eac3to - audio conversion tool

[rapscallion]

Hello,

I am curious about something...

I decoded an ac3-file to pcm wav with eac3to which used the libav/ffmpeg decoder.
Then i did the same thing in Linux, but there I did it straight with ffmpeg (version 0.10.3).

I compared the outputs in a wave editor and discovered that eac3to produces files with 'larger / louder' waves than ffmpeg.

How can this be while eac3to is also using the ffmpeg decoding system?

Thanks in advance for your insights.

EDIT: this is not due to the removal of dialog normalisation; I already ruled that out.
Probably because they're not the same decoders. Eac3to uses a "patched" version that only madshi can supply.

[Jan Marijniszoon]

Perhaps DRC (Dynamic Range Compression) is applied in one and not in the other? Just a wild guess...

Do you perhaps have any insight how to disable it in ffmpeg? I tried the option "-drc_scale 0" but that didn't do much.

Probably because they're not the same decoders. Eac3to uses a "patched" version that only madshi can supply.

And let me guess, madshi's version is much better tuned right?

[Jan Marijniszoon]

I fixed it; I had put the option in the wrong order in the command line.

Using this:

ffmpeg -drc_scale 0.000000 -i sound.ac3 -f wav sound.wav

gave me the exact same wav-file as was produced by eac3to.

Thank you for getting me in the right direction.

[kws53]

I have also noticed a very strange thing with the Brave 3D blu-ray. (I don't have the 2D BD to check if it has the same problem.)

Z:\BDMV\PLAYLIST>eac3to 00800.mpls 1)
M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 1:33:37, 108.607p
1: Chapters, 37 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
4: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
7: AC3, French, 5.1 channels, 512kbps, 48kHz
8: AC3, French, 5.1 channels, 640kbps, 48kHz
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish

As you can see, stream 7 is strange. It has an unusual bit rate, and, as far as I know, that stream does not exist in the BD. There is only one French audio track (listed by eac3to as stream #8). tsMuxeR does NOT show that strange French stream.

When eac3to tries to demux that track, it issues the warning "a07 This track is not clean.", but it continues anyway. The resulting ac3 file plays two times faster than it should in all audio players, and the sound is strange.

Same problem here - did anyone notice the 108.607p in the opening line? My log for the 2D version is similar but produces another bizarre number - 95.907p.

M2TS, 1 video track, 7 audio tracks, 8 subtitle tracks, 1:33:37, 95.907p
1: Chapters, 37 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz (embedded: AC3, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3, French, 5.1 channels, 512kbps, 48kHz
7: AC3, French, 5.1 channels, 640kbps, 48kHz
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: AC3, English, 2.0 channels, 320kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), English
15: Subtitle (PGS), French
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), English

Cheers,
Kurt

[damorsoft]

I have been trying for days to make the code below work.
If I set createnowindow=true and
disable standard output eac3to runs fine in the command box but just locks up when run with the command box hidden.

This same routine works for mkvmerge tsmuxer and eac3to to read the mpls files.\
I have tried read, readline and readtoend all without success.

Any thoughts thanks


Public Function StartFile(ByVal MyApp As String, ByVal MyTail As String) As Integer
Dim myProcess As New Process()
'Dim test As String = Chr(8)
myProcess.StartInfo.WorkingDirectory = Application.StartupPath
myProcess.StartInfo.FileName = MyApp
myProcess.StartInfo.Arguments = MyTail
myProcess.StartInfo.UseShellExecute = False
'myProcess.StartInfo.CreateNoWindow = false
myProcess.StartInfo.CreateNoWindow = True
myProcess.StartInfo.RedirectStandardOutput = True
'myProcess.StartInfo.RedirectStandardError = True
Try
myProcess.Start()
myProcess.PriorityClass = ProcessPriorityClass.Idle
While (myProcess.HasExited = False)
Application.DoEvents()
Dim sLine As String = myProcess.StandardOutput.ReadToEnd
If (Not String.IsNullOrEmpty(sLine)) Then
TextBox14.Text = sLine
End If

If UserCancel Then
myProcess.Kill()
ProcEC = 9
Return ProcEC
End If
End While
ProcEC = myProcess.ExitCode
Catch ex As Exception
MsgBox(" Error(s) with startfile " & MyApp & " " & ex.Message)
ProcEC = 99
End Try
myProcess.Close()
Return ProcEC
End Function

[Soul Taker]

Hi there. I'm a bit confused about something, here it is...

