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#1 | Link |
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Registered User
Join Date: Nov 2023
Posts: 34
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How to decode E-AC3 7.1 to 8 wav files (with correct channel name files)
Hi all,
I've a e-ac3 7.1 file. What is the best/safe way to decode a E-AC3 7.1 file to 8 wavs files? I have:
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed http://www.nero.com/eng/store-blu-ray.html CAUTION: You need Nero 7. Nero 8 won't work with eac3to. ArcSoft DTS Decoder (1.1.0.9) works fine Sonic Audio Decoder (3.47.0.0) doesn't seem to be installed Haali Matroska Muxer doesn't seem to be installed http://haali.net/mkv Nero AAC Encoder could not be located http://www.nero.com/eng/nero-aac-codec.html Copy NeroAacEnc.exe to the eac3to or to the Windows folder. Surcode DTS Encoder doesn't seem to be installed http://www.surcode.com Here https://en.wikibooks.org/wiki/Eac3to/FAQ I read need to register the plugin: ... you can install Nero ShowTime only and of course to register the plug-in. How can I do ? How can I have 8 wav files from the E-AC3 7.1 ? Need also each file have the correct name of the channel. I need this for video editing later... Thanks! PS Or is better via ffmpeg? Code:
ffmpeg -i '.\input.eac3' ` -filter_complex "channelsplit=channel_layout=7.1[FL][FR][FC][LFE][LS][RS][LB][RB]" ` -map "[FL]" -c:a pcm_s24le "output_audio_L.wav" ` -map "[FR]" -c:a pcm_s24le "output_audio_R.wav" ` -map "[FC]" -c:a pcm_s24le "output_audio_C.wav" ` -map "[LFE]" -c:a pcm_s24le "output_audio_LFE.wav" ` -map "[LS]" -c:a pcm_s24le "output_audio_LSR.wav" ` -map "[RS]" -c:a pcm_s24le "output_audio_RSR.wav" ` -map "[LB]" -c:a pcm_s24le "output_audio_LSS.wav" ` -map "[RB]" -c:a pcm_s24le "output_audio_RSS.wav"
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AviSynth+ 3.7.3 Last edited by buddha9; 19th August 2025 at 14:00. |
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#2 | Link |
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Registered User
Join Date: Apr 2006
Posts: 185
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ffmpeg now works. I don't think there is need to use paid software for decoding anymore. Quality-wise the official decoder in Adobe Audition allows to boost extremely low signal without artifacts, while ffmpeg will impart a buzz. The level of very high frequencies seems slightly attenuated in Audition.
Beware that some decoders might swap the surround and rear-surround channel pairs. Here is a clip identifying each channel: http://j7n.sytes.net/misc/71ch_id.ec3 |
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#3 | Link | |
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Registered User
Join Date: Nov 2023
Posts: 34
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Quote:
I've Adobe Audition 24.0.3.3. It doesn't like the e-ac3 files. What version I have to use?
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AviSynth+ 3.7.3 |
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#4 | Link |
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Registered User
Join Date: Nov 2023
Posts: 34
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Anyway, I compared 2 e-ac3 files:
- your "http://j7n.sytes.net/misc/71ch_id.ec3" - the one you can find here https://drive.google.com/drive/folde...roiDOS2vEX7N4b --> "Dolby Digital Plus E-AC-3 7.1.mp4" (demuxed on Dolby Digital Plus E-AC-3 7.1.track_2_und.ec3) Comparing:
Code:
MEDIAINFO: Audio Format : E-AC-3 Format/Info : Enhanced AC-3 Commercial name : Dolby Digital Plus Format settings : Dolby Surround EX Duration : 16 s 0 ms Bit rate mode : Constant Bit rate : 448 kb/s Channel(s) : 8 channels Channel layout : L R C LFE Ls Rs Lb Rb Sampling rate : 48.0 kHz Frame rate : 31.250 FPS (1536 SPF) Compression mode : Lossy Stream size : 875 KiB (100%) Service kind : Complete Main Dialog Normalization : -19 dB compr : -0.28 dB dialnorm_Average : -19 dB dialnorm_Minimum : -19 dB dialnorm_Maximum : -19 dB Code:
PS C:\Users\***\Downloads\e-ac3 channels order\71ch_id_ec3> ffmpeg -hide_banner -i '.\71ch_id.ec3' [eac3 @ 000001e0da61b540] Estimating duration from bitrate, this may be inaccurate Input #0, eac3, from '.\71ch_id.ec3': Duration: 00:00:16.00, start: 0.000000, bitrate: 448 kb/s Stream #0:0: Audio: eac3, 48000 Hz, 7.1, fltp, 448 kb/s
Code:
MEDIAINFO: Audio Format : E-AC-3 Format/Info : Enhanced AC-3 Commercial name : Dolby Digital Plus Duration : 1 min 29 s Bit rate mode : Constant Bit rate : 384 kb/s Channel(s) : 8 channels Channel layout : L R C LFE Ls Rs Lb Rb Sampling rate : 48.0 kHz Frame rate : 31.250 FPS (1536 SPF) Compression mode : Lossy Stream size : 4.08 MiB (100%) Service kind : Complete Main Dialog Normalization : -16 dB compr : 0.53 dB mixlevel : 105 dB dialnorm_Average : -16 dB dialnorm_Minimum : -16 dB dialnorm_Maximum : -16 dB Code:
PS C:\Users\***\Downloads\e-ac3 channels order\e-ac3 from Drive> ffmpeg -hide_banner -i '.\Dolby Digital Plus E-AC-3 7.1.track_2_und.ec3' [eac3 @ 000001dd97ddd440] Estimating duration from bitrate, this may be inaccurate Input #0, eac3, from '.\Dolby Digital Plus E-AC-3 7.1.track_2_und.ec3': Duration: 00:01:29.12, start: 0.000000, bitrate: 384 kb/s Stream #0:0: Audio: eac3, 48000 Hz, 7.1, fltp, 384 kb/s ![]() You can note there are 2 inverted channels. Why ? ![]() Here https://professionalsupport.dolby.co...g000006XdWSAA0 Dolby says "Dolby uses SMPTE channel order (L R C LFE Lss Rss Lrs Rrs, etc). " Thanks!
