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• Where is my problem? The P6X58D Premium Mobo, Windows 7, or other?

  - by Dylan Yaga

  I was having problems with my USB devices for an hour last night, and I am unable to determine the root cause of the problem. The two symptoms are: At seemingly random times (not consistently spaced by time or caused by any detectable event) my USB devices become "detached". Windows will play the USB disconnect sound and then the reconnect sound. The devices disconnected and then reconnected. My USB Keyboard will "stick" on one key for several seconds before processing any other keystroke made. The mouse also does not respond to clicks. I do not lose mouse movement or USB device connectivity. And after a moment of this several beeps will be emitted from the speakers. Hardware Specs: GFX Card: EVGA GeForce GTX 470 Superclocked 1280MB DDR5 PCIe Motherboard: ASUS P6X58D Premium Intel X58 Socket LGA1366 MB Processor: Intel Core i7-920 2.66Ghz 8M LGA1366 CPU Memory: Corsair Dominator 6144MB PC12800 DDR3 Storage: Hitachi 1TB Serial ATA HD 1600MHz 7200/32MB/SATA-3G Cooling: Corsair Hydro H50 CPU Liquid Cooler Case: Corsair Obsidian 800D Full Tower Case Power Supply: Corsair HX1000W 1000W Modular Power Supply Steps I have taken to narrow down the problem: Restarted the computer. - No change Changed USB port the Hub was connected to on the CPU. - No change Removed all devices from USB Hub and connected directly to CPU. - No change Used a different USB keyboard both in USB Hub and directly to CPU. - No change Disconnected and reconnected all cables. - No change Disassembled the Tower and determined if the USB headers were firmly connected. - No change Checked device manager for errors. Checked all USB devices. - Nothing flagged After an hour of frustration trying to narrow down the problem it appeared to disappear. But I am torn between it being a Mobo problem or an OS problem. Is there anything else I can do to narrow down the problem before a reformat and then eventually exchanging the Mobo?

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• How do I configure WakeOnUSB properly?

  - by wishi

  How do I configure Wake-On-USB properly on a 10.04 or 10.10 Ubuntu (2.6.36 and higher if needed)? (Wake-on-USB is when the computer is asleep and for example a USB Keyboard event wakes up the machine!) The notebook is an Acer Aspire Timeline X 1830T. I don't know in which way the Linux Kernel supports the controllers. There are different ways to approach this, for example /proc/acpi/wakeup... or UDEV... or something with HAL? /proc/acpi/wakeup shows every device in S4, but I need S3. Device S-state Status Sysfs node P0P2 S4 *disabled PEGP S4 *disabled P0P1 S0 *disabled pci:0000:00:1e.0 EHC1 S4 *disabled pci:0000:00:1d.0 USB1 S4 *enabled USB2 S4 *disabled USB3 S4 *disabled USB4 S4 *disabled EHC2 S4 *disabled pci:0000:00:1a.0 USB5 S4 *disabled USB6 S4 *disabled USB7 S4 *disabled HDEF S0 *disabled pci:0000:00:1b.0 RP01 S5 *disabled pci:0000:00:1c.0 PXSX S5 *disabled pci:0000:01:00.0 RP02 S0 *disabled pci:0000:00:1c.1 PXSX S5 *disabled pci:0000:02:00.0 RP03 S0 *disabled PXSX S5 *disabled RP04 S0 *disabled PXSX S5 *disabled RP05 S0 *disabled PXSX S5 *disabled RP07 S0 *disabled PXSX S5 *disabled RP08 S0 *disabled PXSX S5 *disabled GLAN S0 *disabled PEG3 S4 *disabled PEG5 S4 *disabled PEG6 S4 *disabled SLPB S3 *enabled S4, which is Suspend-To-Disk afaik... doesn't seem to work either if I echo USB1 into the wakeup table. It just sets an S4 flag. can I get the USB ports in S3? I want to make the machine wakeup from Suspend-To-Ram (S3, ACPI standard) in case a key on my external keyboard is pressed. It only wakes up if a key on the internal Laptop keyboard is pressed... from Suspend To Ram. It seems if I plug in a USB mouse, that the USB port isn't even powered. I have no BIOS option to change this. Further specific information regarding the device: usb-devices T: Bus=01 Lev=02 Prnt=02 Port=01 Cnt=01 Dev#= 13 Spd=1.5 MxCh= 0 D: Ver= 1.10 Cls=00(>ifc ) Sub=00 Prot=00 MxPS= 8 #Cfgs= 1 P: Vendor=04d9 ProdID=1603 Rev=03.10 S: Manufacturer= S: Product=USB Keyboard C: #Ifs= 2 Cfg#= 1 Atr=a0 MxPwr=100mA I: If#= 0 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=01 Prot=01 Driver=usbhid I: If#= 1 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid root@underwater-laptop:/# lsusb [...] Bus 001 Device 013: ID 04d9:1603 Holtek Semiconductor, Inc. Bus 001 Device 004: ID 0bda:0138 Realtek Semiconductor Corp. Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub [...] If this doesn't work I have to properly explain why :( - but I think it is very hard to research this kernel internal. Any hints for good information here? I hope it's possible... I'm just looking for any solution. edit: this, waking up on USB, works on Windows! Thanks a lot, Marius

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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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• 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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• 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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• 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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• 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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• 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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• audio stream sampling rate in linux

  - by farhan

  Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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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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• 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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• 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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• Capturing Mac OS X System Audio output with Python

  - by richbs

  Hello, I've been trying to "hijack" the Mac OS X system audio using PyAudio and save to a wav in python. That is, I do not want to record from an input device such as a microphone. I want to grab the sound output from any or all applications. I have followed the tutorials on the PyAudio site but these do not appear to cover my use case and when I try to read from the output stream I unsurprisingly get the paCanNotReadFromAnOutputOnlyStream exception. Fair enough! Is there a way to do what I am proposing with the PyAudio or other FOSS Python Library?

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