eac3to - audio conversion tool

[madshi]

I've noted two titles that break eac3to both 2_81 and 2_82 as well as 2_83. Seems to relate to handling over audio gaps/overlaps. Eac3 eventually fails with "The temp file could not be interpreted correctly."
This will be fixed in the next build.

I have a disk that -demux es OK with V2.79 and not V2.83
This will also be fixed in the next build.

[madshi]

I have the same disc. v2.83 doesn't find any gaps for the embedded ac3 in the truehd track (#4). Isn't it weird?
I've just tried with the first 4 m2ts parts (not full movie to save processing time) of the unrated cut and I'm already getting an overlap in the embedded AC3 track:

eac3to 00123.m2ts+00141.m2ts+00125.m2ts+00142.m2ts
3: test.ac3
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 0:27:55
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
5: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
6: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
7: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
[a03] Extracting audio track number 3...
[a03] Extracting AC3 stream...
[a03] Creating file "test.ac3"...
[a03] Audio overlaps for 27ms at playtime 0:05:51.
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "test.ac3"...
eac3to processing took 6 minutes, 37 seconds.
Done.
Are you sure that you used the correct command line for testing?

[madshi]

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i.
I've double checked my code and I don't see anything wrong. 1088i should be cropped to 1080i by eac3to (and in my tests it works as expected). Which tool shows you 1084i? Are you sure that that tool is showing the correct information? What does eac3to say about the final demuxed video track? Does it also report 1084i? A sample would be very helpful. Thanks...

[madshi]

got fail sync
This will also be fixed in the next build.

[madshi]

eac3to v2.84 released

http://madshi.net/eac3to.zip

* fixed: 2nd pass gap removal was tried (and failed) for TrueHD+AC3 targets
* fixed: processing aborted when trying to fix gaps in PCM destination files
* fixed: more than one RAW/PCM overlaps resulted in lost sync (since v2.81)
* fixed: demuxing TrueHD+AC3 stream by title number didn't renew the AC3 part
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"
* title sorting criteria changed: resolution is more important than runtime
* new option "-lowPriority" sets eac3to to background/idle priority
* libav warnings are now assigned to the affected audio track
* fixed: "lossless check failed" false alarms for seamless branching movies
* fixed: spike removal filter was not active for the very last overlap/gap
* improved muxing h264 streams which begin with double sequence headers
* source files are now opened with "share read + write access"
* destination files are now opened with "share read access"
Hopefully the new 2 pass processing logic introduced in v2.81 is working reliable now...

The new "-edit" option can be useful to manually sync audio tracks to movies. E.g. if you need to remove 150ms worth of audio data at the runtime of exactly 30 minutes, you can do this: "-edit=0:30:00,-150ms". Or if you need to add/loop 100ms audio data at runtime 15 minutes, 10 seconds and 500 milliseconds, you can do: "-edit=0:15:10.500,+100ms". You can only do one edit at a time, though. The editing feature should work for all supported audio formats.

[jj666]

Editing feature is great news for anyone syncing DVD audio to satellite caps. Thanks very much Madshi! When will you open up a facility for donations ;-)

Cheers,

-jj-

[Thunderbolt8]

sound great! :thanks:

[mrr19121970]

This will also be fixed in the next build.


It works, thanks....

[mrr19121970]

Is there a simple way to get the track details (like this):

M2TS, 1 video track, 8 audio tracks, 25 subtitle tracks, 1:56:52
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
4: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3, Portuguese, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), German
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), German
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), Portuguese
19: Subtitle (PGS), German
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), German
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), German
26: Subtitle (PGS), Spanish
27: Subtitle (PGS), Portuguese
28: Subtitle (PGS), Turkish
29: Subtitle (PGS), French
30: Subtitle (PGS), Dutch
31: Subtitle (PGS), Korean
32: Subtitle (PGS), English
33: Subtitle (PGS), French
34: Subtitle (PGS), Dutch
35: Subtitle (PGS), Korean

without -demux ing, and the pausing ? as I ultimately only want to do this ?

eac3t0 R:\ 1) 2: Movie.h264 3: English.ac3 4: German.ac3

Thanks in advance. Mike.

