eac3to - audio conversion tool

[deathlord]

Hi madshi

Great to read from you!

eac3to uses simple "WriteFile" calls. That's the very core win32 API for writing files. I don't see what I could change there. Maybe I'm writing too big chunks at a time? That's not really my fault, though. If your RAID array can't cope with that, I'd consider that a big bug of your RAID array - or of its driver or something. Maybe you change the drive's cache settings somehow? Don't know...

I have tried everything in this respect. Eac3to can't write if the variable sector size feature is used. I have not had any troule with any other application, ever.
Currently, I do not require eac3to to write to such an array. So personally, I'm ok at this time.
However, since this might become annoying to myself or others in the future, it would be cool if it could be fixed.
Maybe it is a bug in the highpoint driver. But since highpoint do not respond to emails, there is nothing I can do about it.
If you want to track down the error, you could (temporarily) add an option to change the size of the chunks eac3to writes, so I could check if it has an influence. I don't know if this is difficult for you to do, though.

Another thing:
My "CSI: The First Season" Blu-ray contains 1080i60 VC-1. The resulting mkv is stuttering heavily in MPC, as opposed to any other VC-1 (or any Blu-ray i have tried, for that matter). You have mentioned problems with this format, so I guess it is that. Just let me know if you need a sample.

[73ChargerFan]

deathlord, I've just ripped about a dozen 1080i60 "extras" and found that -keepdialnorm to a mkv in eac3to, and then in mkvmerge setting the frame rate for that file to 60/1001 works like a charm. No more problems.

eac3to 1) 1: extra1.mkv 2: extra1.ac3 -keepdialnorm
then mux in mkvmerge

[Snowknight26]

keepDialNorm has nothing to do with successfully remuxing 1080i60 content.

[deathlord]

deathlord, I've just ripped about a dozen 1080i60 "extras" and found that -keepdialnorm to a mkv in eac3to, and then in mkvmerge setting the frame rate for that file to 60/1001 works like a charm. No more problems.

eac3to 1) 1: extra1.mkv 2: extra1.ac3 -keepdialnorm
then mux in mkvmerge

Are you talking about *VC-1* 1080i60 material?
As 73ChargerFan sais, dialnorm has nothing to do with it. But I haven't tried setting the framerate manually in mkvmerge. Does eac3to not write this value to the demuxed mkv file?

[Snowknight26]

It does write it. The only thing you could try is using the -seekToIFrame command, but that'll probably be fruitless. Infact, the issue is probably either a splitter or a decoder issue, not an eac3to issue... but who knows.

[MatMaul]

@madshi : do yo think you can patch the ffmpeg eac3 decoder with this patch to support spectral extension ?
http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/attachments/20090805/381dbc22/attachment.diff

I would like to use eac3to with some of my DVB records but the eac3 tracks have spectral extension.

thanks !

[deathlord]

It does write it. The only thing you could try is using the -seekToIFrame command, but that'll probably be fruitless. Infact, the issue is probably either a splitter or a decoder issue, not an eac3to issue... but who knows.

I will try to use the ffdshow decoder in place of the one from windows. Maybe that helps.
I remember madshi writing something about 1080i60 VC-1 handling being imperfect atm. But I can't find it right now.

[Jeff Flowerday]

I will try to use the ffdshow decoder in place of the one from windows. Maybe that helps.
I remember madshi writing something about 1080i60 VC-1 handling being imperfect atm. But I can't find it right now.

There are definately issues with 1080i VC-1. It wasn't high on madshi's priorities because of it's rarity.

[kypec]

I couldn't figure out how to make bitrate based (not quality) AAC audio through eac3to and NeroAAC encoder with GUIs available here so I made myself a command script/batch for this very purpose. Should work on all Win2K/XP/Vista/7.
Feel free to modify it to your needs.:cool:
Single file usage example: aac.cmd VTS_01_1.VOB
Multiple files usage example: aac.cmd VTS_01_1.VOB+VTS_01_2.VOB please note there MUST NOT be space around plus sign when joining files
@echo off
set decoder="%PROGRAMFILES%\AVTools\eac3to\eac3to.exe"
set encoder="%PROGRAMFILES%\AVTools\eac3to\neroAacEnc.exe"
cls
if '%1'=='' goto error1
echo Analyzing input file(s). Please wait...
%decoder% %1
echo.
:choosetrack
set /p track=Type the number of track you want to convert:
echo Extracting selected track, please wait...
%decoder% %1 %track%:output_%track%.wav -normalize -libav
if %ERRORLEVEL% EQU 0 (goto choosecompression) else (goto error2)
:choosecompression
echo.
echo Option 1 = 96 kbps (2.0 stereo, medium quality)
echo Option 2 = 128 kbps (2.0 stereo, high quality)
echo Option 3 = 256 kbps (5.1 multichannel, medium quality)
echo Option 4 = 320 kbps (5.1 multichannel, high quality)
set /p quality=Choose your desired bitrate option (1, 2, 3, 4):
if not '%quality%'=='' set quality=%quality:~0,1%
if '%quality%'=='1' goto option_a
if '%quality%'=='2' goto option_b
if '%quality%'=='3' goto option_c
if '%quality%'=='4' goto option_d
echo Invalid option specified, try again
goto choosecompression

