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• half-sine pulse shaping

  - by kos

  hi, i wanted to know what is the pulse shape of the modem.oqpskmod? and if it is not half-sine pulse shape, how is it possible to make it half-sine pulse shape as it is stated in ieee 802.15.4(zigbee) standard where it shows it as follows p(t)=sin(pi*t/2*Tc) if 0<=t<=2Tc p(t)=0 if otherwise ? thanks a lot!

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• Switched from DVI to HDMI, possible audio artifacts?

  - by I take Drukqs

  I'm using an ASUS VH236H monitor and an EVGA GeForce 570 GTX both of which are brand new. My monitor has an audio out port for speakers/headphones so I plugged in my headphones and made a random selection from my library when I noticed two things: There are static-like artifacts during "louder" parts of songs. There's what seems to be a volume cap in place. When I crank the volume past 100% in VLC the decibel level does not truly increase but the amount of static does. The cable is not new; I yanked it off of my PS3 when my DVI cable broke. It has been used a good amount on my HDTV and PS3 so I doubt it's a matter of burn-in. I like the way the setup works with an HDMI cable as opposed to DVI because my headphones barely reach my rig whereas I have plenty of slack when they're plugged into my monitor. Thanks in advance for any support. Note: I'm using a high quality HDMI cable from monoprice, AKG K702 headphones, and VLC media player.

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• Audio server with best API?

  - by Wintermute

  I'm a web dev, working in a small studio with a couple of other devs and some crayon-munchers (or, "designers"). Like all the best and trendiest creative studios, we have tunes. Our tunes consists of a set of speakers that whoever wants to can plug into their machine, and DJ their little socks off via iTunes, Spotify, VLC or whatever their music player of choice happens to be. Obviously, this lacks finesse. What we WANT is this: a single, dedicated machine running some sort of audio player (ideally Win-based, but a Linux flavour isn't impossible), that exposes an API. We (ie: me and the other devs) want to write a web-based client onto it, that'll let us remotely do all sorts of funky stuff like generating on-the-fly genre-based playlists, and voting for tracks, and making tea. My question - and please forgive me if this isn't the place for such a question, I was going to ask on Stackoverflow but that didn't seem right either - is this: what's the best player to start with? What can do all of this? I know VLC can function as a streaming server, but know nothing of any API it may have. I'd rather chop my pinky off than use iTunes, but if it does what we want, then... Anyhow, thanks for reading. All comments and suggestions gratefully received.

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• Optical SPDIF audio from motherboard not working with receiver

  - by simon b

  Hi, I hope someone can help; I can't get my SPDIF optical out working through my receiver and all the responses I can see on the web assume you have a sound card, while I settled for the (seemingly high end) sound on my motherboard (Asus P7P55D-E PRO), which appears to limit some of my options. My set-up is a "new out of the box" one and is: *Windows 7 PC (using PowerDVD10 for DVDs/Blurays and Windows media player for music) *Asus P7P55D-E PRO motherboard - has 8-channel audio TRS jacks and SPDIF optical and coaxial out *An old Yamaha receiver, whose only multi-channel input options are optical in and 6 channel RCA in. However, it still can handle DTS and DD *Boston Acoustic Soundware XS 5.1 speakers I've currently got the SPDIF optical out from the motherboard connected to the in on my receiver, have SPDIF enabled in the sound menu and the light is glowing red down the fibre. But I'm getting no sound at all. What I want is to be able to play DVDs/BluRays in 5.1 but also to be able to play music in multi-channel mode (even though I know this will be "fake" multichannel; it's more about where I sit in the room and my requirement to use the sub because the Boston is a satellite/sub set-up) My questions are: *Will optical work at all for multi-channel? THe latest posts I can see suggest it does but some people seem to say optical only outputs stereo. Whom to believe? *Even if it does work, I've read that I have to disable AC-3 decoding, or make various other changes, which don't seem to be possible without the menu options that a sound-card brings. Is the motherboard-only option just too inflexible? *Although my SPDIF device is enabled in the sound menu, it insists under "Jack information" that it is a "rear panel RCA jack", when of course it is not (both TOSLINK and rCA jacks do exist). Has the PC just forgotten that it has an optical? *I think I could relatively easily connect the 8-channel 3.5mm TRS jacks to my receiver 6-ch input jacks by way of TRS/RCA cables, but would that not stop me from being able to play music from media-player in multi-channel mode, as I'm not sure the motherboard can cope *Or do I need to bite the bullet and buy a sound-card? And if so, how can I be sure the one I get doesn't have the same problem? Any thoughts gratefully received, Cheers, simon

