eac3to - audio conversion tool [Archive] - Page 45

madshi

24th December 2007, 10:24

A little off-topic - something strange about Nero Audio Decoder 2 and Blu-ray Disc Dolby TrueHD audio tracks.

I built the following graph with graphedit:

File source (BD m2ts file) ---> Nero Splitter ---> Nero Audio Decoder 2 ---> Renderer.

The graph plays fine, but when I go to the Nero Audio Decoder 2 settings it reports that the bitrate is 640kbps (so I guess it basically gets the DD core).

Is this a mistake or is it really plays the DD core instead of TrueHD audio ?

Thanks.

Edit: I think the problem is even worse.

I took a look at two BD movies with Dolby TrueHD track using TSRemux and it recognized all of the audio streams, including the TrueHD track.

But in Nero splitter and Haali it only shows 4 audio tracks, where there are five actually, so this makes me think that the problem is the splitter.

Which splitter can recognize Dolby TrueHD tracks ?
I'm not sure, maybe the Sonic one. Anyway, that doesn't really belong into this thread. BTW, the Nero Audio Decoder doesn't like Blu-Ray TrueHD streams (because of the interweaved AC3 frames). So even if you found a splitter which exports TrueHD you can still not easily decode it with Nero. Except by using eac3to, of course.

Beastie Boy

24th December 2007, 11:12

Wow, this tool has evolved! Many thanks Madshi for your work on this. It is very much appreciated.

Since the latest update, is it still worth buying the Nero plugin? I have bought a copy of Nero 7 with the intention of adding the plugin, but it seems as though I needn't bother.
What is your recommendation?

Cheers, Beastie.

nautilus7

24th December 2007, 11:18

You "need" nero/plugin if you care about best quality in e-ac3/ac3 decoding, because nero decoder is a reference one and gives max quality.

If not, the free ffmpeg decoder is almost mature and can be used instead. The choice is up to you, i think.

Beastie Boy

24th December 2007, 11:21

Thanks for the reply. Nero it is then as quality is the most important thing for me.

Cheers, Beastie.

TripleH

24th December 2007, 13:09

I'm not sure, maybe the Sonic one. Anyway, that doesn't really belong into this thread. BTW, the Nero Audio Decoder doesn't like Blu-Ray TrueHD streams (because of the interweaved AC3 frames). So even if you found a splitter which exports TrueHD you can still not easily decode it with Nero. Except by using eac3to, of course.

OK. I asked that question because I want to avoid from re-encoding the audio.

Is there any quality loss when going from HD Audio (Nero decoder for DDP/TrueHD and Sonic for DTS-HA MA) to FLAC ?

Thanks.

nautilus7

24th December 2007, 13:38

OK. I asked that question because I want to avoid from re-encoding the audio.

Is there any quality loss when going from HD Audio (Nero decoder for DDP/TrueHD and Sonic for DTS-HA MA) to FLAC ?

Thanks.You are a bit confused... :confused:
eac3to can do exactly what you need.

Dolby TrueHD and DTS-HD Master Audio are lossless formats. Flac is lossless too. So any conversion between these 3 formats is without quality loss.
Dolby E-AC3, AC3, DTS and DTS-HD High Resolution are lossy formats. Decode any of them and encode to a different format will result in quality loss.

But what eac3to does, is to minimize the quality loss by using proper decoders (freeware and not freeware) and various "modifications" so they output the best audio quality.

It seems that you 've never read the 1st post of this thread. Do it and i am sure your questions will disappear.

n_response

24th December 2007, 15:36

hello !

I have a "DTS-HD MA 7.1" audio track demux from blue ray dvd.

I want to convent to ac3

I already install nero 7.

but when I use these comaned line

It shows that:

E:\AUDIO>eac3to 01.dtshd ac3.ac3 -640
DTS Hi-Res, 5.1 channels, 4:49:45, 24 bits, 1676kbit/s, 48khz
Decoding DTS-HD track to raw. Please wait...
Find sync word: 7ffe8001
Find sync extension: 3f
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 9.32 GB.
The expected file size for 24 bit is 13.98 GB.
The real file size is 4.38 GB.

