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
