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• Why is AudioOutputUnitStart freezing my app in iOS 4?

  - by Luke

  Hi guys, I have an audio app which uses the RemoteIO AudioUnit. It works fine on iPhone, iPad, and any flavor of the simulator on 3.2, but when it hits AudioOutputUnitStart (), it freezes. I get the message "AddRunningClient starting device on non-zero client count" in the console, which I'm not sure how to resolve. I stop the unit and dispose of the AudioComponent every time the app closes. The app works fine the first time I run after restarting everything, but freezes every time after that. What's strange is there are no error messages - just an unresponsive interface and a frozen line of code. Thanks for your help. Luke

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• Alternate play / pause button for WordPress wpaudio soundmanager plugin

  - by j-man86

  Hello! I am using the wpaudio plugin to convert mp3 links into a javascript/flash audio player. My problem is that I use this plugin in two areas on my site: one on a black background, and one on a white background. I need to use an alternate set of play/pause buttons for each page (white buttons for the black background and vice versa). I am at a total loss on how to do this. I need to some how incorporate a "if page is..." statement into the wpaudio.js but I don't know how to do this with jQuery. Can anyone help? Thanks so much!

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• Videoconference using Flash and SIP

  - by Júlio Santos

  The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 conference is required.

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• Manipulating multi-track ogg files programatically

  - by Chad Birch

  I'm planning to create a program for manipulating multi-track OGG files, but I don't have any experience with the relevant libraries, so I'm looking for recommendations about which language/library to use for this. I don't really have any preference for the language, I'll happily code it in C, C#, Python, whatever makes things the easiest (or even possible). Perhaps it's even a possibility to automate Audacity somehow? In terms of requirements, I'm not looking for anything particularly fancy. It will probably be a command-line program, I don't need to be able to play the audio, draw image representations of the waveforms, etc. The program will basically be used as a converter, but I need to do some processing before outputting. That is, I need the ability to programatically remove some tracks, set panning per-track, change track volumes, etc. Nothing too complex, just some basic processing, and then output the result in either MP3 or a format easily converted to MP3, such as WAV. Any suggestions or general information would be appreciated, thanks.

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• AudioQueueOfflineRender returning empty data

  - by hyn

  I'm having problems using AudioQueueOfflineRender to decode AAC data. When I examine the buffer after the call, it is always filled with empty data. I made sure the input buffer is valid and packet descriptions are provided. I searched and found that a few others have had the same problem: http://lists.apple.com/archives/Coreaudio-api/2008/Jul/msg00119.html Also, the inTimestamp argument doesn't make sense to me. Why should the renderer care where in the audio the beginning of the buffer corresponds to? The function throws an error if I pass in NULL, so I pass in the timestamp anyway.

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• Play .ts video file on Android?

  - by user359519

  I am pretty new at streaming video, so please bear with me. :) I am trying to port an m3u8 stream over from iPhone to Android. Looking in the m3u8 feed, I found some .ts files. From what I can tell, .ts files are, themselves, wrappers that contain the video stream (Elementary Stream). Is it possible to play a .ts file in Android? (The docs only list 3gp and mp4 as supported formats.) Is there a way to extract the Elementary Stream and just process the video feed? If that is in 3gp or mp4, I should be ok. Will Stagefright handle .ts? Is Stagefright even available? I read that there are/were some problems with it. (As a further caveat, I am not getting much help from my server guys. They are pushing for a Flash player solution, including a proprietary player. They will not provide me with a 3gp or an mp4 feed, but I'm hoping I can find that in the .ts file.) I'm open to other suggestions. Thanks for your patience with this newbie. :)

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• From ASPX to WCF

  - by Barguast

  I'm hoping someone can advise me on how to solve my networking scenario. Both the client and server are to be C# / .NET based. I basically want to invoke some kind of web service from my client in order to retrieve both binary data (e.g. files) and serialised objects and lists of objects (e.g. database query results). At the moment, I'm using ASPX pages, using the query string to provide parameters and I get back either the binary data, or the binary data of the serialised messages. This affords me a lot of flexbility, and I can choose how to transmit the data, perform simulatanous requests, cancel ongoing requests, etc. Since I can control the serialised format, I can also deserialise lists of objects as they are received which is crucial. My problem isn't a problem as such, but this feels a little hack-ish and I can't help but wonder if there are better ways to go about it. I'm considering moving on to WCF or perhaps another technology to see if it helps. However, I need to know if it helps with my scenarios above that is; Can a WCF method return a list of objects, and can the client receive the items of this list as they arrive as opposed to getting the entire list on completion (i.e. streaming). Does anyone know of any examples of this? Am I likely to get any performance benefits from this? I don't know how well ASPX pages are tuned for this, as it surely isn't their primary purpose. Are there any other approaches I should consider? Thanks for your time spent reading this. I hope you can help.

