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Old 23rd July 2010, 01:35   #10361  |  Link
rica
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Quote:
Originally Posted by deado View Post
Using Nero 7 with Win 7 64 here no problems.
Do you mean eac3to can use nero's forced decoder?



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Old 23rd July 2010, 07:37   #10362  |  Link
Snowknight26
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Quote:
Originally Posted by madshi View Post
How can I reproduce this? I've just tried "eac3to some.wav 1.aac 2.aac" and both AAC files are created just fine.
Was able to find a small enough sample to be able to reproduce the issue:

http://stfcc.org/misc/eac3to.aac.issue.zip

Run the bat, wait for eac3to to stop outputting. Might be a while, but it will. Either put the two files in the folder eac3to is in or modify the bat.
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Old 24th July 2010, 16:10   #10363  |  Link
tvjunky
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Hi Madshi!

Maybe this is a bug in eac3to:

I always rip my BRs directly with eac3to like this:
eac3to d: 1) -demux

This was functioning for 100% of my BRs the last years. With the new version 3.22 i get the following error on ~50% of my discs (example):

Quote:
D:\_hd>eac3to y: 1)
Skipping 66240 bytes to work around damaged source file...
Hint: You can use the option "-fix" to clean up the damage.
The format of the source file could not be detected.
The parameter "-fix" does not help. With eac3to version 3.21 all BRs are ok and can be copied to hd.
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Old 27th July 2010, 19:18   #10364  |  Link
jdawn1
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Quote:
Originally Posted by Atak_Snajpera View Post
the same reason why 24bit is converted to 16bit. Human hearing has limitations.
Word. Personally, I stick with as high as possible source sample rate and bit depth, and usually use lossy encoding. Much more efficient use of space, IMO. Especially when you can use something like AAC. I'd rather have a 5.1 96kHz/24bit (sourced) 96kHz AAC @ 650kbit/s then a 5.1 48Khz/16bit FLAC @ 3000kbit/sec. All that extra superfluous audio data is much better spent on the video. By superfluous, I would personally guarantee that no one here could spot the difference between a "high bitrate" AAC and a FLAC from the same source, on your choice of audio equipment. I am talking an AAC file 1/5 to 1/10th the size of the FLAC.

I am still amazed when I see movie encodes people make with 10GB file size: 5GB of video and 5GB of a DTS-HD-MA audio track. Makes no sense whatsoever. Make it 9.2GB of video and 800MB of audio (AAC for PC playback or DTS/AC3 core if you need to "bitstream") and it's overall much better quality.

Quote:
Originally Posted by Midzuki View Post
Digital: In the world of today, 192kHz audio should already be something quite trivial.

Analog: In the world of today, speakers and audio amplifiers that support up to 96kHz still are very rare, and (IMHO) somewhat pointless. ss......
Yeah. If you mean by frequency response, 96KHz could represent 48Khz frequency audio. Given I can't hear much above 16KHz, yeah, speaker reproduction of 48kHz would be pretty stupid.

Quote:
Originally Posted by dansrfe View Post
Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.
Well, speaker support is irrelevant, they're analog. My cheapie receiver supported it just fine, even the original bitstream (DTS-HD-MA 5.1 192 / 24)
Funnily enough, it actually sounds better if I playback a version of it downsampled to 48KHz as my receiver supports Audyssey but can't process 5.1 streams over 48Khz. The audio benefit of running with Audyssey enabled far exceeded listening to the track at 96 or 192 KHz...
My digital copy of the movie is 96/24 audio encoded to AAC, which I have Windows downsample to 48khz (LCPM over HDMI) so the receiver can enable Audyssey.

Quote:
Originally Posted by Killroy™ View Post
OK... I'll be the first to ask... Why would you want to take such a beautiful audio track and throw away 2/3 of the data?
Because it is a pointless waste of space if it's to be stored on a PC. In my case, resampling from 192 to 96 and encoding to AAC (Nero, -q 0.5) resulted in "throwing away" something like 94% of the data, lol. 13.4 GB to less then 1 GB. Yet, you can't hear a damn bit of difference on a nice audio setup; sounds great. I'd dare you to even try.

It's still a beautiful audio track even if you remove a few samples or use a transparent lossy encode. How it SOUNDS is what's important, and what makes a great score. File size has, and should not have, in anyone's mind, anything to do with it.

Quote:
Originally Posted by asarian View Post
Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?
Of course! Just look at the help. It's really easy. I own the blu-ray, and I make .MKV's for my media server. This is the only movie I had to re-sample. I re-sampled to 96KHz / 24bit and encoded to AAC with Nero, file went from 13+ GB (DTS-HD MA) to less then 1 GB and sounds awesome.

The movie went from taking most of a BD-50 to only about 13.4 GB using the original video and AAC audio. Nothing lost quality wise, lol.

