if everyone has such good ideas on how to improve it, I can't wait to see the patch. On Tue, Apr 7, 2009 at 9:42 AM, seven du <seven@idapted.com> wrote:
And, actually the value to be normalized is 32bit and which generally is a sum of two 16bit values( read and write ), so it's probably easy to exceed 32767 if both end speaking at the same time.
On Apr 7, 2009, at 12:15 PM, Nic Bellamy wrote:
Anthony Minessale wrote:
2) the macro is designed to fight clipping: When a sample exceeds the max size of the datatype we set it to the max value then cut the volume in half of that sample so it does not make too much noise. if we divide everything by 2 then all the volume would be reduced even on samples that fell inside the proper range. Hi Anthony, all, this is, in my opinion, actually going to give you far far uglier audio artefacts than simply hard limiting the samples. Rather than try and describe the resulting distortion, it's easier to just listen to it in action...
Each of the following files is about 100kb.
Original file (extracted from freeswitch-sounds-en-us-callie-8000-1.0.7.tar.gz): http://www.vadacom.co.nz/engineering/ivr-sample_submenu.wav
After 30dB gain and hard limiting samples to between -32768 and 32767: http://www.vadacom.co.nz/engineering/ivr-sample_submenu- hardlimited.wav
After 30dB gain and switch_normalize_to_16bit():
http://www.vadacom.co.nz/engineering/ivr-sample_submenu-switch_normalize_to_...
Cheers, Nic.
-- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/
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