And it was fiald!~~

Waiting for the question! And Marry Cristmas!~~

nautilus7

24th December 2007, 17:06

I asked you in the other thread about sonic..
You didn't answer if you have sonic decoders installed and if they are working fine.

Read the 1st post and stop telling that you have nero installed. Nero is not needed in your case.

And you 're waiting for the answer. The question is been made by you.

madshi

24th December 2007, 18:06

The file size of the raw file doesn't seem to fit.
Pleae update your eac3to version. That error message you've posted is from eac3to v1.x. We're already at v2.x now.

madshi

24th December 2007, 18:14

eac3to v2.12 released

http://madshi.net/eac3to.zip

thanks to Ron/drmpeg for all his help

* video resolution, framerate and mode (progressive/interlaced) are displayed
* rewriting timestamps should now always write the correct framerate
* after a full EVO/VOB processing the number of video frames is shown
* EVO 16 bit and 24 bit LPCM demuxing supported now (need samples for 20 bit)
* (E-)AC3 bitstream can be delayed now (similar to delaycut)
* DTS bitstream can be delayed now (similar to delaycut)
* DTS-HD High-Res and Master Audio bitstream can be delayed now
* when demuxing bitstream audio tracks from EVO delay is automatically applied
* some little bugs fixed

nautilus7

24th December 2007, 18:29

Pleae update your eac3to version. That error message you've posted is from eac3to v1.x. We're already at v2.x now.Oh my God!!! :angry:

I couldn't imagine that!

nautilus7

24th December 2007, 18:31

Thunderbolt8

24th December 2007, 19:08

looks great, thanks you two :thanks:

does that mean that eac3to already delays those eac, dts etc. files now automatically (if needed), as with the all the others e.g. truehd, or do we have to do that manually, when taking it from a HD DVD source for example? (am asking because you wrote 'can do now' and not 'does automatically' or something like that)

edit: I dont know exactly what bitstream audios are, but I guess that are those eac3, dts track I was talking about, correct? so according to your update it can do/does both now, automatically if needed and manually, when we feed it with such a track?

nautilus7

24th December 2007, 20:04

I dont know exactly what bitstream audios are, but I guess that are those eac3, dts track I was talking about, correct? so according to your update it can do/does both now, automatically if needed and manually, when we feed it with such a track?
You 're right!

rickardk

24th December 2007, 20:38

Been remuxing HD DVDs all dayt today with not a single error using eac3to. Thanks again for this tool. You have done a fantastic job!

madshi

24th December 2007, 23:01

Merry Cristmas and a happy new year to you too Madshi. :D
Thanks. Same to you and all other Doom9 hotties! :D

madshi

24th December 2007, 23:06

does that mean that eac3to already delays those eac, dts etc. files now automatically (if needed), as with the all the others e.g. truehd, or do we have to do that manually, when taking it from a HD DVD source for example?
Basically eac3to can now apply any delay on any audio track. The only exception is if the source is TrueHD and if you want to keep the audio as TrueHD. That's the only situation where eac3to cannot apply a delay. But as soon as you recode TrueHD to anything else, again eac3to can apply a delay.

am asking because you wrote 'can do now' and not 'does automatically' or something like that
If you feed eac3to EVO files you don't need to care about audio delays, anymore. (Except if you want to keep the TrueHD tracks as external TrueHD files).

Of course that's what I *hope*. It does need to be tested...

madshi

24th December 2007, 23:07

Been remuxing HD DVDs all dayt today with not a single error using eac3to. Thanks again for this tool. You have done a fantastic job!
Thank you. I'm glad it works fine for you. Once you're done with all your HD DVDs, could you please post a movie list here? Also have you had a chance to check whether the audio sync is always correct?

