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23rd June 2007, 17:18 | #221 | Link | |
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so, yes, code to eac3to to remove the least significant byte, when the source TrueHD/MLP is 16bit, seems to be the most proper solution based on the info we now have and we can easily judge, according to the previous test with the compression ratios, which TrueHD track is 24bit and which is 16bit, but still the thing that most least significant bytes with 16bit TrueHD stream are not zero bugs me, because with 16bit MLP most least significant bytes are zero and very few of them are not zero, but my MLP test samples are small, around 100Kb, so maybe then it is normal. Last edited by xkodi; 23rd June 2007 at 17:28. |
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23rd June 2007, 17:27 | #222 | Link |
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I've just checked your files. "164851_nero_decode" and "164851_merge.wav" are 100% identical - if I strip the least significant byte from the "164851_nero_decode" file. And I've done another check: The least significant byte is *ALWAYS* zero in "164851_nero_decode".
Now I'm confused... Where did you find a non-zero least significant byte? |
23rd June 2007, 19:06 | #223 | Link | |
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23rd June 2007, 19:12 | #224 | Link | |
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23rd June 2007, 20:21 | #225 | Link | |
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p.s. Ghost Rider Blu-ray disc seems to have both PCM and TrueHD tracks, so maybe in several weeks, it will be possible for me to do such test, sounds like good test. |
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23rd June 2007, 20:40 | #226 | Link | |
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24th June 2007, 18:46 | #227 | Link |
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I'm currently trying to go from eAC3 to multichannel WMA whilst preserving as much quality as possible, but something isn't right.
I used eAC3to to go from my ec3 file to wav. The resulting wave file would not play using MPC, it simply crashed. Maybe I need another filter? Anyway, undeterred I used Tranzcode to produce 6 mono wave files. These files will play but the sound is garbled, just a series of squarks. I know eAC3to works OK on my machine, because the AC3 files it produces are OK. Any ideas where I am going wrong? Cheers, Beastie. PS. If this problem is not related to eAC3to I'll start a new thread so as not to go off-topic. |
25th June 2007, 08:08 | #228 | Link |
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Latest version of EAC3toGUI (version 0.97)
http://www.sendspace.com/file/zdfq3x Changelog: - Fixed bug that would not allow *.wav files as input. Apologize for any inconvenience this may have caused. The_Keymaker |
25th June 2007, 10:06 | #229 | Link |
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Thx keymaker
Great work! Maybe its better to rename "Orbitlee" to "Sonic" ? I mean Orbitlee DTS/AC3/DD+ Source is only the Source Filter ,the Decoder is SonicCinevisionAudioDecoder4.2 So you can select Nero=NeroAudioDecoder2 or Sonic=SonicCinevisionAudioDecoder4.2 And a .raw input extension would be good Last edited by ACrowley; 25th June 2007 at 10:58. |
25th June 2007, 12:51 | #230 | Link | |
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25th June 2007, 15:40 | #232 | Link |
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Damn. i tried everything
I cant get anything working from a 24bit lpcm from Bluray I tried to demux with SonicHDDemuxer, also with xport. I use this Sox commadline and tried it as little endian or big endian, with -2 and -3 sox.exe -l -r48000 -t .raw -c 6 -3 -s e:\test.raw e:\test.wav nothing! When i use eac3to do it autom. its the same Problem The wave File makes only a heavy loud noise, 6 mono channels ( created with wavewizard or tranzcode) to. The full lpcm 24bit is 7Gb big. The encoded flac from eac3to is "6Gb" and playback fails. Its wrong cause the flac must be around 3Gb. And AC3/ wave output only the loud noise too. A 24-16bit conversion create the same Result When i try it on a 100Mb sample its the same It happens with "all" 24bit lpcm Tracks from Bluray. I really tried everything, checked it it again and again , with no succes LPCM 16 bit works without Problems. I really dont know where the Problem ? Any advice ? I could only image its a Problem caused by demuxing. Heres the same Problem, but no Solution so far i can see ? http://forum.doom9.org/showthread.php?t=124080&page=2 Last edited by ACrowley; 25th June 2007 at 16:24. |
25th June 2007, 15:42 | #233 | Link |
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I have a feature request.
