eac3to - audio conversion tool

[BLKMGK]

Having an issue with a specific movie - The Game Plan. The video appears to rip just fine but the audio track is a mess :( I have gone after the DTS and the AC3 5.1 tracks to no avail. When I rip it I get normal fine audio and then a few minutes into the track it loops back to the beginning! If I FFWD a bit I get normal dialog again - then a few mins later it loops back to the starting music again. After about the 3rd time I gave up :( I have now ripped it 3 times trying various things to include two copies of the movie to no avail. Has anyone else done this one and what issues if any did you encounter? I am on 3.14 and haven't ever seen this before. Ideas or suggestions?

Here's my log -> http://pastebin.com/m2d93e154

[TinTime]

If I feed eac3to a duff audio file it reports "The format of the source file could not be detected." but the return code is 0. Could it be changed to non-zero please? It seems to be rc 1 for other source file errors.

Thanks.

Following on from this I've been trying to work out why eac3to doesn't like some (but not all) of my input files. I used eac3to to create a wav file from a pcm input:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "test.pcm" "E:\Vtemp\test1.wav" -6 -24 -little -96000 -override -log=1.txt
------------------------------------------------------------------------------
RAW/PCM, 5.1 channels, 1:38:26, 24 bits, 13824kbps, 96khz
Reading RAW/PCM...
Writing WAV...
Creating file "E:\Vtemp\test1.wav"...
The original audio track has a constant bit depth of 24 bits.
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
eac3to processing took 9 minutes, 41 seconds.
Done.

If I then run eac3to on the output wav file I get:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Vtemp\test1.wav" -log=t1.txt
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

I've been trying to find out why this file does not work and other apparently similar ones do and the difference seems to be the amount of silence at the beginning of the file. This file contains 11866608 bytes = 6.86725 seconds of silence at the beginning. I tried editing the first six samples to non-zero values and it's then detected by eac3to.

I've created two test files (http://www.sendspace.com/file/2wi285) (the original is 10GB) containing the first 15MB of the wav file. I've edited the first six samples of test2.wav and it can be processed. I've left test3.wav as is and it can't be detected. The lengths in the header remain as they are from the original file. I found that if I change the lengths in the RIFF and data chunks in test3.wav to zero then eac3to will detect it. However this doesn't work with the full sized original.

Thank you!

[tebasuna51]

Yes, seems a bug.

[Snowknight26]

eac3to can't seem to recognize AAC files it made.

[tebasuna51]

eac3to can't seem to recognize AAC files it made.
Eac3to recognize (and can decode with Nero, without plugins) .aac files. Is AAC format with ADTS headers.

Can't recognize AAC inside mp4 container (.mp4 or .m4a).

The use of the faad decoder (aac/m4a, inside libav) or NeroAacDec is in the todo list of madshi like he say.

Edit: By the moment you can use the Faad decoder included in UsEac3to (http://forum.doom9.org/showthread.php?t=145574)

[LeXXuz]

I have Nero7 on my system to use the e-ac3 decoder. I want to get rid of Nero, because I use other tools for burning stuff and Nero just occupies to much space and resources.

How can I get rid of Nero but keep the e-ac3 decoder and/or the necessary libraries/files? Is that possible? :)

[laserfan]

Easiest just to de- and re-install with only Showtime, taking care to skip the Scout & other unneeded junk, though I think there remains an indexing service you still have to Disable by hand.

[Snowknight26]

Can't recognize AAC inside mp4 container (.mp4 or .m4a).

I hadn't even noticed that the files it was making were .m4a. :o

Thanks for the continued support tebasuna51.

Edit: I just noticed that eac3to displays a 2.0 channel AAC track in mkv as being 24KHz when its really 48KHz.

[DragonM]

Hey, I got a 20gig m2ts file containing several episodes in it. I want to demux the video and audio separately to mkv and flac.

Can I split m2ts files by chapters? If so, how?

[tebasuna51]

Edit: I just noticed that eac3to displays a 2.0 channel AAC track in mkv as being 24KHz when its really 48KHz.

Then is a AAC-HE (with Spectral Band Replication) the high frequencies (24-48KHz) are in the SBR part, not always recognized. With the appropriate decoder you can recover the 48 KHz.

[Snowknight26]

It's AAC LC-SBR, 2 channel, 48KHz according to YAMB.

[tebasuna51]

LC-SBR and HE is the same.

The LC part is 24KHz (and some decoders/players only recognize this) and the high frequency go in SBR part.
Is not easy know when a aac audio file have SBR or not, in the ADTS header from aac the frequency are marqued as 24Khz.

BTW the support for AAC in eac3to is still limited.

[narshorn]

Dear Madshi and all sound addicts @ doom9,

Thanks for this wonderful program eac3to. It's a really handsome tool for audio extraction and conversions !
I really like the command line utility. Plus I like the idea of bit perfect decoding from various formats ... :)

However, I have a small problem when opening .wavs created by eac3to in extraction;
when I try to open some 24/96 or 24/192 wavs in r8brain (wavs created by eac3to), r8brain pops up an error.
It's relative with the wav header I guess. When I open the "defective" wav in Wavelab, I have no problem, so I save the file again
in Wavelab without changing the bitrate or sample frequency, i.e. identical file, but now I can open them in r8brain and downsample
them in very high quality mode ... any thoughts on this ? Should I add a particular switch while doing the .wav extraction from .mlp
to get a r8brain compatible header ? Thanks for your thoughts on this.

Best regards,

narshorn

[kypec]

Try to add -simple switch, it should produce old fashioned WAV header, probably better compatible with r8brain.

More info on undocumented switches of this wonderful program can be found here (http://forum.doom9.org/showthread.php?t=145066)

[narshorn]

Try to add -simple switch, it should produce old fashioned WAV header, probably better compatible with r8brain.

More info on undocumented switches of this wonderful program can be found here (http://forum.doom9.org/showthread.php?t=145066)

Thanks for the info :) will try it in my next conversions :)

Regards,

narshorn

[Pl4yit]

Just wanted to post the following issue that I found using eac3to (v3.14). The sample file that I have used can be found in the link here below.

http://rapidshare.de/files/46162759/extract.h264.html

- The sample is a part of a h264 file extracted directly from blu-ray using eac3to
- When performing the following set of commands on this sample:

eac3to extract.h264 extract.mkv
eac3to extract.mkv 1: extractfrommkv.h264


then you will notice that the result file "extractfrommkv.h264" is different from the original "extract.h264".

Looking a bit more in detail there is a sequence of 162 bytes added at specific places in the file, starting with the following 34 bytes:

00 01 27 64 00 29 AC 7B 01 E0 08 9F 97 FF 00 01 00 01 10 00 00 3E 90 00 0B B8 08 40 00 00 00 01 28 EA

The sequence is the same throughout the file and removing it will give back the original "extract.h264".

The issue has been introduced as of v3.07 (v3.06 works fine), and I think it happens only with AVC/H.264 encoded files with 2 Reframes. Did the test on 4 different films (directly from blu-ray), all with 2 reframes, and they all had the issue, i.e. a lot of artifacts during very fast-moving scenes.
Did some other tests with VC1, MPEG2 and higher-frame H.264 encoded files and did not find a difference between the files before and after the 2 eac3to steps provided here above.

Sorry for the long thread, but I hope this is enough information to easily pin-point the problem, otherwise I'm happy to provide more :D

Pl4yit

[shon3i]

I have problem when i try to demux TsMuxer created Blu-Ray.

M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:24:34, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p23.975 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: Subtitle (PGS), English
v02 The video bitstream is encoded in a non-standard framerate.
v02 The video bitstream framerate field doesn't seem to match the timestamps.

but that is not true, i encode to H264 with x264 and input avs have parametar fps=23.976, all programs and players reconize as 23.976, aslo i tryed with checking changefps in TsMuxer and selecting 24000/1001, but still eac3to report message.

Second question is: why eac3to when extracting chapters not include final time anymore?

[BLKMGK]

Having an issue with a specific movie - The Game Plan. The video appears to rip just fine but the audio track is a mess :( I have gone after the DTS and the AC3 5.1 tracks to no avail. When I rip it I get normal fine audio and then a few minutes into the track it loops back to the beginning! If I FFWD a bit I get normal dialog again - then a few mins later it loops back to the starting music again. After about the 3rd time I gave up :( I have now ripped it 3 times trying various things to include two copies of the movie to no avail. Has anyone else done this one and what issues if any did you encounter? I am on 3.14 and haven't ever seen this before. Ideas or suggestions?

Here's my log -> http://pastebin.com/m2d93e154

Any ideas on this guys? I've never had a soundtrack do this and the loops do not appear to be on chapter or segment changes. I'm totally stuck although I have a few (dozen) old versions of eac3to I may try just for grins to see if this is something goofed recently... So far as I can tell this movie hasn't presented AnyDVD-HD any issues.:confused:

[DrNein]

Easiest just to de- and re-install with only Showtime, taking care to skip the Scout & other unneeded junk, though I think there remains an indexing service you still have to Disable by hand.

Alas, all the Nero Home junk (Scout, Indexing, and such) is installed regardless of all the extras being deselected. But yes, it can all be disabled and deleted manually.

However, for the smallest starting point follow my directions here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

After that, Nero Burning ROM could be deleted to save a measly 20MB, essentially just keeping the Common Files including activation and filters (filters could potentially be trimmed down a bit but they are only 8MB as-is).

[setarip_old]

@BLKMGK

Hi!

It appears that you've used the wrong .MPLS- 00139. You should use 00138. Note the playtimes of each:

D:\Video\eac3to>eac3to x:
1) 00139.mpls, 2:07:03
[92+1+93+3+94+5+95+7+96+9+97+11+98+13+99+15+16+100+18+101+20+102+22+103+24+104+26+105+28+106+30+107+32+108+109+35+110
+37+111+39+112+41+113+43+114+45+115+47+116+49+117+51+118+53+119+55+120+57+121+59+122+61+123+63+124+65+125+67+126+69+127+
71+128+73+129+75+130+77+78+131+80+132+82+133+84+134+86+135+88+136+90+137].m2ts
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00138.mpls, 1:50:15
[0+1+2+3+4+5+6+7+8+9+10+11+12+13+14+15+16+17+18+19+20+21+22+23+24+25+26+27+28+29+30+31+32+33+34+35+36+37+38+39+40+41+
42+43+44+45+46+47+48+49+50+51+52+53+54+55+56+57+58+59+60+61+62+63+64+65+66+67+68+69+70+71+72+73+74+75+76+77+78+79+80+81+
82+83+84+85+86+87+88+89+90+91].m2ts
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, English, stereo, 48khz

[laserfan]

...for the smallest starting point follow my directions here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

After that, Nero Burning ROM could be deleted...But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

[BLKMGK]

@BLKMGK

Hi!

It appears that you've used the wrong .MPLS- 00139. You should use 00138. Note the playtimes of each:



Play time is listed as 110minutes on the box. The dialog is perfectly synched and correct up until the glitch occurs. When this occurs the audio from the very beginning of the movie - the Disney fireworks music - plays for a bit and then it jumps back to the correct point in the movie and dialog\effects\etc. is all correct and in synch. It's as if incorrect portions of the movie are being dropped into the midst of correct portions. Honestly I figured one of those features was say the Director's cut and the other was the release cut. I ripped both of them initially but erased them both and reripped with another copy of the media hoping for a fix, I can rip the 2hour version to see what it is easily enough I guess. The box doesn't mention that there are different cuts and I've nto tried it in my PS3 <sigh> In any case - sumthin' ain't right with the way it's ripping....

[DrNein]

But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

Indeed, it is not from Ahead. But like nLite or such, valid licenses are still required.

[NanoBot]

But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

As DrNein already stated, Nero 7 Micro is not an official product from Ahead. Instead, it is a slimmed down installer provided from a third party. If you do not trust this installer, or if you are in doubt about copyright or license problems using such an installer, you possibly prefer to build your own installer using the NeroLiteSDK available here:

http://updatepack.nl/downloads/nero-lite/

With that SDK you are able to generate your own slimmed down installer from the latest official Nero7 update package.

I am using a similar configuration like DrNein described here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

and I neither have problems to use the Nero EAC3 decoder together with eac3to nor do I have any other problems resulting from the use of Nero7Micro.

C.U. NanoBot

[XhmikosR]

Or you can use this (http://updatepack.nl/) version of Nero Micro English which already has all the necessary files to decode True-HD, E-AC3 etc. Of course, a valid license is needed to use it.
The changes in the above version are not in the "official" Lite/Micro, but hopefully they will be added when the creator of Nero Lite has some free time.

EDIT:
And of course the above setup is clean.

EDIT2:
All of the above have been implemented in the "official" Nero Micro/Lite.

[laserfan]

Or you can use this (http://ikwuva.bay.livefilestore.com/y1pla02NXBT_OjrqNpcCkCEcDyxgSOR8MJP9IdIW-h0cfFbcapxmnusVCoR-v8kSurLZyUgldg4g_tIq25ABndmgw/Nero-7.11.10.0_english_micro.exe) version...The version I bought is a little older but I will give this a try, thanks. What is different about this build vs. "official"?

[XhmikosR]

Just the filters needed for E-AC3, True-HD and DTS Express decoding and some minor improvements to the setup script. I have already contacted the creator of Nero Lite in order to include the changes I've made, but he doesn't seem to have a lot of free time.

[shon3i]

I have problem when i try to demux TsMuxer created Blu-Ray.

M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:24:34, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p23.975 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: Subtitle (PGS), English
v02 The video bitstream is encoded in a non-standard framerate.
v02 The video bitstream framerate field doesn't seem to match the timestamps.

but that is not true, i encode to H264 with x264 and input avs have parametar fps=23.976, all programs and players reconize as 23.976, aslo i tryed with checking changefps in TsMuxer and selecting 24000/1001, but still eac3to report message.

Second question is: why eac3to when extracting chapters not include final time anymore?
Did somebody have answers for my questions?

[Chumbo]

Did somebody have answers for my questions?
Did you try forcing the framerate via a CLI switch? What's the exact command line? Always helpful to have the full command line.

[shon3i]

Did you try forcing the framerate via a CLI switch? What's the exact command line? Always helpful to have the full command line.
Yes i tryed but not helped. I use classic cmd eac3to "k:\b13" "k:\video.mkv"

[Chumbo]

Yes i tryed but not helped. I use classic cmd eac3to "k:\b13" "k:\video.mkv"
Put a sample for madshi to look at. Your command line is not workable. It doesn't know what to do if you don't tell it what play list or track and so on. The exact command line is usually at the top of the log file which is helpful to include with any log entries.

[narshorn]

Any reason to why eac3to with -decodeHdcd switch gets 6 dB gain lower than standard decoding ? If so, is it OK to add +6dB as a switch while doing conversions ?

Best regards,

narshorn

[G_M_C]

People, i've got a question, that's (hopefully) just a theoretical one:

Take, for instance, a 5.1 DTS-HD Master Audio stream that has a bitrate of 6900 kbps, with a core DTS stream 5.1 @ 1536 kbps.

The way i understand the DTS-HDMA system works (simplified), is that a 1536 kbps DTS stream is encoded as the core, and that the additional HDMA data contains the differences between the original stream and the DTS stream. When decoded the DTS + the differences end up being completely identical to the source/original stream.

But if I reason this on this / see this correctly (here's the theoretical bit) this would mean that you don't necessarily have to have "the best core track you can make". You could change / master the core-track marginally different, so that the audience that has to use this core-track has the better listening experience. This intentional slight difference can be made up in the (extended) HDMA part of the stream, where you "iron out" this (intentional) slight differences.

So far the theoretical part of this question. The question i'd like to ask is derived from my assumption that a core DTS-stream doesn't necessarily be the best core-track they could make. Wouldn't it be better to fully decode the complete DTS-MA track, and use that (lossless) track to encode a 1536 DTS track yourself ? As opposed to just demuxing the core track off course ?

[easy2Bcheesy]

Hi,

I'm using eac3to to turn .ac3 files into multi-channel WAVs. Decoding is being handled by libav/ffmpeg. The only problem I have is that I'm only getting 2.1 channel output. The command line text refers to remapping channels and reducing depth from 64 to 24-bit.

How can I make libav/ffmpeg give me all six channels instead of just three?

EDIT: A bit of a false alarm. What appears to be happening is that the WAV is absolutely fine. However, when I import it into Adobe Premiere, I am only seeing waveforms on three of the six channels. If I actually play the WAV, I hear it from the centre and rears exactly as you would expect. Even when I edit the audio, export it and re-import it, I still only see three channels, even though sound is coming from all six. Another Adobe cock-up? Is there a freeware tool out there that will show me waveforms on a six channel WAV?

[TinTime]

You can try Audacity (http://audacity.sourceforge.net/). That will show you all six channels.

[StephenB]

Hi,

Thx for the great tool - I've been using it for about a year now, and am quite impressed.

I'm trying to use EAC3TO to process an 8 bit PCM file. The goal is to resample it to 48 kHz and transcode it.

Where I'm having difficulty is that EAC3TO won't expand the bit depth beyond 8 bits. Here's a simple log:

eac3to v3.14
command line: c:\vid2eva\tools\eac3to\eac3to audio_2_l.wav test.flac -16
------------------------------------------------------------------------------
WAV, 1.0 channels, 0:00:10, 8 bits, 88kbps, 11khz
Reading WAV...
Encoding FLAC with libFlac...
The original audio track has a constant bit depth of 8 bits.
Creating file "test.flac"...
eac3to processing took 1 second.
Done.

If I specify the resampling, I end up with a clipped output that is totally unusable.

eac3to v3.14
command line: c:\vid2eva\tools\eac3to\eac3to audio_2_l.wav test.flac -16 -resampleto48000 -r8brain
------------------------------------------------------------------------------
WAV, 1.0 channels, 0:00:10, 8 bits, 88kbps, 11khz
Reading WAV...
Resampling to 48khz...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Creating file "test.flac"...
The original audio track has a constant bit depth of 8 bits.
The processed audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Resampling to 48khz...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
The processed audio track has a constant bit depth of 24 bits.
Creating file "test.flac"...
eac3to processing took 4 seconds.
Done.

This fails in the same way, no matter what output format I choose. It seems to me that EAC3TO is not expanding the bit depth properly.

If I first convert to FLAC with ffmpeg

ffmpeg -i Audio_2_L.wav -acodec flac -f flac test.flac

ffmpeg will convert the bit depth to 16 bits.

Resampling with EAC3TO to 48000 kHz then works ok - so the problem is not with the resampling itself.

[tebasuna51]

Try with the correct parameter to change the bitdepth, because:

...
-down16 downconvert decoded audio data to 14..23 bit
...
-16 PCM file is '16' or '24' bit

[SomeJoe]

But if I reason this on this / see this correctly (here's the theoretical bit) this would mean that you don't necessarily have to have "the best core track you can make". You could change / master the core-track marginally different, so that the audience that has to use this core-track has the better listening experience. This intentional slight difference can be made up in the (extended) HDMA part of the stream, where you "iron out" this (intentional) slight differences.

So far the theoretical part of this question. The question i'd like to ask is derived from my assumption that a core DTS-stream doesn't necessarily be the best core-track they could make. Wouldn't it be better to fully decode the complete DTS-MA track, and use that (lossless) track to encode a 1536 DTS track yourself ? As opposed to just demuxing the core track off course ?

Well, there's two parts to your question here.

First is your question about using virtually any core track along with the HDMA portion of the stream to end up with lossless audio. Yes, that is correct. Theoretically, I could use a 768Kbps core track, and then use more HDMA information and still end up with lossless audio. However, this is not in the best interest.

Your goal with lossless audio is still to reduce the bitrate as much as you can (i.e. get the best compression ratio while still maintaining lossless audio). The core portion of the track is a sophisticated encoding algorithm that implements perceptual encoding, temporal similarities, etc. The HDMA portion is simple by comparison, and just consists of compensation information. Thus, the more of the information that you can put in the core track, the less information that has to be in the HDMA portion, and therefore the higher the compression ratio.

So yes, you could use a DTS 768kbps as the core and expand the HDMA information and still end up with lossless audio. But the bitrate of that resulting DTS-HD MA track would end up being higher than a DTS-HD MA track made with a 1536kbps core.

Thus, it's in the best interest of reducing bitrate to make the best core possible.

The second part of your question is in regards to how to encode if you want to end up with a lossy 1536kbps DTS track: 1) Extract the core, or 2) encode your own 1536kbps DTS track using the decoded lossless audio.

Well, the core was originally made off the studio master. Assuming the decoded DTS-HD MA lossless audio is equal to the studio master, if you encode your own core (with a licensed, official DTS encoder), you should end up with an identical core.

Thus, I think it would be easier to just extract the given core rather than encode your own.

[StephenB]

Try with the correct parameter to change the bitdepth, because:

...
-down16 downconvert decoded audio data to 14..23 bit
...
-16 PCM file is '16' or '24' bit

Using -down16 results in the same behavior. On the first command (w/o resampling) I still get a bit depth of 8 -

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 10s 99ms
Bit rate mode : Variable
Bit rate : 31.4 Kbps
Channel(s) : 1 channel
Sampling rate : 11.024 KHz
Resolution : 8 bits
Stream size : 38.7 KiB (100%)

With resampling, I still get clipping and an unusable output.

[rik1138]

Here's an easy request:

When I convert a 5.1 stream to WAV files, it names the Surround channels .SL.wav and .SR.wav. The encoding software I use for Dolby and DTS expect .LS.wav and .RS.wav (the software will auto-find all the files in their correct channel locations if they are named like this, and it seems to be the industry standard...) Any chance of swapping those letters in a future update? :)

No big deal, obviously... It only takes a few seconds to rename them after all, I just thought I'd suggest it...

I haven't tried to convert a 7.1 to WAVs (assuming eac3to does that), but the encoding software looks for LSS (Left Surround Side), RSS, LSR (Left Surround Rear) and RSR for the surrounds in that case...

Rik

[tebasuna51]

Using -down16 results in the same behavior. On the first command (w/o resampling) I still get a bit depth of 8
Is the expected behaviour, use the minimum bith depth.

With resampling, I still get clipping and an unusable output.
This is a problem. Seems the resample don't work with 8 bit samples.
Use Audacity, WaveWizard or Sox to change the bitdepth, for instance:
sox mono8-11.wav -2 mono16-11.wav

[StephenB]

This is a problem. Seems the resample don't work with 8 bit samples.
Use Audacity, WaveWizard or Sox to change the bitdepth, for instance:
sox mono8-11.wav -2 mono16-11.wav

Or ffmpeg, which is what I am using in the script already.

Though I have a workaround, it seems to me it is a bug, so I am reporting it.

As far as expected behavior goes, it would be useful to be able to specify that you need a 16 or 24 bit depth, without needing to reprocess with another tool.

BTW Madshi, it would be very friendly to batch files if EAC3TO switched to r8brain automatically when ssrc gets tempermental.

[G_M_C]

Well, there's two parts to your question here.