I did keep a copy of that sucker on my server for testing purposes, though. My receiver seems to work just fine with it... a few years old and a cheapie, too. Onkyo 606. Kinda neat to see it say 5.1ch DTS-HD MA 192 khz (bitstreamed from a PC with ATI 5xxx HDMI card). My old PS3 plays it back without a hitch either. 5.1 ch 192 kHz PCM shows up on the receiver, not that I would be able to hear a difference even if it was not fully decoding or down/upsampling, lol.

Last edited by Guest; 28th July 2010 at 02:56. Reason: 6
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Old 28th July 2010, 13:11   #10365  |  Link
lintran
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Hello
I just tried extract DTS-ES Discrete 6.1 to wav(input.dts output.wavs) with Arcsoft Decoder (version 3.0.160) and it produced 7 individual WAV files, but there was something wrong with them. Too much high-frequency and noise.
Same occurs when I try to create an AC3 file.
So please tell me where do i wrong? Or how to do it proper?
Thank you so much.
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Old 28th July 2010, 13:18   #10366  |  Link
Atak_Snajpera
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Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.
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Old 28th July 2010, 18:26   #10367  |  Link
deathlord
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Quote:
Originally Posted by Atak_Snajpera View Post
Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.
Sonic works well on these!

Quote:
Originally Posted by jdawn1 View Post
I re-sampled to 96KHz / 24bit and encoded to AAC with Nero
Why not 48KHz, 96 is a waste of diskspace, too!
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Old 28th July 2010, 18:34   #10368  |  Link
Atak_Snajpera
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Quote:
Why not 48KHz, 96 is a waste of diskspace, too!
because people like in placebo effect
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Old 28th July 2010, 21:40   #10369  |  Link
deathlord
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@ madshi,

Could you allow for longer -edits, in particular silence? At the moment the maximum possible (positive) edit seems to be 4999ms.
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Old 29th July 2010, 07:28   #10370  |  Link
longtom
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I want to extract from a BluRay the 2.0 PCM soundtrack to wav.
The output should be outputchapter1.wav outputchapter2.wav etc...
So that i have all songs in a extra wav files and not in one LARGE output.wav file!
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Old 29th July 2010, 11:23   #10371  |  Link
tebasuna51
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You can use the BeSplit command line, or with CUE_Splitter GUI, to split the full wav file.
For CUE_Splitter you need a .cue file, but maybe you can convert the Chapters file to a cue file easily with a text editor.
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Old 29th July 2010, 12:27   #10372  |  Link
longtom
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Quote:
Originally Posted by tebasuna51 View Post
You can use the BeSplit command line, or with CUE_Splitter GUI, to split the full wav file.
For CUE_Splitter you need a .cue file, but maybe you can convert the Chapters file to a cue file easily with a text editor.
Great, Thanks!
I'll try with Cue_splitter!
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Old 29th July 2010, 12:35   #10373  |  Link
tebasuna51
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WARNING:
- When the language field of a track (in mkv) is empty this mean the language is the default: eng (from MkvMerge v4.0.0)
- eac3to don't manage audio tracks (MP2/MP3/DTS/AC3?) from mkv's generated with MkvMerge v4.1.0 defaults (header compression).
- eac3to crash (without messages) with video track H264 from mkv's generated with MkvMerge v4.2.0 defaults (header compression).

@madshi

Do you have any plan to support these new defaults?
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Old 31st July 2010, 11:41   #10374  |  Link
frank
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Test of mkvtoolnix 4.2.0 header removal compression
(deletes repeating header infos)

Remuxing of a ts Stream to mkv:
8.292.939 kB without header removal compression, like before
8.292.174 kB with Compression

= 0,00923 % difference!!

Wow, what a great saving! Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!

First Aid:
Disable for every track the option Compression: to None.
Not empty.

Last edited by frank; 31st July 2010 at 11:51.
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Old 31st July 2010, 21:08   #10375  |  Link
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@frank

Hi!
Quote:
Wow, what a great saving! Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!
If you read the MKVToolnix thread, you'll find that "bunkus" is adamant that since this is technically correct, it will remain as the far-from-obvious default setting...
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Old 1st August 2010, 21:02   #10376  |  Link
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Does eac3to do encodings to .mp3?
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Old 1st August 2010, 22:51   #10377  |  Link
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dansrfe - Not directly. But you can pipe output to the LAME executable. tebasuna51 has posted a command template several times around here, and I imagine you'll find it pretty easily searching for eac3to lame.exe or something similar,
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Old 4th August 2010, 16:27   #10378  |  Link
mr.duck
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Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?

If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?
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Old 5th August 2010, 13:34   #10379  |  Link
longtom
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Originally Posted by longtom View Post
Great, Thanks!
I'll try with Cue_splitter!
I've splitted it now with foobar!
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Old 6th August 2010, 10:17   #10380  |  Link
LeXXuz
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How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information?

I'd like to make a speedup on an AC3 2.0 Surround f.e.:
eac3to input.ac3 output.ac3 -224 -speedup

However the output file is only AC3 2.0 (when reopening the output file, eac3to reports "AC3 2.0" and not "AC3 2.0 Surround" as for the input file) and my AVR does not show the Surround channel any more.

Is there any switch I need to use to keep the output file as 2.0 surround?
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