eac3to - audio conversion tool - Page 485

tebasuna51 · 11th January 2010, 21:03

Quote:

Originally Posted by PowerGamer

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro?

I never support commercial soft.
If the wav audio data output is different (the header may be different), then the Blu-Ray Demuxer Pro is wrong because the eac3to output is exact (not better, is exact).

Quote:

And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?

The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

PowerGamer · 11th January 2010, 22:53

Quote:

Originally Posted by tebasuna51

The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

Taking that into account (thanks again for explaining) I found out the following (EAC - channel data in .wav file produced by eac3to, BDR - same in Blu-ray Demuxer .wav, PCM - corresponding data in .pcm file):

PCM EAC BDR
0100 0001 0001
0200 0002 0002
0000 0000 0000
ffff 0000 ffff
0000 ffff 0000
0000 0000 0000

ffff ffff ffff
ffff ffff ffff
0000 0000 0000
0100 0001 0001
0000 0001 0000
0100 0000 0001

0000 0000 0000
0000 0000 0000
0000 0000 0000
0000 fffe 0000
0000 0000 0000
feff 0000 fffe

0100 0001 0001
0000 0000 0000
0100 0001 0001
ffff 0003 ffff
0000 ffff 0000
0300 0000 0003

ffff ffff ffff
0100 0001 0001
feff fffe fffe
0000 fffc 0000
0000 0000 0000
fcff 0000 fffc

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).

tebasuna51 · 12th January 2010, 00:52

Quote:

Originally Posted by PowerGamer

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).

Like I say before the audio data in .pcm have different channel order than in wav container:

pcm: FL, FR, FC, BL, BR, LF
wav: FL, FR, FC, LF, BL, BR
then the LF channel in PCM (6th) must go to 4th position, BL (4th) to 5th and BR (5th) to 6th.

Seems BDR don't make the channel arrangement.
Is easy to verify the problem, if you play the BDR.wav with a multichannel capable player must listen the Low Frequency channel in the Back Right speaker.

Stephen R. Savage · 12th January 2010, 04:32

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2

ut.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

TinTime · 12th January 2010, 07:41

@PowerGamer

Specifically the channels must be interleaved in this order for wav pcm.

tebasuna51 · 12th January 2010, 10:11

Quote:

Originally Posted by Stephen R. Savage

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2

ut.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

Something like that:

Code:

v2.84
...
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"

umaximus · 12th January 2010, 11:21

I got loads of lossless check failed errors on Animatrix's TrueHD track while extracting/transcoding. Are this safe to overlook? How much of ms is failed byte?

[HTML]eac3to v3.17
command line: eac3to 1) 2: video.vc1 4: audio.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 1:40:50, 24p /1.001
1: Chapters, 10 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, Japanese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
9: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
10: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Dutch
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Portuguese
18: Subtitle (PGS), Japanese
19: Subtitle (PGS), English
20: Subtitle (PGS), Japanese
21: Subtitle (PGS), French
22: Subtitle (PGS), Italian
23: Subtitle (PGS), Italian
24: Subtitle (PGS), Dutch
25: Subtitle (PGS), Spanish
26: Subtitle (PGS), Portuguese
27: Subtitle (PGS), Japanese
28: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Creating file "video.vc1"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:53:49. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:06:51. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:33:01. <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 43ms at playtime 0:18:49. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:28:12. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:45:06. <WARNING>
[a04] Audio overlaps for 42ms at playtime 0:53:50. <WARNING>
[a04] Audio overlaps for 42ms at playtime 1:06:54. <WARNING>
[a04] Audio overlaps for 43ms at playtime 1:33:01. <WARNING>
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Reducing depth from 24 to 16 bits...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[a04] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 145067 frames.
eac3to processing took 3 hours, 23 minutes.
Done.
[/HTML]

AnryV · 12th January 2010, 14:11

I think that eac3to (or ArcSoft decoder ??) uncorrectly decode 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr scheme.

Quote:

eac3to v3.17
command line: eac3to Ls_for_eac3to.dtshd Ls.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
The original audio track has a constant bit depth of 24 bits.
Creating file "Ls.L.wav"...
Creating file "Ls.R.wav"...
Creating file "Ls.C.wav"...
Creating file "Ls.LFE.wav"...
Creating file "Ls.BL.wav"...
Creating file "Ls.BR.wav"...
Creating file "Ls.SL.wav"...
Creating file "Ls.SR.wav"...
eac3to processing took 4 seconds.
Done.

