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  • TMS320C64x Quick start reference for porgrammers

    - by osgx
    Hello Is thare any quickstart guide for programmers for writing DSP-accelerated appliations for TMS320C64x? I have a program with custom algorythm (not the fft, or usial filtering) and I want to accelerate it using multi-DSP coprocessor. So, how should I modify source to move computation from main CPU to DSPs? What limitations are there for DSP-running code? I have some experience with CUDA. In CUDA I should mark every function as being host, device, or entry point for device (kernel). There are also functions to start kernels and to upload/download data to/from GPU. There are also some limitations, for device code, described in CUDA Reference manual. I hope, there is an similar interface and a documentation for DSP.

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  • Can anyone give me a sample DSP script in C/C++

    - by Andrew
    Im working on a (Audio) DSP project and just wondering if there are any sample (Open source) DSP example that are written in c or c++, for my MSP430 Chip. I just want something as a guideline so i can program my own script using the ACD and DCA on my board for sampling. http://focus.ti.com/docs/toolsw/folders/print/msp-exp430f5438.html Thats my board, MSP430F5438 Experimenter Board, from what i herd it can run dsp script via the USB connection with the computer. Im using CCS ( From TI, code composer studio) and Octave/Matlab. Just any DSP example scripts or sites that will help me create my own would be appreciated. What im tying to do, Partial audio (sampled) track -- Nyquist rate sampling -- over- and undersampling -- reconstruction of the audio track.

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  • Using Cepstrum for PDA

    - by CziX
    Hey, I am currently deleveloping a algorithm to decide wheather or not a frame is voiced or unvoiced. I am trying to use the Cepstrum to discriminate between these two situations. I use MATLAB for my implementation. I have some problems, saying something generally about the frame, but my currently implementation looks like (I'm award of the MATLAB has the function rceps, but this haven't worked for either): ceps = abs(ifft(log10(abs(fft(frame.*window')).^2+eps))); Can anybody give me a small demo, that will convert the frame to the power cepstrum, so a single lollipop at the pitch frequency. For instance use this code to generate the frequency. fs = 8000; timelength = 25e-3; freq = 500; k = 0:1/fs:timelength-(1/fs); s = 0.8*sin(2*pi*freq*k); Thanks.

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  • Simulink sim of rician channel ber process

    - by bob
    Hi, I'm learning simulink and I want to use the rician channle block from the communications blockset. I'm told I need to change the format format. Would anyone have some sample code where they used the rician channels in simulink to model a bit error rate process?

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  • .NET Library to Identify Pitches

    - by Antoni
    I'd like to write a simple program(preferably in C#) to which I sing a pitch using a mic and the program identifies to which musical note that pitch corresponds. Thank you very much for your prompt responses. I clarify: I'd like a (preferably .NET) library that would identify the notes I sing. I'd like that such a library: Identifies a note when I sing(a note from the chromatic scale). Tells me how much I'm off from the closest note. I intend to use such a library to sing one note a time.

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  • Beagleboard: How do I send/receive data to/from the DSP?

    - by snakile
    I have a beagleboard with TMS320C64x+ DSP. I'm working on an image processing beagleboard application. Here's how it's going to work: The ARM reads an image from a file and put the image in a 2D array. The arm sends the matrix to the DSP. The DSP receives the matrix. The DSP performs the image processing algorithm on the received matrix (the algorithm code uses about 5MB of dynamically allocated memory). The DSP sends the processed image (matrix) to the ARM. The arm received the matrix. The arm saved the processed image to a file. I'v already written the code for steps 1,3,5. What is the easiest way to do steps 3+4 (sending the data)? Code examples are welcome.

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  • Using imtophat in Matlab

    - by jaff12
    I'm trying to do top hat filtering in matlab. The imtophat function looks promising, but I have no idea how to use it. I dont have a lot of work with Matlab before. I am trying to look find basically small spots several pixels wide that are local max in my 2 dimensional array.

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  • Convolving two signals

    - by John Elway
    Calculate the convolution of the following signals (your answer will be in the form of an equation): h[n] = delta[n-1] + delta[n+1], x[n] = delta[n-a] + delta[n+b] I'm lost as to what I do with h and x. Do I simply multiply them? h[n]*x[n]? I programmed convolution with several types of blurs and edge detectors, but I don't see how to translate that knowledge to this problem. please help!

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  • What's the fastest way to approximate the period of data using Octave?

