Signal reconstruction

Started by lamerooze in comp.dsp14 years ago 2 replies

Hi guys, I am writing some code in MATLAB in which you can create a new sound sample from two sound samples. The program is able to create a...

Hi guys, I am writing some code in MATLAB in which you can create a new sound sample from two sound samples. The program is able to create a sample such that the sample's envelope will be made up of varying envelope contributions from the two samples (ie. the envelope will be made up of 60% of the envelope of Sample 1 and 40% of the envelope of sample 2). Also, the sample's fine structure will ...


Re: Software PLL (SPLL)

Started by Noway2 in comp.dsp16 years ago

Thank you both. I think I understand things a bit better than before. I think the idea of using a gross zero crossing aproximator to...

Thank you both. I think I understand things a bit better than before. I think the idea of using a gross zero crossing aproximator to drive the PLL into the neighborhood of the incoming frequency and then letting the loop filter / NCO take over sounds like a very workable idea. Tim, as I indicated, I have been planning this as a future sub project for a while now. I have played around w...


Audio expert question please!

Started by slakka in comp.dsp13 years ago 5 replies

Dear Newsgroups, I gotta one of a kind interview on mini dv. When played back it both looks and sounds terrific!! Problem occurs when editing...

Dear Newsgroups, I gotta one of a kind interview on mini dv. When played back it both looks and sounds terrific!! Problem occurs when editing interview onto dvd. Audio becomes very muddy/unusable I could really use any suggestions on how I may cure this please?? Thanks in advance!! pittwindmill@see-below-no-spam hotmail.com


Setup doc/drivers for Loughborough Sound Images PC/C32M dsp card

Started by bouced in comp.dsp11 years ago 2 replies

Hi, I am trying to set-up a TI-TMS320 based DSP card under Windows 98. The card is a Loughborough Sound Images PC/C32M connected through an...

Hi, I am trying to set-up a TI-TMS320 based DSP card under Windows 98. The card is a Loughborough Sound Images PC/C32M connected through an ISA bus. If I plug my card on and start Windows, the device is not detected by the OS. I'm trouble in because I neither have a piece of documentation nor the card drivers. Moreover the card manufacturer no more exists. So, I was wondering if someone ...


Spatial Aliasing and Delay and Sum Beamformer

Started by RIH7777 in comp.dsp8 years ago 6 replies

I'm trying to understand how spatial aliasing will give problems with a simple delay/sum beamformer. I understand the the principle of...

I'm trying to understand how spatial aliasing will give problems with a simple delay/sum beamformer. I understand the the principle of aliasing/Nyquist rate. Lets take simplest case: linear microphone array with 2 mics located d meters apart along the x axis. Sound source is far field and monochromatic with wavelength lambda. Theta is the angle between mic array and sound sour


GSM AMR on blackfin.

Started by Akshay Mishra in comp.dsp15 years ago 4 replies

I am working on a GSM AMR codec. I have optimized the codec for real time operation on a Blackfin DSP on the 533 evaluation board. I capture the...

I am working on a GSM AMR codec. I have optimized the codec for real time operation on a Blackfin DSP on the 533 evaluation board. I capture the input and play it out. I have put the system in two configurations: (a) No codec, plain loopback of the captured sound on the output speaker. I get good sound. (b) When GSM AMR is involved in the operation, the output has the input but it is subdu...


MELP quality

Started by Brian in comp.dsp16 years ago 13 replies

I have listened to MELP vocoder and found that its consonants are sometimes too weak, sometimes it produces hoarse speech and clicks, and its...

I have listened to MELP vocoder and found that its consonants are sometimes too weak, sometimes it produces hoarse speech and clicks, and its pitch shakes/vibrates which makes it sounds unnatural. Is that typical to vocoders at that bitrate (2400 bps)? Can the above be improved or eliminated? Can anyone recommend a better vocoder at that rate? Thanks, BHD


Re: Need som help on QPSK Modulation

Started by Allan Herriman in comp.dsp18 years ago

On 16 Dec 2003 22:14:12 -0800, parth175@yahoo.co.in (Parthasarathy) wrote: > Hello > > I have to perform QPSK Modulation, on a 5...

On 16 Dec 2003 22:14:12 -0800, parth175@yahoo.co.in (Parthasarathy) wrote: > Hello > > I have to perform QPSK Modulation, on a 5 Mbps data using a 70 > Mhz carrier. > Is this possible, Yes. It also sounds reasonable, although if your final Tx frequency is quite high (microwave) and you are using an analog modulator, I suggest starting at a much higher first IF (in upper UHF)


Acoustic Measuerment

Started by Major Misunderstanding in comp.dsp15 years ago 3 replies

There is such a thing as acoustic impedance. Is it possible to have sounds that humans can hear that cannot be picked up with a mic? Could this...

There is such a thing as acoustic impedance. Is it possible to have sounds that humans can hear that cannot be picked up with a mic? Could this be because of a miss-match between acoustic impedances? ie a bit like an amplifier and load.Would using a 'hood' a bit like an ear improve things? M. -- Posted via a free Usenet account from http://www.teranews.com


Re: STUPIDENT question on the difference of the images

Started by rickman in comp.dsp11 years ago

It would seem to me that the need for using anything other than the luminance would depend on the application. You don't provide any application...

It would seem to me that the need for using anything other than the luminance would depend on the application. You don't provide any application information. Why do you think you need to use color for your difference? This sounds a bit like a STUPIDENT question... Not that I'm not willing to help... I just couldn't resist saying that. ;^) Rick


Re: Guidelines for a PhD

Started by Rune Allnor in comp.dsp18 years ago

vimal_bhatia2@yahoo.com (Vimal) wrote in message news: ... > Rune, I have found a course on Financial Topics in Product Design > (talks about...

vimal_bhatia2@yahoo.com (Vimal) wrote in message news: ... > Rune, I have found a course on Financial Topics in Product Design > (talks about profit/loss/cash-flow/budget etc) in my department. > Sounds good, however I may not get paper for doing taking up this > course!! Over here, students that are presently enrolled at the university are


1250MHz ADC

Started by glen herrmannsfeldt in comp.dsp9 years ago 5 replies

While following the discussion about 8b10b coding, I happened to see a page on 10GbaseT. It seems that 1250MHz 7 bit ADCs (and 10 bit DACs) are...

While following the discussion about 8b10b coding, I happened to see a page on 10GbaseT. It seems that 1250MHz 7 bit ADCs (and 10 bit DACs) are required. Much filtering and echo cancellation is done to the signal, so more bits are needed than actually get sent down the line. I haven't followed ADC technology, but that sounds pretty fast. -- glen


impeding DSP lectures

Started by Philip Newman in comp.dsp18 years ago 2 replies

I will be starting my DSP lectures again for this term (I can't wait, honestly!) and the lecturer has promised that he will be teaching...

I will be starting my DSP lectures again for this term (I can't wait, honestly!) and the lecturer has promised that he will be teaching us statistics involved in DSP which sounds marvellous! No doubt he will go about it in his usual drone and then screw us over during exam time. So my question is this: Is there a suitable text that deals with such issues? either book or website would be g...


Measuring Sampling frequency of a Audio signal

Started by rpawade in comp.dsp14 years ago 1 reply

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon...

Hello everybody, I have Denon AVR-2307CI (Audio/Video reciever) with me. I am feeding graphics card's output to the HDMI input for Denon reciever. Now the problem is the sampling frequency of the signal is stuck to 48KHz on Denon screen. I tried changing the sampling frequency of signal using sound card's control panel (Sound cards tried: X-Mystique and M-Audio) but it didnt help . Can any...


How to use a microphones' impulse response to correct recordings made with it (in MATLAB)?

Started by Anonymous in comp.dsp18 years ago 1 reply

I have the impulse response IR (time domain) of a recording microphone (actually a whole set of IRs for the full range of angles of...

I have the impulse response IR (time domain) of a recording microphone (actually a whole set of IRs for the full range of angles of sound incidence). I also have recordings of animal vocalisations (from known angles of sound incidence) made with that microphone. Now I want to use MATLAB to calculate what the initial signal looked like (in the time domain) by removing the amplitude and phase ...


Is IF Data real or complex

Started by b2508 in comp.dsp6 years ago 14 replies

Well, probably sounds stupid but I get quite confused on this. In my specification I get Intermediate Frequency data as real from third party RF...

Well, probably sounds stupid but I get quite confused on this. In my specification I get Intermediate Frequency data as real from third party RF Front end. Isn't RF Data suppose to be real? Then I assume it was mixed with some LO to downconvert it from RF to IF. How this this result in a real signal? --------------------------------------- Posted through http://www.DSPRelated.com


How to decimate/nterpolate in frequency domain and recover a proper causal time signal ?

Started by scoubi in comp.dsp15 years ago 16 replies

Hi everyone ! I'm new in this group and quite intermediate in DSP applications. I'm facing the following problem in the framework of my...

Hi everyone ! I'm new in this group and quite intermediate in DSP applications. I'm facing the following problem in the framework of my study. I've got some 4 seconds recorded sounds (44100Hz, 16bits) for which I'd like to modify the FFT spectrum. Actually, I'd like to replace at a constant rate (e.g. 3 values every four values, from the DC component) some initial values with "new" value...


Convolution via FFT

Started by Robert A. in comp.dsp15 years ago 1 reply

Hi guys, I just added convolution via FFT to my audio program so I can convolve two arbitrary signals (sounds), I have a few questions. In...

Hi guys, I just added convolution via FFT to my audio program so I can convolve two arbitrary signals (sounds), I have a few questions. In Numerical Recipes in C it says to setup the response in wrap-around order, in other tutorials I don't see that being done. Should I be doing that ? Also, if I convolve two signals both in the range [-1.0,1.0] is it normal for the result to be o...


Modern radio

Started by glen herrmannsfeldt in comp.dsp7 years ago 5 replies

From Wikipedia [[Amplitude Modulation]]: "In modern radio systems, modulated signals are generated via digital signal processing...

From Wikipedia [[Amplitude Modulation]]: "In modern radio systems, modulated signals are generated via digital signal processing (DSP)." While this sounds right, how much DSP is really done in radio transmitters? Are they now using DSP modulators in place of mixers for AM and FM broadcasting? Even more, is there a reference for it? -- glen


Code size problem on TigerSHARC

Started by Carlos in comp.dsp18 years ago 7 replies

Hi, I've been asked to look at a problem where developed DSP code will not fit on the target (TigerSharcs, i.e. TS101). What are my...

Hi, I've been asked to look at a problem where developed DSP code will not fit on the target (TigerSharcs, i.e. TS101). What are my options: 1. Rewrite code!! (Not an option I believe) 2. Use memory overlays. OK, I've seen ADI's docs on this. The concept sounds simple, but the examples look "scary". All their examples are in assembler, which I'm not too au fait with. Can you do this in C...