1. Ls.dtshd - is the file encoded by DTS-HD Encoder Suite from wavs from <Original>. Scheme - 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr
2. Ls_for_eac3to.dtshd - the same file but without header which don't recognized by eac3to
3. <Original> dir - contains original 8 wavs (1 second duration)
4. <eac3to> dir - contains 8 wavs decoded from Ls_for_eac3to.dtshd by eac3to
5. <StreamPlayer> dir - contains 8 wavs decoded from Ls.dtshd by DTS-HD StreamPlayer

As result - <StreamPlayer> fully identical <Original> but BL and BR files from <eac3to> contains not Ls and Rs from <Original> but mixed Ls+Lsr and Rs+Rsr

Sample
http://multi-up.com/201196

Thunderbolt8 · 12th January 2010, 16:44

look at the link of post #9676

AnryV · 12th January 2010, 17:38

Quote:

Originally Posted by Thunderbolt8

look at the link of post #9676

Problem, as I do understand, was not solved?

umaximus · 12th January 2010, 21:38

I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?

Thunderbolt8 · 13th January 2010, 02:40

Quote:

Originally Posted by AnryV

Problem, as I do understand, was not solved?

no, apparently there havent been enough discs yet with this problem to justify the work

bassgoonist · 13th January 2010, 20:35

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

Tiziano · 21st January 2010, 13:41

Sorry, perhaps I'm very dum but I need a help. Sorry if I'm disturbing you continuously

I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.

I'm wondered if anybody can help me giving the correct settings or command line in order to get this .WAV correctly converted ...

I thank you very much

utenteanonimo64 · 21st January 2010, 15:43

Quote:

Originally Posted by bassgoonist

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

Why don't you use DVD-A Explorer? You can extract the multichannel data or downmix to 2.0. Once you have the WAV files then eac3to will do whatever you want with them...

tebasuna51 · 21st January 2010, 15:59

Quote:

Originally Posted by Tiziano

...
I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.
...

Aften, the ac3 encoder included with eac3to, isn't a Dolby Digital Plus encoder, you can obtain only 5.1 standard ac3.

tebasuna51 · 21st January 2010, 16:13

Quote:

Originally Posted by bassgoonist

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

To convert a wav file 5.0 you can use WavToAc3Enc or eac3to with external Aften.exe encoder:

eac3to source50 stdout.wav | Aften -pad 0 -readtoeof 1 -exps 32 -s 1 -b 640 - output.ac3

where:
-exps 32 -s 1 -b 640
is the best quality (and slow encode) than Aften can offer.

deathlord · 21st January 2010, 22:32

Hi

Can I have eac3to use the source filename for the destination?
I mean in the command line, like eac3to source.wav dest.wav

twazerty · 22nd January 2010, 19:52

Quote:

Originally Posted by Blue_MiSfit

@madshi:

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?

Noticed that the clipping problem isn't fixed yet. Can we expect a fix?

blackcell · 25th January 2010, 04:55

For some unknown reason, Eac3to is unable to extract the BD disc "The Express". It gets nearly finished then aborts.

Log file:

Code:

eac3to v3.17
command line: "C:\Tools\RipBot\Tools\eac3to\eac3to.exe" "D:\BDMV\STREAM" 1) 1: "C:\tmp\The Express - Chapters.txt" 2: "C:\tmp\The Express 1080p VC-1.mkv" 3: "C:\tmp\The Express - 3 DTS Master.flac" 6: "C:\tmp\The Express - 6 English Subtitle.sup" -log="C:\tmp\The Express - Log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:09:42, 24p /1.001
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
Creating file "C:\tmp\The Express - Chapters.txt"...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[s06] Extracting subtitle track number 6...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\tmp\The Express - 3 DTS Master.flac"...
[s06] Creating file "C:\tmp\The Express - 6 English Subtitle.sup"...
Reading the source file failed.  <ERROR>
Aborted at file position 24368906240.  <ERROR>

Anyone else seen this issue with other BD's? Any workarounds?

