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Old 15th August 2014, 04:13   #12741  |  Link
Thunderbolt8
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whats the correct file format to choose when demuxing DVB subtitles? I tried .srt and .sup, but both gave me errors.
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Old 15th August 2014, 12:41   #12742  |  Link
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I'm not sure eac3to handle DVB-SUB (bitmap based format) but Subtitle Edit does (it needs the TS file).
If you are talking about teletext subtitles format, TSDoctor, CCExtractor and maybe VideoRedo and ProjectX can convert it in srt (they also need the TS file).

Last edited by Music Fan; 15th August 2014 at 12:58.
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Old 15th August 2014, 12:43   #12743  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
I say because the samples in frequency domain are better calculated with a source more precisse.
I had understood but it seems contradictory with the fact that the precision is less than 16 bits in lossy formats.
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Old 15th August 2014, 23:09   #12744  |  Link
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Quote:
Originally Posted by Music Fan View Post
I had understood but it seems contradictory with the fact that the precision is less than 16 bits in lossy formats.
If you original source is 24 bit depth, the encoder will can calculate better because precision is greater than 16 bit depth source.

This and you say is contradictory, but not tebasuna51 says.
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Old 15th August 2014, 23:23   #12745  |  Link
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What is contradictory in my message ? I just asked questions.

Quote:
Originally Posted by tebasuna51 View Post
the samples in frequency domain have a precission equivalent to a bitdepth of 20 bits.
Quote:
Originally Posted by tebasuna51 View Post
the precission is always less than 16 bits in lossy formats
Logical for you ?

Last edited by Music Fan; 15th August 2014 at 23:26.
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Old 16th August 2014, 08:20   #12746  |  Link
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The concept of bit depth just doesn't really translate to the frequency domain transforms used by the lossy codecs. Trying to speak about it will always sound contradictory since it just doesn't fit.
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Old 16th August 2014, 08:29   #12747  |  Link
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When a lossy encoder make the change between time domain to frequency domain not all components fit in the bitrate assigned and discard the less important (for low volume or high frequency), for that you can't recover the original precission.

For example the DTS-MA make the 'core' like lossy standard DTS with a CBR bitrate and all components than not fit are stored in a subframe VBR, now we can recover the original precission.
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Old 20th August 2014, 23:13   #12748  |  Link
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Here's why input bit depth matters. Obviously this is very artificial to make a point, but the higher the bitdepth, the less noise gets sprayed into the rest of the spectrum, which allows the encoder to encode the sound more accurately in less bits.

(One thing I have to comment on, everyone is assuming that all codecs use float internally, but most don't. Quite a few use DCTs/MDCTs optimized specifically for the bit-depth they internally operate at, which is usually 16-bit or 24-bit, though a few implementations can optionally use float. By the same measure, the specs usually only specify output at a certain bit-depth with a certain internal bit-depth; decoders going beyond that can't be "compliant" even if they can be theoretically slightly higher quality. Personally, I feel that the need for optimized audio algorithms is past, when I can do 100x realtime transcoding, and an all-float chain is important, even if it's not perfectly to-spec.)
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Old 21st August 2014, 10:16   #12749  |  Link
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Thanks for this explanation.
You didn't write whats is the difference between these 2 images, I guess the source of the first is 24 bit and 16 for the second, right ?
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Old 21st August 2014, 10:39   #12750  |  Link
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It doesn't even matter much how many bits of resolution there is exactly in these waveforms. May it be only 8 bit on the right side. But what you see is the bad side effect of "quantization". If samples have too little resolution, the conversion to a frequency spectrum will have artefacts. Backwards the same problem: If the frequency spectrum parameters have too low precision, the reconstructed waveform will have artefacts. Lossy compression is mostly about reducing the precision of the frequency spectrum, though, so there is much effort to try to make the resulting artefacts unobvious, or at least not so annoying...
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Old 21st August 2014, 10:53   #12751  |  Link
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The point foxyshadis is making is that the higher the source quality is the easier it is for the lossy encoder to produce a (reasonably) good quality encode with a (relatively) low bitrate. The situation is very different when comparing lossy and lossless encoders: With lossy encoders it's a good idea to feed the encoder the best quality you have. With lossless encoders it's the opposite situation: There higher bitdepths cost additional bitrate. The reason for that is the lossless encoders are not allowed to throw any data away. They have to keep every bit, even noise. And higher bitdepth signals have somewhat random values in the very lowest bits, which are hard to perfectly remember. A lossy encoder can simply throw those away, or encode a signal which only mostly resembles the original. A lossless encoder is not allowed to do such things.

