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Old 20th August 2011, 03:38   #11201  |  Link
ramicio
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No, Arcsoft is the only way to decode DTS-HD Master besides the DTS suite, and the rare and impossible to find Sonic decoder. I want eac3to to mix my stuff because it detects clipping, and it also uses the best possible processing. I also don't care for foobar2000 for anything but playback.
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Old 20th August 2011, 06:09   #11202  |  Link
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Quote:
Originally Posted by ramicio View Post
No, Arcsoft is the only way to decode DTS-HD Master besides the DTS suite, and the rare and impossible to find Sonic decoder.
Fair enough. I was just looking at ffdshow's settings and noticed the Pass-Through option for DTS HD, so wondered if that meant it could also decode it.

Quote:
Originally Posted by ramicio View Post
I want eac3to to mix my stuff because it detects clipping, and it also uses the best possible processing. I also don't care for foobar2000 for anything but playback.
I'm sure there's a rational logic behind all that.
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Old 20th August 2011, 10:34   #11203  |  Link
tebasuna51
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Quote:
Originally Posted by ramicio View Post
... It's these pesky 6.1 I come across now and then. I like eac3to because it detects clipping. Would it be possible for avisynth to turn out 6.1?
For 6.1 in monowavs you can use:
Code:
sl=WavSource("D:\tmp\SL.wav").ConvertAudioToFloat() # Load channels to mix
sr=WavSource("D:\tmp\SR.wav").ConvertAudioToFloat()
bc=WavSource("D:\tmp\BC.wav").ConvertAudioToFloat()
bl=MixAudio(sl, bc, 1.0, 0.7071)                    # Mix half power of BC into SL
br=MixAudio(sr, bc, 1.0, 0.7071)                    # Mix half power of BC into SR

fl=WavSource("D:\tmp\FL.wav").ConvertAudioToFloat() # Load the rest of channels
fr=WavSource("D:\tmp\FR.wav").ConvertAudioToFloat()
fc=WavSource("D:\tmp\FC.wav").ConvertAudioToFloat()
lf=WavSource("D:\tmp\lf.wav").ConvertAudioToFloat()

MergeChannels(fl, fr, fc, lf, bl, br)               # Merge the 6 channels
Normalize()                                         # Normalize to avoid clip
ConvertAudioTo24bit()                               # Change to desired precision
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Old 20th August 2011, 22:47   #11204  |  Link
davidhorman
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From what I've read, eac3to should be able to extract a core AC3 track from a TrueHD file, but I get this:

Code:
M:\TMP\bd>eac3to bd.thd bd.ac3
TrueHD, 7.1 channels, 48kHz, dialnorm: -27dB
AC3 encoding doesn't support back channels. Will mix them into the surround.
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Mixing surround channels...
Encoding AC3 <640kbps> with libAften...
which looks like it's completely re-encoding. Am I missing an option?

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Old 20th August 2011, 22:55   #11205  |  Link
ramicio
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That file doesn't have an AC3 core, it would be listed.

It would look like this:

Code:
d:\>\apps\eac3to\eac3to.exe file.thd
TrueHD/AC3, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
Along with my previous dilemma, I decided to just have 1.1.0.0 and 1.1.0.7 of the Arcsoft filters and flip-flop them depending on what I need.

And why does a line get added to the bottom of the code boxes?

Last edited by ramicio; 20th August 2011 at 23:06.
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Old 20th August 2011, 23:29   #11206  |  Link
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Well, maybe Lord Mulder could want to write a "dtshdma2wavs.exe" , which would always pick the "best" Arcsoft DLL for each case

