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Old 14th April 2017, 08:16   #14201  |  Link
LigH
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@ Brazil2: But this still lets CMD interpret the commands.
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Old 14th April 2017, 11:30   #14202  |  Link
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Originally Posted by LigH View Post
@ Brazil2: But this still lets CMD interpret the commands.
I had the feeling that's what XadoX wanted: to go back to CMD.
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Old 14th April 2017, 22:08   #14203  |  Link
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Quote:
Originally Posted by XadoX View Post
Is there a way while using windows PowerShell to output the eac3to options as list, like usualy with cmd?
I have very little experience with PowerShell, but I believe it can on its own, yes.
Cmd on the other hand doesn't have a replace-tool, so you'd have to use 3rd party software.
My favorite:
Code:
eac3to.exe | xidel.exe -s - -e "replace($raw,'[\b]','','!')" | clip
or
eac3to.exe | xidel.exe -s - -e "replace($raw,'\x08','','!')" | clip
Or Cygwin's sed.exe:
Code:
eac3to.exe | sed.exe "s/\x08//g" | clip
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Old 18th April 2017, 12:10   #14204  |  Link
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Hi, I was under the impression that when converting a 7.1 track to 5.1, eac3to would automatically run a second pass and reduce the volume if clipping was detected. So I just ran this conversion and eac3to didn't generate a second pass. As a test, I ran the same command with -normalize added, and eac3to applied a -0.09db gain, so in other words, there was clipping. So am I supposed to add -normalize to every 7.1 to 5.1 downmix?
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Old 18th April 2017, 12:21   #14205  |  Link
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I can't tell you for sure, but I guess there may be circumstances where eac3to will only extract a 5.1 core and simply omit additional channels. Or it was just matrix encoded, no discrete 7.1 source anyway. You will have to specify more precisely which source format you have and which parameter set you used.
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Old 18th April 2017, 12:28   #14206  |  Link
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Quote:
Originally Posted by LigH View Post
I can't tell you for sure, but I guess there may be circumstances where eac3to will only extract a 5.1 core and simply omit additional channels. Or it was just matrix encoded, no discrete 7.1 source anyway. You will have to specify more precisely which source format you have and which parameter set you used.
Source was DTS-HD MA 7.1 (Zootopia). I just used the stream extractor gui for eac3to, selected the audio track and selected AC3 for output (which would default to a 5.1 640kbps output). The log in both cases say the additional channels were mixed into the surround channels.
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Old 18th April 2017, 12:37   #14207  |  Link
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In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range. Correct me if I'm wrong...
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Old 18th April 2017, 12:55   #14208  |  Link
iSeries
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Quote:
Originally Posted by LigH View Post
In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range. Correct me if I'm wrong...
I just ran the same tests on a completely different movie (but still DTS-HD MA 7.1), and again no messages about clipping. The second test with -normalize added, eac3to again applied a -0.09 gain.

So I guess eac3to just applies a standard -0.09 gain when downmixing 7.1 to 5.1 when the -normalize switch is used, and no clipping is detected?

So now of course I'm wondering, is using the -normalize switch good practice, or should I just let eac3to mix the surrounds and only automatically normalize if there is detected clipping?
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Old 18th April 2017, 13:05   #14209  |  Link
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Quote:
Originally Posted by iSeries View Post
Hi, I was under the impression that when converting a 7.1 track to 5.1, eac3to would automatically run a second pass and reduce the volume if clipping was detected.
Correct.
Quote:
So I just ran this conversion and eac3to didn't generate a second pass. As a test, I ran the same command with -normalize added, and eac3to applied a -0.09db gain, so in other words, there was clipping.
A -0.09db gain can be considered like clipping, because LigH is right when say:
Quote:
Originally Posted by LigH View Post
In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range.
Quote:
So am I supposed to add -normalize to every 7.1 to 5.1 downmix?
Is not recommended.

Even, if the automatic normalize show a big attenuation value, you can consider use a special downmix (MeGUI or BeHappy) than preserve volume in FL,FR,FC and LFE, and only apply attenuation to mix SL+BL, SR+BR.
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Old 18th April 2017, 13:20   #14210  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Correct.