12th January 2010, 04:32	#9684 \| Link
Stephen R. Savage Registered User Join Date: Nov 2009 Posts: 327	Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2ut.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

12th January 2010, 11:21	#9687 \| Link
umaximus Registered User Join Date: Jan 2008 Posts: 34	Animatrix TrueHD I got loads of lossless check failed errors on Animatrix's TrueHD track while extracting/transcoding. Are this safe to overlook? How much of ms is failed byte? [HTML]eac3to v3.17 command line: eac3to 1) 2: video.vc1 4: audio.wavs ------------------------------------------------------------------------------ M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 1:40:50, 24p /1.001 1: Chapters, 10 chapters 2: VC-1, 1080p24 /1.001 (16:9) 3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB) 5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 6: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 7: AC3, Japanese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 8: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB) 9: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 10: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB 11: Subtitle (PGS), English 12: Subtitle (PGS), French 13: Subtitle (PGS), Italian 14: Subtitle (PGS), Italian 15: Subtitle (PGS), Dutch 16: Subtitle (PGS), Spanish 17: Subtitle (PGS), Portuguese 18: Subtitle (PGS), Japanese 19: Subtitle (PGS), English 20: Subtitle (PGS), Japanese 21: Subtitle (PGS), French 22: Subtitle (PGS), Italian 23: Subtitle (PGS), Italian 24: Subtitle (PGS), Dutch 25: Subtitle (PGS), Spanish 26: Subtitle (PGS), Portuguese 27: Subtitle (PGS), Japanese 28: Subtitle (PGS), English [v02] Extracting video track number 2... [a04] Extracting audio track number 4... [a04] Extracting TrueHD stream... [a04] Removing TrueHD dialog normalization... [a04] Decoding with libav/ffmpeg... [a04] Writing WAVs... [v02] Creating file "video.vc1"... [a04] Creating file "audio.R.wav"... [a04] Creating file "audio.L.wav"... [a04] Creating file "audio.LFE.wav"... [a04] Creating file "audio.SR.wav"... [a04] Creating file "audio.C.wav"... [a04] Creating file "audio.SL.wav"... [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING> [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING> [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING> [v02] Video overlaps for 7 frames at playtime 0:53:49. <WARNING> [v02] Video overlaps for 7 frames at playtime 1:06:51. <WARNING> [v02] Video overlaps for 7 frames at playtime 1:33:01. <WARNING> [a04] The original audio track has a constant bit depth of 16 bits. [a04] Audio overlaps for 43ms at playtime 0:18:49. <WARNING> [a04] Audio overlaps for 43ms at playtime 0:28:12. <WARNING> [a04] Audio overlaps for 43ms at playtime 0:45:06. <WARNING> [a04] Audio overlaps for 42ms at playtime 0:53:50. <WARNING> [a04] Audio overlaps for 42ms at playtime 1:06:54. <WARNING> [a04] Audio overlaps for 43ms at playtime 1:33:01. <WARNING> [a04] Superfluous zero bytes detected, will be stripped in 2nd pass. [a04] Starting 2nd pass... [a04] Extracting audio track number 4... [a04] Extracting TrueHD stream... [a04] Removing TrueHD dialog normalization... [a04] Decoding with libav/ffmpeg... [a04] Reducing depth from 24 to 16 bits... [a04] Writing WAVs... [a04] Realizing RAW/PCM gaps... [a04] Creating file "audio.R.wav"... [a04] Creating file "audio.L.wav"... [a04] Creating file "audio.SR.wav"... [a04] Creating file "audio.SL.wav"... [a04] Creating file "audio.C.wav"... [a04] Creating file "audio.LFE.wav"... [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING> [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING> [a04] [libav] End of stream indicated <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING> [a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING> [a04] The processed audio track has a constant bit depth of 16 bits. Video track 2 contains 145067 frames. eac3to processing took 3 hours, 23 minutes. Done. [/HTML]

13th January 2010, 20:35	#9693 \| Link
bassgoonist Registered User Join Date: Sep 2004 Posts: 73	5 channel audio I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

21st January 2010, 22:32	#9698 \| Link
deathlord Registered User Join Date: Mar 2008 Posts: 101	Hi Can I have eac3to use the source filename for the destination? I mean in the command line, like eac3to source.wav dest.wav Last edited by deathlord; 22nd January 2010 at 12:27.

12th January 2010, 07:41	#9685 \| Link
TinTime Registered User Join Date: Jan 2009 Location: UK Posts: 403	@PowerGamer Specifically the channels must be interleaved in this order for wav pcm.

12th January 2010, 16:44	#9689 \| Link
Thunderbolt8 Registered User Join Date: Sep 2006 Posts: 2,197	look at the link of post #9676

12th January 2010, 21:38	#9691 \| Link
umaximus Registered User Join Date: Jan 2008 Posts: 34	I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?

21st January 2010, 13:41	#9694 \| Link
Tiziano Registered User Join Date: Jan 2010 Posts: 15	Sorry, perhaps I'm very dum but I need a help. Sorry if I'm disturbing you continuously I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it. I'm wondered if anybody can help me giving the correct settings or command line in order to get this .WAV correctly converted ... I thank you very much