[pulseaudio-discuss] Re(surrect): [PATCH] vala: Add bindings for libpulse-simple
Hi, There has been a proposed patch sent to this list [1] which has been accepted in email [2] but never committed. Could someone with commit access please commit it? Thanks, Balint [1] http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-January/012731.html [2] http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-February/012896.html ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] On scaling the HRIR in module-virtual-surround-sink
Hello Alexander, On Tuesday 11 March 2014, 15:00:31, Tanu Kaskinen wrote: On Sun, 2014-03-09 at 00:57 +0600, Alexander E. Patrakov wrote: Hello. [Do not blindly apply patches from this e-mail! They mutually exclusive, and I don't have a firm opinion which one of them is correct.] Today I tried to improve the existing module-virtual-surround-sink (but the same issue also affects the IIR-based rewrite that is still sitting on my laptop). The problem is: the current normalization code does not do what it is designed to do. The module clips on some testcases. Let me copy-paste the problematic code for easy discussion. snip Thoughts? My thoughts: The scaling should be put in its own function with a comment that explains why the scaling is done, a high level description of how the scaling algorithm works, and a note that it's unclear whether the algorithm actually makes sense. Something like this could be included too: This algorithm doesn't pretend to be perfect, it's just something that appears to work (not too quiet, no audible clipping) on the material that it has been tested on. If you find a real-world example where this algorithm results in audible clipping, please write a patch that adjusts the scaling factor constants or improves the algorithm (or if you can't write a patch, at least report the problem to the PulseAudio mailing list or bug tracker). Would you like to write such a patch or should I do it? Or do you have an idea for a better heuristic? Regards, Ole ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] On scaling the HRIR in module-virtual-surround-sink
2014-04-02 1:48 GMT+06:00 Niels Ole Salscheider niels_...@salscheider-online.de: Would you like to write such a patch or should I do it? Or do you have an idea for a better heuristic? I do have an idea for a possibly-better heuristic (see below). However, I have caught a cold, and there is also an extraordinary number of DNS queries we (SafeDNS) get from Turkish users right now, leading to server load issues. So, my health and my work are priorities now :( So, please submit a patch. A possibly-better but untested heuristic is as follows, and I do intend to implement it later, after the Turkish YouTube/DNS situation settles. 1. Collect spectrum and inter-channel correlation data from some music DVDs, take an average. 2. Synthesize a short signal that has roughly the same characteristics. 3. Filter it through the A-weighing filter (see the transfer function in Wikipedia, it is a rational function = the filter is implementable as IIR). 4. Store the result as a short sequence of samples in PA source code. 5. When loading a filter from a wav file, apply it to the test signal from step (4). 6. Take the sum of squares of samples sent to the left headphone. This (due to A-weighing) would represent the subjective loudness of the filtered music. 7. Compare that to the sum of squares of samples sent through the conventional downmixer. Compensate accordingly by scaling the loaded filter. The result should be that the audio filtered through your module and through the conventional downmixer (in src/pulsecore/resampler.c), in average for the music material, would be of the same subjective loudness. I am not sure if that would be sufficient to eliminate the clipping, but we can always multiply the result by the fudge factor later if needed. -- Alexander E. Patrakov ___ pulseaudio-discuss mailing list pulseaudio-discuss@lists.freedesktop.org http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
