then eac3to to w64
309 - eac3to to dtsma, then eac3to to dts (core)
309 - eac3to straight to dts (core)

When I extract the video part by itself via eac3to (.264) and then mux with tsmuxer to .m2ts, the miliseconds are 272. Extracting the video with eac3to to .mkv results in the miliseconds being 314.

Does anyone know, which one is actually right???

dansrfe · 1st November 2010, 19:50

This has already been discussed before. When extracting DTS-HDMA to wavs eac3to does NOT fix the delays as reported. Hopefully madshi will come out with an update soon *fingers crossed*. For now you can encode to FLAC then demux the wavs because according to an earlier poster, eac3to DOES fix the delays when encoding from DTS-HDMA to FLAC however I'm still not very keen with the idea and want to see a real fix.

mindbomb · 2nd November 2010, 15:49

In my experience, when converting truehd to .wav, it always reduces bitdepth from 24 to 16 with an additional pass.
So are truehd encoders required to output 24 bit regardless of original or something of that nature?
there isn't a bug with decoding, is there?

dansrfe · 2nd November 2010, 15:54

The truehd streams are padded to 24 bit because, as you said, "truehd encoders [are] required to output 24 bit regardless of original" is true. eac3to doesn't seem the need to leave the padding so it removes it and leaves the job to the decoder to pad it back and output it.

setarip_old · 7th November 2010, 00:59

@Murkurial

Hi!

Are you talking about the "Machete" DVD that came out in 2007, or the one that won't be released until January 2011?

Murkurial · 7th November 2010, 01:30

Alright, I'll give it another shot.

I would simply like to know if anyone has discovered a solution to the problem discussed here? The inability of eac3to to properly identify and convert DTS-MA audio.

jj666 · 7th November 2010, 15:48

Murkurial, that problem was fixed in the last version of EAC3TO, you can process .DTSHD created with DTS Master Audio Suite directly. Recommended is to remove -21ms from the start of the file to stay in sync with the original.

Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with. I don't see his original post to see what he was asking.

Cheers,

-jj-

setarip_old · 7th November 2010, 23:29

@jj666

Hi!

Quote:

Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with.

Snce you've specifically addressed me, if your source of information for your statement is the following remark in Wikipedia:

Quote:

Home video release
Director Robert Rodriguez has expressed in an interview that a more violent director's cut will be released on home media.

The R-rated theatrical version of Machete was released in Russia on DVD and Blu-Ray on September 30, 2010. Because of licensing reasons, these releases only contain Russian and Ukrainian audio and no English-language track.

I'd suggest you read between the lines regarding "Because of licensing reasons." Keep in mind that the first official showing of this movie anywhere was on September 3rd.

Additionally, since you addressed this to me, I did a rather thorough Googlesearch and could find no legitimate sources selling the DVD or Blu-ray discs. If you are aware of any legitimate sources, please be good enough to post them...

Murkurial · 8th November 2010, 06:25

Quote:

Originally Posted by jj666

Murkurial, that problem was fixed in the last version of EAC3TO, you can process .DTSHD created with DTS Master Audio Suite directly. Recommended is to remove -21ms from the start of the file to stay in sync with the original.

Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with. I don't see his original post to see what he was asking.

Cheers,

-jj-

Thanks for that. Can eac3to be used to cut that 21ms from the file or would I need another program to do that? And if you could, how would I do that?

jj666 · 8th November 2010, 08:15

Quote:

Originally Posted by Murkurial

Thanks for that. Can eac3to be used to cut that 21ms from the file or would I need another program to do that? And if you could, how would I do that?

EAC3TO input.dtshd output.dts -21ms should do what you need.

Cheers,

-jj-

follz20 · 8th November 2010, 08:17

Can someone help me out with a query I have?

When I rip an lossless audio track (LPCM or TrueHD) to individual channel .wavs, why is the output bit rate always constant? I've done it multiple times for different movie tracks, but the bit rate is always at 768 Kbps and the output file size is exactly the same for each channel - which I find highly confusing. Shouldn't the bit rate be variable and the output file size for each channel differ greatly?

I'm probably just confused on this, so any help/info would be very much appreciated.

I just want to make sure no information is discarded in the process - I want know for sure that the .wavs are lossless (ie identical to the source).

