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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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  • Determining the magnitude of a certain frequency on the iPhone

    - by eagle
    I'm wondering what's the easiest/best way to determine the magnitude of a given frequency in a sound. It's my understanding that a FFT function will return the magnitudes of all frequencies in a signal. I'm wondering if there is any shortcut I could use if I'm only concerned about a specific frequency. I'll be using the iPhone mic to record the audio. My guess is that I'll be using the Audio Queue Services for recording since I don't need to record the audio to a file. I'm using SDK 4.0, so I can use any of the functions defined in the Accelerate framework (e.g. FFT functions) if needed. Update: I updated the question to be more clear as per Conrad's suggestion.

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  • feature extraction from acoustic signals

    - by Dolphin
    Hi everyone, It's been a while. I found APIs in Java for extracting features from acoustic audio files and symbolic files separately. But now I have a problem in mapping from low level wav audio features to high level midi features. i.e. I need to write the extracted wav audio features on to midi format. But I cannot think of anything even close to it. Can someone pls provide me some insight as in how I can approach this. Greatly appreciate your responses. Advance thanks

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  • How to stream a WAV file?

    - by jonasb
    I'm writing an app where I record audio and upload the audio file over the web. In order to speed up the upload I want to start uploading before I've finished recording. The file I'm creating is a WAV file. My plan was to use multiple data chunks. So instead of the normal encoding (RIFF, fmt , data) I’m using (RIFF, fmt , data, data, ..., data). The first issue is that the RIFF header wants the total length of the whole file, but that is of course not known when streaming the audio (I’m now using an arbitrary number). The other problem is that I'm not sure if it's valid since Audacity doesn't recognise the file, and Windows Media Player opens the file but plays only a very small part. I've been reading WAV specs but haven’t found an answer. Any suggestions?

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  • How to use files/streams as source/sink in PulseAudio

    - by Nilesh
    I'm a PulseAudio noob, and I'm not sure if I'm even using the correct terminology. I've seen that PulseAudio can perform echo cancellation, but it needs a source and a sink to filter from, and a new source and sink. I can provide my mic and my audio-out as the source and sink, right? Now, here's my situation: I have two video streams, say, rtmp streams, or consider two flv files, say at any given moment, stream X is the input stream that's coming from another computer's webcam+mic and stream Y is the output stream that I'm sending, (and it's coming from my computer's webcam+mic). Question: Back to the first paragraph - here's the thing, I don't want to use my mic and my audio-out, instead, I want to use these two "input" and "output" streams as my source and sink so to speak (of course, I'll use xuggler maybe, to extract just the audio from X and Y). It may be a strange question, and I have my reasons for doing this strange this - I need to experiment and verify the results to see.

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  • C# - .WAV Playback Randomly High Pitch

    - by Nate Shoffner
    For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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  • PL/SQL pre-compile and Code Quality checks in an automatted build environment?

    - by Lars Corneliussen
    We build software using Hudson and Maven. We have C#, java and last, but not least PL/SQL sources (sprocs, packages, DDL, crud) For C# and Java we do unit tests and code analysis, but we don't really know the health of our PL/SQL sources before we actually publish them to the target database. Requirements There are a couple of things we wan't to test in the following priority: Are the sources valid, hence "compilable"? For packages, with respect to a certain database, would they compile? Code Quality: Do we have code flaws like duplicates, too complex methods or other violations to a defined set of rules? Also, the tool must run head-less (commandline, ant, ...) we wan't to do analysis on a partial code base (changed sources only) Tools We did a little research and found the following tools that could potencially help: Cast Application Intelligence Platform (AIP): Seems to be a server that grasps information about "anything". Couldn't find a console version that would export in readable format. Toad for Oracle: The Professional version is said to include something called Xpert validates a set of rules against a code base. Sonar + PL/SQL-Plugin: Uses Toad for Oracle to display code-health the sonar-way. This is for browsing the current state of the code base. Semantic Designs DMSToolkit: Quite general analysis of source code base. Commandline available? Semantic Designs Clones Detector: Detects clones. But also via command line? Fortify Source Code Analyzer: Seems to be focussed on security issues. But maybe it is extensible? more... So far, Toad for Oracle together with Sonar seems to be an elegant solution. But may be we are missing something here? Any ideas? Other products? Experiences? Related Questions on SO: http://stackoverflow.com/questions/531430/any-static-code-analysis-tools-for-stored-procedures http://stackoverflow.com/questions/839707/any-code-quality-tool-for-pl-sql http://stackoverflow.com/questions/956104/is-there-a-static-analysis-tool-for-python-ruby-sql-cobol-perl-and-pl-sql