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• How to extract semi-precise frequencies from a WAV file using Fourier Transforms

  - by Seisatsu

  Let us say that I have a WAV file. In this file, is a series of sine tones at precise 1 second intervals. I want to use the FFTW library to extract these tones in sequence. Is this particularly hard to do? How would I go about this? Also, what is the bast way to write tones of this kind into a WAV file? I assume I would only need a simple audio library for the output. My language of choice is C

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• Write wave files to memory in Java

  - by Cliff

  I'm trying to figure out why my servlet code creates wave files with improper headers. I use: AudioSystem.write( new AudioInputStream( new ByteArrayInputStream(memoryBytes), new AudioFormat(22000, 16, 1, true,false), memoryBytes.length ), AudioFileFormat.Type.WAVE, servletOutputStream ); taking a byte array from memory containing raw PCM samples and a servlet output stream that gets returned to the client. In the result I get a normal wave file but with zeros in the chunk size fields. Is the API broken? I would think that the size could be filled in using the size passed in the audio input stream. But now, after typing this out I'm thinking its not making this info available to the outer write() method on AudioSystem. It seems like the AudioSystem.write call needs a size parameter unless it is able to pull the size from the stream... which wouldn't work with an arbitrary sized stream. Does anyone know how to make this example work?

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• Why doesnt R.raw.'songname' not work on android devices?

  - by James Rattray

  I have some media (Audio tracks) on an app... With file path 'R.raw.test' I use some code to get it into a mediaplayer... MediaPlayer.create(Textbox.this, R.raw.fly); And it works PERFECTLY on the Android Emulator... (Plays track on click of button) Can someone explain why, when I put it on my Archos (5 IT) it doesnt work at all? -As soon as the button is clicked, it crashes... Do you have to do something to file paths or what? Please help... Thanks alot... James

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• How to detect when video is buffering?

  - by Leon

  Hi guys, my question today deals with Flash AS3 video buffering. (Streaming or Progressive) I want to be able to detect when the video is being buffered, so I can display some sort of animation letting the user know to wait just a little longer. Currently my video will start up, hold on frame 1 for 3-4 secs then play. Kinda giving the impression that the video is paused or broken :( Update Thanks to iandisme I believe I'm faced in the right direction now. NetStatusEvent from livedocs. It seems to me that the key status to be working in is "NetStream.Buffer.Empty" so I added some code in there to see if this would trigger my animation or a trace statement. No luck yet, however when the Buffer is full it will trigger my code :/ Maybe my video is always somewhere between Buffer.Empty and Buffer.Full that's why it won't trigger any code when I test case for Buffer.Empty? Current Code public function netStatusHandler(event:NetStatusEvent):void { // handles net status events switch (event.info.code) { case "NetStream.Buffer.Empty": trace("¤¤¤ Buffering!"); //<- never traces addChild(bufferLoop); //<- doesn't execute break; case "NetStream.Buffer.Full": trace("¤¤¤ FULL!"); //<- trace works here removeChild(bufferLoop); //<- so does any other code break; case "NetStream.Buffer.Flush": trace("¤¤¤ FLUSH!"); //Not sure if this is important break } }

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• Create mp3 previews from wav and aiff files

  - by August Lilleaas

  I would like to create a program that makes mp3s of the first 30 seconds of an aiff or wav file. I would also like to be able to choose location and length, such as the audio between 2:12 and 2:42. Are there any tools that lets me do this? Shelling out is OK. The application will run on a linux server, so it would have to be a tool that works on linux. I don't mind doing it in two steps - i.e. a tool that first creates the cutout of the aiff/wav, then pass it to a mp3 encoder.