Summary: Feeding a higher quality source into the encoder always improves the overall audio quality, for both lossy and lossless encoders. However, for lossy encoders a higher quality source actually improves the compression efficiency. For lossless encoders a higher quality sources increases the file size instead.
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Old 21st August 2014, 22:05   #12752  |  Link
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Whoops, I should have labeled them; the first is 16-bit and the second 8-bit. Real-life noise is more complicated, and obviously float vs 24 vs 16 is much smaller than 16 vs 8, but this was mostly to make the differences visible. Lower bit-depth always leads to spectral leakage. Of course spectral noise (and regular noise) from lossy compression is even worse, but you may as well not compound the problem when transcoding if given the chance.
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Old 24th August 2014, 12:11   #12753  |  Link
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RedirectStandardError

I've written a vb class to run the eac3to.exe.

In this class I have redirected the error output, but what I get is a truncated string of the full error.

Is there any error documentation/examples available to help me troubleshoot the problem?
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Old 25th August 2014, 10:21   #12754  |  Link
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As far as I remember, eac3to has a strange technique of formatting the text output, using many backspace chars instead of a carriage return. Have a look at the output redirected to a text file in a CLI window, using a hex editor or text viewer supporting non-printable characters.
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Old 25th August 2014, 13:56   #12755  |  Link
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Is there a way to avoid the automatic "Patching bitdepth to 24 bits..." procedure when I use the -core option extracting DTS core from DTS-HD ?

Because sometimes DTS-HD is 16bits and I see this phrase during processing.

Does this conversion lead to better, worse or equal results compared to leaving 16bit in DTS-HD stream ?

Thanks!
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Old 25th August 2014, 15:46   #12756  |  Link
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The core is lossy audio, it has no bitdepth, even if the lossless DTS-HD was 16-bit.
Lossy audio should be decoded to an as high bitdepth as the decoder will give you, and thats with ffmpeg 32-bit floating point, or with the ArcSoft decoder 24-bit integer.
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Old 25th August 2014, 18:41   #12757  |  Link
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But I don't have ArcSoft DTS-HD decoder installed or any other DTS-HD decoder.

Also, I think that the -core process doesn't involve any decoding of DTS-HD stream.

It just extracts the DTS core from DTS-HD without decoding.
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Old 25th August 2014, 22:45   #12758  |  Link
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Quote:
Originally Posted by LigH View Post
As far as I remember, eac3to has a strange technique of formatting the text output, using many backspace chars instead of a carriage return.
Is there any documentation for eac3to and error handling?
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Old 26th August 2014, 02:23   #12759  |  Link
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Quote:
Originally Posted by NikosD View Post
But I don't have ArcSoft DTS-HD decoder installed or any other DTS-HD decoder.

Also, I think that the -core process doesn't involve any decoding of DTS-HD stream.

It just extracts the DTS core from DTS-HD without decoding.
If you're going to throw away the HD correction, let the core be 24-bit. It's only labeled 16-bit because the lossless correction is 16-bit, it'll sound fine at 24-bit.
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Old 26th August 2014, 05:04   #12760  |  Link
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OK.

Let me rephrase my question.

Is the app really doing something during "Patching bitdepth to 24bits..." process ?

If not, why the message ?

If yes, what's the meaning of bitdepth in a lossy (DTS) format and what is technically doing ?

Thanks!
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