Pitch distortion in VoIP?

In

But

I wouldn't even trust VoIP to reproduce a tone that accurately _assuming_ good sampling -- they compress pretty heavily, and I wouldn't be able to tell you if the message that comes across isn't 'tone at around 500Hz' rather than the actual samples.

I _do_ know that one of the oft-used algorithms is SPEEX. Look at the acronym and guess what it's optimized for. You can web search for the spec and look to see what it does to tones.

The frequency you get should only depend on the precision of the oscillator timing the conversion at your side. Encoded signal frequency has nothing to do with transmission speed. Whether you can transfer the signal in real time or not depends on the transmission speed vs. data size, of course.

Transmitting voice over tcp would not be very wise... (retransmissions). Just IP - they probably do it under UDP somehow - should be enough. "Best delivery effort" - which is how IP works - would mean that some lost packet here and there will result in some clicks/noise etc., but given that "most" packets make it the voice will still be recognizable.

Dimiter

------------------------------------------------------ Dimiter Popoff Transgalactic Instruments

------------------------------------------------------

Robert Scott wrote:

In

But

I tend to agree with your point and it is the point I have been promoting in another forum. However it is refuted by direct experiementation by others in that forum where they call up the NIST tones at 303-499-7111 on the VoIP and simulataneously on their regular phone. They report a noticeable phase delay between the two feeds, however they report no beat whatsoever between the two tones. And these people are piano tuners, so they would recognize a beat if it exists, even down to one beat in 10 seconds. So are you sure the VoIP hasn't done something with a software phase-locked loop to generate a local correction to the sample rate difference?

Robert Scott Ypsilanti, Michigan

Absolutely not. It sounds like my concern is quite effectively refuted, at least for the VoIP services that the experiment was conducted with.

I really _don't_ know how VoIP works -- this means I can't say "nay", but that I can think of a lot of reasons not to say "aye" without things like direct experimentation.

There is no beat, so they're on freq; but that doesn't say anything about the phase. I could see syncing up packets to some local clock, like the computer's RTC, just for fidelity's sake; but I wouldn't place any bets on how many cycles of 600Hz' delay there is.

On WWV on the air, there's a voice announcement every minute or hour or something; if that's on VOIP, it'd be interesting to compare the two voice segments. :-)

Cheers! Rich

IIRC VOIP has some means to do bitrate control between both sides so the transmit and receive rates are sort of synchronized (there is probably a lot of phase jitter). This is required in order not to be forced to drop or insert audio frames (which will lead to noticable distortions). So the NIST tone through VOIP should -in theory- be just as precise as the analog version.

Voip must be able to pass DTMF tonesets unaltered in frequency and amplitude (not in phase), so there are some tone encoding features in the algorithm. This is not really for dialing, but for accessing automated answering systems and such.

I unfortunately do not know a lot of details, though. I would think that tones other than the standard DTMF frequencies will likely be passed with unaltered amplitude and frequency as well.

Glenn

For my purposes, I don't care about a fixed phase delay. I only want to use the NIST tones for audio frequency calibration.

I am still looking for definitive evidence that VoIP contains a mechanism to ensure frequency accuracy. Negative evidence of deviation is not proof, although positive evidence of deviaiton would be proof that there is no such mechanism.

Robert Scott Ypsilanti, Michigan

The required frequency accuracy of DTMF tones is not very strict. They would work even if VoIP just used an uncalibrated sound card sample rate.

Robert Scott Ypsilanti, Michigan

I'm not sure that an occasional dropped sample would be all that noticed, especially in speech. I can see how information on the fullness of the FIFO could be used in a low-bandwidth feedback loop to calibrate the rate at which data is extracted from the FIFO, but what could the bandwidth of such a loop be?

Robert Scott Ypsilanti, Michigan

If you can reproduce a particular frequency accurately, depends on the codec. You can force a specific codec from either end, and you could select a-law or u-law PCM, so there is no frequency encoding involved.

The sample rates are negotiated over the network, so both ends use the same rate, as referenced to the two local quartz crystals. The VoIP system compensates against slow drift by inserting or deleting a few ms of signal at a time.

In other words, if you want to calibrate against a tone over VoIP, don't. Spare yourself the effort and just divide down your local quartz timebase to generate the desired tone. Same accuracy.

Kind regards,

Iwo

the

Well, this:

Good Luck! Rich

In a single tone dropping or adding a sample is very noticable. In speech it is not.

I don't think so. The difference is 50 to 100 ppm at most. I don't think the human ear can hear such a small frequency variation.

Google this: voip synchronization

It is more to the point you are seeking. In a good network (synchronized) I don't think you will have a frequency problem, and phase problems should be minimized.

Don

Thanks for the tip. That did indeed lead to a very informative whitepaper on synchronization, and I can see that the issue has been seriously addressed by the designers of VoIP.

It appears that the main concern of synchronization in VoIP is voice latency. Anything more than 150 msec. becomes a noticeable annoyance to the users. The

All that I understand. However I can still see where one particular concern of mine is not addressed in these descriptions. That is the tie-in between time synchronization and audio frequency accuracy. If the raw time-series data were transferred uncompressed, then time-synchronization would automatically imply audio frequency precision. But, as Tim Wescott pointed out, they use some pretty aggresive speech compression in VoIP. If speech is compressed using frequency-domain techniques, then it is entirely possible to have perfect time synchronization, but still have small pitch errors in the playback. The possible disconnect between time and pitch is most easily seen in the systems that create time-compressed audio books without resulting in the "chipmunks" type of pitch raising. That is an extreme example, but it illustrates the potential disconnect. I hope to find that this type of disconnect does not happen in VoIP, but I will have to look a little more to find out for sure.

Robert Scott Ypsilanti, Michigan

I haven't had time to study voip in great detail, but it's not unreasonable to make some assumptions about it....

I believe voip, uses a form of adpcm to code to a low bit-rate. There is nothing in this process that would cause frequency shift in the digitally encoded signal; it only limits the analog bandwidth of the decoded signal while causing uniform delay. However, and this is most unlikely, if every other packet was thrown away then the reconstructed voice signal from the PAM samples, would shift the sound frequency down by half.

I see in today's financial news that Vonage has created a serious non-technical problem by their initial stock offering methods.

Don