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  • How to change the audio output device in Firefox or any other modern browser?

    - by Zanami Zani
    I'm trying to play music through Ventrilo and currently I use Virtual Audio Cable. The way it works is that in foobar2000 (a music playing program) I set the output device in preferences to Virtual Audio Cable. Then in Ventrilo I log in to another name and set the input device to Virtual Audio Cable. This routes the music through the Virtual Audio Cable and allows me to play the music through Ventrilo. However, I would also like to change the output device for Firefox (or any other browser) or "Plugin Container for Firefix" to Virtual Audio Cable so that I could play music from Pandora or YouTube on to Ventrilo. Unfortunately I could not find an option for this anywhere.

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  • Is Windows Media Player able to play DTS audio?

    - by rolgae
    I'm trying to play DTS audio with Windows Media Player 12 on Windows 7. For a MPEG-TS file with video and DTS audio, only video is played. A file containing only a DTS audio stream is rejected. But: WMP is able to play the DTS audio stream of a DVD. So, Is Windows Media Player able to play DTS audio, or not? And if: How do I make him play my DTS files? I did not find any good resources of the supported codecs. Just things like "WMP can play .mpg files, ..." VLC is able to play all of the above files. I do not want to install third party codec packs, thats not the question!

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  • detecting pauses in a spoken word audio file using pymad, pcm, vad, etc

    - by james
    First I am going to broadly state what I'm trying to do and ask for advice. Then I will explain my current approach and ask for answers to my current problems. Problem I have an MP3 file of a person speaking. I'd like to split it up into segments roughly corresponding to a sentence or phrase. (I'd do it manually, but we are talking hours of data.) If you have advice on how to do this programatically or for some existing utilities, I'd love to hear it. (I'm aware of voice activity detection and I've looked into it a bit, but I didn't see any freely available utilities.) Current Approach I thought the simplest thing would be to scan the MP3 at certain intervals and identify places where the average volume was below some threshold. Then I would use some existing utility to cut up the mp3 at those locations. I've been playing around with pymad and I believe that I've successfully extracted the PCM (pulse code modulation) data for each frame of the mp3. Now I am stuck because I can't really seem to wrap my head around how the PCM data translates to relative volume. I'm also aware of other complicating factors like multiple channels, big endian vs little, etc. Advice on how to map a group of pcm samples to relative volume would be key. Thanks!

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  • Any screen capture software that captures webcam, microphone inputs too ?

    - by mohanr
    I am going to conduct a user study. Apart from capturing the screen while the user is interacting with the system, I also want to capture the video/audio of the user. Is there any software that in addition to capturing the screen also overlays it with the webcam/microphone inputs. The goal is to capture the complete experience of the user: key/mouse interactions with the system along with their facial/vocal responses. I know that I can maybe run a screen-capture software and also run a software for capturing webcam audio/video alongside and try to sync/overlay both these streams with timestamps. But I am going to be dealing with probably several hundred hours of data. So I am looking for a tool that can streamline the process for me amap and help me keep my sanity at end of the process. Thanks,

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  • Silverlight 4 - encoding PCM data from the microphone

    - by Richard
    Hi I've written a basic SL4 application to capture audio data from the microphone using CaptureSource. The trouble is, it's raw PCM output - which means huge and uncompressed. Given that I need this application to run purely within a SL4 environment, how can I compress the PCM audio data into something that can be delivered to a remote server more easily? In conversation, people have suggested Speex and WMA for instance, but I haven't found any libraries or examples that work without requiring reference to DLL's that won't work in a SL4 project. Thanks, Richard.

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  • After playing a MediaElement, how can I play it again?