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  • Kde no sound from Phonon or most KDE apps but mplayer,skype and firefox are ok

    - by zeonglow
    Can somebody tell me why I cannot get any sound with most of KDE 4? I'm running a Gentoo box, I'm in both the 'audio' and 'video' groups. I can get sound with mplayer ( but not smplayer ) Firefox and Skype but nothing else. I can't get the test sound to play from the settings window, but Phonon is not whining about broken sound cards when I start up. I have checked with kmix, we seem to be completely unmuted ( and I can get sound with some apps)

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  • Linux: how to use Jellyfish from Jack Meterbridge?

    - by klox
    dear all, i have installed Meterbridge. But,i'm just need to use Jellyfish from this package. I changed the Meterbridge properties become: /usr/bin/meterbridge -t jf alsa_pcm:playback_1 alsa_pcm:playback_2 My problem come here, i can open the Jellyfish window but i can't show the wave from input jack. How should i do? have you ever try this? some tell me to set up the Jack Audio Connection Kit, But i don't understand how to do it because i'm new for this

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  • Flash/Flex: play embedded AAC audio?

    - by aaaidan
    I'm pretty sure of the answer, but just wanted to check with you all. Is it possible to play an embedded AAC file in Flash/Flex somehow? I know you can playback embedded MP3 files, but I hear that you can't do that with AAC. Anyone know any sneaky ways to get around this? By way of illustration, here's come code. [Embed(source='../../audio/music02.m4a', mimeType="audio/aac")] private static const __ExampleMp4File:Class; public var myMp4Sound:Sound = new __ExampleMp4File(); public function EmbeddedAudioTest() { myMp4Sound.play(); }

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  • How do I "link narrations" in PowerPoint 2010 Beta?

    - by Zack Peterson
    I can record narration with PowerPoint 2010, but it seems that it will only embed it into the presentation file. Older versions of PowerPoint allowed the audio to be saved as external sound files. I'd like to perform noise-reduction and minor editing outside of PowerPoint. Has Microsoft removed the "link narrations" option from PowerPoint 2010?

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  • Get 5.1 surround sound from computer through a VCR config?

    - by Wedding Nails
    I'm posting to see if my idea of this setup is right and can be done. I currently have the following "equipment": a JVC VCR -quite old-, which has built in surround sound (aka it has several speaker outputs, which I believe is 5.1 and are connected to several speakers that are in every corner of the room), a computer with SPDIF optical output and a new flat screen TV (with built in HDMI). I want the computer to take advantage of the VCR's surround system (all the speakers in the room) in order to play mainly music and video always with all the speakers (5.1) and with the maximum sound quality. Currently, the computer plays sound only through the front speaker (I connect one output to the on board pc audio input) and the quality is really bad. As a side note, the computer video runs with S-video (old school), and the picture quality as you would imagine, is really bad with the new big LCD screen. My main goals are: to upgrade the picture with a new video card which would support HDMI (my tv has HDMI). to buy a SPDIF optical cable, connect one end to the VCR SPDIF input and the other end to the PC output This is theoretically what I've researched so far, and I came out with several questions: in this case, with the SPDIF cable connected, and all the configurations done in windows allowing the 5.1, will I get every content I play "converted" or played through all of my speakers? (I read this forum post). I already know that in order for this setup to play from all the speakers, the content/audio source has to be 5.1. but my question is, if there is a way to play from all of the speakers no matter what type of content I'm playing (that's why I said conversion there) I already know that HDMI cables carry digital sound. Is there a way I can only use said HDMI cord to the tv, and get sound through the VCR? (I'm not too sure about this, I would have to disable the TVs speakers and use the VCR surround as default, but I have no clue wether this can be done or not). Update: The ultimate question is, do I really have to rely on "sound virtualization" technology to get sound from all the speakers, no matter what content I play? (do I require a newer sound card, like a creative soundblaster with said technology?) Thanks!

