eac3to - audio conversion tool

[jruggle]

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768


The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

That is not true. Though I am listed as the current maintainer of it, I did not write the AC3 encoder in FFmpeg. It was written by Fabrice, who started FFmpeg. I used his code as the starting point for Aften. The original FFmpeg AC3 encoder has not changed much since it was first written. I made a few tweaks and some speed improvements, but nothing significant.

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?
FFmpeg is a simpler encoder than Aften. It only allows 16-bit input, and it uses 16-bit fixed-point for the MDCT and other calculations. It does not have any options other than bit rate and frequency cutoff. Aften allows higher resolution input such as 24-bit and floating-point, it uses floating-point calculations internally, and it has many options. Aften is also multi-threaded.

That said, the quality is not significantly different. There is only so much you can do with the format. The core encoding features are basically the same, the 2 encoders just use different algorithms in several places.

[jruggle]

So what do you guys think... should I use aften and tweak bandwidth or is it a bad idea?
The only downside of increasing the bandwidth is that the encoder has to spread the available bits across more frequency coefficients, so it can decrease overall quality.

[tebasuna51]

That is not true. Though I am listed as the current maintainer of it, I did not write the AC3 encoder in FFmpeg... I made a few tweaks and some speed improvements, but nothing significant...

Sorry, my fault. I was thinking your tweaks like current maintainer was more important.

[Abradoks]

I have recently made an objective comparison between proprietary Dolby encoders and aften. Aften has much worse SNR than DD. Even 224 kbps AC3 encoded with DD beats aften 448 kbps in SNR. When using PEAQ metric aften is closer to DD, but still worse. :(
Also, if you want to have frequency cutoff similar to DD encoders, you should use w=48 for 448 kbps and w=8 for 224 kbps. This tweak gives a little worse SNR but better PEAQ on 448 kbps (w=8 gives worse PEAQ than default settings).
If somebody is interested, I can repost here that comparison. But, it seems that development of AC3 encoders is dead.

[tebasuna51]

...
If somebody is interested, I can repost here that comparison. But, it seems that development of AC3 encoders is dead.
Two post before you have jruggle the Aften developper.

Maybe he need some support to improve the encoder, everybody is interested in a good free ac3 encoder.

But, please, use the Aften thread (http://forum.doom9.org/showthread.php?p=1341620#post1341620) for this.

[shon3i]

When using PEAQ metric aften is closer to DD, but still worse.Nothing than usual ABX comparing is not optimal for making decision which encoder is better, none of methods PEAQ or Spectogram like MokrySedeS posted. Because encoder made decision to cutoff certain frequency range to keep overall quality. That means audio still can be transparent to source, but if you look at spectogram and see that something missing, because forcing to not cutoff (leave frequency range) is not good.

Many people here have a bad decision, rather than listen, they watch the some values and graphs, which is completly wrong.

[Terrachild]

Abradoks,

very interesting.
Could you tell me what commercial encoder you tested. (Surcode?)

How did you measure signal to noise ratio, is there a free program you can point me to that can do that? What about measuring PEAQ?

You said use W=8, what setting is that. Im using "WAV to AC3 Encoder 4.1" where is that setting?

Thanks

[raziel666]

I have a certain DTS-MA audio track from a bluray disc of mine and I want to compress it to FLAC.

I own Sonic Cinemaster Audio Decoder 4.3 and have installed the .dll's of ArcSoft using the trial version as described in other posts in doom9.

Now the problem is this: when trying to encode to FLAC, the ArcSoft decoder gives a file twice the size of that from the Sonic Audio Decoder.

After examining the files with MediaInfo, I found out that the bitrate varied between these two decoders. The ArcSoft one produced a file with 2833kbps and the Sonic one with 1631kbps. Moreover, the file produced with ArcSoft is supposedly 14 minutes longer than the original duration and when I tried to play it back, it sounded slowed down and like crap... :(

Am I doing something wrong? Is it some bug that one of the decoders has?

Thanks in advance for your answers.

[rapscallion]

It's a True-HD Blu-ray track and I'm trying to create lossless wavs. (using eac3to_more) and I'm running into a strange problem.

First, when I create wavs, from a regular AC-3 track, Nero decoder (direct show) does the processing and maintains 24 bit.

However, when I try to do this to the HD track, ffmpeg/Libav somehow does the decoding and after the first pass Libav reports superfluouus zero bytes, eliminates them and drops down to 16 bits.

Is there a way create lossless wavs, maintaining 24bit, from True-HD ?

[Inspector.Gadget]

If eac3to using LAVC says a lossless codec is actually 16-bit, it is. 24-bit decoding for a lossy format like AC3 makes sense because presumably you're going to transcode it later and the additional accuracy is desirable.

[rapscallion]

Yes, however, at the first pass it's reported as 24 bit. And, as I stated, libav is dropping it down from 24 to 16 bit, NOT reporting that it originally is.
What is lavc btw, or did you mean libav?

My goal is to re encode the wavs to DTS-MA.

[Inspector.Gadget]

LAVC = libavcodec. The first pass calls it 24 bit because eac3to has to make sure that the 8 bits of padding lasts the length of the file and there's no actual 24-bit content in there. Once eac3to has verified this, it can discard the padding during decoding.

[rapscallion]

Ok, thanks, I got it now....just two last questions.

Is Nero docoder (from ver 7.10.1) not able to decode TrueHD and that's why eac3to is choosing libav/ffmpeg ?

When I re encode to DTS-MA will the resulting track be back to 24 bit or does it even matter ?

[Abradoks]

I have posted (http://forum.doom9.org/showthread.php?p=1341798#post1341798) comparison in aften thread.

Nothing than usual ABX comparing is not optimal for making decision which encoder is better, none of methods PEAQ or Spectogram like MokrySedeS posted. Because encoder made decision to cutoff certain frequency range to keep overall quality. That means audio still can be transparent to source, but if you look at spectogram and see that something missing, because forcing to not cutoff (leave frequency range) is not good.

Many people here have a bad decision, rather than listen, they watch the some values and graphs, which is completly wrong.
I'm not going to argue about subjective vs. objective tests. If you have enough resources to make ABX comparison, it'll be very interesting to see how they correlate with SNR and PEAQ.
Besides, my tests were started because somebody reported bad listening quality of AC3 track encoded with aften compared to DD encoder. It was suggested that default frequency cutoff causes such effect.

You said use W=8, what setting is that. Im using "WAV to AC3 Encoder 4.1" where is that setting?
You can control encoded frequency range through commandline option "-w" when using aften. I don't know if any GUI can do this.
You can find information about used encoders and measurement tools in the post I linked.

[Inspector.Gadget]

Is Nero docoder (from ver 7.10.1) not able to decode TrueHD and that's why eac3to is choosing libav/ffmpeg ?


I'm not sure, but both Nero and LAVC can decode TrueHD (the MLP part, which is what you're after) bit-perfect.

When I re encode to DTS-MA will the resulting track be back to 24 bit or does it even matter ?

If you're using the DTS HD Encoder Suite, then it will remain 16 bit. I'm not sure what Surcode would do.

[rapscallion]

OK, thanks IG.....and yes DTS MA Suite to encode.

Edit: Both DTS-MA and Surcode encodes result in 16 bit depth.

[TinTime]

I'm not sure, but both Nero and LAVC can decode TrueHD (the MLP part, which is what you're after) bit-perfect.

Nero's limited to 5.1 output though which is why eac3to uses libavcodec by default.

[rapscallion]

If eac3to using LAVC says a lossless codec is actually 16-bit, it is. 24-bit decoding for a lossy format like AC3 makes sense because presumably you're going to transcode it later and the additional accuracy is desirable.

Well, not only are you exactly right, but I found this info, for the audio track I'm processing, on Blu-raystats.com :

Run -Time2h 32m 13s
Language -English
Disc Language -English
Video -1080p 2.35:1
Video Encoding -VC-1 24 Mbps
Audio- Dolby True HD 5.1 16 bits 48 kHz 1505 kbps
Secondary Audio -DD AC3 5.1 640Kbps
Disc- BD50 40.72 GB
Region- A, B, C

I always thought that both DTS-MA and DDTrueHD were 24 bit. Live and learn.

[rapscallion]

Ok guys, I've been playing around with this and now I'm a bit confused.

First time extracting TrueHD to 6 wav files via eac3to_more.

When I do this directly, 2 passes are made, resulting in wav files that are ~800,000KB ea/16bit.

However ,if I process the TrueHD to an AC-3 file and THEN process the AC-3 to wav files, the size of the wavs are ~1,285,000KB ea/24bit.

My reasoning is that the larger wavs are lossless and I can use them to create a DTS-MA audio track ? And I don't know what the smaller wav files are (the 2 pass process) ??

Any explanation would be greatly appreciated.

Another question is, when creating the DTS file should the rear channels (5.1) be attenuated to -3db or should that be left alone ? From what I've read movie theater tracks are boosted by 3db for the rears so theaters then attenuate them by the same. But BD tracks are not boosted, so should be left alone? Is that correct ?

My head hurts !

[nurbs]

TrueHD is a lossless codec that will give you the same output as the source. Lossy codecs like AC-3 and AAC don't store the information at a fixed bitdepth, so you can either decode them to 16 bit or 24 bit. Since the latter is a more accurate representation of the files content eac3to will do that by default.

[rapscallion]

Thanks, I understand that but it doesn't answer my questions re the wav files. Which processing gave me the lossless wavs?

[nurbs]

Both of them in a way. If you decode the TrueHD you get exactly what was in it. If you decode the AC-3 you might not get exactly what's in it (assuming they can be decoded to an arbitrary bitdepth), but 24 bit is close enough as in no audible difference from the source.
Of course if you encoded the AC-3 from the TrueHD there is a loss during the AC-3 encoding.

[Inspector.Gadget]

If you're making a DTS-HD MA track, the you should ONLY go TrueHD -> WAVs -> DTS Encoder. Going TrueHD -> AC3-> WAVs-> DTS encoder is converting lossy to lossless whether you re-encode or take the TrueHD core, because either way you're discarding the lossless MLP data.

[rapscallion]

......which is the logic I was using from the start. Thanks.

However, what really puzzles me here are the substantially smaller wav files (and DTS-MA files- 2.5gb vs 3.8gb) when going from TrueHD -> WAVs -> DTS Encoder, then when going TrueHD -> AC3-> WAVs-> DTS encoder.

Logically, I would think all the files would be larger in the first lossless option. Additionally, the first option results in 16 bit wavs, the second in 24 bit. I think you guys can appreciate my reasoning, even if it's way off base.

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ?

[Inspector.Gadget]

24-bit PCM (16 bit + 8 bits padding) is generated by decoding AC3 to 24 bit, for no quality gain (and in this case, quality loss versus MLP to WAV). Decoding straight to WAVs nets you only the real 16-bit data.

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ?

Yes. Original to lossless to lossy is preferable to original to lossy to lossy.

[nurbs]

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ? You should always use the first option.

About the filesize. IIRC wav filesize is simply bitdepth * sampling rate * duration, so with the same duration and sampling rate a 24 bit file is 50% larger than a 16 bit file. 0.8 GB *1.5 = 1.2 GB


24-bit PCM (16 bit + 8 bits padding)
AFAIK it's not padding, it's actual information, insofar as the 24 bit file is a more accurate representation of the AC-3 file than a 16-bit wav would be even if the source of the AC-3 was only 16 bit. Of course it probably won't make much difference.

