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  1. It says here you can add audio & video separately

    http://dcpomatic.com/manual/html/ch05.html

    But it doesn't say what types are allowed. For example, maybe it has problems with 24bit little endian. Maybe it needs unsigned vs signed etc..



    Originally Posted by El Heggunte View Post
    Proper .WAV files cannot be bigger than 4GB, because the header field that indicates the stream size is a 32-bit integer value

    (yes, Micro$oft is stupid, as usual).

    eac3to can output to .W64 or .RF64 though.


    Yes, but in a MOV container, it makes everything ok... (MOV being anti-MS and all)
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    Just inspected the WAV's and the 6Ch.wav is 9mins long and 4.44gb and the 8ch.wav is 29mins. Can you tell me how to amend the use of different audio codec?
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  3. How did you "inspect" it? For example , what does mediainfo say about it ?

    How about importing 8 individual mono wav files ? Can you import that into the program? ie Just stream copy them from the original mov ? Or will it have problems with that as well ?

    Or what codec / settings / configuration would you like ?
    Last edited by poisondeathray; 15th Aug 2014 at 22:59.
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  4. It's going to be converted back to uncompressed 24bit PCM WAV when it gets packaged in the MXF for DCP, so IMO you might as well leave it like that. You don't want to use some lossy format if you dont' have to

    This will demux (stream copy) them all into mono wavs

    Code:
    ffmpeg -i "input.mov" -map 0:1 -vn -c:a copy FL.wav -map 0:2 -vn -c:a copy FR.wav -map 0:3 -vn -c:a copy FC.wav -map 0:4 -vn -c:a copy LFE.wav -map 0:5 -vn -c:a copy BL.wav -map 0:6 -vn -c:a copy BR.wav -map 0:7 -vn -c:a copy SL.wav -map 0:8 -vn -c:a copy SR.wav

    That is essentially the same thing as this, but the 1st one should be faster, instead of having to sequentially call ffmpeg 8 times

    Code:
    ffmpeg -i "input.mov" -map 0:1 -vn -c:a copy FL.wav
    ffmpeg -i "input.mov" -map 0:2 -vn -c:a copy FR.wav
    ffmpeg -i "input.mov" -map 0:3 -vn -c:a copy FC.wav
    ffmpeg -i "input.mov" -map 0:4 -vn -c:a copy LFE.wav
    ffmpeg -i "input.mov" -map 0:5 -vn -c:a copy BL.wav
    ffmpeg -i "input.mov" -map 0:6 -vn -c:a copy BR.wav
    ffmpeg -i "input.mov" -map 0:7 -vn -c:a copy SL.wav
    ffmpeg -i "input.mov" -map 0:8 -vn -c:a copy SR.wav


    If there is a problem with the mono wavs at that stage - then there is a problem with ffmpeg reading your MOV files, or a problem with your original file. Quicktime pro can demux the audio (but not free). Those mono wavs are 759MiB each according to mediainfo , so well below the 4GB mark should that even be a theoretical problem
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    Thanks. had to get some sleep, my last post was at 4am UK time!

    I'll have sanother look at DCP O Matic to see if it can take seperate video & audio tracks, but I think they need to be muxed together. It can batch convert, but will treat them as seperate DCP's.

    When I said I "inspected" the wavs, I just meant I played them and looked at the running time.

    Here are the MediaInfo reports:

    8ch.wav:

    General
    Complete name : M:\eac3to327\8ch.wav
    Format : Wave
    File size : 5.93 GiB
    Duration : 1h 32mn
    Overall bit rate mode : Constant
    Overall bit rate : 9 216 Kbps
    Writing application : Lavf56.1.100

    Audio
    Format : PCM
    Format settings, Endianness : Little
    Format settings, Sign : Signed
    Codec ID : 00001000-0000-0100-8000-00AA00389B71
    Duration : 1h 32mn
    Bit rate mode : Constant
    Bit rate : 9 216 Kbps
    Channel(s) : 8 channels
    Sampling rate : 48.0 KHz
    Bit depth : 24 bits
    Stream size : 5.93 GiB (100%)

    6ch.wav:
    General
    Complete name : M:\eac3to327\6ch.wav
    Format : Wave
    File size : 4.45 GiB
    Duration : 1h 32mn
    Overall bit rate mode : Constant
    Overall bit rate : 6 912 Kbps