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  • Can I prevent tv-out from disconnecting when monitor sleeps?

    - by Damon
    I have my TV connected to my computer (windows 7) through an HDMI cable that also carries the sound. I like to put on music on my TV, but the problem is when my monitor goes to sleep back in my office, it switches off the hdmi connection completely and the audio goes back to my PC. I would like a way to stop this from happening without turning off sleep mode entirely. I have an ATi Radeon 5700

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  • using addListener with WordPress audio player

    - by Jacob
    Hi, I'm trying to add a listener for the stop event in the word press audio player but usage seems to be undocumented. I'm hoping someone who knows a little flash can look at the code and tell me how it works: In the code at http://tools.assembla.com/1pixelout/browser/audio-player/trunk/source/classes/Application.as I see a snippet with this: ExternalInterface.call("AudioPlayer.onStop", _options.playerID); I was hoping that would let me capture the event in javascript with ("player" is the ID of my player) AudioPlayer.addListener("player", "AudioPlayer.onStop", function() { alert('stopped'); }); But my javascript function never seems to get called

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  • Video/Audio frame as input to OpenCore

    - by Vinay
    I am not able to use MediaPlayer/VideoView to make rtsp to work in Android. So I have created a client to interact with RTSP server, I have succeeded in doing this. I am able to get the video/audio frame from RTSP server (MySpace) in Android. Now I want to play the frames. I have searched OpenCore APIs to play the frames, but didn't get any APIs. My investigation: There is a class PlayerDriver.c It creates two sinks one audio and other video. handleSetVideoSurface handleSetAudioSink Two objects of type PVPlayerDataSinkPVMFNode are created. I suspect this class has a way to give the stream as input, but I am not getting the definition of this class. Can you suggest me is there any class I need to look into it?

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  • Dell Studio 17 with two headphone outs

    - by eych
    Got a Studio 17 with two headphone out jacks. It also has Soundblaster X-Fi Hi Def Audio Software enabled. Would it be possible to use one of the headphone outs as a monitor and the other as a master out with the Soundblaster sofware (for djing), or do you think the two jacks are just one output split into two?

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  • Automatic adaptive gain normalization

    - by Eduardo
    How may I normalize a (voice) audio mp3 or aac file with no loss, having the gain rised as much as possible with little distortion, so in a long conversation people that speak softer can have more gain for their voice and people that speak louder can have less gain?

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  • Make a radio-streaming PC pretend to be a mass-storage USB device

    - by monov
    I'm listening to a net radio on my PC I want the sound to go through my boombox cause it has nice speakers/amp The boombox has no "incoming" audio jack that just plays what comes over the wire However the boombox has a USB jack where you can put a thumbdrive with music. The question: How do I make the PC pretend to be a mass-storage device, and dynamically send all received audiodata to the boombox over a symmetric male USB cable? Failing that, at least tell me how to do it for local files (rather than streams). OS: Vista

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  • Python/Tkinter Audio Player

    - by Nicholas Quirk
    Hey everyone reading this, I've recently got into doing GUI development with Python. Tkinter seems like the easiest and most logical choice starting out. I did a little with wxPython but it was more sophisticated than what I needed. Anyway, I'm developing a media player. Right now it's a simple window with a button to load .wav files. The problem is I would like to implement a pause button now. But, when playing a audio file the GUI isn't accessible again (no buttons can be pushed) till the file is done playing. How can I make the GUI dynamic while an audio file is playing? I was thinking this maybe be because I'm using PyAudio, and their implementation doesn't allow this. Anyway, thanks for any advice before hand.

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  • Why does my 5.1 surround work in testing only?

    - by Mike Pateras
    I've got a 5.1 speaker setup. In both this SoundMax utility (I think it came with my motherboard), and in the Windows 7 sound test, all 5.1 speakers work properly, but that's the only time that I can get audio to come out of the rear speakers and the center channel. When playing games, video, music, etc., I only seem to get 2.1 speakers worth of sound, even though I've configured everything for 5.1 surround sound. How can I get my 5.1 surround sound working during actual use?

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