1: Chapters, 9 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, Japanese, 2.0 channels, 24 bits, 48kHz
4: DTS Master Audio, Japanese, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: TrueHD/AC3, Japanese, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 448kbps, 48kHz)
6: AC3, Japanese, 2.0 channels, 384kbps, 48kHz
7: Subtitle (PGS), Japanese
a04 Extracting audio track number 4...
a04 Decoding with ArcSoft DTS Decoder...
a04 Encoding FLAC with libFlac...
a04 Creating file "C:\Users\ST\Desktop\1_4_audio.flac"...
a04 The original audio track has a constant bit depth of 16 bits.
a04 Superfluous zero bytes detected, will be stripped in 2nd pass.
a04 Starting 2nd pass...
a04 Decoding FLAC...
a04 Reducing depth from 24 to 16 bits...
a04 Encoding FLAC with libFlac...
a04 Creating file "C:\Users\ST\Desktop\1_4_audio.flac"...
a04 The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 133434 frames.
eac3to processing took 10 minutes, 12 seconds.
Done.

-Is reducing the audio depth from 24 to 16 bits... normal? and i have to say, this the first time that happened to me.

[rapscallion]

At first glance, far too many Japanese audio/sub tracks. :D

Seriously though, completely normal when there is padding present. Happens all the time. By eac3to removing it doesn't change the quality at all. Still lossless.

BTW, no one can accuse you of being a post whore, 1 post in 4 years!!

[Soul Taker]

^^^

That's Berserk movie for ya... too much tracks and you only need 2 (facepalm).
LMFAO!! I'm not a forum guy... and i don't know why can't i find it interesting to "camp" 24/7 :p
BTW, i don't even post that much in our own forum "non-american,EU guy"
Anyway, thanks for your reply dude

سلام :D <~ :p

[bilditup1]

Hey all,
I tried about five times now to encode the DTS-MA audio from 'Fellowship of the Ring' using Arcsoft, eac3to, and Nero. Every single time, instead of a 1h:45min multichannel file, I get a 2:00:48 file that's all distorted and useless. I have no idea why this is happening. It's a 6.1 DTS, which there was some info about earlier in the thread about remappings and such from about a year ago. The information does not seem applicable though, since it's about a year old, and since I don't receive any error messages about channel mappings (eac3to automatically detects that it's a 6.1 file and doubles the back channel in order to facilitate compatibility with NeroAacEnc). In any case, I tried what was suggested - manual channel mapping plus the '-double7' switch - and it changed nothing. I also tried to encode just the DTS-ES core - that didn't help at all. Thanks to the Arcsoft filters, the DTS file does play in MPCHC, but a) with terrible seeking b) a reported running time of 05:24:44. I am using Arcsoft 1.1.0.8, eac3to 3.24, and Nero 1.5.4.0.
Anybody know how to encode this thing?
-e

[bilditup1]

Hmm, after tracking down several different versions, this seems to be an issue with any version of Arcsoft above 1.1.0.1 with 6.1 tracks. Getting the old DLL is the key.