:error1
echo You must specify input VOB file(s) on the command line as a parameter!
echo Example (single file input): aac VTS_01_1.vob
echo Example (multiple files input): aac VTS_01_1.vob+VTS_01_2.vob
goto end
:error2
echo Error occurred while extracting track number %track%!
echo Please try again.
goto choosetrack
:error3
echo Error occurred while encoding file output_%track%.wav!
echo Please try again.
goto choosecompression

:option_a
set aac_options=-2pass -br 96000
goto encode
:option_b
set aac_options=-2pass -br 128000
goto encode
:option_c
set aac_options=-2pass -br 256000
goto encode
:option_d
set aac_options=-2pass -br 320000
goto encode

:encode
echo Encoding selected track, please wait...
%encoder% %aac_options% -if output_%track%.wav -of output_%track%.mp4
if %ERRORLEVEL% EQU 0 (exit) else (goto error3)
:end

[RainyDog]

Due to Miramax giving lossless honours to the English dub only on their Hero blu-ray, I've remuxed the original Mandarin DTS-ES track from the R3 DVD with an x264 encode of the blu-ray video. However, I needed to add 7900ms of delay for the A/V to sync up and when I put the DTS track through eac3to for this the resulting track has 7900ms of 'noise' at the beginning instead of silence. Is there anything I can do to avoid this? I simply imported the DTS track into eac3to (using the GUI), added the delay, chose to save as Hero_delay, selected .dts as the output format, and ran the command line.

Thanks in advance for any help.

[73ChargerFan]

Perhaps you can apply no audio delay in eac3to, and instead use apply the delay for that track in mkvmerge?
Or, if this will be the only track, clip off the first 8 seconds of the video (I don't know what tool though.)

[RainyDog]

Perhaps you can apply no audio delay in eac3to, and instead use apply the delay for that track in mkvmerge?
Or, if this will be the only track, clip off the first 8 seconds of the video (I don't know what tool though.)

Thanks for the response, 73ChargerFan. I've already chopped off the first 20 odd seconds of the video (the Miramax logo) using mkvmerge and can't lose anymore without cutting into the film itself. I've also tried adding the audio delay using mkvmerge, which works of course, and could settle on that solution. But I'm just been a perfectionist and looking to have both streams exactly the same length :) I've also tried delaycut which does do the same job as eac3to and without the noise... But unfortunately, that doesn't keep the DTS-ES Matrix header.

[73ChargerFan]

Perfection: Only you know the hacks you did to make it work.
Everyone else "wow, perfect dude!"

[djloewen]

I'm having some trouble figuring out exactly when eac3to does, and does not, correct sync/delay issues.

1.) If I extract a blu-ray audio track as WAV (or mp3 or aac), it applies any necessary delay directly to the WAV file, so if I convert/mux it later I don't need to add any delay at that point - is this correct?
2.) If I extract that same track without doing any conversion (say, as AC3), does it do the same thing? Or might there be a delay issue in this case?

[tebasuna51]

1) Yes
2) Yes. No

[TinTime]

2.) If I extract that same track without doing any conversion (say, as AC3), does it do the same thing? Or might there be a delay issue in this case?

TrueHD won't have the delay applied if you demux it. However the delay will be added to the filename I believe. If you decode it (instead of demux) then the delay will be applied. All other audio formats will be corrected automatically.

[ron spencer]

I am getting the $40000 errors with I am Legend HD DVD...3.16 is fine though 3.17 has the issue

[raquete]

eac3to v3.17
command line: eac3to sampleflac.flac samples.waves
-----------------------------------------------------
FLAC, 5.1 channels, 0:03:11, 24 bits, 4446kbps, 48khz
This audio conversion is not supported. <ERROR>

what i'm doing wrong and getting error?!