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• Audio 2 dj soundcard configuration

  - by Jaroslav

  I have an http://www.native-instruments.com/#/en/products/dj/audio-2-dj/ The problem in settings it only sees one outpout, when there should be two(I need that for mixxx etc.) Also I want to be able set the sample rate to one of these 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

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• choppy streaming audio

  - by user88503

  I could use some help troubleshooting choppy streaming audio. The problem is jerky playback regardless of audio or video with audio. Both Chromium and Firefox have the problem, however files played directly on the machine with Rhythmbox sound just fine. I'm running 12.04 LTS on a C2D T9300. Most of the audio problems others ask about seem to be hardware related, so the following information might be relevant. sudo lshw -c multimedia *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:48 memory:f8400000-f8403fff

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• How to get mixer applet for "Built-in Audio Analog Stereo"

  - by gerrit

  In pavucontrol, I can choose between RV620 HDMI Audio [Radeon HD 3400 Series] and Built-in Audio. When the former is enabled, videos on (among others) Youtube play way too fast, but this answer solved my problem (though I don't know why). However, when I use Built-in Audio instead of RV620 HDMI Audio [Radeon HD 3400 Series], the mixer in my applet appears to be disabled; the icon is replaced by a blank and changing the volume has no effect, as the applet apparently only relays to RV620 HDMI Audio [Radeon HD 3400 Series]. How do I get an applet to control the volume for Built-in Audio instead?

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• Java algorithm for normalizing audio

  - by Marty Pitt

  I'm trying to normalize an audio file of speech. Specifically, where an audio file contains peaks in volume, I'm trying to level it out, so the quiet sections are louder, and the peaks are quieter. I know very little about audio manipulation, beyond what I've learnt from working on this task. Also, my math is embarrassingly weak. I've done some research, and the Xuggle site provides a sample which shows reducing the volume using the following code: (full version here) @Override public void onAudioSamples(IAudioSamplesEvent event) { // get the raw audio byes and adjust it's value ShortBuffer buffer = event.getAudioSamples().getByteBuffer().asShortBuffer(); for (int i = 0; i < buffer.limit(); ++i) buffer.put(i, (short)(buffer.get(i) * mVolume)); super.onAudioSamples(event); } Here, they modify the bytes in getAudioSamples() by a constant of mVolume. Building on this approach, I've attempted a normalisation modifies the bytes in getAudioSamples() to a normalised value, considering the max/min in the file. (See below for details). I have a simple filter to leave "silence" alone (ie., anything below a value). I'm finding that the output file is very noisy (ie., the quality is seriously degraded). I assume that the error is either in my normalisation algorithim, or the way I manipulate the bytes. However, I'm unsure of where to go next. Here's an abridged version of what I'm currently doing. Step 1: Find peaks in file: Reads the full audio file, and finds this highest and lowest values of buffer.get() for all AudioSamples @Override public void onAudioSamples(IAudioSamplesEvent event) { IAudioSamples audioSamples = event.getAudioSamples(); ShortBuffer buffer = audioSamples.getByteBuffer().asShortBuffer(); short min = Short.MAX_VALUE; short max = Short.MIN_VALUE; for (int i = 0; i < buffer.limit(); ++i) { short value = buffer.get(i); min = (short) Math.min(min, value); max = (short) Math.max(max, value); } // assign of min/max ommitted for brevity. super.onAudioSamples(event); } Step 2: Normalize all values: In a loop similar to step1, replace the buffer with normalized values, calling: buffer.put(i, normalize(buffer.get(i)); public short normalize(short value) { if (isBackgroundNoise(value)) return value; short rawMin = // min from step1 short rawMax = // max from step1 short targetRangeMin = 1000; short targetRangeMax = 8000; int abs = Math.abs(value); double a = (abs - rawMin) * (targetRangeMax - targetRangeMin); double b = (rawMax - rawMin); double result = targetRangeMin + ( a/b ); // Copy the sign of value to result. result = Math.copySign(result,value); return (short) result; } Questions: Is this a valid approach for attempting to normalize an audio file? Is my math in normalize() valid? Why would this cause the file to become noisy, where a similar approach in the demo code doesn't?

