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22nd December 2008, 09:16 | #7481 | Link | |||
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What do you mean exactly with "correctly sequenced"? |
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22nd December 2008, 11:56 | #7483 | Link |
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I'm checking my PMs, anyway, so there's no need to flood this thread with such posts.
------- Your DTSWAV sample is strange. It begin with a lot of zero data and then half a DTS frame and then finally with the real DTS data. eac3to cannot automatically detect such a file as a valid DTSWAV file. Don't know, maybe the EAC extra features (which are very useful for ripping normal WAV data) are harmful for DTSWAV tracks? You could try playing around with the EAC options to see whether that makes any difference. Furthermore the DTS data in the DTSWAV file is kind of broken. That's probably not the fault of EAC. The framesize is supposed to be 3585 bytes, but the data only contains 3584 bytes per frame. That's why eac3to didn't accept the DTS file extracted by DTSParser, either. I'll implement a workaround for such broken DTS files into the next eac3to build. So the next eac3to build will be able to detect the DTSParser extracted DTS file as: Code:
DTS, 5.1 channels, 0:03:25, 20 bits, 1235kbps, 44.1khz |
22nd December 2008, 13:27 | #7485 | Link | |
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The core, or a downmix to 5.1 reencoded to a standard DTS (lossy), don't have any bitdepth, the samples are stored in frequency domain with a equivalent precission to 20-24 bits in uncompressed format, not matter if the source is 16 or 24 bits. A -16 switch with a dts output (or other lossy format) have not sense.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 22nd December 2008 at 13:32. |
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22nd December 2008, 17:30 | #7486 | Link | |
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If I can say bit depth is something equivalent to resolution of the sample stored. If the source is 16bit then while re-encoding if it is re-encoded as 24bit would this mean the encoder is filling up the missing data.
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What I am trying to do is to keep everything at the same except downmix to 5.1ch keeping every other same as possible. edit: @madshi it would be great if there was an option to prevent patching and let the default be that the audio be patched. Last edited by krosswindz; 22nd December 2008 at 17:43. |
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22nd December 2008, 18:30 | #7487 | Link | |
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For a lossy output the best procedure is, like madshi know, decode to high resolution (64 or 32 bit float) make functions (mix, speed, resample, ..) with max resolution and, at end, down to the max resolution supported by the encoder (24 bit int for Surcode DTS, 32 bit float for NeroAacEnc, ...). In lossless formats the precision is know by the bitdepth, in lossy formats by the bitrate (forget the bitdepth).
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22nd December 2008, 19:10 | #7488 | Link |
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its basically that the sound quality in it remains at 16-bit, but it seems like the way it is stored can only be 24-bit in that case. so theres no downmixing or artificial upgrading of the original quality, it remains untouched, its only the 'package' which transports that sound content which gets changed. but apparently it cannot be made another way.
(correct me if im wrong) |
22nd December 2008, 20:03 | #7489 | Link |
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madshi
The video runs apparently OK, but the audio track repeats the opening theme over and over and doesn't progress to the dialog etc. So it is an odd one. In a seamless branching title, what is the maximum you allow for/handle? |
22nd December 2008, 23:48 | #7490 | Link | |||
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I remember that there was a Blu-Ray disc which really had a broken audio track just like you describe it. Are you sure that eac3to borked up this audio track? I rather guess that the audio data in the m2ts files is broken. However, IIRC there's another audio track which is working fine. So please check all audio tracks. You may find one which is correctly working... |
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23rd December 2008, 01:30 | #7491 | Link |
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@madshi/tebasuna51 thanks for the explanation, I didnt know that the DTS "bitdepth" header field was only for informational purpose. I know have a better understanding of what you mean.
@madshi: Really love your tool. Is there any future plan for supporting DTS pro series encoders? |
23rd December 2008, 01:43 | #7492 | Link | |
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It is up to the capabilities of mostly decoder, plus encoder. I have lots of dts samples sound better than the original ac3s. Edit: Before getting the objections, i should say: Why do you use an Hi-End CD Player while you can listen the same CD on an ordinary player? Because of its HW decoder (plus of its DAC); correct? So if you can match the Hi-Fi decoder filter with an appropriate encoder filter; why not? This is why i said "decoders are more important than the encoders in transcoding" before... And madshi knows what they are and selected those Hi-Fi decoders as default in his tool... So extracting wavs using an Hi-Fi decoder and re-encoding them with an external Pro encoder would be the best choice. _ _ _ _ _ _ Last edited by rica; 23rd December 2008 at 03:29. |
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23rd December 2008, 03:26 | #7493 | Link | |||
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Well you can filter noise or modify anything wrong in the original. But, with a correct source, isn't a myth.
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23rd December 2008, 04:48 | #7495 | Link |
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Sorry if this has been asked before or if it is a stupid question :
I have been using eac3to to extract video/audio/subs from bluray disks. I noticed something though and I have a question. What exactly does the removal of dialog normalization do? I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything. If I fast forward at like 1.5x realtime where you can still hear the audio its normal level, but as soon as you hit play again it drops the audio down again. I blamed a PS3 firmware update for awhile but I exchanged my PS3 for a panasonic DMP-BD35K bluray player and it has the same behavior. I finally got around to trying the -keepDialnorm and now I can hear the audio with my receiver's volume at a normal range. What exactly is removing the dialog normalization buying me and why is it not recommended to use the -keepDialnorm switch? Is there a way to extract the ac3 audio that already has had its dialog normalization removed from my dvd9s and do the reverse process that removing the dialog normalization does? Anyone else have/seen this issue? Perhaps its my receiver? (Yamaha HTR-5940) Last edited by alc0re; 23rd December 2008 at 05:43. |
23rd December 2008, 10:30 | #7497 | Link | ||
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No, it is not a myth. It's a technical fact that every encoding process to a lossy format throws away information. There may be situations where transcoding from lossy to lossy can result in an improvement. E.g. (as tebasuna51 said) if you post process the audio data in a good way. Or e.g. if one of the decoders in your receiver is broken. But these are really exceptions. It's a fact that every straight transcoding from lossy to lossy format loses audio information. Dialnorm can be set to any value between 0 and 31. According to the AC3 specification both 0 and 31 means: No dialnorm processing. Now any dialnorm processing *lowers* the volume of the audio track. That means removing the dialnorm (which is what eac3to is doing) should result in *higher* volume. Currently eac3to sets dialnorm to 0. Unfortunetely dialnorm set to 1 means lowering volume a lot. So incorrectly working decoders might think that a dialnorm value of 0 means even lower volume than dialnorm 1. But the documentation clearly states that a dialnorm value of 0 shall be treated as "no dialnorm processing" (which means max volume). And all the PC AC3 decoders correctly see value 0 as "dialnorm processing deactivated". Quote:
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23rd December 2008, 10:42 | #7499 | Link | ||||||
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