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AviSynth+ 3.7.3 Last edited by buddha9; 19th August 2025 at 14:09. |
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#5 | Link |
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Registered User
Join Date: Apr 2006
Posts: 185
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This sample is from an official disc "Dolby Labs - Functional Testing". EX supposedly contains an extra back channel within the core 5.1 stream, but I have no means of listening to it. With the extension it decodes to full 7 channels + LFE. It confirms the channel IDs as they are. Not saying that you have them wrong, but that it may happen.
Internally the order is the same as Vorbis: Left, Center, Right, Side Left, Side Right, Rear Left, Rear Right, LFE. Tools should automatically convert it to the SMTPE order. |
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#6 | Link | |
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Registered User
Join Date: Nov 2023
Posts: 34
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Quote:
Format settings : Dolby Surround EX You can verify via MediaInfo. Thanks.
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AviSynth+ 3.7.3 |
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#7 | Link |
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Moderator
![]() Join Date: Feb 2005
Location: Spain
Posts: 7,369
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No mather the internally channel order when it is decoded the channels follow the wav order FL,FR,FC,LFE,LB,RB,LS,RS (Back/Rear channels before Side/Surrround channels) for that the order in the image.
I make this channel test |
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#8 | Link | |
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Join Date: Nov 2023
Posts: 34
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Quote:
![]() So 1) imported your file on Audacity ![]() 2) exported on 8 discrete wav files 3) imported them on DME (v3.7): ![]() 4) split the rendered file via ffmpeg: Code:
ffmpeg -hide_banner -i '.\output.eb3' ` -filter_complex "channelsplit=channel_layout=7.1[FL][FR][FC][LFE][LB][RB][LS][RS]" ` -map "[FL]" -c:a pcm_s24le "output_2_audio_L.wav" ` -map "[FR]" -c:a pcm_s24le "output_2_audio_R.wav" ` -map "[FC]" -c:a pcm_s24le "output_2_audio_C.wav" ` -map "[LFE]" -c:a pcm_s24le "output_2_audio_LFE.wav" ` -map "[LB]" -c:a pcm_s24le "output_2_audio_LS-Back.wav" ` -map "[RB]" -c:a pcm_s24le "output_2_audio_RS-Back.wav" ` -map "[LS]" -c:a pcm_s24le "output_2_audio_LS-Side.wav" ` -map "[RS]" -c:a pcm_s24le "output_2_audio_RS-Side.wav" ![]() ** Note: if you import the .eb3 file Audacity show you only 6 channels ** Conclusion: to "demux" a "e-ac3 7.1" file in the correct way you need to use the ffmpeg above. Right?
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AviSynth+ 3.7.3 |
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#10 | Link |
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Registered User
Join Date: Nov 2023
Posts: 34
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if you have a mkv (for example) which contains:
- 0:0 --> h.264/h.264 ... - 0:1 --> e-ac3 7.1 then this command avoid to extract e-ac3 stream: Code:
ffmpeg -hide_banner -i '.\input.mkv' ` >> -filter_complex "[0:1]channelsplit=channel_layout=7.1[FL][FR][FC][LFE][LB][RB][LS][RS]" ` >> -map "[FL]" -c:a pcm_s24le "output_audio_L.wav" ` >> -map "[FR]" -c:a pcm_s24le "output_audio_R.wav" ` >> -map "[FC]" -c:a pcm_s24le "output_audio_C.wav" ` >> -map "[LFE]" -c:a pcm_s24le "output_audio_LFE.wav" ` >> -map "[LB]" -c:a pcm_s24le "output_audio_LS-Back.wav" ` >> -map "[RB]" -c:a pcm_s24le "output_audio_RS-Back.wav" ` >> -map "[LS]" -c:a pcm_s24le "output_audio_LS-Side.wav" ` >> -map "[RS]" -c:a pcm_s24le "output_audio_RS-Side.wav"
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AviSynth+ 3.7.3 |
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