Last edited by Midzuki; 20th August 2011 at 23:32. Reason: re-wording
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Old 22nd August 2011, 11:47   #11207  |  Link
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Hello any chance to add custom Vorbis encoder (oggenc2) support to eac3to ?
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Old 22nd August 2011, 16:53   #11208  |  Link
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@madshi,
I was doing some batch extracts of audio from my media files when I noticed the following that I had not seen before.
Code:
eac3to v3.24
command line: eac3to  "file1.mkv" 2: "file1.ac3"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:42:01, 24p /1.001
1: h264/AVC, 720p24 /1.001
2: AC3, ???, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB
I tried searching this thread but could not really find anything on the ???.
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Old 22nd August 2011, 17:03   #11209  |  Link
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Quote:
Originally Posted by Chumbo View Post
...
2: AC3, ???, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB
Seems a country code not supported by eac3to.
Try with MediaInfo.
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Old 22nd August 2011, 17:07   #11210  |  Link
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Quote:
Originally Posted by Anakunda View Post
Hello any chance to add custom Vorbis encoder (oggenc2) support to eac3to ?
You can use the external encoder with:
eac3to input stdout.wav | OggEnc2 -q 3 --ignorelength -o output.ogg -
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Old 22nd August 2011, 17:32   #11211  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Seems a country code not supported by eac3to.
Try with MediaInfo.
I had done that already and didn't notice any difference. I only ask because normally I don't see anything in the ??? position, i.e.,
Code:
2: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB, 17ms
Aside from items like duration, file size, etc. MediInfo reports identical results for files with the ??? and without.
Code:
Audio
ID                               : 2
Format                           : AC-3
Format/Info                      : Audio Coding 3
Mode extension                   : CM (complete main)
Codec ID                         : A_AC3
Duration                         : 42mn 10s
Bit rate mode                    : Constant
Bit rate                         : 448 Kbps
Channel(s)                       : 6 channels
Channel positions                : Front: L C R, Side: L R, LFE
Sampling rate                    : 48.0 KHz
Bit depth                        : 16 bits
Compression mode                 : Lossy
Delay relative to video          : 17ms
Stream size                      : 135 MiB (12%)
Language                         : English
I don't see any problems caused by this but I was mostly curious as to what the ??? position is supposed to be reporting.
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Old 23rd August 2011, 02:57   #11212  |  Link
tebasuna51
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Quote:
Originally Posted by Chumbo View Post
...I only ask because normally I don't see anything in the ??? position,...
No?
Maybe your mkv's don't have the language field filled:
Code:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\cono3.mkv" 
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 1 subtitle track, 0:00:22, 25p
1: h264/AVC, English, 720p25
2: DTS, English, 5.1 channels, 24 bits, 755kbps, 48kHz
3: AC3, German, 5.1 channels, 448kbps, 48kHz
4: Subtitle (SRT), Spanish
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Old 23rd August 2011, 03:37   #11213  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
No?
Maybe your mkv's don't have the language field filled:
Code:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\cono3.mkv" 
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 1 subtitle track, 0:00:22, 25p
1: h264/AVC, English, 720p25
2: DTS, English, 5.1 channels, 24 bits, 755kbps, 48kHz
3: AC3, German, 5.1 channels, 448kbps, 48kHz
4: Subtitle (SRT), Spanish
MediaInfo seems to have no issues picking up the language as English and I always set the language on the audio stream specifically to eng using mkvtoolnix when creating my MKV files. Like I said, it's not a big deal, I was just curious about the ???.
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Old 23rd August 2011, 11:13   #11214  |  Link
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Mkvmerge (in recent versions) doesn't write the language field if set to "English" and eac3to doesn't recognize that. That's why eac3to doesn't display any language for English tracks if muxed with such a version. I don't know where the "???" comes from. Look at the value of the field "Language" in mkvinfo - maybe you can spot a difference between files with "" (empty) and "???".
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Old 24th August 2011, 15:42   #11215  |  Link
elmarikon
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Hi guys!

I don't know if this has been asked before, but how can I prevent eac3to from downsampling wavs when I come from a 24bit, 48KHz, 6ch pcm file, trying to extract mono wav files?
I don't know why, but for some reason eac3to from time to time thinks it has to reduce the bit depth to 16bit...

I have tried to define all values, like eac3to in-6ch.pcm out\out.wavs -48000 -24 -6 -little -override (even tried -down24)... still receving 16bit wavs, but I do need 24bit.

Wht can I do, and why s it doing that anyway?

l the best and thanx for the apart from that great tool!

all the best

swk
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Old 24th August 2011, 15:50   #11216  |  Link
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Your input file is 16-bit then.
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Old 24th August 2011, 16:02   #11217  |  Link
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Eac3to detects when a 24-bit file has the 8 least significant bits filled with zeroes (which is easily accomplished by taking a 16-bit file and simply saving it as 24-bit) and drops it to 16 bits. Keeping it 24 bits would be pointless and a waste of space. When eac3to detects that the file is between 17 and 23 bits, it will keep it 24.
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Old 25th August 2011, 08:03   #11218  |  Link
Anakunda
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Quote:
Originally Posted by tebasuna51 View Post
You can use the external encoder with:
eac3to input stdout.wav | OggEnc2 -q 3 --ignorelength -o output.ogg -
this doesnot work for me
Code:
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -
Any idea?

I get the error:
Quote:
Please specify the source and dest files first and then the options.ERROR: Input file "(stdin)" is not a supported format
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Old 25th August 2011, 09:05   #11219  |  Link
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You have to specify "stdout.wav" as a fake output file, like in tebasuna51's example. Try:
Code:
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" stdout.wav -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -
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Old 25th August 2011, 09:11   #11220  |  Link
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If I specify fake output file then the output will go to file and not stdout right?

I tried it now:
Code:
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" $$fake.wav -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -
and still got error:

Quote:
ERROR: Input file "(stdin)" is not a supported format

Sorry I missed the name thankyou!!!

Last edited by Anakunda; 25th August 2011 at 09:28.
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