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  • OSMF seek with Amazon Cloudfront

    - by giorrrgio
    I've written a little OSMF player that streams via RTMP from Amazon Cloudfront. There's a known issue, the mp3 duration is not correctly readed from metadata and thus the seek function is not working. I know there's a workaround implying the use of getStreamLength function of NetConnection, which I successfully implemented in a previous non-OSMF player, but now I don't know how and when to call it, in terms of OSMF Events and Traits. This code is not working: protected function initApp():void { //the pointer to the media var resource:URLResource = new URLResource( STREAMING_PATH ); // Create a mediafactory instance mediaFactory = new DefaultMediaFactory(); //creates and sets the MediaElement (generic) with a resource and path element = mediaFactory.createMediaElement( resource ); var loadTrait:NetStreamLoadTrait = element.getTrait(MediaTraitType.LOAD) as NetStreamLoadTrait; loadTrait.addEventListener(LoaderEvent.LOAD_STATE_CHANGE, _onLoaded); player = new MediaPlayer( element ); //Marker 5: Add MediaPlayer listeners for media size and current time change player.addEventListener( DisplayObjectEvent.MEDIA_SIZE_CHANGE, _onSizeChange ); player.addEventListener( TimeEvent.CURRENT_TIME_CHANGE, _onProgress ); initControlBar(); } private function onGetStreamLength(result:Object):void { Alert.show("The stream length is " + result + " seconds"); duration = Number(result); } private function _onLoaded(e:LoaderEvent):void { if (e.newState == LoadState.READY) { var loadTrait:NetStreamLoadTrait = player.media.getTrait(MediaTraitType.LOAD) as NetStreamLoadTrait; if (loadTrait && loadTrait.netStream) { var responder:Responder = new Responder(onGetStreamLength); loadTrait.connection.call("getStreamLength", responder, STREAMING_PATH); } } }

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  • WPF - Transparency - Stream Desktop Content

    - by Niels Willems
    Greetings I'm in the process of making a Scoreboard for a game (Starcraft II). This scoreboard is being made as a WPF Application with a C# code-behind. I already have a version which works for 90% in WinForms but I lacked the support to easily make it look a lot nicer which are available in WPF. The point of this application will be to form a kind of overlay on top of a running game. This game is in Fulscreen(Windowed Mode) so when in WinForms I coded it so that it should always be on top. It would do so and that was no problem. Since the main look of the app in WPF is based on an image with a transparent background I have set most Background values to Transparent. However when I do this the entire application does not get registered by streaming software. For example it just shows my Desktop or the game I'm playing but not my application even though it IS there. I can see it with my own eyes but the audience on the stream cannot. Does anyone have any experience with this matter because it's really doing my head in. My entire application will be useless if it is not visible on streams. If I have to put the background on a color rather than transparent the UI will be completely demolished as well in terms of looks. I'm basically trying to make a game-overlay in C# & WPF. I have read you can do this on different ways as well but I have little to no knowledge of C++ nor do I know anything about DirectX Thank you for your time reading and your possible insights. Edit: The best solution would be an overlay similar to that one of Steam/Xfire/Dolby Axon. Edit 2: I've had no luck with all the suggestions so I basically made the transparent bits of my image non transparent and let the user decide which one to use depending on what streaming software they would be using.

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  • Syncing two AS3 NetStreams

    - by Lowgain
    I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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  • Creating C++ client app for some abstract windows server - how to manage TCP connection to server speed?

    - by Kabumbus
    So we have some server with some address port and ip. we are developing that server so we can implement on it what ever we need for help. What are standard/best practices for data transfer speed management between C++ windows client app and server (C++)? My main point is in how to get how much data can be uploaded/downloaded from/to client via his low speed network to my relatively super fast server. (I need it for set up of his live stream Audio/Video bit rate) My try on explaining number 3. We do not care how fast is our server. It is always faster than needed. We care about client tyring to stream out to our server his media. he streams encoded (via ffmpeg) live video data to our server. But he has say ADSL with 500kb/s of outgoing traffic. Also he uses some ICQ or what so ever so he has less than 500 kb/s per second. And he wants to stream live video! So we need to set up our ffmpeg to encode video with respect to the bit rate user can provide. We develop server side and client side. We need a way of finding out how much user can upload per second currently (so value can change dynamically over time)

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  • App-Engine Parse a UrlFetch UTF-8 encoded stream

