i have been trying to convert some audio tracks, that i extracted from my bluray collection, to aac using xilisoft audio converter 6.5, and with some tracks i find that only a couple of minutes are converted and the rest is dropped. a file that should have around 100mb only has 30mb, with about 5 minutes of audio. here is the log from xilisoft:
I am not too keen in audio converting so i dont fully grasp the meaing of the above, it says "ERROR: block code look-up failed", which i assume points to some structural problems with the source file? after trying switch audio converter, it also breaks up at the same point. the source audio files in question are dts 5.1. when opening them in mediainfo, i get this:HTML Code:[matroska @ 00D48210]Unknown entry 0xF0 Last message repeated 600 times Input #0, matroska, from 'E:\2.05_2.audio': Duration: 00:20:03.24, start: 0.000000, bitrate: 1510 kb/s, rotation: 0.00?? Stream #0.0(eng): Audio: dca (DTS), 48000 Hz, 5:1, 1536 kb/s Output #0, adts, to 'E:\2.05_2.aac': Stream #0.0(eng): Audio: libfaac, 48000 Hz, 5:1, 112 kb/s Stream mapping: Stream #0.0 -> #0.0 [matroska @ 00D48210]Unknown entry 0x261C34 [matroska @ 00D48210]Unknown entry 0x6120 ERROR: block code look-up failed Last message repeated 22 times [dca @ 00D39CE0]Didn't get subframe DSYNC Last message repeated 49 times [dca @ 00D39CE0]Not a valid DCA frame Error while decoding stream #0.0 Last message repeated 6 times [matroska @ 00D48210]Invalid EBML number size tag 0x0d at pos 64401545 (0x3d6b089) video:0kB audio:28197kB global headers:0kB muxing overhead 0.000000% Conversion finished, total time elapsed: 00:01:02.991
i encoded the blurays with handbrake, but i did an "auto passthrough" on the audio tracks since handbrake has no aac included. so i assume the tracks are in original form like they were on the bluray. when i watch the encoded video with the audio in question (which should be untoched because of auto-passthrough), there are no errors or other problems. it is only during conversion, that these errors occur.HTML Code:Audio ID : 1 Format : DTS Format/Info : Digital Theater Systems Mode : 16 Format settings, Endianness : Big Codec ID : A_DTS Duration : 20mn 3s Bit rate mode : Constant Bit rate : 1 509 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Frame rate : 93.750 fps (512 spf) Bit depth : 16 bits Compression mode : Lossy Stream size : 216 MiB (100%) Title : Englisch Language : English Default : No Forced : No
can anyone give me a hint as to what the problem here is? is there a way to solve this, or is the source material broken?
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ok, thanx for the tip.
i would like to broaden my question to this: what is currently the best audio converter?
that is of course a matter of debate, but i would like to get some opinions with special regard to the following attributes:
- codec quality
- format variety
- ability to control the conversion (eg, more options than small/middle/large)
- compliance with standards (eg, producing files that are compatible with other soft- and hardware and not cause compatibilty issues)
- efficiency of processing
thanx for any opinions
i have tried eac3to, but it didnt really help me. i converted an ac3 to flac, the source ac3 file has 26:38 minutes, the resulting flac has 12:28. here is the log file:
eac3to v3.31 command line: eac3to Bonus_1.audio Bonus_1.flac ------------------------------------------------------------------------------ MKA, 1 audio track, 0:26:38 1: AC3, German, 2.0 channels, 224kbps, 48kHz, dialnorm: -29dB "Deutsch" Track 1 is used for destination file "Bonus_1.flac". [a01] Extracting audio track number 1... [a01] Removing AC3 dialog normalization... [a01] Decoding with libav/ffmpeg... [a01] Reducing depth from 64 to 24 bits... [a01] Encoding FLAC with libFlac... [a01] Creating file "Bonus_1.flac"... [a01] Clipping detected, a 2nd pass will be necessary. <WARNING> [a01] This track is not clean. <WARNING> [a01] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING> [a01] Starting 2nd pass... [a01] Extracting audio track number 1... [a01] Removing AC3 dialog normalization... [a01] Decoding with libav/ffmpeg... [a01] Reducing depth from 64 to 24 bits... [a01] Encoding FLAC with libFlac... [a01] Applying -1.32dB gain... [a01] Creating file "Bonus_1.flac"... [a01] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING> eac3to processing took 1 minute, 37 seconds. Done.
also i have tried to convert the flac to aac with my xilisoft converter, it failed immediately, the resulting file was 0kb.
are all my audio tracks damaged? am i missing something here? as i said, all these tracks are directly from dvd/blurays, no conversion or manipulation happened.
i dont get where all these problems come from...
It looks like there is some corruption. Your original post used dts as as the source, the second uses AC3.
Here is another bug signed off as fixed: https://trac.ffmpeg.org/ticket/5319
Whether or not the fix is complete, I have no idea. Ideally, you want to be able to 'see' the source file.
Try loading the AC3 into Audacity. It can be saved as FLAC from there if it loads/plays well.
handbrake to h265, leaving the audio tracks as they are, since handbrake doesnt handle audio well in general. some of these tracks are ac3, some dts. the ac3 have pretty decent quality/filesize, so basically im ok with letting them the way they are. not so with dts: in my encoded files, the dts-track often holds more than 50% of the entire filesize, which is not acceptable to me. i try to find a good enough compromise between quality and filesize, for audio to video i consider a rate of 1 to 5 acceptable, meaning an audio file can be 20% the size of the video without bugging me
i originally intended to convert both ac3 and dts to aac. i was somewhat fixated on aac, because i read it is the most advanced audio codec and all the videos i find on the internet are aac. also i have been getting pretty good results with my audio converter, but the problems along the way are maddening. so this is obviously not a practical solution, especially considering my goal of automating the whole process. i think i can settle to leave ac3 as is and convert dts to ac3 as well. my devices support aac as well as ac3, so that could be a workable solution.
i have now tried to convert a dts-track to ac3 using the eac3to tool, and it has not only worked without errors but the results are also pretty much equal to my aac tests, as well in quality as in size. another big perk is that i can easily plug this part into my automated process. it will require more testing to see if there are not some other problems popping up just when i start feeling comfortable with the whole thing, but that remains to be seen
the only remaining question i have so far is this: is there a way to (automatically) adjust the volume of an audio track to another one? eg, make track 1 louder or less loud, so that it fits track 2 (i have all the tracks in separate files, so extracting them is not an issue). can i do this with eac3to? that would be christmas and easter in one package
thanks for the help so far...
Last edited by bennyutzer; 2nd Oct 2017 at 17:08.
i am having a weird problem with eac3to here. i am using the following command:
eac3to G:/test/Bonus_1.dts G:/test/Bonus_1.ac3
do i have do escape the destination file name in some way? do i have to use other slashes? \ instead of /?
the source file is found, recognised and converted properly, only the path issue ruins my day
solved it, inexplicably i actually have to use the other slash for the destination file. the correct command is this:
eac3to G:/test/Bonus_1.dts G:\test\Bonus_1.ac3
Never tried to change the volume of an individual track with it. There is a volume change (example: +/-2dB) but relates to all channels I think. You can use the envelope tool in Audacity to alter an individual track: http://wiredpen.com/resources/audacity/adjust-sound-audacity-envelope-tool/