TAudioConverter can convert it to stereo right but fails at 5.1 channel. When converting to 5.1 channel finished it's only 19 minutes long.
mediainfo:Code:Audio Count : 258 Count of stream of this kind : 1 Kind of stream : Audio Kind of stream : Audio Stream identifier : 0 Inform : 384 Kbps, 48.0 KHz, 16 bits, 6 channels, AC-3 Format : AC-3 Format/Info : Audio Coding 3 Commercial name : AC-3 Mode extension : CM (complete main) Format settings, Endianness : Big Codec : AC3 Codec : AC3 Duration : 8635680 Duration : 2h 23mn Duration : 2h 23mn 55s 680ms Duration : 2h 23mn Duration : 02:23:55.680 Bit rate mode : CBR Bit rate mode : Constant Bit rate : 384000 Bit rate : 384 Kbps Channel(s) : 6 Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Channel positions : 3/2/0.1 ChannelLayout : L C R LFE Ls Rs Sampling rate : 48000 Sampling rate : 48.0 KHz Samples count : 414512640 Frame count : 269865 Resolution : 16 Resolution : 16 bits Bit depth : 16 Bit depth : 16 bits Compression mode : Lossy Compression mode : Lossy Stream size : 414512640 Stream size : 395 MiB (100%) Stream size : 395 MiB Stream size : 395 MiB Stream size : 395 MiB Stream size : 395.3 MiB Stream size : 395 MiB (100%) Proportion of this stream : 1.00000 bsid : 8 dialnorm : -31 dialnorm/String : -31 dB compr : -0.28 compr/String : -0.28 dB acmod : 7 lfeon : 1 dialnorm_Average : -31 dialnorm_Average/String : -31 dB dialnorm_Minimum : -31 dialnorm_Minimum/String : -31 dB dialnorm_Maximum : -31 dialnorm_Maximum/String : -31 dB dialnorm_Count : 32 compr_Average : 1.24 compr_Average/String : 1.24 dB compr_Minimum : -0.28 compr_Minimum/String : -0.28 dB compr_Maximum : 2.36 compr_Maximum/String : 2.36 dB compr_Count : 32 dynrng_Average : 0.59 dynrng_Average/String : 0.59 dB dynrng_Minimum : 0.00 dynrng_Minimum/String : 0.00 dB dynrng_Maximum : 1.02 dynrng_Maximum/String : 1.02 dB dynrng_Count : 32
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i forget to say i was trying to convert it to uncompressed WAV and then i tried apple lossless codec(.m4a). neither work.
i want it 5.1 WAV but in meguis "encoders settings" no uncompressed wav. and SplitAc3 log:Code:
FileSize : 414512640 bytes ---------- First valid Header Time eq. : 8635680 ms. SamplCod : 0 (0:48, 1:44.1, 2:32 KHz.) BitRate : 384 Kb/s ChanMode : 7 (1:1/0, 2:2/0, 3:3/0, 4:2/1, 5:3/1, 6:2/2, 7:3/2) FrameSize: 1536 bytes ---------- Process ( 25.000000 fps is used for Trim) Time: 0 ms. Written: 269865 frames 5.1 ( 8635680 ms.) Trim(0, 215891) ---------- End of File Total time: 8635680 ms. at EOF T. written: 269865 frames 5.1. T. written: 0 frames 2.0.
If I want to convert to wave with MeGUI (which can't), I sometimes use it to convert to flac, as it's lossless, then I convert the flac file to wave with something else.
I don't know if it's what's happening in your case, but sometimes the number of channels can change and that can cause decoding/encoding problems. An example might be a TV capture where the audio switches from 5.1ch to stereo and back again etc. Whatever the reason though, if the usual methods fail, I use DirectShow with ffdshow decoding. It'll usually keep going even when other methods fail. You can set DirectShow as the preferred decoder in MeGUI's audio encoder configuration. If ffdshow is decoding, enable it's mixer filter and set it for the desired number of channels (ie 5.1ch). That way, if the channel count changes (or something similar) ffdshow should keep on outputting 5.1ch anyway (even if all but the stereo channels are empty) and the encoder should keep on encoding......
If you can output a complete 5.1ch flac file with MeGUI that way, you can then convert it to a wave file with TAudioConverter.
Last edited by chazz spacey; 27th Dec 2014 at 05:38.
ok. i used avs script with option ConvertAudioTo16bit(). it worked. but i used FFAudioSource. and with ffaudiosource the duration is 32 ms shorter than original.
so how to make megui to use directshow with 16bit depth?
You probably need to set ffdhow to only output 16 bit, assuming you're using it for decoding. Uncheck all the other options under the Output section except for "16 bit integer". If that doesn't work you'll need to convert the 24 bit version to 16 bit with another program.
Do you use foobar2000? It's converter will let you select the bitdepth. It'll load all the common audio types and also some video files such as MKV, MP4 or AVI. It'll also open AVS Scripts and convert the audio. Some of the above may require additional plugins, but they should be on the foobar2000 site.
How did you determine the length of the audio changed? Not that it can't. If there's a gap in the audio stream it won't be taken into account when extracting or converting. If there's a way to account for gaps when using AVISynth, I don't know how to do it. If the audio's in a container eac3to supports (such as MKV) you could try extracting the audio with the HD Streams Extractor under the Tools menu. Eac3to will try to fix problems when it extracts..... which reminds me..... that's another option to try if you're having audio conversion problems. Extract the audio with eac3to first and try converting the extracted version instead.
Don't forget, if there's an audio delay MeGUI usually accounts for it, or some of the decoders do..... I think..... so the length can change as a result.
Oh, and....... just because a flac file is reported as 24 bit, it doesm't mean there's 24 bit audio within. For example if you convert a 16 bit wave file to flac while setting the flac bitdepth to 24, MediaInfo might report 24 bit but the audio will probably still be 16 bit unless you physically convert the bitdepth when encoding. If you convert the 24 bit flac file to a 16 bit flac file and the file size doesn't get noticeably smaller, then the audio within was only 16 bit to begin with.
Last edited by hello_hello; 28th Dec 2014 at 05:58.
mediainfo and then checked the converted file. the difference was 32 ms. and i imported converted file to sony vegas to see if the length is what mediainfo says, mediainfo was right.
MKVExtractGUI2) to WAV but again not fully converted.
And THANK YOU VERY MUCH AGAIN hello_hello