[Midzuki]

raquete wrote:

command line: eac3to sampleflac.flac samples.waves

Try wavs instead of waves.

[raquete]

raquete wrote:



Try wavs instead of waves.

:rolleyes: lol, i knew, was my fault....and is very clever in the help file that i read lots of times. :p
thanks so much Midzuki! :)

[Snowknight26]

I am getting the $40000 errors with I am Legend HD DVD...3.16 is fine though 3.17 has the issue

Redownload eac3to, was fixed a couple pages back...

[honai]

Can someone confirm that the eac3to + Arcsoft 1.1.0.5 (registered manually, legit version) toolchain works fine under Windows 7 Professional?

[ron spencer]

Thanks...did not realize the bugfix.

[Atak_Snajpera]

Can someone confirm that the eac3to + Arcsoft 1.1.0.5 (registered manually, legit version) toolchain works fine under Windows 7 Professional?
http://img266.imageshack.us/img266/6038/new1q.jpg

[Blue_MiSfit]

I can verify this also. No problems whatsoever with any DTS / HD / MA track I've touched in the last ~ 4 months - since RC1 :)

~MiSfit

[honai]

Thanks!

[SomeJoe]

I have tried everything in this respect. Eac3to can't write if the variable sector size feature is used. I have not had any troule with any other application, ever.
Currently, I do not require eac3to to write to such an array. So personally, I'm ok at this time.
However, since this might become annoying to myself or others in the future, it would be cool if it could be fixed.
Maybe it is a bug in the highpoint driver. But since highpoint do not respond to emails, there is nothing I can do about it.
If you want to track down the error, you could (temporarily) add an option to change the size of the chunks eac3to writes, so I could check if it has an influence. I don't know if this is difficult for you to do, though.


I have played with various arrays that support this feature, where the base block size of the device is larger than the standard 512 bytes.

I highly recommend against using this feature in any array. The NTFS file system routines in Windows were never designed to work with block devices with sector sizes other than 512 bytes. I have seen very odd behavior on systems that have one of these arrays installed in this mode.

The proper, known-to-work methods to use an array >2 TB are:

1) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Map the first LUN into the file system on a drive letter. Map other ones into the file system on folders under that drive letter. This allows all the storage to be seen, but each folder's space can't be used by other folders.

2) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Use Windows dynamic disks to pool all the 2TB LUNs together into one dynamic disk.

3) Define a >2TB LUN using 512-byte sectors, using GPT disk layout (requires Windows XP64, Server 2003 SP1 x86 or x64, Server 2008 x86 or x64, Server 2008 R2 x64 only, Vista x86 or x64, or Windows 7 x86 or x64). Format the large LUN with NTFS. This option requires that your drivers and device support 48-bit LBA for IDE/SATA devices or 64-bit LBA / 16-byte CDB for SCSI API-compatible controllers. Not all devices do, so check with the device manufacturer to be sure.

[Thunderbolt8]

got a problem with a H.264 .ts 50i(?) capture. plays allright in its normal .ts container (if you dont use ffmpeg-mt), but sync & playback speed is off when remuxing it .mkv (using the gap file doesnt really seem to make a difference). audio speed seems to be normal, but video playback looks like its running at 2x speed. displayed length of the video in mpc is ~54mins, even though mediainfo tells 1h 48mins and the original .ts video also is 1h 48mins. cannot tell how much the audio is off, as the spoken language is not english.

50mb sample: http://www.sendspace.com/file/qltp8n

and the log: http://www.sendspace.com/file/tnywyl

[73ChargerFan]

I had the exact same experience a few weeks ago with some 60i sources.

-keeppulldown fixed it.

[Thunderbolt8]

what does pulldown mean here?

[73ChargerFan]

-keepPulldown disable removal of pulldown for MPEG2, h264 and VC-1 tracks

Now I'm confused. I had tried a few things, and iirc keeppulldown allowed the file to play correctly. It was with with the extras on Region 1 Superman/Batman Public Enemies BD.

Also when I found video gaps I re-ran the eac3to xxx.m2ts xxx.mkv command which used the gaps file.

EDIT - Sendspace really s**ks. I can't get the TS file.

[deathlord]

I have played with various arrays that support this feature, where the base block size of the device is larger than the standard 512 bytes.

I highly recommend against using this feature in any array. The NTFS file system routines in Windows were never designed to work with block devices with sector sizes other than 512 bytes. I have seen very odd behavior on systems that have one of these arrays installed in this mode.

