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  • xvidcap: Error accessing sound input from /dev/dsp

    - by stivlo
    I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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  • Where is /dev/dsp or /dev/audio?

    - by YumYumYum
    I have to apply sudo chmod a+r /dev/dsp or /dev/audio but in my Ubuntu 12.10 i do not have such. Where is then the PCM sound file for ssh? chmod: cannot access `/dev/dsp': No such file or directory chmod: cannot access `/dev/audio': No such file or directory Follow up: http://superuser.com/questions/244173/missing-dev-dsp-under-ubuntu I want to stream the sound output and input. So that i can capture any audio in/out to a file for recording.

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  • OMAP 3530: How fast can I toggle an IO?

    - by raj.tiwari
    I am putting together an application for OMAP 3530 SoC. This application will run some user interface code on embedded linux and invoke waveform generation code on the DSP. The DSP and Linux sides will interact over DSP/BIOS link. My questions are: What is the highest frequency at which my DSP-side code can toggle a GPIO line? If I want to toggle multiple GPIO lines at this hight rate, how fast can I go? Thanks for any insights. -Raj

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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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  • any good free C DSP library?

    - by Juan
    Hi everybody I am developing an application to process geophysical signals; Right now I have done everything in octave and its digital signal processing toolbox, speed is not bad, however the application specifications say I need to port to the final algorithm to C; I am doing lots of filtering, re-sampling and signal manipulation/characterization with FFTs and cepstrums. do you know a good free C library for DSP packaged with filter design, resampling, fft, etc? Thanks a lot for any suggestion

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  • Generating C code from a Matlab-Simulink model for DSP C6748

    - by Juanma
    I'm trying to generate C code from a Matlab-Simulink simple model (eg.: sine wave generator with a DAC at the output). This code must be executed with Code Composer Studio for TMS320C6748 DSP (Texas Instrument C6748). Specifically, for the development board OMAP-L138 ZOOM ™EVM DEVELOPMENT KIT. For this, I am using the following versions: Simulink (Version 7.7 - R2011a) Embedded Coder (Version 6.0 - R2011a) Code Composer Studio v3.3 I tried several options (with generic modules in Simulink and programming the C6748 timers, configuring a module "Target Preferences" with "OMAP_L138/C6748 EVM"...) but it isn't working. Is it possible to implement this idea? Is there an example working? Thanks

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  • DSP - How are frequency amplitudes modified using DFT?

    - by Trap
    I'm trying to implement a DFT-based equalizer (not FFT) for the sole purpose of learning. To check if it works I took an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. Now I tried to silence some frequency bands, just by setting their amplitudes to zero before resynthesis, but definitely it's not the way to go. What I get is a rather distorted signal. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. I first tried to modify the real part amplitudes only, then modifying both the real and imaginary part amplitudes. I also tried to convert the DFT output to polar notation, then modifying the magnitude and convert back to rectangular notation, but none of this is working. Can someone show me what I'm doing wrong? I tried to find info on this subject in the internet but couldn't find any. Thanks in advance.

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  • DSP - Problems using the inverse Fast Fourier Transform

    - by Trap
    I've been playing around a little with the Exocortex implementation of the FFT, but I'm having some problems. First, after calculating the inverse FFT of an unchanged frequency spectrum obtained by a previous forward FFT, one would expect to get the original signal back, but this is not the case. I had to figure out that I needed to scale the FFT output to about 1 / fftLength to get the amplitudes ok. Why is this? Second, whenever I modify the amplitudes of the frequency bins before calling the iFFT the signal gets distorted at low frequencies. However, this does not happen if I attenuate all the bins by the same factor. Let me put a very simplified example of the output buffer of a 4-sample FFT: // Bin 0 (DC) FFTOut[0] = 0.0000610351563 FFTOut[1] = 0.0 // Bin 1 FFTOut[2] = 0.000331878662 FFTOut[3] = 0.000629425049 // Central bin FFTOut[4] = -0.0000381469727 FFTOut[5] = 0.0 // Bin 3, this is a negative frequency bin. FFTOut[6] = 0.000331878662 FFTOut[7] = -0.000629425049 The output is composed of pairs of floats, each representing the real and imaginay parts of a single bin. So, bin 0 (array indexes 0, 1) would represent the real and imaginary parts of the DC frequency. As you can see, bins 1 and 3 both have the same values, (except for the sign of the Im part), so I guess these are the negative frequency values, and finally indexes (4, 5) would be the central frequency bin. To attenuate the frequency bin 1 this is what I do: // Attenuate the 'positive' bin FFTOut[2] *= 0.5; FFTOut[3] *= 0.5; // Attenuate its corresponding negative bin. FFTOut[6] *= 0.5; FFTOut[7] *= 0.5; For the actual tests I'm using a 1024-length FFT and I always provide all the samples so no 0-padding is needed. // Attenuate var halfSize = fftWindowLength / 2; float leftFreq = 0f; float rightFreq = 22050f; for( var c = 1; c < halfSize; c++ ) { var freq = c * (44100d / halfSize); // Calc. positive and negative frequency locations. var k = c * 2; var nk = (fftWindowLength - c) * 2; // This kind of attenuation corresponds to a high-pass filter. // The attenuation at the transition band is linearly applied, could // this be the cause of the distortion of low frequencies? var attn = (freq < leftFreq) ? 0 : (freq < rightFreq) ? ((freq - leftFreq) / (rightFreq - leftFreq)) : 1; mFFTOut[ k ] *= (float)attn; mFFTOut[ k + 1 ] *= (float)attn; mFFTOut[ nk ] *= (float)attn; mFFTOut[ nk + 1 ] *= (float)attn; } Obviously I'm doing something wrong but can't figure out what or where.

