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  • can not get jplayer plugin to work

    - by Richard
    Hello, I hope somebody has some experience with the jplayer plugin I have been staring at the sourcecode of the demo's and looking in firebug, but I can't see why it is not showing at all. It also try's to use the flash file, but in other examples the embed code does not show up in the container div either. How could I get this to work, or debug? $(document).ready(function(){ $("#jpId").jPlayer( { ready: function () { this.element.jPlayer("setFile", "/mp3/nobodymove.mp3"); // Defines the mp3 } }); }); thanks, Richard

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  • VB FFT - stuck understanding relationship of results to frequency

    - by WaveyDavey
    Trying to understand an fft (Fast Fourier Transform) routine I'm using (stealing)(recycling) Input is an array of 512 data points which are a sample waveform. Test data is generated into this array. fft transforms this array into frequency domain. Trying to understand relationship between freq, period, sample rate and position in fft array. I'll illustrate with examples: ======================================== Sample rate is 1000 samples/s. Generate a set of samples at 10Hz. Input array has peak values at arr(28), arr(128), arr(228) ... period = 100 sample points peak value in fft array is at index 6 (excluding a huge value at 0) ======================================== Sample rate is 8000 samples/s Generate set of samples at 440Hz Input array peak values include arr(7), arr(25), arr(43), arr(61) ... period = 18 sample points peak value in fft array is at index 29 (excluding a huge value at 0) ======================================== How do I relate the index of the peak in the fft array to frequency ?

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  • how to play video from url

    - by priyanka
    i am biginner in android development and try to play video from link but it's give error "sorry,we can't play this video" i tride so many links but for all links its show same error. My code it follwing public class VideoDemo extends Activity { private static final String path ="http://demo.digi-corp.com/S2LWebservice/Resources/SampleVideo.mp4"; private VideoView video; private MediaController ctlr; @Override public void onCreate(Bundle icicle) { super.onCreate(icicle); getWindow().setFormat(PixelFormat.TRANSLUCENT); setContentView(R.layout.videoview); video = (VideoView) findViewById(R.id.video); video.setVideoPath(path); ctlr = new MediaController(this); ctlr.setMediaPlayer(video); video.setMediaController(ctlr); video.requestFocus(); } } thansk in advance

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  • NAudio Mp3 Playback in Console

    - by Kurru
    Hi I'm trying to make a helper dll that will simplify the NAudio framework into a subset of functions I'm likely to need but I've hit a stumbling block right off the bat. I'm trying to use the following code to play an mp3 but I'm not hearing anything at all. Any help would be appreciated! static WaveOut waveout; static WaveStream playback; static System.Threading.ManualResetEvent wait = new System.Threading.ManualResetEvent(false); static void Main(string[] args) { System.Threading.Thread t = new System.Threading.Thread(new System.Threading.ThreadStart(PlaySong)); t.Start(); wait.WaitOne(); System.Threading.Thread.Sleep(2 * 1000); waveout.Stop(); waveout.Dispose(); playback.Dispose(); } static void PlaySong() { waveout = new WaveOut(); playback = OpenMp3Stream(@"songname.mp3"); waveout.Init(playback); waveout.Play(); Console.WriteLine("Started"); wait.Set(); } private static WaveChannel32 OpenMp3Stream(string fileName) { WaveChannel32 inputStream; WaveStream mp3Reader = new Mp3FileReader(fileName); WaveStream pcmStream = WaveFormatConversionStream.CreatePcmStream(mp3Reader); WaveStream blockAlignedStream = new BlockAlignReductionStream(pcmStream); inputStream = new WaveChannel32(blockAlignedStream); return inputStream; }

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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 continuously update data to an asp page?

    - by Lori
    Hi, I have an asp page based on a very simple database. It references a single table of probably 30 records and maybe 12 data fields and everything works great as I am only uploading a new database every week or so. I have a special circumstance where I would like upload new data to the database and display automatically on the page every 20 to 30 seconds without the user having to refresh their screen. I would expect up to 1000 concurrent users accessing the data. I have been manually uploading the database via ftp, which will obviously not work on this timeline and would also run the risk of error pages as the database is being replaced. So, can anyone point me the right direction to setup this scenario? Other details that might be helpful: The database is an Access database (but I could change to another format if needed) Running on Windows platform hosted by an ISP, not my own server Thanks in advance for any help on this! Lori

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  • Sound/Silence in a wav file.

    - by Vivek
    Hi, I am searching for a utility/code that could detect and let me know if my 1 minute wav file contains sound or not ? Other way, if it could detect the duration of the silence(if exists) at any position in the wav file, that would also server the purpose. Does SOX support any command for that ? I tried with Java, but didnt found anything in JMF. Thanks Vivek

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  • Use a "x-dom-event-stream" stream in javascript ?

