Re: [codec] Thresholds and delay.
Michael Knappe <mknappe@juniper.net> Tue, 11 May 2010 18:58 UTC
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From: Michael Knappe <mknappe@juniper.net>
To: "stephen.botzko@gmail.com" <stephen.botzko@gmail.com>, "bmschwar@fas.harvard.edu" <bmschwar@fas.harvard.edu>
Date: Tue, 11 May 2010 11:56:48 -0700
Thread-Topic: [codec] Thresholds and delay.
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Subject: Re: [codec] Thresholds and delay.
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Stephen, Agreed that achieving low enough latencies for sidetone perception should not be a goal of the wg, but we should be aiming if at all possible for better than 250 ms one-way delay in typical (and non-tandemed) deployments. The knee of the one-way delay impairment factor begins rising non-linearly somewhere between 150 and 250 ms. Mike ________________________________ From: codec-bounces@ietf.org <codec-bounces@ietf.org> To: Ben Schwartz <bmschwar@fas.harvard.edu> Cc: codec@ietf.org <codec@ietf.org> Sent: Tue May 11 14:19:22 2010 Subject: Re: [codec] Thresholds and delay. >>> In the presence of echo, round-trip delay must be kept below 30 ms to ensure that the echo is perceived as sidetone, according to the Springer handbook of speech processing: >>> Though true, I don't think this is a mainstream consideration. VOIP phones that are capable of speakerphone operation all have acoustic echo cancelers, and those cancelers are already tuned to deal with internet delays with other voice codecs. Certainly our phones and videoconferencing systems do not have problems with path delays of this order (hundreds of milliseconds). From my own experience (not testing) I agree with Brian's claim that 500 ms round trip is acceptable for most conversation. It does depend on what you are doing, and there are certainly tasks where much lower delays are needed. Regards, Stephen Botzko On Tue, May 11, 2010 at 2:06 PM, Ben Schwartz <bmschwar@fas.harvard.edu<mailto:bmschwar@fas.harvard.edu>> wrote: On Tue, 2010-05-11 at 12:48 -0400, Marshall Eubanks wrote: > As a point of order, I object to any graphs without an available paper > behind them. I have located the first paper mentioned by Christian Hoene at http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=81952 but of course it's paywalled. One test in that paper told trained subjects to "Take turns reading random numbers aloud as fast as possible", on a pair of handsets with narrowband uncompressed audio and no echo. Subjects were able to detect round-trip delays down to 90 ms. Conversational efficiency was impaired even with round-trip delay of 100 ms. Let me emphasize again that these delays are round-trip, not one-way, there is no echo, and the task, while designed to expose latency, is probably less demanding than musical performance. In the presence of echo, round-trip delay must be kept below 30 ms to ensure that the echo is perceived as sidetone, according to the Springer handbook of speech processing: (http://books.google.com/books?id=Slg10ekZBkAC&lpg=PA83&ots=wc9yM9WrCs&dq=sidetone%20delay%2030%20ms&lr&pg=PA84#v=onepage&q&f=false) Such low delays are clearly impossible on many paths, but for Boston to New York City (or London to Paris), ping times can be less than 18 ms, making echo->sidetone conversion just barely possible for a codec with 5ms frames. I accept Brian Rosen's claim that a slow conversation doesn't normally suffer greatly from round-trip latencies up to 500 ms, but under some circumstances much lower latencies are valuable. Let's make sure they're achievable for those who can use them. --Ben _______________________________________________ codec mailing list codec@ietf.org<mailto:codec@ietf.org> https://www.ietf.org/mailman/listinfo/codec
- Re: [codec] Thresholds and delay. Michael Knappe
- Re: [codec] Thresholds and delay. stephen botzko