A -0.09db gain can be considered like clipping, because LigH is right when say:




Is not recommended.

Even, if the automatic normalize show a big attenuation value, you can consider use a special downmix (MeGUI or BeHappy) than preserve volume in FL,FR,FC and LFE, and only apply attenuation to mix SL+BL, SR+BR.
Perfect, thanks guys.
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Old 18th April 2017, 15:05   #14211  |  Link
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Is there a way to avoid the re-encoding of the AC3 embedded track when processing a segmented movie?

Code:
[a03]eac3to v3.31
command line: "D:\Archivos de Programa\EAC32AC3\eac3to.exe"  "I:\TestTHD\BDMV\PLAYLIST\00620.mpls" 1) 3: "D:\\00620.mpls_3eng.thd+ac3" -progressnumbers -log="D:\Archivos de Programa\EAC32AC3\UsEac3To.log"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 4 subtitle tracks, 2:07:47, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz
   (embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, -9ms
5: AC3, French, 5.1 channels, 640kbps, 48kHz, -9ms
6: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
7: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
8: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
[a03] AC3 encoding doesn't support back channels. Will mix them into the surround.
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Mixing surround channels...
[a03] Remapping channels...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "D:\\00620.mpls_3eng.thd+ac3"...
[a03] Audio overlaps for 5ms at playtime 0:15:21.  <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:48:26.  <WARNING>
[a03] The audio gaps/overlaps can't be removed from the TrueHD bitstream.  <WARNING>
[a03] In order to remove them you'll have to transcode to another format.  <WARNING>
[a03] Original audio track, L: max 23 bits, average 19 bits.
[a03] Original audio track, R+BL+BR: max 24 bits, average 18 bits.
[a03] Original audio track, C: max 22 bits, average 19 bits.
[a03] Original audio track, LFE: constant bit depth of 18 bits.
[a03] Original audio track, SL: max 21 bits, average 18 bits.
[a03] Original audio track, SR: max 20 bits, average 18 bits.
[a03] Processed audio track, L: max 23 bits, average 19 bits.
[a03] Processed audio track, R+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, C: max 22 bits, average 19 bits.
[a03] Processed audio track, LFE: constant bit depth of 18 bits.
Video track 2 contains 183851 frames.
eac3to processing took 6 minutes, 58 seconds.
Done.
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Old 7th May 2017, 12:10   #14212  |  Link
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Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.
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Old 7th May 2017, 12:14   #14213  |  Link
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Quote:
Originally Posted by rhaz View Post
Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.
You can delete the WAV file.
it doesn't crash if there's no WAV file.
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Old 8th May 2017, 10:23   #14214  |  Link
rhaz
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Quote:
Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.
Quote:
Originally Posted by Sparktank View Post
You can delete the WAV file.
it doesn't crash if there's no WAV file.
Uhhh okay...
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Old 8th May 2017, 15:55   #14215  |  Link
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Quote:
Originally Posted by rhaz View Post
Uhhh okay...
That is, I don't believe there is a "command" to disable the notification. It's just designed that way.

Hence: Delete the WAV file in the eac3to folder.
No WAV file, no sound.
It doesn't break anything if you delete it.

Try looking inside the folder with eac3to, or whatever GUI/AiO you are using and delete the WAV files.
They're not really needed.
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Old 8th May 2017, 19:23   #14216  |  Link
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I do that right away, after the first success sound reminds me I rename the wavs to nos. It is actually a nice and easy way to configure that feature.
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Old 12th May 2017, 21:12   #14217  |  Link
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Looks like eac3to writes chapters to m4a or is it created by NeroAacEnc?

