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• Noisy audio recording in Ubuntu Linux

  - by Haresh K Miriyala

  I'm unable to record audio without noise on my Ubuntu laptop . It is not a grounding problem . My windows os is able to record voice without the static noise so, my hardware is fine . I read in the Windows forums that the os has software noise filtering mechanism that's automatically enabled. How do I do the same in Ubuntu? If not, please let me know how to filter the static noise. Any help is appreciated. Thanks!

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• Removing surrounding noises from voice recording

  - by Peak Reconstruction Wavelength

  I have a wave file whose frequency spectrum looks like this. http://i.stack.imgur.com/2rRaS.png It contains audio, which I want to keep while removing the rest. The problem is that the surround noise changes, just those distinct voice patterns remain. I marked the voice patterns for clarity: http://i.stack.imgur.com/eLkBl.png What could an algorithm look like / a workflow in adobe audition look like that removes everything but the voice patterns? I think that the main characteristic is the line-shaped form over time. Loudness alone is not enough as the noise is loud aswell.

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• On Windows 7, how can I tell if a recording is multi-channel without third party tools? [migrated]

  - by engineerchuan

  A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

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• On Windows 7, how can I tell if a recording is multi-channel without third party tools?

  - by engineerchuan

  A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

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• same flash file (.swf) downloaded multiple times on a page

  - by Gunjan

  I have a page that has a table with each row corresponding to an audio file. The last cell in each row embeds a simple flash audio player. The problem is that the flash file for the player is being downloaded for each row separately and as soon as rows go beyond 40-50 it crashes the browser. I tried using different players (1pixelout, flash-mp3-player) and the problem is still there, so its not a player specific issue. Is there any way to cache the player so that it is only downloaded once?

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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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• C# Audio - How to time stretch (different tempo, same pitch)

  - by heath

  I'm trying to make a winform app in C# (VS2008) that can load an mp3 (other formats would be nice, but mp3 at a minimum) and be able to adjust the playback speed (tempo) without affecting pitch. I really don't need any other audio effects. I tried using DirectShow but that doesn't seem to offer time stretch capabilities. I was able to incorporate irrklang but that does not seem to have the time stretch capability either. So now I've moved on to SoundTouch. That certainly has the capabilities but I'm very unclear on how to implement in C#. After a few days of this, about all I've accomplished is using DLLImport on the SoundTouch DLL and am able to successfully retrieve a version number. At this point, I'm not even sure if I can do what I'm trying to do with SoundTouch. Can anyone offer some guidance either on how to implement SoundTouch or a different library with the capabilities that I'm looking for? Thank you.

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• Ruby/Rails Audio Conversion Plugins?

  - by coneybeare

  I am looking for a good gem/plugin to convert user-uploaded audio files to different formats. One format in particular that I am interested in is converting to Apple .caf with ima4 compression for inclusion in an iPhone app. I have been using afconvert on my mac for this so far, but I need to do it on my linux box, server-side. Ideally, I would be able to work into paperclip. As an additional solution, ffmpeg could work, but I have not seen any .caf options for it. Anybody know of one?

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• Downsampling and applying a lowpass filter to digital audio

  - by twk

  I've got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I'd like to cut it down to an 11KHz stream. How do I do that? From my days of engineering class many years ago, I know that the stream won't be able to describe anything over 5500Hz accurately anymore, so I assume I want to cut everything above that out too. Any ideas? Thanks. Update: There is some code on this page that converts from 48KHz to 8KHz using a simple algorithm and a coefficient array that looks like { 1, 4, 12, 12, 4, 1 }. I think that is what I need, but I need it for a factor of 4x rather than 6x. Any idea how those constants are calculated? Also, I end up converting the 16 byte samples to floats anyway, so I can do the downsampling with floats rather than shorts, if that helps the quality at all.

