Re: [asterisk-users] Silk for Free
On Fri, 06 Mar 2009 02:15:12 +0800, Steve Underwood wrote: Steve Underwood wrote: They might be doing some kind of fake bandwidth expansion. You can't create something out of nothing, and make the narrowband voice more intelligible, but you can make it sound pretty nice. Jean-Marc Valin (he of speex fame) has a very impressive expander he is working on at the moment. Someone tells me Polycom definitely do some form of bandwidth expansion, though its a lot less realistic than Jean-Marc technique. What do you mean by bandwidth expansion? The term expansion in audio is most commonly used in reference to dynamic range. Expanders (and their inverse, compressors) vary the level of a signal dynamically to effect changed signal.noise ratio. How does one effect frequency domain extension? That seems counterintuitive to me. Frequency response (pass-band) is a function of sampling rate (see Nyquist) and anti-aliasing filtering. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Wilton Helm wrote: 12kHz isn't really enough for high quality voice, and the extra bit rate needed to push the bandwidth to 15kHz is small. Also, a deep man's voice looses something when you cut off at 70Hz. I'm not sure that this isn't stretching things a bit. There are no handsets or headsets (AFAIK) that can do justice to 50 KHz and probably most speakers attached to a PC can't. Likewise, while a deep male voice can go below 70 Hz, few transducers can do justice to those frequencies, either. I don't think the attempt is to reproduce a symphony. The extra bandwidth (even if it is minor) would be hard to justify if one needed $500 speakers to benefit from it. While a number of people might be able to tell the difference in an A B comparison, I suspect few would notice it without direct comparison. I also suspect Skype is correct in that the majority of people, listening to it on typical hardware would like additional low frequencies less than without because of things like distortion in the transducer. So, your approach is to base a high quality codec around the crappiest transducers it will ever meet? :-\ If you want to define standards that stick around, you try to make them work beyond the junk you see today. Making a wideband handset isn't hard. Its a couple of dollars to make a really good one, not $500. People just haven't bothered, because wideband telephony hasn't really existed to make use of it. A 20 cent handset saves them money, but $2 isn't a lot to pay for a handset in a $100 phone. Getting the bandwidth above 3 KHz at the top will improve intelligibility, but somewhere between 5 and 10 KHz that reaches a point of diminishing returns. Likewise, extending the low end below 300 Hz will help naturalness, but that also reaches diminishing returns somewhere around 100 Hz unless all the pieces are very high quality (from the mic to the speaker). It seems to me that they have exceeded those realities by a comfortable margin, which is generally what good engineering is all about. Skype say they are sampling at 24k for their widest bandwidth, so they won't really get 12kHz. More like 11kHz, I guess. Extending voice up to 7kHz pretty much fixes all the intelligibility problems inherent in normal narrowband telephony. However, the difference in sound quality between 7kHz bandwidth and 11kHz is big, and going up to 15kHz adds significantly more. It takes very few extra bits to extend the frequency range. Doubling the bandwidth does not double the bit rate with modern compressed codecs. Good engineering of standards is about building them for the future. Cutting off the bass at 70Hz is far less of a limitation than cutting off the high end at 11kHz, but why do it in the codec? Why not leave the transducers and their amps to do what they can? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Steve Underwood wrote: Good engineering of standards is about building them for the future. Cutting off the bass at 70Hz is far less of a limitation than cutting off the high end at 11kHz, but why do it in the codec? Why not leave the transducers and their amps to do what they can? Along those lines, I've mentioned here before that my wideband Polycom IP650 actually sounds substantially better with G.711 ulaw media streams than any previous phone I've had, and I've had high quality phones before. I can only assume this is due to the higher quality speakers, amps, microphones, and acoustic effects of the phone design. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
On Thu, Mar 5, 2009 at 5:09 PM, Kevin P. Fleming kpflem...@digium.com wrote: Along those lines, I've mentioned here before that my wideband Polycom IP650 actually sounds substantially better with G.711 ulaw media streams than any previous phone I've had, and I've had high quality phones before. I can only assume this is due to the higher quality speakers, amps, microphones, and acoustic effects of the phone design. I've long suspected and we've discussed it on VUC many times, that Polycom uses what used to be know as pre-emphasis and de-emphasis like techniques to make them sound so good. They really are great, even on th elow end, and with g722 enabled, they get even better. We are currently testing a sub branch of the weekly Friday conference using the ZipDX HD conference bridge and it does make a difference. Maybe you'll drop in sometime soon Kevin? Otherwise, probably see you in Rostock in May. Best, randy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Kevin P. Fleming wrote: Steve Underwood wrote: Good engineering of standards is about building them for the future. Cutting off the bass at 70Hz is far less of a limitation than cutting off the high end at 11kHz, but why do it in the codec? Why not leave the transducers and their amps to do what they can? Along those lines, I've mentioned here before that my wideband Polycom IP650 actually sounds substantially better with G.711 ulaw media streams than any previous phone I've had, and I've had high quality phones before. I can only assume this is due to the higher quality speakers, amps, microphones, and acoustic effects of the phone design. They might be doing some kind of fake bandwidth expansion. You can't create something out of nothing, and make the narrowband voice more intelligible, but you can make it sound pretty nice. Jean-Marc Valin (he of speex fame) has a very impressive expander he is working on at the moment. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Steve Underwood wrote: Kevin P. Fleming wrote: Steve Underwood wrote: Good engineering of standards is about building them for the future. Cutting off the bass at 70Hz is far less of a limitation than cutting off the high end at 11kHz, but why do it in the codec? Why not leave the transducers and their amps to do what they can? Along those lines, I've mentioned here before that my wideband Polycom IP650 actually sounds substantially better with G.711 ulaw media streams than any previous phone I've had, and I've had high quality phones before. I can only assume this is due to the higher quality speakers, amps, microphones, and acoustic effects of the phone design. They might be doing some kind of fake bandwidth expansion. You can't create something out of nothing, and make the narrowband voice more intelligible, but you can make it sound pretty nice. Jean-Marc Valin (he of speex fame) has a very impressive expander he is working on at the moment. Someone tells me Polycom definitely do some form of bandwidth expansion, though its a lot less realistic than Jean-Marc technique. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Dean Collins wrote: http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news any thoughts? They have said it will be royalty free, but they have said little else. From discussions with Skype people in the last few days they seem very reluctant to hand out source code, so it looks like they will provide binary blobs for whatever platforms they choose to support. They are clearly eager to get Skype broadly connected to corporate networks, but if they don't get this codec into a broad range of phones its a waste of time. Transcoding looses too much quality.. If they don't hand out the source, or at least provide a rigorous spec, I don't think this will fly. Even rigorous specs aren't really enough. Pretty much all modern codecs are defined by their reference implementation. The bit rate is supposed to dynamically adapt to network conditions, when the code is used in conjunction with a suitable network performance monitor. Exactly what those bit rates are, however, still seems to be a mystery. They claim audio up to 12kHz, and specifically say they are suppressing the bass end below 70Hz as it just sounds nasty. That's sad. 12kHz isn't really enough for high quality voice, and the extra bit rate needed to push the bandwidth to 15kHz is small. Also, a deep man's voice looses something when you cut off at 70Hz. You really want the bass to extend to 40Hz or 50Hz. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
Dean Collins wrote: http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news any thoughts? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). Cheaper to give away for hopes of proliferation what you've already implemented versus having someone else get theirs proliferated and popular first and then you are strapped with the cost of implementation of someone else's popular and free codec? -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
BJ Weschke wrote: Cheaper to give away for hopes of proliferation what you've already implemented versus having someone else get theirs proliferated and popular first and then you are strapped with the cost of implementation of someone else's popular and free codec? Polycom's Siren7 (G.722.1) is already 'free' under basically the same terms and is being implemented in endpoints currently. Siren14 (G.722.1 Annex C) is in essentially the same situation, and provides even higher audio bandwidth. The selling points for SILK are primarily the network bandwidth optimization features, but as Steve Underwood already posted, that requires the implementation to have access to network monitoring information so that it can proactively make bandwidth changes (as opposed to just waiting until the packet loss reaches unacceptable levels and audio quality is already suffering). It will be interesting to see where this goes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
12kHz isn't really enough for high quality voice, and the extra bit rate needed to push the bandwidth to 15kHz is small. Also, a deep man's voice looses something when you cut off at 70Hz. I'm not sure that this isn't stretching things a bit. There are no handsets or headsets (AFAIK) that can do justice to 50 KHz and probably most speakers attached to a PC can't. Likewise, while a deep male voice can go below 70 Hz, few transducers can do justice to those frequencies, either. I don't think the attempt is to reproduce a symphony. The extra bandwidth (even if it is minor) would be hard to justify if one needed $500 speakers to benefit from it. While a number of people might be able to tell the difference in an A B comparison, I suspect few would notice it without direct comparison. I also suspect Skype is correct in that the majority of people, listening to it on typical hardware would like additional low frequencies less than without because of things like distortion in the transducer. Getting the bandwidth above 3 KHz at the top will improve intelligibility, but somewhere between 5 and 10 KHz that reaches a point of diminishing returns. Likewise, extending the low end below 300 Hz will help naturalness, but that also reaches diminishing returns somewhere around 100 Hz unless all the pieces are very high quality (from the mic to the speaker). It seems to me that they have exceeded those realities by a comfortable margin, which is generally what good engineering is all about. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silk for Free
http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio _codec.html?tk=rss_news any thoughts? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silk for Free
On Tue, Mar 03, 2009 at 11:51:56PM -0500, Dean Collins wrote: http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio _codec.html?tk=rss_news any thoughts? What patents does Skype have that are required for implementing this codec? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