The proper, known-to-work methods to use an array >2 TB are:

1) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Map the first LUN into the file system on a drive letter. Map other ones into the file system on folders under that drive letter. This allows all the storage to be seen, but each folder's space can't be used by other folders.

2) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Use Windows dynamic disks to pool all the 2TB LUNs together into one dynamic disk.

3) Define a >2TB LUN using 512-byte sectors, using GPT disk layout (requires Windows XP64, Server 2003 SP1 x86 or x64, Server 2008 x86 or x64, Server 2008 R2 x64 only, Vista x86 or x64, or Windows 7 x86 or x64). Format the large LUN with NTFS. This option requires that your drivers and device support 48-bit LBA for IDE/SATA devices or 64-bit LBA / 16-byte CDB for SCSI API-compatible controllers. Not all devices do, so check with the device manufacturer to be sure.

Thanks for the info!

[73ChargerFan]

got a problem with a H.264 .ts 50i(?) capture. plays allright in its normal .ts container (if you dont use ffmpeg-mt), but sync & playback speed is off when remuxing it .mkv (using the gap file doesnt really seem to make a difference). audio speed seems to be normal, but video playback looks like its running at 2x speed. displayed length of the video in mpc is ~54mins, even though mediainfo tells 1h 48mins and the original .ts video also is 1h 48mins. cannot tell how much the audio is off, as the spoken language is not english.

50mb sample: http://www.sendspace.com/file/qltp8n

and the log: http://www.sendspace.com/file/tnywyl
I've got it figured out:

eac3to test.ts -demux
(version 3.17, ignore erroneous gap warnings)

mkvmerge GUI v2.9.8:
- add test.h264, select stream, go to Format specific options tab, FPS "50000/1001", display width/height set to 1920x1080
- add test.ac3 stream
- merge to test.mkv

mpc-hc v1299 plays test.ts & test.mkv correctly.

BUG - 1 minute 50i stream "eac3to test.ts 1: test.mkv" detects gaps incorrectly. When run a second time, the gaps file generates a test.mkv video file that is over 4 minutes in length.

[heerschop]

Hello madshi,

I use Eac3to to quickly convert my mp3 files to ac3. Recently I have noticed that some ac3 files were distorted.
So I converted the mp3 to wav and they were also distorted.
After a bit experimenting I discovered the problem.
It turns out that libav/ffmpeg reports a wrong bit depth to eac3to.
The bit depth of the mp3 is 16 bits as reported by nero but libav/ffmpeg think it is 32 bits. This is causing the distorting. The problem is salved when I use nero to decode.

Hope you can fix the problem.

Greetings,
Ben

Used Eac3to 3.17

Let nero decode:

C:\tools\eac3to\eac3to.exe J:\term.mp3 J:\termb.ac3 –nero –down16

MP3, 2.0 channels, 1:57:18, 128kbps, 48khz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 2.0 channels, 16 bits, 48khz
Encoding AC3 <448kbps> with libAften...
Creating file "J:\termb.ac3"...

Let libav/ffmpeg decode:

C:\tools\eac3to\eac3to.exe J:\term.mp3 J:\termb.ac3 -down16

MP3, 2.0 channels, 1:57:18, 128kbps, 48khz
Decoding with libav/ffmpeg...
Reducing depth from 32 to 16 bits...
Encoding AC3 <448kbps> with libAften...
Creating file "J:\termb.ac3"...

[Snowknight26]

MP3 files have no bit-depth so ffmpeg uses 32-bit floating point by default. -full which uses 64-bit floating point and -down16 which downsamples the audio to 16 bits still causes the issue, so it's not a bit-depth issue. All that eac3to probably needs is an updated libavcodec.

[Thunderbolt8]

I've got it figured out:

eac3to test.ts -demux
(version 3.17, ignore erroneous gap warnings)

mkvmerge GUI v2.9.8:
- add test.h264, select stream, go to Format specific options tab, FPS "50000/1001", display width/height set to 1920x1080
- add test.ac3 stream
- merge to test.mkv

mpc-hc v1299 plays test.ts & test.mkv correctly.

BUG - 1 minute 50i stream "eac3to test.ts 1: test.mkv" detects gaps incorrectly. When run a second time, the gaps file generates a test.mkv video file that is over 4 minutes in length.
thanks for testing!
so does the error occur, because eac3to mistakes the framerate as 25i instead of 50i?
as the log said: [v01] The video bitstream framerate field doesn't seem to match the timestamps. <WARNING>

[73ChargerFan]

I don't know where the bug is, but I think the gap warnings are incorrect.
Using the gaps file doubles the length of the video, more or less.
Ignoring the gaps yielded the correct result.