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• Architecture of chatroulette

  - by user317163

  Could somebody explain to me the architecture behind chatroulette? I was thinking about a similar project that would only implement Audio support (for starters). Is the best way to set this up a flash server? If so, how should I go about getting into flash, will I need flex 4? I have some beginner experience with c++, c# and java but I have never developed anything for the web. I was also wondering how the randomizer matches up the participants. How would you code something like this. Im obviously pretty clueless here and I'd greatly appreciate some advice regarding this problem -- I don't expect copy and paste solutions. It would just be nice to hear how you guys would tackle this problem. Thank you very much

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• Detecting when Bluetooth is disabled on iOS5

  - by Non Umemoto

  I'm developing blog speaker app. I wanna pause the audio when bluetooth is disabled like iPod app. I thought it's not possible without using private api after reading this. Check if Bluetooth is Enabled? But, my customer told me that Rhapsody and DI Radio apps both support it. Then I found iOS5 has Core Bluetooth framework. https://developer.apple.com/library/ios/documentation/CoreBluetooth/Reference/CoreBluetooth_Framework/CoreBluetooth_Framework.pdf CBCentralManagerStatePoweredOff status seems like the one. But, the description says this api only supports Bluetooth 4.0 low energy devices. Did anyone try doing the same thing? I want to support current popular bluetooth headsets, or bluetooth enabled steering wheel on the car. I don't know if it's worth trying when it only supports some brand new bluetooth.

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• About data size filled in the buffer

  - by Bohan Lu

  I need low-latency audio in my project, and I know Android 2.3 supports OpenSL ES. I have read documents and sample code and I decide to use Android simple buffer queue to do the play and record. I now try to write a simple application to do the test. However, I have some questions about recording. If I set the recorder stop when it is recording, how do I know the exact number of bytes filled in the last buffer if it is not filled up ? In 1.1 version, the callback function has some parameters about buffer and its filled data, but there is no such parameters in version 1.0.1. Is there any way to get this information ? Any suggestion would be greatly appreciated !

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• Making of a "Babbelbox" where you can speak to for partys

  - by Spidfire

  Ive got a project to make for a party, its called in holland a "Babbelbox". its a computer with a webcam and microphone that can be used to make a kind of video log of everyone who wants to say something about the party. But the problem is that i dont know where to start. ive made a kind of video show system in c but i cant save any data to a good format so it wont jam my harddisk in one hour full. Requirements: Record video + audio Recoding has to start after pressing a button Good compression over the recorded videos (would be even better if it can to be read by final cut pro or premiere pro) Light wight programm would be nice but i could scale up the computer power

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• For the iPad or iPhone, how do you control the system Volume? For example, have a button that mutes

  - by SolidSnake4444

  I would like to make a button in my iPad app (probably will be similar to iPhone apps) that when I push this button, all audio is muted, even when you exit the app. I don't see anyway that you can control the volume, although I'm sure other apps have that I have seen in the app store for the iPhone. I also read some places that doing this would reject you from the app store. How could I go about lowering, or highering the volume of the iPad from an app that works even when the app closes? Thank you!

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• MPMoviePlayerContentPreloadDidFinishNotification seems more reliable than MPMoviePlayerLoadStateDidChangeNotification

  - by user567889

  I am streaming small movies (1-3MB) off my website into my app. I have a slicehost webserver, I think it's a "500MB slice". Not sure off the top of my head how this translates to bandwidth, but I can figure that out later. My experience with MPMoviePlayerLoadStateDidChangeNotification is not very good. I get much more reliable results with the old MPMoviePlayerContentPreloadDidFinishNotification If I get a MPMoviePlayerContentPreloadDidFinishNotification, the movie will play without stuttering, but if I use MPMoviePlayerLoadStateDidChangeNotification, the movie frequently stalls. I'm not sure which load state to check for: enum { MPMovieLoadStateUnknown = 0, MPMovieLoadStatePlayable = 1 << 0, MPMovieLoadStatePlaythroughOK = 1 << 1, MPMovieLoadStateStalled = 1 << 2, }; MPMovieLoadStatePlaythroughOK seems to be what I want (based on the description in the documentation): MPMovieLoadStatePlaythroughOK Enough data has been buffered for playback to continue uninterrupted. Available in iOS 3.2 and later. but that load state NEVER gets set to this in my app. Am I missing something? Is there a better way to do this?