Currently I'm running eac3to on a windows virtual machine in which it stores everything (source + raw + wav + ac3/flac) to a network share. For multiple reasons, I'd like to proceed on linux after the raw file has been dumped. Seeing that most the tools used (all but the directsource stuff?) are available there, I'd like a simple parameter (like -n for make) that spits out the commands it would run after the directshow dump to raw, so that I can run sox and aften in linux. Thats basically everything after: Code:
E-AC3, 5.1 channels, 1:59:02, 1536kbit/s, 48khz Muxing eac3 file to raw. Please wait... Audio Decoder output was 24 bit this time. Yes, I could just dump to raw myself, then do strings eac3to.exe and figure out the right commands, but i'm lazy, and it's probably really simple to add |
26th June 2007, 09:10 | #236 | Link | |
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EDIT : I have it! Very strange cause i tried all theye combinations before...mhh Problem : even the 24bit lpcm was demxued with graphedit, it works only with Big Endian Switch in Sox 1 .sox.exe -B -r48000 -t .raw -c 6 -3 -s e:\test.raw e:\test.wav 2. wavewizard 0 1 2 5 3 4 3. encode to whatever @Madshi Ofcourse i tried all Combinations mayn Times before. I can say :eac3to fails (for me)on these 24bit lpcm , But it works manually . Even with Big or Little Endian switch, nothing! No working wave, no flac,no ac3. outputs is a loud noise. eac3to e:\test.pcm e:\lpcm.wav -blu-ray -24 ( i tried with -B / -L all combinations) Heres the sample. Its LPCM 24Bit 5.1 from BluRay "Apocalypto" http://rapidshare.com/files/39409043...24Bit.rar.html Last edited by ACrowley; 26th June 2007 at 16:55. |
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26th June 2007, 16:15 | #237 | Link |
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First thanks for this great prog
I don't know if this is a bug or if I'm doing something wrong but I discovered this because some one said that you can create smaller flacs with the -8 option (default is -5 see http://flac.sourceforge.net/document...coding_options). When I use the following command "eac3to.exe input.ddp output.flac -nero" and I want to compress the flac some more with the command "flac output.flac -o output.reenc.flac -f -8". I get the following error "output.flac: ERROR: FLAC input has STREAMINFO with unknown total samples which is not supported" With "metaflac.exe output.flac --list" I get this. Code:
METADATA block #0 type: 0 (STREAMINFO) is last: false length: 34 minimum blocksize: 4096 samples maximum blocksize: 4096 samples minimum framesize: 0 bytes maximum framesize: 0 bytes sample_rate: 48000 Hz channels: 6 bits-per-sample: 24 total samples: 0 MD5 signature: 00000000000000000000000000000000 METADATA block #1 type: 4 (VORBIS_COMMENT) is last: false length: 40 vendor string: reference libFLAC 1.1.4 20070213 comments: 0 METADATA block #2 type: 1 (PADDING) is last: true length: 8192 Now that someone who told me about the -8 option also gave me some pointers to fix this. Remember the sample_rate, channels and bits-per-sample. Create a raw file from the corrupt flac. "flac -d output.flac -o temp.raw --force-raw-format --endian=little --sign=signed" Next create the flac from the raw with the information from metaflac (with the -8 option if you want) "flac -8 --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed -o "output.fixed.flac" - < temp.raw" Now you can see that when you use metaflac on the new file it has the correct streaminfo, but I think eac3to should set the correct streaminfo to begin with . Code:
METADATA block #0 type: 0 (STREAMINFO) is last: false length: 34 minimum blocksize: 4096 samples maximum blocksize: 4096 samples minimum framesize: 417 bytes maximum framesize: 42568 bytes sample_rate: 48000 Hz channels: 6 bits-per-sample: 24 total samples: 310497536 MD5 signature: c5fd63920d96db81ec780bacb00ca4eb METADATA block #1 type: 4 (VORBIS_COMMENT) is last: false length: 40 vendor string: reference libFLAC 1.1.4 20070213 comments: 0 METADATA block #2 type: 1 (PADDING) is last: true length: 8192 Last edited by kweek; 26th June 2007 at 16:21. |
26th June 2007, 17:02 | #238 | Link | |
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-5 should output a liottle bit bigger File with less compression and lower Quality eac3to use -8 too, i would only use 8 for "HD Audio" Last edited by ACrowley; 26th June 2007 at 17:21. |
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26th June 2007, 19:35 | #239 | Link | |
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Do you mean that eac3to uses the -8 option for flac? I'm using eac3to mainly for highdef audio. |
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26th June 2007, 20:30 | #240 | Link | |
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yes eac3to should use -8 The flac output has the same Size compared to a media coder -8 output. ################ ABout Sonic Cinevision 2 beta : There a a few new Decoder Versions. Located in C:\xxSonic/Cinevision/Filters 1. Video Decoder (I think based on Elecard/Moonlight) all are .ax Files and you can reg/use them all -Sonic AVC/H264 Decoder -Sonic VC1 Decoder -Sonic MPEG Demuxer -no AudioDecoder 2. The Cinemaster HD 4.3 Decoder ,all are .dll Files -SonicHDDemxuer 4.3 -SonicCinemsterVideoDecoder 4.3 -SonicCinemsterÁudioDecoder 4.3 The .ax Filter are working very good on BluRay H264/VC1 Video. No Artefacts on H264 PAFF etc. More interesting for this Thread is ,that the 4.3 CinemasterAudioDecoder connects to TrueHD! The TrueHD is playing ,but you can hear a noise in the backround. But basicly it should work now. I Tried it on Matrix HJDDVD TrueHD 24bit The other Thing is the Cinevision 2 has no installation fo the Cinemaster 4.2 Decoder. When i try to reg the manually it fails with a Error Message. You can overwrite the 4.2 dlls with the new 4.3. But Avisynth wont connect anymore. So, it would be nice if somebody can find out a workaround how to properly reg. the Cinemaster HD 4.3 Decoder Last edited by ACrowley; 26th June 2007 at 20:46. |
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