[rebkell]

Editing feature is great news for anyone syncing DVD audio to satellite caps. Thanks very much Madshi! When will you open up a facility for donations ;-)

Cheers,

-jj-

yes, that is a great feature, I've got a few captures right now, I'll probably try to get it working on, They are pretty close, but they need a tweak here and there.

[cavediver]

1) Yes: eac3to inputfile outputfile (*.dtshd/dts) -both will give the same result except you use -core option. "-core" extracts the core not DTS-HD.
2) -If not: Arcsoft TMT. (or Sonic Cinemaster)

Thank you.

[madshi]

Is there a simple way to get the track details [...] without -demux ing [...]?
Yes:

eac3to R:\ 1)

[frenchglen]

Has anyone tried decoding the 6.1ch DTS-HD MA track on the Wall-e Blu-ray?
Sonic thinks it's only 5.1ch. :(

[asarian]

Has anyone tried decoding the 6.1ch DTS-HD MA track on the Wall-e Blu-ray?
Sonic thinks it's only 5.1ch. :(

Probably because it is 5.1. ;)

eac3to 20000.m2ts
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3 Surround, 2.0 channels, 192kbps, 48khz
6: AC3 Surround, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)

[Thunderbolt8]

perhaps it only switches to 6.1 at a later stage of the movie.

[asarian]

perhaps it only switches to 6.1 at a later stage of the movie.

I doubt it. That's the m2ts for the main title. I don't see how it could change to 6.1 midstream.

EDIT: Could be they released a disc with different tracks in France, of course.

[rebkell]

eac3to v2.84 released

http://madshi.net/eac3to.zip

The new "-edit" option can be useful to manually sync audio tracks to movies. E.g. if you need to remove 150ms worth of audio data at the runtime of exactly 30 minutes, you can do this: "-edit=0:30:00,-150ms". Or if you need to add/loop 100ms audio data at runtime 15 minutes, 10 seconds and 500 milliseconds, you can do: "-edit=0:15:10.500,+100ms". You can only do one edit at a time, though. The editing feature should work for all supported audio formats.

I assume this will work like the delay, if I have an AC3@48KHz, it would drop 3 32msec frames if I put in -100ms, and add 96ms if I put in +100ms. I really love this feature, I've needed this for a long time. Now I've got something to fix my encodes that end with random sync problems because of corrupted ts streams, now I just have to figure out how much I need to tweak to get things back in sync.

[madshi]

I assume this will work like the delay, if I have an AC3@48KHz, it would drop 3 32msec frames if I put in -100ms, and add 96ms if I put in +100ms.
Yes, that's correct. The new "edit" feature is basically just reusing the same code which is fixing gaps/overlaps in seamless branching movies.

[odin24]

This might be a matter of preference, what would produce better quality audio... A HDDVD with EAC3 @ 1536kb/s with dialnorm or that same track transcoded to DTS @1536kb/s for Blu-ray. Would there would be some quality loss in the conversion or would all of the data be retained?

Thanks.

[Atak_Snajpera]

It's like asking which track sounds better mp3 @ 320 kbps or aac @ 320 kbps (or which is better AVC @ 50 Mbps or VC-1 @ 50Mbps?) Conclusion is that you won't hear any difference. (assuming that your source is uncompressed)
BTW. Transcoding from one lossy format to another always means loss in quality.

[williewonton]

@madshi

Version 2.84 works a treat. Thanks.

[nautilus7]

I've just tried with the first 4 m2ts parts (not full movie to save processing time) of the unrated cut and I'm already getting an overlap in the embedded AC3 track:

eac3to 00123.m2ts+00141.m2ts+00125.m2ts+00142.m2ts
3: test.ac3
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 0:27:55
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
5: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
6: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
7: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
[a03] Extracting audio track number 3...
[a03] Extracting AC3 stream...
[a03] Creating file "test.ac3"...
[a03] Audio overlaps for 27ms at playtime 0:05:51.
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "test.ac3"...
eac3to processing took 6 minutes, 37 seconds.
Done.
Are you sure that you used the correct command line for testing?
Tested again, but with 2.84. And it finds many gaps, so i guess there's nothing to worry about now.