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  • DSP - Filtering frequencies using DFT

    - by Trap
    I'm trying to implement a DFT-based 8-band equalizer for the sole purpose of learning. To prove that my DFT implementation works I fed an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. This method calculates the real and imaginary parts both N/2 + 1 samples in length. To attenuate a frequency I'm just doing: float atnFactor = 0.6; Re[k] *= atnFactor; Im[k] *= atnFactor; where 'k' is an index in the range 0 to N/2, but what I get after resynthesis is a slighty distorted signal, especially at low frequencies. The input signal sample rate is 44.1 khz and since I just want a 8-band equalizer I'm feeding the DFT 16 samples at a time so I have 8 frequency bins to play with. Can someone show me what I'm doing wrong? I tried to find info on this subject on the internet but couldn't find any. Thanks in advance.

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  • DSP - Filtering in the frequency domain via FFT

    - by Trap
    I've been playing around a little with the Exocortex implementation of the FFT, but I'm having some problems. Whenever I modify the amplitudes of the frequency bins before calling the iFFT the resulting signal contains some clicks and pops, especially when low frequencies are present in the signal (like drums or basses). However, this does not happen if I attenuate all the bins by the same factor. Let me put an example of the output buffer of a 4-sample FFT: // Bin 0 (DC) FFTOut[0] = 0.0000610351563 FFTOut[1] = 0.0 // Bin 1 FFTOut[2] = 0.000331878662 FFTOut[3] = 0.000629425049 // Bin 2 FFTOut[4] = -0.0000381469727 FFTOut[5] = 0.0 // Bin 3, this is the first and only negative frequency bin. FFTOut[6] = 0.000331878662 FFTOut[7] = -0.000629425049 The output is composed of pairs of floats, each representing the real and imaginay parts of a single bin. So, bin 0 (array indexes 0, 1) would represent the real and imaginary parts of the DC frequency. As you can see, bins 1 and 3 both have the same values, (except for the sign of the Im part), so I guess bin 3 is the first negative frequency, and finally indexes (4, 5) would be the last positive frequency bin. Then to attenuate the frequency bin 1 this is what I do: // Attenuate the 'positive' bin FFTOut[2] *= 0.5; FFTOut[3] *= 0.5; // Attenuate its corresponding negative bin. FFTOut[6] *= 0.5; FFTOut[7] *= 0.5; For the actual tests I'm using a 1024-length FFT and I always provide all the samples so no 0-padding is needed. // Attenuate var halfSize = fftWindowLength / 2; float leftFreq = 0f; float rightFreq = 22050f; for( var c = 1; c < halfSize; c++ ) { var freq = c * (44100d / halfSize); // Calc. positive and negative frequency indexes. var k = c * 2; var nk = (fftWindowLength - c) * 2; // This kind of attenuation corresponds to a high-pass filter. // The attenuation at the transition band is linearly applied, could // this be the cause of the distortion of low frequencies? var attn = (freq < leftFreq) ? 0 : (freq < rightFreq) ? ((freq - leftFreq) / (rightFreq - leftFreq)) : 1; // Attenuate positive and negative bins. mFFTOut[ k ] *= (float)attn; mFFTOut[ k + 1 ] *= (float)attn; mFFTOut[ nk ] *= (float)attn; mFFTOut[ nk + 1 ] *= (float)attn; } Obviously I'm doing something wrong but can't figure out what. I don't want to use the FFT output as a means to generate a set of FIR coefficients since I'm trying to implement a very basic dynamic equalizer. What's the correct way to filter in the frequency domain? what I'm missing? Thanks in advance.