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• What is the best software to help record the auido of MagicJack?

  - by Daniel

  What is the best software to help record the auido of MagicJack? I would love to record audio off of conference calls. This question is for both Windows and Macs.

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• How do audio based games such as Audiosurf and Beat Hazard work?

  - by The Communist Duck

  Note: I am not asking how to make a clone of one of these. I am asking about how they work. I'm sure everyone's seen the games where you use your own music files (or provided ones) and the games produce levels based on them, such as Audiosurf and Beat Hazard. Here is a video of Audiosurf in action, to show what I mean. If you provide a heavy metal song, you would get a completely different set of obstacles, enemies, and game experience from something like Vivaldi. What does interest me is how these games work. I do not know much about audio (well, data-side), but how do they process the song to understand when it is settling down or when it's speeding up? I guess they could just feed the pitch values (assuming those sorts of things exist in audio files) to form a level, but it wouldn't fully explain it. I'm either looking for an explanation, some links to articles about this sort of thing (I'm sure there's a term or terms for it), or even an open-source implementation of this kind of thing ;-) EDIT: After some searching and a little help, I found out about FFT (Fast Fourier Transform). This maybe a step in the right direction, but it is something that does not make any sense to me..or fits with my physics knowledge of waves.

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• Is it possible for a faulty processor to cause audio static/noise?

  - by Tom

  I have a Core 2 Extreme processor I received from a friend and have set up an XBMC box using it. However, I constantly get audio static whenever playing any music or videos. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 I have tried replacing everything short of the case and the processor, including cables, audio interfaces, operating systems, ram, etc, leading me to think it might be either the case shorting out the motherboards I have tried or a faulty processor. Is it possible for a faulty processor to cause audio static/noise? Any feedback would be appreciated. Edit - Here's a list of things I have tried: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Switching Power Cable Plugging in through surge protector Plugging into different outlet on separate circuit

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• [SOLVED} How do I restore my audio after uninstalling Ventrilo?

  - by Marcx

  Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. Update: Here's a sample of how I hear others people voices.. Audio Sample After some tests, also the desktop has the same problem. (I tried TeamSpeak3) Here are some details on my laptop and desktop Laptop Dell Studio 1555 Core 2 Duo P8600 2.4Ghz 4Gb Ram Dual Channel Ati HD 4570 512Mb dedicated (up to 2048) IDT High Definition Audio Desktop Motherboard Asus P5KPL-AM Dual Core CPU E5200 2.50Ghz 2x2GB PC6400 Dual Channel Ati Radeon HD 4650 512MB VIA High Definition Audio Both computers have Windows 7 Professional 64Bit. So how do I restore my audio? SOLVED The problem was in router firmware, there was a bug that recognized VoIP traffic as a DOS attack and the router grambled every packet... I've installed the newest firmware and everything is fine :)

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• How to write C++ audio processing applications?

