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Old 11th December 2008, 22:34   #7321  |  Link
digitlman
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Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks
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Old 12th December 2008, 10:44   #7322  |  Link
deathlord
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madshi,
Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):
Code:
eac3to 1) 2: g:\Simpsons\Video.mkv 3: g:\Simpsons\Engli
sh_5.1_dts-hdma.flac -down16
M2TS, 1 video track, 9 audio tracks, 24 subtitle tracks, 1:41:28
1: Chapters, 25 chapters
2: h264/AVC, 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, French, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
6: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, Finnish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, Swedish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
11: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), French
16: Subtitle (PGS), German
17: Subtitle (PGS), German
18: Subtitle (PGS), Dutch
19: Subtitle (PGS), Dutch
20: Subtitle (PGS), Finnish
21: Subtitle (PGS), Finnish
22: Subtitle (PGS), Swedish
23: Subtitle (PGS), Swedish
24: Subtitle (PGS), English
25: Subtitle (PGS), French
26: Subtitle (PGS), German
27: Subtitle (PGS), Dutch
28: Subtitle (PGS), Finnish
29: Subtitle (PGS), Swedish
30: Subtitle (PGS), English
31: Subtitle (PGS), French
32: Subtitle (PGS), German
33: Subtitle (PGS), Dutch
34: Subtitle (PGS), Finnish
35: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "g:\Simpsons\English_5.1_dts-hdma.flac"...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.
Do you need a sample, and if yes, what exactly should I make the sample from?

Also, this movie is supposed to have force subs. But how do I see which it is?
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Old 12th December 2008, 11:41   #7323  |  Link
madshi
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Quote:
Originally Posted by Thunderbolt8 View Post
got a problem with a TrueHD track here:

50mb sample: http://www.sendspace.com/file/y2diub

Code:
[a03] This track is not clean.
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.
Obviously the source file is not clean. The libav decoder doesn't seem to be able to handle this situation. Are you sure that the rip is not corrupt? I'd strongly suggest reripping the disk.

Quote:
Originally Posted by Thunderbolt8 View Post
and when I do it with -nero (nero also decodes TrueHD 5.1 completely lossless, right?) then I get:

and after re-running the cmd line:

Code:
Audio gap description file detected, will be used for processing...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, LFE: max 23 bits, average 16 bits.
so regarding the difference between the original and the processed track with nero at the re-run, is the resulting flac not lossess or whats going on there?
I've recently added a post processing filter which smooths the audio cut points where gaps/overlaps are fixed to remove audio spikes. This post processing filter creates 24bit samples. That's why the bitdepth analyzation reports different results after the gaps have been fixed. In your specific case that's no problem cause the stream was overall (up to) 24bit, anyway. So it's still practically lossless. But for 16bit the behavior of the smoothing filter is not very good. So I'll change it for the next build so that it only creates 16bit samples.

Quote:
Originally Posted by Thunderbolt8 View Post
and when gaps are detected, is there a way to find out if the delay needed is positive or negative? then it would be possible to add that delay to the demuxed truehd track at a later point.
When there's a gap you need to apply a positive delay. For overlaps you need to apply a negative delay. The gap/overlap amount (in milliseconds) is listed in the eac3to log of the first run.

Quote:
Originally Posted by 73ChargerFan View Post
Report on Mask BD, Extra stream 00024.m2ts using eac3to v2.78
This is the first time I've encountered this, so I thought I'd post it here.

Code:
2: VC-1, 480p24 /1.001 (3:2) with pulldown flags
gave 9684 "video overlaps" messages, or 1 every 10 seconds,
and a playtime of 0:34:06, which is precisely 30/24 times the correct playtime 0:27:17.
I'm aware of this problem. It occurs with VC-1 extras which are encoded as 480p30 with pulldown flags to 480i60. eac3to incorrectly identifies these as "480p24 with pulldown flags". So basically a wrong framerate is detected which results in the problems you noticed. I have on my to do list to fix this. It only occurs with some VC-1 extras, though, so it's not too bad...