[kypec]

Hmm, after tracking down several different versions, this seems to be an issue with any version of Arcsoft above 1.1.0.1 with 6.1 tracks. Getting the old DLL is the key.
Yes, exactly. I'm using version 1.1.0.0 and this works just fine, see the log below:eac3to v3.24
command line: "C:\Program Files (x86)\AVTools\eac3to\eac3to.exe" "I:\TLOTR\FOTR Part 1\BDMV\" 1) 3: D:\Source\TLOTR_EE\fotr1_eng.w64 -down16 -down6 -progressnumbers -log="fotr1_eng.log"
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 24 subtitle tracks, 1:45:43, 24p /1.001
1: Chapters, 27 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3 EX, Polish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3 EX, Thai, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
10: Subtitle (PGS), English
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Chinese
14: Subtitle (PGS), Czech
15: Subtitle (PGS), Hebrew
16: Subtitle (PGS), Polish
17: Subtitle (PGS), Romanian
18: Subtitle (PGS), Thai
19: Subtitle (PGS), English
20: Subtitle (PGS), Chinese
21: Subtitle (PGS), Korean
22: Subtitle (PGS), English
23: Subtitle (PGS), Chinese
24: Subtitle (PGS), Korean
25: Subtitle (PGS), English
26: Subtitle (PGS), Chinese
27: Subtitle (PGS), Korean
28: Subtitle (PGS), English
29: Subtitle (PGS), Chinese
30: Subtitle (PGS), Korean
31: Subtitle (PGS), English
32: Subtitle (PGS), Polish
33: Subtitle (PGS), Thai
[a03] Extracting audio track number 3...
[a03] Remapping channels...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Mixing surround channels...
[a03] Reducing depth from 24 to 16 bits...
[a03] Writing W64...
[a03] Creating file "D:\Source\TLOTR_EE\fotr1_eng.w64"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 152090 frames.
eac3to processing took 8 minutes, 17 seconds.
Done.

[Brom]

What's the proper way to transcode DTS-ES 5.1 (matrixed 6.1) to FLAC?

Is there any decoder which outputs DTS-ES 5.1 to 6.1?
Is the matrix known?

[sneaker_ger]

6.1 channel order is not defined in FLAC, so there is no "proper way" to do it.
http://flac.sourceforge.net/format.html#frame_header

[Brom]

More interested in getting the correct 7 channel wavs from a DTS-ES 5.1 Matrix.


But the reference FLAC library does the following since FLAC 1.1.3 (27-Nov-2006):
Now properly supports AIFF and WAVEFORMATEXTENSIBLE multichannel input, performing necessary channel reordering both for encoding and decoding. WAVEFORMATEXTENSIBLE channel mask is also saved to a tag on encoding and restored on decoding for situations when there is no natural mapping to FLAC channel assignments.http://flac.sourceforge.net/changelog.html

[DarkSpace]

What's the proper way to transcode DTS-ES 5.1 (matrixed 6.1) to FLAC?

Is there any decoder which outputs DTS-ES 5.1 to 6.1?
Is the matrix known?
I'm not sure if there's any decoder that natively outputs 6.1 audio for DTS-EX 5.1 (DTS-ES means 6.1 discrete channels, whereas DTS-EX means the back channel is matrix-encoded into the 5.1 channels present), but if you're using ArcSoft DTS Decoder in eac3to, you could try the -7 switch which (undocumented) asks ArcSoft to output 6.1 (7 channels). I am however not sure if that will restore the 7th channel, though, it might just as well output a mix of the present channels or add a silent back channel.

EDIT
More interested in getting the correct 7 channel wavs from a DTS-ES 5.1 Matrix.
if it's just the mono wavs for each channel you want, then you can also try decoding into those directly, using a commandline like
eac3to "input" 1:"output.wavs" -7
under the assumtion that the track number is 1.

[quartex]

noob question:
convert an AC3 448 Kbps track to an AC3 640Kbps track improve quality?

[LigH]

Of course not. Dolby Digital (AC3) is a lossy format which stores frequencies of the audible spectrum with a limited precision and suppresses frequencies it calculates as subjectively inaudible. The source (with 448 kbps) already lost some frequency parts of the original sound which was recorded on the movie set. Raising the bitrate afterwards doesn't bring these lost frequencies back, they will still be lost. You cannot "invent" more quality than the source contained. And each conversion to a lossy format — even with a higher bitrate than the source — reduces the quality even more.

[sporic]

Any fix for Brave and Total Recall yet?