    - by Edward Tanguay
    I have a variable MediaElement variable named TestAudio in my Silverlight app. When I click the button, it plays the audio correctly. But when I click the button again, it does not play the audio. How can I make the MediaElement play a second time? None of the tries below to put position back to 0 worked: private void Button_Click_PlayTest(object sender, RoutedEventArgs e) { //TestAudio.Position = new TimeSpan(0, 0, 0); //TestAudio.Position = TestAudio.Position.Add(new TimeSpan(0, 0, 0)); //TestAudio.Position = new TimeSpan(0, 0, 0, 0, 0); //TestAudio.Position = TimeSpan.Zero; TestAudio.Play(); }

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  • Is it possible to have AVFramework and AudioToolbox framework in one app?

    - by Satyam
    I'm trying to write develop audio related application. In that, I'm using AudioToolBox framework for recording the sound. And I'm using AVFramework to play soudns. When app is stared, it will play some mp3 file using AVFramework. And also initializes Audiotoolbox. In simulator, I'm able to record and play. But when I'm testing it on iPhone, I'm getting following error for initializing AudioToolBox. 2009-12-11 22:25:51.599 StoryBook[807:207] AudioRecorder init AudioSessionInitialize failed with error: 1768843636 Can some one tell me whether we can use both AV as well as Audio Toolbox frame works in one application? Why I'm getting that error?

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  • Changing volume in Java when using JLayer.

    - by Penchant
    I'm using JLayer to play an inputstream of mp3 data from the internet. How do i change the volume of the output? I'm using this code to play it: URL u = new URL(s); URLConnection conn = u.openConnection(); conn.setConnectTimeout(Searcher.timeoutms); conn.setReadTimeout(Searcher.timeoutms); bitstream = new Bitstream(conn.getInputStream()/*new FileInputStream(quick_file)*/); System.out.println(bitstream); decoder = new Decoder(); decoder.setEqualizer(equalizer); audio = FactoryRegistry.systemRegistry().createAudioDevice(); audio.open(decoder); for(int i = quick_positions[0]; i > 0; i--){ Header h = bitstream.readFrame(); if (h == null){ return; } bitstream.closeFrame();

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  • Play a beep that loop and change the frequency/speed

    - by Bono
    Hi all, I am creating an iphone application that use audio. I want to play a beep sound that loop indefinitely. I found an easy way to do that using the upper layer AVAudioPlayer and the numberOfLoops set to "-1". It works fine. But now I want to play this audio and be able to change the rate / speed. It may works like the sound played by a car when approaching an obstacle. At the beginning the beep has a low frequency and this frequency accelerate till reaching a continuous sound biiiiiiiiiiiip ... It seems this is not feasible using the high layer AVAudioPlayer, but even looking at AudioToolBox I found no solution. Does anybody have informations about how to do that? Thanks a lot for helping me!

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  • Novocaine - How to loop file playback? (iOS)

    - by lppier
    I'm using Novocaine by alexbw Novocaine for my audio project. I'm playing around with the example code here for file reading. The file plays back with no problem. I would like to loop this recording with the gap between the loops - any suggestion as to how I can do so? Thanks. Pier. // AUDIO FILE READING OHHH YEAHHHH // ======================================== NSArray *pathComponents = [NSArray arrayWithObjects: [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject], @"testrecording.wav", nil]; NSURL *inputFileURL = [NSURL fileURLWithPathComponents:pathComponents]; NSLog(@"URL: %@", inputFileURL); fileReader = [[AudioFileReader alloc] initWithAudioFileURL:inputFileURL samplingRate:audioManager.samplingRate numChannels:audioManager.numOutputChannels]; [fileReader play]; [fileReader setCurrentTime:0.0]; //float duration = fileReader.getDuration; [audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels) { [fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels]; NSLog(@"Time: %f", [fileReader getCurrentTime]); }];

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  • How Do I Convert text to a WAV file With Inaudible Waveform?

    - by Scott
    I am trying to create an audio watermarking system. I figure the best solution is to create an audio file (WAV) based on a unique string of text and then combine this with the original wav. The part that makes this tricky (for me anyway) is: How do I convert the text string to a wav? How do I ensure that the resulting WAV form is inaudible (or at least barely noticeable to the listener). I would prefer this be done server side (via PHP, etc) but if the processing load isn't too much then would be ok with something in Flash or Javascript. I'd be willing to pay someone to create me a workable solution (complete source code that functions as described). Thanks, Scott!