Quote:
Originally Posted by digitlman View Post
I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio.
So please ask Syabas and Popcorn Hour to add support for MKV files with multichannel FLAC in it. It's about time they finally support that!

Quote:
Originally Posted by digitlman View Post
i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?
You can use mkvextract the demux the video and audio tracks. Then you can use eac3to to convert the FLAC to whatever format you need. Can't help you with m2ts muxing. That's outside the scope of this thread.

Quote:
Originally Posted by deathlord View Post
Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):
Code:
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.
That's weird. Especially the PTS break before the muxing failure report. My best guess is that your source is corrupt and that the corruption is responsible for the problems. Can you please rerip the disk? Please rip to harddisk first and then run eac3to on the folder on harddisk. This is known to be the most stable way if you have problems with corruption.

Has anyone else remuxed the Simpsons movie with eac3to yet? Did it work ok for you?
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Old 12th December 2008, 17:24   #7324  |  Link
Jeff Flowerday
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Quote:
Originally Posted by digitlman View Post
Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks
If your flac was created from the lossless soundtrack at this point your only choice is to convert the audio to LPCM before stuffing it in a m2ts container. Unfortunaty your file size will increase dramatically, I've seen 4-7GB per file.

I ended up dumping the popcorn hour and putting a HTPC in it's place. With madshi's madflac directshow filter, my MKV's work flawlessly and the Media Portal interface is sooooooo much richer than the popcorn hour and/or the YAMJ plugin for iHome.
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Old 12th December 2008, 23:25   #7325  |  Link
rack04
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Madshi,

Could you lend any advice to neuron2 regarding how to detect dts-hd ma, dts-hd hr, and truehd?

http://forum.doom9.org/showthread.ph...45#post1223545

Thanks.
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Old 13th December 2008, 00:28   #7326  |  Link
rica
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Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.
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Old 13th December 2008, 00:49   #7327  |  Link
nautilus7
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I don't think it can be done with eac3to. Use a hex editor instead.
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Old 13th December 2008, 00:56   #7328  |  Link
rica
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Oooo, i hate hex editors.
Thanks anyways.

Last edited by rica; 13th December 2008 at 01:06.
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Old 13th December 2008, 03:54   #7329  |  Link
banker_rishad
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Madshi sir

Madshi sir i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high. what to do. plz advise.

Last edited by banker_rishad; 14th October 2014 at 15:54.
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Old 13th December 2008, 04:31   #7330  |  Link
yonta
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Quote:
Originally Posted by rica View Post
Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.
tail.exe can do that.
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Old 13th December 2008, 11:03   #7331  |  Link
Snowknight26
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Or clip.exe.
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Old 13th December 2008, 12:15   #7332  |  Link
asarian
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Quote:
Originally Posted by tebasuna51 View Post
Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.
This LPCM conversion is very exciting. I'm still not getting it entirely right. Here's an example:

Code:
eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm
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)
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "c:\video\wall-e.pcm"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 140154 frames.
Video track 2 contains 140154 frames.
Video track 3 contains 140154 frames.
eac3to processing took 13 minutes, 20 seconds.
Done.
So far so good (?). Now:

Code:
Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -c 6
This kinda works. I get an audible 5.1 track (of around 5G), but a lot seems missing: like if the front channels were gone. Mind you, there's sound in the front channels, but it's almost as if things are miswired, as you can hear voices only very faintly, and other major stuff. Weird.

Maybe this has to do with the -16 switches (both for eac3to and Pcm2Tsmu). But if I use those, like:

Code:
eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm -16
Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -i 16 -c 6
Then my PS3 just hangs at the start of the movie (doesn't even start playing the video). So, my question is, how is this down-converting to 16bit supposed to go then? Or, in case down-converting isn't necessary, why am I missing so much audio on the 24bit mix? Other than clearly some major channels/sound missing, what is being output actually sound superb, btw. So I guess I'm close.

