Search Results

Search found 5304 results on 213 pages for 'audio streaming'.

Page 47/213 | < Previous Page | 43 44 45 46 47 48 49 50 51 52 53 54  | Next Page >

  • Method for launching audio player on Android from web page for streaming media

    - by Brad
    To link to SHOUTcast/HTTP internet radio streams, traditionally you would link to a playlist file, such as an M3U or PLS. From there, the browser would launch the audio player registered to handle the playlist. This works great on any PC, Palm, Blackberry, and iPhone. This method does not work in Android without installing extra software. Sure, Just Playlists or StreamFurious can handle it just fine, but I am assuming there has to be a way to invoke the audio or video player commonly installed by default on Android installations. By default, no audio player is capable of handling M3U or PLS. The player seems to open it, but says "Unsupported Media Type". To make this more annoying, the browser is capable of streaming MP3 audio over HTTP, simply by opening a link to an MP3 file. I have tried simply linking directly to the MP3 stream hosted by SHOUTcast, which should end up in the same result, but SHOUTcast detects "Mozilla" in the user-agent string, and instead of sending the stream, it sends the information page for the station. How should I link to a SHOUTcast stream on Android, from a normal mobile site, without using extra applications?

    Read the article

  • Recording slow web stream

    - by Budric
    I'm trying to record an mpeg2 video stream from a website that doesn't have the greatest bandwidth. The video often buffers. I want to download the stream and watch it offline. The extract stream format received is: Stream #0.0[0x44]: Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s Stream #0.1[0x45]: Video: mpeg2video (Main), yuv420p, 704x576 [PAR 16:11 DAR 16:9], 15000 kb/s, 27.19 fps, 25 tbr, 90k tbn, 50 tbc I use the following tool to transocde the stream: ffmpeg -i "http://url" -y -vcodec libx264 -b 3000k -acodec copy /tmp/stream.mp4 Unfortunately after a few seconds ffmpeg stops recording with an error [mpegts @ 0x1f0b9c0] PES packet size mismatch [mp2 @ 0x1f14640] incomplete frame Error while decoding stream #0.0 [mpeg2video @ 0x1f16860] ac-tex damaged at 0 26 [mpeg2video @ 0x1f16860] Warning MVs not available I've tried encoding with vlc as well with similar issues. Although vlc doesn't stop encoding, the output video has regions where it hangs. vlc -I dummy "http://url" --network-caching="1000" --sout="#transcode{vcodec=h264,vb=3000,acodec=mp3,ab=192}:std{access=file,mux=mp4,dst=/tmp/stream.mp4}" [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 9 33 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 132 DC, 132 AC, 132 MV errors [mpeg2video @ 0x7f2d4c001e20] ac-tex damaged at 16 17 [mpeg2video @ 0x7f2d4c001e20] Warning MVs not available [mpeg2video @ 0x7f2d4c001e20] concealing 836 DC, 836 AC, 836 MV errors libdvbpsi error (PSI decoder): TS discontinuity (received 4, expected 3) for PID 0 I also tried flv transcoding and it shows up with its own set of issues, like output flv file hangs in certain parts. Anyone know what's wrong or how to fix this?

    Read the article

  • How to send audio stream via UDP in java?

    - by Nob Venoda
    Hi to all :) I have a problem, i have set MediaLocator to microphone input, and then created Player. I need to grab that sound from the microphone, encode it to some lower quality stream, and send it as a datagram packet via UDP. Here's the code, i found most of it online and adapted it to my app: public class AudioSender extends Thread { private MediaLocator ml = new MediaLocator("javasound://44100"); private DatagramSocket socket; private boolean transmitting; private Player player; TargetDataLine mic; byte[] buffer; private AudioFormat format; private DatagramSocket datagramSocket(){ try { return new DatagramSocket(); } catch (SocketException ex) { return null; } } private void startMic() { try { format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 8000.0F, 16, 2, 4, 8000.0F, true); DataLine.Info info = new DataLine.Info(TargetDataLine.class, format); mic = (TargetDataLine) AudioSystem.getLine(info); mic.open(format); mic.start(); buffer = new byte[1024]; } catch (LineUnavailableException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } private Player createPlayer() { try { return Manager.createRealizedPlayer(ml); } catch (IOException ex) { return null; } catch (NoPlayerException ex) { return null; } catch (CannotRealizeException ex) { return null; } } private void send() { try { mic.read(buffer, 0, 1024); DatagramPacket packet = new DatagramPacket( buffer, buffer.length, InetAddress.getByName(Util.getRemoteIP()), 91); socket.send(packet); } catch (IOException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void run() { player = createPlayer(); player.start(); socket = datagramSocket(); transmitting = true; startMic(); while (transmitting) { send(); } } public static void main(String[] args) { AudioSender as = new AudioSender(); as.start(); } } And only thing that happens when I run the receiver class, is me hearing this Player from the sender class. And I cant seem to see the connection between TargetDataLine and Player. Basically, I need to get the sound form player, and somehow convert it to bytes[], therefore I can sent it as datagram. Any ideas? Everything is acceptable, as long as it works :)

    Read the article

  • Is the Finch audio library for iPhone capable of doing this?

