5.1 AC3 encoding problem

[chazz spacey]

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:

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

[chazz spacey]

TAudioConverter logs:tac_logs.zip

[Cornucopia]

It might be having trouble "converting" to 5.1ch because it already IS 5.1ch.

That 6th channel (LFE) is really the .1ch. That's just how MediaInfo reports it.

Scott

[chazz spacey]

i forget to say i was trying to convert it to uncompressed WAV and then i tried apple lossless codec(.m4a). neither work.

[chazz spacey]

and i converted other 5.1 AC3 files without any problem to uncompressed 5.1 WAV with TAudioConverter.

[ndjamena]

https://forum.videohelp.com/threads/367876-Error-joining-AC3-audio-in-ts-files?p=235213...=1#post2352136
https://forum.videohelp.com/threads/367907-Problem-with-Audio-Compressing-an-avi-with-A...=1#post2352542

[chazz spacey]

i want it 5.1 WAV but in meguis "encoders settings" no uncompressed wav. and SplitAc3 log:

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.

[hello_hello]

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.

[chazz spacey]

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.

yeah i converted to flac with DirectShow before you wrote. and it worked. but bit depth is 24. original bit depth is 16. how to fix it?
this is the settings:

[chazz spacey]

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.
the script:
LoadPlugin("C:\Users\User\Desktop\ffms2-2.20-icl\x86\ffms2.dll")
FFaudioSource("C:\Users\User\Desktop\1.ac3")
ConvertAudioto16bit()

so how to make megui to use directshow with 16bit depth?

[chazz spacey]

anybody?

[hello_hello]

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.

[chazz spacey]

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".

i already did it. didn't work.

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.

no i never used it. WAV didn't work. it didn't fully convert the file. last 20 minutes of the audio is gone. but flac worked without any problem.

How did you determine the length of the audio changed?

i checked original file with 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.

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.

i'll try. btw i converted the file with eact3to(file was already extracted with MKVExtractGUI2) to WAV but again not fully converted.

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.

i am sure that it was physically converted to 24 bit. because when i converted original file(ac3 16 bit) to 16 bit flac with same decoder with avs script and with option ConvertAudioTo16bit(), size was 2 times smaller.

And THANK YOU VERY MUCH AGAIN hello_hello