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16th November 2015, 09:10 | #13681 | Link |
German doom9/Gleitz SuMo
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Regarding DVD Video, that was one of the reasons to extract "the movie PGC" instead of processing VOB sequences as authored: to avoid trailer audio issues like asynchronity.
Removing trailers from BD playlists may be a different topic... |
17th November 2015, 06:51 | #13682 | Link |
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are DTS:X headphone (2.0) tracks which are currently detected as DTS track by eac3to actually lossy or lossless tracks, like other DTS-HD MA tracks? and are DTS:X tracks are not denoted as such by eac3to yet?
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17th November 2015, 08:04 | #13683 | Link |
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Could you provide a sample file of DTS:X headphone?
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Want to know all about your media files? http://mediaarea.net/MediaInfo |
17th November 2015, 10:43 | #13684 | Link |
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If anyone is interested here's the latest unofficial MediaInfo dll:
http://www.mediafire.com/download/5y.../MediaInfo.zip This is what it gets you: Code:
General Unique ID : 219291506723631580012532402152276823948 (0xA4FA02080541F5119808D773964D0F8C) Complete name : D:\Zilla.mkv Format : Matroska Format version : Version 4 / Version 2 File size : 36.8 GiB Duration : 2h 18mn Overall bit rate mode : Variable Overall bit rate : 37.9 Mbps Movie name : Zilla Encoded date : UTC 2015-11-17 09:05:56 Writing application : mkvmerge v8.5.1 ('Crosses') 64bit Writing library : libebml v1.3.3 + libmatroska v1.4.4 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : High@L4.1 Format settings, CABAC : Yes Format settings, ReFrames : 2 frames Format settings, GOP : M=1, N=10 Codec ID : V_MPEG4/ISO/AVC Duration : 2h 18mn Bit rate mode : Variable Bit rate : 34.7 Mbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate mode : Constant Frame rate : 23.976 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.698 Stream size : 33.6 GiB (92%) Language : English Default : No Forced : No Audio ID : 2 Format : FLAC Format/Info : Free Lossless Audio Codec Codec ID : A_FLAC Duration : 2h 18mn Bit rate mode : Variable Bit rate : 3 202 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 24 bits Detected bit depth : 20 bits Stream size : 3.10 GiB (8%) Writing library : libFLAC 1.2.1 (UTC 2007-09-17) Language : English Default : Yes Forced : No Text #1 ID : 3 Format : PGS Muxing mode : zlib Codec ID : S_HDMV/PGS Codec ID/Info : Picture based subtitle format used on BDs/HD-DVDs Duration : 2h 1mn Bit rate : 11.4 Kbps Count of elements : 2084 Stream size : 9.90 MiB (0%) Language : English Default : No Forced : No Text #2 ID : 4 Format : PGS Muxing mode : zlib Codec ID : S_HDMV/PGS Codec ID/Info : Picture based subtitle format used on BDs/HD-DVDs Duration : 1h 23mn Bit rate : 198 bps Count of elements : 36 Stream size : 121 KiB (0%) Language : English Default : No Forced : No Text #3 ID : 5 Format : PGS Muxing mode : zlib Codec ID : S_HDMV/PGS Codec ID/Info : Picture based subtitle format used on BDs/HD-DVDs Duration : 2h 9mn Bit rate : 7 053 bps Count of elements : 2460 Stream size : 6.52 MiB (0%) Language : English Default : No Forced : No Menu 00:00:00.000 : en:Start 00:07:48.092 : en:Collision Course 00:17:25.502 : en:Worm Guy 00:25:28.026 : en:"Gojira" 00:36:53.461 : en:Inside a Footprint 00:41:47.880 : en:Insurance Claim 00:48:19.479 : en:Caught on Something 01:01:35.900 : en:"I Got a Bite" 01:11:05.803 : en:New Kid in Town 01:21:38.685 : en:23Rd St. Station 01:30:26.087 : en:Drawing Him Out 01:39:13.989 : en:Fire at Will: One 01:50:23.867 : en:Copter Chase 01:52:28.658 : en:"He's Pregnant" 02:00:36.771 : en:Shaken, Not Stirred 02:05:56.882 : en:Section Five Subtitle track one has 2084 elements in it, subtitle two has only 36, that one must be forced subtitles, the last subtitle has 2460 elements, which is more than the first, so the first must be the regular subtitles and the last must be SDH. Oh, right, the flac was muxed straight into the file using MKVMerge, so the UID is 3419832279421249968 and yet its statistics tags are still being applied to it correctly. And then there's this: Code:
General Unique ID : 215636814642891718231343589359444197377 (0xA23A23CCC54ADBF1948103B2D1428001) Complete name : D:\DTS-X.mka Format : Matroska Format version : Version 4 / Version 2 File size : 123 MiB Duration : 3mn 32s Overall bit rate : 4 861 Kbps Encoded date : UTC 2015-11-15 15:35:39 Writing application : mkvmerge v8.5.1 ('Crosses') 64bit Writing library : libebml v1.3.3 + libmatroska v1.4.4 Audio ID : 1 Format : DTS Format/Info : Digital Theater Systems Format profile : X / MA / Core Mode : 16 Format settings, Endianness : Big Codec ID : A_DTS Duration : 3mn 32s Bit rate mode : Variable / Variable / Constant Bit rate : 4 859 Kbps / 4 859 Kbps / 1 509 Kbps Channel(s) : Object Orientated / 8 channels / 6 channels Channel positions : Object Orientated / Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, LFE Sampling rate : / 48.0 KHz / 48.0 KHz Bit depth : / 24 bits / 24 bits Compression mode : / Lossless / Lossy Stream size : 123 MiB (100%) Language : English Default : Yes Forced : No and this: Code:
