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  • Learn mp3 format and audio signal processing

    - by Shankhoneer Chakrovarty
    I am trying to learn the following things: How mp3 file looks like internally? I found this: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html but it seems old. Is there any recent changes to the format? I couldnt find any. How to open a mp3 file in java and look for bytes? I tried using audiostream but I am getting a lot of zeros and signed short integers which nowhere resemble the header/body format as mentioned in the above link. Am I wrong in interpreting the bytes? How to get amplitude, frequency and pitch of a mp3 file? No idea. Can you please suggest some book or tutorial? Can you please help me in getting the solution for the above questions? I am sorry if some questions appear to be naive, I am a just begun to learn mp3. Thanks

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  • How to change default audio input device programatically

    - by f34r
    I am looking for a way to set/change default input device inside my application. I have several different recording devices and it is very anoying to go into the control panel and change default recording device. I was looking around and I did not find anything that could help me with the problem. Application is written in c# and it is targeted for Windows Vista / Windows 7.

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  • Synchronizing Java Visualizer Audio and Visual

    - by Matt
    I've run into a problem creating a visualizer for .mp3 files in Java. My goal is to create a visualization that runs in time with the .mp3 file being played. I can currently visualize an .mp3 OR play it, but not both at the same time. I am using libraries which may make this trickier than necessary. I currently: Read in the .mp3 as a FileInputStream. a) Convert the FileInputStream into a Bitstream and run the Visualizer OR b) Pass the FileInputStream to a library Play method where it converts it into a Bitstream, decodes it, and plays it. I am using the JLayer library to play and decode the .mp3. My question is: how do I synchronize the two actions so that I can run both at the same time AND they line up (so my visualizations correspond to the changing frequencies). This implies that they finish at the same time as well.

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  • Audio/Voice Visualization

    - by Neurofluxation
    Hey you Objective-C bods. Does anyone know how I would go about changing (transforming) an image based on the input from the Microphone on the iPhone? i.e. When a user speaks into the Mic, the image will pulse or skew. Thanking you!!

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  • How to release audio properly? (AVAudioPlayer)

    - by Aluminum
    Hello everyone! I need help with my iOS application ^^,. I want to know if I'm releasing AVAudioPlayer correctly. MyViewController.h #import <UIKit/UIKit.h> @interface MyViewController : UIViewController { NSString *Path; } - (IBAction)Playsound; @end MyViewController.m #import <AVFoundation/AVAudioPlayer.h> #import "MyViewController.h" @implementation MyViewController AVAudioPlayer *Media; - (IBAction)Playsound { Path = [[NSBundle mainBundle] pathForResource:@"Sound" ofType:@"wav"]; Media = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:Path] error:NULL]; [Media play]; } - (void)dealloc { [Media release]; [super viewDidUnload]; } @end

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  • Non intrusive notification without audio?

    - by acidzombie24
    i have a C# app that registers a protocol. When you click BLAH://djfhgjfdghjkd in a browser it launches my app. However you can click multiple links and each link is a note added into the app. How can i inform the user that he did fully click the link? Right now i have a console app showing up for 1sec (basically pops up and goes away as fast as possible) which felt better then a hidden console since you are unsure if it went through. The 1 second takes a lot of time when you are trying to rapidly click many notes/links and the console gets in the way. What can i do that is noticeable? I'm thinking have a box that comes up (and is semi transparent) but the click passes through it. Maybe there is a better way? Also i wouldnt know where to start with transparent windows or pass through clicks

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  • How to control wether or not my app will stop any iPod music playing?

    - by mystify
    When my app launches, there are rumors that I can decide if any currently playing music is stopped or not. My goal is to use Audio Queue Services, because I believe that's the most powerful audio technology in iPhone OS. So, could I really decide that when my app launches? How? Which one of the many audio technologies on the iPhone OS is responsible for managing this?

