Re: Re: Sample Rate [7:36566]
This is OT, but the upper limit of human hearing is actually around 20KHz at best and usually drops to around 16KHz or so. If your upper limit starts to drop below that you'll start to notice that it's difficult to hear clearly. (Sorry, in my other life I'm a sound engineer and musician.) I've heard that the 4KHz limit is because there is a low-pass filter used for voice. I can't remember the exact reason, but that information plugged into the Nyquist theorem explains--as Priscilla mentions--why a DS0 is 64Kbps. Okay, time to do some serious studying once I'm through being lazy and drinking this coffee... John On Tue, 26 Feb 2002, Priscilla Oppenheimer ([EMAIL PROTECTED]) wrote: > At 08:06 PM 2/26/02, Rafay wrote: > >How do you describe Sample Rate.? > > In what context? The term is sometimes used when describing the analog > to > digital process, for example when digitizing voice. Voice produces an > analog wave as your lungs and tongue press against the air. An analog > wave > has infinite possible values. Computers can't deal with infinity. They > work > with discreet numbers. The solution is to sample the analog voice many > times per second. Sampling means to take a snapshot. > > The sample rate is how often the analog wave is sampled. Nyquist showed > that you have to sample at twice the rate of the highest frequency that > may > occur in the original data. Most humans don't output (and can't hear) > anything about 4 KHz. So sample 8,000 times per second (8Khz) and the > result will be good enough. When using a sample rate of 8,000 KHz, if > each > sample is saved in an 8-bit byte, the resulting data rate is 64 Kbps. > That's one DS0. Compression allows us to use a smaller data rate, with > some > loss in fidelity. > > Priscilla > > > Priscilla Oppenheimer > http://www.priscilla.com [EMAIL PROTECTED] > > Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36587&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
64 kbps comes about from sampling 8 bits at 8khz, 8x8000=64000 Bri On Tue, 26 Feb 2002, John Neiberger wrote: > This is OT, but the upper limit of human hearing is actually > around 20KHz at best and usually drops to around 16KHz or so. > If your upper limit starts to drop below that you'll start to > notice that it's difficult to hear clearly. (Sorry, in my > other life I'm a sound engineer and musician.) > > I've heard that the 4KHz limit is because there is a low-pass > filter used for voice. I can't remember the exact reason, but > that information plugged into the Nyquist theorem explains--as > Priscilla mentions--why a DS0 is 64Kbps. > > Okay, time to do some serious studying once I'm through being > lazy and drinking this coffee... > > John > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > ([EMAIL PROTECTED]) wrote: > > > At 08:06 PM 2/26/02, Rafay wrote: > > >How do you describe Sample Rate.? > > > > In what context? The term is sometimes used when describing > the analog > > to > > digital process, for example when digitizing voice. Voice > produces an > > analog wave as your lungs and tongue press against the air. > An analog > > wave > > has infinite possible values. Computers can't deal with > infinity. They > > work > > with discreet numbers. The solution is to sample the analog > voice many > > times per second. Sampling means to take a snapshot. > > > > The sample rate is how often the analog wave is sampled. > Nyquist showed > > that you have to sample at twice the rate of the highest > frequency that > > may > > occur in the original data. Most humans don't output (and > can't hear) > > anything about 4 KHz. So sample 8,000 times per second (8Khz) > and the > > result will be good enough. When using a sample rate of 8,000 > KHz, if > > each > > sample is saved in an 8-bit byte, the resulting data rate is > 64 Kbps. > > That's one DS0. Compression allows us to use a smaller data > rate, with > > some > > loss in fidelity. > > > > Priscilla > > > > > > Priscilla Oppenheimer > > http://www.priscilla.com > [EMAIL PROTECTED] > > > > > > > > Get your own "800" number > Voicemail, fax, email, and a lot more > http://www.ureach.com/reg/tag Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36597&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
Exactly, that's what Priscilla and I both just said. :-) What I'm trying to find out is why the original 4KHz limit on voice calls was put into place. It sounds like it was simply an arbitrary decision. 4KHz is sufficient for a telephone call and to provide clear calls that included higher frequencies might have added some technical complexities, perhaps. They also added a high-pass filter around 400Hz since most telephones can't reproduce low frequencies well and it also filters out some harmonics of 50-60Hz hum that might show up from time to time. That is concrete reason for including a high-pass filter and I wondered if there was a concrete technical reason for including the 4KHz low-pass filter. From the sounds of it there really isn't a technical issue, 4K is just a nice round number. :-) I've actually read that they limit it to around 3.4KHz, but if you sample that at 8KHz you'd be well above the Nyquist limit. John Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag On Tue, 26 Feb 2002, Brian ([EMAIL PROTECTED]) wrote: > 64 kbps comes about from sampling 8 bits at 8khz, 8x8000=64000 > > > Bri > > On Tue, 26 Feb 2002, John Neiberger wrote: > > > This is OT, but the upper limit of human hearing is actually > > around 20KHz at best and usually drops to around 16KHz or so. > > If your upper limit starts to drop below that you'll start to > > notice that it's difficult to hear clearly. (Sorry, in my > > other life I'm a sound engineer and musician.) > > > > I've heard that the 4KHz limit is because there is a low- pass > > filter used for voice. I can't remember the exact reason, but > > that information plugged into the Nyquist theorem explains-- as > > Priscilla mentions--why a DS0 is 64Kbps. > > > > Okay, time to do some serious studying once I'm through being > > lazy and drinking this coffee... > > > > John > > > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > > ([EMAIL PROTECTED]) wrote: > > > > > At 08:06 PM 2/26/02, Rafay wrote: > > > >How do you describe Sample Rate.? > > > > > > In what context? The term is sometimes used when describing > > the analog > > > to > > > digital process, for example when digitizing voice. Voice > > produces an > > > analog wave as your lungs and tongue press against the air. > > An analog > > > wave > > > has infinite possible values. Computers can't deal with > > infinity. They > > > work > > > with discreet numbers. The solution is to sample the analog > > voice many > > > times per second. Sampling means to take a snapshot. > > > > > > The sample rate is how often the analog wave is sampled. > > Nyquist showed > > > that you have to sample at twice the rate of the highest > > frequency that > > > may > > > occur in the original data. Most humans don't output (and > > can't hear) > > > anything about 4 KHz. So sample 8,000 times per second (8Khz) > > and the > > > result will be good enough. When using a sample rate of 8,000 > > KHz, if > > > each > > > sample is saved in an 8-bit byte, the resulting data rate is > > 64 Kbps. > > > That's one DS0. Compression allows us to use a smaller data > > rate, with > > > some > > > loss in fidelity. > > > > > > Priscilla > > > > > > > > > Priscilla Oppenheimer > > > http://www.priscilla.com > > [EMAIL PROTECTED] > > > > > > > > > > > > > > Get your own "800" number > > Voicemail, fax, email, and a lot more > > http://www.ureach.com/reg/tag [EMAIL PROTECTED] Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36599&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: Re: Sample Rate [7:36566]
In the Cvoice class I attended, my instructor answered the reason why it was 4K as there were 2 groups wanting different rates, so the compromise between them was 4k. None of my IP Telephony course books or Cisco Press books for the class re-iterate that, but I recall hearing it... Larry Letterman Cisco Systems [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of John Neiberger Sent: Tuesday, February 26, 2002 9:23 PM To: [EMAIL PROTECTED] Subject: Re: Re: Sample Rate [7:36566] Exactly, that's what Priscilla and I both just said. :-) What I'm trying to find out is why the original 4KHz limit on voice calls was put into place. It sounds like it was simply an arbitrary decision. 4KHz is sufficient for a telephone call and to provide clear calls that included higher frequencies might have added some technical complexities, perhaps. They also added a high-pass filter around 400Hz since most telephones can't reproduce low frequencies well and it also filters out some harmonics of 50-60Hz hum that might show up from time to time. That is concrete reason for including a high-pass filter and I wondered if there was a concrete technical reason for including the 4KHz low-pass filter. From the sounds of it there really isn't a technical issue, 4K is just a nice round number. :-) I've actually read that they limit it to around 3.4KHz, but if you sample that at 8KHz you'd be well above the Nyquist limit. John Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag On Tue, 26 Feb 2002, Brian ([EMAIL PROTECTED]) wrote: > 64 kbps comes about from sampling 8 bits at 8khz, 8x8000=64000 > > > Bri > > On Tue, 26 Feb 2002, John Neiberger wrote: > > > This is OT, but the upper limit of human hearing is actually > > around 20KHz at best and usually drops to around 16KHz or so. > > If your upper limit starts to drop below that you'll start to > > notice that it's difficult to hear clearly. (Sorry, in my > > other life I'm a sound engineer and musician.) > > > > I've heard that the 4KHz limit is because there is a low- pass > > filter used for voice. I can't remember the exact reason, but > > that information plugged into the Nyquist theorem explains-- as > > Priscilla mentions--why a DS0 is 64Kbps. > > > > Okay, time to do some serious studying once I'm through being > > lazy and drinking this coffee... > > > > John > > > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > > ([EMAIL PROTECTED]) wrote: > > > > > At 08:06 PM 2/26/02, Rafay wrote: > > > >How do you describe Sample Rate.? > > > > > > In what context? The term is sometimes used when describing > > the analog > > > to > > > digital process, for example when digitizing voice. Voice > > produces an > > > analog wave as your lungs and tongue press against the air. > > An analog > > > wave > > > has infinite possible values. Computers can't deal with > > infinity. They > > > work > > > with discreet numbers. The solution is to sample the analog > > voice many > > > times per second. Sampling means to take a snapshot. > > > > > > The sample rate is how often the analog wave is sampled. > > Nyquist showed > > > that you have to sample at twice the rate of the highest > > frequency that > > > may > > > occur in the original data. Most humans don't output (and > > can't hear) > > > anything about 4 KHz. So sample 8,000 times per second (8Khz) > > and the > > > result will be good enough. When using a sample rate of 8,000 > > KHz, if > > > each > > > sample is saved in an 8-bit byte, the resulting data rate is > > 64 Kbps. > > > That's one DS0. Compression allows us to use a smaller data > > rate, with > > > some > > > loss in fidelity. > > > > > > Priscilla > > > > > > > > > Priscilla Oppenheimer > > > http://www.priscilla.com > > [EMAIL PROTECTED] > > > > > > > > > > > > > > Get your own "800" number > > Voicemail, fax, email, and a lot more > > http://www.ureach.com/reg/tag [EMAIL PROTECTED] Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36604&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