[Chumbo]

I'm looking for some help on a strange one that I've been unable to mux. Basically, when I load the M2V file into MPC-HC and let the player load the accompanying AC3 audio track, the movie plays fine and stays in sync all the way through. However, when I mux it, the audio/video go out of sync. I've tried everything I can think of, i.e., mux to MKV, TS and M2TS containers plus tried different frame rates and so on but it's always just off at the end. Probably around 3 seconds off when using 59.940 fps.

I also no longer have the original TS file which irritates me that I deleted it. Below is the eac3to log from when it was demuxed and you can obviously see the video overlaps are many. For brevity, I've removed the myriad of lines, but essentially they appear all the way through every 3 and 4 seconds apart.eac3to v3.17
command line: eac3to "l:\hd movies\my.movie_720p_dd51.ts" -demux
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:50:49, 59.960p
1: MPEG2, 720p60 /1.001 (16:9)
2: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "my.movie_720p_dd51 - 1 - MPEG2, 720p60.m2v"...
[a02] Creating file "my.movie_720p_dd51 - 2 - AC3, 5.1 channels, 384kbps, 48khz.ac3"...
[v01] Video overlaps for 1 frames at playtime 0:01:42. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:46. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:49. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:53. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:56. <WARNING>
...
[v01] Video overlaps for 1 frames at playtime 1:50:33. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:37. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:40. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:44. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:47. <WARNING>
Video track 1 contains 398661 frames.
eac3to processing took 9 minutes, 53 seconds.
Done.
Any ideas on remuxing this to an MKV container would be greatly appreciated. And yes I did try the listed frame rate of 59.960 and that didn't help either. :(

[EDIT] Wouldn't you know it, after I post this I realized something. The one thing I didn't try is tsmuxer with a custom frame rate setting in a meta file and running the command-line. The UI does NOT allow you to enter a custom FPS setting nor does it let you edit the Meta file section so I was limited to 59.9401. Any how, setting the fps to 59.960 did work. I'm not sure why I don't get the same result for an MKV, but I'm happy with having a playable M2TS file. Whew...

[EDIT2] I got the MKV mux working correctly now. The timecodes file alone with 59.960 wasn't enough. I also manually added 60020/1001 to the FPS input on the "Format specific options" tab for the M2V stream and that did the trick.

[Snowknight26]

I accidentally encoded a file with the wrong framerate, so when I tried to change it (and demux the audio at the same time), eac3to spat out a lot of video overlaps that shouldn't be there:

G:\Encoding Tools\eac3to>eac3to.exe ..\temp\ipcressfile.done.mkv 1: ..\temp\ipcr
essfile.h264 -changeto24.000 2: ..\temp\ipcressfile.ac3
MKV, 1 video track, 1 audio track, 1:47:40, 24p
1: h264/AVC, English, 1920x816 24p /1.001
2: AC3, English, 5.1 channels, 640kbps, 48khz
"AC3, 5.1 channels, 640kbps"
v01 The video bitstream framerate field doesn't match the container framerate.
v01 Extracting video track number 1...
a02 Extracting audio track number 2...
v01 Writing new framerate "24fps" to bitstream.
v01 Creating file "..\temp\ipcressfile.h264"...
a02 Creating file "..\temp\ipcressfile.ac3"...
v01 Video overlaps for 4 frames at playtime 0:05:27.
v01 Video overlaps for 2 frames at playtime 0:05:46.
v01 Video overlaps for 2 frames at playtime 0:05:51.
v01 Video overlaps for 3 frames at playtime 0:05:57.
v01 Video overlaps for 11 frames at playtime 0:06:05.
v01 Video overlaps for 1 frames at playtime 0:06:38.
v01 Video overlaps for 15 frames at playtime 0:06:40.
v01 Video overlaps for 4 frames at playtime 0:06:42.
v01 Video overlaps for 13 frames at playtime 0:07:38.
v01 Video overlaps for 1 frames at playtime 0:08:16.
v01 Video overlaps for 1 frames at playtime 0:08:19.
v01 Video overlaps for 2 frames at playtime 0:08:21.
v01 Video overlaps for 15 frames at playtime 0:08:29.
v01 Video overlaps for 2 frames at playtime 0:09:12.
v01 Video overlaps for 1 frames at playtime 0:09:18.
[...]

