eac3to - audio conversion tool

[raquete]

http://www.true-audio.com/
What ??? :confused:
Are you sure you weren't using the «lossy» compression mode ?
Also, if the answer is "no", are you sure the sound editors you used were decompressing the lossless .WV files correctly???


first, thanks for the link.
i really don't know TTA and need to read about this format. :)

WV lossy? :confused:
i don't knew that one lossless compression can be "lossy". :p

adjusted "lossless(high)" to encode (no hibrid) 32 bit float
and adobe audition to open and check WV.

example from audition waveform statistics
(selected one channel where have one clip):
Mono
Min Sample Value: 21785
Max Sample Value: 32767
Peak Amplitude: -13.87 dB
Possibly Clipped: 1
DC Offset: 79.736
Minimum RMS Power: 996.99 dB
Maximum RMS Power: -inf dB
Average RMS Power: -3.01 dB
Total RMS Power: -18.65 dB
Actual Bit Depth: 16 Bits

Using RMS Window of 50 ms

[raquete]

tebasuna51,
after 8 minutes with ~60% of the file encoded i got this message:
Error: System.IO.IOException: O pipe foi finalizado.
Starting job 01wave.wav->01wave.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=96000Hz
encoder\flac.exe --force -o "H:\eac3to\01wave.flac"
--silent --force-raw-format --endian=little
--channels=6 --bps=24 --sample-rate=96000 --sign=signed -
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: O pipe foi finalizado.

[kypec]

Another "undocumented" option of the flac applications? :(
No, it's not undocumented. FLAC has very extensive support of tagging, look in the documentation, --help switch is your friend.
I wonder what's different about your test files, raquete and tebasuna51, that produces such dramatic differences?
In my case the size difference wasn't that dramatic at all.
435 306 274 : 435 640 726 = 99.923%, the real difference is less than 0.1%, no big deal.

[tebasuna51]

tebasuna51,
after 8 minutes with ~60% of the file encoded i got this message:
Error: System.IO.IOException: O pipe foi finalizado.

Yep, this is a know problem with flac.exe. Can't write flac files >2GB
Maybe in next v1.2.2...

See:
http://sourceforge.net/tracker/index.php?func=detail&aid=1834949&group_id=13478&atid=363478
http://sourceforge.net/tracker2/?func=detail&aid=1800350&group_id=13478&atid=113478

Until next flac.exe version you must use only eac3to if flac file output is >2GB.

Edit: With an audio track tested don't exist many difference:
8032850564 Pulp.wav
3918536382 Pulp_120.flac 48.8%
3893160387 Pulp_121.flac 48.5%

[Midzuki]

No, it's not undocumented. FLAC has very extensive support of tagging, look in the documentation, --help switch is your friend.

metaflac - Command-line FLAC metadata editor version 1.2.1
Copyright (C) 2001,2002,2003,2004,2005,2006,2007 Josh Coalson

Usage:
metaflac [options] [operations] FLACfile [FLACfile ...]

Use metaflac to list, add, remove, or edit metadata in one or more FLAC files.
You may perform one major operation, or many shorthand operations at a time.

Options:
--preserve-modtime Preserve the original modification time in spite of edits
--with-filename Prefix each output line with the FLAC file name
(the default if more than one FLAC file is specified)
--no-filename Do not prefix each output line with the FLAC file name
(the default if only one FLAC file is specified)
--no-utf8-convert Do not convert tags from UTF-8 to local charset,
or vice versa. This is useful for scripts, and setting
tags in situations where the locale is wrong.
--dont-use-padding By default metaflac tries to use padding where possible
to avoid rewriting the entire file if the metadata size
changes. Use this option to tell metaflac to not take
advantage of padding this way.

Shorthand operations:
--show-md5sum Show the MD5 signature from the STREAMINFO block.
--show-min-blocksize Show the minimum block size from the STREAMINFO block.
--show-max-blocksize Show the maximum block size from the STREAMINFO block.
--show-min-framesize Show the minimum frame size from the STREAMINFO block.
--show-max-framesize Show the maximum frame size from the STREAMINFO block.
--show-sample-rate Show the sample rate from the STREAMINFO block.
--show-channels Show the number of channels from the STREAMINFO block.
--show-bps Show the # of bits per sample from the STREAMINFO block.
--show-total-samples Show the total # of samples from the STREAMINFO block.