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• Java playback of 24 bit audio is incorrect

  - by Paul Hampson

  I am using the javax sound API to implement a simple console playback program based on http://www.jsresources.org/examples/AudioPlayer.html. Having tested it using a 24 bit ramp file (each sample is the last sample plus 1 over the full 24 bit range) it is evident that something odd is happening during playback. The recorded output is not the contents of the file (I have a digital loopback to verify this). It seems to be misinterpreting the samples in some way that causes the left channel to look like it is having some gain applied to it and the right channel looks like it is being attenuated. I have looked into whether the PAN and BALANCE controls need setting but these aren't available and I have checked the windows xp sound system settings. Any other form of playback of this ramp file is fine. If I do the same test with a 16bit file it performs correctly with no corruption of the stream. So does anyone have any idea why the Java Sound API is modifying my audio stream?

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• Iphone progressive download audio player

  - by joynes

  Hi! Im trying to implement a progressive download audio player for the iphone, ie using http and fixed size mp3-files. I found the AudioStreamer project but it seems very complicated and works best with endless streams. I need to be able to find out the total length of audiofiles and I also need to be able to seek in the files. I found a hacked deviation from AudioStreamer but it doesnt seem to work very well for me. http://www.saygoodnight.com/?p=14 Im wondering if there is a more simple way to achieve my goals or if there are some better working samples out there? I found the bass library but not much documentation about it. /Br Johannes

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• Audio -- How much performance improvement can I expect from from reducing function calls by using bu

  - by morgancodes

  I'm working on an audio-intensive app for the iPhone. I'm currently calling a number of different functions for each sample I need to calculate. For example, I have an envelope class. When I calculate a sample, I do something like: sampleValue = oscilator->tic() * envelope->tic(); But I could also do something like: for(int i = 0; i < bufferLength; i++){ buffer[i] = oscilatorBuffer[i] * evelopeBuffer[i]; } I know the second will be more efficient, but don't know by how much. Are function calls expensive enough that I'd be crazy not to use buffers if I care event a tiny bit about performance?

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• Streaming audio - where to start?

  - by Adam Davis

  I need to develop an embedded audio streaming server. Requirements: Voice quality or better Intended for low power wifi transmission Broad support in existing software and devices (ie, windows media player, quicktime, vlc, iPhone, Android, etc). Royalty/patent free, or cheap to license Preferences: Low overhead TCP/IP based streaming protocol Voice grade codec (easy to implement in software, no DSP, 32bit CPU if needed) Would be nice if it supported HTML5 browsers, but is there any codec (such as raw) that is supported by the latest browsers that is lower overhead than MP3? Therefore: What are the relevant streaming protocols I should be looking at? What are the relevant codecs I should be looking at? What transport streams should I be looking at? What am I missing, or where else should I be looking for this type of need?

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• Android - Audio recorder FileNotFound

  - by david

  Hi, I'm trying to record audio this.recorder = new android.media.MediaRecorder(); this.recorder.setAudioSource(android.media.MediaRecorder.AudioSource.MIC); this.recorder.setOutputFormat(android.media.MediaRecorder.OutputFormat.DEFAULT); this.recorder.setAudioEncoder(android.media.MediaRecorder.AudioEncoder.DEFAULT); this.recorder.setOutputFile("pruebaAudioRecorder.mp4"); this.recorder.prepare(); this.recorder.start(); but when i call prepare method throws the FileNotFound exception. Should I create the file before prepare method? something like new File(...) If so, which should be the file path? thx a lot.

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• Client-side framework for web-app with good audio support

  - by Poita_

  I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

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• iPhone game audio and background music

  - by Boon

  Have a few questions related to adding sounds to my game, specifically intro music (for splash), background music (loop) and button event sounds. Hope you can share your knowledge on this. 1) Should I use compressed sounds or uncompressed sounds? Or perhaps a combination of the two? Are there any limitations on the iPhone hardware that I should be aware of -- for example, the ability to play multiple compressed sounds? 2) What's the best audio format for my purpose? 3) For background music, I am thinking of using AVAudioPlayer. For button event sounds, I am thinking of using AudioServicesPlaySystemSound, what do you think? 4) Any other issues I should be aware of? Thank you!

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• directx audio video error message in debugmode

  - by clamp

  I have a c#/winforms application that uses directx to play some video and audio. whenever i start my application in debugmode i get this annoying message. i can click "continue" and everything seems to work fine. but i still want to get rid of this message. it does not show up in releasemode. Managed Debugging Assistant 'LoaderLock' has detected a problem in 'C:\pathtoexe.exe'. Additional Information: DLL 'C:\WINDOWS\assembly\GAC\Microsoft.DirectX.AudioVideoPlayback\1.0.2902.0__31bf3856ad364e35\Microsoft.DirectX.AudioVideoPlayback.dll' is attempting managed execution inside OS Loader lock. Do not attempt to run managed code inside a DllMain or image initialization function since doing so can cause the application to hang.