[odin24]

It's like asking which track sounds better mp3 @ 320 kbps or aac @ 320 kbps (or which is better AVC @ 50 Mbps or VC-1 @ 50Mbps?) Conclusion is that you won't hear any difference. (assuming that your source is uncompressed)
BTW. Transcoding from one lossy format to another always means loss in quality.


OK then, thanks for the reply.

[krosswindz]

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.

[Chrishel]

First off, I want to thank madshi for creating such a robust conversion program! I'm trying to archive my high-resolution audio collection, and eac3to works great for converting mlps from DVD-Audios, but I can't seem to get it to work easily with my DTS CDs.

I start off by ripping the DTS CDs to dtswavs using Exact Audio Copy. The subsequent dtswavs are detected as having DTS streams and play as they should through directshow filters, but when I try to use eac3to to convert them, it treats them as 2 channel wav files. Here is the contents of the log file eac3to produces:


eac3to v2.84
command line: "\Program Files (x86)\eac3to\eac3to.exe" "01 - Tequila Sunrise.dtswav" "01 - Tequila Sunrise.flac"
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:03:26, 16 bits, 1411kbps, 44.1khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "01 - Tequila Sunrise.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 6 seconds.
Done.


If I open the dtswav in DTSParser, it detects the file as having an open bitrate, with a format of DTS 20/44.1 in 6.1. If I use DTSParser to rebuild the stream and save a .dts file, the subsequent dts file also plays fine, but eac3to gives the error "The format of the source file could not be detected.". In case it is relevant, the original dtswav is 34.6 MB and the rebuilt dts file is 30.2 MB.

The only way I've found to come close to completing this conversion is to use VLC to save a multichannel wav file, and then use eac3to to convert that. However, the channels from that appear to be 16-bit, and according to DTSParser, the original is 20-bit, so I'd rather not lose the quality if I can avoid it.

Do you have any suggestions as to anything I could try to get eac3to to detect the dtswavs properly. Is it possible Exact Audio Copy is adding or removing some header information which is interfering with eac3to's ability to detect them?

[Rectal Prolapse]

madshi, can you enter a more finely-grained timestamp for the audio editing? For example, to the nearest 100th of a millisecond? Instead of "01:30:02" can it be "01:30:02.250"?

Thanks!

[Snowknight26]

Should probably read his post again.
http://forum.doom9.org/showpost.php?p=1226437&postcount=7460

[bmnot]

madshi, eac3to is hands down one of the best designed and supported pieces of software I've ever used. Is there anyway I can donate a little something for all the hard work you put into it?

[williewonton]

madshi

While 2.84 isn't broken by the title "Game Plan" I've found that the audio isn't correctly sequenced. I'll do some more looking and see if I can make sense of it.

[Rectal Prolapse]

Should probably read his post again.
http://forum.doom9.org/showpost.php?p=1226437&postcount=7460

Hmm somehow I missed that. Everything afterwards never used that example. (Signal-to-noise is kind of low in this thread lol)

Thank you, snow kiniggit! You're my *hero*. :devil:

[madshi]

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.
Why does the patching bother you?

Do you have any suggestions as to anything I could try to get eac3to to detect the dtswavs properly.
Yes. Make a sample available to me... ;)

madshi, eac3to is hands down one of the best designed and supported pieces of software I've ever used. Is there anyway I can donate a little something for all the hard work you put into it?
Thanks! And no, right now there's no way to donate, cause such a donation would be difficult for me to handle tax wise. But I might change my mind next year...

While 2.84 isn't broken by the title "Game Plan" I've found that the audio isn't correctly sequenced. I'll do some more looking and see if I can make sense of it.
What do you mean exactly with "correctly sequenced"?