--show-vendor-tag Show the vendor string from the VORBIS_COMMENT block.
--show-tag=NAME Show all tags where the the field name matches 'NAME'.
--remove-tag=NAME Remove all tags whose field name is 'NAME'.
--remove-first-tag=NAME Remove first tag whose field name is 'NAME'.
--remove-all-tags Remove all tags, leaving only the vendor string.
--set-tag=FIELD Add a tag. The FIELD must comply with the Vorbis comment
spec, of the form "NAME=VALUE". If there is currently
no tag block, one will be created.
--set-tag-from-file=FIELD Like --set-tag, except the VALUE is a filename
whose contents will be read verbatim to set the tag value.
Unless --no-utf8-convert is specified, the contents will
be converted to UTF-8 from the local charset. This can
be used to store a cuesheet in a tag (e.g.
--set-tag-from-file="CUESHEET=image.cue"). Do not try
to store binary data in tag fields! Use APPLICATION
blocks for that.
--import-tags-from=FILE Import tags from a file. Use '-' for stdin. Each line
should be of the form NAME=VALUE. Multi-line comments
are currently not supported. Specify --remove-all-tags
and/or --no-utf8-convert before --import-tags-from if
necessary. If FILE is '-' (stdin), only one FLAC file
may be specified.
--export-tags-to=FILE Export tags to a file. Use '-' for stdout. Each line
will be of the form NAME=VALUE. Specify
--no-utf8-convert if necessary.
--import-cuesheet-from=FILE Import a cuesheet from a file. Use '-' for stdin.
Only one FLAC file may be specified. A seekpoint will be
added for each index point in the cuesheet to the
SEEKTABLE unless --no-cued-seekpoints is specified.
--export-cuesheet-to=FILE Export CUESHEET block to a cuesheet file, suitable
for use by CD authoring software. Use '-' for stdout.
Only one FLAC file may be specified on the command line.
--import-picture-from=FILENAME|SPECIFICATION Import a picture and store it in a
PICTURE block. Either a filename for the picture file or
a more complete specification form can be used. The
SPECIFICATION is a string whose parts are separated by |
characters. Some parts may be left empty to invoke
default values. FILENAME is just shorthand for
"||||FILENAME". The format of SPECIFICATION is:
[TYPE]|[MIME-TYPE]|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILE
TYPE is optional; it is a number from one of:
0: Other
1: 32x32 pixels 'file icon' (PNG only)
2: Other file icon
3: Cover (front)
4: Cover (back)
5: Leaflet page
6: Media (e.g. label side of CD)
7: Lead artist/lead performer/soloist
8: Artist/performer
9: Conductor
10: Band/Orchestra
11: Composer
12: Lyricist/text writer
13: Recording Location
14: During recording
15: During performance
16: Movie/video screen capture
17: A bright coloured fish
18: Illustration
19: Band/artist logotype
20: Publisher/Studio logotype
The default is 3 (front cover). There may only be one picture each
of type 1 and 2 in a file.
MIME-TYPE is optional; if left blank, it will be detected from the
file. For best compatibility with players, use pictures with MIME
type image/jpeg or image/png. The MIME type can also be --> to
mean that FILE is actually a URL to an image, though this use is
discouraged.
DESCRIPTION is optional; the default is an empty string
The next part specfies the resolution and color information. If
the MIME-TYPE is image/jpeg, image/png, or image/gif, you can
usually leave this empty and they can be detected from the file.
Otherwise, you must specify the width in pixels, height in pixels,
and color depth in bits-per-pixel. If the image has indexed colors
you should also specify the number of colors used.
FILE is the path to the picture file to be imported, or the URL if
MIME type is -->
--export-picture-to=FILE Export PICTURE block to a file. Use '-' for stdout.
Only one FLAC file may be specified. The first PICTURE
block will be exported unless --export-picture-to is
preceded by a --block-number=# option to specify the exact
metadata block to extract. Note that the block number is
the one shown by --list.
--add-replay-gain Calculates the title and album gains/peaks of the given
FLAC files as if all the files were part of one album,
then stores them in the VORBIS_COMMENT block. The tags
are the same as those used by vorbisgain. Existing
ReplayGain tags will be replaced. If only one FLAC file
is given, the album and title gains will be the same.
Since this operation requires two passes, it is always
executed last, after all other operations have been
completed and written to disk. All FLAC files specified
must have the same resolution, sample rate, and number
of channels. The sample rate must be one of 8, 11.025,
12, 16, 22.05, 24, 32, 44.1, or 48 kHz.
--remove-replay-gain Removes the ReplayGain tags.
--add-seekpoint={#|X|#x|#s} Add seek points to a SEEKTABLE block
# : a specific sample number for a seek point
X : a placeholder point (always goes at the end of the SEEKTABLE)
#x : # evenly spaced seekpoints, the first being at sample 0
#s : a seekpoint every # seconds; # does not have to be a whole number
If no SEEKTABLE block exists, one will be created. If
one already exists, points will be added to the existing
table, and any duplicates will be turned into placeholder
points. You may use many --add-seekpoint options; the
resulting SEEKTABLE will be the unique-ified union of
all such values. Example: --add-seekpoint=100x
--add-seekpoint=3.5s will add 100 evenly spaced
seekpoints and a seekpoint every 3.5 seconds.
--add-padding=length Add a padding block of the given length (in bytes).
The overall length of the new block will be 4 + length;
the extra 4 bytes is for the metadata block header.