General Unique ID : 186823894588399821734555224452256288412 (0x8C8CF929A176D63082E6B973CE30CE9C) Complete name : D:\DTS ES.mka Format : Matroska Format version : Version 4 / Version 2 File size : 6.80 MiB Duration : 4s 11ms Overall bit rate : 14.2 Mbps Encoded date : UTC 2015-11-13 23:30:23 Writing application : mkvmerge v8.5.1 ('Crosses') 64bit Writing library : libebml v1.3.3 + libmatroska v1.4.4 Audio #1 ID : 1 Format : DTS Format/Info : Digital Theater Systems Format profile : MA / ES Matrix / Core Mode : 16 Format settings, Endianness : Big Codec ID : A_DTS Duration : 4s 10ms Bit rate mode : Variable / Constant / Constant Bit rate : 2 812 Kbps / 1 509 Kbps / 1 509 Kbps Channel(s) : 8 channels / 7 channels / 6 channels Channel positions : Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 24 bits Compression mode : Lossless / Lossy / Lossy Stream size : 1.34 MiB (20%) Language : English Default : Yes Forced : No Audio #2 ID : 2 Format : DTS Format/Info : Digital Theater Systems Format profile : MA / ES Discrete / Core Mode : 16 Format settings, Endianness : Big Codec ID : A_DTS Duration : 4s 0ms Bit rate mode : Variable / Constant / Constant Bit rate : 2 269 Kbps / 1 509 Kbps / 1 509 Kbps Channel(s) : 7 channels / 7 channels / 6 channels Channel positions : Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 24 bits Compression mode : Lossless / Lossy / Lossy Stream size : 1.08 MiB (16%) Language : English Default : No Forced : No Audio #3 ID : 3 Format : DTS Format/Info : Digital Theater Systems Format profile : ES Discrete / Core Mode : 16 Format settings, Endianness : Big Codec ID : A_DTS Duration : 4s 0ms Bit rate mode : Constant Bit rate : 1 509 Kbps Channel(s) : 7 channels / 6 channels Channel positions : Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Bit depth : 24 bits Compression mode : Lossy Stream size : 737 KiB (11%) Language : English Default : No Forced : No Last edited by ndjamena; 17th November 2015 at 14:51. |
19th November 2015, 07:23 | #13685 | Link |
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I have a question regarding a 2.0 24-bit DTS-HD MA file. I've decoded it with dcadec (eac3to v3.31) and the Arcsoft decoder (eac3to v3.28) and the file decoded with the Arcsoft decoder is quieter.
According to eac3to the DTS file has a Dialnorm value of -4dB. If my assumptions about Dialnorm are correct, that means that a gain has to be applied, since neither flac nor wav support Dialnorm. I thought that maybe the dialogue normalization wasn't taken into account while using the Arcsoft decoder, so I encoded it again with the "+4dB" option and tested the difference to the dcadec version. In order to do so I opened both audio streams in Reaper and aplied phase inversion to one of the streams. Now if they are identical they should cancel each other out resulting in complete silence. There was still a sound at about -120dB (deviation from the peak), which is absolutely inaudible, though. Since I don't know much about Dialnorm I want to ask which version is correct, the one without the gain or the one with it and the dcadec version. Also, is there a reason there is still a small measurable difference between the Arcsoft decoded file with the gain and the one decoded by dcadec, shouldn't they be the same if the gain is applied manually? And finally, is there a better way to check if two audio files are identical? On a side note, I did the same test with Reaper using a 5.1 24-bit DTS-HD MA track without a Dialnorm value. There was no measurable difference between Arcsoft and dcadec, meaning they are in fact completely identical. Last edited by Frechdachs; 19th November 2015 at 07:35. |
19th November 2015, 11:02 | #13686 | Link |
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@Frechdachs
- Dialog Normalization was a interesting concept if all sounds respect it. But there are many (TV advertisement, remastered CD's Loudness war, ...) than try to offer the max digital volume in the wrong concept than loud volume is better quality. Thats enforce to use the Volume knob between normalized and not normalized sounds. eac3to is a transcoding tool and, by default, try, if can do, to not apply Dialog Normalization to offer the original sound without attenuation. What is correct? Is your choice. - There are other methods to compare wav's but for this pourpose your method is correct (I also use it). Take in mind than apply -4dB gain (DialNorm), and after a manual +4dB gain, is not a lossless operation and your can lose a bit of precission. The dcadec output is the lossless source.