[Chrishel]

Yes. Make a sample available to me... ;)

Check your PMs. :-)

[madshi]

Check your PMs. :-)
I'm checking my PMs, anyway, so there's no need to flood this thread with such posts.

-------

Your DTSWAV sample is strange. It begin with a lot of zero data and then half a DTS frame and then finally with the real DTS data. eac3to cannot automatically detect such a file as a valid DTSWAV file. Don't know, maybe the EAC extra features (which are very useful for ripping normal WAV data) are harmful for DTSWAV tracks? You could try playing around with the EAC options to see whether that makes any difference.

Furthermore the DTS data in the DTSWAV file is kind of broken. That's probably not the fault of EAC. The framesize is supposed to be 3585 bytes, but the data only contains 3584 bytes per frame. That's why eac3to didn't accept the DTS file extracted by DTSParser, either. I'll implement a workaround for such broken DTS files into the next eac3to build. So the next eac3to build will be able to detect the DTSParser extracted DTS file as:

DTS, 5.1 channels, 0:03:25, 20 bits, 1235kbps, 44.1khz

Also decoding will be possible with ArcSoft, libav and Nero, but not with Sonic.

[Chrishel]

Thanks so much!

[tebasuna51]

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.

Maybe you are confused, only DTS MA (lossless) can have a exact bitdepth.

The core, or a downmix to 5.1 reencoded to a standard DTS (lossy), don't have any bitdepth, the samples are stored in frequency domain with a equivalent precission to 20-24 bits in uncompressed format, not matter if the source is 16 or 24 bits.

A -16 switch with a dts output (or other lossy format) have not sense.

[krosswindz]

Why does the patching bother you?


If I can say bit depth is something equivalent to resolution of the sample stored. If the source is 16bit then while re-encoding if it is re-encoded as 24bit would this mean the encoder is filling up the missing data.

Maybe you are confused, only DTS MA (lossless) can have a exact bitdepth.

The core, or a downmix to 5.1 reencoded to a standard DTS (lossy), don't have any bitdepth, the samples are stored in frequency domain with a equivalent precission to 20-24 bits in uncompressed format, not matter if the source is 16 or 24 bits.

A -16 switch with a dts output (or other lossy format) have not sense.

Very true I am confused on this.

What I am trying to do is to keep everything at the same except downmix to 5.1ch keeping every other same as possible.

edit: @madshi it would be great if there was an option to prevent patching and let the default be that the audio be patched.

[tebasuna51]

If I can say bit depth is something equivalent to resolution of the sample stored. If the source is 16bit then while re-encoding if it is re-encoded as 24bit would this mean the encoder is filling up the missing data.

Still confused. After a lossy encode forget the source resolution, if we can recover the source resolution (even if is poor) we have a lossless encoder.

For a lossy output the best procedure is, like madshi know, decode to high resolution (64 or 32 bit float) make functions (mix, speed, resample, ..) with max resolution and, at end, down to the max resolution supported by the encoder (24 bit int for Surcode DTS, 32 bit float for NeroAacEnc, ...).

In lossless formats the precision is know by the bitdepth, in lossy formats by the bitrate (forget the bitdepth).

[Thunderbolt8]

its basically that the sound quality in it remains at 16-bit, but it seems like the way it is stored can only be 24-bit in that case. so theres no downmixing or artificial upgrading of the original quality, it remains untouched, its only the 'package' which transports that sound content which gets changed. but apparently it cannot be made another way.

(correct me if im wrong)

[williewonton]

madshi

The video runs apparently OK, but the audio track repeats the opening theme over and over and doesn't progress to the dialog etc. So it is an odd one. In a seamless branching title, what is the maximum you allow for/handle?

[madshi]

If I can say bit depth is something equivalent to resolution of the sample stored.
If you're talking about a LPCM or WAV track then you're right. If you're talking about a lossy DTS track then you're flat out wrong. As tebasuna51 already explained, DTS tracks do not store samples in any specific bitdepth. The DTS "bitdepth" header field only has pure informational character and tells us which bitdepth the original PCM audio master had which was fed into the DTS encoder. But the DTS track itself is not bound to any specific bitdepth.