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Audio encoding using eac3to 3.31 x86
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

D:\Projekte\VS\VB\StaxRip\bin\Apps\eac3to\eac3to.exe "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2 44ms.flac" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" -quality=0.35 -normalize +44ms -progressnumbers

FLAC, 2.0 channels, 0:00:41, 24 bits, 827kbps, 44.1kHz
Decoding FLAC...
Applying RAW/PCM delay...
Writing WAV...
Creating file "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.35> with NeroAacEnc...
Applying 0.05dB gain...
The processed audio track has a constant bit depth of 32 bits.
eac3to processing took 1 second.
Done.

Start: 21:53:24
End: 21:53:25
Duration: 00:00:01

General
Complete name : D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a
Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42 (M4A /mp42/isom)
File size : 318 KiB
Duration : 41 s 215 ms
Overall bit rate mode : Variable
Overall bit rate : 63.2 kb/s
Encoded date : UTC 2017-05-12 19:53:25
Tagged date : UTC 2017-05-12 19:53:25

Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 41 s 215 ms
Bit rate mode : Variable
Bit rate : 61.4 kb/s
Maximum bit rate : 70.0 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 kHz
Frame rate : 43.066 FPS (1024 spf)
Compression mode : Lossy
Stream size : 309 KiB (97%)
Writing library : Nero AAC codec 1.5.4.0
Encoding settings : -q 0.35
Encoded date : UTC 2017-05-12 19:53:25
Tagged date : UTC 2017-05-12 19:53:25

Menu

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Muxing using mkvmerge 11.0.0 pre x64
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

D:\Projekte\VS\VB\StaxRip\bin\Apps\MKVToolNix\mkvmerge.exe -o "D:\Video\Samples\MKV\720p SBR_new.mkv" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264" --audio-tracks 0 --language 0:ger --default-track 0:0 "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" --global-tags "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_tags.xml" --ui-language en

mkvmerge v11.0.0 ('Alive') 64bit
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264': Using the demultiplexer for the format 'AVC/h.264'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a': Using the demultiplexer for the format 'QuickTime/MP4'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264' track 0: Using the output module for the format 'AVC/h.264 (unframed)'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a' track 0: Using the output module for the format 'AAC'.
The file 'D:\Video\Samples\MKV\720p SBR_new.mkv' has been opened for writing.
The cue entries (the index) are being written...
Multiplexing took 0 seconds.

Start: 21:53:36
End: 21:53:37
Duration: 00:00:00

General
Complete name : D:\Video\Samples\MKV\720p SBR_new.mkv
Format : Matroska
Format version : Version 4 / Version 2
File size : 11.4 MiB
Duration : 41 s 156 ms
Overall bit rate : 2 324 kb/s
Encoded date : UTC 2017-05-12 19:53:36
Writing application : mkvmerge v11.0.0 ('Alive') 64bit
Writing library : libebml v1.3.4 + libmatroska v1.4.5

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Baseline@L2.1
Format settings, CABAC : No
Format settings, ReFrames : 1 frame
Codec ID : V_MPEG4/ISO/AVC
Duration : 41 s 66 ms
Bit rate : 2 264 kb/s
Width : 480 pixels
Height : 272 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 30.000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.578
Stream size : 11.1 MiB (97%)
Writing library : x264 core 148 r2762 90a61ec
Default : Yes
Forced : No

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : A_AAC
Duration : 41 s 146 ms
Bit rate : 61.5 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 kHz
Frame rate : 43.066 FPS (1024 spf)
Compression mode : Lossy
Delay relative to video : 10 ms
Stream size : 309 KiB (3%)
Language : German
Default : No
Forced : No

Menu
00 : 00:00.059 : en:00:00:00.059
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Old 12th May 2017, 21:46   #14218  |  Link
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Written by NeroAacEnc to signal AAC encoder delay.
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Old 12th May 2017, 21:54   #14219  |  Link
stax76
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And mkvmerge accounts this delay? It's shown by a player as chapters (Editions):

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Old 12th May 2017, 22:15   #14220  |  Link
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NeroAacEnc writes the delay info 2 times:
A) as chapter
B) as proprietary iTunSMPB tag

Mkvmerge reads and applies delay from B automatically. A is treated like a regular chapter. So probably best to set --no-chapters for these m4a files.
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