[ntp:questions] Accuracy of audio tones via VOIP

Robert Scott no-one at notreal.invalid
Mon Jul 15 17:32:49 UTC 2013


On Mon, 15 Jul 2013 15:51:25 GMT, unruh <unruh at invalid.ca> wrote:

>On 2013-07-15, Robert Scott <no-one at notreal.invalid> wrote:
>> On Wed, 10 Jul 2013 17:57:40 GMT, unruh <unruh at invalid.ca> wrote:
>>
>>>
>>>But again. why does the OP not just measure it? I guess he is hoping
>>>that someone already had done his work for him.
>>>
>>
>> There are several reasons why I have not "just measured it".  One is
>> that I do not have a Skype subscription.  Another reason is that a
>> single measurement on a single computer means nothing.  If the sound
>> card in that computer happens to have a crystal oscillator that is set
>> very close to its nominal frequency, the frequency reproduction error
>> would be nearly zero, even if Skype did nothing to compensate for it.
>
>Skype nor anything will compensate for the sound card frequency. Noone
>in that field cares about a few cents ( 100ths of a semitone) difference
>in frequencies.

I agree that pitch accuracy itself is not important to Skype et. al.
But indirectly a mismatch between the originating audio sampling rate
and the playback sampling rate results in an ever increasing buffer
overflow or underflow that could only be corrected by dropping packets
or inserting extra packets.  Perhaps you are right and that is all
they do.  I suppose if the packets are small enough that would not
interfere noticeably with speech.  But for my purposes it would result
in occasional phase jumps in the recovered tones, which is just as
problematic for me as an inaccuracte playback sample rate.  I am
leaning heavily toward simply recommending that my users not use any
VOIP connection to do their calibration.  That would be consistent
will all the advice I have gotten here so far.


>What in the world makes you think skype
>corrects anything? They have a desperate time of it getting most of the
>packets to you, never mind correcting for frequency errors in a
>soundcard. (and how would they know that the soundcard had frequency
>errors to correct?)
>And what is the range of frequency errors in the soundcards?

I described earlier the one possible method whereby they might know
the soundcard playback rate error.  If their software running on my
computer is able to monitor the long-term trend of the number samples
in their buffer, that trend would directly correlate with the audio
playback sampling rate error.  The trouble is the fluctuations in the
number of samples buffered is, in the short run, more dependent on
random internet latency than it is on playback rate error.  Only over
very long time periods would the internet randomness be dominated by
the playback rate error.  But most phone calls do not last long enough
to establish any meaningful measurment, which is why I doubt that they
use such a method.  However I am not so bold as to assume that just
because I am not clever enough to figure out how VOIP might correct
for playback rate errors, that there is no such method, thus my
questions here.

As for your last question, I have measured sound boards that are 11
cents off from their nominal playback rate (22200 sps instead of the
nominal 22050 sps).  Most soundcards are less than 1 cent off.  But
the standard among my competitors in the field of electronic aids to
piano tuning is 0.02 cents.  So if I want to compete then my app must
be able to calibrate to that accuracy too.

>> I am the developer of a professional piano tuning app for
>> smarthphones.  My customers are professional piano tuners.  Although
>> the sound sampling rate on most smartphones is very close to nominal,
>> there is as need to calibrate each device on which the app runs.  The
>> means that I offer in my app is to have the user call up the NIST
>> tones over the telephone and let the app listen to the 500 Hz or 600
>> Hz tones for 30 seconds.  The call is placed using a different phone
>> and the sound is transferred acoustically by careful positioning of
>> the microphone.  With this method I have been able to achieve a
>> frequency calibration of 6 ppm.  This worked fine when everyone had
>
>You know this how? You tested the frequency calibration with some
>independent way? Of is this a theoretical  estimate based on how you
>detect the frequency?

I tested the calibration against the audio of WWV received over
shortwave radio during times when a strong signal was received with no
fading.

Robert Scott
Hopkins, MN



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