[bilditup1]

Hmm...so I have a DTS track with 5 channels. It isn't a theater DTS track...just DTS from a commercial Blu-Ray that has no LFE (Adventures of Tintin series from the early 90s). I want to encode this to AAC with a fake LFE so that the usual suspects won't choke on it (iOS hardware decoding). Using the -6 option with eac3to yields a '6-channel' track consisting of the right and left channels and 4 channels of silence, so that's not it. Is there some easy switch that'll enable me to do this with eac3to, or is a more manual approach required?

ED: Looking into it more, seems to be that the commercial DVD itself only had LPCM at 2.0, and the release group botched the audio encode somehow.

[tebasuna51]

I have a DTS track with 5 channels ...from a commercial Blu-Ray that has no LFE... Using the -6 option with eac3to yields a '6-channel' track consisting of the right and left channels and 4 channels of silence...
Please put the log.

BTW, seems your DTS 5.0 have 3 channels of silence.
Maybe you can try something like:

eac3to input.dts stdout.wav | sox -t wav - -t wav - remix -m 1 2 | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

[jpsdr]

Does someone know how i can extract the 6 channels from a .m2ts file when audio is PCM ? I want to have all 6 channels in separate files, raw format is fine.

Thanks.

[bilditup1]

Please put the log.

BTW, seems your DTS 5.0 have 3 channels of silence.
Maybe you can try something like:

eac3to input.dts stdout.wav | sox -t wav - -t wav - remix -m 1 2 | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

Right, I figured out that it was a bad encode. I don't think the resulting 2ch files (which register as 6ch, with 4ch of silence) are any bigger than normal 2ch files. They're just recognized as 6ch files (as they should be) Will the above code take only the front two channels? Will using -down2 instead of that in this case muck up the audio or something?

[tebasuna51]

Will the above code take only the front two channels?
Yes.

Will using -down2 instead of that in this case muck up the audio or something?
Nope, because -down2 can't be applied to a 5.0 source, must be 5.1.

[tebasuna51]

Does someone know how i can extract the 6 channels from a .m2ts file when audio is PCM ? I want to have all 6 channels in separate files, raw format is fine.

eac3to input.m2ts PCM_TRACK: output.wavs

[madshi]

eac3to v3.25 released

http://madshi.net/eac3to.zip

* fixed: option "-down6" didn't handle 6.1 tracks properly
* fixed: option "-down6" for 7.1 content lowered side channels in volume
* fixed: 6.1 DTS/E-AC3 libav decoding resulted in incorrect 70f channel mask
* fixed: a couple of funny AC3 channel configs resulted in swapped channels
* fixed: option "-check" failed for SRT subtitles
* fixed: MKV tracks without language information didn't default to "English"
* fixed: 16bit MLP decoding didn't work at all
* fixed: DTS 2.1 decoding had empty sub channel when using ArcSoft decoder
* fixed: muxing h264 video streams without VUI eventually crashed
* fixed: SMPTE 302M 20bit PCM tracks didn't extract properly
* improved AC3 seamless branching handling by detection of identical frames
* improved DTS seamless branching handling by detection of identical frames
* updated to latest ffmpeg/libav revision
* added support for libav DTS 6.1 and 7.1 decoding (only in core, no DTS-HD)
* added support for encoding all AC3 channel configurations
* added support for demuxing Blu-Ray primary E-AC3 tracks (AC3 core)
* libav is now the default decoder for AC3 and E-AC3
* option "-down2" is now deprecated, but still supported
* option "-downDpl" mixes down to Dolby Pro Logic II
* option "-downStereo" mixes down to standard stereo
* added support for mixing any channel configuration to stereo/DPL
* DPL downmixing doesn't lower volume, anymore (except when clipping occurs)
* editing audio tracks by adding silence is now less restrictive
* trying to mux right eye stream to MKV errors out now, instead of crashing
* doing "eac3to 1) movie.mkv" for 3D BDs now properly skips right eye stream
* FLAC encoding is now done with optimized block sizes for "LossyWAV" files
* encoder bitdepth for lossy DTS tracks is no longer displayed
* option "-test" no longer checks mkvtoolnix
Thanks to tebasuna51 for moderating the bug/improvement list, and for investigating the "duplicate frame" stuff for seamless branching Blu-Rays. I didn't know that overlapping AC3 and DTS frames were actually bit-wise identical! With this information eac3to can now discard such frames immediately, which means that it's no longer necessary to do a 2nd pass for seamless branching Blu-Rays to fix audio overlaps. This new logic should also avoid any audio glitches at the seamless branching edit points - because simply no edits are necessary, anymore, at all. Please note that this only applies to AC3 and DTS (and DTS-HD/MA) tracks. It does not apply to LPCM and TrueHD tracks. However, overlaps with LPCM and TrueHD tracks are rather small, so overlap editing is often not necessary for such tracks, anyway. The new logic also only works if the overlapping AC3 and DTS frames are truely identical. They have been with the 2 Blu-Rays I've tested with. But I don't know for sure if that's the case for *all* seamless branching Blu-Rays. So there's a chance the old overlap logic might still be used for some Blu-Rays.