[Thunderbolt8]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?

[honai]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?

Yes, gap detection means that eac3to looks for audio gaps in relation to the video stream (i.e. container timecodes), so once you demux audio there is no reference video stream (i.e. container timecodes) anymore.

[rapscallion]

In case you all missed this last question in an earlier post today :

"Another question is, when encoding a DTS file should the rear channels (5.1) be attenuated to -3db or should that be left alone (unchecked) ? From what I've read movie theater tracks are boosted by 3db for the rears so theaters then attenuate them by the same. But BD tracks are not boosted, so should be left alone? Is that correct ?"

Edit : Found this in the Surcde (which I don't use) manual :
7.2
The Attenuate Rear Channels 3 dB Option

The original standard for Surround Sound in movie theaters was to attenuate
the rear channels by 3 dB.

So, in mixing for theaters, studios have their rear channel monitors attenuated 3 dB.

Home theaters, on the other hand, have the rear channels at unity gain.
So, a mix that was made with the rear monitors attenuated 3 dB will
have rear channel levels that are 3 dB too high for home theater.

This option takes care of that difference. So, in general, you use this option
if creating a DVD-Video using a Surround master that was originally mixed for movie theaters

So, for tracks from BD discs everyone leaves the -3db option unchecked, right ?

[Snowknight26]

eac3to incorrectly reports the resolution for one of the playlists on a Blu-ray of mine:

C:\unzipped\eac3to>eac3to.exe E: 1)
M2TS, 2 video tracks, 8 audio tracks, 4 subtitle tracks, 1:36:07, 112.806p
1: Chapters, 35 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
(core: DTS, 2.0 channels, 24 bits, 1509kbps, 48khz)
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
7: AC3 EX, French, 5.1 channels, 640kbps, 48khz
8: AC3 EX, Spanish, 5.1 channels, 640kbps, 48khz
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3 Surround, French, 2.0 channels, 192kbps, 48khz
11: AC3 Surround, Spanish, 2.0 channels, 192kbps, 48khz
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), English

[honai]

Is that some CGI movie from Pixar?

[Snowknight26]

It is. Seamlessly branched, too.

[Abradoks]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?

[honai]

It is. Seamlessly branched, too.

Then take a look at the last few lines after muxing, eac3to 3.17 should be reporting the correct fps there, nothing to worry about.

[honai]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?

I believe you are the first to mention this.

[Thunderbolt8]

Yes, gap detection means that eac3to looks for audio gaps in relation to the video stream (i.e. container timecodes), so once you demux audio there is no reference video stream (i.e. container timecodes) anymore.is there any chance to save a gap file which then can be used when I process only the audio track alone later on?

[Snowknight26]

Then take a look at the last few lines after muxing, eac3to 3.17 should be reporting the correct fps there, nothing to worry about.

A cosmetic issue is still an issue.

[tebasuna51]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?
Welcome to lossy encoders/decoders.
Yes it is know, to avoid overflows, when convert from frequency domain to time domain, many decoders attenuate a little bit the output.

You can use the -normalize parameter if you want.
And, of course, the use of ArcSoft decoder is always recommended.

Decoding ac3 sometimes eac3to detect overflows and make a second pass attenuating the signal.

[Thunderbolt8]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?
for some strange reason the subtitle file for this movie is about 4 seconds out of sync after remuxing. the movie consists of 2 m2ts files, the first one only being 7-8 seconds long. still, no apparent relation to those 4 seconds of subtitle delay. the fps rate of this movie is listed as 26.744fps (even though the value added to the header is 24/1.001), maybe this affects the subtitles here?

[crasus]

v02 The video framerate is correct, but rather unusual.
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Applying (E-)AC3 delay...
a03 A remaining delay of -13ms could not be fixed.

The BluRay has two audio tracks. An ac3 640kbps one and a DTS-HD one with a core of 1536kbps. I get a delay could not be fixed message on both. Is there any way to correct it? ArcSoft DTS is installed and fully functional with eac3to (latest version).

Thank you!

[Snowknight26]

No, because of the size of AC3/DTS frames. I can't remember if AC3 is 32ms per frame or not, but if you have an AC3 track that has a -40ms delay, adding 32ms would yield an unfixable -8ms delay unless you reencode the audio. Same goes for DTS.

Not like you'd be able to hear the difference with an -8ms delay anyway.

madshi, would it be possible to parallelize audio conversion and gap fixing? I noticed that when converting a track to AAC and extracting a DTS track that had gaps, the 2nd pass for the DTS gap fixing only started after the 2nd pass of the AAC encoding/gap fixing finished.

[crasus]

eac3to reports that both tracks have a 83ms delay on the BluRay. Should I try to encode the DTS-HD to an ac3 5.1 448kbps track?

[Snowknight26]

Why are you trying to reencode? If it's to remove that miniscule delay, I suggest you rethink what you're doing.

[dcmo]

I'm pretty new to this, but I have a pretty big issue with the sound quality which I am getting. I'm using eac3to, pcm2tsmu, and ts muxer to get my backup. When running an
eac3to test it shows I have the Arcsoft Decoder, Haali Matroska Muxer, Nero AAC Endoder, and the MkvToolnix as up to date or working fine. The picture comes out great but the sound is definitely lacking. The channels being mapped wrong on some of the movies is merely an annoyance, I can just switch the cables; but the poor quality coming out of the center channel is almost a deal breaker. I'm re-encoding them all as lossless pcm to stream to a PS3, for that matter they may still be lossless. However the center sounds boxy, or boomy; after searching thru this thread it could be it has a kind of mono sound to it. I don't know what to call it, but I do know that it pales in comparison to the disc. I've got a pretty nice little set-up and I don't want to lose anything quality wise. Is there something I'm doing wrong while backing these movies up. Any help is much appreciated on this end.

[Inspector.Gadget]

Post the exact command line you're using, including the output of eac3to and what it reports about the source and destination files.

[tebasuna51]

I'm pretty new to this, but I have a pretty big issue with the sound quality which I am getting. I'm using eac3to, pcm2tsmu, and ts muxer to get my backup.
Pcm2Tsmu now is obsolete, the last TsMuxer accept .w64 files to avoid the problem with .wav files greater than 4GB.

Use eac3to to output .w64 files instead .pcm+pcm2tsmu

I'm re-encoding them all as lossless pcm to stream to a PS3, for that matter they may still be lossless. However the center sounds boxy, or boomy; ...

I think PS3 only support PCM 2.0, if you have an audio equipment with 5.1 speakers use ac3 640 Kb/s instead PCM.

If you want only stereo sound you need use :
eac3to input.file output.w64 -down2 -normalize
to obtain a PCM 2.0 with all the channels mixed.

[KidKappa]

i'm having troubles getting eac3to to recognize my arcsoft dts decoders/encoders, i have TotalMedia Theatre (2.1.6.131) and eac3to (v3.17) and running on Windows 7 64bit, when I run the -test i get that arcsoft is not installed and and error saying that magcore.dll cannot be found. i've looked through this thread but alot of it is over my head. i noticed that the arcsoft common files are in the "program files (x86)" folder and not in the "program files" folder. Also the missing magcore.dll is in there as well. could someone please give me a nice step by step on how to get this working. thank you.

[dcmo]

Post the exact command line you're using, including the output of eac3to and what it reports about the source and destination files.

eac3to v3.17
command line: eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.pcm
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 1:42:50, 24p /1.001
1: Chapters, 16 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: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48khz
6: AC3, English, 2.0 channels, 224kbps, 48khz
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Spanish
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[v02] Extracting video track number 2...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "C:\movie\HIC.pcm"...
[v02] Creating file "C:\movie\HIC.h264"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] Audio overlaps for 9ms at playtime 0:21:31. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:23:05. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:39:41. <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:40:32. <WARNING>
[a03] Audio overlaps for 10ms at playtime 1:03:57. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:16:14. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:20:34. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:25:34. <WARNING>
[a03] Starting 2nd pass...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Realizing RAW/PCM gaps...
[a03] Creating file "C:\movie\HIC.pcm"...
[a03] The processed audio track has a constant bit depth of 24 bits.
Video track 2 contains 147951 frames.
eac3to processing took 51 minutes, 59 seconds.
Done.

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\HIC.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\HICUPDATE.pcm"


SmartLabs tsMuxeR. Version 1.10.6 http://www.smlabs.net
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 stream does not contain fps field. Muxing fps=23.976
Decoding LPCM stream (track 2): Bitrate: 9216Kbps Sample Rate: 48KHz Channels: 7.1 Bits per sample: 24bit
Processed 147951 video frames
Mux successful complete.
Muxing time: 8 min 43 sec


I've only backed up a few movies so far, this is the only one where the audio was overlapping. I used Clown BD to do this one the first time, said the blu-ray structure wasn't recognized when I tried to use Eac3to with it the first time. The log files indicated wav files with Clown BD so I re-ran it.

I think PS3 only support PCM 2.0, if you have an audio equipment with 5.1 speakers use ac3 640 Kb/s instead PCM.

If you want only stereo sound you need use :
eac3to input.file output.w64 -down2 -normalize
to obtain a PCM 2.0 with all the channels mixed.

I've got multiple channels working now out of the PS3 with this, would the other channels be active if the PS3 only supported 2.0. Downgrading my sound quality is unacceptable, I enjoy the sound more than I do the picture. There are times when I will just shut my eyes and let the sound sink in. If this can't work I will just rip to a back up disc with BD Rebuilder.

[Inspector.Gadget]

What about taking the DTS core? You'll have known surround compatibility at the cost of the lossless data that you probably can't ABX against the core. That said, tebasuna 51 knows more than I do about the PS3, so I'll defer to him.

[dcmo]

A little more follow-up. I have a 7.2 system set-up and was testing (receiver set to MCH PCM) out the new results this morning. I had to switch the center and the RF channel to get the dialog coming out of the center channel. All speakers were active except for the LS (for further clarification not the LBS speaker) speaker, which was completly dead. As I had suggested earlier about the sound not sounding right when coming out of the center (possibly being mono) I wanted to hear what the mono would actually sound like so I changed my selection on my receiver to mono. While listening to it in mono I don't believe it is quite mono sounding when it is set to MCH PCM, and more than that the dialog changed from the center channel back to the RF channel with everything coming out of the RF speaker. So to sum it all up:

MCH PCM: have to switch C and RF speakers to get dialog out of center.

Mono: With the C and RF speakers switched (have to switch them to get MCH PCM), the sound now comes out the RF.

[tebasuna51]

command line: eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.pcm
...
[a03] Creating file "C:\movie\HIC.pcm"...
...

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\HIC.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\HICUPDATE.pcm"

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

[dcmo]

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

I've been converting with pcm2tsmu. My question now is is
.w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality. I'm really not interested in a 2.0 solution or any loss in sound quality.

[nurbs]

It's going to be exactly the same as the source.

[tebasuna51]

I've been converting with pcm2tsmu.
Please put command line
My question now is is .w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality.
PCM in WAV or W64 container and raw LPCM have absolutely the same uncompressed audio data. You can verify the sizes, only differents by a few bytes for the header.