Thunderbolt8

24th December 2007, 23:36

ive tested it with the pianist and bourne identity so far (only main track though). and both, dts-hd ma (pianist) and eac3 (bourne), were fine (though in both cases no delay had to be applied anyway)

Snowknight26

25th December 2007, 00:03

If you feed eac3to EVO files you don't need to care about audio delays, anymore. (Except if you want to keep the TrueHD tracks as external TrueHD files).

Of course that's what I *hope*. It does need to be tested...

So lets say that audio track 1 (DTS) has a delay or 1001ms. If you demux that audio track, and want to mux it with the video track into an mkv container, you don't need to add the 1001ms delay because eac3to automatically applied it when it demuxed it?

n_response

25th December 2007, 00:47

thanks to madshi & nautilus7

I use the last version. It's Done.

Have a nice day~~~~

Biggiesized

25th December 2007, 00:53

Hello, I'm having trouble converting a .dtshd 5.1 and 7.1 48/24 track that I have. I have the Sonic Audio Decoder (4.3.0.169) installed but I always get an error when I run the command line.

"The format of the source file could not be detected."

Any idea why this is happening? I can play the file fine in Windows Media Player. I can upload the track (it's short--about 2:00) if anyone wants to play with it.

By the way, I just downloaded the latest version of this program today. I'm using the GUI, but I could just as easily use the command line version (since I have that too).

nautilus7

25th December 2007, 01:37

So lets say that audio track 1 (DTS) has a delay or 1001ms. If you demux that audio track, and want to mux it with the video track into an mkv container, you don't need to add the 1001ms delay because eac3to automatically applied it when it demuxed it?

Yes, that's right. Unless there's a bug in the new version. :p
But you have a very big delay value, so you'll be able to notice if something wrong.

nautilus7

25th December 2007, 01:41

Biggiesized

25th December 2007, 02:10

Yes, I'm positive it's a DTS-HD file--otherwise it wouldn't have had a .dtshd extension and play correctly.

I'm uploading the 5.1 mix since it's only 20 MB. I'll upload the 7.1 longer mix later if necessary(it's about 90 MB).

http://www.sendspace.com/file/f7bw5d

EDIT: I'm trying to transcode it to .flac or .wav or even .ac3 if I can get it to work.

nautilus7

25th December 2007, 02:37

Your file is a dts-hd track that has only the dts core with no extensions in. This means it's a normal dts file.
Except from the "garbage" it has in the beginning and the end. That's why eac3to can't recognize it.

Change the extension to .dts and use latest delaycut (v1.3.0.0) to fix the track (remove the garbage). Then it 'll work. I guess the same applies to the 7.1 track you have.

rickardk

25th December 2007, 02:57

Thank you. I'm glad it works fine for you. Once you're done with all your HD DVDs, could you please post a movie list here? Also have you had a chance to check whether the audio sync is always correct?

Yes I will post a list...done with 25 titles so far (using two computers) without a single problem (Goodfellas is the exeption, but I guess the disc is corrupt. Will try a fresh disc on thursday).

I'm doing a fast check on audio sync (5 minutes from start and 15 minutes from end). The TrueHD track on Training Day was out of sync. BUT it's out of sync when played back in PowerDVD also. So I guess it's a bad authoring.

I hope to go through the whole collection before the end of the year.

Then I will move on to the Blu-ray titles. I don't really know how to handle audio delay on movies spread over multiple m2ts files yet. But I guess that's a headache to handle later on.

One thing that's bother me though is if I should use 23.976 as timecode or 24000/1001.

I can't get perfect smooth playback on some titles (don't know if the timecode provided when muxing the mkvs will make any diffrence). And I'm not sure if it's going to influence audio sync at all (tiny diffrence),

My TV (Pioneer LX608) accepts 24p. And the combination of this TV and a nvidia 8600gts (Vista 32) I can output 23, 24, 25, 29, 50, 59, 60.

When using 59 (I guess it's 59.97Hz) the audio falls behind (don't know if it's related to the 3:2 pulldown).
Choosing 23 (don't know how exact this is but I guess 23.976Hz) results in perfect audio sync but some titles stutters sometimes.