  - by cesko82

  Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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• Looping HTML5 audio on the iPhone

  - by Peeps

  I'm trying to make a HTML5 webapp that simply plays a sound over and over and over again, on my iPhone. I don't know any Obj-C to do it natively. What I have works fine, but the sound only plays once: <!DOCTYPE html> <html> <head> <title>noisemaker!</title> <meta http-equiv="content-type" content="text/html; charset=utf-8" /> <meta name="viewport" content="maximum-scale=1, minimum-scale=1, width=device-width, user-scalable=no" /> <meta name="apple-mobile-web-app-capable" content="yes" /> </head> <body> <audio src="noise.mp3" autoplay controls loop></audio> </body> </html> Is there a way to either bypass the QuickTime audio screen and loop it in the webpage, or get the QuickTime audio screen to loop the sound?

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• Virtual audio driver (microphone)

  - by Dalamber

  Hello guys, I want to develop a virtual microphone driver. Please, do not say anything about DirectShow - that's not "the way". I need a solution that will work with any software including Skype and MSN. And DirectShow doesn't fit these requirements. I found AVStream Filter-Centric Simulated Capture Driver (avssamp.sys) in Windows 7 WDK. What I need is an audio part of it. By default it reads avssamp.wav and plays it. But this driver is registered as WDM streaming capture device. And I want it in Audio Capture Device. There are some posts in the web but they are all the same: http://www.tech-archive.net/Archive/Development/microsoft.public.development.device.drivers/2005-05/msg00124.html http://www.winvistatips.com/problem-installing-avssamp-audio-capture-sources-category-t184898.html I think registering this filter-driver as audio capture device will make Skype recognize it as a microphone and thefore I will be able to push any PCM file as if it goes from mic. If someone already faced this problem before, please help. Thanks in advance.

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• No suitable decoder module: VLC does not support the audio or video format "undf". Unfortunately the

  - by metal gear solid

  How to fix this VLC Media player error? I'm running a Video which is in dual-audio. It runs fine with one track(Language) but when i change to second audio track . then it doesn't play second audio and giving this error. "No suitable decoder module: VLC does not support the audio or video format "undf". Unfortunately there is no way for you to fix this."

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• How do I restore my audio after uninstalling Ventrilo?

  - by Marcx

  Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. So how do I restore my audio?

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• How do I use different audio devices for different apps in Windows 8?

  - by Eclipse

  Besides switching the default audio device, how can I send the audio from one app (say x-box music) to one audio device, and another (say the video app) to another audio device? Edit: Looking further, I found this: http://channel9.msdn.com/Events/BUILD/BUILD2011/APP-408T At 16:16, he demonstrates exactly what I'm wanting to do, but when I go to the devices charm, I get a message: "You don't have any devices that can receive content from Music".

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• Best way to learn iphone audio queue services, step by step tutorial

  - by optician

  Hi Everyone, I'm trying to learn how to handle audio at a fairly low level with audio queue services. I have been progrmaing in memory managed languages for quite a while, and have just completed the c programing tutorial by vtc (2007). This has left me comfortable with the understanding of pointers and memory allocation, but the apple documention still leaves me wanting for a simpler implenation and explaination. Maybe I need to learn objective c and cocoa better. I have heard that this book is good. Cocoa(R) Programming for Mac(R) OS X (3rd Edition) Could someone suggest a learning path that is going to help me get an better understanding of working with audio and an iphone. I want to be able to play mp3 files back and also alter the pitch of them as they are playing. I am prepared that I may have to temporarily convert the mp3 files into pcm files to do things like that to them. Thanks everyone.

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• Which audio library to use?

  - by Jeb

  I want to build a .Net application for processing audio, and distribute it using ClickOnce deployment. I need access to a raw audio pipeline. Which audio library should I be using? I've heard the managed libraries for DirectSound are a dead end. I need as little as possible to be installed on the client's machine. Anything outside of the ClickOnce process isn't going to work. NAudio might be a possibility, but isn't there potentially a separate driver install? There's also SlimDX. It's a shame -- the managed DirectX libraries seem to work nicely and from what I've read, DirectX can be included in the ClickOnce install.