Major operations:
--version
Show the metaflac version number.
--list
List the contents of one or more metadata blocks to stdout. By default,
all metadata blocks are listed in text format. Use the following options
to change this behavior:

--block-number=#[,#[...]]
An optional comma-separated list of block numbers to display. The first
block, the STREAMINFO block, is block 0.

--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
An optional comma-separated list of block types to be included or ignored
with this option. Use only one of --block-type or --except-block-type.
The valid block types are: STREAMINFO, PADDING, APPLICATION, SEEKTABLE,
VORBIS_COMMENT. You may narrow down the types of APPLICATION blocks
displayed as follows:
APPLICATION:abcd The APPLICATION block(s) whose textual repre-
sentation of the 4-byte ID is "abcd"
APPLICATION:0xXXXXXXXX The APPLICATION block(s) whose hexadecimal big-
endian representation of the 4-byte ID is
"0xXXXXXXXX". For the example "abcd" above the
hexadecimal equivalalent is 0x61626364

NOTE: if both --block-number and --[except-]block-type are specified,
the result is the logical AND of both arguments.

--application-data-format=hexdump|text
If the application block you are displaying contains binary data but your
--data-format=text, you can display a hex dump of the application data
contents instead using --application-data-format=hexdump

--remove
Remove one or more metadata blocks from the metadata. Unless
--dont-use-padding is specified, the blocks will be replaced with padding.
You may not remove the STREAMINFO block.

--block-number=#[,#[...]]
--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
See --list above for usage.

NOTE: if both --block-number and --[except-]block-type are specified,
the result is the logical AND of both arguments.

--remove-all
Remove all metadata blocks (except the STREAMINFO block) from the
metadata. Unless --dont-use-padding is specified, the blocks will be
replaced with padding.

--merge-padding
Merge adjacent PADDING blocks into single blocks.

--sort-padding
Move all PADDING blocks to the end of the metadata and merge them into a
single block.

As you ¿can? see, there is no mentioning of "WaveFormatExtensible", nor "Channel_Mask".

[raquete]

Yep, this is a know problem with flac.exe. Can't write flac files >2GB
Maybe in next v1.2.2...

See:
http://sourceforge.net/tracker/index.php?func=detail&aid=1834949&group_id=13478&atid=363478
http://sourceforge.net/tracker2/?func=detail&aid=1800350&group_id=13478&atid=113478

Until next flac.exe version you must use only eac3to if flac file output is >2GB.