[dcmo]

Been pretty busy working today, just realized what you meant by command line. I just assumed it was pretty universal on pcm2tsmu input. Here is what was input:
pcm2tsmu.exe C:\movie\hic.pcm C:\movie\hicupdated.pcm -i 24 -c 8 -s 48000

Getting ready to give w64 a try now, hope it comes thru for me.

[arty]

D:\util\eac3to>eac3to i:\temp\movie.mkv 3:d:\trans.flac
MKV, 1 video track, 2 audio tracks, 2:30:25, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
"movie h264 stream"
2: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz
"dts core track"
3: FLAC, English, 5.1 channels, 2:30:25, 24 bits, 48khz
"FLAC created by eac3to3.17 + Sonic 4.3.0 filters"
a03 Extracting audio track number 3...
a03 Decoding FLAC...
a03 Encoding FLAC with libFlac...
a03 Creating file "d:\trans.flac"...
----------------------------------

is this "demux" method lossless? couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(

[Snowknight26]

is this "demux" method lossless?
Of course it is, you're going from lossless -> lossless.

couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(
You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

[arty]

Of course it is, you're going from lossless -> lossless.


You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

thank you!

ahh, the --no-ogg param, so true :)

[dcmo]

.w64 was a complete bust for me, TS Muxer locked up on two different movies on me (one with TrueHD and the other with DTS HD MA). Both times with this:

LPCM bad frame detected. Resync stream.

Settings for TS Muxer:

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\ppl.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\ppl.w64"

SmartLabs tsMuxeR. Version 1.10.6 http://www.smlabs.net
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 stream does not contain fps field. Muxing fps=23.976
Decoding LPCM stream (track 2): Bitrate: 6912Kbps Sample Rate: 48KHz Channels: 5.1 Bits per sample: 24bit

Actually though it wasn't a complete bust, it did about 35% of the THD version and about 25% of the DTS before it locked up on me. Both played fine on the computer, haven't checked it out on the PS3 as of yet.

Okay, played it on the PS3 and it came out great for what was there, does anyone know what caused TS Muxer to stop muxing. Or rather why a bad frame was detected. How do I solve this problem.

On what was there, there didn't appear to be any mapping problems (I really like that). The sound sounded pretty much identical, I think I would need two audio systems to be able to check it out.
However the tv upstairs still didn't play it properly, it was like the center channel (main dialog) was missing although on the system downstairs it came thru properly. It's an older receiver (5 years maybe, no hdmi, no advanced audio). Should I swap cables around or do I just need a new receiver.

Thanks to all for you help!

[tebasuna51]

...
LPCM bad frame detected. Resync stream.
...

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

[mikelebron]

Guys.. Is it possible for me to remux my MKVs that have FLAC for audio and change the AUDIO to LPCM? Reason why I am asking is I want to stream my MKVs to the PS3... Am I going about it the wrong way?

[Inspector.Gadget]

Mikelebron: When you create the Matroska file from the original Blu-ray, just output PCM (WAV) audio instead of FLAC. Nothing requires you to use FLAC when demuxing a disc with eac3to...

[dcmo]

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

So I should take this to the TS Muxer section? I saw this problem described over there (in April), but there never was really an answer for it.

I was reading in the thread about first converting it to WAV, then to W64; do you think this would help.

If so how does one go about converting it from wav to w64; I know that eac3to.exe E: 1) 2: C:\movie\movie.h264 3: C:\movie\movie.wav would get it to a wav file (although I've read that wav has problems with over 4G files), but how do you then convert that wav file to a w64 file. I assume it would be a command line, what would that line look like?

I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.

[tebasuna51]

...
I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.
TsMuxer don't accept standard LPCM (.pcm from eac3o), for that I wrote Pcm2tsmu.

TsMuxer always accept .wav files but only <4GB
Last versions of TsMuxer say accept .w64 files (same data than .wav but with a header to support >4GB).

I don't know how solve your problem.
My recommendation is use AC3 640 Kb/s, only with very good audio equipment in a conditioned room you can listen some difference.

[dcmo]

I was going to use pcm2tsmu just to see if it was w64 that was causing the problems and not the discs. I'm sure it's w64, I just wanted to be sure.

After reading some more, what about splitting the file and rejoining them later (that way I could do it in a wav file).

I just can't downgrade the sound; if the choice is between streaming with downgraded sound or not streaming at all then not streaming at all is hands down the winner (it's really not even an option).

I will just go the re-build route and burn it to a physical disc. Anyway can I split them and rejoin them, would I be able to tell where it was happening when I watched it.

Thanks again.

[utenteanonimo64]

I have just discovered eac3to and I am really impressed. In a matter of days it has become a central part of my audio processing workflow. There is only one thing I am missing (or I haven't been able to find out): I would like an option to have the audio output from a bluray stream split into several files one for each chapter.
Is Madshi still supporting this software and adding new features? Or is there an automated way to achieve this?
I have seen that someone has written a GUI called "HDConcertRipper" for this purpose but it gives me all sorts of errors on most of my discs...

[tebasuna51]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

[deebo]

some weird feature with powershell in windows7 (works fine without the -quality switch):

PS D:\> eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
Track 2 is used for destination file ".35".
This audio conversion is not supported.

works fine with cmd.exe:


D:\>eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
a08 Extracting audio track number 8...
a08 Patching bitdepth to 24 bits...
a08 Decoding with libav/ffmpeg...
a08 Reducing depth from 64 to 32 bits...
a08 Encoding AAC <0.35> with NeroAacEnc...
---

[utenteanonimo64]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

Well I wish Madshi changed his mind.... personally it's the only feature I miss...

[AnryV]

eac3to can't decode 1.0 DTS-HD MA
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

[dcmo]

Actually I got it figured out. It wouldn't do a wav or w64 file, but it would do a pcm (with pcm2tsmu) file. After having checked that out, I converted the original audio to a wav and then to w64. Turned out perfect, sounds great. As far as the issue upstairs, picked up a cheap Insignia receiver (HDMI input with HD audio codecs) on sale at Best Buy which solved the sound issue upstairs.

[tebasuna51]

eac3to can't decode 1.0 DTS-HD MA

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

[Inspector.Gadget]

AnryV, maybe i'm missing something, but could the error be the result of the wavs switch for a mono source. What happens when you output to audio.flac instead?

[AnryV]

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

http://rapidshare.de/files/48730597/sample_Tonton.dtshd.html

This occurs not only with this sound.

[tebasuna51]

This occurs not only with this sound.

Yes, seems a bug in ArcSoft decoder (same error with output to flac or wav).
I don't have Sonic to test.

BTW, the core bitrate, 768 Kb/s, is more than enough for a mono audio. Must be dificult diference from original.

[AnryV]

Yes, seems a bug in ArcSoft

And Sonic?
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

TMT3 play it normally. I think that this is a bug of eac.

[xkodi]

TMT3 play it normally. I think that this is a bug of eac.

and why you are so sure that TMT3 plays the DTS-HD MA extension rather than DTS core, which i'm sure eac3to can also extract and decode correctly.

[AnryV]

and why you are so sure ...
Not sure... now. :(

[dcmo]

Been trying to get Ice Age 3 done, but Arc Soft is responding with strange set-up. The completed file outputs the center out the right channel, so a couple questions please. When channel remapping which channels do -0,1,2,3,4,5 correspond to? I take this is for 5.1, what happens when you have a 7.1 track? I assume you run the -log to find out what the channels are listed at, is this a correct assumption?

[Thunderbolt8]

And Sonic?


TMT3 play it normally. I think that this is a bug of eac.
afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files

[AnryV]

afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files
What about Sonic decoder?

[asc28]

Is there any way to recalculate and add back dialog normalization to AC3 tracks, after eac3to has stripped them?

[edit] realized that many studios use an arbitrary figure, so might not be able to recalculate. But how about using a ReplayGain like technique to find the optimal dB to be added and then adding it to an already encoded AC3 track?

Also, I know this has been asked a couple times before, but no one seems to know how exactly to go about reinserting dialnorm metadata without re-encoding.

[Esurnir]

I'm going to sound dumb, but which product from arcsoft still include the DTS-HD filter decoder? Does TotalMedia Theater 3 work ? Any special editions? (platinum?)

[Thunderbolt8]

What about Sonic decoder?
dont know to be frank.

[IanD]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable by Goldwave, Cooledit or CDwave as it seems they are "raw". No matter what bit depth, signature or sample rate I choose, I only get noise. Strangely I am able to play the wav files with Directshow no problem.

How do I make the wav files readable in the sound editor applications?

It's frankly annoying that the output isn't wav standard.

[kypec]

Try to add switch -simple and see if it helps your applications to recognize such stereo WAV files. If yes then you should blame those apps that they don't support standard WAV headers.

[tebasuna51]

GoldWave (from v5.02, 2004? ) read WAVE_FORMAT_EXTENSIBLE headers without problems.

[73ChargerFan]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable
Perhaps the DTS core is being converted to a dts-wav file, instead of being decoded?

eac3to (DTS or DD file) filename.wavs should decode, one channel per file.

[IanD]

Using -simple seems to have done the trick and both converted DTS HD MA and DD2.0 source files are now recognised as stereo wav.

However, I'm experiencing problems trying to synchronise the tracks: the DTS HD MA soundtrack is identified by eac3to as 24fps (not 23.976fps), whilst the commentary track I'm trying to synchronise to it is from a PAL source. Using the switches -25.000 and -ChangeTo24.000 results in the commentary being about 500ms longer than the soundtrack when the beginnings are synchronised.

I thought it might be possible to use the source switch -24.99x to fine tune the commentary conversion, in case the PAL source is not quite 25fps, but eac3to refuses to accept anything other than -25.000

Any thoughts?

Unfortunately I don't have access to an NTSC commentary, only a PAL one. It's quite close at the moment but not close enough because the commentary also has parts of the soundtrack that need to be in-sync with the on-screen action.

[Vick]

Many thanks to Christopher Key for providing the HDCD decoder!

Hi madshi

Whether Chris has given you the HDCD decoder source code?
He does not respond to email (cjk32 cam ac uk) :confused:

Thnx
Vick
p.s. sorry for my poor english

[pcordes]

I'm an old-school GNU/Linux command line junkie. Running eac3to inside
screen (http://en.wikipedia.org/wiki/GNU_Screen), the output comes out like this:


$ alias eac3to='wine /usr/local/.../eac3to.exe'
$ eac3to foo.eac3
fixme:mixer:ALSA_MixerInit No master control found on HDA ATI HDMI, disabling mixer
fixme:reg:GetNativeSystemInfo (0x32fc80) using GetSyE-AC3, 5.1 channels, 0:31:59, 1536kbps, 48khz
# or, discarding wine's messages: eac3to 2>/dev/null
$ eac3to 2(1) Haali Matroska Muxer


I spent a while looking for --help output, until I eventually realized that its output was designed for a Windows command window, not a VT100 terminal emulator. It happens to work outside of screen, in just gnome-terminal, and in native xterm, though.

eac3to outputs a bunch of ^h (ascii 0x08) characters at the start of every line, and maybe under screen, reverse-line-wrap happens, and the cursor backspaces up to the previous line, so every line of text overwrites the one before. If I redirect the output to a file,

$ eac3to > eac3to.txt 2>/dev/null
$ file eac3to.txt
eac3to.txt: ASCII English text, with CRLF line terminators, with overstriking
$ hexdump -C -v eac3to.txt
00000000 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000010 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000020 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000030 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000040 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000050 20 20 20 20 20 20 20 20 20 20 20 20 20 20 0d 0a | ..|
00000060 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000070 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000080 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000090 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000b0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000c0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000d0 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 08 | ........|
000000e0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000f0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000100 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000110 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000120 08 08 08 08 08 08 08 08 08 08 65 61 63 33 74 6f |..........eac3to|
00000130 20 76 33 2e 31 37 2c 20 66 72 65 65 77 61 72 65 | v3.17, freeware|
00000140 20 62 79 20 6d 61 64 73 68 69 2e 6e 65 74 20 20 | by madshi.net |
00000150 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000160 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000170 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 | .......|
00000180 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000190 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001b0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001c0 08 08 08 08 08 08 08 08 08 08 08 20 20 20 20 20 |........... |
000001d0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001e0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001f0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000200 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000210 20 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 | ......|
...