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• Naudio - putting audio stream into values [-1,1]

  - by denonth

  Hi all I need to put my audio stream into values of [-1,1]. Can someone tell me a good approach. I was reading byte array and float array from stream but I don't know what to do next. Here is my code: float[] bytes=new float[stream.Length]; float biggest= 0; for (int i = 0; i < stream.Length; i++) { bytes[i] = (byte)stream.ReadByte(); if (bytes[i] > biggest) { biggest=bytes[i]; } } and I don't know how to put values into stream. Because byte is only positive values. And I need to have from [-1,1] for (int i = 0; i < bytes.Count(); i++) { bytes[i] = (byte)(bytes[i] * (1 / biggest)); }

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• audio power on AudioQueue

  - by Tomoyuki

  Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

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• Toggling audio on click?

  - by angela

  please look at this fiddle http://jsfiddle.net/rabelais/yLdkj/1/ The above fiddle shows three bars that on hover play audios. How do I change this so the music plays and pauses on click instead. Also if one audio is playing and another is clicked how can the already playing song pause? $("#one").mouseenter(function () { $('#sound-1').get(0).play(); }); $("#one").mouseleave(function () { $('#sound-1').get(0).pause(); }); $("#two").mouseenter(function () { $('#sound-2').get(0).play(); }); $("#two").mouseleave(function () { $('#sound-2').get(0).pause(); }); $("#three").mouseenter(function () { $('#sound-3').get(0).play(); }); $("#three").mouseleave(function () { $('#sound-3').get(0).pause(); });

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• High level audio crossfading library for python

  - by tcoopman

  I am looking for a high level audio library that supports crossfading for python (and that works in linux). In fact crossfading a song and saving it is about the only thing I need. I tried pyechonest but I find it really slow. Working with multiple songs at the same time is hard on memory too (I tried to crossfade about 10 songs in one, but I got out of memory errors and my script was using 1.4Gb of memory). So now I'm looking for something else that works with python. I have no idea if there exists anything like that, if not, are there good command line tools for this, I could write a wrapper for the tool.

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• audio error in vmware running mac os x

  - by PenguinSource

  simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound. the error is display bottom right, saying 'error in creating sound stream; sound is disconnected' i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version. before i get that error, the sound on the system works just fine (youtube, etc) the exact code im calling is.. [CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID]; [[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes]; soundEngine = [SimpleAudioEngine sharedEngine]; [soundEngine preloadBackgroundMusic:@"somemp3.mp3"]; [soundEngine playBackgroundMusic:@"somemp3.mp3"]; maybe the bit rate is too high .. ? thanks

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• How to calculate the audio file duration in core audio?

  - by mystify

  I have this info variable which is of this type: struct AudioStreamBasicDescription { Float64 mSampleRate; UInt32 mFormatID; UInt32 mFormatFlags; UInt32 mBytesPerPacket; UInt32 mFramesPerPacket; UInt32 mBytesPerFrame; UInt32 mChannelsPerFrame; UInt32 mBitsPerChannel; UInt32 mReserved; }; How could I calculate the total duration of the audio file, in seconds?

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• Seeking not working in HTML5 audio tag

  - by lord_wilmore

  I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

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• Advice for building a browser-based audio mixer up to 32 tracks

  - by Jonathan P.

  As a personal hobby I am looking to build an online audio mixer where I can upload individual instrument tracks, control individual volumes of each track, and export the mixed down version. I've been trying (and have come pretty close) with javascript. I really would like to stay away from flash if possible, but I'm really looking for suggestions for technologies to try. If anyone has any suggestions on languages that are good at stuff like this or libraries that I am missing, please let me know! I have a test environment that I have been using: http://driverstestpractice.com/sandbox Currently all tracks on the site are set to the click track in order to test the track sync (which as you can tell is a little off)! Thanks!

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