Does the timecode I set when remuxing affect this in anyway?
Right now I'm starting to believe that 23.976 gives the best result (over 24000/1001) but I'm not sure if it's just my imagination.

I know it's the wrong place for this discussion. But it would be great to hear if someone know how the timecode affects audio sync AND smooth video playback.

rickardk

25th December 2007, 03:04

nautilus7

25th December 2007, 03:04

nautilus7

25th December 2007, 03:07

Where do I find delaycut v1.3.0.0? Can just find v1.2.1.2.
v1.3.0.0 is made by madshi (not the original author of delaycut), thus it can be found on this forum, in delaycut thread (use search).

EDIT: here's the link http://madshi.net/delaycut.rar

rickardk

25th December 2007, 03:09

I have to ask this, sorry...
Does such thing as 23,976 really exist? Where does it come from? I believe it's 24000/1001(23,9760XXXXX) that is called 23,976 in sort. Nothing more. But maybe i just said a nonsense.

Don't know. 24000/1001 is the standard for what most titles are mastered at I think. But I guess it's a rounding thing. And that's why I ask. If it will make any diffrence at all on smooth video playback and audio sync.

nautilus7

25th December 2007, 03:17

Exactly. So when 23,976 is used instead of 24000/1001 is wrong. Of course the difference is tiny, but there shouldn't be a debate whether the one or the other should be chosen.

rickardk

25th December 2007, 03:22

Exactly. So when 23,976 is used instead of 24000/1001 is wrong. Of course the difference is tiny, but there shouldn't be a debate whether the one or the other should be chosen.

Exactly what I think. But why does some titles playback perfect when timecode are set to 23.976 but NOT when set to 24000/1001. That's what I can't accept. One example is The Last Samurai HD DVD.
I guess I have a bad understanding in how the frames are handled...

nautilus7

25th December 2007, 03:32

You mean .mkv? Don't have a clue. Maybe an audio problem.
I have this hd dvd, maybe i try something tomorrow.

rickardk

25th December 2007, 03:40

You mean .mkv? Don't have a clue. Maybe an audio problem.
I have this hd dvd, maybe i try something tomorrow.

Yes mkv... Syriana is another title with the same problem (?).

madshi

25th December 2007, 11:23

One thing that's bother me though is if I should use 23.976 as timecode or 24000/1001.
The differerence is extremely small. But the more "correct" value is 24000/1001 and that's what the latest eac3to version is using.

I can't get perfect smooth playback on some titles
But audio sync is correct? This could be a software setup related problem.

My TV (Pioneer LX608) accepts 24p. And the combination of this TV and a nvidia 8600gts (Vista 32) I can output 23, 24, 25, 29, 50, 59, 60.
23 and 59 sound strange to me. It's strange to name "23.976" as "23". But who knows... Maybe PowerStrip would be worth a try for you?

Choosing 23 (don't know how exact this is but I guess 23.976Hz) results in perfect audio sync but some titles stutters sometimes.
You mean the video stutters sometimes? Is this with MPEG2, VC-1 or h264 (or with all three)? Maybe your PC is not fast enough to handle some of those movies?

Does the timecode I set when remuxing affect this in anyway?
You set a timecode when remuxing? How/where?

buzzqw

25th December 2007, 15:05

just playing with this tools.. great work!

just to add to wisth list
output to stdout (for using with neroaac/lame/oggenc...)
parsing of ts files

thanks!

BHH

nautilus7

25th December 2007, 15:18

I was about to ask for mp3, aac, vorbis support.

I know this tools focuses on HD DVD/Blu-ray audio, but these formats are useful to convert the audio commentaries to.

madshi

25th December 2007, 15:50

Which format do these tools expect via stdout?

nautilus7

25th December 2007, 16:36

I guess the q is for buzzqw. Don't know what stdout is anyway. :D

rickardk

25th December 2007, 16:37

But audio sync is correct? This could be a software setup related problem.

Yes it's in sync. I really think it's a frame rate/refresh rate missmatch problem.

23 and 59 sound strange to me. It's strange to name "23.976" as "23". But who knows... Maybe PowerStrip would be worth a try for you?

Powerstrip does not work well with the new line of nvidia cards. Don't know why they are naming the refresh rates like that.

You mean the video stutters sometimes? Is this with MPEG2, VC-1 or h264 (or with all three)? Maybe your PC is not fast enough to handle some of those movies?

It looks like video drops some frames (or not sync them with the refresh rate). The EVR renderer says no frames are dropped though. It's a very very subtle stutter. I can't spot it when I feed my display with 59.97 (the 3:2 pulldown mask small problems). I have a Q6600 at 3GHz. Should be enough. Yes I have seen this with all three codecs. The strange thing is that by setting a less exact (23.976) timecode thoose titles will playback perfect.

The example mentioned above The Last Samurai will have small micro stutters like once every minute when played back with the timecode for the mkv set to 24000/1001 (or if I play back the disc in PowerDVD). But when I rewrite the mkv with the timecode set to the less exact 23.976, the mkv plays perfect smooth all through the movie.

Don't know how the renderer is working but it may be correcting something to match audio and video sync when the timecode is set to 24000/1001.

To make it even more strange. NO title (yet) is experience any problem (the other way around) by using the less exact timecode.
So it may be better to have eac3to use 23.976 by default.

If someone (who like me have a 24p capable TV or projector and a graphic card that can output 24p) can test this it would be great!!

Done so much testing on this tonight that my eyes are bleeding...

You set a timecode when remuxing? How/where?
Don't know how to set 24000/1001 with a tmecode file in mkvmerge (can just set decimal values...any ideas?).

But to be able to compare I let eac3to set the timecode to 24000/1001 (don't know how you do this). Then I make a copy and run it through mkvmerge with a timecode file loaded with assume 23.976.

In mkvinfo the default duration for video frame shows the exact same value for mkvs with 24000/1001 and 23.976. I assumed that this would tell me that they should playback in the exact same way. But I guess I REALLY don't understand how the frames are handled.

For testing of Blu-rays I compare playback of the original m2ts with a mkv with timecode set to 23.976.

madshi

25th December 2007, 18:20

I guess the q is for buzzqw. Don't know what stdout is anyway. :D
Yeah, the q was for buzzqw. I could allow eac3to to output audio data through "stdout". If another program is written to accept data through "stdin" eac3to could pass audio data to the other program without having to write the data to a temporary file first.

Thunderbolt8

25th December 2007, 18:26

Don't know how to set 24000/1001 with a timecode file in mkvmerge (can just set decimal values...any ideas?).yes, you can only set decimal values, and not the "/". so what works best is calc, then calculate 24000/1001 and just copy & paste the result into that .txt file (and exchange the comma for a dot!)

madshi

25th December 2007, 18:32

It looks like video drops some frames (or not sync them with the refresh rate). The EVR renderer says no frames are dropped though. It's a very very subtle stutter. I can't spot it when I feed my display with 59.97 (the 3:2 pulldown mask small problems). I have a Q6600 at 3GHz. Should be enough. Yes I have seen this with all three codecs.
If you've seen it with all 3 codecs then it's most probably not caused by a too slow PC.

The strange thing is that by setting a less exact (23.976) timecode thoose titles will playback perfect.

The example mentioned above The Last Samurai will have small micro stutters like once every minute when played back with the timecode for the mkv set to 24000/1001 (or if I play back the disc in PowerDVD). But when I rewrite the mkv with the timecode set to the less exact 23.976, the mkv plays perfect smooth all through the movie.

Don't know how the renderer is working but it may be correcting something to match audio and video sync when the timecode is set to 24000/1001.