Also, if you Ctrl+C at the wrong time, the colors stay in the command prompt, but I'm not sure if theres a fix for that.
http://i38.tinypic.com/fvcsh0.jpg

[ACrowley]

I had Overlaps on some3 DTS HD Tracks too.

But the Output from eac3to was perfectly fine after 2nd Pass.

[Terrachild]

I have two questions for you:

1) I've recently gotten into a debate with some people online that use the Surcode plug-in. I need some facts. If I use eac3to to encode an ac3 file, is the quality substantially different then Surcode? Is FFmpeg or Aften better?

2) I created a 6 track wave file in audacity, when I try to convert it with eac3to to an ac3 file, I get an error that says : "

eac3to v3.17
command line: "C:\Users\Admin\Desktop\eac3to\eac3to.exe" "C:\Users\Admin\Desktop\Surround Test2.wav" "C:\Users\Admin\Desktop\Surround Test2.ac3"-448
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:00:31, 16 bits, 4608kbps, 48khz
This audio conversion is not supported. <ERROR>"


I've tried different bitrates out of Audacity, but nothing works. What am I doing wrong.

[Beastie Boy]

eac3to v3.17
command line: "C:\Users\Admin\Desktop\eac3to\eac3to.exe" "C:\Users\Admin\Desktop\Surround Test2.wav" "C:\Users\Admin\Desktop\Surround Test2.ac3"-448
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:00:31, 16 bits, 4608kbps, 48khz
This audio conversion is not supported. <ERROR>"


I've tried different bitrates out of Audacity, but nothing works. What am I doing wrong.

Try inserting a space before -448. It could be a syntax error.

Cheers, Beastie.

[Terrachild]

Thanks Beastie, that was it.
I'm using the GUI by The_Keymaker, version 2.00.
In the command line Preview it shows a space where you said to put it, but when the command window opens there is no space there. What's going on? Is that a bug in the GUI? Is there a better GUI to use?

[Beastie Boy]

Personally I use the command line, but this GUI (http://forum.doom9.org/showthread.php?t=141829) by Greif is a good option.

[tebasuna51]

1) I've recently gotten into a debate with some people online that use the Surcode plug-in. I need some facts. If I use eac3to to encode an ac3 file, is the quality substantially different then Surcode?

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768

Is FFmpeg or Aften better?.
The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

[Terrachild]

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768


The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

Audacity uses FFmpeg for encoding ac3. Should I stick with that, or is there an advantage to eac3to with Aften?

Do they come close to Surcode for quality?

[leeperry]

AC3 640 > DTS 1536
got any source about this please? the more I upgrade my sound system, the less I stand 1.5mbit DTS :mad:

[Terrachild]

To answer another debate I've entered into, could someone tell me if it's legal to use an open-source ac3 encoder for non-commercial use. Just a guy sitting at home testing out surround sound. Does the act of simply making an ac3 break any laws in the U.S.?

[tebasuna51]

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

Audacity uses FFmpeg for encoding ac3. Should I stick with that, or is there an advantage to eac3to with Aften?
Well, ffmpeg have many forks, branches, compiles, ... then I can't know what is your ffmpeg and can't say nothing about.

I can't test all ffmpeg versions, I always use Aften.

Do they come close to Surcode for quality?

I don't know blind test between Aften and Surcode.
For me, with my audio equipment and my ears, sound the same.

Does the act of simply making an ac3 break any laws in the U.S.?
I don't know in U.S. but we are talking about audio encoded (A/52) AC3 compatible, not Dolby Digital.

[MokrySedeS]

if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

I've made a test but still... I'm not sure what is better.


Here it is:

Original wav............................................ Slowed down wav.................................... eac3to encode......................................... Aften encode
http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/wavoryg.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/wavoryg.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/wavslowed.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/wavslowed.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/eac3to.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/eac3to.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/aften.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/aften.png)

Original wav is encoded using eac3to from 448kbps 25fps AC3 track.
Slowed down wav is made the same way original wav was, just slowed to 23.976fps.
eac3to encode is made from slowed down wav and the only used setting was bitrate at 448kbps.
Aften encode is made from slowed down wav, bitrate at 448kbps and tweaked -w option (bandwidth).

When I did an aften encode leaving -w set at default, the result was exactly the same as eac3to's.

So what do you guys think... should I use aften and tweak bandwidth or is it a bad idea?