    - by Davidrd91
    I am trying to parse an XML from a URL using the xml.sax parser. I know there are other libraries to use but coming from Java this is the one I am most familiar with and seems the least complicated to me. The code I'm using to parse is as follows: parser = xml.sax.make_parser() handler = MangaHandler() parser.setContentHandler(handler) url = urlfetch.Fetch('http://www.mangapanda.com/alphabetical', allow_truncated = False, follow_redirects = False, deadline = False) xml.sax.parseString(url.content, handler) This returns a SaxException (invalid token) once the parser reaches the first & sign: SAXParseException: <unknown>:582:34: not well-formed (invalid token) Because urlfetch returns a string and not a stream I cannot use the parse() (which only works with streams) and am left to use parseString() instead. To see if parsing as a stream would fix this I tried: parser.parse(io.StringIO(url.content).encode('utf-8')) but this returns: TypeError: initial_value must be unicode or None, not str I have also tried to use the urllib2 libraries which do return a stream instead of urlfetch but the file is too large and is automatically truncated, leaving me with missing data. Any Sort of work-around for this would be greatly appreciated as I've spent days getting around one obstacle just to be stopped by another.

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  • Loading saved byte array to memory stream causes out of memory exception

    - by user2320861
    At some point in my program the user selects a bitmap to use as the background image of a Panel object. When the user does this, the program immediately draws the panel with the background image and everything works fine. When the user clicks "Save", the following code saves the bitmap to a DataTable object. MyDataSet.MyDataTableRow myDataRow = MyDataSet.MyDataTableRow.NewMyDataTableRow(); //has a byte[] column named BackgroundImageByteArray using (MemoryStream stream = new MemoryStream()) { this.Panel.BackgroundImage.Save(stream, ImageFormat.Bmp); myDataRow.BackgroundImageByteArray = stream.ToArray(); } Everything works fine, there is no out of memory exception with this stream, even though it contains all the image bytes. However, when the application launches and loads saved data, the following code throws an Out of Memory Exception: using (MemoryStream stream = new MemoryStream(myDataRow.BackGroundImageByteArray)) { this.Panel.BackgroundImage = Image.FromStream(stream); } The streams are the same length. I don't understand how one throws an out of memory exception and the other doesn't. How can I load this bitmap? P.S. I've also tried using (MemoryStream stream = new MemoryStream(myDataRow.BackgroundImageByteArray.Length)) { stream.Write(myDataRow.BackgroundImageByteArray, 0, myDataRow.BackgroundImageByteArray.Length); //throw OoM exception here. }

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  • Spawning vim from a node git hook

    - by Lawrence Jones
    I've got a project purely in coffeescript, with git hooks for deployment also written in cs. I don't really want to break away from the language just to use bash for a quick commit message formatter, but I've got a problem spawning vim from the commit-msg hook. I've seen here that when piping to vim, the stdio is not necessarily set correctly to the tty streams. I get how that could cause a problem, but I don't exactly know how to get vim to load correctly using nodes spawn command. At the moment I have... vim = (require 'child_process').spawn('vim', [file], stdio: 'inherit') vim.on 'exit', (err) -> console.log "Exited! [#{err}]" cb?() ...which works fine to spawn a vim process that can r/w from the parents stdio, but when I use this in the hook things go wrong. Vim states that the stdio is not from terminal, and then once opened typing causes escape characters to pop up all over the place. Backspace for example, will produce ^?. Any help would be appreciated!

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  • Generating jquery 'rules' from business model to UI in asp.net mvc

    - by jim
    Hi all, I've had a good look around and am certain that there's no matching question on SO, so here goes. Has anyone created a 'helper' method on their model that generates jquery (or plain javascript) rules validation dynamically, based on the criteria/rules that are contained within the object and taken from a repository (i.e. DB). What i'm thinking of is a discrete set of partial views (and associated models) that have rules at the business logic 'level' and rather than (or in combination with) validating the rule(s) at postback, translating the same rules into tightly focussed jquery methods that work identically at client (js) and server (c#) levels. I can see benefits here re performance. Also, the rules definitions could be created in a single place (in c#) and the jquery generated off of that, thus allowing single edits to update both code streams. I appreciate that there would be limitations imposed by language specific contstraints but the general principle could be quite interesting if used appropriately. I'm also aware that testibility could be an issue when using two different language structures and hoping to achieve similar test outcomes - but those aside... any thoughts or experiences of similar out there?? cheers jimi

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  • Using stringstream instead of `sscanf` to parse a fixed-format string

    - by John Dibling
    I would like to use the facilities provided by stringstream to extract values from a fixed-format string as a type-safe alternative to sscanf. How can I do this? Consider the following specific use case. I have a std::string in the following fixed format: YYYYMMDDHHMMSSmmm Where: YYYY = 4 digits representing the year MM = 2 digits representing the month ('0' padded to 2 characters) DD = 2 digits representing the day ('0' padded to 2 characters) HH = 2 digits representing the hour ('0' padded to 2 characters) MM = 2 digits representing the minute ('0' padded to 2 characters) SS = 2 digits representing the second ('0' padded to 2 characters) mmm = 3 digits representing the milliseconds ('0' padded to 3 characters) Previously I was doing something along these lines: string s = "20101220110651184"; unsigned year = 0, month = 0, day = 0, hour = 0, minute = 0, second = 0, milli = 0; sscanf(s.c_str(), "%4u%2u%2u%2u%2u%2u%3u", &year, &month, &day, &hour, &minute, &second, &milli ); The width values are magic numbers, and that's ok. I'd like to use streams to extract these values and convert them to unsigneds in the interest of type safety. But when I try this: stringstream ss; ss << "20101220110651184"; ss >> setw(4) >> year; year retains the value 0. It should be 2010. How do I do what I'm trying to do? I can't use Boost or any other 3rd party library, nor can I use C++0x.

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  • How to pass value in url using web request GET method asp.net

    - by Narasi
    Am new to .Net and web service..i like to pass id through url..how to do that?whether it was done post or get method?guide me string url = "http://XXXXX//"+id=22; WebRequest request = WebRequest.Create(url); request.Proxy.Credentials = new NetworkCredential(xxxxx); request.Credentials = CredentialCache.DefaultCredentials; //add properties request.Method = "GET"; request.ContentType = "application/json"; //convert byte[] byteArray = Encoding.UTF8.GetBytes(data); request.ContentLength = byteArray.Length; //post data Stream streamdata = request.GetRequestStream(); streamdata.Write(byteArray, 0, byteArray.Length); streamdata.Close(); //response WebResponse response = request.GetResponse(); // Get the stream containing content returned by the server. Stream dataStream = response.GetResponseStream(); // Open the stream using a StreamReader for easy access. StreamReader reader = new StreamReader(dataStream); // Read the content. string responseFromServer = reader.ReadToEnd(); // Clean up the streams and the response. reader.Close(); response.Close(); Thanks in advance

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  • How to use libavformat for a separate encoder?

    - by Brendon Tsai
    I've build a encoder based on the sample of QUALCOMM, which captures the video and compresses it into h264 file. I am using Android 4.2.2. Now I want to add a mp4 muxer into this encoder(actually, just video will be fine, I don't need audio). I want to use FFMpeg. But after I read the example, I found out that the muxer was using the encoder of FFMpeg. I don't know how to use the muxer part for another encoder. I've read this post, but I don't understand how the code provide video stream to the muxer. I think that mainly because I don't understand these code: AVCodecContext * strmCodec = oFmtCtx->streams[0]->codec; // Fill the required properties for codec context. // *from the documentation: // *The user sets codec information, the muxer writes it to the output. // *Mandatory fields as specified in AVCodecContext // *documentation must be set even if this AVCodecContext is // *not actually used for encoding. my_tune_codec(strmCodec); Can anyone give me a hint? Thank you!

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  • deserializing multiple types from a stream

    - by clanier9
    I have a card game program, and so far, the chat works great back and forth over the TCPClient streams between host and client. I want to make it do this with serializing and deserializing so that I can also pass cards between host and client. I tried to create a separate TCPClient stream for the passing of cards but it didn't work and figured it may be easier to keep one TCPClient stream that gets the text messages as well as cards. So I created a class, called cereal, which has the properties for the cards that will help me rebuild the card from an embedded database of cards on the other end. Is there a way to make my program figure out whether a card has been put in the stream or if it's just text in the stream so I can properly deserialize it to a string or to a cereal? Or should I add a string property to my cereal class and when that property is filled in after deserializing to the cereal, i'll know it's just text (if that field is empty after deserializing i'll know it's a card)? I'm thinking a try catch, where it tries to deserialize to a string, and if it fails it will catch and cast as a cereal. Or am I just way off base with this and should choose another route? I'm using visual studio 2011, am using a binaryformatter, and am new to serializing/deserializing.

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  • std::ifstream buffer caching

    - by ledokol
    Hello everybody, In my application I'm trying to merge sorted files (keeping them sorted of course), so I have to iterate through each element in both files to write the minimal to the third one. This works pretty much slow on big files, as far as I don't see any other choice (the iteration has to be done) I'm trying to optimize file loading. I can use some amount of RAM, which I can use for buffering. I mean instead of reading 4 bytes from both files every time I can read once something like 100Mb and work with that buffer after that, until there will be no element in buffer, then I'll refill the buffer again. But I guess ifstream is already doing that, will it give me more performance and is there any reason? If fstream does, maybe I can change size of that buffer? added My current code looks like that (pseudocode) // this is done in loop int i1 = input1.read_integer(); int i2 = input2.read_integer(); if (!input1.eof() && !input2.eof()) { if (i1 < i2) { output.write(i1); input2.seek_back(sizeof(int)); } else input1.seek_back(sizeof(int)); output.write(i2); } } else { if (input1.eof()) output.write(i2); else if (input2.eof()) output.write(i1); } What I don't like here is seek_back - I have to seek back to previous position as there is no way to peek 4 bytes too much reading from file if one of the streams is in EOF it still continues to check that stream instead of putting contents of another stream directly to output, but this is not a big issue, because chunk sizes are almost always equal. Can you suggest improvement for that? Thanks.

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  • Reading and writing in parallel

    - by Malfist
    I want to be able to read and write a large file in parallel, or if not in parallel, at least in blocks so that I don't use up so much memory. This is my current code: // Define memory stream which will be used to hold encrypted data. MemoryStream memoryStream = new MemoryStream(); // Define cryptographic stream (always use Write mode for encryption). CryptoStream cryptoStream = new CryptoStream(memoryStream, encryptor, CryptoStreamMode.Write); //start encrypting using (BinaryReader reader = new BinaryReader(File.Open(fileIn, FileMode.Open))) { byte[] buffer = new byte[1024 * 1024]; int read = 0; do { read = reader.Read(buffer, 0, buffer.Length); cryptoStream.Write(buffer, 0, read); } while (read == buffer.Length); } // Finish encrypting. cryptoStream.FlushFinalBlock(); // Convert our encrypted data from a memory stream into a byte array. //byte[] cipherTextBytes = memoryStream.ToArray(); //write our memory stream to a file memoryStream.Position = 0; using (BinaryWriter writer = new BinaryWriter(File.Open(fileOut, FileMode.Create))) { byte[] buffer = new byte[1024 * 1024]; int read = 0; do { read = memoryStream.Read(buffer, 0, buffer.Length); writer.Write(buffer, 0, read); } while (read == buffer.Length); } // Close both streams. memoryStream.Close(); cryptoStream.Close(); As you can see, it reads the entire file into memory, encrypts it, then writes it out. If I happen to be encrypting files that are very large (2GB+) it tends not to work, or at the very least, consumes ~97% of my memory. How could I do it in a more effective manner?

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  • Which options do I have for Java process communication?

    - by Dmitriy Matveev
    We have a place in a code of such form: void processParam(Object param) { wrapperForComplexNativeObject result = jniCallWhichMayCrash(param); processResult(result); } processParam - method which is called with many different arguments. jniCallWhichMayCrash - a native method which is intended to do some complex processing of it's parameter and to create some complex object. It can crash in some cases. wrapperForComplexNativeObject - wrapper type generated by SWIG processResult - a method written in pure Java which processes it's parameter by creation of several kinds (by the kinds I'm not meaning classes, maybe some like hierarchies) of objects: 1 - Some non-unique objects which are referencing each other (from the same hierarchy), these objects can have duplicates created from the invocations of processParam() method with different parameter values. Since it's costly to keep all the duplicates it's necessary to cache them. 2 - Some unique objects which are referencing each other (from the same hierarchy) and some of the objects of 1st kind. After processParam is executed for each of the arguments from some set the data created in processResult will be processed together. The problem is in fact that jniCallWhichMayCrash method may crash the entire JVM and this will be very bad. The reason of crash may be such that it can happen for one argument value and not for the other. We've decided that it's better to ignore crashes inside of JVM and just skip some chunks of data when such crashes occur. In order to do this we should run processParam function inside of separate process and pass the result somehow (HOW? HOW?! This is a question) to the main process and in case of any crashes we will only lose some part of data (It's ok) without lose of everything else. So for now the main problem is implementation of transport between different processes. Which options do I have? I can think about serialization and transmitting of binary data by the streams, but serialization may be not very fast due to object complexity. Maybe I have some other options of implementing this?