To make it even more strange. NO title (yet) is experience any problem (the other way around) by using the less exact timecode.
So it may be better to have eac3to use 23.976 by default.
Well, I'm not so sure about that. Obviously your graphics card is set to 23.976 and not to 24/1.001. Now I don't know why that is the case. Maybe your display is reporting that it wants 23.976 instead of 24/1.001. Or maybe your graphics card isn't able to output 24/1.001, but only 23.976. Either way it could be related to your specific display. Another person with different hardware could have an experience which is exactly the opposite of yours. One thing is for sure: 24/1.001 is more correct than 23.976 (although the difference is really small). If that wasn't the case I'd change back to 23.976 right now. But since 24/1.001 is the more correct value, I don't really like the idea of going back to 23.976 because of one single experience report, only. No offense! I do appreciate the feedback very much.

If someone (who like me have a 24p capable TV or projector and a graphic card that can output 24p) can test this it would be great!!
Good call! We do need testers here...

rickardk

25th December 2007, 20:04

I agree to 100%. But I'm starting to think that 3 decimals may be a maximum of what is used of modern graphic cards. I don't really have a clue, just speculations. But my display (TV) is the latest from Pioneer and should handle 24p perfect.
Actuallt I know it does, because I tested with a stand alone Pioneer Blu-ray a month ago with perfect result.

But what does that mean. I have searched the net all day for spec on what refresh rates modern 24p capable TVs and projectors expects. And that refresh rate that stand alone players actually outputs.

Also if someone know and could explain how the frames are handled by the renderer it would be great. Where does the rounding and calculation on how long to show each frame take placè? Does it take the refresh rate output into tte equation (matching)? I know that the use of reclock was essential before we had EVR.

As I'm going to´remux my whole collection it would be great to do it "right".

madshi

25th December 2007, 20:11

Ok, now if it's one of those new Kuro Pioneers then I might reconsider, since one of those may end up in my home, too, sooner or later... :D

rickardk

25th December 2007, 20:36

Ok, now if it's one of those new Kuro Pioneers then I might reconsider, since one of those may end up in my home, too, sooner or later... :D

It's a 60 inch 1080p Pioneer LX608 (G8 KURO)...
I'm about to test with my ATI2600 now...If it's related to the graphic card. I hope not because PQ is better on my nvidia 8600.

buzzqw

25th December 2007, 20:54

@madshi

Which format do these tools expect via stdout?

varius.. from raw pcm (aften) to WAV file, from mono to multi channels, sample rate can be 8000 to 48000., from 8 to 32 floating bit...
oggenc can parse
OggEnc input files must currently be 32, 24, 16, or 8 bit PCM WAV, AIFF, or AIFF/C files, or 32 bit IEEE floating point WAV. Files may be mono or stereo (or more channels) and any sample rate.
neroaacenc
The file must be in Microsoft WAV format and contain PCM data.
for enc_aacplus can be wav or raw

a messy.. .. anyway 16 bit wav file is accepted by all
if sample rate, bit depth, wav or raw pcm and channels can be specified, any external encoder will accept

about ts files (both mpeg and avc) ?

thanks for your interest

BHH

tebasuna51

25th December 2007, 20:59

Which format do these tools expect via stdout?

Same than wav files with standard header and:

Lame (.mp3): only mono or stereo and int samples 16, 24 or 32 bits (don't support float)

NeroAacEnc (.mp4): support 5.1 and any bitdepth 16, 24, 32 int or 32 float. Support big files > 4GB (with -ignorelength) if the value in field RiffLength is 36 + DataLength.

OggEnc2 (.ogg): same as NeroAacEnc (support for big files not tested)

EDIT: Hello buzzqw, we have crosspost. And yes Enc_aacplus to use CT aac encoder (from winamp) only work with 16 bit int.

MuteyM

25th December 2007, 21:39

Yeah, it seems to be a bug in the libav decoder. My guess is that the decoder believes that the truehd stream is done and finished and then surprisingly there's more truehd data coming in. And the decoder doesn't seem to like that. Should be easy to fix, though. Give the decoder developer a few days. His replies sometimes take a few days, but he always comes back with a fix.

Hi Madshi, since it might take awhile to get libav updated, any chance you can update eac3to to not delete the output file upon libav error?