Edit: With an audio track tested don't exist many difference:
8032850564 Pulp.wav
3918536382 Pulp_120.flac 48.8%
3893160387 Pulp_121.flac 48.5%

no problems, eac3to is cool, work perfectly and easy to use.

thanks so much for the links, hints, suport, big patience and friendship. :)

[Midzuki]

first, thanks for the link.
i really don't know TTA and need to read about this format. :)

WV lossy? :confused:
i don't knew that one lossless compression can be "lossy". :p

WavPack is a very-versatile format, indeed. :D

adjusted "lossless(high)" to encode (no hibrid) 32 bit float
and adobe audition to open and check WV.

example from audition waveform statistics
(selected one channel where have one clip):
Mono
Min Sample Value: 21785
Max Sample Value: 32767
Peak Amplitude: -13.87 dB
Possibly Clipped: 1
DC Offset: 79.736
Minimum RMS Power: 996.99 dB
Maximum RMS Power: -inf dB
Average RMS Power: -3.01 dB
Total RMS Power: -18.65 dB
Actual Bit Depth: 16 Bits

Using RMS Window of 50 ms

Have you tried: 1) decompress the .WV file to a .WAV copy of the original source; 2) bit-compare the original .WAV and the copied .WAV; ???

I don't use Adobe Audition, so I can just suppose it doesn't deal very well with 32-bit .WV files...


BTW: Did you generate those 32-bit .WV archives from 32-bit sources --- or not ???

I mean: regardless of the possible bugs/imperfections of «WavPack.exe»,
I see no point in creating 32-bit .WV archives only because Adobe Audition is your favorite sound editor. :)

[raquete]

WavPack is a very-versatile format, indeed.
lol

audition always open as wave 32bit float but i don't remember if i generate those 32-bit WavPack archives from 32-bit sources ...i don't use WV no more after see that clips.
sound forge show exactly the same clips as audition but audition is my favorite, have amazing differents features.
i'm using now flacs or apes.
flac is my preference.

cheers!

[yesgrey]

I've tested with Baraka, the biggest movie audio file I have, 5.1 96kHz 24bit.
wav : 9,452,814,404
1.2.0: 5,279,451,099 (55.85%)
1.2.1: 5,247,885,754 (55.51%)

That's not a very big difference. I'm not sure it's worth the effort to reencode all my backups with 1.2.1, I think I will only use it with the new ones...

[kypec]

As you ¿can? see, there is no mentioning of "WaveFormatExtensible", nor "Channel_Mask".
Of course there isn't. They are just METADATA and therefore can have arbitrary names and values. It's up to the application processing such FLAC files to honor/ignore such tags. These tags are not the same as RIFF data structures which are mandatory in WAVE headers for example.
I just included one of that tags manually in order to obtain one FLAC file as closely similar to another one as possible for testing purposes.

[leeperry]

i'm using now flacs or apes.
flac is my preference.
APE compresse way better than FLAC for stereo, and WavPack is usually slightly better than FLAC for 5.1 audio.

there really isn't anything good about FLAC, except hardware compatiblity..

plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.

[raquete]

APE compresse way better than FLAC for stereo, and WavPack is usually slightly better than FLAC for 5.1 audio.

there really isn't anything good about FLAC, except hardware compatiblity..

plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.as i posted, i have some problems like clips with WV then was left alone and what i know and was posted here too is that eac3to use flac max compression.
as my target is compress 5.1 multichannel, ape can't be used and WV even giving more compression than flac give me problems compressing stereo, i never did tests with 5.1 multichannel.
Paul McCartney E.Arguments are in flacs too then is cool, agree? :)

http://flac.sourceforge.net/news.html

[Midzuki]

raquete wrote:

Wavpack always give me problems when i encode waves with high volumes(levels), some peaks appear clippeds(saturations, straight lines/squares). i can see in any wave editor doing analysis.

Are you sure that your troubles with WavPack did not come from those excessively-loud sources ???

P.S.: Did you read the PM I sent you @ VideoHelp ?

[raquete]

raquete wrote:
Are you sure that your troubles with WavPack did not come from those excessively-loud sources ???

