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8th March 2013, 16:34 | #12161 | Link |
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Wov it works with 3.27 now why the temporary wav file for 2hour 8 channel audio track only has about 8GiB when the same 6 channel temporary wav from same track had almost 20gigs with ffmpeg? Does Arscoft decode lossy? This is what mediainfo writes about it:
Audio Format : PCM Format settings, Endianness : Little Format settings, Sign : Signed Codec ID : 00001000-0000-0100-8000-00AA00389B71 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 And why the encoded track is so quiet? This in log: eac3to v3.27 command line: D:\media\eac3to\eac3to.exe Source.dtshd stdout.wav -normalize ------------------------------------------------------------------------------ DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48kHz (core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz) CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING> Decoding with ArcSoft DTS Decoder... Applying RAW/PCM delay... Writing WAV... Creating file "stdout.pass1.wav"... The original audio track has a constant bit depth of 24 bits. Can this be fixed please? Last edited by Anakunda; 8th March 2013 at 16:51. |
8th March 2013, 17:38 | #12162 | Link | |
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Those files, from Ver v1.1.0.0, work perfectly. Anything newer, not so much. Plus, 20gb for just an audio track. That's not right, considering the complete file, with video, is 27gb ! Edit: is source.dtshd a result of encoding via DTS Master Audio Suite? If so, you have to remove the header hex by demuxing, via Tsmuxer , before running through eac3to.
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"Talk to me Goose" Last edited by rapscallion; 9th March 2013 at 15:42. |
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8th March 2013, 17:45 | #12163 | Link | |
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And where can I get the old AS decoder files which "work perfectly"? I extracted the mines from Totalmediatheatre 6.0.1.119 installation. Are these ok? No this was extracted right from bluray disc by eac3to. Do I need to remove the header anyway? Last edited by Anakunda; 8th March 2013 at 17:48. |
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8th March 2013, 20:29 | #12165 | Link |
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I just realized you were decoding to wav/pcm, so 8gb would not be unusual.
What are you trying to accomplish ?
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"Talk to me Goose" Last edited by rapscallion; 9th March 2013 at 15:42. |
9th March 2013, 06:26 | #12166 | Link | |
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(From TMT 2.1.6.120) Good: v1.1.0.0 25/04/2008 Bad: v1.1.0.0 21/04/2008 More info here: http://forum.doom9.org/showthread.php?p=1266679#post1266679
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9th March 2013, 12:56 | #12167 | Link |
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I just did some rough calculations, but (8 GB / 24 bit) * 64 bit equals roughly 21 GB, so I'd guess that ffmpeg decodes the lossy DTS core track at 64 bit floating point precision whereas ArcSoft decodes the lossless DTS-HD track at its specified precision of 24 bit. Comparing file sizes is not always a reliable way to find out which file is better, it only serves to show which file holds more (theoretical) information.
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9th March 2013, 13:11 | #12168 | Link |
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I guessed that libavcodec uses floating point too, this may explain the huge filesize difference. But I still get awfully sounding conversion on output with
Code:
eac3to Source.dtshd stdout.wav -normalize | qaac --ignorelength -o Destination.m4a - What I don't understand if I open the dtshd file in foobar2000 using eac3to as decoder wrapper, the file plays very good, no annoying hiss and not quiet even in 7.1 layout. Simply no artifacts. Why the conversion is so poor, I use the same command line??? |
9th March 2013, 15:47 | #12169 | Link |
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As I asked above, just what is it that you're trying to accomplish????
@Sparktank, he has v1.1.0.0 25/04/2008
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"Talk to me Goose" Last edited by rapscallion; 9th March 2013 at 15:54. |
9th March 2013, 16:07 | #12171 | Link |
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So, I assume, that you can play the BD on an Apple device.
I use Arcsoft Media Coverter for that. One step process and works every time. Any further with eac3to, maybe someone else can chime in.
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9th March 2013, 16:14 | #12173 | Link |
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I have sometimes had problems with outputting stuff if ffdshow is set to output 32bit floating point audio, there's static and distortion. Have you checked that the problem is not in the decoder of the result instead of something else. Maybe you could put a sample somewhere so we can test it too.
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9th March 2013, 16:52 | #12174 | Link | |
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Quote:
Code:
eac3to "Source.dtshd" "Intermediate.wav" -normalize -full qaac --ignorelength -o "Destination.m4a" "Intermediate.wav" |
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9th March 2013, 19:16 | #12176 | Link |
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I have a few dts files I'm converting to ac3 but the ArcSoft decoder is reporting an error. What are the best ways to get the file converted using an alternate method? For now, I'm basically checking if a failure occurs on the default setting, then use -sonic and if that fails, use -nero. Is this a good approach? Seems to be working fine. Am I using the recommended next-in-line to the ArcSoft decoder by going to sonic first and then nero? Thanks.
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11th March 2013, 01:12 | #12177 | Link | |
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(Best avoid nero altogether for any DTS-HD) |
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11th March 2013, 12:54 | #12179 | Link | |
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The dtshd track reported length by MI is none. In foobar2000 it's reported only 1h02m which is even not a half of true length. ediainfo for dtshd: Code:
Audio Format : DTS Format/Info : Digital Theater Systems Format profile : MA / Core Mode : 16 Format settings, Endianness : Big Bit rate mode : Variable Bit rate : Unknown / 1 509 Kbps Channel(s) : 8 channels / 6 channels Channel positions : Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 24 bits Compression mode : Lossless / Lossy and got this 22Gigs big WAV file. If I open it in foobar2000 it seems to play all channels properly but is reported length only 10m29s Code:
Audio Format : PCM Format profile : Float Codec ID : 00001000-0000-0300-8000-00AA00389B71 Codec ID/Hint : IEEE Duration : 2h 6mn Bit rate mode : Constant Bit rate : 24.6 Mbps Channel(s) : 8 channels Channel positions : Front: L C R, Side: L R, Back: L R, LFE Sampling rate : 48.0 KHz Bit depth : 64 bits Stream size : 21.8 GiB (100%) |
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11th March 2013, 15:09 | #12180 | Link | ||
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Quote:
Quote:
Same as above, WAV files greater than 2 GB may be problematic. However, before I mention that you may need to look for a switch to convert the whole file to AAC, make sure the --ignorelength parameter is present. Alternatively, you can also use the .w64 approach mentiond above, if qaac can handle .w64 files. Last edited by DarkSpace; 11th March 2013 at 21:18. |
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