It seems that mono TrueHD decoding is currently broken when using libav/ffmpeg decoding, but I'm not 100% sure because I only have one such sample and it's pretty short. Maybe you guys can double check (and provide me with a longer sample)? This looks like a libav/ffmpeg bug to me, though, if it's really a bug at all. At least Brave TrueHD decoding seems to be working fine now.

Please note that although eac3to now supports parsing, demuxing and core-extraction for Blu-Ray primary E-AC3 tracks, currently no decoder is available which can decode the full audio information with all 6.1/7.1 channels. Currently only the AC3 core can be decoded which results in 5.1 output only.

[sneaker_ger]

Thanks, that was faster than expected.

[Adub]

Woot! Thanks for the release madshi! You've got quite the plethora of fixes there!

[LigH]

Happy New Year, madshi - and thank you for a first of many new versions of eac3to... ;)

[Boulder]

Thank you, it's been a while :)

[sl1pkn07]

zankius!

[nickintheforest]

Hello, madshi!
First of all, big THANK YOU for eac3to, especially for support and update this great program!
And I'd like to ask you one question: why you didn't add azid.dll to eac3to? There is some troubles to decode DD, because you have to demux DD with eac3to and decode it with azid or nicaudio for correctly decoding DD. You know, nero don't ignore DRC (http://forum.doom9.org/showthread.php?p=1404212#post1404212) and libav downturn of high frequencies after 17 kHz. You can see (E-)AC3 decoders comparison here (http://forum.doom9.org/showthread.php?t=161654).
It would be very nice, if we can decode ac3 correctly only by eac3to, like "eac3to test.ac3 test.wavs -azid".
Thanks.

[soneca]

Thanks madshi!

[LigH]

:goodpost: nickintheforest ... the azid.dll should still be available from many BeSweet related sources.

[madshi]

first of many new versions of eac3to... ;)
Haha! My focus is still on madVR for the time being...

And I'd like to ask you one question: why you didn't add azid.dll to eac3to? There is some troubles to decode DD, because you have to demux DD with eac3to and decode it with azid or nicaudio for correctly decoding DD. You know, nero don't ignore DRC (http://forum.doom9.org/showthread.php?p=1404212#post1404212) and libav downturn of high frequencies after 17 kHz. You can see (E-)AC3 decoders comparison here (http://forum.doom9.org/showthread.php?t=161654).
It would be very nice, if we can decode ac3 correctly only by eac3to, like "eac3to test.ac3 test.wavs -azid".
Sorry, no support for azid planned. The test you're linking to is 1.5 years old and the guy reponsible for the libav AC3 decoder has seen and commented on the test results those 1.5 years ago, too, so I would say a retest would be needed to make sure those results are still valid today. And if they are, a bug report should be created for libav/ffmpeg, so that the problem is fixed.