But the overstriking doesn't have any effect anywhere: xterm or gnome-terminal. (it visually looks the same as regular text). In VT100, ^h is cursor-movement, not highlighting. If eac3to is doing that itself on purpose, please don't, at least under wine. The CRLF line endings are not a problem.

I'd also love to have an option to not play sound. I haven't tried just renaming the .wav files. Maybe you could add a -unix option, or check an environment variable, to turn off sound and not print funky text. It's harder to deal with options at the end, instead of as the first arguments (e.g. an alias doesn't work), but a wrapper script like

#!/bin/sh
wine /.../eac3to.exe "$@" -unix 2>/dev/null
# or use a wine cmdline arg to filter out the messages it always prints, instead of redirecting to /dev/null

would do the trick.

apparently, wine devs
don't want apps to detect wine (http://forum.winehq.org/viewtopic.php?p=25906&sid=4a6025570ec9cb97aa5e4d952cf6ad88), so the -unix option would be a nice feature. Or have a -nosound option, or a -quiet option. (OTOH, -quiet usually means no messages, not literally quiet.) I'm worried that -nosound would cause confusion, though, since someone might think it means exclude audio tracks. But it's -nosound, not -noaudio...

Anyway, that's my feature request, please and thank you.

[samepaul]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?
Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

[tebasuna51]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?

You can add the parameter:
-no2ndpass
but is a very little diferent volume.

Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

Is the size of the audio sample, trust in eac3to, NeroAacEnc don't support 64 bit float audio samples.

[samepaul]

You can add the parameter:
-no2ndpass
but is a very little diferent volume.


Thank you! It did the magic - volume remained good. Seems anti-clipping algorithm requires some improvements :)

[tebasuna51]

... Seems anti-clipping algorithm requires some improvements :)

Then there are big volume difference ...
Please, can you upload a sample?

[samepaul]

Seems you're right. I tried to notch the place where clipping occurs, cut it out and encoded with and without 2nd pass.
There was difference in volume, but not so dramatic as it was when whole track was encoded. So I don't think that these samples are good for debugging.

I think it's not completely fault of encoder. I guess the track has places with really hard overload, but when eac3to performs 2nd pass it lowers volume of whole track evenly. Of course, after this previously clipped parts (blows, shootings etc) sound better, but "normal" parts where people talk become indiscernible. If I have tool to detect exact places where eac3to detects maximal clipping I would prepare sample for you. But without this I'd have to submit whole 700mb track , which I don't know how to do.
Btw, I've mistaken when I wrote "TrueHD". It was actually AC3.

But nevertheless, regarding to volume optimization, I think if eac3to performs 2 passes it can normalize levels non-linearly.

[honai]

I think if eac3to performs 2 passes it can normalize levels non-linearly.

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

[samepaul]

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

Uhum... and making audio quiet so much that you hadly can distinguish what people say - this is what you call "high fidelity"? If so, I'd stick with normal fidelity ;)

1st of all, DRC does not lower fidelity more than any volume adjustment, which is done by eac3to anyway. And really negligible comparing to quality loss due to audiocompression.
2nd, what I propose is not really DRC. Range compression only performed around overloaded zones, leaving ~90% (depending on content) of audiotrack intact.

[IanD]

The switches -25.000 (source) and -ChangeTo24.000 imply that they are floating point values that "may" accept any floating point value, but in fact seem to be hardcoded to -23.976, -24.000 and -25.000 and nothing else is accepted.

Is there any way to define variable floating point values to finetune these parameters?

Is there an alternate way to perform non-standard resampling and time expansion/compression?

[tebasuna51]

...
Is there any way to define variable floating point values to finetune these parameters?
To make the job eac3to use SSRC libraries to change the audio samplerate.
And not all conversion are possible. You can know available values here: http://avisynth.org/mediawiki/SSRC

Is there an alternate way to perform non-standard resampling and time expansion/compression?
There are commercial software like TimeFactory (http://www.prosoniq.com/editing-products/timefactory-2/)

And also AviSynth methods like TimeStretch (http://avisynth.org/mediawiki/TimeStretch)

[samepaul]

I think I found bug.
eac3to 00001.m2ts 1: chapters.txt 3: audio1.m4a -no2ndpass 6: audio2.m4a -quality=0.6 -no2ndpass 9: english.sup
It starts, spawns 2 nero encoders, processes whole input and then one Nero finishes, but 2nd does not and stays idle. I mean really idle - 0% CPU and no byte read or written.
As result 1st m4a file is ok and 2nd is unreadable, although is not empty.
No problem occurs if only one track is recoded at a time.

[Foofaraw]

So, any plans to support wmv/a? I need to get some wma10 out of a wmv and down to something sensible, like say ac3 :)

[tebasuna51]

To convert wma audio, in wmv container, to ac3 you can try several methods:
- Foobar and Aften
- dshow2wav (http://forum.doom9.org/showthread.php?t=141441) and pipe the output to Aften.
- AviSynth methods (Soundout, BeHappy, bepipe, wavi) with DirectshowSource decoder.

For more detailed explanation please open a new thread.

[b66pak]

off topic

@tebasuna51 it is not so simple...most of the time you will get an audio oos...
_

[Midzuki]

b66pak wrote:

off topic

it is not so simple...most of the time you will get an audio oos...

That's true. To whom this may interest, here goes my "method":

1) copy the Windows Media Audio stream to a standalone ASF file (*.wma) with AsfBin ;

2a) convert to a .WAV file via Avisynth + BassAudioSource() ;

2b) OR... playback the intermediate WMA file with Winamp's Disk Writer plugin.

So far, it's worked fine for stereo Windows Media Audio,
I still haven't found problematic multichannel WMAs on which I should test this workaround.

:helpful:

[hoju3508]

Is there a way to preserve the HD audio when slowing it from 25fps to 23.976fps? Once it's slowed down through eac3to, the HD streams are gone. It only uses the core of the track.

:thanks:

[tebasuna51]

Use flac output, is also lossless, or buy a TrueHD/DTS-HD encoder. But change the samplerate isn't a lossless operation.

[Mark_A_W]

I would use FLAC, but leave the file alone and change the speed on the during playback with Reclock.

You don't lose anything permanently that way.

[hoju3508]

Thanks for the feedback.

Use flac output, is also lossless, or buy a TrueHD/DTS-HD encoder. But change the samplerate isn't a lossless operation.


I would use FLAC, but leave the file alone and change the speed on the during playback with Reclock.

You don't lose anything permanently that way.

[sucker]

suppose i have a 1h30min track and i want to cut it, so that i only have a 1h25min track, what would be the commandline to achieve this?

tia

[jmone]

Given we are seeing advances in Filters supporting HD codec playback (eg FFDSHOW can now decode most and is working at bitstreaming)....is it worth transcoding the Audio or just Remux the Original Audio when using MKV containers?

[jmone]

wrong thread - sorry

[Blue_MiSfit]

IMO, most stuff can safely be transcoded to AAC with perfect transparency. I do this most of the time.

If AC3 is the best quality audio source on a given movie, then I just keep it. Why reinvent the wheel?

If a movie has an incredibly good audio track, or is a musical performance, then I either keep it as-is (if it's TrueHD), or just transcode to FLAC.

Of course, I don't have a fancy receiver, or even a 5.1ch system, so I don't mind terribly :) Headphones suit me just fine!

~MiSfit

[MrVideo]

There is a situation, which I've hit a few times, that causes eac3to to hang in a loop of some sort, forever.

It hits a bad section of the video, and if it is bad enough, it will not get through it, nor will it error out. This is what is on the screen when it happens:

a03 0:43:17 The source file seems to be damaged (transport error indicated).
v01 0:43:17 The source file seems to be damaged (transport error indicated).
a05 0:43:17 The source file seems to be damaged (transport error indicated).
a04 0:43:17 The source file seems to be damaged (transport error indicated).
a03 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 0:43:17 The source file seems to be damaged (discontinuity).
a04 0:43:17 The source file seems to be damaged (discontinuity).
a03 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 0:43:17 The source file seems to be damaged (discontinuity).
a04 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 This track is not clean.
a03 This track is not clean.
a04 This track is not clean.

I run via the command line (Zshell script within cygwin).

An update: I was able to take what was completed and finish the job. I then looked at the output and the real nasty damage is at 58:00 minutes. There was no error output placed onto the screen, eac3to just went into a loop and hung at that point in the file.

[Carpo]

I have a little issue whereby the encode i have done is just a little bit (i would say 100meg or so) big to fit on a dvd9 disc, i know i could use 2 dvd5s but was thinking it would be easier to just drop the rate of the ac3, i have the ac3 on the drive (used tsmuxer to remove true-hd part) so its sitting as a 640kbps ac3 file.

To use eac3to i would just do

eac3to gran.torino.640.ac3 gran.torino.448.ac3 -448

is that correct? or would people suggest i copy the true-hd file from the bluray disc and use ac3 to do the lot manually skipping tsmuxer down converting to ac3?

[Snowknight26]

To get the best quality, use the TrueHD track and go straight to the 448kbps AC3.

[Carpo]

Thought as much, so its just the line above i need? or do i need to add the frame rate as well?

Does this program still require external decoders to do its thing or is it all done by eac3to?

[Snowknight26]

The line above would work. For TrueHD, replace the source file name with the TrueHD track. You don't have to worry about any kind of 'rates' unless you're changing the video.
eac3to can decode TrueHD and it can encode AC3 without any additional extras.

[Carpo]

thanks :) - i have to change the rate on the ac3 otherwise i cant get the backup i made to fit on a dvd9 - its slightly oversized, if i drop the rate from 640 to 448 it should fit nicely :)

[Snowknight26]

Bitrate, not rate.

[Carpo]

same thing for me, im a lazy typist ;)

[Abradoks]

I have tested Nero for AC3 decoding and it seems that DRC isn't removed correctly. Here is source (http://rghost.ru/716070), AC3 (DRC: None) (http://rghost.ru/716075), AC3 (DRC: Speech) (http://rghost.ru/716076). When comparing SNR of decoded files (I use CompAudio) Nero gets same SNR as libav on file encoded without DRC, but on second file Nero gets much worse SNR, while libav has nearly same numbers.
I'm not the only one who gets this issue, it was previously reported on another forum.

