On Sat, Nov 03, 2007 at 01:40:37PM +0100, Karel Kulhavy wrote: > On Wed, Oct 31, 2007 at 05:48:20PM +0100, Alexandre Ratchov wrote: > > On Wed, Oct 31, 2007 at 11:23:31AM -0400, Nick Guenther wrote: > > > On 10/31/07, Brian A Seklecki (Mobile) > > > <[EMAIL PROTECTED]> wrote: > > > > Some *BSD systems are adjusting PCM driver support to allow multiple > > > > process to open /dev/dsp / /dev/audio multiple times in-exclusively, > > > > mitigating the needs for piss-poor software API multiplex'ing solutions > > > > a-la ARTS/ESD. > > > > > > Oh awesome! Is /Open/BSD one of those? > > > > > > > no; character devices (such as /dev/audio) keep per-unit state > > (encoding, rate, ...). To mix multiple audio streams per-stream > > state must be kept. That's why arts/esd/jack/... exist. > > You don't need arts/esd/jack because of this. This can be solved in kernel. > The kernel opens the audio device for the highest common sampling rate > from those requested, or, if the rate cannot be switched without an > audible glitch, for the highest hardware available (48 or 96kHz, or > user-configured if it should be too much burden). > > Then if the kernel has it 48 and the app opens 44.1, the kernel resamples. > According to Nyquist theorem, you first need to emulate the reconstruction > lowpass filter with 22.05kHz programatically, and then resample the > output stream at 48kHz and send it out.
that's not exact; you're talking about upsampling. So you need to resample first and then you apply the pass-low filter. But we're going off-topic. > This requires an intermediate stream at least common multiple, > for this ugly case it's 2352 kHz. But you actually don't have to > shove data at 2352 kHz sampling rate in the kernel, since you use > only every 49th sample for the output. So some nifty math or > multipass filter does the job, for the expense of your brain > exploding. > > Now how the lowpass is done. It has to be really sharp otherwise you get > an 8-bit-era-like ringing in the sound. And it must not have large delay > (you don't want to hear the explosion in your Quake a second later) and > also not computationally expensive. > > The suitable type of filter is called finite impulse response (FIR) and > it's just a naiive convolution with a short kernel. Now how to calculate > this kernel to get the best response? You make your kernel and imagine > it's cyclically wrapped. Then you calculate through FFT the ideal response > into it - that's perfectly sharp. But now since the response won't be > cyclically wrapped but occurs just once in the time and have zero strecthing > into both infinities, we have to fix this. > that isn't necessary because we don't need FFT to obtain filter coefficients (they can be determined at initialization). But I'm again off-topic. > You take a Hann window http://en.wikipedia.org/wiki/Hann_window and apply that > and you got it. Hann window is just one cycle of a sine wave plus minus some > pushing around. You don't even need a sin for this, you can calculate it with > complex multiplication of one pre-calculated complex number. > > How long do you make your kernel? The longer the kernel, the more > computationally intensive, but the sharper the transition so you lose less of > the high frequencies. > afaik, most of us have followed math curses too and understand pretty well Fourier transforms and fitlering and we know how to implement this stuff too. I'm not against resampling and mixing but audio(4) driver is the wrong place for it. -- Alexandre