    - by mystify
    I need to: - start / stop sounds with lengths between 0.1 and 10 seconds - change the playback volume I want to / would like to / would be nice to have to: - change the playback speed - change the playback pitch / frequency - pause an sound and resume playing it later - play a sound backwards Is Finch my best friend here?

    Read the article

  • Is there simple way to play an rtp video/audio stream in WPF?

    - by Robin
    I need to create a WPF control that will play an rtp stream with the requirement that the latency needs to be as low as possible. I've looked at the following two projects: http://vlcdotnet.codeplex.com/ http://wpfmediakit.codeplex.com/ As far as I know, I can't use VLC because we're shipping a commercial application with a more restrictive license than GPL (i.e. we can't ship our source). Wpf media kit is nice, but I can't seem to find a good/free rtp directshow source filter and I wanted to ask if there is a simpler solution out there that I'm missing before I jump into writing my own. Any ideas?

    Read the article

  • how to get audio frequency data from a wave file?

    - by potlee
    I want to build a speech recognition engine in ruby. I know i'll never get there, doing it just for fun. I need to get data for the frequencies of the sound stored in a wav file to compare with data i already have of different sounds that i want to recognize. I will write the code in ruby but i dont think there are any libraries for this written in ruby, they would be too slow if there were any anyway. The good thing about ruby is I'll be able to use libraries for .net via IronRuby or Java via Jruby. How can i get the frequency data?

    Read the article

  • Re-streaming RTMP stream

    - by Yvan JANSSENS
    I have a set of local RTMP stream servers in my network, but I want them to be reachable outside. The bandwidth is too narrow to serve multiple clients on the streamservers of my network, so the idea is to pull the local RTMP streams on a computer serving as a gateway, which pushes them on his turn to a hosted streaming provider. It is not possible to let the sources of the stream push their stream directly to the server outside due to network policy restrictions. Scheme of what I'm trying to accomplish: Internal network | External network ------------ ------------ ----------------------- | internal | <---- | Gateway | ------> | streamserver outside| | streams | ------------ ----------------------- ------------ | ^ | | | ----------- | | clients | | ----------- My question now is: which application which can pull a live stream from an RTMP source (Flash Media Server) and push it to another one (Flash Media Server at hosting provider).

    Read the article

  • Audio and video streaming using Network Simulator in Linux

    - by Parth_90
    I am working on a project which is to show the simulation of streaming of audio and video data in wireless networks. I want to show the simulation that involves a base station, with few wireless stations. The base station should start sending data once it computes a certain value . On receiving the data, each wireless must begin communicating with the base station. I have gone through basic NS-2 tutorials from over here but I am not getting how to go about integrating it with my project. Can anyone tell me how to do it using NS2 or any other network simulator?

    Read the article

  • Creating a custom NAS compatible with the Mac Time machine and for media streaming

    - by Bobby Alexander
    I am planning to assemble a custom NAS machine using an Intel Atom processor. I need the NAS for the following purposes: It should be accessible from by Windows PC so that I can dump data on the NAS (installations, media etc) It should be accessible from my Macbook for the above use. I should be able to use it with the Mac time machine software for backup. The media should be available to my PS3 for streaming. I should be able to access it from my iphone. All the above features should be available over wireless. The time machine feature is very important. Is this even possible? Can someone provide resources on how I can assemble such a machine and setup the required software on it? Much appreciated.