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  • aplay -l says no soundcards found; alsaconf says no supported cords; yet /proc/asound contains cards

    - by nimasmi
    I am trying to get HDMI output using a Gainward Nvidia 210 512 MB on Ubuntu 10.04 Lucid Lynx. I have upgraded alsa-driver, alsa-lib and alsa-utils to 1.0.24 by building from source, thanks to this blog post. Some relevant output... user@box:~$ lspci | grep Audio 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 01:09.0 Multimedia video controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder (rev 05) 01:09.2 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [MPEG Port] (rev 05) 01:09.4 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [IR Port] (rev 05) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) user@box:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Sep 15 2012 for kernel 2.6.32-42-generic (SMP). user@box:~$ ls /proc/asound` card0 cards hwdep NVidia oss seq version card1 devices modules NVidia_1 pcm timers user@box:~$ aplay -l aplay: device_list:240: no soundcards found... user@box:~$ sudo /sbin/alsa-utils start * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1403: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... amixer: Invalid command! ...done. Any help appreciated. PS my video card is connected only through the PCI-E slot. I assume there is no extra audio connection required.

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  • NAudio demos not working anymore

    - by Kurru
    I just tried to run the NAudio demos and I'm getting a weird error: System.BadImageFormatException: Could not load file or a ssembly 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' or one o f its dependencies. An attempt was made to load a program with an incorrect form at. File name: 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' at NAudioWpfDemo.AudioGraph..ctor() at NAudioWpfDemo.ControlPanelViewModel..ctor(IWaveFormRenderer waveFormRender er, SpectrumAnalyser analyzer) in C:\Users\Admin\Downloads\NAudio-1.3\NAudio-1-3 \Source Code\NAudioWpfDemo\ControlPanelViewModel.cs:line 23 at NAudioWpfDemo.MainWindow..ctor() in C:\Users\Admin\Downloads\NAudio-1.3\NA udio-1-3\Source Code\NAudioWpfDemo\MainWindow.xaml.cs:line 15 WRN: Assembly binding logging is turned OFF. To enable assembly bind failure logging, set the registry value [HKLM\Software\M icrosoft\Fusion!EnableLog] (DWORD) to 1. Note: There is some performance penalty associated with assembly bind failure lo gging. To turn this feature off, remove the registry value [HKLM\Software\Microsoft\Fus ion!EnableLog]. Since the last time I used NAudio demos I have changed from 32bit Windows XP to 64bit Windows 7. Would this cause this issue? Its very annoying as I was about to try my hand at audio in C# again

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  • How can I find the song position of a song being played with XACT?

    - by DJ SymBiotiX
    So I'm making a game in XNA and I need to use XACT for my songs (rather than media player). I need to use XACT because each song will have multiple layers that combine when played at the same time (bass, lead, drums) etc. I cant use the media player because the media player can only play one song at a time. Anyways, so lets say I have a song playing with XACT in my project with the following code public SongController() { audioEngine = new AudioEngine(@"Content\Song1\Song1.xgs"); waveBank = new WaveBank(audioEngine, @"Content\Song1\Layers.xwb"); soundBank = new SoundBank(audioEngine, @"Content\Song1\SongLayers.xsb"); songTime = new PlayTime(); Vox = soundBank.GetCue("Vox"); BG = soundBank.GetCue("BG"); Bass = soundBank.GetCue("Bass"); Lead = soundBank.GetCue("Lead"); Other = soundBank.GetCue("Other"); Vox.SetVariable("CueVolume", 100.0f); BG.SetVariable("CueVolume", 100.0f); Bass.SetVariable("CueVolume", 100.0f); Lead.SetVariable("CueVolume", 100.0f); Other.SetVariable("CueVolume", 100.0f); _bassVol = 100.0f; _voxVol = 100.0f; _leadVol = 100.0f; _otherVol = 100.0f; Vox.Play(); BG.Play(); Bass.Play(); Lead.Play(); Other.Play(); } So when I look at the variables in Vox, or BG (they are Cue's btw) I cant seem to find any play position in them. So I guess the question is: Is there a variable I can query to find that data, or do I need to make my own class that starts counting up from the time I start the song? Thanks

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  • SoundManager2 has irregular latency

