On Thu, May 08, 2014 at 02:25:40PM +0200, Niels Möller wrote: > [email protected] (Niels Möller) writes: > > > It seems to give reasonable output for the "Master Audio 7.1.dts" file, > > except that the LFE channel is messed up in some places. > > I suspect this is not a bug in my code... > > Let's I look at one sample where there's a large error, index 461360. > For the reference flac file, I have sample values > > -134990 -39862 -1678929 -248163 -116804 -112124 -1656 -34272 > > With avconv master I get > > -95623 -28254 -1187410 -159055 -83522 -103577 > > This is the core channels only, downmix version of the 7.1 data, and > without applying any lossless residuals. The inverse downmix scales all > channels by sqrt(2), and then undoes some mixing of the last two > surround channels. Ignore the surround stuff, and just scale the first > four channels. This gives > > -135231 -39959 -1679253 -224938 > > with errors > > -241 -97 -324 23225 > > So the final channel, the LFE, gets a much worse error than the three > normal channels. In dcadec.c, there's a comment > > /* Generate LFE samples for this subsubframe FIXME!!! */ > > which looks like it predates the libav/ffmpeg fork. Does anyone remember > what the problem was?
It comes from the original libdts (now known as libdca) so no. But IIRC D** decided to use IIR-based interpolation for LFE in case it's not core-only so the problem lies in Calabasas, California. _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