> John Neiberger wrote: > What I'm trying to find out is why the original 4KHz limit on > voice calls was put into place. It sounds like it was simply > an arbitrary decision. 4KHz is sufficient for a telephone call > and to provide clear calls that included higher frequencies > might have added some technical complexities, perhaps. > > They also added a high-pass filter around 400Hz since most > telephones can't reproduce low frequencies well and it also > filters out some harmonics of 50-60Hz hum that might show up > from time to time. That is concrete reason for including a > high-pass filter and I wondered if there was a concrete > technical reason for including the 4KHz low-pass filter. From > the sounds of it there really isn't a technical issue, 4K is > just a nice round number. :-) I used three sources to answer John's query: "Voice over IP Fundamentals", "Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and Data Networks". These are great books for anyone wanting to know more about voice technologies. Interesting Facts and Ideas I came across: 1) Human hearing is in the range of 200 Hz to 20,000 Hz 2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the information comes from the middle frequencies. According to Nyquist, "Human voice contains sounds that are more often Middle-pitched frequencies than either High or Low pitched frequencies. 3) Frequencies greater than 4,000 Hz are filter out to limit crosstalk. 4) During the Analog to Digital conversion voice samples are put though a process called Quantization. Quantization is the process of rounding sampled values to the nearest predefined discreet value. Pulse Code Modulation (PCM) is a Quantization process. PCM is also used to achieve 12 to 13 bits of voice information in 8 bit words. Two commonly used PCM's are: mu-law (North America), and a-law (Europe). What you hear is not someone's voice, but a representation of their voice. 5) Noise is a major issue when talking about voice quality. Noise is constant problem for Analog signals. What is signal and what is Noise? When a Analog signal is amplified so is the Noise, which in turn makes the quality of Analog calls worst as the distance increases. Digital Calls are less suitable to Noise than Analog calls. 6) Delay is a major issue when talking about conversation flow for two reasons: 1) For a conversation to flow normally, the delay is receiving the voice information must be less than 250ms. When the delay is more than 250ms, the human receiving the voice message will start to talk thinking the human sending the voice message is at a breaking point in the conversation, i.e.. both people are talking at the same time similar to a collision in Ethernet. Delay is also important in how the voice packets are filled during the Analog to Digital conversion. That is why ATM (ATM cell is 53 octets, 5 octets are header and 48 octets are payload) is a good method for transporting voice packets because the delay to fill the payload section is smaller than with other cell/packet types. Answer: It does indeed seem that the 4,000 Hz mark was arbitrary in nature; 3,500 Hz or 5,000 Hz would work also. It is a "nice round" number to work with. Simplies any math work. Middle frequencies carry the bulk of the information and Human speech upper limit is 10,000 Hz amd 4,000 is near the middle. The low filter is also to reduce the frequencies that carry less information. Hope this helps. "Through Complexity there is Simplicity, Through Simplicity there is Complexity" David L. Blair - CCNP, CCNA, MCSE, CBE, A+, 3Wizard Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36655&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
Thanks, that's exactly the sort of thing I was hoping to find. It's also interesting to note that the human ear is most sensitive to midrange frequencies, say between 1000Hz and 4000Hz. So, even if you filter out what's above 4KHz, you can make up for the lack of clarity with a little amplitude. However, it's this filter that can make it difficult to distinguish between an F and an S sound. The frequencies most necessary to hear those sounds clearly are above 4KHz. And no, amplitude is NOT a measure of how much air a snowboarder gets above the half pipe rim! :-) John >>> "David L. Blair" 2/27/02 8:58:07 AM >>> > John Neiberger wrote: > What I'm trying to find out is why the original 4KHz limit on > voice calls was put into place. It sounds like it was simply > an arbitrary decision. 4KHz is sufficient for a telephone call > and to provide clear calls that included higher frequencies > might have added some technical complexities, perhaps. > > They also added a high-pass filter around 400Hz since most > telephones can't reproduce low frequencies well and it also > filters out some harmonics of 50-60Hz hum that might show up > from time to time. That is concrete reason for including a > high-pass filter and I wondered if there was a concrete > technical reason for including the 4KHz low-pass filter. From > the sounds of it there really isn't a technical issue, 4K is > just a nice round number. :-) I used three sources to answer John's query: "Voice over IP Fundamentals", "Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and Data Networks". These are great books for anyone wanting to know more about voice technologies. Interesting Facts and Ideas I came across: 1) Human hearing is in the range of 200 Hz to 20,000 Hz 2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the information comes from the middle frequencies. According to Nyquist, "Human voice contains sounds that are more often Middle-pitched frequencies than either High or Low pitched frequencies. 3) Frequencies greater than 4,000 Hz are filter out to limit crosstalk. 4) During the Analog to Digital conversion voice samples are put though a process called Quantization. Quantization is the process of rounding sampled values to the nearest predefined discreet value. Pulse Code Modulation (PCM) is a Quantization process. PCM is also used to achieve 12 to 13 bits of voice information in 8 bit words. Two commonly used PCM's are: mu-law (North America), and a-law (Europe). What you hear is not someone's voice, but a representation of their voice. 5) Noise is a major issue when talking about voice quality. Noise is constant problem for Analog signals. What is signal and what is Noise? When a Analog signal is amplified so is the Noise, which in turn makes the quality of Analog calls worst as the distance increases. Digital Calls are less suitable to Noise than Analog calls. 6) Delay is a major issue when talking about conversation flow for two reasons: 1) For a conversation to flow normally, the delay is receiving the voice information must be less than 250ms. When the delay is more than 250ms, the human receiving the voice message will start to talk thinking the human sending the voice message is at a breaking point in the conversation, i.e.. both people are talking at the same time similar to a collision in Ethernet. Delay is also important in how the voice packets are filled during the Analog to Digital conversion. That is why ATM (ATM cell is 53 octets, 5 octets are header and 48 octets are payload) is a good method for transporting voice packets because the delay to fill the payload section is smaller than with other cell/packet types. Answer: It does indeed seem that the 4,000 Hz mark was arbitrary in nature; 3,500 Hz or 5,000 Hz would work also. It is a "nice round" number to work with. Simplies any math work. Middle frequencies carry the bulk of the information and Human speech upper limit is 10,000 Hz amd 4,000 is near the middle. The low filter is also to reduce the frequencies that carry less information. Hope this helps. "Through Complexity there is Simplicity, Through Simplicity there is Complexity" David L. Blair - CCNP, CCNA, MCSE, CBE, A+, 3Wizard Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36661&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: Re: Sample Rate [7:36566]
All this voice stuff goes back to copper lines. Some random thoughts that I heard circa 1970 while working for Ma Bell and T-1 was the new kid on the block. When extending trunks the two wire line went through a hybrid that converted it to four wire. Two for transmit and two for receive. Amplifiers were used. Higher frequencies attenuate at a greater rate than lower frequencies. The signal was filtered into bands with each band amplified at a different gain so that at the receiving end it sounded normal. Choices had to be made about how high a frequency to use. A copper pair has resistance (length) and capacitance (each pair is twisted together to minimize crosstalk.) To offset the effects of capacitance load coils (inductance) was added. Thus an RCL tuned circuit. It was optimized for frequencies below 4khz. This work predates digital carrier. Side note: Analog frequency seperated carriers used inband signalling. Various devices had different functions but the common point was the use of a 2600 Hz tone. Some folks had too much of that frequency in their voice and so disconnected themselves. > -Original Message- > From: David L. Blair [mailto:[EMAIL PROTECTED]] > Sent: Wednesday, February 27, 2002 9:58 AM > To: [EMAIL PROTECTED] > Subject: Re: Re: Sample Rate [7:36566] > > > > John Neiberger wrote: > > What I'm trying to find out is why the original 4KHz limit on > > voice calls was put into place. It sounds like it was simply > > an arbitrary decision. 4KHz is sufficient for a telephone call > > and to provide clear calls that included higher frequencies > > might have added some technical complexities, perhaps. > > > > They also added a high-pass filter around 400Hz since most > > telephones can't reproduce low frequencies well and it also > > filters out some harmonics of 50-60Hz hum that might show up > > from time to time. That is concrete reason for including a > > high-pass filter and I wondered if there was a concrete > > technical reason for including the 4KHz low-pass filter. From > > the sounds of it there really isn't a technical issue, 4K is > > just a nice round number. :-) > > I used three sources to answer John's query: "Voice over IP > Fundamentals", > "Cisco Voice over Frame Relay, ATM, and IP", and Integrating > Voice and Data > Networks". These are great books for anyone wanting to know > more about > voice technologies. > > Interesting Facts and Ideas I came across: > > 1) Human hearing is in the range of 200 Hz to 20,000 Hz > > 2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > information comes from the middle frequencies. According to > Nyquist, "Human > voice contains sounds that are more often Middle-pitched > frequencies than > either High or Low pitched frequencies. > > 3) Frequencies greater than 4,000 Hz are filter out to limit > crosstalk. > > 4) During the Analog to Digital conversion voice samples are > put though a > process called Quantization. Quantization is the process of rounding > sampled values to the nearest predefined discreet value. Pulse Code > Modulation (PCM) is a Quantization process. PCM is also used > to achieve 12 > to 13 bits of voice information in 8 bit words. Two commonly > used PCM's are: > mu-law (North America), and a-law (Europe). What you hear is > not someone's > voice, but a representation of their voice. > > 5) Noise is a major issue when talking about voice quality. Noise is > constant problem for Analog signals. What is signal and what > is Noise? > When a Analog signal is amplified so is the Noise, which in > turn makes the > quality of Analog calls worst as the distance increases. > Digital Calls are > less suitable to Noise than Analog calls. > > 6) Delay is a major issue when talking about conversation flow for two > reasons: 1) For a conversation to flow normally, the delay is > receiving the > voice information must be less than 250ms. When the delay is > more than > 250ms, the human receiving the voice message will start to > talk thinking the > human sending the voice message is at a breaking point in the > conversation, > i.e.. both people are talking at the same time similar to a > collision in > Ethernet. Delay is also important in how the voice packets are filled > during the Analog to Digital conversion. That is why ATM > (ATM cell is 53 > octets, 5 octets are header and 48 octets are payload) is a > good method for > transporting voice packets because the delay to fill the > payload section is > smaller than with other cell/packet types. > > > Answer: It does indeed seem that the 4,0