    Audio
    Format : PCM
    Format settings, Endianness : Little
    Format settings, Sign : Signed
    Codec ID : 00001000-0000-0100-8000-00AA00389B71
    Duration : 1h 32mn
    Bit rate mode : Constant
    Bit rate : 6 912 Kbps
    Channel(s) : 6 channels
    Channel positions : Front: L C R, Side: L R, LFE
    Sampling rate : 48.0 KHz
    Bit depth : 24 bits
    Stream size : 4.45 GiB (100%)
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    Originally Posted by poisondeathray View Post
    It's going to be converted back to uncompressed 24bit PCM WAV when it gets packaged in the MXF for DCP, so IMO you might as well leave it like that. You don't want to use some lossy format if you dont' have to

    This will demux (stream copy) them all into mono wavs

    Code:
    ffmpeg -i "input.mov" -map 0:1 -vn -c:a copy FL.wav -map 0:2 -vn -c:a copy FR.wav -map 0:3 -vn -c:a copy FC.wav -map 0:4 -vn -c:a copy LFE.wav -map 0:5 -vn -c:a copy BL.wav -map 0:6 -vn -c:a copy BR.wav -map 0:7 -vn -c:a copy SL.wav -map 0:8 -vn -c:a copy SR.wav
    Thabnks for your continued help. I did the above and now have the eight separate audio tracks. I've put them into DCP O MATIC along with the original video, but unfortunately it still sees them as individual files, so it will try and make nine DCPs, one with video and eight sound only.

    So I need some way to extract the video and then mux it together with the eight sound tracks. I'm sure this is possible, but could do with some guidance, as every step takes so long, and the actual DCP bit took 17hrs last time, and I need this ready first thing on Monday!
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  7. So what mediainfo says above doesn't correlate with what you see for those 2 wavs ? Filesizes aren't 5.93Gib and 4.45Gib ? That suggests something aborted the process

    Were there any other errors in the log file when processing ?

    What does mediainfo say about the final MOV ?

    How are you "playing" these ?



    The problem is , all freeware will be using the same routines / libraries as ffmpeg (libav) . In fact DCP O MATIC uses ffmpeg as well.
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    At the bottom of the page http://en.wikipedia.org/wiki/Digital_Cinema_Package ,
    there is a list which includes several freeware DCP applications, so maybe you should try them too,
    IF dcp-o-matic is not working as expected

    Regarding the JPEG2000 compression especifically ---- SFAIK,
    the open-source implementations of the format are rather incomplete
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    Thanks, just had to pop out for a couple of hours.

    This sizes of the wavs are as in the log. I played them with vlc and Windows Media Player, and they behaved as if they were 9 and 29 min long files.

    I didn't notice any other errors.

    Heres the MedfiaInfo, has has the audio as lasting 9mins.

    General
    Complete name : H:\output.mov
    Format : MPEG-4
    Format profile : QuickTime
    Codec ID : qt
    File size : 109 GiB
    Duration : 1h 32mn
    Overall bit rate mode : Variable
    Overall bit rate : 170 Mbps
    Writing application : Lavf56.1.100

    Video
    ID : 1
    Format : ProRes
    Format version : Version 0
    Format profile : High
    Codec ID : apch
    Duration : 1h 32mn
    Bit rate mode : Variable
    Bit rate : 169 Mbps
    Width : 1 920 pixels
    Height : 1 080 pixels
    Display aspect ratio : 16:9
    Frame rate mode : Constant
    Frame rate : 23.976 fps
    Color space : YUV
    Chroma subsampling : 4:2:2
    Scan type : Progressive
    Bits/(Pixel*Frame) : 3.396
    Stream size : 109 GiB (100%)
    Writing library : Apple
    Language : English
    Color primaries : BT.709
    Transfer characteristics : BT.709
    Matrix coefficients : BT.709

    Audio
    ID : 2
    Format : PCM
    Format settings, Endianness : Little
    Format settings, Sign : Unsigned
    Codec ID : in24
    Duration : 9mn 16s
    Bit rate mode : Constant
    Bit rate : 6 912 Kbps
    Channel(s) : 6 channels
    Channel positions : Front: L C R, Side: L R, LFE
    Sampling rate : 48.0 KHz
    Bit depth : 24 bits
    Stream size : 459 MiB (0%)
    Language : English

    Other
    ID : 3
    Type : Time code
    Format : QuickTime TC
    Duration : 1h 32mn
    Time code of first frame : 00:00:00:00
    Time code settings : Striped
    Language : English

    I could try this again, but might get the same result, or if theres a way to mux them all together that might be better at this stage.
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    Originally Posted by El Heggunte View Post
    At the bottom of the page http://en.wikipedia.org/wiki/Digital_Cinema_Package ,
    there is a list which includes several freeware DCP applications, so maybe you should try them too,
    IF dcp-o-matic is not working as expected

    Regarding the JPEG2000 compression especifically ---- SFAIK,
    the open-source implementations of the format are rather incomplete
    I've tried a few different solutions in the past and DCP O Matic is the only one that has produced satisfactory results. I'll have another look now though.
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  11. That 1st step WAS muxing them all together (at least it was supposed to be). It's combining 8 mono to a single 8ch WAV.