    - by Stefan Monov
    I'm playing some notes at regular intervals. Each one is delayed by a random number of milliseconds, creating a jarring irregular effect. How do I fix it? Note: I'm OK with some latency, just as long as it's consistent. Answers of the type "implement your own small SoundManager2 replacement, optimized for timing-sensitive playback" are OK, if you know how to do that :) but I'm trying to avoid rewriting my whole app in flash for now. For an example of app with zero audible latency see the flash-based ToneMatrix. Testcase (see it here live or get it in an zip): <head> <title></title> <script type="text/javascript" src="http://www.schillmania.com/projects/soundmanager2/script/soundmanager2.js"> </script> <script type="text/javascript"> soundManager.url = '.' soundManager.flashVersion = 9 soundManager.useHighPerformance = true soundManager.useFastPolling = true soundManager.autoLoad = true function recur(func, delay) { window.setTimeout(function() { recur(func, delay); func(); }, delay) } soundManager.onload = function() { var sound = soundManager.createSound("test", "test.mp3") recur(function() { sound.play() }, 300) } </script> </head> <body> </body> </html>

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  • How do I play back a WAV in ActionScript?

    - by Jeremy White
    Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

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  • How can a silverlight app download and play an mp3 file from a URL?

    - by Edward Tanguay
    I have a small Silverlight app which downloads all of the images and text it needs from a URL, like this: if (dataItem.Kind == DataItemKind.BitmapImage) { WebClient webClientBitmapImageLoader = new WebClient(); webClientBitmapImageLoader.OpenReadCompleted += new OpenReadCompletedEventHandler(webClientBitmapImageLoader_OpenReadCompleted); webClientBitmapImageLoader.OpenReadAsync(new Uri(dataItem.SourceUri, UriKind.Absolute), dataItem); } else if (dataItem.Kind == DataItemKind.TextFile) { WebClient webClientTextFileLoader = new WebClient(); webClientTextFileLoader.DownloadStringCompleted += new DownloadStringCompletedEventHandler(webClientTextFileLoader_DownloadStringCompleted); webClientTextFileLoader.DownloadStringAsync(new Uri(dataItem.SourceUri, UriKind.Absolute), dataItem); } and: void webClientBitmapImageLoader_OpenReadCompleted(object sender, OpenReadCompletedEventArgs e) { BitmapImage bitmapImage = new BitmapImage(); bitmapImage.SetSource(e.Result); DataItem dataItem = e.UserState as DataItem; CompleteItemLoadedProcess(dataItem, bitmapImage); } void webClientTextFileLoader_DownloadStringCompleted(object sender, DownloadStringCompletedEventArgs e) { DataItem dataItem = e.UserState as DataItem; string textFileContent = e.Result.ForceWindowLineBreaks(); CompleteItemLoadedProcess(dataItem, textFileContent); } Each of the images and text files are then put in a dictionary so that the application has access to them at any time. This works well. Now I want to do the same with mp3 files, but all information I find on the web about playing mp3 files in Silverlight shows how to embed them in the .xap file, which I don't want to do since I wouldn't be able to download them dynamically as I do above. How can I download and play mp3 files in Silverlight like I download and show images and text?

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  • Silverlight MediaElement Position Property Weirdness