Re: Re: Sample Rate [7:36566]
You are right, John. With digital telephony, the analog speech signal is filtered before sampling. High and low frequency components are removed. I think it was just a tradeoff. We don't expect the human voice to sound that great over the phone anyway, and by filtering we can reduce bandwidth requirements. In fact, the human voice doesn't sound too great over the phone. We've just gotten used to it. Music on hold over the phone sounds pretty bad. Priscilla At 10:25 PM 2/26/02, John Neiberger wrote: >This is OT, but the upper limit of human hearing is actually >around 20KHz at best and usually drops to around 16KHz or so. >If your upper limit starts to drop below that you'll start to >notice that it's difficult to hear clearly. (Sorry, in my >other life I'm a sound engineer and musician.) > >I've heard that the 4KHz limit is because there is a low-pass >filter used for voice. I can't remember the exact reason, but >that information plugged into the Nyquist theorem explains--as >Priscilla mentions--why a DS0 is 64Kbps. > >Okay, time to do some serious studying once I'm through being >lazy and drinking this coffee... > >John > > On Tue, 26 Feb 2002, Priscilla Oppenheimer >([EMAIL PROTECTED]) wrote: > > > At 08:06 PM 2/26/02, Rafay wrote: > > >How do you describe Sample Rate.? > > > > In what context? The term is sometimes used when describing >the analog > > to > > digital process, for example when digitizing voice. Voice >produces an > > analog wave as your lungs and tongue press against the air. >An analog > > wave > > has infinite possible values. Computers can't deal with >infinity. They > > work > > with discreet numbers. The solution is to sample the analog >voice many > > times per second. Sampling means to take a snapshot. > > > > The sample rate is how often the analog wave is sampled. >Nyquist showed > > that you have to sample at twice the rate of the highest >frequency that > > may > > occur in the original data. Most humans don't output (and >can't hear) > > anything about 4 KHz. So sample 8,000 times per second (8Khz) >and the > > result will be good enough. When using a sample rate of 8,000 >KHz, if > > each > > sample is saved in an 8-bit byte, the resulting data rate is >64 Kbps. > > That's one DS0. Compression allows us to use a smaller data >rate, with > > some > > loss in fidelity. > > > > Priscilla > > > > > > Priscilla Oppenheimer > > http://www.priscilla.com >[EMAIL PROTECTED] > > > > > > > >Get your own "800" number >Voicemail, fax, email, and a lot more >http://www.ureach.com/reg/tag Priscilla Oppenheimer http://www.priscilla.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36713&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
Female opera singers probably hate it when people ask them to sing over the phone!? OK, have we distracted you enough, John? ;-) Seriously, I think this was a great discussion. Thanks to everyone who contributed. Priscilla At 10:58 AM 2/27/02, David L. Blair wrote: > > John Neiberger wrote: > > What I'm trying to find out is why the original 4KHz limit on > > voice calls was put into place. It sounds like it was simply > > an arbitrary decision. 4KHz is sufficient for a telephone call > > and to provide clear calls that included higher frequencies > > might have added some technical complexities, perhaps. > > > > They also added a high-pass filter around 400Hz since most > > telephones can't reproduce low frequencies well and it also > > filters out some harmonics of 50-60Hz hum that might show up > > from time to time. That is concrete reason for including a > > high-pass filter and I wondered if there was a concrete > > technical reason for including the 4KHz low-pass filter. From > > the sounds of it there really isn't a technical issue, 4K is > > just a nice round number. :-) > >I used three sources to answer John's query: "Voice over IP Fundamentals", >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and Data >Networks". These are great books for anyone wanting to know more about >voice technologies. > >Interesting Facts and Ideas I came across: > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the >information comes from the middle frequencies. According to Nyquist, "Human >voice contains sounds that are more often Middle-pitched frequencies than >either High or Low pitched frequencies. > >3) Frequencies greater than 4,000 Hz are filter out to limit crosstalk. > >4) During the Analog to Digital conversion voice samples are put though a >process called Quantization. Quantization is the process of rounding >sampled values to the nearest predefined discreet value. Pulse Code >Modulation (PCM) is a Quantization process. PCM is also used to achieve 12 >to 13 bits of voice information in 8 bit words. Two commonly used PCM's are: >mu-law (North America), and a-law (Europe). What you hear is not someone's >voice, but a representation of their voice. > >5) Noise is a major issue when talking about voice quality. Noise is >constant problem for Analog signals. What is signal and what is Noise? >When a Analog signal is amplified so is the Noise, which in turn makes the >quality of Analog calls worst as the distance increases. Digital Calls are >less suitable to Noise than Analog calls. > >6) Delay is a major issue when talking about conversation flow for two >reasons: 1) For a conversation to flow normally, the delay is receiving the >voice information must be less than 250ms. When the delay is more than >250ms, the human receiving the voice message will start to talk thinking the >human sending the voice message is at a breaking point in the conversation, >i.e.. both people are talking at the same time similar to a collision in >Ethernet. Delay is also important in how the voice packets are filled >during the Analog to Digital conversion. That is why ATM (ATM cell is 53 >octets, 5 octets are header and 48 octets are payload) is a good method for >transporting voice packets because the delay to fill the payload section is >smaller than with other cell/packet types. > > >Answer: It does indeed seem that the 4,000 Hz mark was arbitrary in nature; >3,500 Hz or 5,000 Hz would work also. It is a "nice round" number to work >with. Simplies any math work. Middle frequencies carry the bulk of the >information and Human speech upper limit is 10,000 Hz amd 4,000 is near the >middle. The low filter is also to reduce the frequencies that carry less >information. > >Hope this helps. > > >"Through Complexity there is Simplicity, >Through Simplicity there is Complexity" > >David L. Blair - CCNP, CCNA, MCSE, CBE, A+, 3Wizard Priscilla Oppenheimer http://www.priscilla.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36715&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
Yes, this was very distracting! :-) I didn't get any studying done at all last night! Between checking and answering email, looking for Clannad MP3s, reading about the Gaelic language just for fun, and looking up telecom stuff it's a wonder I even powered up a router. I was able to boot up six routers, erase their configs, and recable them in preparation for a lab scenario tonight. Not bad for three hours work. heh heh... Thanks, John >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM >>> Female opera singers probably hate it when people ask them to sing over the phone!? OK, have we distracted you enough, John? ;-) Seriously, I think this was a great discussion. Thanks to everyone who contributed. Priscilla At 10:58 AM 2/27/02, David L. Blair wrote: > > John Neiberger wrote: > > What I'm trying to find out is why the original 4KHz limit on > > voice calls was put into place. It sounds like it was simply > > an arbitrary decision. 4KHz is sufficient for a telephone call > > and to provide clear calls that included higher frequencies > > might have added some technical complexities, perhaps. > > > > They also added a high-pass filter around 400Hz since most > > telephones can't reproduce low frequencies well and it also > > filters out some harmonics of 50-60Hz hum that might show up > > from time to time. That is concrete reason for including a > > high-pass filter and I wondered if there was a concrete > > technical reason for including the 4KHz low-pass filter. From > > the sounds of it there really isn't a technical issue, 4K is > > just a nice round number. :-) > >I used three sources to answer John's query: "Voice over IP Fundamentals", >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and Data >Networks". These are great books for anyone wanting to know more about >voice technologies. > >Interesting Facts and Ideas I came across: > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the >information comes from the middle frequencies. According to Nyquist, "Human >voice contains sounds that are more often Middle-pitched frequencies than >either High or Low pitched frequencies. > >3) Frequencies greater than 4,000 Hz are filter out to limit crosstalk. > >4) During the Analog to Digital conversion voice samples are put though a >process called Quantization. Quantization is the process of rounding >sampled values to the nearest predefined discreet value. Pulse Code >Modulation (PCM) is a Quantization process. PCM is also used to achieve 12 >to 13 bits of voice information in 8 bit words. Two commonly used PCM's are: >mu-law (North America), and a-law (Europe). What you hear is not someone's >voice, but a representation of their voice. > >5) Noise is a major issue when talking about voice quality. Noise is >constant problem for Analog signals. What is signal and what is Noise? >When a Analog signal is amplified so is the Noise, which in turn makes the >quality of Analog calls worst as the distance increases. Digital Calls are >less suitable to Noise than Analog calls. > >6) Delay is a major issue when talking about conversation flow for two >reasons: 1) For a conversation to flow normally, the delay is receiving the >voice information must be less than 250ms. When the delay is more than >250ms, the human receiving the voice message will start to talk thinking the >human sending the voice message is at a breaking point in the conversation, >i.e.. both people are talking at the same time similar to a collision in >Ethernet. Delay is also important in how the voice packets are filled >during the Analog to Digital conversion. That is why ATM (ATM cell is 53 >octets, 5 octets are header and 48 octets are payload) is a good method for >transporting voice packets because the delay to fill the payload section is >smaller than with other cell/packet types. > > >Answer: It does indeed seem that the 4,000 Hz mark was arbitrary in nature; >3,500 Hz or 5,000 Hz would work also. It is a "nice round" number to work >with. Simplies any math work. Middle frequencies carry the bulk of the >information and Human speech upper limit is 10,000 Hz amd 4,000 is near the >middle. The low filter is also to reduce the frequencies that carry less >information. > >Hope this helps. > > >"Through Complexity there is Simplicity, >Through Simplicity there is Complexity" > >David L. Blair - CCNP, CCNA, MCSE, CBE, A+, 3Wizard Priscilla Oppenheimer http://www.priscilla.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36724&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]