    If the demuxed mono wavs are ok (in filesize, don't abort when playing), You can try wavewizard to stitch them together
    http://forum.doom9.org/showthread.php?t=95265

    There is a link to a newer version at the end of the thread. It has channel mapping matrix as well. In the preferences you can set it to "merge" files
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    Originally Posted by poisondeathray View Post
    That 1st step WAS muxing them all together (at least it was supposed to be). It's combining 8 mono to a single 8ch WAV.

    If the demuxed mono wavs are ok (in filesize, don't abort when playing), You can try wavewizard to stitch them together
    http://forum.doom9.org/showthread.php?t=95265

    There is a link to a newer version at the end of the thread. It has channel mapping matrix as well. In the preferences you can set it to "merge" files
    Can you give me some guidance as to how to use this? All I've succeeded in doing so far is produce 8 more Wav files.
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  13. In the preferences, make sure stream manipulation is set to "merge files" instead of "mono streams" , push apply
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    Ok, done that. It appears to create a 1k file, then batch manager opens. I managed to work out that I need to point it towards Wavepack.exe, but it only seems to momentarily create a small file, then it disappears.

    Oh, and the de muxed mono files all appear to be exactly the same size?
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  15. Demuxed mono wavs are supposed to be exactly the same size (they are uncompressed wave)

    When merged together, they should be the sum of all 8 (or very close, might be a few bytes difference from header) . Or if you downmixed using the channel matrix to 6ch it should be exactly 3/4 of the sum

    I don' t know, it works here. Do you have start batcher when finished checkmarked ? It should automatically do the conversion
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    Ok, sorry to be pedantic, but I need step by step instructions, there are so many options here and i'm just blindly faffing with it while also trying to suss out another DCP software, attempting to convert it to another more familiar file format, burn a DVD, find out if it will play on an xbox (will it?), find a downloadable version, go out and re film the thing myself be kind rewind stlyee or just bash my head against a brick wall!
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  17. It sounds like it's not going to work for you, for whatever reason

    Add, the files, push convert. That's it.

    The only difference is in the preferences settings as described above - you have to change it to merge instead of mono, and don't forget to push apply . That gives me a merged 8ch wav in filesize, that is the sum of the 8 mono channels. If you want to enable channel mapping you can, but if it fails even on that simple conversion, then it's probably not going to work when you add in other complexities




    The ffmpeg merge worked ok here too. It doesn't make sense that it shouldn't for you, since the extracted mono wavs are good (aren't they ) ? If ffmpeg was able to extract it, it should be able to merge them them to an 8ch WAV you would think.

    You can try merging with ffmpeg from the mono wavs, instead of directly from the MOV, but I doubt it will make a difference. I really don't know why none of these things are working for you
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    Thanks for your patience. I'm just giving this another go now.

    I think I need to have a break after that, I'm tired and frustrated. But I'd like to set something running while I do. To compound everything, my PC seems have developed as problem in the last 24 hrs (before any of the actions in this thread) so I will give it a go on my laptop instead (I've been doing the Wavewizard on here as well). Which step would you consider the best to try again, post 8?
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    Originally Posted by poisondeathray View Post
    It sounds like it's not going to work for you, for whatever reason

    Add, the files, push convert. That's it.

    The only difference is in the preferences settings as described above - you have to change it to merge instead of mono, and don't forget to push apply . That gives me a merged 8ch wav in filesize, that is the sum of the 8 mono channels. If you want to enable channel mapping you can, but if it fails even on that simple conversion, then it's probably not going to work when you add in other complexities
    I got a slightly different result this time, a 4mb file. Are you saying that the batch process shouldn't come into this at all? It comes up after I press convert, but doesn't seem to do anything. It might just be a quirk of this software, but "apply" doesn't grey out after its been pressed, so I'm not 100% sure it worked, the merge box is still ticked when I reopen it though.
    Last edited by mrbliss; 16th Aug 2014 at 16:11.
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  20. It's not actually run through conversion batcher when merging. If you open up the conversion batcher, you confirm this by can disable everything, all the encoders , wavpack, flac, mp3 etc.... so nothing is enabled. Mine is all blank, no paths are set. That batcher is used when you want to convert with other tools to different formats and you want to queue up a list of jobs. If you uncheckmark send jobs to batcher, uncheckmark start batcher when finished, only wave wizard is being used to merge the file
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  21. If you have the mono wavs, and you're sure they are good, you can try merging them instead of reading from the MOV. It shouldn't make any difference, but at this point it's worth a try