[samepaul]

I'm sorry if I overlooked - I tried to find answer, but with no luck so far... I wonder where sources of eac3to available? As long as I understand this project uses some LGPL code/libraries and thus fall under LGPL and would have it's sources published. I wanted actually contribute in further eac3to development.

[kypec]

I wonder where sources of eac3to available? As long as I understand this project uses some LGPL code/libraries and thus fall under LGPL and would have it's sources published. I wanted actually contribute in further eac3to development.
You have it misunderstood I guess. Quoting the wikipedia info on LGPL (http://en.wikipedia.org/wiki/LGPL):The main difference between the GPL and the LGPL is that the latter can be linked to (in the case of a library, 'used by') a non-(L)GPLed program, and regardless of whether it is free software or proprietary software.
eac3to is free yet proprietary software (until now) and its sources are not publicly available. It uses a lot of LGPL libraries though which are bundled in the download package.

[samepaul]

hm.. i rather misunderstood the "linked library" concept. But yes, even so it's not "derivative work" as it's defined in the LGPL.
Anyway, I'm not pursuing any legal issues anyway. I just wanted to contribute and help with bugfixes, since I consider this project an excellent multimedia integration effort. Actually the best among all I ever seen on Doom9.

[madshi]

The whole point of LGPL is that closed source projects like eac3to can use LGPL libraries without having to go open source. That's the key difference between LGPL and GPL.

Thanks for the offer to help with development, and for the nice feedback, I appreciate that. I'd rather keep eac3to closed source, though, and do the bug fixing myself. I'm not aware of any major bugs (except maybe the Nero DRC bug, if that report is actually correct). So I'm not in a hurry to release a new version. Too busy with other stuff right now...

If you have some specific "big" new features you want to add to eac3to, you could either:
(1) Write some Delphi (!!) code and send it to me for integration into eac3to.
(2) Or you could create a little standard win32 dll, which eac3to would then link to.

[Carpo]

will you be adding support for nero 9 - or are you going to leave it with 7.x - a version you know works :p

[brasch]

On both Leaving Las Vegas and 3 days of the condor I can't choose any subtitles, even though they are there. Anyone know a fix for the problem?

[mikelebron]

Question... I noticed in some movies (so far only trueHD) once in a while at the same spot in a movie I get a little bit of static... What is that? How can i prevent it?

UPDATE: I tried redoing the file.. this time putting -keepDialNorm bu this did not help.... in the EXACT spot where the crackling was had the exact same issue... any other ideas? BTW: Its star trek first contact.. the scene is at the end where the vulcans are landing on earth for the first time... The crackle happens when the vulcan ship touches down..

[Carpo]

could this be an issue with the actual disc? or does this happen only after you have copied it to the drive?

[mikelebron]

Played right from the disc using all the same player (MPC-HC), codecs (FFDSHOW video and audio) and renderer (reclock) NO ISSUES....

what I did notice was the volume (level) was MUCH MUCH higher from the eac3to ripped version... Is this normal? I suspect in the audio chain (renderer, driver, windows mixer, etc.) is being over modulated... is there a way to have the volume level unmolested? (I know I can reduce it by -3db but I prefer it to just be like whats on the disc..)

could this be an issue with the actual disc? or does this happen only after you have copied it to the drive?

[Blue_MiSfit]

It's obviously not an issue with the disc.

Do you hear the crackling from the TrueHD file, simply extracted with eac3to, or is this in a transcoded file from a TrueHD source?

I'd try demuxing the TrueHD with TSMuxeR, and comparing the results (though I can't imagine there will be any).

Curious problem..

~MiSfit

[tebasuna51]

Played right from the disc using all the same player (MPC-HC),...

what I did notice was the volume (level) was MUCH MUCH higher from the eac3to ripped version... Is this normal? I suspect in the audio chain (renderer, driver, windows mixer, etc.) is being over modulated...
Maybe is a problem with TrueHD libav decoder, please put the log file at conversion.

If you can extract (in the source) the problematic fragment and upload to analyze ...

[umaximus]

Right, I have bunch of trueHD soundtracks that were demuxed in past with eac3to. Since I got a capable HD receiver I want to mux HD audio back. Doing this with TsMuxeR I have no problems with DTS-HD MA, but TsMuxer wont accept trueHD streams and is giving an error of 'Unsupported format' (cant detect stream type) on every single trueHD stream that I have (+100). Im sure there is difference in header info of how eac3to is demuxing trueHD but I have no clue how to 'repair' it.

[mikelebron]

Whats the best way for me to cut that one scene out to upload it?

Maybe is a problem with TrueHD libav decoder, please put the log file at conversion.

If you can extract (in the source) the problematic fragment and upload to analyze ...

[mikelebron]

Ok.. I used Tsmuxer to see if I had the same issue and it was there... same static while I hear the audio(not crackle)... that lead me to try again in MPC-HC and play the movie right from the disc. Low and behold... it was there from the disc. Sorry about that guys.. I dont know how I missed it. It must be something then with ffdshow decoder

[stax76]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

http://i.imagehost.org/t/0466/aaa.jpg (http://i.imagehost.org/view/0466/aaa)

[honai]

Your console runs in ANSI mode, but eac3to probably assumes a Unicode codetable (which is the default for most Windows installations).

[Yraen]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

Just a thought, are you trying to read the STDERR? Try just STDOUT if you are.

[stax76]

I'm using simple .NET code which works for all other applications, .NET and WinAPI are both pretty much unicode if you don't count the legacy stuff, the .NET lib is pretty powerful so maybe there is a simple solution, if not interfacing with native code would be no problem for me. I assume it has something to do with the colors eac3to outputs which is pretty cool if you just use the command shell. Both console and text encoding are not areas I have much experience. I've tried stderr first since it uses the same code running BeSweet and that use stderr, eac3to did however not write anything to stderr if I didn't miss something, I'm reading eac3to output from stdout.

[stax76]

It's some control chars appearing right in front of colored output, happens also when you make a bat file and redirect the output to a text file using: eac3to.exe test.wav test.ac3 > test.txt

Is there a switch to disable the colors? Otherwise I need to try trimming these control chars.

[Yraen]

I need to get around to trying to learn .Net. I know from autoit, when trying to read the stderr along with stdout I got the exact same result as you. When I went to stdout only I got the results I was looking for.

Edit: I'm not aware of any switches to turn off the colors.

[stax76]

I tried both and it only wrote something to stdout. I've checked the WinAPI console documentation and the eac3to documentation, seems I don't have luck here so I have to try to trim the 'garbage'. Thanks for the help.

[stax76]

If you look at the screenshot I've posted you can see BS BS BS BS etc., it means Backspace! I've checked it with a hex editor, where it was 08 08 08 so I went to ASCII at wikipedia and it said 08 is Backspace. I'm such a noob when it comes to binary data and console. :)

[Abradoks]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

http://i.imagehost.org/t/0466/aaa.jpg (http://i.imagehost.org/view/0466/aaa)
It was already discussed a few times (1 (http://forum.doom9.org/showthread.php?p=1333829#post1333829), 2 (http://forum.doom9.org/showthread.php?p=1348250#post1348250)).
Adding -simpleoutput switch into eac3to looks like the best solution.

[mrr19121970]

It's really OT, but quite straightforward. See here (http://msdn.microsoft.com/en-us/library/ms682073%28VS.85%29.aspx). AttachConsole, CreateFile, GetConsoleScreenBufferInfo, ReadConsoleOutputCharacter & FreeConsole when you're done.

[stax76]

You receive lines from a callback using .NET, this lines can be cleaned with simple string processing.

[honai]

Your screenshot shows that your console assumes ANSI (in the lower right corner), not Unicode. ;)

[stax76]

From a text file you cannot make a relation to a console, this text file was a .NET string object which is UTF16 converted to ANSI when saving to file. I should save it better as UTF8. For the backspace (0x08) control char issue text encoding isn't relevant. eac3to is the only console application I know writing backspace to the console, other applications probably mostly write/read lines but since I never did a console application I cannot tell sure. I can handle it, it took me a while but I've learned something.

[Roscoe62]

I just bought the Clone Wars Season one box set.

After ripping with AnyDVD HD, I use eac3to to select one of the episodes and rip out the video, audio & chapter data and process further from there.

However, eac3to is only showing 1 "episode" (for lack of a better term) which I don't believe is actually an episode, and there's only video - duration 15:02. No audio, no chapters.

I know there are 7 separate episodes on this disc but, for some reason, eac3to is not finding them.

Anyone else struck this yet and, if so, found a way around it?

[73ChargerFan]

Try BDInfo, it will decode all MPLS files and give you the track lengths. I'm guessing you'll find seven 20 minute playlists. BDInfo does make mistakes sometimes, though.

[Killroy™]

I just bought the Clone Wars Season one box set.

After ripping with AnyDVD HD, I use eac3to to select one of the episodes and rip out the video, audio & chapter data and process further from there.

However, eac3to is only showing 1 "episode" (for lack of a better term) which I don't believe is actually an episode, and there's only video - duration 15:02. No audio, no chapters.

I know there are 7 separate episodes on this disc but, for some reason, eac3to is not finding them.

Anyone else struck this yet and, if so, found a way around it?

Good luck with this. I found the same thing until I found sort of a solution:

I had to hunt down the .mpls that showed the actual playlist and found that the "Play All" is listed on 00020.mpls but I found another surprise:

c:\eac3to317>eac3to f:\swcwdisc1\bdmv\playlist\00200.mpls
1) 00200.mpls, 2:41:03
[95+96+97+96+98+96+99+96+101+96+102+96+103+96].m2ts
- Chapters, 35 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Japanese, multi-channel, 48khzDo you see that repeating 00096.m2ts file? Well, that little sucker is a 5 minute copyright placecard...yes, 5 minutes repeated after each episode.

Even if you decide to leave it in place and just skip it with the "Next Chapter" button (if you include chapters) you will find that those 5 minute files really screw up the synch for each episode after it. The first episode is fine but later on it gets worse and worse.

The only way to do it is to remux each individual episode as its own file. At least you know the correct order by looking at the mpls file.

[73ChargerFan]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

[Roscoe62]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

Thanks for this. I'm just going to do this individually for each episode, but so far this procedure is working fine. Thanks again! :)

[umaximus]

Is it possible that TrueHD track doesn't have audio gaps/overlaps on seamless branching disc while AC3 has?