P.S.: Did you read the PM I sent you @ VideoHelp ?
PM in videohelp?!? no, i will read after this post.

excessively-loud sources not means clips but something like loudness war(is what you mean? )
as i encode my own cds, is impossible that EACopy or any wave editor(audition/sound forge) used to rip will encrease the levels of the source.
i check everything with big patience(sorry, i'm really bored with audio) : the waveforms after extracteds and after encodeds...and lots more details.
i posted the result of audition waveform statistics showing "Possibly Clipped: 1" because was selected few seconds around the first clip that i found from WV but the wave source don't have this clip. the top of the waveform turn from sine to straight line. :eek:
do you want samples and pictures from this waveforms with clips? i can post.
sometimes i need one entire month to encode a single album in 5.1(DVD-A & DVD-Video with audio too), i don't like of "more or less" results.

now answer me: if you find clips in the results of any encoder, what you will do?

cheers! :)

edit:
idea...as what we are talking about now nothing have to do with eac3to, we must open a new thread about encoders, his features, problems and advantages to don't mess this thread as the program here works very very fine, agree? :)

[kypec]

there really isn't anything good about FLAC, except hardware compatiblity..Maybe, but it is THE FACTOR for many people around here I guess. Also it is open source, cross-platform compatible, HW decoding is quite simple...
plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.
No, it's stuck on 8=best as been stated by madshi before and proven true by testers (like me and raquette) also.

[leeperry]

oh ok?! but last time I transcoded it to "8" w/ dBPowerAmp and the file was smaller :confused:

well DVDA-Explorer works just fine w/ WavPack and crashes w/ FLAC for me, so that kinda closes the deal(files are not >2GB)...FLAC for HD audio BD tracks(in eac3to), WavPack for DVD-A and APE for CDDA :)

comparing TrueHD to 384kb/s AC3 is like comparing 64kbit MP3 to WAV...I'm such a lossless whore these days :D

[kypec]

oh ok?! but last time I transcoded it to "8" w/ dBPowerAmp and the file was smaller :confused:That's because eac3to uses older version 1.2.0 of FLAC library. Replace it with newer 1.2.1 as suggested by me & raquette and you'll get smaller FLAC files also! :p

[yesgrey]

If you don't know how, you need to get the libflac pack for developers here (http://sourceforge.net/projects/flac/files/flac-win/flac-1.2.1-win/flac-1.2.1-devel-win.zip/download).
Then, you grab the libflac.dll contained in the pack (inside Lib dir) and substitute the one in eac3to's dir.

[Atak_Snajpera]

Replace it with newer 1.2.1 as suggested by me & raquette and you'll get smaller FLAC files also!
http://img175.imageshack.us/img175/3171/new1.th.png (http://img175.imageshack.us/i/new1.png/)

[raquete]

Atak_Snajpera,
your wave 5.1 with 1h 35mn have round 3Gb size?
or better, what is the size of your source that result 798Mb flac??

[kypec]

Atak_Snajpera,
it's not always that you get smaller files with 1.2.1 vs 1.2.0
Try with some 24-bit resolution samples and perhaps you'll see some improvement.
It also happened to me that 1.2.0 produced same size as 1.2.1 when compressing down-sampled 16-bit sources.

[Atak_Snajpera]

your wave 5.1 with 1h 35mn have round 3Gb size?
or better, what is the size of your source that result 798Mb flac??
It was encoded directly from DTSMA via eac3to with ArcSoft DTS decoder installed.

[raquete]

Atak_Snajpera,
i asked because seems too short the flac final size 798MB "only" as your source is wave 1h 35 min with 6.1 channels can have round 3GB, i mean must be too big size source to result in 798MB.
i'm losing details somewhere, i can be wrong and i'm confused.

cheers. :)

[Atak_Snajpera]

MY SOURCE IS NOT WAVE 6.1! It was encoded directly from DTSMA without any temporary wave files! What part you don't understand now? 16 bit files compress alot better than 24bit.

[raquete]

MY SOURCE IS NOT WAVE 6.1! It was encoded directly from DTSMA without any temporary wave files! What part you don't understand now? 16 bit files compress alot better than 24bit.

ah, sure...no 6.1(was a typo, i was thinking to write 5.1)....and because i don't know what is DTSMA. :confused:
of course 16b encode alot better than 24b.

in the end, what size have your source? :p (have a way to measure i think)
after know the size source we can tell about flac compression, right?