Speaking of sample rates, I am playing with the idea of offering audio training using MP3 files. I have prepared such a training on WAN Troubleshooting. I'd love to get some feedback. This audio training will help people studying for the Support test especially. It's 40 minutes. This means the file is huge, so don't try this at home on a modem line. The file is available for download here: http://www.troubleshootingnetworks.com/audio.html Please send me some feedback. Would you find such a product line helpful? Would you pay for MP3 audio training files? Thanks. Priscilla Oppenheimer At 03:32 PM 2/27/02, John Neiberger wrote: >Yes, this was very distracting! :-) I didn't get any studying done at >all last night! Between checking and answering email, looking for >Clannad MP3s, reading about the Gaelic language just for fun, and >looking up telecom stuff it's a wonder I even powered up a router. I >was able to boot up six routers, erase their configs, and recable them >in preparation for a lab scenario tonight. Not bad for three hours >work. heh heh... > >Thanks, >John > > >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM > >>> >Female opera singers probably hate it when people ask them to sing over >the >phone!? > >OK, have we distracted you enough, John? ;-) Seriously, I think this >was a >great discussion. Thanks to everyone who contributed. > >Priscilla > >At 10:58 AM 2/27/02, David L. Blair wrote: > > > John Neiberger wrote: > > > What I'm trying to find out is why the original 4KHz limit on > > > voice calls was put into place. It sounds like it was simply > > > an arbitrary decision. 4KHz is sufficient for a telephone call > > > and to provide clear calls that included higher frequencies > > > might have added some technical complexities, perhaps. > > > > > > They also added a high-pass filter around 400Hz since most > > > telephones can't reproduce low frequencies well and it also > > > filters out some harmonics of 50-60Hz hum that might show up > > > from time to time. That is concrete reason for including a > > > high-pass filter and I wondered if there was a concrete > > > technical reason for including the 4KHz low-pass filter. From > > > the sounds of it there really isn't a technical issue, 4K is > > > just a nice round number. :-) > > > >I used three sources to answer John's query: "Voice over IP >Fundamentals", > >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and >Data > >Networks". These are great books for anyone wanting to know more >about > >voice technologies. > > > >Interesting Facts and Ideas I came across: > > > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > > > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > >information comes from the middle frequencies. According to Nyquist, >"Human > >voice contains sounds that are more often Middle-pitched frequencies >than > >either High or Low pitched frequencies. > > > >3) Frequencies greater than 4,000 Hz are filter out to limit >crosstalk. > > > >4) During the Analog to Digital conversion voice samples are put >though a > >process called Quantization. Quantization is the process of rounding > >sampled values to the nearest predefined discreet value. Pulse Code > >Modulation (PCM) is a Quantization process. PCM is also used to >achieve 12 > >to 13 bits of voice information in 8 bit words. Two commonly used >PCM's are: > >mu-law (North America), and a-law (Europe). What you hear is not >someone's > >voice, but a representation of their voice. > > > >5) Noise is a major issue when talking about voice quality. Noise is > >constant problem for Analog signals. What is signal and what is >Noise? > >When a Analog signal is amplified so is the Noise, which in turn makes >the > >quality of Analog calls worst as the distance increases. Digital >Calls are > >less suitable to Noise than Analog calls. > > > >6) Delay is a major issue when talking about conversation flow for >two > >reasons: 1) For a conversation to flow normally, the delay is >receiving the > >voice information must be less than 250ms. When the delay is more >than > >250ms, the human receiving the voice message will start to talk >thinking the > >human sending the voice message is at a breaking point in the >conversation, > >i.e.. both people are talking at the same time similar to a collision >in > >Ethernet. Delay is also important in how the voice packets are >filled > >during the Analog to Digital conversion. That is why ATM (ATM cell is >53 > >octets, 5 octets are header and 48 octets are payload) is a good >method for > >transporting voice packets because the delay to fill the payload >section is > >smaller than with other cell/packet types. > > > > > >Answer: It does indeed seem that the 4,000 Hz mark was arbitrary in >nature; > >3,500 Hz or 5,000 Hz would work also. It is a "nice round" number to >work > >with. Simplies any math work. Middle frequencies carry the bulk of >the > >information
RE: Re: Sample Rate [7:36566]
Priscilla, Wow what a great idea. I have been looking into programs that 'read' text on the computer, but they all sound like r2d2, so this is just what I need for drive to and from home. I hope you do more like it. Thanks Rah -Original Message- From: Priscilla Oppenheimer [mailto:[EMAIL PROTECTED]] Sent: 01 March 2002 20:15 To: [EMAIL PROTECTED] Subject: Re: Re: Sample Rate [7:36566] Speaking of sample rates, I am playing with the idea of offering audio training using MP3 files. I have prepared such a training on WAN Troubleshooting. I'd love to get some feedback. This audio training will help people studying for the Support test especially. It's 40 minutes. This means the file is huge, so don't try this at home on a modem line. The file is available for download here: http://www.troubleshootingnetworks.com/audio.html Please send me some feedback. Would you find such a product line helpful? Would you pay for MP3 audio training files? Thanks. Priscilla Oppenheimer At 03:32 PM 2/27/02, John Neiberger wrote: >Yes, this was very distracting! :-) I didn't get any studying done at >all last night! Between checking and answering email, looking for >Clannad MP3s, reading about the Gaelic language just for fun, and >looking up telecom stuff it's a wonder I even powered up a router. I >was able to boot up six routers, erase their configs, and recable them >in preparation for a lab scenario tonight. Not bad for three hours >work. heh heh... > >Thanks, >John > > >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM > >>> >Female opera singers probably hate it when people ask them to sing over >the >phone!? > >OK, have we distracted you enough, John? ;-) Seriously, I think this >was a >great discussion. Thanks to everyone who contributed. > >Priscilla > >At 10:58 AM 2/27/02, David L. Blair wrote: > > > John Neiberger wrote: > > > What I'm trying to find out is why the original 4KHz limit on > > > voice calls was put into place. It sounds like it was simply > > > an arbitrary decision. 4KHz is sufficient for a telephone call > > > and to provide clear calls that included higher frequencies > > > might have added some technical complexities, perhaps. > > > > > > They also added a high-pass filter around 400Hz since most > > > telephones can't reproduce low frequencies well and it also > > > filters out some harmonics of 50-60Hz hum that might show up > > > from time to time. That is concrete reason for including a > > > high-pass filter and I wondered if there was a concrete > > > technical reason for including the 4KHz low-pass filter. From > > > the sounds of it there really isn't a technical issue, 4K is > > > just a nice round number. :-) > > > >I used three sources to answer John's query: "Voice over IP >Fundamentals", > >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and >Data > >Networks". These are great books for anyone wanting to know more >about > >voice technologies. > > > >Interesting Facts and Ideas I came across: > > > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > > > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > >information comes from the middle frequencies. According to Nyquist, >"Human > >voice contains sounds that are more often Middle-pitched frequencies >than > >either High or Low pitched frequencies. > > > >3) Frequencies greater than 4,000 Hz are filter out to limit >crosstalk. > > > >4) During the Analog to Digital conversion voice samples are put >though a > >process called Quantization. Quantization is the process of rounding > >sampled values to the nearest predefined discreet value. Pulse Code > >Modulation (PCM) is a Quantization process. PCM is also used to >achieve 12 > >to 13 bits of voice information in 8 bit words. Two commonly used >PCM's are: > >mu-law (North America), and a-law (Europe). What you hear is not >someone's > >voice, but a representation of their voice. > > > >5) Noise is a major issue when talking about voice quality. Noise is > >constant problem for Analog signals. What is signal and what is >Noise? > >When a Analog signal is amplified so is the Noise, which in turn makes >the > >quality of Analog calls worst as the distance increases. Digital >Calls are > >less suitable to Noise than Analog calls. > > > >6) Delay is a major issue when talking about conversation flow for >two > >reasons: 1) For a conversation
Audio Training, was Re: Re: Sample Rate [7:36566]
