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31st May 2010, 15:28 | #10041 | Link |
マジンさん
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Hello, since this is my first time dealing with DTS, I wanted to ask some questions here before doing anything wrong...
eac3to reports that my Blu-ray has "LPCM 16 bit 5.1", "DTS Master Audio 16 bit 5.1" and "LPCM 16 bit 2.0" audio tracks. I want to extract the best quality one, but I'm a bit confused since eac3to reports that the DTS MA track is 16bit (which should decompressed be 1:1 to the first LPCM one, shouldn't it?) while its DTS core is 24bit (MediaInfo says the same). Can that be correct? Shouldn't the DTS MA also be 24 bit or the DTS core 16 bit? |
31st May 2010, 21:56 | #10044 | Link | |||
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If your source claims to be 24bit, and if you use -down16, then eac3to will apply dithering - even if it's only really 16bit. The reason for that is that eac3to doesn't know which bitdepth the source has until processing is complete. You surely don't want eac3to to scan the whole file first, everytime, before starting processing, do you? |
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31st May 2010, 22:07 | #10045 | Link |
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eac3to v3.19 released
http://madshi.net/eac3to.zip Code:
* added support for 3D Blu-Rays (playlists, detection & demuxing) Code:
D:\BDMV\PLAYLIST>eac3to 00001.mpls 1) 00001.mpls, 00001.m2ts, 0:00:09 - h264/AVC (left eye), 1080p24 /1.001 (16:9) - h264/AVC (right eye), 1080p24 /1.001 (16:9) - DTS, English, multi-channel, 48kHz Code:
M2TS, 2 video tracks, 1 audio track, 0:00:09, 24p /1.001 1: h264/AVC (left eye), 1080p24 /1.001 (16:9) 2: h264/AVC (right eye), 1080p24 /1.001 (16:9) 3: DTS, English, 5.1 channels, 16 bits, 1509kbps, 48kHz a03 Extracting audio track number 3... v01 Extracting video track number 1... v01 Extracting video track number 1... a03 Patching bitdepth to 24 bits... v01 Creating file "c:\desktop\test - 2 - h264 (right eye), 1080p24.h264"... v01 Creating file "c:\desktop\test - 1 - h264 (left eye), 1080p24.h264"... a03 Creating file "c:\desktop\test - 3 - DTS, English, 5.1 channels, 16 bits, 15 09kbps, 48kHz.dts"... Video track 1 contains 211 frames. Video track 2 contains 211 frames. eac3to processing took 2 seconds. Done. (Before anybody asks: The "left eye" stream is a normal h264 stream, which every h264 decoder supports. The "right eye" stream is a special h264 extension stream which current h264 PC decoders don't understand.) |
31st May 2010, 22:55 | #10046 | Link |
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"If your source claims to be 24bit, and if you use -down16, then eac3to will apply dithering - even if it's only really 16bit."
So, I'll get better quality on padded sources if I first demux/convert to .wav? That way eac3to can work out if its padded or true 24bit. If its padded it'll do a second pass and remove the padding and if its 24bit it will stay as 24bit and I can then run it through eac3to again with -down16? (yeah more time consuming but its the computer doing the work, not me ) |
31st May 2010, 23:19 | #10048 | Link | |
Matroska find' ich toll
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thats wonderful news. I hope this issue with the "right eye" stream will be fixed at someone next time. thanks for your working on eac3to. |
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1st June 2010, 04:03 | #10050 | Link | |
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If eac3to kept it flagged as 16-bit like the source, some decoders might only decode in 16-bit, so madshi smartly made eac3to flag all DTS as 24-bit just in case |
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1st June 2010, 06:53 | #10051 | Link | |
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1st June 2010, 07:14 | #10052 | Link | |
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1st June 2010, 07:22 | #10053 | Link |
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It is on my to do list - like so many other things I'd like to add to eac3to. I'm currently lacking time, though. And the time I can find for HTPC development mostly go into madVR at this point, since madVR is still a work in progress, while eac3to is pretty stable. There probably will come a time when I come back to eac3to with more development resources, but it's not here yet right now. (3D was important and easy to add.)
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1st June 2010, 08:18 | #10054 | Link |
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So are there any decoders which decode non 24-bit padded 16-bit DTS properly and account for the variations in frequency domain representation when the PCM audio is recreated during decoding? Does ffdshow audio decoder properly do that with the 24-bit padding? Also I've always been curious to know why are there different outputs for the different bit integers. I have checked only 32 bit floating point because I assumed it was the "best" option and gives the decoder lots of "room" for whatever it needs to do though it might unnecessarily raise cpu usage since the decoding actually only needs 24-bit for 16-bit Digital Dolby or DTS audio.
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1st June 2010, 08:28 | #10055 | Link | |
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Standard DTS is never padded 16bit. It's floating point. DTS-MA is usually also never padded 16bit. It's either 16bit, and then the decoder outputs 16bit. Or it's 24bit and then the decoder outputs 24bit. The only situation where 16bit -> 24bit padding usually occurs is with TrueHD/MLP tracks. I've seen tons of DTS-MA tracks and their bitdepth information is usually correct. I've only ever seen one single DTS-MA track which was flagged as 24bit, but only contained 16bit data, while I've seen hundreds of DTS-MA tracks which were flagged correctly. |
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1st June 2010, 10:23 | #10058 | Link |
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Standard DTS. The link in some posts above said that when standard dts is converted to PCM via decoding it is 8 bits more therefore some decoders need the 24-bit padding in order to have all the bits that's why I was wondering if there are any decoders that can play without the padding and still anticipate 8 bits more when decoding.
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1st June 2010, 10:25 | #10059 | Link |
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I don't know for sure which decoders are outputting what, if eac3to doesn't patch the bitdepth to 24bit. ArcSoft outputs 16bit without the patching, I think. To which bitdepth libav/ffmpeg decodes, depends on the libav/ffmpeg version and the decoder/media player which uses it. If you decode by using eac3to, you will always get full bitdepth.
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1st June 2010, 12:25 | #10060 | Link |
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madshi
A bit more info - I've been looking further into the problem I reported with reducing the bitdepth of 24 bit audio. It seems to be near peak values that don't necessarily have their depth reduced. Presumably dithering means that a value can be rounded up or down. However near the peak I suppose the value should always be rounded down - ie truncated. For example the centre channel in the test file I posted contains the following sequence of samples (as wav): Code:
24 bit in 20 bit out EC 65 77 F0 65 77 FC FF 7F FF FF 7F FE FF 7F FF FF 7F FD FF 7F F0 FF 7F 51 E8 5C 50 E8 5C Code:
eac3to.exe "test1 24.flac" "test1 20.flac" -1dB -down20 Last edited by TinTime; 1st June 2010 at 13:03. Reason: Initial post incorrect |
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