@madshi it would be great if there was an option to prevent patching
No, because IMO such an option would be totally useless. If you find a technically correct argument for adding such an option, then please let me know. But right now I don't see any such argument...

The video runs apparently OK, but the audio track repeats the opening theme over and over and doesn't progress to the dialog etc. So it is an odd one. In a seamless branching title, what is the maximum you allow for/handle?
There's no max. You can have thousands of m2ts parts, no problem, just takes longer.

I remember that there was a Blu-Ray disc which really had a broken audio track just like you describe it. Are you sure that eac3to borked up this audio track? I rather guess that the audio data in the m2ts files is broken. However, IIRC there's another audio track which is working fine. So please check all audio tracks. You may find one which is correctly working...

[krosswindz]

@madshi/tebasuna51 thanks for the explanation, I didnt know that the DTS "bitdepth" header field was only for informational purpose. I know have a better understanding of what you mean.

@madshi: Really love your tool. Is there any future plan for supporting DTS pro series encoders?

[rica]

Transcoding from one lossy format to another always means loss in quality.

Not always, this is a myth which everybody believes in.

It is up to the capabilities of mostly decoder, plus encoder.

I have lots of dts samples sound better than the original ac3s.

Edit: Before getting the objections, i should say:
Why do you use an Hi-End CD Player while you can listen the same CD on an ordinary player?
Because of its HW decoder (plus of its DAC); correct?
So if you can match the Hi-Fi decoder filter with an appropriate encoder filter; why not?
This is why i said "decoders are more important than the encoders in transcoding" before...
And madshi knows what they are and selected those Hi-Fi decoders as default in his tool...
So extracting wavs using an Hi-Fi decoder and re-encoding them with an external Pro encoder would be the best choice.
_ _ _ _ _ _

[tebasuna51]

Not always, this is a myth which everybody believes in.
Well you can filter noise or modify anything wrong in the original. But, with a correct source, isn't a myth.
It is up to the capabilities of mostly decoder, plus encoder.
No.
I have lots of dts samples sound better than the original ac3s.
Please check your players settings.
"decoders are more important than the encoders in transcoding"
No.

[rica]

"No" would not explain anything.
Except telling the same story...

[alc0re]

Sorry if this has been asked before or if it is a stupid question :

I have been using eac3to to extract video/audio/subs from bluray disks.

I noticed something though and I have a question.

What exactly does the removal of dialog normalization do?

I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything. If I fast forward at like 1.5x realtime where you can still hear the audio its normal level, but as soon as you hit play again it drops the audio down again.

I blamed a PS3 firmware update for awhile but I exchanged my PS3 for a panasonic DMP-BD35K bluray player and it has the same behavior.

I finally got around to trying the -keepDialnorm and now I can hear the audio with my receiver's volume at a normal range. What exactly is removing the dialog normalization buying me and why is it not recommended to use the -keepDialnorm switch? Is there a way to extract the ac3 audio that already has had its dialog normalization removed from my dvd9s and do the reverse process that removing the dialog normalization does? Anyone else have/seen this issue? Perhaps its my receiver? (Yamaha HTR-5940)

[Mercury_22]

Sorry for n00b question but How can I demux to specific location using "-demux " command ?

:helpful: :thanks:

[madshi]

Is there any future plan for supporting DTS pro series encoders?
There is AGM output support (for older DTS pro encoders). Don't know if anything more than that will come.

Not always, this is a myth which everybody believes in.
No, it is not a myth. It's a technical fact that every encoding process to a lossy format throws away information. There may be situations where transcoding from lossy to lossy can result in an improvement. E.g. (as tebasuna51 said) if you post process the audio data in a good way. Or e.g. if one of the decoders in your receiver is broken. But these are really exceptions. It's a fact that every straight transcoding from lossy to lossy format loses audio information.