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  • Audio Streaming Latency

    - by killianmcc
    I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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  • What is PHP like as a programming language?

    - by seanlinmt
    I am not really familiar with PHP, but I get the impression that it is like JavaScript (syntax-wise). What are the benefits of a dynamically typed language, when compared to a strongly typed language like C# or Java, and how would this help in the context of web development? What would make a dynamically typed language so attractive? Or, does the popularity of PHP have more to do with it being free? Okay, I think I better give a little more background to get more meaningful answers, because I am not wanting a flame war. I come from a C background, and when I moved into C# and Visual Studio. Having code completion, integration with an SQL database, huge existing class libraries and easy to access documentation, as well as new tools such as LINQ and ReSharper was like heaven. I didn't enjoy JavaScript before JQuery, but now I love it as well. Recently, I ported a PHP project over to C# and I used Zend to help me debug and understand more while porting - instead of maintaining two code streams. That also cut down on the cost of the server and maintenance. Getting into PHP would be nice. I think that Visual Studio has spoiled me - but again Eclipse is also equally spoiling. It would be nice to have an answer from someone who has experience developing both under PHP and .NET.

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  • BluetoothChat doesn't work

    - by jes
    Hello I want to make conversation between android devices. I use BluetoothChat to do this but it doesn't work I can't read correctly data from another device. Conversation is : Me: privet Device: p Device: rivet Can you help me? private class ConnectedThread extends Thread { private final InputStream mmInStream; private final OutputStream mmOutStream; public ConnectedThread(BluetoothSocket socket) { Log.d(TAG, "create ConnectedThread"); mmSocket = socket; //InputStream tmpIn = null; OutputStream tmpOut = null; BufferedInputStream tmpIn=null; int INPUT_BUFFER_SIZE=32; // Get the BluetoothSocket input and output streams try { //tmpIn = socket.getInputStream(); tmpOut = socket.getOutputStream(); tmpIn = new BufferedInputStream(socket.getInputStream(),INPUT_BUFFER_SIZE); } catch (IOException e) { Log.e(TAG, "temp sockets not created", e); } mmInStream = tmpIn; mmOutStream = tmpOut; } public void run() { Log.i(TAG, "BEGIN mConnectedThread"); byte[] buffer = new byte[1024]; int bytes; // Keep listening to the InputStream while connected while (true) { try { // Read from the InputStream bytes = mmInStream.read(buffer); // Send the obtained bytes to the UI Activity mHandler.obtainMessage(BluetoothChat.MESSAGE_READ, bytes, -1, buffer) .sendToTarget(); } catch (IOException e) { Log.e(TAG, "disconnected", e); connectionLost(); break; } } }

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  • C#, How to download file into string with progress callback?

    - by Kaminari
    I would like to use the WebClient (or there is another better option?) but there is a problem. I understand that opening up the stream takes some time and this can not be avoided. However, reading it takes a strangely much more amount of time compared to read it entirely immediately. Of course it's not working good because i'm not so familiar with streams. Is there a best way to do this? I mean two ways, to string and to file. Progress is my own delegate and it's working good. FIRST UPDATE: Ok, now i got something like this and it seems to work but still slow: System.Net.WebClient client = new System.Net.WebClient(); System.IO.Stream streamRemote = client.OpenRead(new Uri(URL)); if (savePath == null) { StreamReader reader = new StreamReader(streamRemote); int iByteSize = 0; byte[] byteBuffer = new byte[iSize]; char[] charBuffer = new char[iSize]; StringBuilder sb = new StringBuilder(); while ((iByteSize = reader.Read(charBuffer, 0, iSize)) > 0) { sb.Append(charBuffer, 0, iByteSize); iRunningByteTotal += iByteSize; float dIndex = (float)(iRunningByteTotal); float dTotal = (float)byteBuffer.Length; float dProgressPercentage = (dIndex / dTotal); float iProgressPercentage = (dProgressPercentage * 100); if (Progress != null) Progress(iProgressPercentage); } result = sb.ToString(); } Im wondering about DownloadStringAsync method?