I know that I like this sort of product and I wish I'd had more of this sort of thing available when I first started out in this field. For certain types of learners, hearing someone discuss the topic allows it to sink in better than reading alone. I'm definitely that type of learner. One problem I've seen with some audio-based training is the pricing. IMO, they are often priced too high. While audio training is quite nice to have, it's not necessarily as portable as books, for instance, although that is changing quickly. Pretty soon everyone will have a portable MP3 player and/or CD burners and they'd easily be able to go mobile instead of sitting glued in front of their PCs. Regards, John >>> "Priscilla Oppenheimer" 3/1/02 1:15:08 PM >>> Speaking of sample rates, I am playing with the idea of offering audio training using MP3 files. I have prepared such a training on WAN Troubleshooting. I'd love to get some feedback. This audio training will help people studying for the Support test especially. It's 40 minutes. This means the file is huge, so don't try this at home on a modem line. The file is available for download here: http://www.troubleshootingnetworks.com/audio.html Please send me some feedback. Would you find such a product line helpful? Would you pay for MP3 audio training files? Thanks. Priscilla Oppenheimer At 03:32 PM 2/27/02, John Neiberger wrote: >Yes, this was very distracting! :-) I didn't get any studying done at >all last night! Between checking and answering email, looking for >Clannad MP3s, reading about the Gaelic language just for fun, and >looking up telecom stuff it's a wonder I even powered up a router. I >was able to boot up six routers, erase their configs, and recable them >in preparation for a lab scenario tonight. Not bad for three hours >work. heh heh... > >Thanks, >John > > >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM > >>> >Female opera singers probably hate it when people ask them to sing over >the >phone!? > >OK, have we distracted you enough, John? ;-) Seriously, I think this >was a >great discussion. Thanks to everyone who contributed. > >Priscilla > >At 10:58 AM 2/27/02, David L. Blair wrote: > > > John Neiberger wrote: > > > What I'm trying to find out is why the original 4KHz limit on > > > voice calls was put into place. It sounds like it was simply > > > an arbitrary decision. 4KHz is sufficient for a telephone call > > > and to provide clear calls that included higher frequencies > > > might have added some technical complexities, perhaps. > > > > > > They also added a high-pass filter around 400Hz since most > > > telephones can't reproduce low frequencies well and it also > > > filters out some harmonics of 50-60Hz hum that might show up > > > from time to time. That is concrete reason for including a > > > high-pass filter and I wondered if there was a concrete > > > technical reason for including the 4KHz low-pass filter. From > > > the sounds of it there really isn't a technical issue, 4K is > > > just a nice round number. :-) > > > >I used three sources to answer John's query: "Voice over IP >Fundamentals", > >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice and >Data > >Networks". These are great books for anyone wanting to know more >about > >voice technologies. > > > >Interesting Facts and Ideas I came across: > > > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > > > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > >information comes from the middle frequencies. According to Nyquist, >"Human > >voice contains sounds that are more often Middle-pitched frequencies >than > >either High or Low pitched frequencies. > > > >3) Frequencies greater than 4,000 Hz are filter out to limit >crosstalk. > > > >4) During the Analog to Digital conversion voice samples are put >though a > >process called Quantization. Quantization is the process of rounding > >sampled values to the nearest predefined discreet value. Pulse Code > >Modulation (PCM) is a Quantization process. PCM is also used to >achieve 12 > >to 13 bits of voice information in 8 bit words. Two commonly used >PCM's are: > >mu-law (North America), and a-law (Europe). What you hear is not >someone's > >voice, but a representation of their voice. > > > >5) Noise is a major issue when talking about voice quality. Noise is > >constant problem for Analog signals. What is signal and what is >Noise? > >When a Analog signal is amplified so is the Noise, which in turn makes >the > >quality of Analog calls worst as the distance increases. Digital >Calls are > >less suitable to Noise than Analog calls. > > > >6) Delay is a major issue when talking about conversation flow for >two > >reasons: 1) For a conversation to flow normally, the delay is >receiving the > >voice information must be less than 250ms. When the delay is more >than > >250ms, the human receiving the voice message will start to talk >thinking the > >human sendi
Re: Re: Sample Rate [7:36566]--long reply [7:36588]
All right, John-- A couple of years ago (discreet cough), Cisco gave away copies of books as promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw Hill, 2000). GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate and voice sampling: "...Thus, if an analog voice signal reaching up to 3400Hz is to be sampled at the Nyquist rate, the sampling frequency must be at least twice that, or 6800Hz, or samples per second. "Sampling does not have to be done at the Nyquist rate. The Nyquist rate is a minimal requirement to reproduce the input waveform, but sampling can be done at rates higher or lower than the Nyquist rate. If sampling takes place at rates lower than the Nyquist rate, the result is distortion of the waveform known as (italics) aliasing. Aliasing just means that there is more than one output waveform that fits the 'connect the dots' pattern of the samples. There is no aliasing ast the Nyquist rate and above." They go on to point out that, by sampling at a rate above the Nyquist rate, you have more than the minimum required information to reliably reconstruct the voice signal at the destination. This allows you to lose a few samples in transit (not that such things would ever happen, of course) and still have only one possible reconstruction. Sampling at 8000Hz means there is a 4000Hz voice bandwidth (overly generous but convenient because 4 is a power of 2 and that makes it easier to code in a binary system). And from the 8000 samples/sec, each of which sends 1 8-bit word, we have the DS0 of 64000 bps (why only 56000 bps may be usable is a separate issue, having to do with signaling on telephone links). Annlee ""John Neiberger"" wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > This is OT, but the upper limit of human hearing is actually > around 20KHz at best and usually drops to around 16KHz or so. > If your upper limit starts to drop below that you'll start to > notice that it's difficult to hear clearly. (Sorry, in my > other life I'm a sound engineer and musician.) > > I've heard that the 4KHz limit is because there is a low-pass > filter used for voice. I can't remember the exact reason, but > that information plugged into the Nyquist theorem explains--as > Priscilla mentions--why a DS0 is 64Kbps. > > Okay, time to do some serious studying once I'm through being > lazy and drinking this coffee... > > John > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > ([EMAIL PROTECTED]) wrote: > > > At 08:06 PM 2/26/02, Rafay wrote: > > >How do you describe Sample Rate.? > > > > In what context? The term is sometimes used when describing > the analog > > to > > digital process, for example when digitizing voice. Voice > produces an > > analog wave as your lungs and tongue press against the air. > An analog > > wave > > has infinite possible values. Computers can't deal with > infinity. They > > work > > with discreet numbers. The solution is to sample the analog > voice many > > times per second. Sampling means to take a snapshot. > > > > The sample rate is how often the analog wave is sampled. > Nyquist showed > > that you have to sample at twice the rate of the highest > frequency that > > may > > occur in the original data. Most humans don't output (and > can't hear) > > anything about 4 KHz. So sample 8,000 times per second (8Khz) > and the > > result will be good enough. When using a sample rate of 8,000 > KHz, if > > each > > sample is saved in an 8-bit byte, the resulting data rate is > 64 Kbps. > > That's one DS0. Compression allows us to use a smaller data > rate, with > > some > > loss in fidelity. > > > > Priscilla > > > > > > Priscilla Oppenheimer > > http://www.priscilla.com > [EMAIL PROTECTED] > > > > > > > > Get your own "800" number > Voicemail, fax, email, and a lot more > http://www.ureach.com/reg/tag Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36588&t=36588 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
I understand all that, but what I don't remember is why there is a 4KHz low-pass filter on voice lines. I know I've read the reason before but I just can't recall what it was. Was it simply arbitrary? A 4KHz upper limit is obviously sufficient for voice quality. Did someone just pick that limit and filter out everything above it, possibly to filter noise or something? Hmm...this is bugging me now. :-) But I can't be distracted right now, I'm trying to study...which explains why I keep taking time out to check my email and search the internet for MP3s of Clannad. :-) I did just find a killer sampler of Celtic stuff. Very relaxing... John Get your own "800" number Voicemail, fax, email, and a lot more http://www.ureach.com/reg/tag On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED]) wrote: > All right, John-- > > A couple of years ago (discreet cough), Cisco gave away copies of books > as > promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw Hill, > 2000). > GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate and voice > sampling: > > "...Thus, if an analog voice signal reaching up to 3400Hz is to be > sampled > at the Nyquist rate, the sampling frequency must be at least twice that, > or > 6800Hz, or samples per second. > > "Sampling does not have to be done at the Nyquist rate. The Nyquist rate > is > a minimal requirement to reproduce the input waveform, but sampling can > be > done at rates higher or lower than the Nyquist rate. If sampling takes > place > at rates lower than the Nyquist rate, the result is distortion of the > waveform known as (italics) aliasing. Aliasing just means that there is > more > than one output waveform that fits the 'connect the dots' pattern of the > samples. There is no aliasing ast the Nyquist rate and above." > > They go on to point out that, by sampling at a rate above the Nyquist > rate, > you have more than the minimum required information to reliably > reconstruct > the voice signal at the destination. This allows you to lose a few > samples > in transit (not that such things would ever happen, of course) and still > have only one possible reconstruction. Sampling at 8000Hz means there is > a > 4000Hz voice bandwidth (overly generous but convenient because 4 is a > power > of 2 and that makes it easier to code in a binary system). > > And from the 8000 samples/sec, each of which sends 1 8-bit word, we have > the > DS0 of 64000 bps (why only 56000 bps may be usable is a separate issue, > having to do with signaling on telephone links). > > Annlee > ""John Neiberger"" wrote in message > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > This is OT, but the upper limit of human hearing is actually > > around 20KHz at best and usually drops to around 16KHz or so. > > If your upper limit starts to drop below that