[Snowknight26]

For best AC3, E-AC3 and AAC decoding you need:
(1) Nero 7 (Nero 8 won't work!)
(2) Nero HD DVD / Blu-Ray plugin
Is that still accurate or is libavcodec on par now?

-phaseShift shift phase (when doing stereo downmixing, see "down2")
-mixlfe mix LFE in (when doing stereo downmixing, see "down2")

Should say downStereo instead of down2.

[LigH]

Not downStereo, but rather downDpl. Back channel phase shift is a typical feature of Dolby ProLogic downmixing; basic stereo downmixing would not care much about the back channels.

[88keyz]

Curious, why the change from Nero to libav as the preferred DD decoder?

[bilditup1]

This is a significant update. I'll have to redo audio for all my recent encodes. Thank you madshi!

[Snowknight26]

Errors that libavcodec spits out break the progress bar:

v02 Extracting video track number 2...
a03 Extracting audio track number 3...
v02 Muxing video to Matroska...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 24 to 16 bits...
a03 Encoding FLAC with libFlac...
a03 Creating file "G:\brave.flac"...
a03 Skipping identical AC3 frames (seamless branching)...
-[truehd @ 009471a0] Lossless check failed - expected 00, calculated 15.
[truehd @ 009471a0] Lossless check failed - expected 00, calculated e7.
---[truehd @ 009471a0] End of stream indicated.
[truehd @ 009471a0] Lossless check failed - expected 00, calculated 91.

[robpdotcom]

Thanks for the update, madshi. :thanks:

[bilditup1]

eac3to v3.25 released


* added support for mixing any channel configuration to stereo/DPL



Hey there madshi,
This still doesn't appear to work with 5.0 DTS files.

[Guest]

Be a sport and give madshi a stream that he can use to reproduce that. :thanks:

[bilditup1]

Be a sport and give madshi a stream that he can use to reproduce that. :thanks:

Aye aye cap'n. It's an edge-case to be sure.
[removed link by tebasuna51]

[sundance]

@madshi:

Options:
-downDpl downmix multi channel audio to Dolby Pro Logic II
-downStereo downmix multi channel audio to simple Stereo
-phaseShift shift phase (when doing stereo downmixing, see "down2")
-mixlfe mix LFE in (when doing stereo downmixing, see "down2")
Since "down2" is deprecated, you might want to remove the references (see "down2") to avoid a newbie's confusion...
Anyway, thanks a lot for this invaluable tool!

[tebasuna51]

We can't support downloaded (and fake) files.

But is't true eac3to 3.25 don't downmix 5.0 files (tested DTS, AC3 and WAV).

[get8p]

madshi, thanks a lot for numerous fixes and improvements.

I asked that before and I don't want to annoy anyone, but I wanted to ask you, as a dev, personally - is there any chances for these ipms in a future versions?
• avi, mp4, mov, mpg, flv demuxing(maybe even muxing, that would be fantastic). Basically, avi/mp4 support is most important feature I think eac3to is lacking, along with
• Proper ac3 decoding. Whilst a freeware prog like azid can properly decode ac3, it's a real shame such powerful instrument as eac3to uses nero/libav for that, because both decoders has defects (http://forum.doom9.org/showthread.php?t=161654).
• and also, downmix to mono and ac3/dts dialnorm applying - minor, but still useful features.

/Thank you.

[phate89]

For noob people like me, what's the difference between downmix to dolby plII and to stereo?

[kukushka]

+1 for azid, thanks (anyway)

[tebasuna51]

@all
To report bugs please use the thread http://forum.doom9.org/showthread.php?t=166487

@get8p
please read http://forum.doom9.org/showthread.php?p=1608655#post1608655
and http://forum.doom9.org/showthread.php?t=166487
Use -keepDialnorm.

@phate89
a mix DPL II can recover the surround (more or less) with the appropiated decoder in audio receivers.

[phate89]

@phate89
a mix DPL II can recover the surround (more or less) with the appropiated decoder in audio receivers.
But if my goal is to convert the track to aac or mp3 through wav it's useless and i should use -downtostereo right?