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Old 18th January 2009, 10:19   #7901  |  Link
madshi
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Quote:
Originally Posted by starkhouse33 View Post
would it be possible to add a switch into eac3to that can reset/change the video full range flag? some .ts files display the wrong black levels when played on nmt.
Can you give me more information about this? Which flag do you mean exactly and is it set wrong in those TS files?

Quote:
Originally Posted by n0mag!c View Post
Presence of "-silence" makes no difference. Is this a bug?
The behaviour is as intended by me, so I don't consider it a bug. Do you honestly have a real life situation where you need to add 960ms inside of the first second runtime of an audio track?

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Can't recall fully (I'm just the messenger)
I refuse to add any features if I don't even know exactly what they're good for.
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Old 18th January 2009, 10:48   #7902  |  Link
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Quote:
Originally Posted by madshi View Post
The behaviour is as intended by me, so I don't consider it a bug. Do you honestly have a real life situation where you need to add 960ms inside of the first second runtime of an audio track?
This IS a real life situation. I need to apply delay to the beginning of audio track. (I don't like delaycut interface )
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Old 18th January 2009, 11:10   #7903  |  Link
dat720
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Quote:
Originally Posted by madshi View Post
It's already there, just do "eac3to -test".
Nice one, i was using 2.87 and hadn't noticed that.... is it posible to make it do it during normal runtime? (i know it's possible but will you do it?) like show the file version in the normal output, kinda like mplayer does? that way you don't have to run -test to check the version....
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Old 18th January 2009, 11:22   #7904  |  Link
starkhouse33
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Quote:
Originally Posted by madshi View Post
Can you give me more information about this? Which flag do you mean exactly and is it set wrong in those TS files?
h.264 streams broadcast in Europe have the video flag set wrong in their encoder. using eac3to to slow down the video I would then have to run the file through ts4np and check change video full range flag in the options to have the video play correctly on my popcorn hour.

here is a quote that explains in detail what is wrong from a popcorn hour user.
Quote:
Sky and probably other European companies have a bug in their H264 encoder. The sequence header indicates that they use full color range when in reality they are encoding using limited range. Their own STB ignores the flag so their normal customers are not affected.This is not a Sigma or NMT bug and they are doing the correct thing.
I suggest patching the original source files to correct the flag.

Broadcast digital video uses 16-235 levels, whilst PC video uses 0-255 levels. Therefore video black is at 16, and white at 235, whilst PC black is at 0 and white is at 255. Therefore video at 16 is black, and white at 235, whilst the PC is at 0 black and white is at 255th

If you get PC and Broadcast levels confused - or feed broadcast levels to a PC display or vice versa you can end up with crushed blacks, blown out whites, or nasty washed out pictures. If you get PC and broadcast levels confused - or broadcast feed levels to a PC display or vice versa you can end up with crushed blacks, whites blown out, or washed out nasty pictures.

This can be caused by the wrong format being flagged in a header - or a header being mis-interpreted.

All satellite receivers will assume 16-235 - but the video content could be flagged as 0-255 and ignored. All satellite receivers will assume 16-235 - but the video content could be flagged as 0-255 and ignored.
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Old 18th January 2009, 11:46   #7905  |  Link
Thunderbolt8
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Quote:
Originally Posted by madshi View Post
eac3to v3.04 released
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Old 18th January 2009, 12:26   #7906  |  Link
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Quote:
Originally Posted by dat720 View Post
What about the AV recievers that can natively decode the DTS and AC3 varients?
You always can play original BD or at least copy the original audio tracks.
Quote:
Not everyone watches movies on a PC....
FLAC is supported by new HD players like PopCorn Hour and variants.
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Old 18th January 2009, 12:31   #7907  |  Link
jfcarbel
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A few questions:

Quote:
- can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks
- can extract DTS core from DTS-HD tracks
Do the above features require Nero or ArcSoft software? And is trial version of ArcSoft adequate to test it on my machine before I buy?

Quote:
- can extract TrueHD stream from Blu-Ray TrueHD/AC3 tracks
Is this just mean extract the original HD audio stream with no changes? And does this require any 3rd party software?

Last edited by jfcarbel; 18th January 2009 at 13:03.
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Old 18th January 2009, 12:45   #7908  |  Link
Jom
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Quote:
Originally Posted by tebasuna51 View Post
You always can play original BD or at least copy the original audio tracks.

FLAC is supported by new HD players like PopCorn Hour and variants.
Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo
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Old 18th January 2009, 13:35   #7909  |  Link
madshi
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Quote:
Originally Posted by n0mag!c View Post
This IS a real life situation. I need to apply delay to the beginning of audio track. (I don't like delaycut interface )
Well, there is a separate way to do simple audio delays. Simply do e.g. "eac3to source.ac3 dest.ac3 +960ms". The "edit" option is not meant to be used for simple delays.

