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7th January 2016, 10:46 | #13781 | Link | |||
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Thanks for the kind reply, tebasuna51.
1. Quote:
Code:
eac3to IN.aac stdout.wav | Aften -b 192 -pad 0 -readtoeof 1 - OUT.ac3 2. Quote:
So if the following command is run, the output delay is either -37ms or +5ms depending on the content. What might be causing such unusual behavior? Code:
eac3to IN.mkv 2: OUT.ac3 Quote:
Any reason? Many thanks and best regards. |
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7th January 2016, 11:19 | #13782 | Link | |
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Quote:
The AC3 encoder inside ffmpeg is better than Aften encoder, both free soft, but maybe there are better certified Dolby Digital commercial encoders. I say "maybe" because I don't know test about that.
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7th January 2016, 15:06 | #13783 | Link | |||
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Quote:
libAften is only a AC3 encoder, can't decode aac, that only work if you have installed the directshow Nero 7 decoder. Recommended command line: eac3to IN.aac -192 OUT.ac3 or eac3to IN.mkv 2: -192 OUT.ac3 Both produce an AC3 with firts 21 ms cutted and 5 ms delayed (16 ms shorter). You can add the parameter +16ms to both command lines to obtain the first 21 ms replaced by silence. Quote:
ffmpeg -i IN.mkv -map 0:1 -c:a ac3 -b:a 192k OUT.ac3 ffmpeg -i IN.mp4 -map 0:1 -c:a ac3 -b:a 192k OUT.ac3 All AC3 with the standard ac3 5 ms delay of silence. Quote:
And in all my test opening IN.aac or IN.mkv or IN.mp4 the decoded aac is perfect, without any cut or delay. I don't know how you obtain these data. Please explain your workflow.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 7th January 2016 at 15:18. Reason: typo |
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9th January 2016, 00:03 | #13784 | Link |
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madshi, could we please get an update for dcadec? it has reached v0.2. I know we could do it ourselves but as others have reported for some reason the .dll we produce seems to work slower than yours.
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9th January 2016, 08:50 | #13785 | Link | ||
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Thanks for the kind reply,
1. Quote:
2. Quote:
Code:
eac3to IN.mkv 2: OUT.ac3 I have uploaded two 10sec samples for your testing pleasure. The first has the remarkable ability to survive Nero's molestation, while the other has a sampling frequency of 48000/24000, which may explain the 42ms truncation instead of 21ms. Perhaps a spectrum analyzer would tell if it is indeed band limited. Many thanks and best regards. |
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9th January 2016, 11:34 | #13786 | Link | |
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Quote:
A last question, why do you use "-center_mixlev 0.707 -surround_mixlev 0.707" with ffmpeg ? |
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9th January 2016, 13:49 | #13787 | Link | |
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At least is not ducumented that option:
https://ffmpeg.org/ffmpeg-all.html#a...-ac3_005ffixed Quote:
Even with other samples until 54 ms cut. Then I edited my post http://forum.doom9.org/showthread.ph...12#post1747412 and now must be: - The directshow Nero 7 decoder, used by eac3to to decode .aac, make unpredictables cuts (from 0 to 54 ms at least) in 2.0 and is broken for 5.1. But this is still valid: - eac3to works fine extracting AAC from MKV/TS/M2TS containers, I obtain the same aac than I muxed previously. Then, to avoid problems, you can use eac3to to extract and Qaac or ffmpeg to decode (or LWLibavAudioSource inside AviSynth).
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9th January 2016, 14:05 | #13788 | Link | ||
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Quote:
Quote:
I recommend use -center_mixlev 0.707 to avoid the low dialog volume when is downmixing to stereo. You can let the default -surround_mixlev or even 0.000 at your preference.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 9th January 2016 at 14:10. |
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11th January 2016, 11:10 | #13790 | Link | ||
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Thanks for the kind reply, tebasuna51.
1. Quote:
2. Quote:
Many thanks and best regards. |
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11th January 2016, 18:37 | #13791 | Link | |
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You can search at same doc page.
For add +delay: https://ffmpeg.org/ffmpeg-all.html#toc-adelay To add 500 ms delay to a 6 channel audio add this to command line: -af adelay=500|500|500|500|500|500 For add -delay: https://ffmpeg.org/ffmpeg-all.html#toc-atrim To cut fisrt 50 ms to an audio add this to command line: -af atrim=0.05 Quote:
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12th January 2016, 19:24 | #13792 | Link |
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Just out of curiosity, can you try and test here ?
I've change some compiler options, to make a build, theoricaly, more efficient. The Intel version is compiled with Intel compiler, and needs a CPU with AVX2 instructions. |
15th January 2016, 05:57 | #13793 | Link |
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Is edit limited to one socond precision? -edit=h:mm:ss,+-delayms
Or is there any way to work with a higher precision? Sth. like -edit:=h:mm:ss:msm,+-delayms? In several scenaerios, I just want to cut out one or two AC3 frames at a very precise position or insert some silent frames (looping is actually not always an option), but there seem to be no tools out there for such tasks. |
15th January 2016, 13:03 | #13796 | Link | |
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Quote:
But remember than eac3to (or DelayCut) only work adding/deleting frames (32 ms for samplerate 48 KHz). Then the edit point, and delay value, are rounded to near value (precision +- 16 ms) Is not possible better precision without recode.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 15th January 2016 at 13:08. |
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20th January 2016, 00:59 | #13797 | Link |
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Due to the circumstances, I encountered yet another movie example where I wonder if the decoding is correctly done by eac3to/ffmpeg or not.
"Everest" comes with a 7.1 TrueHD Atmos track which is, decoded to 7.1 FLAC full of clipping in certain scenes. For the sake of comparison, I extracted the AC3 core and let that one decode to 5.1 PCM. While eac3to suggests a gain of -6.2dB after encountering clipping and recognizing it as such, the result looks just as bad as the "lossless" one (as I learned, this seems to be a relative term when it comes to multichannel audio in conjunction with TrueHD and DTS-HD MA), only more quiet of course. Here is twice the left channel of the soundtrack. Now I wonder: is that particular clipping already contained in the mix (and maybe even intended) or is it screwed up during decoding? In other words: how can I be sure which type of clipping I'm dealing with - the immanent one or the artificial one introduced by bugs or difficulty of the decoding itself thanks to the floating point vs. integer dilemma? |
20th January 2016, 08:56 | #13798 | Link |
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The foobar2000 developer for the DTS plugin had abandoned support of dcadec and reverted back to the old DTS codec because dca was too slow and buggy at the time. Can somebody please make a fork or something with the latest dcadec so I can run some tests with it?
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20th January 2016, 12:43 | #13799 | Link | ||
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Quote:
Quote:
The old BeSplit can do the job without problems. Maybe this is important for Foobar2000 but not here, and the old DTS decoder foobar plugin don't support DTS-HD at all, at least in my test.
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20th January 2016, 13:05 | #13800 | Link | |
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Quote:
Yes, that's it. It had serious issues with DTS CDs. And there is an old foobar plugin for DTS-HD. Okay, I guess I'll ask about it on Hydrogenaudio instead of here. Thanks! |
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