    Read the article

  • Use MTOM/streaming from C# calling a webservice in java exposed via jaxws

    - by raticulin
    We have this webservice created with jax-ws @WebService(name = "Mywebser", targetNamespace = "http://namespace") @MTOM(threshold = 2048) @SOAPBinding(style = SOAPBinding.Style.DOCUMENT, use = SOAPBinding.Use.LITERAL, parameterStyle = SOAPBinding.ParameterStyle.WRAPPED) public class Mywebser { @WebMethod(operationName = "doStreaming", action = "urn:doStreaming") @WebResult(name = "return") public ResultInfo doStreaming(String username, String pwd, @XmlMimeType("application/octet-stream") DataHandler data, boolean overw){ ... } } The generated client side looks like this: @WebMethod(action = "urn:doStreaming") @WebResult(targetNamespace = "") @RequestWrapper(localName = "doStreaming", targetNamespace = "http://namespace", className = "com.mypack.client.doStreaming") @ResponseWrapper(localName = "doStreamingResponse", targetNamespace = "http://namespace", className = "com.mypack.client.doStreamingResponse") public ResultInfo doStreaming( @WebParam(name = "arg0", targetNamespace = "") String arg0, @WebParam(name = "arg1", targetNamespace = "") String arg1, @WebParam(name = "arg2", targetNamespace = "") DataHandler arg2, @WebParam(name = "arg3", targetNamespace = "") boolean arg3); By using it this way it uses streaming properly (verified we can pass an argument of 80mb when the jvm had less allowed. MywebserService serv = ...; Mywebser wso = serv.getMywebserPort(new MTOMFeature()); Map<String, Object> ctxt = ((BindingProvider) wso).getRequestContext(); ctxt.put(JAXWSProperties.HTTP_CLIENT_STREAMING_CHUNK_SIZE, 8192); DataHandler dataHandler = new DataHandler(new FileDataSource("c:\\temp\\A.dat")); arcres = wso.doStreaming("a", "b", dataHandler, true); We generate a clienet for .net, with VS2008, using "Add Web Reference", we get this C# code: [System.Web.Services.Protocols.SoapDocumentMethodAttribute("urn:doStreaming",RequestNamespace="http://namespace",ResponseNamespace="http://namespace",Use=System.Web.Services.Description.SoapBindingUse.Literal,ParameterStyle=System.Web.Services.Protocols.SoapParameterStyle.Wrapped)] [return: System.Xml.Serialization.XmlElementAttribute("return",Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] public ResultInfo doStreaming( [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] string arg0, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] string arg1, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified,DataType="base64Binary")] byte[] arg2, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] bool arg3) Apparently this is not using streaming? The type base64Binary of arg2 seems not the right one? In java it's a DataHandler. By testing it with low memory on the java side we can see it is not using streaming as it fails with OOM. Does someone knows if this is possible, and if so how? Our environment: server: jdk1.6, jaxws 2.1.7 client: C# 2.0, visual studio 2008

    Read the article

  • .net IHTTPHandler Streaming SQL Binary Data

    - by Yisman
    Hello everybody I am trying to implement an ihttphandeler for streaming files. files may be tiny thumbnails or gigantic movies the binaries r stored in sql server i looked at a lot of code online but something does not make sense isnt streaming supposed to read the data piece by piece and move it over the line? most of the code seems to first read the whole field from mssql to memory and then use streaming for the output writing wouldnt it b more eficient to actually stream from disk directly to http byte by byte (or buffered chunks?) heres my code so far but cant figure out the correct combination of the sqlreader mode and the stream object and the writing system Public Sub ProcessRequest(ByVal context As HttpContext) Implements IHttpHandler.ProcessRequest context.Response.BufferOutput = False Dim FileField=safeparam(context.Request.QueryString("FileField")) Dim FileTable=safeparam(context.Request.QueryString("FileTable")) Dim KeyField=safeparam(context.Request.QueryString("KeyField")) Dim FileKey=safeparam(context.Request.QueryString("FileKey")) Using connection As New SqlConnection(ConfigurationManager.ConnectionStrings("Main").ConnectionString) Using command As New SqlCommand("SELECT " & FileField & "Bytes," & FileField & "Type FROM " & FileTable & " WHERE " & KeyField & "=" & FileKey, connection) command.CommandType = Data.CommandType.Text enbd using end using end sub please be aware that this sql command also returns the file extension (pdf,jpg,doc...) in the second field of the query thank you all very much

    Read the article

  • How to route KVM virtual machine audio to Ubuntu 11.10 host using virt-manager?

    - by iGadget
    I've been using KVM in combination with Virt-Manager and Remmina at a fair success up until now. The issue I need to solve now is to get audio from a virtualized Windows XP and make it audible on the Ubuntu 11.10 host. Remmina / RDP works for 'simple' audio (system sounds and such), but when the source gets trickier (e.g. Flash audio), Remmina / RDP messes up. So I figured I'd just connect to the machine directly using Virt-Manager. Unfortunately, it seems that even though I have successfully configured the AC97 audio device on WinXP, it's unable to get it's output to the Ubuntu host. This is probably because Virt-Manager uses VNC (and AFAIK, VNC doesn't transport audio). Does anyone know if there is a solution to fix this? I've heard of Spice, but the installation required so much voodoo last time I checked, I figured I'd let that solution boil to maturity a little longer ;) But perhaps there are other options I haven't thought of yet (which don't require switching to VirtualBox / VMware)...

    Read the article

  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

    Read the article

  • Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

    - by Ryan Thompson
    To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

    Read the article

< Previous Page | 43 44 45 46 47 48 49 50 51 52 53 54  | Next Page >