    - by BarrettJ
    I have a MediaElement that is reporting its position incorrectly and weirdly, but consistently. It seems like when it gets to the last second of the audio (and it's always the last second, regardless if the sound is two seconds or 10), it doesn't update it's position until it finishes. Example output: Playback Progress: 0/3.99 - 0 Playback Progress: 0.01/3.99 - 0 Playback Progress: 0.03/3.99 - 0 Playback Progress: 0.06/3.99 - 1 Playback Progress: 0.07/3.99 - 1 Playback Progress: 0.08/3.99 - 2 Playback Progress: 0.11/3.99 - 2 Playback Progress: 0.14/3.99 - 3 Playback Progress: 0.19/3.99 - 4 Playback Progress: 0.23/3.99 - 5 Playback Progress: 0.25/3.99 - 6 Playback Progress: 0.28/3.99 - 7 Playback Progress: 0.3/3.99 - 7 Playback [SNIP] Playback Progress: 2.8/3.99 - 70 Playback Progress: 2.83/3.99 - 70 Playback Progress: 2.88/3.99 - 72 Playback Progress: 2.9/3.99 - 72 Playback Progress: 2.91/3.99 - 72 Playback Progress: 2.92/3.99 - 73 Playback Progress: 2.99/3.99 - 74 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3.99/3.99 - 100 That is the result of: WriteLine("Playback Progress: " + Position + "/" + LengthInSeconds + " - " + (int)((Position / LengthInSeconds) * 100)); public double Position { get { return my_media_element != null ? my_media_element.Position.TotalSeconds : 0; } } public double LengthInSeconds { get { return my_media_element != null ? my_media_element.NaturalDuration.TimeSpan.TotalSeconds : 0; } } Anyone have any ideas why this is occurring?

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  • How to get a volume measurement of iPhone recording in dB, with a limit of at least 120dB

    - by Cyber
    Hello, I am trying to make a simple volume meter for the iPhone. I want the volume displayed in dB. When using this turorial, I am only getting measurements up to 78 dB. I've read that that is because the dBFS spectrum for 16 bit audio recordings is only 96 dB. I tried modifying this piece of code in the init funcyion: dataFormat.mSampleRate = 44100.0f; dataFormat.mFormatID = kAudioFormatLinearPCM; dataFormat.mFramesPerPacket = 1; dataFormat.mChannelsPerFrame = 1; dataFormat.mBytesPerFrame = 2; dataFormat.mBytesPerPacket = 2; dataFormat.mBitsPerChannel = 16; dataFormat.mReserved = 0; I changed the value of mBitsPerChannel, hoping to increase the bit value of the recording. dataFormat.mBitsPerChannel = 32; With that variable set to 32, the "mAveragePower" function returns only 0. So, how can i measure more decibels? All my code is practically the same as in the tutorial i posted above. Thanks in advance, Thomas

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  • Can't play wav file from Javascript in Firefox for Mac

    - by Mike Royle
    I have the following html file that plays a wav file when the user hovers over the 'Play' anchor tag. It works perfectly on IE, Chrome, Firefox, Opera, Safari on both Windows and Mac - except for Firefox on the Mac which does not play the file. <html> <head> <title></title> <script> function PlayAudio() { var s = document.getElementById("soundFile"); s.Play(); } </script> </head> <body> <embed src="MySound.wav" enablejavascript="true" type="audio/wav" autostart="false" width="0" height="0" id="soundFile" /> <a href="#" onmouseover="PlayAudio()">Play</a> </body> </html> If the autostart attribute of the embed tag is set to true then the wav file plays as expected in Firefox for Mac, but not on the mouseover of the anchor tag. Any ideas?

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  • Debug NAudio MP3 reading difference?

    - by Conrad Albrecht
    My code using NAudio to read one particular MP3 gets different results than several other commercial apps. Specifically: My NAudio-based code finds ~1.4 sec of silence at the beginning of this MP3 before "audible audio" (a drum pickup) starts, whereas other apps (Windows Media Player, RealPlayer, WavePad) show ~2.5 sec of silence before that same drum pickup. The particular MP3 is "Like A Rolling Stone" downloaded from Amazon.com. Tested several other MP3s and none show any similar difference between my code and other apps. Most MP3s don't start with such a long silence so I suspect that's the source of the difference. Debugging problems: I can't actually find a way to even prove that the other apps are right and NAudio/me is wrong, i.e. to compare block-by-block my code's results to a "known good reference implementation"; therefore I can't even precisely define the "error" I need to debug. Since my code reads thousands of samples during those 1.4 sec with no obvious errors, I can't think how to narrow down where/when in the input stream to look for a bug. The heart of the NAudio code is a P/Invoke call to acmStreamConvert(), which is a Windows "black box" call which I can't think how to error-check. Can anyone think of any tricks/techniques to debug this?

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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