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• "Winamp style" spectrum analyzer

  - by cvb

  I have a program that plots the spectrum analysis (Amp/Freq) of a signal, which is preety much the DFT converted to polar. However, this is not exactly the sort of graph that, say, winamp (right at the top-left corner), or effectively any other audio software plots. I am not really sure what is this sort of graph called (if it has a distinct name at all), so I am not sure what to look for. I am preety positive about the frequency axis being base two exponential, the amplitude axis puzzles me though. Any pointers?

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• How to receive a datastream from a device on your computer, in C#

  - by WebDevHobo

  I plan to build a small audio-recorder app in C#. My laptop has a built in Microphone that's always active, so I want to use that as an early-stage test. I would simply start recording, save the file as a .wav or even use the LAME dll to make it into an MP3. The problem is, I don't know how to contact that microphone. Do I use a library that can detect a device, or do I just catch a stream of bytes from the port that the device is on? I don't have any experience with receiving data from connected devices. I suppose that I'll need to enter all the data into a byte array and then Serialize that into a WAV file, but I'm not sure. Can I get some pointers on this subject?

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• Java stop MIDI playback

  - by user456268

  Hi I have java application which plays midi messages from sequence. I'm doing this using jfugue library. the problem is when I'm tryingto stop playback with stop button (which call sequencer.stop() and sequencer.close()) the last played note is sound all of rest time, and I can't stop it. So I'm asking about solution about stopping all audio and MIDI too! sound playback from java application. Notice: If you want propose just mute volume, you need to know that I want end-use will be able to press play button again and hear the sound again, so muting volumr will be not a solution, or explain please. Thank you!

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• Python frequency detection

  - by Tsuki

  Ok what im trying to do is a kind of audio processing software that can detect a prevalent frequency an if the frequency is played for long enough (few ms) i know i got a positive match. i know i would need to use FFT or something simiral but in this field of math i suck, i did search the internet but didn not find a code that could do only this. the goal im trying to accieve is to make myself a custom protocol to send data trough sound, need very low bitrate per sec but im also very limited on the transmiting end so the recieving software will need to be able custom (cant use an actual hardware/software modem) also i want this to be software only (no additional hardware except soundcard) thanks alot for the help.

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• How to replicate Google "Hangouts On Air" stream combining functionality?

  - by Rob Olmos

  I've been researching this one for quite a bit but haven't found any solid leads. I have a Wowza/Flash app with video chatroom functionality and would like to combine the streams server-side into one video/audio stream in order to be sent to a live Youtube channel. I've found a couple projects such as jMixer and some helpful keywords such as "vision mixer" to help with my search but looking for any previous experience or new ideas. The other option is building something like it myself with a commercial video decoding/encoding library to raw frames, stitching the frames together, then encoding it. I was originally going down this route but put project on hold. What are some ideas, keywords, or existing software (open source preferred) to take those live streams and combine them into one in real-time? Or is coding it myself the required route? Thanks!

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• Flash video plays well, but time and time-remaining are out of sync.

  - by sparkyfied

  Hey guys, Could be a known issue, an issue with my code or an oversight on my part. I have created a video player in flash. I have got it playing progressive and streaming over rtmp/rtmpt so that is all fine. My only issue is that when the video's are playing, the time-codes for time played and time remaining are not synced. So, if my video is 20 secs long and 5 have been played, time played will be 5secs, time remaining will be 16secs until it updates about half a second later. So even though they are both being set with the same line of code, there are not changing at the same time. The time played changes, then a split second later the time remaining changes. Anyone got any idea what this could be. Maybe a miscalculation on my part. Maybe I need to round up or down the remaining time. How can I sync the two times. I understand this is probably an tough question to answer, I have done my best to explain it. Thanks in advance.

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• Can I play any Buffer only once at a given time?

  - by mystify

  From the OpenAL documentation: The basic OpenAL objects are a Listener, a Source, and a Buffer. There can be a large number of Buffers, which contain audio data. Each buffer can be attached to one or more Sources My problem is, that I have one sound file which I need to play multiple times per second, at the same time. The sound is 2 seconds long. So it will overlap. Would I need multiple filled buffers for this (= multiple times that sound in memory)? If I would attach one Buffer to multiple Sources, would I be able to play the sound 10 times, overlapping itself, with just one copy in memory? Or would I still have to deal with 10 copies of that sound in memory?

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