[opus] Antw: Re: Antw: [EXT] Opus merging streams
andrew at safehear.fr
Wed Apr 6 10:55:17 UTC 2022
Thanks for the answer and thanks for the warning about complex
algorithms as I was starting to check how conferencing application were
My signal will approximately be the same on each channel as I only send
voice data. Also, I already send each channel on a different queue as to
decode and synchronize them correctly and independently.
So if I resume, it's better to decode, then use the sum of each channel
scaled by the square root of the number of channels.
Is that right ? Or is there some kind of trickier and better algorithms
that I'm not aware of ?
Sorry if my questions seems childish, I don't have that much experience
in audio based algorithms and it's kind of fascinating how much it can
be tricky to adjust correctly some parameters or even find the correct
algorithms for a certain use case.
Le 06/04/2022 à 12:37, Sampo Syreeni a écrit :
> On 2022-04-06, Andrew Sonzogni wrote:
>> I've used this one but it's not THAT good: (chan1 + chan2 + chan3) / 3
>> The output signal may peak or be buggy some times.
> That is an age old question. In general, if you assume the channels
> are statistically independent, and you want each channel to contribute
> equally and maximally, you scale each by the square root of the number
> of channels. That solution mathematically mostly goes through under
> the central limit theorem, and exactly so if each of the sources are
> Gaussian distributed. In general you can assume that with sampled real
> life signals, because of reverberation and another application of the
> central limit theorem, but in case of "digital" signals and in
> particular if you sum many channels which are coherent (contain more
> or less the same stuff), you'll fall off the ladder.
> Personally I'd advocate for a pure sum with no scaling. But that sort
> of thing is kind of opinionated: it only really works if you have
> dynamics to spare and also an absolute scaling of the incoming
> waveforms over your whole signal chain. That is, each of the source
> waveforms at digital full scale are supposed to represent an absolute
> amplitude of, say, 96 dB(Z).
> When you do this sort of a gain architecture, anything coming in goes
> out just as it was, unless modified. If it comes in at 24-bit
> fullscale, it'll literally break not your ears but also your windows.
> So you'll never actually put in anything near fullscale. You *might*
> put in combinations of sounds which then -- if incoherent with each
> other -- will usually scale by the square root law. Which is fine, as
> long as you keep your output fullscale to something which will break
> stuff. Say, do 24 bits and set the digital fullscale at something like
> 130dB SPL in your monitors.
> This by the way is how it was and is done in cinematic audio editing.
> The full dynamics shatter your bones and ass, in absolute SPL range.
> Anything lesser is just added linearly into the mix. The added bonus
> of this setup is that you actually get to use amplitude as an extra
> variable which isn't normalized away by someone turning a knob or
> putting in an extra compressor in the signal pathway. You can actually
> compose with volume, instead of assuming everybody takes your dynamics
> away at will. (Obviously in many cases they will do just that, but
> trust you me, composing in absolute amplitude leaves *so* much more
> dynamics and nuance for the rest of them idjits to churn up.)
>> For your information, I'm using an ARM M4F with Opus configured like
>> this (40ms, 16kHz, 16 bitrate, 0 compres).
> Just unpack and sum. You really, *really* don't want to see how an
> optimized algorithm for something like this would be. It'd take a
> cross-compile into a fully functional language, some heady CS-minded
> meta-coding concepts, and a week of computation to find the precise
> weaving-in of the functional forms. At best. It's nowhere *near* the
> best use of your time, especially since you can optimize the simple
> decode-sum-recode(?) cycle already.
> As an example of the latter kind of optimization, if you have a number
> of unsynchronized opus streams decoding, you can rather easily land
> the output in circular buffers of double the block size. The code for
> updating such a buffer is classic, as is reading from it with
> arbitrary delay upto half the buffer size. The code to SIMD-vectorize
> arbitrary byte spanning reads from such a buffer is also classic, and
> can be found even from the better implementations of memcpy(). The
> only thing you need to do then is to weave your code together so that
> it more or less advances a word/cacheline/whatever-unit a time in each
> buffer, linearly, and sums over the inputs synchronously, without
> unduly polluting your CPU's data cache. It's nasty even then, but
> highly doable, and that's about as fast you can go with any library
> imaginable (any deeper algorithm would require recoding the Opus
> library in a functional language, and weaving its internals together
> with your buffering algo).
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