Uhh, can't replicate bug here (freshly built ffmpeg, just applied this patch only), (md5 for file = 473edd68b91123c3a9c1825271012357). tried other files and they encode and play fine. Spectrum also looks fine. I also tested other random files out of the samples and they all seem file. Know of any other problematic files?
On 14 April 2015 at 16:51, Claudio Freire <klaussfre...@gmail.com> wrote: > On Mon, Apr 13, 2015 at 8:33 PM, Rostislav Pehlivanov > <atomnu...@gmail.com> wrote: > > This commit implements the perceptual noise substitution AAC extension. > This is a proof of concept implementation, and as such, is not enabled by > default. This is the second revision of this patch, made after some > discussion via non-public email due to a mistake. Any changes made since > the first revision have been indicated. > > > > In order to extend the encoder to use an additional codebook, the array > holding each codebook has been modified with two additional entries - 13 > for the NOISE_BT codebook and 12 which has a placeholder function. The cost > system was modified to skip the 12th entry using an array to map the input > and outputs it has. It also does not accept using the 13th codebook for any > band which is not marked as containing noise, thereby restricting its > ability to arbitrarily choose it for bands. The use of arrays allows the > system to be easily extended to allow for intensity stereo encoding, which > uses additional codebooks. > > > > The 12th entry in the codebook function array points to a function which > stops the execution of the program by calling an assert with an always > 'false' argument. After a discussion, it was pointed out in an email > discussion with Claudio Freire that having a 'NULL' entry can result in > unexpected behaviour and could be used as a security hole. There is no > danger of this function being called during encoding due to the codebook > maps introduced. > > > > Another change from version 1 of the patch is the addition of an > argument to the encoder, '-aac_pns' to enable and disable the PNS. This > currently defaults to disable the PNS, as it is experimental. The switch > will be removed in the future, when the algorithm to select noise bands has > been improved. The current algorithm simply compares the energy to the > threshold (multiplied by a constant) to determine noise, however the > FFPsyBand structure contains other useful figures to determine which bands > carry noise more accurately. > > > > Finally, the way energy values are converted to scalefactor indices has > changed since the first commit, as per the suggestion of Claudio Freire. > This may still have some drawbacks, but unlike the first commit it works > without having redundant offsets and outputs what the decoder expects to > have, in terms of the ranges of the scalefactor indices. > > > > Some spectral comparisons: https://0x0.st/T7.png (original), > https://0x0.st/Th.png (encoded without PNS), https://0x0.st/A1.png > (encoded with PNS, const = 1.2), https://0x0.st/Aj.png (spectral > difference). The constant is the value which multiplies the threshold when > it gets compared to the energy, larger values means more noise will be > substituded by PNS values. Example when const = 2.2: https://0x0.st/Ae.png > > > > Comments, tips, feedback and criticism are welcome. > > > This commandline: > > /home/claudiofreire/src/ffmpeg/ffmpeg -i > /home/claudiofreire/tmp/audiosamples/ffsamples/aac/ct_faac-adts.aac > -strict -2 -c:a aac -b:a 48k -cutoff 22050 -f adts -aac_pns 1 -y > test.adts > > Produces: > > Assertion diff >= 0 && diff <= 120 failed at libavcodec/aacenc.c:398 > Aborted > > This will probably relate to the fact that noise scalefactors need to > be clamped to a range of SCALE_MAX_DIFF (though independently of > regular scalefactors). > > I would suggest that, at the end of twoloop, you measure the minimum > noise scalefactor, and clamp in the range minscaler to > minscaler+SCALE_MAX_DIFF. > > You can get the ffsamples folder by configuring with > --samples=/home/claudiofreire/tmp/audiosamples/ffsamples (or whatever > path works for you), and then make fate-rsync > _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