Quote:
Originally Posted by dat720 View Post
Nice one, i was using 2.87 and hadn't noticed that.... is it posible to make it do it during normal runtime? (i know it's possible but will you do it?) like show the file version in the normal output, kinda like mplayer does? that way you don't have to run -test to check the version....
I don't know. Fetching the latest version number from the internet does cost a bit of time. Can be up to multiple seconds, if the doom9 server is slow. I don't like the idea of having every eac3to run be slowed down by trying to get online...

Quote:
Originally Posted by starkhouse33 View Post
h.264 streams broadcast in Europe have the video flag set wrong in their encoder. using eac3to to slow down the video I would then have to run the file through ts4np and check change video full range flag in the options to have the video play correctly on my popcorn hour.
Ok, I see. I'll add "full range" patching to the next eac3to version. Also I'll post a warning whenever eac3to sees a h264 stream with "full range" set.

Quote:
Originally Posted by jfcarbel View Post
Do the above features require Nero or ArcSoft software?

Is this just mean extract the original HD audio stream with no changes? And does this require any 3rd party software?
All of this just extracts without changes (apart from eventually removing dialnorm information) and doesn't require any 3rd party software.
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Old 18th January 2009, 14:29   #7910  |  Link
tebasuna51
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Quote:
Originally Posted by Jom View Post
Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo
I don't have PCH to answer you but is a firmware question.
Of course can't be passed by SPDIF but can be send by HDMI or multichannel analogic output (Kaiboer)
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Old 18th January 2009, 15:25   #7911  |  Link
bold1342
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PAL speedup bug?

madshi

Excellent tool thank you.

I think I may have encountered a bug. I always speedup the blu ray audio 24 to 25fps.

The processed audio file is always higher bit depth than the original track. For example Harry Potter and the Prizoner of Azkaban blu ray has a 16 bit RAW audio track, when processed with eac3to the finished file reports that it is 32 bit?

Please see log file below

eac3to v3.04
command line: "C:\Users\dmb_laptop\Desktop\eac3to 3.04\eac3to.exe" "F:\POTTER 3\00103.m2ts" 3: "F:\POTTER 3\harry .dts" -speedup
------------------------------------------------------------------------------
M2TS, 1 video track, 14 audio tracks, 29 subtitle tracks, 2:21:42, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: RAW/PCM, 5.1 channels, 16 bits, 48khz
4: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
12: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
13: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
14: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
15: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
16: Subtitle (PGS)
17: Subtitle (PGS)
18: Subtitle (PGS)
19: Subtitle (PGS)
20: Subtitle (PGS)
21: Subtitle (PGS)
22: Subtitle (PGS)
23: Subtitle (PGS)
24: Subtitle (PGS)
25: Subtitle (PGS)
26: Subtitle (PGS)
27: Subtitle (PGS)
28: Subtitle (PGS)
29: Subtitle (PGS)
30: Subtitle (PGS)
31: Subtitle (PGS)
32: Subtitle (PGS)
33: Subtitle (PGS)
34: Subtitle (PGS)
35: Subtitle (PGS)
36: Subtitle (PGS)
37: Subtitle (PGS)
38: Subtitle (PGS)
39: Subtitle (PGS)
40: Subtitle (PGS)
41: Subtitle (PGS)
42: Subtitle (PGS)
43: Subtitle (PGS)
44: Subtitle (PGS)
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Changing FPS from 23.976 to 25.000...
[a03] Reducing depth from 64 to 32 bits...
[a03] Writing WAVs...
[a03] Creating file "F:\POTTER 3\harry .SL.wav"...
[a03] Creating file "F:\POTTER 3\harry .R.wav"...
[a03] Creating file "F:\POTTER 3\harry .LFE.wav"...
[a03] Creating file "F:\POTTER 3\harry .SR.wav"...
[a03] Creating file "F:\POTTER 3\harry .L.wav"...
[a03] Creating file "F:\POTTER 3\harry .C.wav"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] The processed audio track has a constant bit depth of 32 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...


Any help would be appreciated.


Thanks


David

Last edited by bold1342; 18th January 2009 at 17:08.
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Old 18th January 2009, 17:10   #7912  |  Link
madshi
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Quote:
Originally Posted by bold1342 View Post
I think I may have encountered a bug. I always speedup the blu ray audio 24 to 25fps.