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  • R from java with no graphics: is it worth moving to JRI

    - by LH
    I have a system set up that's been happily running R from a java servlet, spawning processed & hooking into the process's stdin, stdout, and stderr streams, as in the second andwer to this question. After a system upgrade (that included glibc), the input is no longer reaching the R process.* Until now, 'R --vanilla --slave -f [file] ...' was working fine for me. I also have no swing dependencies right now, so I'm somewhat reluctant to add them. (I may actually not be able to add swing dependencies; am I right that using REngine automatically brings swing in? The examples import all of swing.) Are there advantages to switching to JRI? What changes would I need to make to my R script? (It currently reads from stdin and writes to stdout). I'm not finding the provided examples terribly helpful for how to use JRI in this situation. Thanks for your help & comments. *I can't even tell if the problem is data being written too soon or too late, but that's a separate issue/question; if I move to JRI I'm hoping it all becomes moot.

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  • ffmpeg hangs when creating a video

    - by FearUs
    I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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  • Having trouble Getting "RTSP over HTTP"

    - by Muhammad Adeel Zahid
    There is an axis camera that is connected to our site (camba.tv) through axis one click connection component (which acts as proxy). We can communicate with this camera only through http by setting the proxy to our OCCC server's address. If we want to get RTSP streams (h.264) we are only left with "RTSP over HTTP" option. For this I have followed axis VAPIX 3 documentation section 3.3. I issue requests through fiddler but don't get any response. But when i put the URL (axrtsphttp://1.00408CBEA38B/axis-media/media.amp) in windows media player (with proxy set to OCCC server 212.78.237.156:3128) the player is able to get RTSP stream over HTTP after logging in. I have created a request trace of communication between camera and windows media player through wireshark and the request that brings the stream looks like http://1.00408cbea38b/axis-media/media.amp HTTP/1.1 x-sessioncookie: 619 User-Agent: Axis AMC Host: 1.00408CBEA38B Proxy-Connection: Keep-Alive Pragma: no-cache Authorization: Digest username="root",realm="AXIS_00408CBEA38B",nonce="000a8b40Y0100409c13ac7e6cceb069289041d8feb1691",uri="/axis-media/media.amp",cnonce="9946e2582bd590418c9b70e1b17956c7",nc=00000001,response="f3cab86fc84bfe33719675848e7fdc0a",qop="auth" HTTP/1.0 200 OK Content-Type: application/x-rtsp-tunnelled Date: Tue, 02 Nov 2010 11:45:23 GMT RTSP/1.0 200 OK CSeq: 1 Content-Type: application/sdp Content-Base: rtsp://1.00408CBEA38B/axis-media/media.amp/ Date: Tue, 02 Nov 2010 11:45:23 GMT Content-Length: 410 v=0 o=- 1288698323798001 1288698323798001 IN IP4 1.00408CBEA38B s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:* a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:30.0 a=transform:1,0,0;0,1,0;0,0,1 a=control:trackID=1 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeNQFAe2AtwEBAaQeJEV,aM48gA== RTSP/1.0 200 OK CSeq: 2 Session: 3F4763D8; timeout=60 Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=060922C6;mode="PLAY" Date: Tue, 02 Nov 2010 11:45:24 GMT RTSP/1.0 200 OK CSeq: 3 Session: 3F4763D8 Range: npt=0- RTP-Info: url=rtsp://1.00408CBEA38B/axis-media/media.amp/trackID=1;seq=7392;rtptime=4190934902 Date: Tue, 02 Nov 2010 11:45:24 GMT [Binary Stream Content] But when i copy this request to fiddler, I only get 200 status code with content-type set to application/x-rtsp-tunneled and there is no stream data. The only thing i do different with stream is to use Basic in authorization header instead of Digest and I do not get 401 (Un authorized) status code. Can anyone explain what's happening here? How can I write request sequences to get stream in fiddler? If it is needed, I can upload the wireshark request dump somewhere.

