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Old 12th April 2012, 14:32   #11601  |  Link
frumble
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Thank you for your help! So I made WAV files with eac3to and want to convert them to WavPack, but the Encoder says: "can't handle .WAV files larger than 4 GB (non-standard)!" - sadly booth streams are over 4 GB. Also Monkey's audio can't encode them: "Error: 1002".
Export to AC3 with ea3to works fine but I absolutely want a lossless codec (it feels better ). In my desperation I even tried to merge the movie MKV with the WAVE files but MKVmerge GUI gives the error "87".
Both WAVE streams seams to be a little bit strange: Non of my audio players can play more than a few minutes (even they display only a few minutes of playtime) and also Audacity can only show me a few seconds!
The link from @Sparktank is very confusing for me. Maybe I try it later.
Downmixing the stream would be the last option but I would be very pleased if it's possible to preserve the original channels...
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Old 12th April 2012, 15:10   #11602  |  Link
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^ @ frumble:

1) forget Monkey Audio, it doesn't support multichannel ;

2) try "wavpack --help" in the command prompt
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Old 12th April 2012, 15:18   #11603  |  Link
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Quote:
Originally Posted by frumble View Post

Off this topic...

It's really great that there are freeware tools like eac3to but I can't understand why the authors of such media helpers don't make the source code available...
But since 2010 no progress with eac3to. In this time the program could have been matured and bugs could have been fixed from others but they can't do it because they don't have the source code.
Sadly you're not entirely correct. Open source-code is NO guarantee that there will ever be other capable programmers interested in fixing the bugs of the software. Take a look at MaestroSBT, K-Meleon, mpeg2enc, mkisofs, etc.
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Old 12th April 2012, 16:04   #11604  |  Link
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@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
You are right, the fact that a software is OSS doesn't mean that there are other developers capable with the task. But look at ffmpeg/libav, x264, VLC and mplayer in the media field. It's not soo bad. And consider the chance @madshi, eac3to's author can't work on freeware projects anymore or dies. Then the community would be forced to rewrite the tool from scratch. OSS is freedom and opportunity, not constraint.
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Old 12th April 2012, 16:16   #11605  |  Link
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Quote:
Originally Posted by frumble View Post
@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
...
Code:
-i                      ignore length in wav header
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Old 12th April 2012, 22:35   #11606  |  Link
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eac3to bug when decoding DTS-ES 6.1 using libav

Hi, I am decoding a DTS-ES 6.1 soundtrack to wav format. If I use the Sonic decoder then everything works correctly. However if I use libav, then I end up with a wav file with a corrupt channel mask. It looks like eac3to is not taking into account libav's lack of back channel decoding:

Code:
eac3to j:\backup 1) 4: c:\temp\orig.wav -libav
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 1:38:52, 24p /1.001
1: Chapters, 20 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB
5: Subtitle (PGS), English
6: Subtitle (PGS), French
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), English
a04 The libav DTS decoder doesn't decode the back channels.
a04 Extracting audio track number 4...
a04 Removing DTS dialog normalization...
a04 Removing XCh extension...
a04 Decoding with libav/ffmpeg...
a04 Reducing depth from 64 to 24 bits...
a04 Writing WAV...
a04 Creating file "c:\temp\orig.wav"...
I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1). This confuses the heck out of any software that attempts to play it

I thought using -down6 might force eac3to to work properly, but this ended up with the same problem. So I think the solution is for madshi to update eac3to to write the correct channel mask when libav decodes 6.1 (and probably 7.1) DTS files.
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Old 13th April 2012, 03:05   #11607  |  Link
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Quote:
Originally Posted by MuteyM View Post
I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1).
Yes, is a know bug.
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Old 13th April 2012, 18:41   #11608  |  Link
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@Midzuki Thank you very much! The conversion worked with all streams and the end MKVs play nice with the WavPack codec!
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Old 13th April 2012, 18:51   #11609  |  Link
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^ Glad to see it worked

Now let's spread the word,
WavPack is the future!
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Old 20th April 2012, 00:31   #11610  |  Link
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Is it possible to convert TrueHD to DTS-HDMA (TrueHD > WAVS > DTS HD Encoder) and keep the same sound for the TrueHD tracks that apply DRC to improve the sound? (at least I think it's the DRC that's doing this ?).

This problem arises with the Transformers 3 TrueHD 7.1 track (and people tell me Iron Man is probably the only other movie doing the same). The converted DTS-HDMA track does not sound the same as the original TrueHD. Some sounds at different moments in the movie have more presence with the original TrueHD track. It makes the sound better, more dynamic.

I think this is caused by the DRC (?) that's applied differently on the different channels at different moments.

The TrueHD track sounds better with PowerDVD 12 compared to the converted DTS-HDMA, but with MPC I don't seem to hear a difference (this leads me to think that MPC doesn't apply the effects and so it doesn't handle the TrueHD metadata correctly (?).

PCM and DTS-HDMA can't modify the sound at playback time in the way TrueHD does. How can I save the wavs with the full TrueHD sound experience?

To summarize, I would like to convert TrueHD to DTS-HDMA and have it sound as good on hardware receivers and PowerDVD (now with eac3to the audio source quality is the same, but many sounds are lacking presence in different parts of the movie).


(This is the first movie I am having this problem with. It seems the Transformers 3 sound engineers wanted to TrueHD track to be reproduced with improvements applied to the Lossless track it contains - and the whole experience is indeed more enjoyable to my ears with these improvements).



PS: sorry mods, I have opened this thread before, but I think here is the right place as it concerns eac3to

Last edited by Bigmango; 20th April 2012 at 00:52.
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Old 20th April 2012, 10:11   #11611  |  Link
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Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month.

eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.
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Old 21st April 2012, 14:40   #11612  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month.

eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.
Ok so if it's not DRC it's another effect that's applied to the sound. (continued in my other thread).

Regarding eac3to
: Can eac3to be used with another decoder that can save the "complete" TrueHD to wavs? It seems the open source decoders don't fully support TrueHD as they only extract the lossless track without the effects that enhance it by giving more presence to specific sounds at specific times (most movies only play the lossless track as is, so only a handful of movies seem to be concerned by this issue).

Thanks.
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Old 22nd April 2012, 12:13   #11613  |  Link
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eac3to save the complete TrueHD to wav's. Don't exist "effects that enhance it by giving more presence to specific sounds".
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Old 26th April 2012, 11:29   #11614  |  Link
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I'm converting 5.1 AC3 from DVD (25.000 -> 23.976), Is it normal that eac3to remaps channels?

Code:
eac3to f:\al.ac3 g:\al.ac3 -slowdown
AC3, 5.1 channels, 1:26:43, 384kbps, 48kHz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Creating file "g:\al.ac3"...
eac3to processing took 3 minutes, 42 seconds.
Done.
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Old 26th April 2012, 15:49   #11615  |  Link
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Yes, internal AC3 channel order is different than standard channel order used to resample and send to encoder.
Don't worry.
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Old 5th May 2012, 22:22   #11616  |  Link
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Hello, I am having the following problem:
Code:
eac3to v3.24
command line: "c:\Program Files\Utils\eac3to\eac3to.exe" "track01.dts" "track01.wav" -libav -simple
------------------------------------------------------------------------------
VOB, 1 audio track, 0:08:57
1: DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz
Track 1 is used for destination file "track01.wav".
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Creating file "track01.wav"...
[a01] Clipping detected, a 2nd pass will be necessary.  <WARNING>
[a01] Starting 2nd pass...
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Applying -0.73dB gain...
[a01] Creating file "track01.wav"...
eac3to processing took 1 minute, 27 seconds.
Done.
The wav file seems fine, I can see it plays all 5.1 channels
But look at the file sizes:

Code:
>>dir track01*.*
 Volume in drive K is Music
 Volume Serial Number is 884C-4DCD

 Directory of K:\DTS

29/04/2012  14:09       105,261,056 track01.dts
05/05/2012  23:54               987 track01 - Log.txt
05/05/2012  23:54       463,804,460 track01.wav
               3 File(s)    569,066,503 bytes
               0 Dir(s)  88,401,051,648 bytes free
Is this correct, WAV has a factor of 4.5 times the DTS size?
I don't know, other WAV DTS files I have are much smaller (given the average MB/min of audio, this one is 8:56 min)
Is there any command line switch to keep the WAV size smaller?
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Old 6th May 2012, 02:49   #11617  |  Link
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Quote:
Originally Posted by ilomambo View Post
...
Is this correct, WAV has a factor of 4.5 times the DTS size?
eac3to/libav decode your DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz to a WAV 5.1 channels, 24 bits, 48kHz with a bitrate of:
6 channels x 24 bits x 48KHz = 6912 kb/s

6912/1510 = 4,57

Then, yes, is correct.

Quote:
Is there any command line switch to keep the WAV size smaller?
You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05
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Old 7th May 2012, 09:53   #11618  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05
Thanks.

I am dumb regarding the technical details. I think the DTS file has the same information as the final WAV, that's why the increase in size seemed too much.
But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it?

EDIT:
I just looked in another song I have in DTS 5.1 WAV format (in the AC3 filter properties while it was playing) and it showed this:

Code:
Decoder:
Stream format: DTS 3/2.1 (5.1) 44100Hz
Bitstream type: 14bit low endian
Frame size: free format
Samples: 1024
Bitrate: unknown
SPDIF stream type: 0xc
Frame interval: 4096
Actual bitrate: 1411kbps
DTS
speakers:  3/2.1 (5.1)
sample rate: 44100Hz
bitrate: 1411kbps
stream: 14bit LE
frame size: 3584 bytes
nsamples: 1024
amode: 9
No CRC
Tebasuna51, If I follow your math 6ch x 14bit x 44KHz = 3696 kbps != 1411 kbps reported by AC3
It is 13:40 min song and the file only takes 141MB
Something is not fitting, according to my understanding

On the other hand when I play the file created by eac3to, AC3 filter shows this:

Code:
Input format: PCM24 3/2.1 (5.1) 48000
User format: PCM16 - 0
Output format: PCM16 3/2.1 (5.1) 48000
That's why I think I am using the wrong tool. I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.

Last edited by ilomambo; 7th May 2012 at 10:13.
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Old 7th May 2012, 11:33   #11619  |  Link
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Quote:
Originally Posted by ilomambo View Post
I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.
WAV is not a wrapper, it is a format, just like MPEG-2 and MPEG-4 are video formats, that get wrapped into various containers. Neither DTS or WAV are containers.

DTS is a compressed audio format, while WAV is not. Hence the reason that the WAV file is larger after you uncompressed the DTS file.
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Old 7th May 2012, 13:45   #11620  |  Link
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Quote:
Originally Posted by MrVideo View Post
...

DTS is a compressed audio format, while WAV is not.
But yes, .WAV is a RIFF-based container, and may contain compressed audio. Regarding DTS-in-WAV especifically, there are two types, 1) normal, without SPDIF-padding, and with a .dca TwoCC, and 2) hacky, with SPDIF-padding, disguised as stereo PCM @ 32 / 44.1 / 48 kHz.
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