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• Unexpected behavior with AudioQueueServices callback while recording audio

  - by rcw3

  I'm recording a continuous stream of data using AudioQueueServices. It is my understanding that the callback will only be called when the buffer fills with data. In practice, the first callback has a full buffer, the 2nd callback is 3/4 full, the 3rd callback is full, the 4th is 3/4 full, and so on. These buffers are 8000 packets (recording 8khz audio) - so I should be getting back 1s of audio to the callback each time. I've confirmed that my audio queue buffer size is correct (and is somewhat confirmed by the behavior). What am I doing wrong? Should I be doing something in the AudioQueueNewInput with a different RunLoop? I tried but this didn't seem to make a difference... By the way, if I run in the debugger, each callback is full with 8000 samples - making me think this is a threading / timing thing.

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• Link to audio in XHTML/EPUB

  - by wxs

  I'm looking into synchronizing an ebook in epub format (so the content is in XHTML) to an audio file. I'm thinking of putting something along the lines of: <a class="audiolink" href="sound.ogg?t=1093"></a> into the body of the document, and then have a custom epub reader that recognizes those tags and synchronizes the audio accordingly. That does seem like a bit of a hack to me though, especially the use of a special class name. Does anyone have any pointers to how this may be done in a more standards-compliant manner (or somewhere where it has been done before)? Ebooks with audio annotation seem like an idea that may already be out there.

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• Trying to build automatic audio-conferencing capability into a WebApp

  - by Keller

  Hey all, I'm working with a team of relatively novice programmers, and we are trying to create a site that will have audio-conferencing capabilities such that whenever someone visits the page, they will immediately have audio-conferencing capabilities with everyone else on the page (5 people max). Can anyone point us in a general direction? Should we be looking into building a custom app, leveraging audio conferencing software, or trying to mimic a webex program? Would Adobe Stratus be useful in getting this kind of functionality? Does anyone have any ideas about how we would design something like this on a macro level? Sorry for the noobish question, but any guidance would be deeply appreciated. Thanks, Keller

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• Seeking through a streamed MP3 file with HTML5 <audio> tag

  - by Kyle Slattery

  Hopefully someone can help me out with this. I'm playing around with a node.js server that streams audio to a client, and I want to create an HTML5 player. Right now, I'm streaming the code from node using chunked encoding, and if you go directly to the URL, it works great. What I'd like to do is embed this using the HTML5 <audio> tag, like so: <audio src="http://server/stream?file=123"> where /stream is the endpoint for the node server to stream the MP3. The HTML5 player loads fine in Safari and Chrome, but it doesn't allow me to seek, and Safari even says it's a "Live Broadcast". In the headers of /stream, I include the file size and file type, and the response gets ended properly. Any thoughts on how I could get around this? I certainly could just send the whole file at once, but then the player would wait until the whole thing is downloaded--I'd rather stream it.

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• Suggestion for creating custom sound recognition software to toggle audio

  - by Parrot owner

  I need to develop a program that toggles a particular audio track on or off when it recognizes a parrot scream or screech. The software would need to recognize a particular range of sounds and allow some variations in the range (as a parrot likely won't replicate its sreeches EXACTLY each time). Example: Bird screeches, no audio. Bird stops screeching for five seconds, audio track praising the bird plays. Regular chattering needs to be ignored completely, as it is not to be discouraged. I've heard of java libraries that have speech recognition with dictionaries built in, but the software would need to be taught the particular sounds that my particular parrot makes - not words or any random bird sound. In addition as I mentioned above, it would need to allow for slight variation in the sound, as the screech will likely never be 100% identical to the recorded version. What would be the best way to go about this/what language should I look into?

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