23rd October 2010, 17:43	#10665 \| Link
Thunderbolt8 Registered User Join Date: Sep 2006 Posts: 2,197	when having a 32-bit float wavepack file and I want to convert it losslessly to .flac, how to do that best? when converting what 32-bit float file to .wav and then to flac (because wavepack input is not supported by eac3to), then I get the message "reducing bithdepth from 64 to 24-bits" and the final flac file will have 24-bits which is not lossless any more, right? same when I convert the 32-bit float file to 32-bit non floating .wav (is this step still lossless btw?) and then to flac, then I get "reducing bitdepth from 32 to 24 bits" and the final flac file isnt lossless any more either, correct? if so, why is the creation of 32-bit flac files not supported? according to wikipedia, flac supports up to 32-bit: "It can handle any PCM bit resolution from 4 to 32 bits per sample" Last edited by Thunderbolt8; 23rd October 2010 at 17:48.

25th October 2010, 09:23	#10666 \| Link
flyingernst Registered User Join Date: Oct 2008 Posts: 34	Hi guys. I have a problem System Win 7 x64 + newest eac3to + Arcsoft tmt3 3.0.1.180 I cant encode a .dtsma to flac because eac3to didn´t find the arcsoft decoder. What did I wrong? Thank, greetings, flyingernst Edit: I tryed this: http://forum.doom9.org/showpost.php?...postcount=9076 but I get an error edit edit: i installed Version 2, now it works __________________ Popcorn Hour B-110 Onkyo TX-SR 805 Last edited by flyingernst; 25th October 2010 at 10:45.*

27th October 2010, 11:30	#10668 \| Link
mrr19121970 Registered User Join Date: Mar 2008 Posts: 305	It will extract the core, to be 100% sure of this add -core. Code: eac3to 1) 3: d:\audio.ac3 -core __________________ Blu-Ray Movie Only Copies : Clown_BD Blu-Ray Movie Only DVD Conversion : Clown_BD Blu-Ray Copies With Menus : Clown_BD BD Copier

7th November 2010, 00:59	#10674 \| Link
setarip_old Registered User Join Date: Aug 2005 Posts: 16,267	@Murkurial Hi! Are you talking about the "Machete" DVD that came out in 2007, or the one that won't be released until January 2011? Last edited by setarip_old; 7th November 2010 at 04:14. Reason: Omitted title

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23rd October 2010, 12:32	#10664 \| Link
Thunderbolt8 Registered User Join Date: Sep 2006 Posts: 2,197	just tick "display all 7.1 channel layouts" under preferences

1st November 2010, 19:50	#10671 \| Link
dansrfe Registered User Join Date: Jan 2009 Posts: 1,210	This has already been discussed before. When extracting DTS-HDMA to wavs eac3to does NOT fix the delays as reported. Hopefully madshi will come out with an update soon fingers crossed. For now you can encode to FLAC then demux the wavs because according to an earlier poster, eac3to DOES fix the delays when encoding from DTS-HDMA to FLAC however I'm still not very keen with the idea and want to see a real fix.

2nd November 2010, 15:49	#10672 \| Link
mindbomb Registered User Join Date: Aug 2010 Posts: 576	In my experience, when converting truehd to .wav, it always reduces bitdepth from 24 to 16 with an additional pass. So are truehd encoders required to output 24 bit regardless of original or something of that nature? there isn't a bug with decoding, is there?

2nd November 2010, 15:54	#10673 \| Link
dansrfe Registered User Join Date: Jan 2009 Posts: 1,210	The truehd streams are padded to 24 bit because, as you said, "truehd encoders [are] required to output 24 bit regardless of original" is true. eac3to doesn't seem the need to leave the padding so it removes it and leaves the job to the decoder to pad it back and output it.

7th November 2010, 01:30	#10675 \| Link
Murkurial Registered User Join Date: Aug 2007 Posts: 10	Alright, I'll give it another shot. I would simply like to know if anyone has discovered a solution to the problem discussed here? The inability of eac3to to properly identify and convert DTS-MA audio.

7th November 2010, 15:48	#10676 \| Link
jj666 Registered User Join Date: May 2003 Posts: 153	Murkurial, that problem was fixed in the last version of EAC3TO, you can process .DTSHD created with DTS Master Audio Suite directly. Recommended is to remove -21ms from the start of the file to stay in sync with the original. Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with. I don't see his original post to see what he was asking. Cheers, -jj-

8th November 2010, 08:17	#10680 \| Link
follz20 Registered User Join Date: Jan 2010 Posts: 19	Can someone help me out with a query I have? When I rip an lossless audio track (LPCM or TrueHD) to individual channel .wavs, why is the output bit rate always constant? I've done it multiple times for different movie tracks, but the bit rate is always at 768 Kbps and the output file size is exactly the same for each channel - which I find highly confusing. Shouldn't the bit rate be variable and the output file size for each channel differ greatly? I'm probably just confused on this, so any help/info would be very much appreciated. I just want to make sure no information is discarded in the process - I want know for sure that the .wavs are lossless (ie identical to the source).