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  • How to configure the framesize using AudioUnit.framework on iOS

    - by Piperoman
    I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

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  • Signal amplitude against time in Java

    - by wsr74ws84
    I'm racking my brain in order to solve a knotty problem (at least for me). While playing an audio file (using Java) I want the signal amplitude to be displayed against time. I mean I'd like to implement a small panel showing a sort of oscilloscope (spectrum analyzer). The audio signal should be viewed in the time domain (vertical axis is amplitude and the horizontal axis is time). Does anyone know how to do it? Is there a good tutorial I can rely on? Since I know very little about Java, I hope someone can help me.

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  • Prefered method for looping sound flash as3

    - by Brian Heylin
    Hi there, I'm having some issues with looping a sound in flash AS3, in that when I tell the sound to loop I get a slight delay at the end/beginning of the audio. The audio is clipped correctly and will play without a gap on garage band. I know that there are issues with sound in general in flash, bugs with encodings and the inaccuracies with the SOUND_COMPLETE event (And Adobe should be embarrassed with their handling of these issues) I have tried to use the built in loop argument in the play method on the Sound class and also react on the SOUND_COMPLETE event, but both cause a delay. But has anyone come up with a technique for looping a sound without any noticeable gap?

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  • Html5 - Callback when media is ready on iPad wont work

    - by Kap
    I'm trying to add a callback to a HTML5 audio element on an iPad. I added an eventlistener to the element, the myOtherThing() starts but there is no sound. If I pause and the play the sound again the audio starts. This works in chrome. Does anyone have an idea how I can do this? myAudioElement.src = "path_to_file"; addEventListener("canplay", function(){ myAudioElement.play(); myOtherThing.start(); }); SOLVED Just wanted to share my solution here, just in case someone else needs it. As far as I understand the iPad does not trigger any events without user interactions. So to be able to use "canply", "playing" and all the other events you need to use the built in media controller. Once you press play in that controller, the events gets triggered. After that you can use your custom interface.

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  • Question on ExtAudioFileRead and AudioBuffer for iPhone SDK

    - by backspacer
    I'm developing an iPhone app that uses the Extended Audio File Services. I try to use ExtAudioFileRead to read the audio file, and store the data in an AudioBufferList structure. AudioBufferList is defined as: struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[1]; }; typedef struct AudioBufferList AudioBufferList; and AudioBuffer is defined as struct AudioBuffer { UInt32 mNumberChannels; UInt32 mDataByteSize; void* mData; }; typedef struct AudioBuffer AudioBuffer; I want to manipulate the mData but I wonder what does the void* mean. Why is it void*? How can I decide what data type is actually stored in mData?

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  • Can FLV AAC stream be played in Android

    - by HariKJ
    Hi, I'm trying to build a radio player and the client is providing a stream which is a FLV container with the audio being AAC When I read the headers it shows up as audio/aacp. I have tried all possible ways such as using the 1) Streaming through mediaplayer (Does not work) 2) Use the NPR mode of using a proxy stream (I get a broken pipe exception) 3) Play it in chunks ( Plays but I need the SDCard and the playback is not very great) 4) Use the GPL'd FAAD2 Library but I would have to pay the royalty fee Can some one help me out on figuring this issue out. The last option that I have is to have my client change the stream to mp3 container (which I know that it works) Regards, Hari

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  • sound loop breaks after some time in background music in iphone app

    - by amy
    I am playing sounds in loop in my app. So it should continue playing through out the app. but sometimes it stops after playing sound for 3/4 times.I don't understand whats happening. I am using audio-toolbox framework for playing sound. creating audio queue and then playing sounds in loop. I am also playing sound from ipod library using mediaplayer. Same thing happening with song from ipod. I have set [musicPlayer setRepeatMode: MPMusicRepeatModeOne]; but still it stops after 3/4 times.

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