Pineapple Express

eac3to v3.17
command line: eac3to 1) 2: video.h264 3: audio.wav 5: audio.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 15 subtitle tracks, 1:57:26, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
4: TrueHD/AC3, French, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a05] Extracting audio track number 5...
[a03] Decoding with libav/ffmpeg...
[a05] Removing AC3 dialog normalization...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[v02] Creating file "video.h264"...
[a05] Creating file "audio.ac3"...
[a03] [libav] End of stream indicated <WARNING>
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Caution: The WAV file is bigger than 4GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Audio overlaps for 7ms at playtime 0:10:27. <WARNING>
[a05] Audio overlaps for 8ms at playtime 0:27:14. <WARNING>
[a05] Audio overlaps for 10ms at playtime 0:32:23. <WARNING>
[a05] Audio overlaps for 18ms at playtime 0:44:12. <WARNING>
[a05] Audio overlaps for 31ms at playtime 0:44:54. <WARNING>
[a05] Audio overlaps for 28ms at playtime 0:56:01. <WARNING>
[a05] Audio overlaps for 9ms at playtime 1:02:00. <WARNING>
[a05] Audio overlaps for 15ms at playtime 1:10:16. <WARNING>
[a05] Audio overlaps for 14ms at playtime 1:12:12. <WARNING>
[a05] Audio overlaps for 32ms at playtime 1:43:47. <WARNING>
[a05] Audio overlaps for 25ms at playtime 1:52:10. <WARNING>
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Reading WAV...
[a03] Stripping zero bytes...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[a03] Caution: The WAV file is bigger than 2GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "audio.ac3"...
Video track 2 contains 168949 frames.
eac3to processing took 47 minutes, 28 seconds.
Done.

[nautilus7]

Yes, that's very usual. TrueHD frames are ~3ms, while AC3 ones are 32ms long. IIRC, eac3to won't report any overlaps if they are too short.

[wolfbane5]

I was checking out We Were Soldiers and noticed that it fronts a DTS-ES audio track. When I used the -down6 option, it created the .wavs but said that Surcode wasn't installed. It is installed, however, I've encountered this problem before and have simply ignored it since I've been unable to find a solution to it. My question is: does the -down6 option mix/rewrite the channels so that the BC .wav track is somehow included in the other 6 channels or does it simply ignore the BC .wav track created when you convert the .dts into .wavs? If it's the latter, then I can simply take the long route and create the .wavs from the .dts, ignoring the BC .wav track and then use Surcode to create a 5.1 dts audio track.

Thanks for the help in advance.

[nautilus7]

From 1st post:

-down6 --> downmix 7 or 8 channels to 6 channels

[wolfbane5]

Alright, so if that's the case, then I need to use -down6. But this leads to my problem of eac3to thinking Surcode isn't installed. Is there a way to use the Arcsoft DTS Decoder manually?

[TinTime]

If you already have a DTS track why decode it and encode to DTS again? Why not just keep it?

[SomeJoe]

1. Why the need to convert DTS-ES into something else? The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions.

2. DTS-ES can come in two forms:

A) DTS-ES 5.1 Matrix, which derives the back center channel from matrix encoding on the Left Surround/Right Surround channels. This is the same way that Dolby Digital EX works. There is no difference between a DTS-ES 5.1 Matrix stream and a standard DTS 5.1 stream -- both contain 5.1 discrete channels of audio.

B) DTS-ES 6.1 Discrete, which actually has a discrete back center channel in the DTS stream. This is not nearly as common as the DTS-ES 5.1 Matrix described above. But this stream will still play back properly on 5.1 equipment, just without the back center channel.

If your current stream is DTS-ES 5.1 Matrix (likely), then there is no need for -down6, since the stream is already 5.1 channels.

[raquete]

The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions. seems that have space for my fast question.
i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
(with or without video, doesn't matter, the target is audio)

Happy New Year ! :)

[wolfbane5]

SomeJoe,

I was converting the DTS-ES into DTS since I don't have a receiver and was planning on watching the movie on my computer, which has 5.1 audio.

As for Matrix or Discrete, MediaInfo is telling me this:

General
Format : DTS
Format/Info : Digital Theater Systems
File size : 1.46 GiB
Duration : 2h 18mn
Overall bit rate : 1 510 Kbps

Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : ES
Duration : 2h 18mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 7 channels
Channel positions : Front: L C R, Rear: L C R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 1.46 GiB (100%)

Am I correct in assuming that's DTS 6.1 Discrete?

[tebasuna51]

You only need eac3to to know this. Check the DTS with eac3to:

DTS-ES, 5.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Matrix)
DTS-ES, 6.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Discrete)

[tebasuna51]

i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded)

[ACrowley]

one question :

I never thought about but can ec3to outout 32bit float .wav and .wavs from any Source ?
Maybe -down32 works as a override ?

EDIT: no...doesnt work.

[TinTime]

You can try using -full.

However I don't think eac3to will ever output at a higher bit depth (i.e. pad with zeroes) than is necessary.

So with the Nero decoder...
eac3to input.ac3 output.wav -full
...will result in 24bit output but...
eac3to input.ac3 output.wav -25.000 -changeTo24.000 -full
...will result in 64bit float output.

[raquete]

The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded) very clever with full details.

thank you so much! :)

[Thunderbolt8]

what does it mean again when I have a 7.1 track with a "strange setup" note? anything I have to look out for? and what was it again with a DTS-HD MA track with -2db dialnorm & eac3to saying "decoding this track with arcsoft resulsts in low volume" (its the same track in both cases)

guess I have a problem now, because arcsoft results in low volume and sonic can only do 5.1?

[xkodi]

Arcsoft can't do bit-perfect decode of DTS-HD MA 7.1 "strange setup" files, while Sonic can but only 5.1 channels, look at this posts and all links in it to understand more:

http://forum.doom9.org/showthread.php?p=1179218#post1179218

[Thunderbolt8]

hm so whats the standing regarding this problem, is madshi willing to do this patch so that arcsoft can properly recognize the channel order?

and does the low volume decoding problem result from this strange channel order or from those -2db dialnorm value?

additionally, the acrsoft decoder says that track has constant 24-bit, while sonic says the 5.1 track is only 16-bit?

[SpaceAgeHero]

Hey folks,

I am having some troubles getting the Arcsoft DTS Decoder to work with eac3to.
Recently I installed Arcsoft Total Media Theatre and updated to 3.0.1.160.
I don't know what version of the dts decoder is included in this package.
Eac3to simply won't recognize the decoder. I'm currently running Windows 7 x64 Ultimate.
I also tried adding "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\" and/or "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\Codec\" to my environment path. c:\Program Files\Common Files\ArcSoft is not there!

Perhaps someone can help me out via E-Mail or private messages.
I do not want to bloat this thread with this issue.
Thank you!

[Leiw]

great program :) works fine for all my conversion needs

[Capsbackup]

@SpaceAgeHero;
Try this thread out, should help you.
http://forum.doom9.org/showthread.php?t=148324&page=2

[PowerGamer]

I have "Pixar Short Films Collection" Bluray. "Mike's New Car" movie on this bluray has the following tracks:

eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls"
1) 00008.mpls, 00006.m2ts, 0:03:49
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
...

I demuxed video and audio tracks with eac3to, tsmuxer and xport. Audio track in .pcm format created by eac3to and xport are bitwise identical. But the same audio track in .wav format produced by eac3to and tsmuxer are different.

Command lines used:

eac3to vs. xport:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" -demux
xport.exe -h "PIX1EGD1\BDMV\STREAM\00006.m2ts" 1 1 1

eac3to vs. tsmuxer:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" 2: audio_eac3to.wav
tsMuxeR.exe 1.meta .\

1.meta file content:
MUXOPT --demux
A_LPCM, "PIX1EGD1\BDMV\STREAM\00006.m2ts", track=4352, lang=eng, mplsFile=00008

Upon further analysis of .wav files created by eac3to and tsmuxer I discovered that these two .wav files contain different value in the dwChannelMask field in the .wav file header: tsmuxer .wav file has 0x0000003F and eac3to .wav file has 0x0000060F. Otherwise headers of both .wav files contain bitwise identical information. In the audio data contained within these two .wav files there were about 880 miscompared bytes (see wav_diff.txt attached to this post).

According to http://www.microsoft.com/whdc/device/audio/multichaud.mspx:

dwChannelMask=0x0000003F (tsmuxer):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, BACK_LEFT, BACK_RIGHT

dwChannelMask=0x0000060F (eac3to):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, SIDE_LEFT, SIDE_RIGHT.

My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?

[Abradoks]

neroAacEnc expects floating-point number as "-q" parameter while eac3to accepts only two decimal places. It is important because bitrate may vary significantly over 0.01, e.g. q=0.37 -> 240kbps, q=0.38 -> 370kbps.

[tebasuna51]

...
My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?
I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

[AnryV]

eac3to v3.17
command line: 00000_1_02.dtshd dtshd.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
Creating file "dtshd.C.wav"...
Creating file "dtshd.R.wav"...
Creating file "dtshd.SR.wav"...
Creating file "dtshd.LFE.wav"...
Creating file "dtshd.L.wav"...
Creating file "dtshd.SL.wav"...
Creating file "dtshd.BR.wav"...
Creating file "dtshd.BL.wav"...


00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?

[tebasuna51]

00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?
SL (Side Left) is Lss (Left Surround Side?), SR is Rss.
BL (Back Left) is Lsr (Left Surround Rear?), BR is Rsr.

[PowerGamer]

I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

Tebasuna51, thanks for your explanations. Meanwhile I tryed to demux that audio track using Blu-Ray Demuxer Pro (http://dvd-logic.com/bddemuxerpro.php) and much to my dismay found out that it produces exactly the same .pcm file as eac3to/xport but completely different .wav file that practically in every byte (apart from header which is the same as tsmuxer's .wav file) differs from both eac3to and tsmuxer versions. Still all .wav files have exactly the same size and play fine in media player.

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro? And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?

[tebasuna51]

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro?
I never support commercial soft.
If the wav audio data output is different (the header may be different), then the Blu-Ray Demuxer Pro is wrong because the eac3to output is exact (not better, is exact).

And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?
The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

[PowerGamer]

The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

Taking that into account (thanks again for explaining) I found out the following (EAC - channel data in .wav file produced by eac3to, BDR - same in Blu-ray Demuxer .wav, PCM - corresponding data in .pcm file):

PCM EAC BDR
0100 0001 0001
0200 0002 0002
0000 0000 0000
ffff 0000 ffff
0000 ffff 0000
0000 0000 0000

ffff ffff ffff
ffff ffff ffff
0000 0000 0000
0100 0001 0001
0000 0001 0000
0100 0000 0001

0000 0000 0000
0000 0000 0000
0000 0000 0000
0000 fffe 0000
0000 0000 0000
feff 0000 fffe

0100 0001 0001
0000 0000 0000
0100 0001 0001
ffff 0003 ffff
0000 ffff 0000
0300 0000 0003

ffff ffff ffff
0100 0001 0001
feff fffe fffe
0000 fffc 0000
0000 0000 0000
fcff 0000 fffc

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).

[tebasuna51]

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).
Like I say before the audio data in .pcm have different channel order than in wav container:

pcm: FL, FR, FC, BL, BR, LF
wav: FL, FR, FC, LF, BL, BR
then the LF channel in PCM (6th) must go to 4th position, BL (4th) to 5th and BR (5th) to 6th.

Seems BDR don't make the channel arrangement.
Is easy to verify the problem, if you play the BDR.wav with a multichannel capable player must listen the Low Frequency channel in the Back Right speaker.

[Stephen R. Savage]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

[TinTime]

@PowerGamer

Specifically the channels must be interleaved in this order (http://www.microsoft.com/whdc/device/audio/multichaud.mspx#EKLAC) for wav pcm.