Thanks
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Old 13th December 2008, 15:27   #7333  |  Link
asarian
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^^ Seems I'm in bad luck. I send the LPCM over a Toslink, and apparently that only does stereo (I thought it would do 5.1 LPCM as well). Alas.

Interestingly, though, the PS3 actually lists the audio as "Linear PCM, 5.1ch, 6.9Mbps", while streaming (via Twonky). The latter is remarkable, as I always thought, until now, that 640kbps AC3 was the limit for streaming on the PS3. And now it turns out it does full 5.1 LPCM, and at near 7Mbps even! (converted from the DTS-HD track).

@Madshi: still, I'm not a man without hope, as I can now still try my Dolby ProLogic II trick; this time, however, not with a measly 640kbps AC3 stream, but with a 7Mbps LPCM stereo stream: 11x as much bandwidth.

Still, seems the PS3 streams a whole lot more than 640kbps AC3, after all. And that is good news. That I just need to buy a better amp is irrelevant here.
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Old 13th December 2008, 18:27   #7334  |  Link
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Quote:
Originally Posted by rica View Post
Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.
To cut last part of any file you can use CutTools.
It works in primitive binary mode and doesn't know anything about frames or packets of specific file types. You can calculate cut point for yourself if you know frame or packet length.
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Old 13th December 2008, 18:33   #7335  |  Link
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some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???

screen
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Old 13th December 2008, 20:44   #7336  |  Link
madshi
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Quote:
Originally Posted by banker_rishad View Post
i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high.
I don't really know what you mean. And posting attachments to this forum doesn't make sense, unfortunately, because it usually takes years until they are approved.

Quote:
Originally Posted by asarian View Post
This LPCM conversion is very exciting. I'm still not getting it entirely right.
Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.

Quote:
Originally Posted by telmoMRC View Post
some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???
It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".
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Old 13th December 2008, 20:46   #7337  |  Link
madshi
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Quote:
Originally Posted by komisar View Post
Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:
Code:
eac3to.exe input.ac3 output.aac -quality=0.23 -lc
Not sure how many funny parameters neroaacenc has. Right now I'd have to implement every single one by hand to make them all work. That's no fun, so I'm probably not gonna do it. I might sooner or later find a way to automatically pass all unknown parameters to the encoder module. Then what you're looking for would automatically work.
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Old 13th December 2008, 23:41   #7338  |  Link
telmoMRC
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Quote:
Originally Posted by madshi View Post
It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".

i try that and doesn´t work what can i do ?

Last edited by telmoMRC; 13th December 2008 at 23:51.
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Old 14th December 2008, 01:06   #7339  |  Link
rica
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Quote:
Originally Posted by yonta View Post
tail.exe can do that.
Quote:
Originally Posted by Snowknight26 View Post
Thanks guys.

Can you please give me a command line sample for tail.exe and clip.exe?

Code:
usage: tail <infile> <outfile> <length>
For tail i gave it a go with:

Code:
C:\>clip\tail  E:\BDMV\stream\00181.m2ts C:\HD\out.m2ts 500MB
But nothing has happened.

For clip.exe...???

Code:
usage: clip <infile> <outfile> <start offset> <length>
***madshi, thanks for your understanding.

Last edited by rica; 14th December 2008 at 01:31.
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Old 14th December 2008, 02:12   #7340  |  Link
asarian
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Quote:
Originally Posted by madshi View Post

Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.
Fortunately, for me, I have neither a tsMuxeR, nor a Pcm2Tsmu issue. I just like to know one eac3to thing; when I do this:

eac3to 20000.m2ts 4: wall-e.pcm -down2

eac3to lowers the bitrate to about 2Mbps. Any way I can keep the original bitrate for the two channels?
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