    Code:
    ffmpeg -i FL.wav -i FR.wav -i FC.wav -i LFE.wav -i BL.wav -i BR.wav -i SL.wav -i SR.wav -vn -filter_complex "[0:0][1:0][2:0][3:0][4:0][5:0][6:0][7:0]amerge=inputs=8[aout]" -map "[aout]" -c:a pcm_s24le 8ch_new.wav
    If the mono wavs were 759MiB, then 8x759MiB should be = 6072MiB (or very close to)


    The last thing I can think of in terms of freeware you can try is audacity. It's an audio editor. But it's based on ffmpeg as well. So if ffmpeg fails, chances are it will fail as well
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    Originally Posted by poisondeathray View Post
    If you have the mono wavs, and you're sure they are good, you can try merging them instead of reading from the MOV. It shouldn't make any difference, but at this point it's worth a try

    Code:
    ffmpeg -i FL.wav -i FR.wav -i FC.wav -i LFE.wav -i BL.wav -i BR.wav -i SL.wav -i SR.wav -vn -filter_complex "[0:0][1:0][2:0][3:0][4:0][5:0][6:0][7:0]amerge=inputs=8[aout]" -map "[aout]" -c:a pcm_s24le 8ch_new.wav
    If the mono wavs were 759MiB, then 8x759MiB should be = 6072MiB (or very close to)


    The last thing I can think of in terms of freeware you can try is audacity. It's an audio editor. But it's based on ffmpeg as well. So if ffmpeg fails, chances are it will fail as well
    Remind me, do i put the FFMpeg.exe in the same dir as the wavs then run it?
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  23. Yes, that's the easiest way. But you can specify the exact path e.g. "c:\path\ffmpeg.exe" , or setup an environment variable
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    Originally Posted by poisondeathray View Post
    Yes, that's the easiest way. But you can specify the exact path e.g. "c:\path\ffmpeg.exe" , or setup an environment variable
    I got it to work by putting the wavs in the bin folder, running ffprompt and entering the code. Will Report back in an hour or whenever its finished.
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    Results similar to before, file nearly the size you predicted, but only seems to have 30mins of audio. MediaInfo thinks its 1hr 32 though:

    General
    Complete name : F:\PKG - KNOCKED_FOR_SIX_FEATURE_ONLY\ffmpeg-20140816-git-65f05ef-win64-static\bin\8ch_new.wav
    Format : Wave
    File size : 5.93 GiB
    Duration : 1h 32mn
    Overall bit rate mode : Constant
    Overall bit rate : 9 216 Kbps
    Writing application : Lavf56.1.100

    Audio
    Format : PCM
    Format settings, Endianness : Little
    Format settings, Sign : Signed
    Codec ID : 00001000-0000-0100-8000-00AA00389B71
    Duration : 1h 32mn
    Bit rate mode : Constant
    Bit rate : 9 216 Kbps
    Channel(s) : 8 channels
    Channel positions : Front: L C R, Side: L R, Back: L R, LFE
    Sampling rate : 48.0 KHz
    Bit depth : 24 bits
    Stream size : 5.93 GiB (100%)

    Here's the FFMpeg incase that sheds any light on it:

