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#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 |...

#2027: multistream_encode(_float) fails on hard cbr encodes in libopus v1.1 -----------------------------------------+------------------------------ Reporter: thinktink | Owner: jmvalin@? Type: defect | Status: closed Priority: P3 |...

...will take care > of resampling for you. Just feed in any PCM wav file and it > should be fine. > > If instead you're writing a program yourself that uses the reference > libopus API, then yes, you need to resample 44.1 kHz audio > to 48 kHz before submitting it to opus_encode(_float). > > -r > > > > ------------------------------ > > _______________________________________________ > opus mailing list > opus at xiph.org > http://lists.xiph.org/mailman/listinfo/opus > > > End of opus Digest, Vol 45, Issue 5 > ************************...

Hi, The attached patch includes a fully working patch for the resampler, including manual SSE optimizations for the single target. I've tested up and downsampling, changing the filter quality on the fly, changing sampling speed on the fly, multichannel resampling, _float and _int versions in both floating and fixed point and resetting the resampler. They all give the same results as the original one. The next step would be SSE2 optimization for quality >= 9 (using double instead of float); but I think I've done enough for now. On a sidenote; I'm mi...

Hi , I have read that it is posible to encode higher sample rates like 96 khz or 192khz? and the output is 48 khz, the resample is internally.? http://wiki.xiph.org/OpusFAQ But it is possible to encode? 44.1khz. It is resampled to 48khz or I have to make the resample by myself and then encode it with opus. thnx, arctor -------------- next part -------------- An HTML attachment was scrubbed...

Ok. I did some quality tests. First off; never do quality tests with ints. I had serious problems interpreting my results until it dawned on me that the signal differences were just 0 or 1. So, after a lot of scratching my head, these are done comparing the result from the _float versions (which is how both resamplers work internally anyway). What I did was this: Load speex_wb.wav as one large chunk of data. Pad data with as many zeroes as there are samples. Convert to long double. Use one long double FFT for the entire thing. Insert or chop off zeroes so the new length i...

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block