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From: disney_snoopy on 30 Nov 2009 03:07 On Nov 30, 2:20 pm, glen herrmannsfeldt <g... (a)ugcs.caltech.edu> wrote:> Randy Yates <ya... (a)ieee.org> wrote:> > glen, even his specification is incomplete. The input audio should > > be STEREO, i.e., left and right. > > I thought about mentioning that, but decided not to. > > Otherwise, it is much easier to create the stereo signal with > a mono source... > > -- glen Coding below is i use to code the pilot freq. x3 = ((0.5*sin(2*pi*19000*t1))+ (randn(size(t1)))); [f3] = pmtm(x3,4,65536,fs); While for the L+R i cant use this to code.
From: Randy Yates on 30 Nov 2009 10:41 disney_snoopy <disney_snoopy (a)hotmail.com> writes:> On Nov 30, 1:42 pm, Randy Yates <ya... (a)ieee.org> wrote:>> glen herrmannsfeldt <g... (a)ugcs.caltech.edu> writes:>> > disney_snoopy <disney_sno... (a)hotmail.com> wrote:>> >> >> my input audio signal is 1kHz and my sampling freq is 200kHz. >> >> Please help! >> >> > Generate sum and difference, generate pilot signal, generate AMSC >> > subcarrier with the appropriate phase, add all three together. >> >> > A little more complicated since 200 isn't a multiple of 38. >> >> glen, even his specification is incomplete. The input audio should >> be STEREO, i.e., left and right. >> -- >> Randy Yates % "Maybe one day I'll feel her cold embrace, >> Digital Signal Labs % and kiss her interface, >> mailto://ya... (a)ieee.org % til then, I'll leave her alone."http://www.digitalsignallabs.com% 'Yours Truly, 2095', *Time*, ELO > > Hi Glen and Randy, > > This is my DSP project. > what i am doing is use a software to prove the FM stereo is work which > i > can get the outlook structure to this FM stereo broadcast. After get > the > FM stereo structured, apply an audio frequency to this FM stereo and > calculate out the THD. > > What i know from here is the L+R region is 50Hz to 15kHz. How to > implement > it to the Matlab i really no idea. Hence, now im finding for help or > comment on which command i should use to implement the L+R region. > I tried to use multiband coding for this region but i found that i > cant code my pilot frequency. > The only thing that i can do in this Matlab now is only the pilot > freq. > > So now do you all get what i mean and what i want? Not really. I'm not good at inferring meaning from poorly-constructed sentences. However, I'm almost positive that if you read this: http://en.wikipedia.org/wiki/FM_broadcasting#Modulation_characteristics and can really understand it, especially the section entitled "FM stereo," then you will have answered many of your own questions. It seems you've put the cart before the horse and you are trying to implement a system that you really don't understand - that will never work. -- Randy Yates % "And all you had to say Digital Signal Labs % was that you were mailto://yates (a)ieee.org % gonna stay."http://www.digitalsignallabs.com % Getting To The Point', *Balance of Power*, ELO
From: Randy Yates on 30 Nov 2009 10:56 disney_snoopy <disney_snoopy (a)hotmail.com> writes:> On Nov 30, 2:20 pm, glen herrmannsfeldt <g... (a)ugcs.caltech.edu> wrote:>> Randy Yates <ya... (a)ieee.org> wrote:>> > glen, even his specification is incomplete. The input audio should >> > be STEREO, i.e., left and right. >> >> I thought about mentioning that, but decided not to. >> >> Otherwise, it is much easier to create the stereo signal with >> a mono source... >> >> -- glen > > Coding below is i use to code the pilot freq. > > x3 = ((0.5*sin(2*pi*19000*t1))+ (randn(size(t1)))); > > [f3] = pmtm(x3,4,65536,fs); [Pardon me, glen, for stepping in here...] You probably don't want to do things this way. What you probably want to do is first create the sum (s(t)), difference (d(t)), and pilot (p(t)) signals, then sum them to get a composite baseband signal: b(t) = s(t) + d(t) + p(t) Then FM-modulate b(t). Then add the (possibly complex) noise as the final step. Rethinking this whole situation, this problem is probably too advanced for you at this stage. You need to try a few simpler experiments in Matlab first and learn some of the terininology, concepts, and techniques, before tackling a full FM stereo modulator. -- Randy Yates % "Maybe one day I'll feel her cold embrace, Digital Signal Labs % and kiss her interface, mailto://yates (a)ieee.org % til then, I'll leave her alone."http://www.digitalsignallabs.com % 'Yours Truly, 2095', *Time*, ELO
From: Jerry Avins on 30 Nov 2009 11:26 Randy Yates wrote: > disney_snoopy <disney_snoopy (a)hotmail.com> writes:> >> On Nov 30, 1:42 pm, Randy Yates <ya... (a)ieee.org> wrote:>>> glen herrmannsfeldt <g... (a)ugcs.caltech.edu> writes:>>>> disney_snoopy <disney_sno... (a)hotmail.com> wrote:>>>>> my input audio signal is 1kHz and my sampling freq is 200kHz. >>>>> Please help! >>>> Generate sum and difference, generate pilot signal, generate AMSC >>>> subcarrier with the appropriate phase, add all three together. >>>> A little more complicated since 200 isn't a multiple of 38. >>> glen, even his specification is incomplete. The input audio should >>> be STEREO, i.e., left and right. >>> -- >>> Randy Yates % "Maybe one day I'll feel her cold embrace, >>> Digital Signal Labs % and kiss her interface, >>> mailto://ya... (a)ieee.org % til then, I'll leave her alone."http://www.digitalsignallabs.com% 'Yours Truly, 2095', *Time*, ELO >> Hi Glen and Randy, >> >> This is my DSP project. >> what i am doing is use a software to prove the FM stereo is work which >> i >> can get the outlook structure to this FM stereo broadcast. After get >> the >> FM stereo structured, apply an audio frequency to this FM stereo and >> calculate out the THD. >> >> What i know from here is the L+R region is 50Hz to 15kHz. How to >> implement >> it to the Matlab i really no idea. Hence, now im finding for help or >> comment on which command i should use to implement the L+R region. >> I tried to use multiband coding for this region but i found that i >> cant code my pilot frequency. >> The only thing that i can do in this Matlab now is only the pilot >> freq. >> >> So now do you all get what i mean and what i want? > > Not really. I'm not good at inferring meaning from poorly-constructed > sentences. > > However, I'm almost positive that if you read this: > > http://en.wikipedia.org/wiki/FM_broadcasting#Modulation_characteristics > > and can really understand it, especially the section entitled "FM > stereo," then you will have answered many of your own questions. > > It seems you've put the cart before the horse and you are trying > to implement a system that you really don't understand - that will > never work. I have to second that on the evidence that L+R is the mono signal that any receiver will pick up. The multiplexed part riding on the 38 KHz carrier is L-R. Stereo vinyl disks are the same way. L+R is the horizontal movement of the stylus that a mono player will respond to, while L-R is the encoded in vertical motion. That's what makes it _compatible_ stereo. Jerry -- Engineering is the art of making what you want from things you can get. ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
From: glen herrmannsfeldt on 30 Nov 2009 14:35
Jerry Avins <jya (a)ieee.org> wrote:(snip) > I have to second that on the evidence that L+R is the mono signal that > any receiver will pick up. The multiplexed part riding on the 38 KHz > carrier is L-R. Stereo vinyl disks are the same way. L+R is the > horizontal movement of the stylus that a mono player will respond to, > while L-R is the encoded in vertical motion. That's what makes it > _compatible_ stereo. The story I remember being that horizontal motion for mono records (vinyl now, but I believe originally something else) such that bearing noise was not picked up. Somehow that problem was solved before stereo came out. The other description is that one wall of the groove has the left channel, the other the right channel. As I understand it, the record cutters use L+R and L-R, but playback is done L and R with the appropriate coordinate rotation. Also, FM stereo is described with the 38kHz subcarrier, but, since the amplitude of the subcarrier is just right, also as switching between the two channels at 38kHz. (Filter out any higher order components.) With a sample rate a multiple of 38kHz it would be especially easy to generate the signal. -- glen |