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buddhablack (Customer) asked a question.

November 4, 2024 at 5:27 PM
Dolby E-AC3 7.1 and audio channels order
Hi all!
I’ve just droped an eac3 7.1 file on last Audacity version.

These info are from MediaInfo:

Audio
Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Format profile : Blu-ray Disc
Duration : 2 h 31 min
Bit rate mode : Constant
Bit rate : 896 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 : 973 MiB (100%)
Service kind : Complete Main
Dialog Normalization : -28 dB
compr : -0.28 dB
cmixlev : -3.0 dB
surmixlev : -6 dB
dmixmod : Lo/Ro
ltrtcmixlev : -3.0 dB
ltrtsurmixlev : -4.5 dB
lorocmixlev : -3.0 dB
lorosurmixlev : -4.5 dB
dialnorm_Average : -28 dB
dialnorm_Minimum : -28 dB
dialnorm_Maximum : -28 dB

Now I see correctly 8 clips on Audacity.
Which channels are they? I mean, track 1 → L, track 2 → R, ecc

Thanks
  • 11 answers

  • BennettS (Dolby Labs)

    Hi,

    The answer is in the media info text you copied: "Channel layout : L R C LFE Ls Rs Lb Rb". Lb and Rb = Lrs Rrs.

    It is not recommended to edit or re-encode already encoded files, as the encoding process involves compression and anything else you do to the audio could result in double processing a reduction in quality. EAC3 was designed to be decoded by a playback device, not imported into a DAW.

    -B
    Expand Post
    Selected as Best

  • BennettS (Dolby Labs)

    Hi,

    The answer is in the media info text you copied: "Channel layout : L R C LFE Ls Rs Lb Rb". Lb and Rb = Lrs Rrs.

    It is not recommended to edit or re-encode already encoded files, as the encoding process involves compression and anything else you do to the audio could result in double processing a reduction in quality. EAC3 was designed to be decoded by a playback device, not imported into a DAW.

    -B
    Expand Post
    Selected as Best

      • BennettS (Dolby Labs)

        Dolby uses SMPTE channel order (L R C LFE Lss Rss Lrs Rrs, etc).

        What exactly are you trying to do with the file? As mentioned above, you should not use an encoded file as the source, as it is compressed and any re-encoding will double process the encoding and compression.

  • buddhablack (Customer)

    In short, take these 2 files:

    -     http://j7n.sytes.net/misc/71ch_id.ec3
    -     https://drive.google.com/drive/folders/1JxmeedtAtgmoafXv9rroiDOS2vEX7N4b --> "Dolby Digital Plus E-AC-3 7.1.mp4" (demuxed on Dolby Digital Plus E-AC-3 7.1.track_2_und.ec3)

    Open them with 2 Audacity instances.
    Why there are 2 channels (right-center) in different positions?

    If you open 71ch_id.ec3 with MediaInfo you can read "Format settings : Dolby Surround EX".
    "Dolby Digital Plus E-AC-3 7.1.track_2_und.ec3" instead doesn't have this thing.

    Are there 2 different "E-AC3 7.1" versions?

    Thanks!

    What the right way to identify each channel inside to a e-ac3 7.1 ?
    Expand Post
    • Compare

    • BennettS (Dolby Labs)

      Were these 2 files encoded using the same master file? Did you encode them? 7.1 core EAC3 and Dolby Surround EX are legacy codecs and indicate that these files are most likely old and/or were encoded using an old tool.

      Current EAC3 files will use a 5.1 core with SMPTE layout (L R C LFE Ls Rs) which will then be decoded as 7.1 if the playback system is 7.1.

      Channel order for Dolby codecs is SMPTE.

      Either these files were encoded from a master file that used the wrong channel order or audacity is improperly decoding the channel order on import.

      What is the end goal with these files?
      Expand Post

      • buddhablack (Customer)

        Thanks BennetS for your patience and suport.

        The goal of this process is take a "E-AC3 7.1" file and obtain 8 discrete wav files (each one with the correct file name of his channel)
        How can I do this?

        Can I use this FFMPEG command ?

        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"

        Or which tool do you recommend?
        Cheers,
        Expand Post

  • buddhablack (Customer)

    I did some test.

    The zip file contains:

    - test.eac3
    - inport_on_Audacity.jpg
    - DME_v3.7_encoding.jpg
    - inport_on_Audacity_after_DME_encoding.jpg

    1) import on Audacity "test.eac3" (see "inport_on_Audacity.jpg")

    2) export it to 8 discrete wav files

    3) import them on DME and encode a eb3 file (see "DME_v3.7_encoding.jpg")

    4) "wave converting" via ffmpeg
    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"

    5) import all these new files on Audacity (see "import_on_Audacity_after_DME_encoding.jpg")

    6) you can see that "inport_on_Audacity.jpg" and "nport_on_Audacity_after_DME_encoding.jpg" have the same order of channels

    So, to "wave converting" a "e-ac3 7.1" can I use this process? Is it safe about "channel order" ?

    Thanks,
    Expand Post

    • BennettS (Dolby Labs)

      Back to my earlier question - what is your end goal with this process? As mentioned in an earlier reply, importing an encoded file into Audacity is not a supported workflow, and the audio be compressed compressed twice (once in the initial encode and again in the re-encode).

      If you are trying to create test signals I would recommend starting with PCM source material and creating the file using SMPTE channel order from scratch.

      If you do not have access to the source assets that were used to create the encodes with different channel orders it is likely that the source material is using the wrong channel order.

      -B
      Expand Post

      • buddhablack (Customer)

        Hi BennettS,
        the main goal is to fix (cut/join some parts) some wedding videos with e-ac3 7.1.
        I need to know exactly how extract 8 channels from "e-ac3 7.1" to 8 wav files.
        I know it's possible with ffmpeg for example. Maybe with this?

        ffmpeg -hide_banner -i '.\input.eac3' `
        -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"

        Thanks!
        Expand Post
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