    >ffmpeg -i FL.wav -i FR.wav -i FC.wav -i LFE.wav -i BL.wav -i BR.wav -i SL.wav -
    i SR.wav -vn -filter_complex "[0:0][1:0][2:0][3:0][4:0][5:0][6:0][7:0]amerge=inp
    uts=8[aout]" -map "[aout]" -c:a pcm_s24le 8ch_new.wav
    ffmpeg version N-65672-g65f05ef Copyright (c) 2000-2014 the FFmpeg developers
    built on Aug 15 2014 22:08:53 with gcc 4.8.3 (GCC)
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
    isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
    le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
    enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib
    modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw
    b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge
    r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en
    able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
    --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-
    libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
    libavutil 54. 3.100 / 54. 3.100
    libavcodec 56. 0.101 / 56. 0.101
    libavformat 56. 1.100 / 56. 1.100
    libavdevice 56. 0.100 / 56. 0.100
    libavfilter 5. 0.100 / 5. 0.100
    libswscale 3. 0.100 / 3. 0.100
    libswresample 1. 0.100 / 1. 0.100
    libpostproc 53. 0.100 / 53. 0.100
    Input #0, wav, from 'FL.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels
    (FL), s32, 1152 kb/s
    Input #1, wav, from 'FR.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #1:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels
    (FR), s32, 1152 kb/s
    Input #2, wav, from 'FC.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #2:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32,
    1152 kb/s
    Input #3, wav, from 'LFE.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #3:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels
    (LFE), s32, 1152 kb/s
    Input #4, wav, from 'BL.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #4:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels
    (BL), s32, 1152 kb/s
    Input #5, wav, from 'BR.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #5:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels
    (BR), s32, 1152 kb/s
    Guessed Channel Layout for Input Stream #6.0 : mono
    Input #6, wav, from 'SL.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #6:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels,
    s32, 1152 kb/s
    Guessed Channel Layout for Input Stream #7.0 : mono
    Input #7, wav, from 'SR.wav':
    Metadata:
    encoder : Lavf56.1.100
    Duration: 01:32:07.69, bitrate: 1152 kb/s
    Stream #7:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 1 channels,
    s32, 1152 kb/s
    [Parsed_amerge_0 @ 0000000002b0ae60] No channel layout for input 1
    [Parsed_amerge_0 @ 0000000002b0ae60] Input channel layouts overlap: output layou
    t will be determined by the number of distinct input channels
    Output #0, wav, to '8ch_new.wav':
    Metadata:
    ISFT : Lavf56.1.100
    Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32, 9
    216 kb/s (default)
    Metadata:
    encoder : Lavc56.0.101 pcm_s24le
    Stream mapping:
    Stream #0:0 (pcm_s24le) -> amerge:in0
    Stream #1:0 (pcm_s24le) -> amerge:in1
    Stream #2:0 (pcm_s24le) -> amerge:in2
    Stream #3:0 (pcm_s24le) -> amerge:in3
    Stream #4:0 (pcm_s24le) -> amerge:in4
    Stream #5:0 (pcm_s24le) -> amerge:in5
    Stream #6:0 (pcm_s24le) -> amerge:in6
    Stream #7:0 (pcm_s24le) -> amerge:in7
    amerge -> Stream #0:0 (pcm_s24le)
    Press [q] to stop, [?] for help
    size= 6218650kB time=01:32:07.68 bitrate=9216.0kbits/s
    video:0kB audio:6218650kB subtitle:0kB other streams:0kB global headers:0kB muxi
    ng overhead: 0.000002%
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    I've had a play with Audacity and have successfully made a 1hr 32min stereo WAV, I think that will do the trick if DCP O Matic knows how to handle stereo. How do I go about removing the video from the original and replacing it with this?
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  27. You can set audacity to output 5.1 or 7.1 , custom channel mapping as well





    But to copy the video, and mux in the new audio:

    Where "original.mov" is the original video, "audacity.wav" is the audio export from audacity. As usual , you can specify paths for each

    Code:
    ffmpeg -i "original.mov" -i "audacity.wav" -map 0:0 -map 1:0 -c:v copy -c:a copy "newoutput.mov"
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  28. I bet you're probably going down to stereo 16 bit . At least make sure the project is set to 48000, I think it defaults to 44100. If you can't get it working with the better audio , I guess that stereo, 16 bit should be plan B unless you've already run out of time


    In audacity:

    edit => preferences => import/export => use custom mix (for example to export a 5.1 multichannel file)

    under file=>export if you select other uncompressed files, you have a bunch of other options, if you push options and from the drop down menus. You can try different types of wav, e.g. w64

    Or you might try something like flac instead if it accepts flac (lossless compression)
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  29. Member
    Join Date
    Oct 2006
    Location
    United Kingdom
    Search Comp PM
    I'll have another look at the audacity options, may as well try and get it perfect if we've come this far.

    I'm using VLC at the moment to save the video stream only. I'll look at the FFMpeg option afterwards. Its going to be another very late night for me!
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  30. Member
    Join Date
    Oct 2006
    Location
    United Kingdom
    Search Comp PM
    Is there any way to change the ordering in Audacity? I think this can be done in DCP O Matic, but would be good to get that correct at this stage.

    EDIT
    Don't worry I've found out how to do it.
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