What exactly does the removal of dialog normalization do?
Dialnorm can be set to any value between 0 and 31. According to the AC3 specification both 0 and 31 means: No dialnorm processing. Now any dialnorm processing *lowers* the volume of the audio track. That means removing the dialnorm (which is what eac3to is doing) should result in *higher* volume. Currently eac3to sets dialnorm to 0. Unfortunetely dialnorm set to 1 means lowering volume a lot. So incorrectly working decoders might think that a dialnorm value of 0 means even lower volume than dialnorm 1. But the documentation clearly states that a dialnorm value of 0 shall be treated as "no dialnorm processing" (which means max volume). And all the PC AC3 decoders correctly see value 0 as "dialnorm processing deactivated".

I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything. If I fast forward at like 1.5x realtime where you can still hear the audio its normal level, but as soon as you hit play again it drops the audio down again.

I blamed a PS3 firmware update for awhile but I exchanged my PS3 for a panasonic DMP-BD35K bluray player and it has the same behavior.
IMHO the decoders in the PS3 and Panasonic are not working correctly. Or maybe my AC3 specification is outdated? Anyway, the documentation clearly says that dialnorm 0 is "reserved". So I think it's not really good that eac3to uses it. That means I'll change it to 31 in the next build. I think that should fix the problem you're seeing. However, I believe to remember that some Sony Blu-Rays had a dialnorm value of 0, too. Well, anyway...

[madshi]

Sorry for n00b question but How can I demux to specific location using "-demux " command ?
By changing the command prompt "current directory" to the wanted location. (e.g. "d:", followed by "cd movies").

[madshi]

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i. Why is that? Is there a way to leave it alone or else change to 1080?
I've done several checks:

(1) MediaInfo reports 1920x1088 for the original video track. And it reports 1920x1080 for the eac3to processed track.

(2) When playing back the original TS file with the Cyberlink h264 decoder, the video renderer gets 1920x1088 pixels. When playing back the eac3to processed video track with the same decoder, the video renderer gets 1920x1080 pixels.

(3) Same as (2), but with Sonic h264 decoder.

(4) Same as (2), but with ffmpeg/libav h264 decoder.

I think that are enough proofs that eac3to's processing is 100% correct. The cropping down to 1080 from 1088 is a (surprisingly) complicated calculation and tsMuxeR evidently does it in the wrong way. It's so complicated because the cropping size must be multiplied several times, depending on some conditions. E.g. with your sample, eac3to writes a cropping value of "2" into the video bitstream. This must be multipled by 2 twice to get to the correct cropping size of "8". tsMuxeR seems to forget one of the multiplications, that's why it incorrectly reports 1084i instead of 1080i.

Short summary: This is a(nother) bug in tsMuxeR.
Well, I suppose that is sort of good news then ;)

Hopefully, tsMuxeR will be corrected for this and a few other things soon (such as 25.01 FPS output and THD handling). In the meantime, do you think it is safe to use tsMuxeR to mux such files? That is, is it only reporting wrong but not altering them?
I don't know. Why do people keep asking me questions about tsMuxeR?

Do you plan to add TS and/or M2TS output to eac3to?
No.

Also, I have not really noticed any problem with playback of most most 1088 AVC files however 1088 MPEG-2 with DXVA often display a grey bar in MPC-HC but not in PowerDVD. Can eac3to correct those MPEG-2?
No. In MPEG2 it's not as easy as in h264.

Is there any potential drawback to eac3to automatically processing to 1080 (with either codec)?
I see none. Actually I think it has several advantages. E.g. with the original file the video renderer actually gets 1088 lines. What will the renderer do with that? Display all of them? Then it must downscale the image, which means we don't have 1:1 pixel mapping, anymore. With the eac3to cropped video bitstream the video renderer is only getting 1080 lines, so we get perfect 1:1 pixel mapping. You can see that for yourself: If you play the original TS file, you'll see some garbage lines at the bottom of the screen. With the eac3to processed stream these garbage lines are gone.

[Mercury_22]

By changing the command prompt "current directory" to the wanted location. (e.g. "d:", followed by "cd movies").

Thanks !
:thanks: