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  • How to show a video stored on server on iphone

    - by Amitkumar
    Hi, I have a query regarding showing a video (which is stored on server) on iPhone. I want show a video in an iPhone Application. This is not live streaming. So how the video can be shown? I have read the Apple's documentation for HTTP streaming of video. Do I need to call a Web Service? Is there any tutorial for this? Thanks in advance..

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  • Linux application that bundles multiple incoming audio and video streams into one container file?

    - by StackedCrooked
    I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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  • Play and record streaming audio

    - by Igor
    I'm working on an iPhone app that should be able to play and record audio streaming data simultaneously. Is it actually possible? I'm trying to mix SpeakHere and AudioRecorder samples and getting an empty file with no audio data... Here is my .m code: import "AzRadioViewController.h" @implementation azRadioViewController static const CFOptionFlags kNetworkEvents = kCFStreamEventOpenCompleted | kCFStreamEventHasBytesAvailable | kCFStreamEventEndEncountered | kCFStreamEventErrorOccurred; void MyAudioQueueOutputCallback( void* inClientData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, const AudioTimeStamp inStartTime, UInt32 inNumberPacketDescriptions, const AudioStreamPacketDescription inPacketDesc ) { NSLog(@"start MyAudioQueueOutputCallback"); MyData* myData = (MyData*)inClientData; NSLog(@"--- %i", inNumberPacketDescriptions); if(inNumberPacketDescriptions == 0 && myData-dataFormat.mBytesPerPacket != 0) { inNumberPacketDescriptions = inBuffer-mAudioDataByteSize / myData-dataFormat.mBytesPerPacket; } OSStatus status = AudioFileWritePackets(myData-audioFile, FALSE, inBuffer-mAudioDataByteSize, inPacketDesc, myData-currentPacket, &inNumberPacketDescriptions, inBuffer-mAudioData); if(status == 0) { myData-currentPacket += inNumberPacketDescriptions; } NSLog(@"status:%i curpac:%i pcdesct: %i", status, myData-currentPacket, inNumberPacketDescriptions); unsigned int bufIndex = MyFindQueueBuffer(myData, inBuffer); pthread_mutex_lock(&myData-mutex); myData-inuse[bufIndex] = false; pthread_cond_signal(&myData-cond); pthread_mutex_unlock(&myData-mutex); } OSStatus StartQueueIfNeeded(MyData* myData) { NSLog(@"start StartQueueIfNeeded"); OSStatus err = noErr; if (!myData-started) { err = AudioQueueStart(myData-queue, NULL); if (err) { PRINTERROR("AudioQueueStart"); myData-failed = true; return err; } myData-started = true; printf("started\n"); } return err; } OSStatus MyEnqueueBuffer(MyData* myData) { NSLog(@"start MyEnqueueBuffer"); OSStatus err = noErr; myData-inuse[myData-fillBufferIndex] = true; AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; fillBuf-mAudioDataByteSize = myData-bytesFilled; err = AudioQueueEnqueueBuffer(myData-queue, fillBuf, myData-packetsFilled, myData-packetDescs); if (err) { PRINTERROR("AudioQueueEnqueueBuffer"); myData-failed = true; return err; } StartQueueIfNeeded(myData); return err; } void WaitForFreeBuffer(MyData* myData) { NSLog(@"start WaitForFreeBuffer"); if (++myData-fillBufferIndex = kNumAQBufs) myData-fillBufferIndex = 0; myData-bytesFilled = 0; myData-packetsFilled = 0; printf("-lock\n"); pthread_mutex_lock(&myData-mutex); while (myData-inuse[myData-fillBufferIndex]) { printf("... WAITING ...\n"); pthread_cond_wait(&myData-cond, &myData-mutex); } pthread_mutex_unlock(&myData-mutex); printf("<-unlock\n"); } int MyFindQueueBuffer(MyData* myData, AudioQueueBufferRef inBuffer) { NSLog(@"start MyFindQueueBuffer"); for (unsigned int i = 0; i < kNumAQBufs; ++i) { if (inBuffer == myData-audioQueueBuffer[i]) return i; } return -1; } void MyAudioQueueIsRunningCallback( void* inClientData, AudioQueueRef inAQ, AudioQueuePropertyID inID) { NSLog(@"start MyAudioQueueIsRunningCallback"); MyData* myData = (MyData*)inClientData; UInt32 running; UInt32 size; OSStatus err = AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &running, &size); if (err) { PRINTERROR("get kAudioQueueProperty_IsRunning"); return; } if (!running) { pthread_mutex_lock(&myData-mutex); pthread_cond_signal(&myData-done); pthread_mutex_unlock(&myData-mutex); } } void MyPropertyListenerProc( void * inClientData, AudioFileStreamID inAudioFileStream, AudioFileStreamPropertyID inPropertyID, UInt32 * ioFlags) { NSLog(@"start MyPropertyListenerProc"); MyData* myData = (MyData*)inClientData; OSStatus err = noErr; printf("found property '%c%c%c%c'\n", (inPropertyID24)&255, (inPropertyID16)&255, (inPropertyID8)&255, inPropertyID&255); switch (inPropertyID) { case kAudioFileStreamProperty_ReadyToProducePackets : { AudioStreamBasicDescription asbd; UInt32 asbdSize = sizeof(asbd); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataFormat, &asbdSize, &asbd); if (err) { PRINTERROR("get kAudioFileStreamProperty_DataFormat"); myData-failed = true; break; } err = AudioQueueNewOutput(&asbd, MyAudioQueueOutputCallback, myData, NULL, NULL, 0, &myData-queue); if (err) { PRINTERROR("AudioQueueNewOutput"); myData-failed = true; break; } for (unsigned int i = 0; i < kNumAQBufs; ++i) { err = AudioQueueAllocateBuffer(myData-queue, kAQBufSize, &myData-audioQueueBuffer[i]); if (err) { PRINTERROR("AudioQueueAllocateBuffer"); myData-failed = true; break; } } UInt32 cookieSize; Boolean writable; err = AudioFileStreamGetPropertyInfo(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, &writable); if (err) { PRINTERROR("info kAudioFileStreamProperty_MagicCookieData"); break; } printf("cookieSize %d\n", cookieSize); void* cookieData = calloc(1, cookieSize); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, cookieData); if (err) { PRINTERROR("get kAudioFileStreamProperty_MagicCookieData"); free(cookieData); break; } err = AudioQueueSetProperty(myData-queue, kAudioQueueProperty_MagicCookie, cookieData, cookieSize); free(cookieData); if (err) { PRINTERROR("set kAudioQueueProperty_MagicCookie"); break; } err = AudioQueueAddPropertyListener(myData-queue, kAudioQueueProperty_IsRunning, MyAudioQueueIsRunningCallback, myData); if (err) { PRINTERROR("AudioQueueAddPropertyListener"); myData-failed = true; break; } break; } } } static void ReadStreamClientCallBack(CFReadStreamRef stream, CFStreamEventType type, void *clientCallBackInfo) { NSLog(@"start ReadStreamClientCallBack"); if(type == kCFStreamEventHasBytesAvailable) { UInt8 buffer[2048]; CFIndex bytesRead = CFReadStreamRead(stream, buffer, sizeof(buffer)); if (bytesRead < 0) { } else if (bytesRead) { OSStatus err = AudioFileStreamParseBytes(globalMyData-audioFileStream, bytesRead, buffer, 0); if (err) { PRINTERROR("AudioFileStreamParseBytes"); } } } } void MyPacketsProc(void * inClientData, UInt32 inNumberBytes, UInt32 inNumberPackets, const void * inInputData, AudioStreamPacketDescription inPacketDescriptions) { NSLog(@"start MyPacketsProc"); MyData myData = (MyData*)inClientData; printf("got data. bytes: %d packets: %d\n", inNumberBytes, inNumberPackets); for (int i = 0; i < inNumberPackets; ++i) { SInt64 packetOffset = inPacketDescriptions[i].mStartOffset; SInt64 packetSize = inPacketDescriptions[i].mDataByteSize; size_t bufSpaceRemaining = kAQBufSize - myData-bytesFilled; if (bufSpaceRemaining < packetSize) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; memcpy((char*)fillBuf-mAudioData + myData-bytesFilled, (const char*)inInputData + packetOffset, packetSize); myData-packetDescs[myData-packetsFilled] = inPacketDescriptions[i]; myData-packetDescs[myData-packetsFilled].mStartOffset = myData-bytesFilled; myData-bytesFilled += packetSize; myData-packetsFilled += 1; size_t packetsDescsRemaining = kAQMaxPacketDescs - myData-packetsFilled; if (packetsDescsRemaining == 0) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } } } (IBAction)buttonPlayPressedid)sender { label.text = @"Buffering"; [self connectionStart]; } (IBAction)buttonSavePressedid)sender { NSLog(@"save"); AudioFileClose(myData.audioFile); AudioQueueDispose(myData.queue, TRUE); } bool getFilename(char* buffer,int maxBufferLength) { NSArray paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString docDir = [paths objectAtIndex:0]; NSString* file = [docDir stringByAppendingString:@"/rec.caf"]; return [file getCString:buffer maxLength:maxBufferLength encoding:NSUTF8StringEncoding]; } -(void)connectionStart { @try { MyData* myData = (MyData*)calloc(1, sizeof(MyData)); globalMyData = myData; pthread_mutex_init(&myData-mutex, NULL); pthread_cond_init(&myData-cond, NULL); pthread_cond_init(&myData-done, NULL); NSLog(@"Start"); myData-dataFormat.mSampleRate = 16000.0f; myData-dataFormat.mFormatID = kAudioFormatLinearPCM; myData-dataFormat.mFramesPerPacket = 1; myData-dataFormat.mChannelsPerFrame = 1; myData-dataFormat.mBytesPerFrame = 2; myData-dataFormat.mBytesPerPacket = 2; myData-dataFormat.mBitsPerChannel = 16; myData-dataFormat.mReserved = 0; myData-dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked; int i, bufferByteSize; UInt32 size; AudioQueueNewInput( &myData-dataFormat, MyAudioQueueOutputCallback, &myData, NULL /* run loop /, kCFRunLoopCommonModes / run loop mode /, 0 / flags */, &myData-queue); size = sizeof(&myData-dataFormat); AudioQueueGetProperty(&myData-queue, kAudioQueueProperty_StreamDescription, &myData-dataFormat, &size); CFURLRef fileURL; char path[256]; memset(path,0,sizeof(path)); getFilename(path,256); fileURL = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8*)path, strlen(path), FALSE); AudioFileCreateWithURL(fileURL, kAudioFileCAFType, &myData-dataFormat, kAudioFileFlags_EraseFile, &myData-audioFile); OSStatus err = AudioFileStreamOpen(myData, MyPropertyListenerProc, MyPacketsProc, kAudioFileMP3Type, &myData-audioFileStream); if (err) { PRINTERROR("AudioFileStreamOpen"); return 1; } CFStreamClientContext ctxt = {0, self, NULL, NULL, NULL}; CFStringRef bodyData = CFSTR(""); // Usually used for POST data CFStringRef headerFieldName = CFSTR("X-My-Favorite-Field"); CFStringRef headerFieldValue = CFSTR("Dreams"); CFStringRef url = CFSTR(RADIO_LOCATION); CFURLRef myURL = CFURLCreateWithString(kCFAllocatorDefault, url, NULL); CFStringRef requestMethod = CFSTR("GET"); CFHTTPMessageRef myRequest = CFHTTPMessageCreateRequest(kCFAllocatorDefault, requestMethod, myURL, kCFHTTPVersion1_1); CFHTTPMessageSetBody(myRequest, bodyData); CFHTTPMessageSetHeaderFieldValue(myRequest, headerFieldName, headerFieldValue); CFReadStreamRef stream = CFReadStreamCreateForHTTPRequest(kCFAllocatorDefault, myRequest); if (!stream) { NSLog(@"Creating the stream failed"); return; } if (!CFReadStreamSetClient(stream, kNetworkEvents, ReadStreamClientCallBack, &ctxt)) { CFRelease(stream); NSLog(@"Setting the stream's client failed."); return; } CFReadStreamScheduleWithRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); if (!CFReadStreamOpen(stream)) { CFReadStreamSetClient(stream, 0, NULL, NULL); CFReadStreamUnscheduleFromRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); CFRelease(stream); NSLog(@"Opening the stream failed."); return; } } @catch (NSException *exception) { NSLog(@"main: Caught %@: %@", [exception name], [exception reason]); } } (void)viewDidLoad { [[UIApplication sharedApplication] setIdleTimerDisabled:YES]; [super viewDidLoad]; } (void)didReceiveMemoryWarning { [super didReceiveMemoryWarning]; } (void)viewDidUnload { } (void)dealloc { [super dealloc]; } @end

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  • Is it possible to play multiple audio streams from one "jukebox" to multiple Airport Express devices?

    - by Alex Reynolds
    I have set up a Mac mini as a jukebox that streams audio to an Airport Express in another room in the house, using the AirPlay/AirTunes feature in iTunes. I control this with the iOS Remote app, and this works great. At the present time, it looks like the Mac mini's copy of iTunes gets taken over by the Remote app, while streaming. If I set up a second Airport Express in room B, is there a way to set it up (as well as the jukebox) so that it can receive and play its own unique music stream ("stream B"), separate from what's going on at the Mac mini, or in room A, which is playing stream A? To accomplish this, I would be happy to buy a copy of Rogue Amoeba's AirFoil if it will allow sending multiple, separate audio streams from one computer to the multiple wireless bridges, while using the Remote app (or a Rogue Amoeba equivalent for iOS). However, it is unclear to me from their site documentation, whether that is possible or not. I'd prefer to give the points to an answer that solves this problem. If you don't know if it can be done, or do not think it can be done, please allow others to answer. I appreciate your help. Thanks for your advice.

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  • Play audio over network with Windows 7?

    - by Josh
    I have a unique situation where I'd like to stream audio (ALL audio, not just mp3s, etc) from my laptop to another computer over the network. I live in a studio apartment and my laptop is my main computer but I'd like it's audio to play on my htpc with a nice stereo system. Since it's a studio, both computers are in the same room so I don't want 2 sets of speakers. I want my computer to directly play back through the stereo. I used to do this with pulseaudio but my job now requires that I run Windows full time. I'm aware of Shoutcast and other similar streaming solutions but I don't want any transcoding done. It's a waste of CPU and not to mention my laptop fans, and I don't mind the network bandwidth that uncompressed audio requires. Is there a way to run Shoutcast without encoding? Also, I know that Windows Remote Desktop can play audio over the network pretty easily. Is this part of .Net that I could just code a simple app that streams the audio without RD'ing in? I also don't want to run it over a physical wire. :)

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  • Stream video file in debian?

    - by Rob
    I've tried ffserver with ffmpeg, I've tried VLC, and I'm not sure what else to try or what I've done wrong. I've gone through, with VLC +-[ robert@s10 ]--[ ~ ] +[#!]¬ vlc --version VLC media player 2.0.0 Twoflower (revision 2.0.0-0-g421a4fc) VLC version 2.0.0 Twoflower (2.0.0-0-g421a4fc) Compiled by buildd on biber.debian.org (Mar 1 2012 22:21:37) Compiler: gcc version 4.6.2 (Debian 4.6.2-14) This program comes with NO WARRANTY, to the extent permitted by law. You may redistribute it under the terms of the GNU General Public License; see the file named COPYING for details. Written by the VideoLAN team; see the AUTHORS file. and tried everything I could in the streaming section, but I can't get the stream to actually work. Looking around, apparently debian strips the encoders from the package? I want to do share some videos I've made with friends on IRC, and it would be easiest if I could just stream it so we can all watch at the same time and critique parts of it in real time. Has anyone done something similar? Linux s10 3.2.0-2-686-pae #1 SMP Tue Mar 20 19:48:26 UTC 2012 i686 GNU/Linux Basic home network, I am behind a NAT (192.168.1.*) and have dynamic DNS set up. That doesn't really matter too much, I can figure that out, but it's not even working locally. I have a file server set up and could just share the files that way, but I'd rather have everyone watching at the same time (or just about). Not worried about installing new packages or building something from source, that's not a big issue, just want to get it working. Big plus if I can do it from command line.

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  • Setup for a live (low-latency) audio video broadcast over Wi-Fi?

    - by Majal Mirasol
    The Upgrade We are capturing audio (from mixer) and video (from a camera) from a main auditorium and passing it to separate rooms within the building. We used to have done this via manual audio/video cables and wires. We wanted to "upgrade" the system and wirelessly broadcast the stream via Wi-Fi. The Problem In our current setup (Wirecast running on A10 on a Wireless-N network), we have the problem of delay. Our streams are delayed from a minute up to five minutes on the clients (laptop/iPad/Android). This had not been a problem from the previous wired connections. Since the wireless network is local, we thought that a delay of less than a second should be achievable. Our Question And so it goes. Anybody there who has any experience for a setup that has both low latency and at the same time user-friendly to clients streaming in the program? Any recommendations would be highly appreciated. (Our current setup in on Windows 7, but setup on a dedicated Linux box is preferred, if achievable.)

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  • Why are FMS logs filled with 'play' event status code 408 for a failed webcast?

    - by Stu Thompson
    Recently we had a live webcast event go horribly wrong. I'm doing the technical post-mortem, with limited information. We know that the hardware encoder (a Digital Rapid Touch Stream Web HDI) was unable to send upstream at a sustained reliable high rate. What we don't know is if the encoder's connection was problematic (Zürich), or that of the streaming server (in Frankfurt). Unfortunately, I've got three different vendors all blaming each other (the CDN who runs the server, the on-site ISP and the on-site encoding team.) In the FMS log files I see a couple of interesting things: Zillions of Status Code 408 on play event entries for clients. Adobe's documentation stats that this "Stream stopped because client disconnected". ("Zillions" would be a ratio of 10 events for every individual IP address.) Several unpublish / (re)publish events per hour for the encoder I'd like to know if all those 408s could tell me with authority that the FMS server was starved for bandwidth, or that the encoding signal was starved (and hence the server was disconnecting clients.) Any clues?

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  • Windows Media Based video audio converter?

    - by acidzombie24
    This may seem like an odd question. Right now ONLY windows media player, VLC and media player classic opens and plays my audio video correctly. Virtualdub plays it back with the wrong framerate and losses the audio, Avidemux 2.5 seems to be able to dump the audio/video but the video (like all other apps) is either a bad framerate or is wrong (glitches and bad framerate or bad dump). Nothing recognizes the audio file and when playing the video Avidemux (and most other things) die. FFMPEG cant seem to split the video or audio (using copy -an and etc) and this is getting me very angry. VLC dumps the video incorrectly when i try dumping it with that too. What can i use to convert the video? its streaming so it starts at 26mins in and ends at 28 (this is where apps have the problem. They dont know this and fudge everything or crash). I manage to dump the audio with Avidemux but virtualdub and ffmpeg says unreconized codec. Even if i cant convert it (it seems compressed enough) i want to at least attach it back into an AVI.

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  • L'HADOPI aimerait surveiller les plateformes de streaming, quelles mesures répressives pourraient en découler ?

    L'HADOPI aimerait surveiller les plateformes de streaming, quelles mesures répressives pourraient en découler ? Alors que l'Hadopi a déjà fait moult mécontents, ce chiffre pourrait encore augmenter. La Haute Autorité est en effet consciente que son arrivée à poussé un grand nombre d'internautes vers le streaming. Or, elle n'a de pouvoir d'action que sur les réseaux P2P. De quoi pousser largement la communauté on-line à fuir ces plateformes, et faire naître de nouvelles préoccupations pour le gouvernement. « Pour l'instant, ce qui se dit c'est qu'il y a une migration. Est-ce qu'on l'a constaté ? Non. Dire qu'il y a une migration, ne veut pas dire qu'il y a un effet Hadopi chez le téléchargeur illégal. Cela veut dire ...

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  • All the Gear and No Idea: Suggestions for re-designing my home/office/entertainment network

    - by 5arx
    Help/ Advice/ Suggestions please: I have a load of kit that I love but which currently operate in disconnected, sometimes counter-productive way. Because I never really had a masterplan I just added these things one after another and connected them up in ad hoc ways. Since I bought my Macbook I've found I spend much less time on the MacPro that was until then my main machine. Perversely, as my job involves writing .Net software, I spend a lot of Mac time actually inside a Windows 7 VM. I stream media from the HP box to the PS3 and thus to the TV, but its not without its limitations/annoyances. We listen to each other's iTunes libraries but the music files are all over the place and it would be good to know they were all safely in one location (and fully backed up). I need to come up with a strategy that will allow me to use all the kit for work, play (recording live music, making tunes, iMovie work), pushing/streaming media to the TV and sharing files with my other half (she uses a Windows laptop and her iPod touch). Ideally I'd like to be able to work on any of the machines and have a shared homedrive that was visible to all machines so all my current files were synced up wherever i was. It would be great if I could access everything securely and quickly over the web. I'd also like to be able to set up a background backup process. The kit list thus far: Apple MacPro 8GB/3x250GB RAID0 + 1TB Apple MacBook Pro 13" 8GB/250GB - I spend a lot of my work time on a Windows 7 VM on this. Crappy Acer laptop (for children's use - iPlayer, watching movies/tv files) HP Proliant Server 4GB/80GB+160GB+300GB Sun Ultra 10 2 x 80GB (old, but in top-notch condition) PS3 160GB iPod Classic 2 x 8GB iPod Touch Observations: Part of the problem is our dual use of Windows and OS X - we can't go for a pure NT style roaming profile. Because the server is also used for hosting test/beta applications and a SQL Server db, it can't be dedicated to file serving. The two Macs really could do with sharing a roaming profile or similar. I'd love to be able to do something useful with the Ultra 10. My other half has been trying to throw it away for over five years now and regularly ask what function it serves in my study :-( I've got no shortage of 500GB external USB hard drives iMovie files are very large and ideally would be processed on a RAID system. Apple's TimeMachine isn't so great. If anyone could suggest all or part of a setup that would fulfil some of my requirements I'd be very grateful. I am willing to consider purchasing one or two more bits of kit (an Apple TV and a Squeezebox have been moted by friends) if they will help make efficiencies rather than add to the chaos and confusion. Thanks for looking.

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  • Java - Save video stream from Socket to File

    - by Alex
    I use my Android application for streaming video from phone camera to my PC Server and need to save them into file on HDD. So, file created and stream successfully saved, but the resulting file can not play with any video player (GOM, KMP, Windows Media Player, VLC etc.) - no picture, no sound, only playback errors. I tested my Android application into phone and may say that in this instance captured video successfully stored on phone SD card and after transfer it to PC played witout errors, so, my code is correct. In the end, I realized that the problem in the video container: data streamed from phone in MP4 format and stored in *.mp4 files on PC, and in this case, file may be incorrect for playback with video players. Can anyone suggest how to correctly save streaming video to a file? There is my code that process and store stream data (without errors handling to simplify): // getOutputMediaFile() returns a new File object DataInputStream in = new DataInputStream (server.getInputStream()); FileOutputStream videoFile = new FileOutputStream(getOutputMediaFile()); int len; byte buffer[] = new byte[8192]; while((len = in.read(buffer)) != -1) { videoFile.write(buffer, 0, len); } videoFile.close(); server.close(); Also, I would appreciate if someone will talk about the possible "pitfalls" in dealing with the conservation of media streams. Thank you, I hope for your help! Alex.

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  • Reading data from an open HTTP stream

    - by allenjones
    Hi, I am trying to use the .NET WebRequest/WebResponse classes to access the Twitter streaming API here "http://stream.twitter.com/spritzer.json". I need to be able to open the connection and read data incrementally from the open connection. Currently, when I call WebRequest.GetResponse method, it blocks until the entire response is downloaded. I know there is a BeginGetResponse method, but this will just do the same thing on a background thread. I need to get access to the response stream while the download is still happening. This just does not seem possible to me with these classes. There is a specific comment about this in the Twitter documentation: "Please note that some HTTP client libraries only return the response body after the connection has been closed by the server. These clients will not work for accessing the Streaming API. You must use an HTTP client that will return response data incrementally. Most robust HTTP client libraries will provide this functionality. The Apache HttpClient will handle this use case, for example." They point to the Appache HttpClient, but that doesn't help much because I need to use .NET. Any ideas whether this is possible with WebRequest/WebResponse, or do I have to go for lower level networking classes? Maybe there are other libraries that will allow me to do this? Thx Allen

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  • How can I eliminate latency in quicktime streamed video

    - by JJFeiler
    I'm prototyping a client that displays streaming video from a HaiVision Barracuda through a quicktime client. I've been unable to reduce the buffer size below 3.0 seconds... for this application, we need as low a latency as the network allows, and prefer video dropouts to delay. I'm doing the following: - (void)applicationDidFinishLaunching:(NSNotification *)aNotification { NSString *path = [[NSBundle mainBundle] pathForResource:@"haivision" ofType:@"sdp"]; NSError *error = nil; QTMovie *qtmovie = [QTMovie movieWithFile:path error:&error]; if( error != nil ) { NSLog(@"error: %@", [error localizedDescription]); } Movie movie = [qtmovie quickTimeMovie]; long trackCount = GetMovieTrackCount(movie); Track theTrack = GetMovieTrack(movie,1); Media theMedia = GetTrackMedia(theTrack); MediaHandler theMediaHandler = GetMediaHandler(theMedia); QTSMediaPresentationParams myPres; ComponentResult c = QTSMediaGetIndStreamInfo(theMediaHandler, 1,kQTSMediaPresentationInfo, &myPres); Fixed shortdelay = 1<<15; OSErr theErr = QTSPresSetInfo (myPres.presentationID, kQTSAllStreams, kQTSTargetBufferDurationInfo, &shortdelay ); NSLog(@"OSErr %d", theErr); [movieView setMovie:qtmovie]; [movieView play:self]; } I seem to be getting valid objects/structures all the way down to the QTSPres, though the ComponentResult and OSErr are both returning -50. The streaming video plays fine, but the buffer is still 3.0seconds. Any help/insight appreciated. J

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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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  • Icecast/shoutcast streaming on Android

    - by Alvin
    Is there a way to stream shoutcast/icecast on the android? Passing the icecast URL to the mediaplayer does not work and after researching the topic it seems it is because android can't play raw aac files without a media container. What can I do to get around this? Thanks

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  • Streaming a webcam from Silverlight 4 (Beta)

    - by Ken Smith
    The new webcam stuff in Silverlight 4 is darned cool. By exposing it as a brush, it allows scenarios that are way beyond anything that Flash has. At the same time, accessing the webcam locally seems like it's only half the story. Nobody buys a webcam so they can take pictures of themselves and make funny faces out of them. They buy a webcam because they want other people to see the resulting video stream, i.e., they want to stream that video out to the Internet, a lay Skype or any of the dozens of other video chat sites/applications. And so far, I haven't figured out how to do that with It turns out that it's pretty simple to get a hold of the raw (Format32bppArgb formatted) bytestream, as demonstrated here. But unless we want to transmit that raw bytestream to a server (which would chew up way too much bandwidth), we need to encode that in some fashion. And that's more complicated. MS has implemented several codecs in Silverlight, but so far as I can tell, they're all focused on decoding a video stream, not encoding it in the first place. And that's apart from the fact that I can't figure out how to get direct access to, say, the H.264 codec in the first place. There are a ton of open-source codecs (for instance, in the ffmpeg project here), but they're all written in C, and they don't look easy to port to C#. Unless translating 10000+ lines of code that look like this is your idea of fun :-) const int b_xy= h->mb2b_xy[left_xy[i]] + 3; const int b8_xy= h->mb2b8_xy[left_xy[i]] + 1; *(uint32_t*)h->mv_cache[list][cache_idx ]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[0+i*2]]; *(uint32_t*)h->mv_cache[list][cache_idx+8]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[1+i*2]]; h->ref_cache[list][cache_idx ]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[0+i*2]>>1)]; h->ref_cache[list][cache_idx+8]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[1+i*2]>>1)]; The mooncodecs folder within the Mono project (here) has several audio codecs in C# (ADPCM and Ogg Vorbis), and one video codec (Dirac), but they all seem to implement just the decode portion of their respective formats, as do the java implementations from which they were ported. I found a C# codec for Ogg Theora (csTheora, http://www.wreckedgames.com/forum/index.php?topic=1053.0), but again, it's decode only, as is the jheora codec on which it's based. Of course, it would presumably be easier to port a codec from Java than from C or C++, but the only java video codecs that I found were decode-only (such as jheora, or jirac). So I'm kinda back at square one. It looks like our options for hooking up a webcam (or microphone) through Silverlight to the Internet are: (1) Wait for Microsoft to provide some guidance on this; (2) Spend the brain cycles porting one of the C or C++ codecs over to Silverlight-compatible C#; (3) Send the raw, uncompressed bytestream up to a server (or perhaps compressed slightly with something like zlib), and then encode it server-side; or (4) Wait for someone smarter than me to figure this out and provide a solution. Does anybody else have any better guidance? Have I missed something that's just blindingly obvious to everyone else? (For instance, does Silverlight 4 somewhere have some classes I've missed that take care of this?)

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  • WCF streaming on asmx ?

    - by phenevo
    Hi, I'he got wcf service for wcf straming. I works. But I must integrate it with our webserice. is there any way, to have webmethod like this: [webmethod] public Stream GetStream(string path) { return Iservice.GetStream(path); } I service is a class which I copy from WCF service to my asmx. And is there any way to integrate App.config from wcf with web.config ?

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  • Node.js as a custom (streaming) upload handler for Django

    - by Gijs
    I want to build an upload-centric app using Django. One way to do this is with nginx's upload module (nonblocking) but it has its problems. Node.js is supposed to be a good candidate for this type of application. But how can I make node.js act as an upload_handler() for Django (http://docs.djangoproject.com/en/1.1/topics/http/file-uploads/#modifying-upload-handlers-on-the-fly) I'm not sure where to look for examples?

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  • FMOD.net streaming, callback and exinfo parameters

    - by Tesserex
    I posted a question on gamedev about how to play nsf files (NES console music) in FMOD. It didn't get any results, but since then I made some progress. I decided that the easiest method was just to compile an existing player into a dll and then call it from C# to populate my buffer. The problem now is getting it to sound right, and making sure all my paremeters are correct. Here are the facts so far: The nsf dll is dealing with shorts, so the data is PCM16. The sample nsf I'm using has a playback rate of 60 Hz. Just for playing around now, I'm using a frequency of 48000. Based on 2 and 3, the dll calculates a necessary buffer size of 48000 / 60hz = 800. This means it will render 800 shorts worth of buffer for every simulated NES frame. I've so far got my C# code to play the nsf, at the correct pitch and tempo, but it's very grainy / fuzzy, which I'm attributing to the fact that the FMOD read callback is giving a data length of 1600, whereas I should be expecting 800. I've tried playing around with all the numbers and it either crashes, or the music changes pitch, tempo, or both. Here's some of my C# code: uint channels = 1, frequency = 48000; FMOD.MODE mode = (FMOD.MODE.DEFAULT | FMOD.MODE.OPENUSER | FMOD.MODE.LOOP_NORMAL); FMOD.Sound sound = new FMOD.Sound(); FMOD.CREATESOUNDEXINFO ex = new FMOD.CREATESOUNDEXINFO(); ex.cbsize = Marshal.SizeOf(ex); ex.fileoffset = 0; ex.format = FMOD.SOUND_FORMAT.PCM16; // does this even matter? It doesn't change my results as long as it's long enough for one update ex.length = frequency; ex.numchannels = (int)channels; ex.defaultfrequency = (int)frequency; ex.pcmreadcallback = pcmreadcallback; ex.dlsname = null; // eventually I will calculate this with frequency / nsf hz, but I'm just testing for now ex.decodebuffersize = 800; // from the dll load_nsf_file("file.nsf", 8, (int)frequency); // 8 is the track number to play var result = system.createSound( (string)null, (mode | FMOD.MODE.CREATESTREAM), ref ex, ref sound); channel = new FMOD.Channel(); result = system.playSound(FMOD.CHANNELINDEX.FREE, sound, false, ref channel); private FMOD.RESULT PCMREADCALLBACK(IntPtr soundraw, IntPtr data, uint datalen) { // from the dll process_buffer(data, (int)800); // if I use datalen, it usually crashes (I can't get datalen to = 800 safely) return FMOD.RESULT.OK; } So here are some of my questions: What is the relationship between exinfo.decodebuffersize, frequency, and the datalen parameter of the read callback? With this code sample, it's coming in as 3200. I don't know where that factor of 4 between it and the decodebuffersize comes from. Is datalen in the callback referring to number of bytes, or shorts? The process_buffer function takes a short array and its length. I would expect fmod is talking about shorts as well because I told it PCM16. Maybe my playback quality is bad for some totally different reason. If so I have no idea where to begin solving that. Any ideas there?

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  • WCF streaming files

    - by Pinu
    I need to pass a memory stream to the WCF server , how do i need to add this data type in my data contract. I will eventually need to convert this to a memory stream and pass it on to my service layer. datacontact[DataMember] Stream str = null; public Stream File { get { return str; } set { str = value; } }

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  • MPMoviePlayerController - streaming works on 3GS, not on anything pre-3GS

    - by Canada Dev
    I am having some serious issues and annoyances with MPMoviePlayerController. In my app you can watch trailers for some movies in .mov format. I have tested with a friend and had users report that it does not work on their device, which are all 3G. I have tested on my own, a 3GS and playback works fine. I have tried on a 1st gen iPhone and it doesn't work. So I am lead to believe it's a memory issue, and that it's simply stopping the playback and returning to the previous screen. Below is the code I use to launch the player, which is straight out of the MoviePlayer example from Apple. MPMoviePlayerController *mp = [[MPMoviePlayerController alloc] initWithContentURL:[NSURL URLWithString:trailerURL]]; if (mp) { self.moviePlayer = mp; [mp release]; [self.moviePlayer play]; } I have tried to check the NSError from the notifications, but the only thing I get is "An unknown playback error occurred" for both the localizedDescription and localizedRecoverySuggestion, making it impossible to figure out exactly why it's not working. I have seen many examples of people who just have issues with the movie player, but it's starting to annoy me that it sometimes seem to work fine and other times it just doesn't (again, appearing like a memory issue). Thanks for any help/feedback provided

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  • PHP Streaming CSV always adds UTF-8 BOM

    - by Mustafa Ashurex
    The following code gets a 'report line' as an array and uses fputcsv to tranform it into CSV. Everything is working great except for the fact that regardless of the charset I use, it is putting a UTF-8 bom at the beginning of the file. This is exceptionally annoying because A) I am specifying iso and B) We have lots of users using tools that show the UTF-8 bom as characters of garbage. I have even tried writing the results to a string, stripping the UTF-8 BOM and then echo'ing it out and still get it. Is it possible that the issue resides with Apache? If I change the fopen to a local file it writes it just fine without the UTF-8 BOM. header("Content-type: text/csv; charset=iso-8859-1"); header("Cache-Control: no-store, no-cache"); header("Content-Disposition: attachment; filename=\"report.csv\""); $outstream = fopen("php://output",'w'); for($i = 0; $i < $report-rowCount; $i++) { fputcsv($outstream, $report-getTaxMatrixLineValues($i), ',', '"'); } fclose($outstream); exit;

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  • Streaming webcam video in Flash using MP4 encoding

    - by Herms
    One of the features of the Flash app I'm working on is to be able to stream a webcam to others. We're just using the built-in webcam support in Flash and sending it through FMS. We've had some people ask for higher quality video, but we're already using the highest quality setting we can in Flash (setting quality to 100%). My understanding is that in the newer flash players they added support for MPEG-4 encoding for the videos. I created a simple test Flex app to try and compare the video quality of the MP4 vs FLV encodings. However, I can't seem to get MP4 to work at all. According to the Flex documentation the only thing I need to do to use MP4 instead of FLV is prepend "mp4:" to the name of the stream when calling publish: Specify the stream name as a string with the prefix mp4: with or without the filename extension. The prefix indicates to the server that the file contains H.264-encoded video and AAC-encoded audio within the MPEG-4 Part 14 container format. When I try this nothing happens. I don't get any events raised on the client side, no exceptions thrown, and my logging on the server side doesn't show any streams starting. Here's the relevant code: // These are all defined and created within the class. private var nc:NetConnection; private var sharing:Boolean; private var pubStream:NetStream; private var format:String; private var streamName:String; private var camera:Camera; // called when the user clicks the start button private function startSharing():void { if (!nc.connected) { return; } if (sharing) { return; } if(pubStream == null) { pubStream = new NetStream(nc); pubStream.attachCamera(camera); } startPublish(); sharing = true; } private function startPublish():void { var name:String; if (this.format == "mp4") { name = "mp4:" + streamName; } else { name = streamName; } //pubStream.publish(name, "live"); pubStream.publish(name, "record"); }

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