    - by rnaud
    Hello, HTML5 draft contains an API called EventSource to stream data (notifications) trough javascript using only one server call. Looking it up, I found an exemple on Opera Labs of the javascript part : document.getElementsByTagName("event-source")[0] .addEventListener("server-time", eventHandler, false); function eventHandler(event) { // Alert time sent by the server alert(event.data); } and the server side part : <?php header("Content-Type: application/x-dom-event-stream"); while(true) { echo "Event: server-time\n"; $time = time(); echo "data: $time\n"; echo "\n"; flush(); sleep(3); } ?> But as of today, it seems only Opera has implemented the API, neither Chrome nor Safari have a working version (Am I wrong here ?) So my question is, is there any other way in javascript, maybe more complex, to use this one stream to get data ?

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  • Actionscript: NetStream stutters after buffering.

    - by meandmycode
    Using NetStream to stream content from http, I've noticed that esp with certain exported h264's, if the player encounters an empty buffer, it will stop and buffer to the requested length (as expected). However once the buffer is full, the playback doesn't resume, but instead jumps ahead, as such- instantly playing the buffered duration in a brief moment, and thusly triggering an empty buffer again.. this will then continue over and over. Presumably when the netstream pauses to buffer, the playhead position continues, and the player is attempting to snap to that position on resume- however given it could take 5 seconds to build a 2 second buffer- it ends up with a useless buffer again.. (this is an assumption) I've attempted to work around this by listening for an empty buffer netstatus event, pausing the stream, and at the same time setting up a loop to check the current buffer length vs the requested buffer length.. and resuming once the buffer length is greater than or equal to the requested buffer.. however this causes problems when there isn't enough of the video remaining.. for example, a 10 second buffer with only 5 seconds remaining, the loop just sits there waiting for a buffer length of 10 seconds when theres only 5 left... You would think that you could simply check which was smaller, the time left or the requested buffer length.. however the times flash gives are not accurate.. If you add the net streams current time index, plus the buffered time, the total is not the entire duration of the movie (when at the end).. it is close but not the same. This brings me back to the original problem, and if there is another way to fix this, clearly flash knows when the buffer is ready, so how can i get flash pause when it buffers, and resume once the buffer is ready? currently it doesn't.. it pauses and then once the buffer is full- it plays the entire buffered content in about .1 of a second. Thanks in advance, Stephen.

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  • Detect and record a sound with python

    - by Jean-Pierre
    I'm using this program to record a sound in python: import pyaudio import wave import sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = "output.wav" p = pyaudio.PyAudio() stream = p.open(format = FORMAT, channels = CHANNELS, rate = RATE, input = True, frames_per_buffer = chunk) print "* recording" all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) print "* done recording" stream.close() p.terminate() write data to WAVE file data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() I want to change the program to start recording when sound is detected by the sound card input. Probably should compare the input sound level in Chunk, but how do this?

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  • J2ME Camera and Sound Recorder Access On A Windows Mobile

    - by Steven Knox
    I'm currently involved in a research project that requires me to access a Windows Mobile Camera and sound recorder with J2ME to, well take pictures and record sound... the phone has to be a windows mobile for some reason that has nothing to do with me and the software has to be written in Java, also not my decision. So I need to try and find a phone that supports this (if one exists) so I'd like to know if anyone has found one? Thank You For Your Help. (Note the phone supporting MMAPI (JSR 135) does not imply that you can use the camera and sound recorder, our current phone has this and has not access).

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  • Video encoding Help

    - by Pedro
    Hi guys, I'm doing one research on video encoding tools for flv. I tested flvtool2 and Yamddi, but I'm losing lots of quality of video. Does anyone recommend any other tool or algorithm to keep the maximum quality of the movie in flv? Regards, Pedro

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  • Is there a way to make PHP progressively output as the script executes?

    - by Iain Fraser
    So I'm writing a disposable script for my own personal single use and I want to be able see how the process is going. Basically I'm processing a couple of thousand media releases and sending them to our new CMS. So I don't hammer the CMS, I'm making the script sleep for a couple of seconds after every 5 requests. I would like - as the script is executing - to be able to see my echos telling me the script is going to sleep or that the last transaction with the webservice was successful. Is this possible in PHP? Thanks for your help! Iain

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  • Playing sounds in iPhone SDK?

    - by seanny94
    Does anyone have a snippet that uses the AudioToolBox framework that can be used to play a short sound? I would be grateful if you shared it with me and the rest of the community. Everywhere else I have looked doesn't seem to be too clear with their code. Thanks!