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  • Loose component cables causing HDMI video problems

    - by jwir3
    I'm not sure this is the correct forum, but I'll ask anyway. I have an A/V setup at home that has something like the following: Five Components (actually a few more, like a CD player, but they don't really relate to this question): Older Pioneer Receiver Digital Set Top Box Sony BluRay Player Samsung Plasma TV Speakers The reason for the receiver is so that all the sound can go through the speakers, rather than some going to the TV speakers and some to the external speakers. They are connected as follows: Digital Set Top Box connects via component video to Samsung TV directly via Component 2 (audio goes to Older Pioneer Receiver). Sony BluRay player is connected via HDMI 1 to TV, but audio goes to the receiver. Now, the problem I'm having is that when I have the digital set top box connected, there are times when the Netflix or Hulu streams I watch through the Sony BluRay player (it's connected to a router for internet access) will lose video. What I mean by this is that the sound of the episode will keep playing, but the screen will go black. If I jiggle the component cables, it will often come back. If I disconnect the component cables, it will always come back. I've noticed that one of the connections (the red component cable) doesn't like to sit very well in the component socket in the back of the digital set top box. It seems like there is a bad connection here, but it doesn't seem like this should be affecting the HDMI input at all. What I've noticed, though, is that when I disconnect the digital set top box completely (i.e. remove the component cable from the back of the TV), the problem seems to resolve itself. I'm not talking about actually removing the cable physically, because I thought perhaps the cables were mashing against one another, and possibly jiggling each other loose. To correct this possible problem, I took the component cable completely out of the cable ties it was in in the back of my entertainment center, as well as pulled the digital set top box out from the entertainment center altogether. It's now connected directly to the TV, without any other cables touching it to cause some kind of weird interference or just physical pulling on the cable. Same problem. If, however, I disconnect the component cable and just leave it sitting behind the TV, then the problem goes away. So, my question is this - what could be causing this? Is it a case where it's an improperly shielded component cable that's causing interference with the HDMI input, or something that's wrong with the TV? It's an intermittent problem, so it's difficult to track down. The TV isn't that old, so it's probably still under warranty. I'm just wondering if there is something else I can do that might reduce this problem without having to haul a massive television set out of my house to get repaired/replaced.

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  • How can I simulate blocking RTMP over port 80 on Windows?

    - by Christian Nunciato
    It seems like this should be so simple, but since this isn't my area of expertise, I'm having a hell of a time figuring out how to do it. Basically, I have a Flash app and I'm connecting to a Flash Media Server to stream some content. The URL I'm using to do this, for example, looks like this: rtmp://someserver.com/some/path/mp3:somefile Everything works -- but that's sort of the problem. When I'm trying to do is simulate my users attempting to play back my media under more restrictive conditions than the ones I have here (i.e., none) -- namely being stuck behind firewalls or proxy servers that block access to RTMP streams. Flash, according to Adobe, is equipped to handle proxy servers and firewalls automatically, like so (from the docs): When you do not specify a port number in an RTMP address, Flash will attempt to connect to port 1935. If it fails it will then try to connect to port 443; if that fails, it will try port 80. [And if that fails, it will attempt to connect via RTMPT (i.e., HTTP tunneling) on port 80.] So no coding is required to access ports 1935, 443, or port 80 if you do not specify a port in the RTMP address. The problem I'm having is setting up a reliable environment in which to test that this behavior actually happens. I'm on a Windows machine, for example, so with Windows Firewall, I can block certain ports and protocols (1935, 443), but I don't want to block port 80, because the final fallback protocol (RTMPT) is supposed to run on port 80, and Windows Firewall only gives me enough granularity (as far as I know, anyway) to block "all outbound TCP traffic to remote port 80" -- that is, I can't, apparently, block "all outbound RTMP traffic to port 80" while leaving RTMPT traffic to port 80 unaffected. My understanding thus far is that I'll probably need to set up a proxy server to do this. Is this correct? Or is there a simpler way (on Win 7, at least) to filter out RTMP to 1935, RTMP to 443, RTMP to 80, but still allow RTMPT to 80 (where all four hostnames are identical)? And if I do have to set up a proxy server, what's the simplest way to go on Windows? I've set up WinProxy, which seems a bit janky but apparently works -- but then what I can't figure out is how to tell Windows to force all TCP traffic (including RTMP, RTMPT and HTTO) through this proxy server so I can turn around and reject the requests for RTMP. Any help would be hugely appreciated. This isn't my realm of expertise and I've alreasdy spent more time on it than I probably should. :)

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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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