you'll start to > > notice that it's difficult to hear clearly. (Sorry, in my > > other life I'm a sound engineer and musician.) > > > > I've heard that the 4KHz limit is because there is a low- pass > > filter used for voice. I can't remember the exact reason, but > > that information plugged into the Nyquist theorem explains-- as > > Priscilla mentions--why a DS0 is 64Kbps. > > > > Okay, time to do some serious studying once I'm through being > > lazy and drinking this coffee... > > > > John > > > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > > ([EMAIL PROTECTED]) wrote: > > > > > At 08:06 PM 2/26/02, Rafay wrote: > > > >How do you describe Sample Rate.? > > > > > > In what context? The term is sometimes used when describing > > the analog > > > to > > > digital process, for example when digitizing voice. Voice > > produces an > > > analog wave as your lungs and tongue press against the air. > > An analog > > > wave > > > has infinite possible values. Computers can't deal with > > infinity. They > > > work > > > with discreet numbers. The solution is to sample the analog > > voice many > > > times per second. Sampling means to take a snapshot. > > > > > > The sample rate is how often the analog wave is sampled. > > Nyquist showed > > > that you have to sample at twice the rate of the highest > > frequency that > > > may > > > occur in the original data. Most humans don't output (and > > can't hear) > > > anything about 4 KHz. So sample 8,000 times per second (8Khz) > > and the > > > result will be good enough. When using a sample rate of 8,000 > > KHz, if > > > each > > > sample is saved in an 8-bit byte, the resulting data rate is > > 64 Kbps. > > > That's one DS0. Compression allows us to use a smaller data > > rate, with > > > some > > > loss in fidelity. > > > > > > Priscilla > > > > > > > > > Priscilla Oppenheimer > > > http://www.priscilla.com > > [EMAIL PROTECTED] > > > > > > > > > > > > > > Get your own "800" number > > Voicemail, fax, email,
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
It's the average frequency of human voice. if you look at a conversation on a osciliscope, it averages out to 4k, so you double that and get the 8k sample rate. -- RFC 1149 Compliant. ""John Neiberger"" wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > I understand all that, but what I don't remember is why there > is a 4KHz low-pass filter on voice lines. I know I've read the > reason before but I just can't recall what it was. Was it > simply arbitrary? A 4KHz upper limit is obviously sufficient > for voice quality. Did someone just pick that limit and filter > out everything above it, possibly to filter noise or something? > > Hmm...this is bugging me now. :-) > > But I can't be distracted right now, I'm trying to > study...which explains why I keep taking time out to check my > email and search the internet for MP3s of Clannad. :-) I did > just find a killer sampler of Celtic stuff. Very relaxing... > > John > > > > Get your own "800" number > Voicemail, fax, email, and a lot more > http://www.ureach.com/reg/tag > > > On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED]) > wrote: > > > All right, John-- > > > > A couple of years ago (discreet cough), Cisco gave away > copies of books > > as > > promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw > Hill, > > 2000). > > GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate > and voice > > sampling: > > > > "...Thus, if an analog voice signal reaching up to 3400Hz is > to be > > sampled > > at the Nyquist rate, the sampling frequency must be at least > twice that, > > or > > 6800Hz, or samples per second. > > > > "Sampling does not have to be done at the Nyquist rate. The > Nyquist rate > > is > > a minimal requirement to reproduce the input waveform, but > sampling can > > be > > done at rates higher or lower than the Nyquist rate. If > sampling takes > > place > > at rates lower than the Nyquist rate, the result is > distortion of the > > waveform known as (italics) aliasing. Aliasing just means > that there is > > more > > than one output waveform that fits the 'connect the dots' > pattern of the > > samples. There is no aliasing ast the Nyquist rate and above." > > > > They go on to point out that, by sampling at a rate above the > Nyquist > > rate, > > you have more than the minimum required information to > reliably > > reconstruct > > the voice signal at the destination. This allows you to lose > a few > > samples > > in transit (not that such things would ever happen, of > course) and still > > have only one possible reconstruction. Sampling at 8000Hz > means there is > > a > > 4000Hz voice bandwidth (overly generous but convenient > because 4 is a > > power > > of 2 and that makes it easier to code in a binary system). > > > > And from the 8000 samples/sec, each of which sends 1 8-bit > word, we have > > the > > DS0 of 64000 bps (why only 56000 bps may be usable is a > separate issue, > > having to do with signaling on telephone links). > > > > Annlee > > ""John Neiberger"" wrote in message > > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > > This is OT, but the upper limit of human hearing is actually > > > around 20KHz at best and usually drops to around 16KHz or > so. > > > If your upper limit starts to drop below that you'll start > to > > > notice that it's difficult to hear clearly. (Sorry, in my > > > other life I'm a sound engineer and musician.) > > > > > > I've heard that the 4KHz limit is because there is a low- > pass > > > filter used for voice. I can't remember the exact reason, > but > > > that information plugged into the Nyquist theorem explains-- > as > > > Priscilla mentions--why a DS0 is 64Kbps. > > > > > > Okay, time to do some serious studying once I'm through > being > > > lazy and drinking this coffee... > > > > > > John > > > > > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > > > ([EMAIL PROTECTED]) wrote: > > > > > > > At 08:06 PM 2/26/02, Rafay wrote: > > > > >How do you describe Sample Rate.? > > > > > > > > In what context? The term is sometimes used when > describing > > > the analog > > > > to > > > > digital process, for example when digitizing voice. Voice > > > produces an > > > > analog wave as your lungs and tongue press against the > air. > > > An analog > > > > wave > > > > has infinite possible values. Computers can't deal with > > > infinity. They > > > > work > > > > with discreet numbers. The solution is to sample the > analog > > > voice many > > > > times per second. Sampling means to take a snapshot. > > > > > > > > The sample rate is how often the analog wave is sampled. > > > Nyquist showed > > > > that you have to sample at twice the rate of the highest > > > frequency that > > > > may > > > > occur in the original data. Most humans don't output (and > > > can't hear) > > > > anything about 4 KHz. So sample 8,000 times per second > (8Khz) > > > and the > > > > result will be good enough. When
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
Well, according to a voice manual I have, "Lower bandwidths need less
expensive equipment, which affects the sound quality... this was
determined to be the best cost-performance trade-off".
So I think it was reasonably arbitrary. My ancient but still useful
Tanenbaum (Computer Networks) doesn't seem to give any reason, which sort
of supports that. Horak's "Communications Systems and Networks" agrees
that it "is considered sufficient for voice communications and certainly
is more cost-effective than if full-fidelity voice were to be supported."
And by the way, most references I have refer to the range as being 200 Hz
to 3500 Hz, not 4000Hz - for noise I think - Nyquist assumes a noise-less
channel.
So... you're trying to study, I'm about to give up trying to work today
;-)
JMcL
- Forwarded by Jenny Mcleod/NSO/CSDA on 27/02/2002 05:15 pm -
"John Neiberger"
Sent by: [EMAIL PROTECTED]
27/02/2002 03:17 pm
Please respond to "John Neiberger"
To: [EMAIL PROTECTED]
cc:
Subject:Re: Re: Sample Rate [7:36566]--long reply [7:36566]
I understand all that, but what I don't remember is why there
is a 4KHz low-pass filter on voice lines. I know I've read the
reason before but I just can't recall what it was. Was it
simply arbitrary? A 4KHz upper limit is obviously sufficient
for voice quality. Did someone just pick that limit and filter
out everything above it, possibly to filter noise or something?
Hmm...this is bugging me now. :-)
But I can't be distracted right now, I'm trying to
study...which explains why I keep taking time out to check my
email and search the internet for MP3s of Clannad. :-) I did
just find a killer sampler of Celtic stuff. Very relaxing...
John
Get your own "800" number
Voicemail, fax, email, and a lot more
http://www.ureach.com/reg/tag
On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED])
wrote:
> All right, John--
>
> A couple of years ago (discreet cough), Cisco gave away
copies of books
> as
> promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw
Hill,
> 2000).
> GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate
and voice
> sampling:
>
> "...Thus, if an analog voice signal reaching up to 3400Hz is
to be
> sampled
> at the Nyquist rate, the sampling frequency must be at least
twice that,
> or
> 6800Hz, or samples per second.
>
> "Sampling does not have to be done at the Nyquist rate. The
Nyquist rate
> is
> a minimal requirement to reproduce the input waveform, but
sampling can
> be
> done at rates higher or lower than the Nyquist rate. If
sampling takes
> place
> at rates lower than the Nyquist rate, the result is
distortion of the
> waveform known as (italics) aliasing. Aliasing just means
that there is
> more
> than one output waveform that fits the 'connect the dots'
pattern of the
> samples. There is no aliasing ast the Nyquist rate and above."
>
> They go on to point out that, by sampling at a rate above the
Nyquist
> rate,
> you have more than the minimum required information to
reliably
> reconstruct
> the voice signal at the destination. This allows you to lose
a few
> samples
> in transit (not that such things would ever happen, of
course) and still
> have only one possible reconstruction. Sampling at 8000Hz
means there is
> a
> 4000Hz voice bandwidth (overly generous but convenient
because 4 is a
> power
> of 2 and that makes it easier to code in a binary system).
>
> And from the 8000 samples/sec, each of which sends 1 8-bit
word, we have
> the
> DS0 of 64000 bps (why only 56000 bps may be usable is a
separate issue,
> having to do with signaling on telephone links).
>
> Annlee
> ""John Neiberger"" wrote in message
> [EMAIL PROTECTED]">news:[EMAIL PROTECTED]...
> > This is OT, but the upper limit of human hearing is actually
> > around 20KHz at best and usually drops to around 16KHz or
so.
> > If your upper limit starts to drop below that you'll start
to
> > notice that it's difficult to hear clearly. (Sorry, in my
> > other life I'm a sound engineer and musician.)
> >
> > I've heard that the 4KHz limit is because there is a low-
pass
> > filter used for voice. I can't remember the exact reason,
but
> > that information plugged into the Nyquist theorem explains--
as
> > Priscilla mentions--why a DS0 is 64Kbps.
> >
> > Okay, time to do some serious studying once I'm through
being
> > lazy and drinking this coffee...