The processed audio file is always higher bit depth than the original track. For example Harry Potter and the Prizoner of Azkaban blu ray has a 16 bit RAW audio track, when processed with eac3to the finished file reports that it is 32 bit?
That's just fine. Speeding up audio properly requires a full floating point resampling process, which even ends up with 64bit data. eac3to then reduces the 64bit floating point data to whatever seems most appropriate. If you want to end up with only 16bit data, you can use "-down16". However, that doesn't make any sense if you want to use Surcode to encode to DTS, because when doing DTS encoding the higher the bitdepth you feed into the encoder, the better. That results in higher audio quality without any size increase...
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Old 18th January 2009, 17:40   #7913  |  Link
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i have quite an interesting lossless track where the bitdepth analyser doesn't tell the whole truth unless i convert it to separate waves and process those one by one.. i'm not sure though if anyone except me actually cares about that sort of thing :P

Code:
C:\Program Files\eac3to>eac3to E:\aviator.flac f:\aviator.wavs
FLAC, 5.1 channels, 2:50:07, 24 bits, 1691kbps, 48khz
Decoding FLAC...
Writing WAVs...
Creating file "f:\aviator.L.wav"...
Creating file "f:\aviator.LFE.wav"...
Creating file "f:\aviator.SR.wav"...
Creating file "f:\aviator.R.wav"...
Creating file "f:\aviator.C.wav"...
Creating file "f:\aviator.SL.wav"...
Original audio track: max 24 bits, average 18 bits, most common 16 bits.
eac3to processing took 4 minutes, 28 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\aviator.L.wav aviator.L.wav
WAV, 1.0 channels, 2:50:07, 24 bits, 1152kbps, 48khz
Reading WAV...
Writing WAV...
Creating file "aviator.L.wav"...
Original audio track: max 24 bits, average 17 bits, most common 16 bits.
eac3to processing took 43 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\aviator.LFE.wav aviator.LFE.wav
Original audio track: max 24 bits, average 16 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.SR.wav aviator.SR.wav
Original audio track: max 24 bits, average 18 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.R.wav aviator.R.wav
Original audio track: max 24 bits, average 17 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.C.wav aviator.C.wav
The original audio track has a constant bit depth of 24 bits.

C:\Program Files\eac3to>eac3to f:\aviator.SL.wav aviator.SL.wav
Original audio track: max 24 bits, average 18 bits, most common 16 bits.
it's from the french aviator bluray. what makes it even more interesting is how much bitrate it takes on the disc.

DTS-HD Master Audio English 5.1 / 48 kHz / 3545 kbps / 24-bit (DTS Core: 5.1 / 48 kHz / 1509 kbps / 24-bit)

i was really surprised how much lower the bitrate was when re-encoded to flac.. unless this is some sort of freak unique case, it would seem that dts-hd ma is really bad at compressing variable-bitdepth material.
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Old 18th January 2009, 19:02   #7914  |  Link
Jeff Flowerday
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Quote:
Originally Posted by Jom View Post
Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo
It doesn't support multichannel flac in mkv at all.
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Old 18th January 2009, 19:05   #7915  |  Link
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Quote:
Originally Posted by madshi View Post
What did madFlac do? It should play the file just fine - just as 5.0 and the audio renderer may not like this format. You can try chaining the ffdshow raw audio processor into your playback chain and letting is add an empty LFE channel or something like that...
Nothing really. If I drop the mkv in graphedit I get an error that it doesn't know which filter to use on the 5.0 flac. PS) I had the same issue with 6.1 flac I had to redoo the flac as 7.1

I guess I should post this kind of stuff in your madflac thread.
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Old 18th January 2009, 19:22   #7916  |  Link
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I'm going to second the request for a -plain switch to chapters extraction. The reason is that tsMuxer likes just the timestamp in each chapter line when muxing. This would make muxing a lot quicker if we didn't have to edit out each non-timestamp line (ex Chapter1=)
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Old 18th January 2009, 19:25   #7917  |  Link
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Hi everyone!

I'm trying to cut the first 28.7 seconds out of an audio file, so I'm doing this:

eac3to input.wav output.aac -edit=0:00:00,28700ms

eac3to complains that this is an "invalid edit format", though. Can anyone tell me what the correct syntax is? I'm using madshi's latest (3.04).

Thanks!
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Old 18th January 2009, 19:29   #7918  |  Link
Thunderbolt8
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as posted before, when you want to cut/insert something at the beginning, then just use the delay function and not the cut function.
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Old 18th January 2009, 19:32   #7919  |  Link
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Originally Posted by Thunderbolt8 View Post
as posted before, when you want to cut/insert something at the beginning, then just use the delay function and not the cut function.
I see - I thought a "delay" was different from a "cut" (you normally wouldn't lose any audio with a "delay" - it would simply be moved to a later starting point within the stream).

I'll give it a try, thanks.
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Old 18th January 2009, 19:36   #7920  |  Link
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Quote:
Originally Posted by omf View Post
I see - I thought a "delay" was different from a "cut" (you normally wouldn't lose any audio with a "delay" - it would simply be moved to a later starting point within the stream).

I'll give it a try, thanks.
Actually, a negative delay will cut off the first part of an audio stream and a positive delay will pad silence at the start of an audio file.
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