First is your question about using virtually any core track along with the HDMA portion of the stream to end up with lossless audio. Yes, that is correct. Theoretically, I could use a 768Kbps core track, and then use more HDMA information and still end up with lossless audio. However, this is not in the best interest.

Your goal with lossless audio is still to reduce the bitrate as much as you can (i.e. get the best compression ratio while still maintaining lossless audio). The core portion of the track is a sophisticated encoding algorithm that implements perceptual encoding, temporal similarities, etc. The HDMA portion is simple by comparison, and just consists of compensation information. Thus, the more of the information that you can put in the core track, the less information that has to be in the HDMA portion, and therefore the higher the compression ratio.

So yes, you could use a DTS 768kbps as the core and expand the HDMA information and still end up with lossless audio. But the bitrate of that resulting DTS-HD MA track would end up being higher than a DTS-HD MA track made with a 1536kbps core.

Thus, it's in the best interest of reducing bitrate to make the best core possible.

The second part of your question is in regards to how to encode if you want to end up with a lossy 1536kbps DTS track: 1) Extract the core, or 2) encode your own 1536kbps DTS track using the decoded lossless audio.

Well, the core was originally made off the studio master. Assuming the decoded DTS-HD MA lossless audio is equal to the studio master, if you encode your own core (with a licensed, official DTS encoder), you should end up with an identical core.

Thus, I think it would be easier to just extract the given core rather than encode your own.

I've tried the 2 methods on a DTS-MA track (extract core and remuxing myself). The reencoded channel seemd to be better, better balanced between channels, and more pronounced LFE response. But i'm keeping the both to let someone else choose between the two, without letting them know wich is wich.

[evdberg]

Regarding below problem, where to get older versions of eac3to? Just to see when the problem is introduced and see if eac3to is the source of the problem at all.

I am not sure since what version the following problem is (I had V3.12 and just tried V3.14), but eac3to can not read m2ts files anymore. So when I point to the root of the ripped BD, I get the report as usual. But when I add the title index, eac3to reports there is an error reading xxxxx.m2ts. I am using CrossOver on Mac OS-X 10.5. For the record, this has always worked perfectly fine with older versions. CrossOver has not been updated in the meantime.

[Thunderbolt8]

arcsoft totalmedia theatre 3 is out, does anyone know if there could be benefits for eac3to from it?

[honai]

Regarding below problem, where to get older versions of eac3to? Just to see when the problem is introduced and see if eac3to is the source of the problem at all.

http://www.videohelp.com/tools/eac3to/old-versions#download

[deathlord]

How do I disable the 2nd pass? I have a file with discontinuities. Eac3to doesn't care and finishes processing. The result has minor picture and audio dropouts I can easily live with. However, eac3to erroneously sees an audio delay and starts a second pass. Afterwards the audio is no longer in sync from the first discon on. I am pretty sure this would not happen without the second pass. But that is automatically executed and I don't know how to prevent that.
Unfortunately the newest version without automatic 2nd pass (2.80) stops at the discons (even with -ignorediscon switch).

[evdberg]

@honai,
Thanks! After some testing it turns out the problem is introduced in V3.06

@Madshi,
What is changed in V3.06 that can explain this behavior? So in short, eac3to can perfectly read the contents of a BD, but it can not read the m2ts files (using CrossOver on Mac OS-X).

[IanD]

Is there any way to demux a PIP video and its associated soundtrack and keep them in-sync?

Whenever I have attempted to do this, the audio is always wildly out of sync. I'm assuming that either the video or audio is continuous, whilst the corresponding audio or video isn't: often there are long stretches of blank video between the PIP segments.

[saint-francis]

Got an issue with Terminator, the SaraConnor Chronicles. eac3to failed to detect an entire episode. There are three on there and eac3to only found 2. I managed to demux the m2ts file for the episode, but it lacked a chapter file.

PLAYLIST and CLIPINF folder can be found here. (http://www.mediafire.com/?sharekey=910d230ea6085fa036df4e8dca1419699b7d209600d73463b8eada0a1ae8665a)

Thanks madshi for all your work.