    - by John
    I have a set of data that is periodic (but not sinusoidal). I have a set of time values in one vector and a set of amplitudes in a second vector. I'd like to quickly approximate the period of the function. Any suggestions? Specifically, here's my current code. I'd like to approximate the period of the vector x(:,2) against the vector t. Ultimately, I'd like to do this for lots of initial conditions and calculate the period of each and plot the result. function xdot = f (x,t) xdot(1) =x(2); xdot(2) =-sin(x(1)); endfunction x0=[1;1.75]; #eventually, I'd like to try lots of values for x0(2) t = linspace (0, 50, 200); x = lsode ("f", x0, t) plot(x(:,1),x(:,2)); Thank you! John

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  • BlackBerry Technical Specification

    - by Sam
    I'm having trouble locating BlackBerry techical specifications and their website is a mess. They also don't have a number that I can use to easily contact them. This isn't exactly a coding question, but what does the BlackBerry audio API look like, and where can I get technical specifications on audio? Specifically, I'm trying to find out more information on Audio-In, specifically, through the Mic-In on the 3.5 mm jack. Unfortunately, before I can proceed, I need to know such things like sampling rate, data width, etc. Direction to the right resource or if you know off of the top of your head is appreciated.

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  • How to process audio in real time?

    - by user1756648
    I am giving some audio input through microphone. I recorded it in Audacity, it looks something like as shown below. I want to process this audio in real time. I mainly want to do this. 1) see real time audio amplitude vs time graph 2) perform some actions based on some thing (like if a specific type of hike is seen in audio, then do something, else do something else) Is there any python module or C library that can allow me to do this ?

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  • Why is object destructor not called when script terminates ?

    - by planetp
    I have a test script like this: package Test; sub new { bless {} } sub DESTROY { print "in DESTROY\n" } package main; my $t = new Test; sleep 10; The destructor is called after sleep returns (and before the program terminates). But it's not called if the script is terminated with Ctrl-C. Is it possible to have the destructor called in this case also?

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  • How to add a slot to my main window in Qt builder?

    - by George Edison
    I am using Qt Builder to create a simple window. I used the menu editor to add a menu. Now, I figured out how to connect one of the menu items to the close() method of the main window. My problem is how to add a slot to the main window. Here is what I have: private slots: void OnAbout(); However, I can't get this method to show up in the 'Signals and Slots Editor'. How can I get it to show up?

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  • "Voice trigger" detection

    - by sehugg
    I have a voice application that would be much-improved if there was the ability to use a "trigger word" to start recording audio. I don't need a full speech-text engine, just the ability to reliably/efficiently detect the trigger word. I am wondering if there are any specialized speech engines that support this specific use case, or any libraries/methods to developing such a single-purpose detection engine. Ideally I'd like it to work in noisy environments, but it can be trained for a single user's voice. Pointers to research papers / topics would also be appreciated so I know what to ask for.

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  • Compare two audio files of beat/tempo and rating in iphone

    - by Senthil Kumar
    Hello, I want to develop iPhone application should have the ability to count the number of phrases that are received when user sing on mic. This application should also have the ability to decipher whether the users phrases are in or out of cadence with a preset beat.When user sing on mic Instrumental music only play. So I have to merge the User Recorded voice with Instrumental music this is one Audio file.Already i have on original Song file.I have to compare both and give the Rating to users. [Note: Instrumental music is without vocal of Original Song file] Can you please help me?. Thanks Vadivelu

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  • Linear Interpolation. How to implement this algorithm in C ? (Python version is given)

    - by psihodelia
    There exists one very good linear interpolation method. It performs linear interpolation requiring at most one multiply per output sample. I found its description in a third edition of Understanding DSP by Lyons. This method involves a special hold buffer. Given a number of samples to be inserted between any two input samples, it produces output points using linear interpolation. Here, I have rewritten this algorithm using Python: temp1, temp2 = 0, 0 iL = 1.0 / L for i in x: hold = [i-temp1] * L temp1 = i for j in hold: temp2 += j y.append(temp2 *iL) where x contains input samples, L is a number of points to be inserted, y will contain output samples. My question is how to implement such algorithm in ANSI C in a most effective way, e.g. is it possible to avoid the second loop? NOTE: presented Python code is just to understand how this algorithm works. UPDATE: here is an example how it works in Python: x=[] y=[] hold=[] num_points=20 points_inbetween = 2 temp1,temp2=0,0 for i in range(num_points): x.append( sin(i*2.0*pi * 0.1) ) L = points_inbetween iL = 1.0/L for i in x: hold = [i-temp1] * L temp1 = i for j in hold: temp2 += j y.append(temp2 * iL) Let's say x=[.... 10, 20, 30 ....]. Then, if L=1, it will produce [... 10, 15, 20, 25, 30 ...]

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  • Finding center of fingerprints.

    - by an_ant
    If we suppose that every fingerprint is made of concentric curves (ellipses or circles) - and I'm aware of the fact that not every fingerprint is - how can I find center of those concentric curves? Let's take this "ideal" fingerprint and try to find out its center ... My approaches were to try: Find the spectrum through columns/rows of the image and try to find columns/rows that maximize particular band of the spectrum. I thought that column going through the center would have most regular pattern of changing amplitudes - therefore, most recognizible harmonic. My second approach was to try to count the changes of black-and-white also through the columns and rows, and to maximize that amount among rows and columns also. While these methods work to the some extant, with some additional filtering, they fail, when fingerprint is "not ideal as this one is". Can you think of any different approach? Are there standard ways to do it?

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  • Comapring pitches with digital audio

    - by user2250569
    I work on application which will compare musical notes with digital audio. My first idea was analyzes wav file (or sound in real-time) with some polyphonic pitch algorithms and gets notes and chords from this file and subsequently compared with notes in dataset. I went through a lot of pages and it seems to be a lot of hard work because existing implementations and algorithms are mainly/only focus on monophonic sound. Now, I got the idea to do this in the opposite way. In dataset I have for example note: A4 or better example chord: A4 B4 H4. And my idea is make some wave (or whatever I don't know what) from this note or chord and then compared with piece of digital audio. Is this good idea? Is it better/harder solution? If yes can you recommend me how to do it?

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  • What does Ubuntu do when I signal undocking to a laptop?

    - by Seppo Erviälä
    It seems that Ubuntu runs some script or command when I signal that I want to undock my laptop by pressing the undock button on the dock. Most visible thing that happens is that resolution on external display is changed. After prepearing for undock my laptop is still connected to power, VGA-output and audio jacks through dock but not to any usb devices or optical drive. I'm running 11.04 on a ThinkPad X61s with X6 UltraBase. What happens when I signal undocking? This is what dmesg says after pressing undock button: [81459.990682] ata1.00: disabled [81459.990727] ata1.00: detaching (SCSI 0:0:0:0) [81459.991722] ACPI: \_SB_.GDCK - undocking [81460.009462] ehci_hcd 0000:00:1a.7: power state changed by ACPI to D0 [81460.020252] ehci_hcd 0000:00:1a.7: BAR 0: set to [mem 0xfe226c00-0xfe226fff] (PCI address [0xfe226c00-0xfe226fff]) [81460.020265] ehci_hcd 0000:00:1a.7: power state changed by ACPI to D0 [81460.020281] ehci_hcd 0000:00:1a.7: restoring config space at offset 0xf (was 0x300, writing 0x30b) [81460.020309] ehci_hcd 0000:00:1a.7: restoring config space at offset 0x1 (was 0x2900000, writing 0x2900102) [81460.020338] ehci_hcd 0000:00:1a.7: PME# disabled [81460.020346] ehci_hcd 0000:00:1a.7: power state changed by ACPI to D0 [81460.020352] ehci_hcd 0000:00:1a.7: power state changed by ACPI to D0 [81460.020363] ehci_hcd 0000:00:1a.7: PCI INT C -> GSI 22 (level, low) -> IRQ 22 [81460.020372] ehci_hcd 0000:00:1a.7: setting latency timer to 64 [81460.020432] ehci_hcd 0000:00:1d.7: power state changed by ACPI to D0 [81460.040071] ehci_hcd 0000:00:1d.7: BAR 0: set to [mem 0xfe227000-0xfe2273ff] (PCI address [0xfe227000-0xfe2273ff]) [81460.040085] ehci_hcd 0000:00:1d.7: power state changed by ACPI to D0 [81460.040104] ehci_hcd 0000:00:1d.7: restoring config space at offset 0xf (was 0x400, writing 0x40b) [81460.040133] ehci_hcd 0000:00:1d.7: restoring config space at offset 0x1 (was 0x2900000, writing 0x2900102) [81460.040170] ehci_hcd 0000:00:1d.7: PME# disabled [81460.040178] ehci_hcd 0000:00:1d.7: power state changed by ACPI to D0 [81460.040184] ehci_hcd 0000:00:1d.7: power state changed by ACPI to D0 [81460.040195] ehci_hcd 0000:00:1d.7: PCI INT D -> GSI 19 (level, low) -> IRQ 19 [81460.040204] ehci_hcd 0000:00:1d.7: setting latency timer to 64 [81460.040503] ehci_hcd 0000:00:1d.7: PCI INT D disabled [81460.040552] ehci_hcd 0000:00:1d.7: PME# enabled [81460.061657] ehci_hcd 0000:00:1d.7: power state changed by ACPI to D3 [81460.200414] usb 1-4: USB disconnect, address 14 [81462.220088] ehci_hcd 0000:00:1a.7: PCI INT C disabled [81462.220169] ehci_hcd 0000:00:1a.7: PME# enabled [81462.240115] ehci_hcd 0000:00:1a.7: power state changed by ACPI to D3

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