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  • hosting environment for delivering FLVs [closed]

    - by Gotys
    What would be the ideal hardware setup for pushing lots of bandwith on a tube site? We have ever-expanding cloud storage where users upload the movies, then we have these web-delivery machines which cache the FLV files on its local harddrives and deliver them to users. Each cache machine can deliver 1200 mbits/s , if it has SAS 8 harddrives. Such a cache machine costs us $550/month for 8x160gb -- so each machine can cache only 160GB at any given time. If we want to cache more then 160gb , we need to add another machine..another $550/month..etc. This is very un-economical so I am wondering if we have any experts here who can figure out a better setup. I've been looking into "gluster FS", but I am not sure if this thing can push a lot of bandwith. Any ideas highly appreciated. Thank you!

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  • Man pages for libvlc

    - by mawia
    Hi! all. Though I am not sure wether this question belongs to here but pardon me if not so. Can you people guide me to the manual page of libvlc functions like(just give a kind of pointer where these functions are described in detail) void libvlc_playlist_pause( libvlc_instance *, libvlc_exception ) mtime_t libvlc_input_get_length( libvlc_input_t *, libvlc_exception ) mtime_t libvlc_input_get_time( libvlc_input_t *, libvlc_exception ) void libvlc_input_set_time( libvlc_input_t *, mtime_t , libvlc_exception ) float libvlc_input_get_position( libvlc_input_t *, libvlc_exception ) void libvlc_input_set_position( libvlc_input_t *, float , libvlc_exception ) void libvlc_set_rate( libvlc_input_t *, float rate, libvlc_exception ) float libvlc_get_rate( libvlc_input_t *, libvlc_exception ) libvlc_input_get_information( libvlc_input_t *, libvlc_exception ) In particular can you please describe the functioning of libvlc_playlist_pause.I am using this in my aplication to run a video stream.My video is running but since video file is coming over a network I need to pause the player for a particular amount till enough data is buffered. With regards Mawia

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  • How to stream your images/files with VLC? (C# .Net)

    - by Ole Jak
    So I know there are lot of wrapers of libVLC.dll . But I just do not know what one is ready to do what I need... What I need is simple... in my C# programm I create some bitmap (once or twice per second)... I now want to stream bitmaps live as video (in some format VLC can to offer me) to some http:localhost:port/ using VLC... What libvlc.dll wrapper can help me with that?

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  • (Android SDk 2.1) Getting error when I use setAudioSource and setVideoSource

    - by Rainfer
    I got the follow error when I run setAudioSource and setVideoSource. 03-16 10:26:25.302: ERROR/audio_input(52): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value 03-16 10:26:25.302: ERROR/audio_input(52): VerifyAndSetParameter failed 03-16 10:26:25.302: ERROR/CameraInput(52): Unsupported parameter(x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value) 03-16 10:26:25.302: ERROR/CameraInput(52): VerifiyAndSetParameter failed on parameter #0 This error happen on both emulator and the device. (I am using Google nexus one) I have set the CAMERA and RECORD_AUDIO user permission already. I spent many days but I still cannot figure out what is the cause of this runtime error.

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  • Displaying Video using a Window Handle

    - by fergs
    I'm working on a C# wrapper for Dallmeier camera's and currently have a working wrapper. I can connect to a camera via passing the window handle (in my application its a picture box handle), this is used to send video and messages. Once connected I can then send the StartLiveView command and then a live stream video will be shown in the picture box. Can someone explain how this works by just giving the window handle? And how can I grab an Image from this stream when Picturebox1.Image is null?

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  • NAudio playback wont stop successfully

    - by Kurru
    Hi When using NAudio to playback an mp3 [in the console], I cant figure out how to stop the playback. When I call waveout.Stop() the code just stops running and waveout.Dispose() never gets called. Is it something to do with the function callback? I dont know how to fix that if it is. static string MP3 = @"song.mp3"; static WaveOut waveout; static WaveStream playback; static void Main(string[] args) { waveout = new WaveOut(WaveCallbackInfo.FunctionCallback()); playback = OpenMp3Stream(MP3); waveout.Init(playback); waveout.Play(); Console.WriteLine("Started"); Thread.Sleep(2 * 1000); Console.WriteLine("Ending"); if (waveout.PlaybackState != PlaybackState.Stopped) waveout.Stop(); Console.WriteLine("Stopped"); waveout.Dispose(); Console.WriteLine("1st dispose"); playback.Dispose(); Console.WriteLine("2nd dispose"); } private static WaveChannel32 OpenMp3Stream(string fileName) { WaveChannel32 inputStream; WaveStream mp3Reader = new Mp3FileReader(fileName); WaveStream pcmStream = WaveFormatConversionStream.CreatePcmStream(mp3Reader); WaveStream blockAlignedStream = new BlockAlignReductionStream(pcmStream); inputStream = new WaveChannel32(blockAlignedStream); return inputStream; }

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