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  • DSP - Filter sweep effect

    - by Trap
    I'm implementing a 'filter sweep' effect (I don't know if it's called like that). What I do is basically create a low-pass filter and make it 'move' along a certain frequency range. To calculate the filter cut-off frequency at a given moment I use a user-provided linear function, which yields values between 0 and 1. My first attempt was to directly map the values returned by the linear function to the range of frequencies, as in cf = freqRange * lf(x). Although it worked ok it looked as if the sweep ran much faster when moving through low frequencies and then slowed down during its way to the high frequency zone. I'm not sure why is this but I guess it's something to do with human hearing perceiving changes in frequency in a non-linear manner. My next attempt was to move the filter's cut-off frequency in a logarithmic way. It works much better now but I still feel that the filter doesn't move at a constant perceived speed through the range of frequencies. How should I divide the frequency space to obtain a constant perceived sweep speed? Thanks in advance.

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  • wxCam can't open /dev/dsp

    - by SIJAR
    I try to run wxCam through PulseAudio using the following command: $ padsp -d wxcam Although wxCam started ok, wxCam is been set to use xdiv format to enable sound during recording, but while recording the I get the error: Cannot open /dev/dsp. Video file will be recorded without audio track Please help me in fixing this issue. Below is some debug information: $ padsp -d wxcam Determining video4linux API version... Using video4linux 2 API VIDIOC_ENUM_FRAMESIZES: Invalid argument V4L2_CID_GAMMA is not supported Determining pixel format... pixel format: YUV 4:2:2 (YUYV) Found V4L2_PIX_FMT_YUYV pixel format pixel format: MJPEG Found V4L2_PIX_FMT_MJPEG pixel format --DEBUG: [wxcam] Generating standard Huffman tables for this frame. Corrupt JPEG data: 2 extraneous bytes before marker 0xd2 ... repeats a couple of times ... open of failed: No such file or directory --DEBUG: [wxcam] Generating standard Huffman tables for this frame. Corrupt JPEG data: 2 extraneous bytes before marker 0xd4 ... repeats a couple of times .... /home/sij/Videos/Webcam/video.avi written: 640x480, 1334396 bytes --DEBUG: [wxcam] Generating standard Huffman tables for this frame. Corrupt JPEG data: 2 extraneous bytes before marker 0xd3 ... repeats a couple of times ...

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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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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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  • Is Objective C fast enough for DSP/audio programming

    - by morgancodes
    I've been making some progress with audio programming for iPhone. Now I'm doing some performance tuning, trying to see if I can squeeze more out of this little machine. Running Shark, I see that a significant part of my cpu power (16%) is getting eaten up by objc_msgSend. I understand I can speed this up somewhat by storing pointers to functions (IMP) rather than calling them using [object message] notation. But if I'm going to go through all this trouble, I wonder if I might just be better off using C++. Any thoughts on this?

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  • Carousel not working in IE7/8

    - by user515990
    I am working with jquery.carouFredSel-4.0.3-packed.js for the carousal and it works good with IE9 and mozilla,but in IE7/8, it says "LOG: carouFredSel: Not enough items: not scrolling " whenever i am seeing it is not the case. The code i am using is <div class="carousel-wrapper"> <div class="mask"> <a class="arrow left off"><-</a> <a class="arrow left on" href="javascript:void(0);"><-</a> <ul> <dsp:droplet name="ForEach"> <dsp:param name="array" value="${listRecommended}"/> <dsp:oparam name="empty">no recommended apps</dsp:oparam> <dsp:oparam name="output"> <li> <a href="javascript:void(0);"><img src="${resourcePath}/images/apps/carousel-image1.jpg" alt="bakery story"/></a> <a href="javascript:void(0);"><dsp:valueof param="element.displayName"/></a><br/> <dsp:getvalueof var="averageRating" param="element.averageRating"/> <dsp:getvalueof var="rating" param="count"/> <div class="rating"> <div class="medium"> <dsp:droplet name="For"> <dsp:param name="howMany" value="${averageRating}"/> <dsp:oparam name="output"> <input checked="checked" class="star {split:1}" disabled="disabled" name="product-similar-'${rating}'" type="radio"> </dsp:oparam> </dsp:droplet> </div> </div> </li> </dsp:oparam> </dsp:droplet> </ul> </div> <a class="arrow right off">-></a> <a class="arrow right on" href="javascript:void(0);">-></a> </div> and javascript library is jquery.carouFredSel-4.0.3-packed.js. Please let me know if someone has faced similar problem. thanks in advance Hemish