> >
> > John
> >
> > On Tue, 26 Feb 2002,
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
>All right, John-- > >A couple of years ago (discreet cough), Cisco gave away copies of books as >promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw Hill, 2000). >GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate and voice >sampling: Well, if it comes from MANY years ago, before even MY time, Nyquist started publishing his work in the 1928 Bell System Technical Journal. I don't have a copy handy, but, if anyone really cares, I do have a copy somewhere of Shannon's 1948 book "The Mathematical Theory of Communications," itself an extension of a BSTJ article, that I think quotes Nyquist. *sigh* and people around here think one is ancient when one refers to a mainframe, or bisync, or analog... > >"...Thus, if an analog voice signal reaching up to 3400Hz is to be sampled >at the Nyquist rate, the sampling frequency must be at least twice that, or >6800Hz, or samples per second. > >"Sampling does not have to be done at the Nyquist rate. The Nyquist rate is >a minimal requirement to reproduce the input waveform, but sampling can be >done at rates higher or lower than the Nyquist rate. If sampling takes place >at rates lower than the Nyquist rate, the result is distortion of the >waveform known as (italics) aliasing. Aliasing just means that there is more >than one output waveform that fits the 'connect the dots' pattern of the >samples. There is no aliasing ast the Nyquist rate and above." > >They go on to point out that, by sampling at a rate above the Nyquist rate, >you have more than the minimum required information to reliably reconstruct >the voice signal at the destination. This allows you to lose a few samples >in transit (not that such things would ever happen, of course) and still >have only one possible reconstruction. Sampling at 8000Hz means there is a >4000Hz voice bandwidth (overly generous but convenient because 4 is a power >of 2 and that makes it easier to code in a binary system). > >And from the 8000 samples/sec, each of which sends 1 8-bit word, we have the >DS0 of 64000 bps (why only 56000 bps may be usable is a separate issue, >having to do with signaling on telephone links). Nyquist's model refers to PCM encoding, representing any sample in 8 bits. Even before we get into compression, there are more bandwidth-efficient, standardized encodings, such as ADPCM at 32 Kbps or less. > >Annlee >""John Neiberger"" wrote in message >[EMAIL PROTECTED]">news:[EMAIL PROTECTED]... >> This is OT, but the upper limit of human hearing is actually >> around 20KHz at best and usually drops to around 16KHz or so. >> If your upper limit starts to drop below that you'll start to >> notice that it's difficult to hear clearly. (Sorry, in my >> other life I'm a sound engineer and musician.) >> >> I've heard that the 4KHz limit is because there is a low-pass >> filter used for voice. I can't remember the exact reason, but >> that information plugged into the Nyquist theorem explains--as >> Priscilla mentions--why a DS0 is 64Kbps. >> >> Okay, time to do some serious studying once I'm through being >> lazy and drinking this coffee... >> >> John >> >> On Tue, 26 Feb 2002, Priscilla Oppenheimer >> ([EMAIL PROTECTED]) wrote: >> >> > At 08:06 PM 2/26/02, Rafay wrote: >> > >How do you describe Sample Rate.? >> > >> > In what context? The term is sometimes used when describing >> the analog >> > to >> > digital process, for example when digitizing voice. Voice >> produces an >> > analog wave as your lungs and tongue press against the air. >> An analog >> > wave >> > has infinite possible values. Computers can't deal with >> infinity. They >> > work >> > with discreet numbers. The solution is to sample the analog >> voice many >> > times per second. Sampling means to take a snapshot. >> > >> > The sample rate is how often the analog wave is sampled. >> Nyquist showed >> > that you have to sample at twice the rate of the highest >> frequency that >> > may >> > occur in the original data. Most humans don't output (and >> can't hear) > > > anything about 4 KHz. So sample 8,000 times per second (8Khz) >> and the >> > result will be good enough. When using a sample rate of 8,000 >> KHz, if >> > each >> > sample is saved in an 8-bit byte, the resulting data rate is >> 64 Kbps. >> > That's one DS0. Compression allows us to use a smaller data >> rate, with >> > some >> > loss in fidelity. >> > > > > Priscilla Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36629&t=36566 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
Two questions to prolongue the distraction: 1. If a 4KHz signal is sampled at 8Khz, it is sampled twice per cycle. Once in the positive half and once in the negative. Doesn't this mean that the value is dependant totally on the point at which it is sampled (and if you sample it as it passes zero both times it won't be heard)? 2. If with ISDN you have 56Kb usable in US and we have 64Kb usable in UK, if I called from UK to you in US, would you still be listening after I hung up. :-) (After this question you'd probably hang up as soon as I called - I know) Gaz (Long week and 4 beers deep - Is it Friday yet?) ""Steven A. Ridder"" wrote in message news:[EMAIL PROTECTED].; > It's the average frequency of human voice. if you look at a conversation on > a osciliscope, it averages out to 4k, so you double that and get the 8k > sample rate. > > -- > > RFC 1149 Compliant. > > > ""John Neiberger"" wrote in message > news:[EMAIL PROTECTED].; > > I understand all that, but what I don't remember is why there > > is a 4KHz low-pass filter on voice lines. I know I've read the > > reason before but I just can't recall what it was. Was it > > simply arbitrary? A 4KHz upper limit is obviously sufficient > > for voice quality. Did someone just pick that limit and filter > > out everything above it, possibly to filter noise or something? > > > > Hmm...this is bugging me now. :-) > > > > But I can't be distracted right now, I'm trying to > > study...which explains why I keep taking time out to check my > > email and search the internet for MP3s of Clannad. :-) I did > > just find a killer sampler of Celtic stuff. Very relaxing... > > > > John > > > > > > > > Get your own "800" number > > Voicemail, fax, email, and a lot more > > http://www.ureach.com/reg/tag > > > > > > On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED]) > > wrote: > > > > > All right, John-- > > > > > > A couple of years ago (discreet cough), Cisco gave away > > copies of books > > > as > > > promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw > > Hill, > > > 2000). > > > GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate > > and voice > > > sampling: > > > > > > "...Thus, if an analog voice signal reaching up to 3400Hz is > > to be > > > sampled > > > at the Nyquist rate, the sampling frequency must be at least > > twice that, > > > or > > > 6800Hz, or samples per second. > > > > > > "Sampling does not have to be done at the Nyquist rate. The > > Nyquist rate > > > is > > > a minimal requirement to reproduce the input waveform, but > > sampling can > > > be > > > done at rates higher or lower than the Nyquist rate. If > > sampling takes > > > place > > > at rates lower than the Nyquist rate, the result is > > distortion of the > > > waveform known as (italics) aliasing. Aliasing just means > > that there is > > > more > > > than one output waveform that fits the 'connect the dots' > > pattern of the > > > samples. There is no aliasing ast the Nyquist rate and above." > > > > > > They go on to point out that, by sampling at a rate above the > > Nyquist > > > rate, > > > you have more than the minimum required information to > > reliably > > > reconstruct > > > the voice signal at the destination. This allows you to lose > > a few > > > samples > > > in transit (not that such things would ever happen, of > > course) and still > > > have only one possible reconstruction. Sampling at 8000Hz > > means there is > > > a > > > 4000Hz voice bandwidth (overly generous but convenient > > because 4 is a > > > power > > > of 2 and that makes it easier to code in a binary system). > > > > > > And from the 8000 samples/sec, each of which sends 1 8-bit > > word, we have > > > the > > > DS0 of 64000 bps (why only 56000 bps may be usable is a > > separate issue, > > > having to do with signaling on telephone links). > > > > > > Annlee > > > ""John Neiberger"" wrote in message > > > news:[EMAIL PROTECTED].; > > > > This is OT, but the upper limit of human hearing is actually > > > > around 20KHz at best and usually drops to around 16KHz or > > so. > > > > If your upper limit starts to drop below that you'll start > > to > > > > notice that it's difficult to hear clearly. (Sorry, in my > > > > other life I'm a sound engineer and musician.) > > > > > > > > I've heard that the 4KHz limit is because there is a low- > > pass > > > > filter used for voice. I can't remember the exact reason, > > but > > > > that information plugged into the Nyquist theorem explains-- > > as > > > > Priscilla mentions--why a DS0 is 64Kbps. > > > > > > > > Okay, time to do some serious studying once I'm through > > being > > > > lazy and drinking this coffee... > > > > > > > > John > > > > > > > > On Tue, 26 Feb 2002, Priscilla Oppenheimer > > > > ([EMAIL PROTECTED]) wrote: > > > > > > > > > At 08:06 PM 2/26/02, Rafay wrote: > > > > > >How do you describe Sample Rate.? > > > > > > > > > > In what con
Re: Re: Sample Rate [7:36566]--long reply [7:36566]
The first question is pretty interesting. It seems that if you took a 1Hz signal and sampled it twice per second but you kept sampling the points where the level was zero, it does at least seem possible. However, that would only be with sine waves, not complex waves. Still, it's an interesting point! :-) John >>> "Gaz" 2/28/02 3:52:35 PM >>> Two questions to prolongue the distraction: 1. If a 4KHz signal is sampled at 8Khz, it is sampled twice per cycle. Once in the positive half and once in the negative. Doesn't this mean that the value is dependant totally on the point at which it is sampled (and if you sample it as it passes zero both times it won't be heard)? 2. If with ISDN you have 56Kb usable in US and we have 64Kb usable in UK, if I called from UK to you in US, would you still be listening after I hung up. :-) (After this question you'd probably hang up as soon as I called - I know) Gaz (Long week and 4 beers deep - Is it Friday yet?) ""Steven A. Ridder"" wrote in message news:[EMAIL PROTECTED].; > It's the average frequency of human voice. if you look at a conversation on > a osciliscope, it averages out to 4k, so you double that and get the 8k > sample rate. > > -- > > RFC 1149 Compliant. > > > ""John Neiberger"" wrote in message > news:[EMAIL PROTECTED].; > > I understand all that, but what I don't remember is why there > > is a 4KHz low-pass filter on voice lines. I know I've read the > > reason before but I just can't recall what it was. Was it > > simply arbitrary? A 4KHz upper limit is obviously sufficient > > for voice quality. Did someone just pick that limit and filter > > out everything above it, possibly to filter noise or something? > > > > Hmm...this is bugging me now. :-) > > > > But I can't be distracted right now, I'm trying to > > study...which explains why I keep taking time out to check my > > email and search the internet for MP3s of Clannad. :-) I did > > just find a killer sampler of Celtic stuff. Very relaxing... > > > > John > > > > > > > > Get your own "800" number > > Voicemail, fax, email, and a lot more > > http://www.ureach.com/reg/tag > > > > > > On Tue, 26 Feb 2002, Annlee Hines ([EMAIL PROTECTED]) > > wrote: > > > > > All right, John-- > > > > > > A couple of years ago (discreet cough), Cisco gave away > > copies of books > > > as > > > promos. One was _IP Telephony_ by Gorlaski and Kolon (McGraw > > Hill, > > > 2000). > > > GOOD BOOK. On pp 77-78 is an explanation of the Nyquist rate > > and voice > > > sampling: > > > > > > "...Thus, if an analog voice signal reaching up to 3400Hz is > > to be > > > sampled > > > at the Nyquist rate, the sampling frequency must be at least > > twice that, > > > or > > > 6800Hz, or samples per second. > > > > > > "Sampling does not have to be done at the Nyquist rate. The > > Nyquist rate > > > is > > > a minimal requirement to reproduce the input waveform, but > > sampling can > > > be > > > done at rates higher or lower than the Nyquist rate. If > > sampling takes > > > place > > > at rates lower than the Nyquist rate, the result is > > distortion of the > > > waveform known as (italics) aliasing. Aliasing just means > > that there is > > > more > > > than one output waveform that fits the 'connect the dots' > > pattern of the > > > samples. There is no aliasing ast the Nyquist rate and above." > > > > > > They go on to point out that, by sampling at a rate above the > > Nyquist > > > rate, > > > you have more than the minimum required information to > > reliably > > > reconstruct > > > the voice signal at the destination. This allows you to lose > > a few > > > samples > > > in transit (not that such things would ever happen, of > > course) and still > > > have only one possible reconstruction. Sampling at 8000Hz > > means there is > > > a > > > 4000Hz voice bandwidth (overly generous but convenient > > because 4 is a > > > power > > > of 2 and that makes it easier to code in a binary system). > > > > > > And from the 8000 samples/sec, each of which sends 1 8-bit > > word, we have > > > the > > > DS0 of 64000 bps (why only 56000 bps may be usable is a > > separate issue, > > > having to do with signaling on telephone links). > > > > > > Annlee > > > ""John Neiberger"" wrote in message > > > news:[EMAIL PROTECTED].; > > > > This is OT, but the upper limit of human hearing is actually > > > > around 20KHz at best and usually drops to around 16KHz or > > so. > > > > If your upper limit starts to drop below that you'll start > > to > > > > notice that it's difficult to hear clearly. (Sorry, in my > > > > other life I'm a sound engineer and musician.) > > > > > > > > I've heard that the 4KHz limit is because there is a low- > > pass > > > > filter used for voice. I can't remember the exact reason, > > but > > > > that information plugged into the Nyquist theorem explains-- > > as > > > > Priscilla mentions--why a DS0 is 64Kbps. > > > > > > > >
Re: Audio Training, was Re: Re: Sample Rate [7:36566]
I just heard Priscilla's audio training on WAN Troubleshooting and I think it's great. I spent the time listening to the audio file while organizing my desk at work; very convenient!!. I'd definitely pay for audio training if it was available. Priscilla, do you have any other audio training files besides WAN Troubleshooting? ""John Neiberger"" wrote in message news:[EMAIL PROTECTED].; > I know that I like this sort of product and I wish I'd had more of this > sort of thing available when I first started out in this field. For > certain types of learners, hearing someone discuss the topic allows it > to sink in better than reading alone. I'm definitely that type of > learner. > > One problem I've seen with some audio-based training is the pricing. > IMO, they are often priced too high. While audio training is quite nice > to have, it's not necessarily as portable as books, for instance, > although that is changing quickly. Pretty soon everyone will have a > portable MP3 player and/or CD burners and they'd easily be able to go > mobile instead of sitting glued in front of their PCs. > > Regards, > John > > >>> "Priscilla Oppenheimer" 3/1/02 1:15:08 PM > >>> > Speaking of sample rates, I am playing with the idea of offering audio > > training using MP3 files. I have prepared such a training on WAN > Troubleshooting. I'd love to get some feedback. This audio training > will > help people studying for the Support test especially. It's 40 minutes. > This > means the file is huge, so don't try this at home on a modem line. The > file > is available for download here: > > http://www.troubleshootingnetworks.com/audio.html > > Please send me some feedback. Would you find such a product line > helpful? > Would you pay for MP3 audio training files? > > Thanks. > > Priscilla Oppenheimer > > > > > > At 03:32 PM 2/27/02, John Neiberger wrote: > >Yes, this was very distracting! :-) I didn't get any studying done > at > >all last night! Between checking and answering email, looking for > >Clannad MP3s, reading about the Gaelic language just for fun, and > >looking up telecom stuff it's a wonder I even powered up a router. I > >was able to boot up six routers, erase their configs, and recable > them > >in preparation for a lab scenario tonight. Not bad for three hours > >work. heh heh... > > > >Thanks, > >John > > > > >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM > > >>> > >Female opera singers probably hate it when people ask them to sing > over > >the > >phone!? > > > >OK, have we distracted you enough, John? ;-) Seriously, I think this > >was a > >great discussion. Thanks to everyone who contributed. > > > >Priscilla > > > >At 10:58 AM 2/27/02, David L. Blair wrote: > > > > John Neiberger wrote: > > > > What I'm trying to find out is why the original 4KHz limit on > > > > voice calls was put into place. It sounds like it was simply > > > > an arbitrary decision. 4KHz is sufficient for a telephone call > > > > and to provide clear calls that included higher frequencies > > > > might have added some technical complexities, perhaps. > > > > > > > > They also added a high-pass filter around 400Hz since most > > > > telephones can't reproduce low frequencies well and it also > > > > filters out some harmonics of 50-60Hz hum that might show up > > > > from time to time. That is concrete reason for including a > > > > high-pass filter and I wondered if there was a concrete > > > > technical reason for including the 4KHz low-pass filter. From > > > > the sounds of it there really isn't a technical issue, 4K is > > > > just a nice round number. :-) > > > > > >I used three sources to answer John's query: "Voice over IP > >Fundamentals", > > >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice > and > >Data > > >Networks". These are great books for anyone wanting to know more > >about > > >voice technologies. > > > > > >Interesting Facts and Ideas I came across: > > > > > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > > > > > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > > >information comes from the middle frequencies. According to > Nyquist, > >"Human > > >voice contains sounds that are more often Middle-pitched > frequencies > >than > > >either High or Low pitched frequencies. > > > > > >3) Frequencies greater than 4,000 Hz are filter out to limit > >crosstalk. > > > > > >4) During the Analog to Digital conversion voice samples are put > >though a > > >process called Quantization. Quantization is the process of > rounding > > >sampled values to the nearest predefined discreet value. Pulse Code > > >Modulation (PCM) is a Quantization process. PCM is also used to > >achieve 12 > > >to 13 bits of voice information in 8 bit words. Two commonly used > >PCM's are: > > >mu-law (North America), and a-law (Europe). What you hear is not > >someone's > > >voice, but a representation of their voice. > > > > > >5) Noise is a major issue when talking about voice qua
Re: Audio Training, was Re: Re: Sample Rate [7:36566]
As you know with MP3, you won't really make much money because it's so easily copy-able. You'd be on Napster, Morpheus, WinMX in no time. If you don't mind the piracy, I think it would be a great idea! -- RFC 1149 Compliant. ""Audy Bautista"" wrote in message news:[EMAIL PROTECTED].; > I just heard Priscilla's audio training on WAN Troubleshooting and I think > it's great. I spent the time listening to the audio file while organizing > my desk at work; very convenient!!. I'd definitely pay for audio training > if it was available. > > Priscilla, do you have any other audio training files besides WAN > Troubleshooting? > > > ""John Neiberger"" wrote in message > news:[EMAIL PROTECTED].; > > I know that I like this sort of product and I wish I'd had more of this > > sort of thing available when I first started out in this field. For > > certain types of learners, hearing someone discuss the topic allows it > > to sink in better than reading alone. I'm definitely that type of > > learner. > > > > One problem I've seen with some audio-based training is the pricing. > > IMO, they are often priced too high. While audio training is quite nice > > to have, it's not necessarily as portable as books, for instance, > > although that is changing quickly. Pretty soon everyone will have a > > portable MP3 player and/or CD burners and they'd easily be able to go > > mobile instead of sitting glued in front of their PCs. > > > > Regards, > > John > > > > >>> "Priscilla Oppenheimer" 3/1/02 1:15:08 PM > > >>> > > Speaking of sample rates, I am playing with the idea of offering audio > > > > training using MP3 files. I have prepared such a training on WAN > > Troubleshooting. I'd love to get some feedback. This audio training > > will > > help people studying for the Support test especially. It's 40 minutes. > > This > > means the file is huge, so don't try this at home on a modem line. The > > file > > is available for download here: > > > > http://www.troubleshootingnetworks.com/audio.html > > > > Please send me some feedback. Would you find such a product line > > helpful? > > Would you pay for MP3 audio training files? > > > > Thanks. > > > > Priscilla Oppenheimer > > > > > > > > > > > > At 03:32 PM 2/27/02, John Neiberger wrote: > > >Yes, this was very distracting! :-) I didn't get any studying done > > at > > >all last night! Between checking and answering email, looking for > > >Clannad MP3s, reading about the Gaelic language just for fun, and > > >looking up telecom stuff it's a wonder I even powered up a router. I > > >was able to boot up six routers, erase their configs, and recable > > them > > >in preparation for a lab scenario tonight. Not bad for three hours > > >work. heh heh... > > > > > >Thanks, > > >John > > > > > > >>> "Priscilla Oppenheimer" 2/27/02 3:04:13 PM > > > >>> > > >Female opera singers probably hate it when people ask them to sing > > over > > >the > > >phone!? > > > > > >OK, have we distracted you enough, John? ;-) Seriously, I think this > > >was a > > >great discussion. Thanks to everyone who contributed. > > > > > >Priscilla > > > > > >At 10:58 AM 2/27/02, David L. Blair wrote: > > > > > John Neiberger wrote: > > > > > What I'm trying to find out is why the original 4KHz limit on > > > > > voice calls was put into place. It sounds like it was simply > > > > > an arbitrary decision. 4KHz is sufficient for a telephone call > > > > > and to provide clear calls that included higher frequencies > > > > > might have added some technical complexities, perhaps. > > > > > > > > > > They also added a high-pass filter around 400Hz since most > > > > > telephones can't reproduce low frequencies well and it also > > > > > filters out some harmonics of 50-60Hz hum that might show up > > > > > from time to time. That is concrete reason for including a > > > > > high-pass filter and I wondered if there was a concrete > > > > > technical reason for including the 4KHz low-pass filter. From > > > > > the sounds of it there really isn't a technical issue, 4K is > > > > > just a nice round number. :-) > > > > > > > >I used three sources to answer John's query: "Voice over IP > > >Fundamentals", > > > >"Cisco Voice over Frame Relay, ATM, and IP", and Integrating Voice > > and > > >Data > > > >Networks". These are great books for anyone wanting to know more > > >about > > > >voice technologies. > > > > > > > >Interesting Facts and Ideas I came across: > > > > > > > >1) Human hearing is in the range of 200 Hz to 20,000 Hz > > > > > > > >2) Human speech is in the range of 250 Hz to 10,000 Hz. Most of the > > > >information comes from the middle frequencies. According to > > Nyquist, > > >"Human > > > >voice contains sounds that are more often Middle-pitched > > frequencies > > >than > > > >either High or Low pitched frequencies. > > > > > > > >3) Frequencies greater than 4,000 Hz are filter out to limit > > >crosstalk. > > > > > > > >4) During the Analog to Digital conversion voice sample