[yesgrey]

i don't know what is DTSMA.
DTS-HD Master Audio. Is the lossless format from DTS. One of the formats used in Blu-ray. If you want to know more go to the dts website...

[Atak_Snajpera]

n the end, what size have your source? (have a way to measure i think)
after know the size source we can tell about flac compression, right?

http://img382.imageshack.us/img382/5386/capturegfw.th.png (http://img382.imageshack.us/i/capturegfw.png/)

[raquete]

as you know, i'm not familiar with DTSMA (thanks for dts website hint) but for me from 1.28GB(5.1 dtsma variable bit rate) to 798MB flac seems very good compression.

[yesgrey]

DTS-HD MA is not very efficient, because it consists of a lossy dts core at 1536kbps and another part that contain the difference between the original and the lossy core.

[Revgen]

^FLAC even beats True-HD when it comes to lossless compression. True-HD, to my knowledge, does not have a lossy core. The only thing it has that FLAC doesn't is metadata telling True HD decoders how to downmix. I can't imagine that the metadata would put so much size on a file.

[leeperry]

you know, i'm not familiar with DTSMA (thanks for dts website hint)
the worst is DTS-HD HR, it was just as big as DTS-MA on the few discs I encountered it...and yet, it's lossy.

a sneaky way to still provide customers w/ lossy audio in my book..

yes, FLAC beats TrueHD :cool:

[M-Blaster]

Hey, when demuxing a movie with DTS HD master audio
"eac3to l:\mainevo.evo e:\ren.mkv -core"
I am getting only big flac audio files. that can not integrated in the mkv movie.

How can I get dts files (1508kbit core only ) from an movie with DTS Master?

[tebasuna51]

1) flac is supported in mkv and is more efficient than DTS-MA, then is the default option.

2) To extract the core of the track 2 (if is the DTS-MA you want)

eac3to l:\mainevo.evo 2: e:\ren.dts -core

[raquete]

another question for you tebasuna.
how to eac3to command line to extract dts.wav (5.1 44.1/16b) to waves (seperateds channels in 48k/24b) ?
thanks in advance.

[tebasuna51]

another question for you tebasuna.
how to eac3to command line to extract dts.wav (5.1 44.1/16b) to waves (seperateds channels in 48k/24b) ?
thanks in advance.

¿dts.wav (5.1 44.1/16b)?

I don't know that. I know:
- dts.wav (2.0, 44.1, 16b) if the fake wav header is readed
- dts.wav (5.1, 44.1, some bitrate) the dts in the fake wav container without bitdepth but bitrate

To obtain mono wavs 48KHz 24 bits you can use:

eac3to dts.wav mono.wavs -resampleTo48000

(24 bits is the default)

[raquete]

yes, dts.wav is what i get ripping dts-cds with EAC....

to burn multichannel dts-cd is needed 5.1 44.1k 16b files(burn as audio cd)and ripping dts.wav from dts-cds give 5.1 44.1k 16b too!

as 24 bits is the default when extract with eac3to the command line posted is perfect.
Thanks so much.

edit:
and how to(command line) extract from the same source in 96k/24b?

[Snowknight26]

It's documented when you run the program and in the first post.

[raquete]

It's documented when you run the program and in the first post.

ah...thanks...great document, i read all.
is cool for who knows command lines. :(

following tebasuna example seems easy to use, i only have doubt what came first in the command line...the bitrate or the (up)sample?!? (i can't find this detail in the documment)

[Snowknight26]

Well, seeing as -changeTo48000 changed the sampling rate to 48KHz....

[raquete]

correct. :)
edit:
and how to(command line) extract from the same source in 96k/24b?
what came first in the command line...the bitrate or the (up)sample?!? (i can't find this detail in the documment)

i know that 24b is default but and if i want 32b or 8b?
the doubt remains Snowknight26.