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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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  • iTunes Visualizer Plugin in C# - Energy Function

    - by James D
    Hi, iTunes Visualizer plugin in C#. Easiest way to compute the "energy level" for a particular sample? I looked at this writeup on beat detection over at GameDev and have had some success with it (I'm not doing beat detection per se, but it's relevant). But ultimately I'm looking for a stupid-as-possible, quick-and-dirty approach for demo purposes. For those who aren't familiar with how iTunes structures visualization data, basically you're given this: struct VisualPluginData { /* SNIP */ RenderVisualData renderData; UInt32 renderTimeStampID; UInt8 minLevel[kVisualMaxDataChannels]; // 0-128 UInt8 maxLevel[kVisualMaxDataChannels]; // 0-128 }; struct RenderVisualData { UInt8 numWaveformChannels; UInt8 waveformData[kVisualMaxDataChannels][kVisualNumWaveformEntries]; // 512-point FFT UInt8 numSpectrumChannels; UInt8 spectrumData[kVisualMaxDataChannels][kVisualNumSpectrumEntries]; }; Ideally, I'd like an approach that a beginning programmer with little to no DSP experience could grasp and improve on. Any ideas? Thanks!

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  • Implementation of FIR filter in C#

    - by user261924
    Hi, at the moment im trying to implement a FIR lowpass filter on a wave file. The FIR coefficients where obtained using MATLAB using a 40 order. Now i need to implement the FIR algorithm in C# and im finding it difficult to implement it. Any help? Thanks

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  • VB FFT - stuck understanding relationship of results to frequency

    - by WaveyDavey
    Trying to understand an fft (Fast Fourier Transform) routine I'm using (stealing)(recycling) Input is an array of 512 data points which are a sample waveform. Test data is generated into this array. fft transforms this array into frequency domain. Trying to understand relationship between freq, period, sample rate and position in fft array. I'll illustrate with examples: ======================================== Sample rate is 1000 samples/s. Generate a set of samples at 10Hz. Input array has peak values at arr(28), arr(128), arr(228) ... period = 100 sample points peak value in fft array is at index 6 (excluding a huge value at 0) ======================================== Sample rate is 8000 samples/s Generate set of samples at 440Hz Input array peak values include arr(7), arr(25), arr(43), arr(61) ... period = 18 sample points peak value in fft array is at index 29 (excluding a huge value at 0) ======================================== How do I relate the index of the peak in the fft array to frequency ?

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  • Saturated addition of two signed Java 'long' values

    - by finnw
    How can one add two long values (call them x and y) in Java so that if the result overflows then it is clamped to the range Long.MIN_VALUE..Long.MAX_VALUE? For adding ints one can perform the arithmetic in long precision and cast the result back to an int, e.g.: int saturatedAdd(int x, int y) { long sum = (long) x + (long) y; long clampedSum = Math.max((long) Integer.MIN_VALUE, Math.min(sum, (long) Integer.MAX_VALUE)); return (int) clampedSum; } or import com.google.common.primitives.Ints; int saturatedAdd(int x, int y) { long sum = (long) x + (long) y; return Ints.saturatedCast(sum); } but in the case of long there is no larger primitive type that can hold the intermediate (unclamped) sum. Since this is Java, I cannot use inline assembly (in particular SSE's saturated add instructions.) It can be implemented using BigInteger, e.g. static final BigInteger bigMin = BigInteger.valueOf(Long.MIN_VALUE); static final BigInteger bigMax = BigInteger.valueOf(Long.MAX_VALUE); long saturatedAdd(long x, long y) { BigInteger sum = BigInteger.valueOf(x).add(BigInteger.valueOf(y)); return bigMin.max(sum).min(bigMax).longValue(); } however performance is important so this method is not ideal (though useful for testing.) I don't know whether avoiding branching can significantly affect performance in Java. I assume it can, but I would like to benchmark methods both with and without branching. Related: http://stackoverflow.com/questions/121240/saturating-addition-in-c

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  • i2s0: transmitter underrun (0)

    - by tbarbe
    were doing some audio stuff and I keep seeing this in the Organizer Console. Sun May 2 20:16:48 unknown kernel[0] : i2s0: transmitter underrun (0) Are these transmitter underruns bad? I think its just when were shutting down audio input...but could a few of these cause some issues later on?

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