[tebasuna51]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

Something like that:

v2.84
...
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"

[umaximus]

I got loads of lossless check failed errors on Animatrix's TrueHD track while extracting/transcoding. Are this safe to overlook? How much of ms is failed byte?

eac3to v3.17
command line: eac3to 1) 2: video.vc1 4: audio.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 1:40:50, 24p /1.001
1: Chapters, 10 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, Japanese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
9: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
10: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Dutch
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Portuguese
18: Subtitle (PGS), Japanese
19: Subtitle (PGS), English
20: Subtitle (PGS), Japanese
21: Subtitle (PGS), French
22: Subtitle (PGS), Italian
23: Subtitle (PGS), Italian
24: Subtitle (PGS), Dutch
25: Subtitle (PGS), Spanish
26: Subtitle (PGS), Portuguese
27: Subtitle (PGS), Japanese
28: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Creating file "video.vc1"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:53:49. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:06:51. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:33:01. <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 43ms at playtime 0:18:49. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:28:12. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:45:06. <WARNING>
[a04] Audio overlaps for 42ms at playtime 0:53:50. <WARNING>
[a04] Audio overlaps for 42ms at playtime 1:06:54. <WARNING>
[a04] Audio overlaps for 43ms at playtime 1:33:01. <WARNING>
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Reducing depth from 24 to 16 bits...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[a04] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 145067 frames.
eac3to processing took 3 hours, 23 minutes.
Done.

[AnryV]

I think that eac3to (or ArcSoft decoder ??) uncorrectly decode 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr scheme.


eac3to v3.17
command line: eac3to Ls_for_eac3to.dtshd Ls.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
The original audio track has a constant bit depth of 24 bits.
Creating file "Ls.L.wav"...
Creating file "Ls.R.wav"...
Creating file "Ls.C.wav"...
Creating file "Ls.LFE.wav"...
Creating file "Ls.BL.wav"...
Creating file "Ls.BR.wav"...
Creating file "Ls.SL.wav"...
Creating file "Ls.SR.wav"...
eac3to processing took 4 seconds.
Done.


1. Ls.dtshd - is the file encoded by DTS-HD Encoder Suite from wavs from <Original>. Scheme - 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr
2. Ls_for_eac3to.dtshd - the same file but without header which don't recognized by eac3to
3. <Original> dir - contains original 8 wavs (1 second duration)
4. <eac3to> dir - contains 8 wavs decoded from Ls_for_eac3to.dtshd by eac3to
5. <StreamPlayer> dir - contains 8 wavs decoded from Ls.dtshd by DTS-HD StreamPlayer

As result - <StreamPlayer> fully identical <Original> but BL and BR files from <eac3to> contains not Ls and Rs from <Original> but mixed Ls+Lsr and Rs+Rsr

Sample
http://multi-up.com/201196

[Thunderbolt8]

look at the link of post #9676

[AnryV]

look at the link of post #9676
Problem, as I do understand, was not solved?

[umaximus]

I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?

[Thunderbolt8]

Problem, as I do understand, was not solved?no, apparently there havent been enough discs yet with this problem to justify the work

[bassgoonist]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

[Tiziano]

Sorry, perhaps I'm very dum but I need a help. Sorry if I'm disturbing you continuously

I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.

I'm wondered if anybody can help me giving the correct settings or command line in order to get this .WAV correctly converted ...

I thank you very much

[utenteanonimo64]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

Why don't you use DVD-A Explorer? You can extract the multichannel data or downmix to 2.0. Once you have the WAV files then eac3to will do whatever you want with them...

[tebasuna51]

...
I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.
...

Aften, the ac3 encoder included with eac3to, isn't a Dolby Digital Plus encoder, you can obtain only 5.1 standard ac3.

[tebasuna51]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

To convert a wav file 5.0 you can use WavToAc3Enc or eac3to with external Aften.exe encoder:

eac3to source50 stdout.wav | Aften -pad 0 -readtoeof 1 -exps 32 -s 1 -b 640 - output.ac3

where:
-exps 32 -s 1 -b 640
is the best quality (and slow encode) than Aften can offer.

[deathlord]

Hi

Can I have eac3to use the source filename for the destination?
I mean in the command line, like eac3to source.wav dest.wav

[twazerty]

@madshi:

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?

Noticed that the clipping problem isn't fixed yet. Can we expect a fix?

[blackcell]

For some unknown reason, Eac3to is unable to extract the BD disc "The Express". It gets nearly finished then aborts.

Log file:
eac3to v3.17
command line: "C:\Tools\RipBot\Tools\eac3to\eac3to.exe" "D:\BDMV\STREAM" 1) 1: "C:\tmp\The Express - Chapters.txt" 2: "C:\tmp\The Express 1080p VC-1.mkv" 3: "C:\tmp\The Express - 3 DTS Master.flac" 6: "C:\tmp\The Express - 6 English Subtitle.sup" -log="C:\tmp\The Express - Log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:09:42, 24p /1.001
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
Creating file "C:\tmp\The Express - Chapters.txt"...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[s06] Extracting subtitle track number 6...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\tmp\The Express - 3 DTS Master.flac"...
[s06] Creating file "C:\tmp\The Express - 6 English Subtitle.sup"...
Reading the source file failed. <ERROR>
Aborted at file position 24368906240. <ERROR>

Anyone else seen this issue with other BD's? Any workarounds?

[flyingernst]

Hi,

for a hardwaretest I need a 6.1 Flac (in mkv container with video).
Is there any possibility to get a 100mb sample?

I tried to make a own one, but my mit dts-es 6.1 file got decoded to 5.1 flac, I think because I only use the dts-lib that comes with eac3to

thank you very much

greetings, flyingernst

PS: I tried a 5(6).1 ac3-EX File to, but eac3to don´t finds my nero decoder (BD/HD-DVD Plugin is installed, has a key)

[Inspector.Gadget]

flyingernst, you can make a 6.1 AVS script with Avisynth's SoundOut(), which has its own FLAC export plugin.

[setarip_old]

@blackcell

Hi!

I believe that error would be indicative of a disc that's either dirty or physically damaged. Try cleaning the disc...

[jasonwc]

I'm attempting to convert a DTS-HD MA 7.1 24/96 track to FLAC, but eac3to reports a discontinuity error, and aborts. I know that another individual attempted to convert the same file, and received an identical error. I have done a hash check to ensure that the file is not corrupted on my end. Apparently, the discontinuity is only 11 ms, so I am attempting to override the error, and encode to FLAC anyway.

I found this option that seems promising:

"eac3to v2.34 released

http://madshi.net/eac3to.zip

Code:

* TS/M2TS: new "-ignoreDiscon" option makes eac3to ignore discontinuity errors

As usual eac3to demands the source files to be ultra clean. If eac3to finds any stream errors it will at once stop processing. Since it happens rather often that there are some discontinuity errors in the stream I've now added a new option called "-ignoreDiscon". This way you can force eac3to to continue processing despite of eventual discontinuity errors. I have to warn you, though: When there are discontinuity errors, there's a good chance that there will be artifacts in the video/audio. So use this option with care..."

However, I can't seem to get the option to work. I just get "Command Line paramater "ignoreDiscon" is unknown" when I attempt to demux the m2ts or encode the DTS-HD MA track to FLAC.

I can extract the 24/48 DTS core, but thus far, I've had no luck getting the lossless dtsma track.

Sorry for my ignorance on this topic. I'm new to HD audio. While I haven't had an issue with other disks, this one appears to be corrupt and is therefore giving me issues.

[setarip_old]

@jasonwc

Hi!

If you look at Post #1 of this thread, you'll see that the latest version is v.3.17 - Perhaps your problem no longer exists...

[jasonwc]

@jasonwc

Hi!

If you look at Post #1 of this thread, you'll see that the latest version is v.3.17 - Perhaps your problem no longer exists...

Sorry, I should have been more clear. I am using eac3to 3.17. I was pasting from the change log that added the feature to which I referred.

[tebasuna51]

Read also in the changelog:

v2.45
...
* removed "-ignoreDiscon" from help; hint is shown when a discontinuity occurs

[jasonwc]

I saw that, but I get no hint. It just fails.


eac3to v3.17
command line: eac3to "J:\VIVALDI\BDMV\PLAYLIST\00000.mpls" 3: "G:\vivaldi.flac"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1:11:53, 60i /1.001
1: Chapters, 31 chapters
2: MPEG2, 1080i60 /1.001 (16:9)
3: DTS Master Audio, , 7.1 channels, 24 bits, 96khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\vivaldi.flac"...
[a03] [0:36:07] The source file seems to be damaged (discontinuity). <WARNING>
[a03] The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 3571449856. <ERROR>

[Snowknight26]

Because eac3to can't force the ArcSoft DTS decoder to ignore it. eac3to ignored it, hence the warning.

[umaximus]

Ok, I have found my problem with trueHD tracks. Im using Pioneer receiver and when I was demuxing my trueHD tracks I didnt use -keepdialnorm switch. Now I have over 50 unplayable tracks that I cant use. Is there any software that could re-apply dialnorm?

[Xorp]

Did eac3to mess up the channel mapping when converting to flac:

before: 4: RAW/PCM, Japanese, 3/1 channels, 16 bits, 48khz

after: FLAC, 2/2 channels, 1:02:46, 16 bits, 771kbps, 48khz

Shouldn't it keep it as 3/1? I get also "a04 The AC3 encoder received a non-supported data format (pcm, 4, 16, -)." when trying to convert the both original pcm and the flac to ac3.

[Snowknight26]

FLAC has no 3/1 channel configuration so the closest one (2/2) is used.

eac3to's Aften implementation doesn't support strange channel configurations. Use eac3to's stdout.wav feature and pipe it to Aften. Example here (http://forum.doom9.org/showthread.php?p=1283462#post1283462).

[deathlord]

I saw that, but I get no hint. It just fails.

You could try demuxing the track using xport or tsmuxer, then convert using eac3to. It's quite likely the error will occur again, but maybe it's still worth a try?

[b66pak]

or you can use "-libav" to bypass the arcsoft decoder...
_

[Snowknight26]

...which would only decode the core.

[deathlord]

...which would only decode the core.

...or -sonic, which would decode only 5.1.
But the best bet in such situations is probably reripping the disc, the error might be gone.

[yesgrey]

When I used XP, I was able to run eac3to while I was doing other things. Now, with W7 RC1, I cannot, my computer stalls. I have to start Task Manager and change eac3to's process priority to low, and then it works a little better. Though, I still feel that with XP my computer did not stall at all.
Can any of W7 users share their experience? It might be some problem with multi-tasking handling that only occurs in RC1...

[Snowknight26]

Your HDD is probably being thrashed.

[newday_99]

Feature Request:

Don't suppose you could extract 'S_HDMV/PGS' subtitles to '.SUP' files

(eac3to title00.mkv movie.mkv)

like mkvextract (mkvtoolnix-3.1.0) can now demux/convert ????

(mkvextract tracks title00.mkv 3:track3.sup)

Thanks.

[yesgrey]

Your HDD is probably being thrashed.
Sorry, but I did not understand.:o
Can you elaborate a little more?

[datman]

I just started using this tool with anotherEAC3to and first off it works really good. I only wish I had the wits to use it before a GUI was developed.

I was able to get it to use Archsoft's DTS decoders but not Nero 7's AC3 decoders. I'm not sure how to get it to use the decoder?


Is this still one of the better ways to go?