[tebasuna51]

@raquete
You can use:
eac3to dts.wav mono.wavs -down16 -resampleTo48000
---------------------------------------------------------
DTSWAV, 5.1 channels, 0:02:07, 24 bits, 1235kbps, 44.1khz
Reading DTSWAV...
Decoding with ArcSoft DTS Decoder...
Resampling to 48khz...
Reducing depth from 64 to 16 bits...
Writing WAVs...

Always the process (here only decode and resample) is done with the high precission (here 64 bits float), at the end the depth is changed to the desired.

No matter the parameter order.

[raquete]

no matter the parameter order, very nice, very clever!
i have dozen works to start now.

thank you so much tebasuna51.

[M-Blaster]

1) flac is supported in mkv and is more efficient than DTS-MA, then is the default option.

2) To extract the core of the track 2 (if is the DTS-MA you want)

eac3to l:\mainevo.evo 2: e:\ren.dts -core

Error with abort : This audio conversion is not supported.

[tebasuna51]

Error with abort : This audio conversion is not supported.

Please put a MediaInfo report of your evo file

[NanoBot]

Hi,

Error with abort : This audio conversion is not supported.


tebasuna51 accidentally added a white space in his suggested command line. If you entered the command exactly like suggested by copy and paste, it cannot work.

The correct command line is

eac3to l:\mainevo.evo 2:e:\ren.dts -core

without the white space between 2: and e:\ren.dts -core

C.U. NanoBot

[Snowknight26]

No, the space there doesn't matter. Since it doesn't matter, then the only logical explanation is that track #2 doesn't have a DTS core.

[Chumbo]

@madshi,
When you have time to get back to this, would you consider adding support for FLV (Flash Video) please? It would be great if eac3to can demux and give info on this file type. Hopefully this will help. I use a tool called FLV Extract but would rather use eac3to. It's available here with source: http://www.moitah.net/. The direct file link is: http://www.moitah.net/download/latest/FLV_Extract.zip. Thanks for considering it.

[Bozster]

Can someone help me out. I have Face Off HD DVD which has 2 audio streams worth looking at.

One is E-AC3 (DD+) 1536kbps 5.1 24bit 48khz

the other is

DTS-ES 1509kbps 6.1 24bit 48khz

I'm guessing that DTS-ES would be better considering it has 6.1 channels and is close to the bitrate of E-AC3

However I'm not sure.. Which is better for me to extract for my home library?


Second problem I have is when I extract the DTS track I get only 5.1 reported in TSMuxer or 6 channels in RipBot 264 .. not sure why I'm not getting 6.1.

When I do run eac3to audio.dts again it shows DTS-ES 6.1.

Will RipBot encode this DTS track into MKV as 6.1 if it says X.X Stream (muxing in)?


I extracted the DTS track with

eac3to 1) 5: audio.dts ..

also tried:

eac3to 1) 5: audio.dts -1536

in hopes that SurCode DTS encoder would recognize and create a proper DTS-ES 6.1 stream.. no luck..

Is it just reporting issue with aforementioned TSMuxer and RipBot or eac3to is not extract DTS stream properly?

My filters installed are as follows:

- Arcsoft
- Surcode


Thanks

[tebasuna51]

E-AC3 (DD+) 1536kbps is much better quality than DTS-ES 1509kbps.
Other question is if you have a 6.1 audio system with a DTS-ES decoder

If eac3to shows DTS-ES 6.1 then is DTS-ES 6.1, don't worry.

[Bozster]

E-AC3 (DD+) 1536kbps is much better quality than DTS-ES 1509kbps.
Other question is if you have a 6.1 audio system with a DTS-ES decoder

If eac3to shows DTS-ES 6.1 then is DTS-ES 6.1, don't worry.

Yes I do... I have a 6.1 Klipsch DIY custom speakers setup and Denon 2809ci AVR.

That's why I have a dillema.. I think that 6.1 benefit over 5.1 would be better then somewhat better compression.

I archive all my movies as DTS 1.5mbps because I can't hear the difference above that even with DTS-MA or TrueHD. The DD (480,640) and DTS (768) are different story. I can hear those being softer when compared to HD audio but anything at 1.5mbps I can't hear it.