Quote from: googlebot on 29 March, 2010, 12:47:07 PMThe spec's bad choice of the very long minimum block size of 256 samples makes it pretty hard to improve things as transient handling above a certain point. Even if you employ the best programmers on earth.That would be one of those assertions that needs to be supported by some real world data. Where's the AC3 listening test performed with the latest-greatest encoders from Dolby?Otherwise, you're basically messing with TOS 8.
The spec's bad choice of the very long minimum block size of 256 samples makes it pretty hard to improve things as transient handling above a certain point. Even if you employ the best programmers on earth.
Once your partition length is above that, the distribution of quantization noise is not just a complex, but solvable, technical problem anymore, but impossible, as I understand it.
Do you have any reference for 2x128 blocks? I could not find anything.
In the AC-3 transform block switching procedure, a block length of either 512 or 256 samples (time resolution of 10.7 or 5.3 ms for sampling frequency of 48 kHz) can be employed.
Normal blocks are of length 512 samples. When a normal windowed block is transformed, the result is 256 unique frequency domain transform coefficients. Shorter blocks are constructed by taking the usual 512 sample windowed audio segment and splitting it into two segments containing 256 samples each. The first half of an MDCT block is transformed separately but identically to the second half of that block. Each half of the block produces 128 unique non-zero transform coefficients representing frequencies from 0 to fs/2, for a total of 256. This is identical to the number of coefficients produced by a single 512 sample block, but with two times improved temporal resolution. Transform coefficients from the two half-blocks are interleaved together on a coefficient-by-coefficient basis to form a single block of 256 values.
Well Arnold, we've got to make do with the data we have here. The reason that I started this thread was precisely because there seems to be so little information about audio quality of the top AC3 implementations. And if the companies won't offer their encoders for ABX tests, the next best thing we can do is reason through the limits of the format.
[...] That should be half the maximum temporal resolution of AAC at the same sample rate
or does AAC use blocks of 128 spectral coefficients for blocks of 128 samples?
Quote from: sauvage78 on 19 February, 2010, 02:32:48 PMYou can hardly rip a blu-ray with a mono-core barton sempron 3000+ so I gave up anyway.You don't need a powerful computer, just a big hard drive and the tools (which I think would violate TOS if I mentioned). There's no transcoding necessary, and there are now many (though mostly unreliable due to bugs) HDMI solutions that will bitstream the bluray codecs even from individual files.
You can hardly rip a blu-ray with a mono-core barton sempron 3000+ so I gave up anyway.
Quote from: googlebot on 30 March, 2010, 07:56:14 AMAC3 is unable to satisfy such a capability with 5.3 ms long samples. As said, no programmer or "latest-greatest encoder" from Dolby can go beyond that.This is nonsense. You're implying that a codec using a filterbank with 256 banks isn't able to represent arbitrarily small time shifts. This is wrong for the same reason why this is wrong: people claiming -- just because PCM is sampled -- it's not able to represent sub-sample time shifts. Yes it is. And so is AC3. Sampling just limits the signal bandwidth, nothing else.
AC3 is unable to satisfy such a capability with 5.3 ms long samples. As said, no programmer or "latest-greatest encoder" from Dolby can go beyond that.