[consultant]

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

[datman]

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

anotherAC3to GUI will do it http://www.curtpalme.com/forum/viewtopic.php?t=17002&sid=6aa1e7a252df759c18ed801496533911
uses free decoders too

[tebasuna51]

TrueHD to FLAC? Possible?
Yes.
Please put your log file:logfile:

[Beastie Boy]

Your HDD is probably being thrashed.

Using W7 Resource Monitor, I have noticed that eac3to seems to continually write to and read from the swap file, regardless off how much RAM is available. If your swap file is on a drive that you are using, then this causes a bottleneck.

Cheers, Beastie.

[TinTime]

The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound.

Have you tried a different media player? Try MPC with or without the madFlac decoder, or try playing the flac file in foobar.

[mikelebron]

All my TrueHD movies are FLAC.. if your using RipBot then your doing something wrong as Ripbot will convert all HD audio formats to FLAC as it uses eac3to under the covers to demux the video and audio (then re-encodes to flac for the audio)...

Ask in the RipBot Forum to see what you are doing wrong...

If you dont car about compressing the video then just use eac3to command line.. its a multi step process if your dealing with a branching movie..


If you dont want to learn the eac3to command line then use
for mkv

HD DVD/Blu-Ray Stream Extractor - place in directory where eac3to is located. You will need to use mkvmerge (alsno known as mkvtoolnix) after the demux.
http://forum.doom9.org/showthread.php?t=141829


for m2ts, etc. (NOT MKV) - self containing solution
Clown BD:
http://forum.slysoft.com/showthread.php?t=25818

MKV has slightly better file size and allows chapters.. m2ts typically is more fogiving when steaming and can re-assembled into a bluray structure for an ISO.

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

[consultant]

I don't see how I could be doing something wrong since there are only one option in RipBot, check the FLAC box or don't check it. lol

The fact that I demuxed the m2ts and converted the TrueHD to FLAC using USEac3to (select the m2ts, set command line for the TrueHD (3) track to FLAC) and the FLAC file still wouldn't play is indicating to me Eac3to may have an issue with the TrueHD encoding on the Star Trek original movies.

What is interesting is that I can convert the TrueHD to WAV and it plays fine. For kicks, I tried converting the WAV back to FLAC and it won't play. I tried playing it in MediaMonkey also just to make sure it wasn't VLC having the problem.

[stax76]

HD DVD/Blu-Ray Stream Extractor appear to be very buggy with nobody having an interest to fix the bugs.

[consultant]

MPC has killed two birds with one stone. It has less stuttering issues than VLC AND it plays the FLAC audio track correctly! Hurray!

Thanks for your help.

P.S. AnotherAC3To too buggy for me. Couldn't handle directories with spaces in the names for one. Got exception error, etc.

[mikelebron]

Using VLC.. yea that player is garbage... MPC is much better player...

MPC has killed two birds with one stone. It has less stuttering issues than VLC AND it plays the FLAC audio track correctly! Hurray!

Thanks for your help.

P.S. AnotherAC3To too buggy for me. Couldn't handle directories with spaces in the names for one. Got exception error, etc.

[MuLTiTaSK]

UsEac3to: a different GUI for eac3to (bit.ly/cgukqK)

[yesgrey]

Using W7 Resource Monitor, I have noticed that eac3to seems to continually write to and read from the swap file, regardless off how much RAM is available. If your swap file is on a drive that you are using, then this causes a bottleneck.
Thanks for your post, but with my 4GB memory I am running W7 without any swap file, and I also disabled the superfetch service. With XP I also don't use any swap file, but the usage is a lot better...

[Blue_MiSfit]

Erm.. you do know that running any Windows OS without the swap file enabled is generally a bad idea, right? I would suggest keeping it turned on, unless you have solid reproducible results to the contrary (and all your software works without issues)

~MiSfit

[raquete]

yesgrey3,
in my understanding Win7(x86) don't recognize 4Gb Ram but only round 3,25Gb.
even with XP or Win7 is good adjust the swap file....to tell the true, i don't know how to deal/adjust swap file in Win7.

[yesgrey]

Erm.. you do know that running any Windows OS without the swap file enabled is generally a bad idea, right? I would suggest keeping it turned on, unless you have solid reproducible results to the contrary (and all your software works without issues)
I don't have any problems working without the swap file. Only once when I was performing some tests with an application I'm developing and I was allocating a lot of memory. I enabled the swap file and it worked ok.

yesgrey3,
in my understanding Win7(x86) don't recognize 4Gb Ram but only round 3,25Gb.
even with XP or Win7 is good adjust the swap file....to tell the true, i don't know how to deal/adjust swap file in Win7.
I'm Using W7(x64), so it's really 4GB.;)
Adjusting the swap file in W7 is exactly the same as it is in XP.
Control Panel->System->Advanced Settings->Performance->Advanced->Virtual Memory

I think that the problem I'm having should be some kind of bug in W7 RC1 (probably it was already fixed on the retail version), because I already had some problems with my application cr3dlut.
To avoid computer stalling, I have set it's priority to "BELOW_NORMAL_PRIORITY_CLASS". With XP it worked fine. When I tried it in W7, it didn't. I had to set the priority to "IDLE_PRIORITY_CLASS" to avoid the stalling.
That's what I have to do now with eac3to, I have to go to the task manager and set it manually.:(
OTOH, they process handling in W7 is different from XP's, so it could also be something related to that... that's why I asked here if anyone else with W7 retail are getting the same results when using eac3to...

[raquete]

i'm newby in W7 and i'm installing/adjusting in 4 pcs at the same time, thanks for your hints.

i only use pc for audio and never could 'stand' XP, i still use Win2000 and with the rights adjusts in swap file 'everything' run fine, even with some programs that need XP as min system...don't ask me why but i can put some forbidden programs for Win2000 working fine here.
is hard to left what works...
for example: after install W7 with roung 9.3Gb in hd, the system don't recognize my wireless...gee, 9,3Gb around and still need a single dll to put a single wireless working...i can't stand news systems...lol.
my bootalble dvd of Linux Ubuntu recognize the wireless and all cards in my pcs without install anything in hd, how can it be? bootable Ubuntu free versus W7 Ultimate ~U$350,00(here) installing 9,3GB.

in the other side my nephew is video designer of one great tv channel and he told me that using W7(x64) have some problems, some programs don't run and he need to use 2 pcs: one with x86 and other with x64...and with 12GB Ram in each!
then, if you have 2 pcs test one with x86, maybe yours issues can end.

[yesgrey]

i only use pc for audio and never could 'stand' XP, i still use Win2000
I also cannot stand XP GUI. Those very highly saturated colors are irritating. The trick is using the classic GUI theme, which will give you the Win2000 GUI in XP.;)

if you have 2 pcs test one with x86, maybe yours issues can end.
Thanks, but I don't think this is a x86 vs x64 problem. Also, I will not go back to x86, I want a 64bit OS. It's preferable for software developing purposes.

[raquete]

I also cannot stand XP GUI. Those very highly saturated colors are irritating. The trick is using the classic GUI theme, which will give you the Win2000 GUI in XP.;)
:eek: oh...classic GUI theme? really?
i never heard and knew about that possibility. :cool:
can you pm me with the hints to do that please?

cheers!

[BLKMGK]

Has anyone used eac3to to create branching Matroska files? I'm led to believe it's possible using eac3to by a posting to the MakeMKV support forum (http://www.makemkv.com/forum2/viewtopic.php?f=1&t=897) but I cannot figure out how to do it. I've asked elsewhere in the Matroska forums (http://forum.doom9.org/showthread.php?t=72700) for help as well. Currently I rip a BD that has multiple versions multiple times but this really seems silly. MKV apparently supports the ability to branch much like BD does and I'd like to try and take advantage of that as well as urge the XBMC developers to support it. However if it cannot be easily built I won't be able to provide examples nor will anyone be able to use it. :(

Thoughts?

[Snowknight26]

eac3to would have nothing to do with that.

[setarip_old]

@BLKMGK

Hi!

Perhaps you're referring to this post:

http://forum.doom9.org/showpost.php?p=1129751&postcount=4465

But I believe it refers to BD...

[yesgrey]

:eek: oh...classic GUI theme? really?
i never heard and knew about that possibility. :cool:
can you pm me with the hints to do that please?

I've created a thread (http://forum.doom9.org/showthread.php?t=152748) with instructions for both XP and 7, in case other people also do not know how to do it.

[BLKMGK]

eac3to would have nothing to do with that.

Actually yes it would since eac3to is capable of pulling out specific sections and is capable of ripping the various branches. If you had followed the post I linked on the MakeMKV forums you'd have noted that the author specifically mentioned eac3to as one of the tools he used to facilitate the process but provided no details on his process. eac3to builds MKV files, I use it pretty much weekly to do so, I am simply asking if anyone has used it to pull out branches and facilitate creating a file that allowed the various branches to live in one file....

@BLKMGK
Hi!

Perhaps you're referring to this post:

http://forum.doom9.org/showpost.php?p=1129751&postcount=4465

But I believe it refers to BD...

No, this refers to the ability eac3to possesses to follow a specific branch in a BD structure. This is pretty handy if you've got a BD with multiple versions and want just one. What I want is all of them but without the overhead of storing the redundant parts - just like a BD does it...

[n3rdy]

Hello,

In case I would ask a stupid question... please forgive me.
When using eac3to to demux my HDTV records it says

"a remaining delay of -11ms seconds could not be fixed"

So in order to fix that delay in my re-encode, do i have to type :
DelayAudio(-0.011) or DelayAudio(0.011)

(which obviously is a part of my avisynth script)


Thank You !

[tebasuna51]

If you need recode the audio you can use DelayAudio(-0.011), but you can preserve the original audio without recode, -11ms isn't a problem.

[zeropc]

question: i'd like to cut a 22sec and 160ms at the beginning from a dts-hd file. what cmd would i need to use to achieve this with eac3to?
i'm not sure if the delay option is the one i need to use.

thanks for the help :)

[xxx666yyy777]

I normally use v3.14 because latest version has some type of problem where the last line of output to stdout is not present, so my front end cannot parse certain data.

But I have also tried v3.16 and get the same issue.

Apparently, my AV receiver is highly sensitive to a TrueHD stream. Unless -keepDialnorm is used, TrueHD output is very distorted.

It is no big deal, I have dialogue normalization turned off on the AV receiver anyway.

Does anyone know, whether the TrueHD/DialNorm issue still persists with the new Pioneer Receivers (1019 series for example)?

Thx

[stax76]

There is a new GUI in StaxRip:

Download: http://www.stax76.bplaced.net/files/applications/StaxRip/StaxRip_1.1.5.1.7z

http://f.imagehost.org/0756/Source_Files-2010-02-15_04_33_31.png (http://f.imagehost.org/view/0756/Source_Files-2010-02-15_04_33_31)

http://i.imagehost.org/0810/StaxRip-2010-02-15_04_34_06.png (http://i.imagehost.org/view/0810/StaxRip-2010-02-15_04_34_06)

http://f.imagehost.org/0415/SS-2010-02-15_04_38_14.png (http://f.imagehost.org/view/0415/SS-2010-02-15_04_38_14)

[Snowknight26]

I hope you're only demonstrating your GUI's capabilities and not actually trying to output DTS-HD with those options.