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19th November 2015, 14:57 | #13687 | Link | |
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Quote:
Code:
E:\>md5sum Tomorrowland*.wav 8e8d58cc32c6432a0c5222c7086e6444 *Tomorrowland_ArcSoft_decoder_eac3to-3.31.wav 8e8d58cc32c6432a0c5222c7086e6444 *Tomorrowland_default_eac3to-3.31.wav Code:
8 989 397 060 Tomorrowland_ArcSoft_decoder_eac3to-3.31.wav 8 989 397 060 Tomorrowland_default_eac3to-3.31.wav |
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19th November 2015, 17:18 | #13688 | Link |
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@Frechdachs
Assuming you want the best possible result (sorry), the correct "version" is the one produced by dcadec, since it does not apply 4db of gain reduction. As tebasuna51 mentioned, the Arcsoft file will be lower quality since the LSBs (the noisefloor @ 120db) get trashed as your source file is integer precision format. Last edited by AlexKane; 19th November 2015 at 17:21. |
19th November 2015, 21:52 | #13690 | Link |
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Come to think of it. I don't think I've ever seen my surround sound amplifier respond to a DTS-HD MA dialnorm flag. I've only ever seen it respond to a Dolby Digital dialnorm flag
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19th November 2015, 22:06 | #13691 | Link |
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The Dolby Digital spec requires Dolby Digital decoders to apply dialnorm by default, unless the user specifically disables it. DTS is not as stupid, fortunately, so DTS tracks have much less problems with that. When using libdcadec, dialnorm is by default ignored, so eac3to decoding no longer has any problems with that, anyway. eac3to still cannot fully remove dialnorm from DTS-HD tracks, though, because doing so would require to rewrite the whole HD frame structure, including CRCs etc, which is very complicated.
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19th November 2015, 22:48 | #13692 | Link |
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does detecting DTS:X headphone tracks work differently from detecting norma DTS:X tracks? from what Ive heard using this info here https://github.com/foo86/dcadec/issues/37 does not lead to success in cases of headphone tracks.
maybe the dcadec developer could be able to figure something out as well in this case?
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19th November 2015, 23:19 | #13693 | Link | ||
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Quote:
Quote:
So if I get it straight this time, it meanst that when I use the Arcsoft decoder with the "+4dB" option, there is first a negative gain of -4dB applied (because of dialnorm) and then a positive gain again, which is retarded. Is this correct? You mean when input and output is both DTS? If encoding to flac for example, either dialnorm has to be ignored or there has to be a gain applied, because flac doesn't support dialnorm. Or does it? |
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19th November 2015, 23:24 | #13694 | Link |
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When eac3to decodes the DTS track to convert it to FLAC, dialnorm is simply ignored.
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20th November 2015, 08:41 | #13695 | Link | |
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Quote:
A dialnorm flag of -4db means that the decoder needs to apply 4db of gain reduction during decoding. In your case, Arcsoft respects the dialnorm flag while dcadec does not. If all you want is "Extract the audio as it is stored" and encode it in another format, you need to use dcadec. You can then apply any level adjustment during playback. Last edited by AlexKane; 20th November 2015 at 08:48. |
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20th November 2015, 09:30 | #13696 | Link | |
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Quote:
But not retarded, when you recode the time to do the operations is not important. Is other problem: You lose precission in round to int values. For instance you have a sample with a int 24 bits volume value of 16777211. When you apply -4dB you obtain a real value than you must round to int: Code:
Int value -4dB Int Up Int Down --------- ------------- -------- -------- 16777211 10585704,5003 10585705 10585704 Code:
Int value +4dB Int --------- ------------- -------- 10585705 16777211,7919 16777212 10585704 16777210,2070 16777210
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20th November 2015, 21:58 | #13697 | Link | |
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Thank you for all the replys. Everything makes sense now.
Quote:
That's what I meant when I said that it would be kind of stupid by me to do that. But thanks for the explaination on how the precision loss comes to be. Much appreciated. |
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21st November 2015, 00:16 | #13698 | Link |
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https://github.com/MediaArea/MediaIn...4d5b9e34f94144
Code:
if (HD_SubStreams_Count == 4) Fill(Stream_Audio, 0, Audio_Codec, "ATMOS"); Is there anything else I should look out for? |
21st November 2015, 16:56 | #13699 | Link |
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eac3to + Nero Audio Decoder (Nero 7) are dropping first aac frame on decoding. always. along with faad. ffmpeg & Nero AAC Decoder 1.5.1.0 are doing it correctly. can be verified as example with extracted aac from qaac --no-delay encoding
possible solution - to feed first frame twice to decoder? oh, and as i've heard, 5.1 aac decoding is broken in eac3to after 3.0.1 version, but it's probably nothing new with it /just sayin' Last edited by kukushka; 21st November 2015 at 17:25. |
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eac3to |
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