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  • How do I maximize a window in one monitor and do other things on the other monitor

    - by RoboShop
    Hi, I have windows 7 installed on my laptop which is a Dell studio XPS 16. I recently set up a second monitor for it. I've noticed that when I load up "full maximisation" applications (I don't know the term) like a game or media center on one monitor, I can't seem to do anything with the other monitor. With the game, I can see the other monitor, I can play a movie on it even, but if I click the mouse button on it, it'll stop the game and alt-tab away. In Media Center, I can't even cross my mouse over the other monitor. So my question is, in Windows 7, is it possible to run programs like this that are designed to be run in one maximised window and still be able to do things on the other window?

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  • Recovered video files won't play

    - by BioGeek
    I have an SD card with pictures and video which malfunctioned. I was able to recover the files with Photorec. The pictures are OK, but wen I try to open the vide files (*.mov extension) in get the following errors when I try to open them in the following programs Windows Media player: "Windows Media Player encountered a problem while playing the file" Quicktime: "Error -2048: Couldn't open the file because it is not a file that QuickTime understands" VLC: it shows the first frame of the video and the sound is just white noise The filesizes look correct so I presume the data is still in there. Is there any way to fix these recovered video files?

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  • youtube video playing without sound when downloaded?

    - by On a serious note
    I have been looking for this since a long long time. I have downloaded a few videos from you-tube using IDM(all mp4). When I opened the video in VLC media player,a black screen appeared and started flickering.The video just does not start. What all I things I did-: updated my VLC Reinstalled my VLC downloaded another media player(wondershare--very very beautiful i must say).It played the video but audio was missing. converted mp4 to flv and then played in both players,still no sound. Its very frustrating as I want to complete my tutorial series asap.FLV works well,but mp4s fall flat. What should I do?

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  • My new laptop won't let me play any songs I download

    - by ryuuko
    I recently got a new laptop. I reinstalled everything because my old computer is loaded with viruses. I downloaded a song to play on iTunes but it just will not play. I tried it with Windows Media Player and it just says it encountered a problem while playing the file. It's the same with every kind of media player on my laptop. It can only play the sample music provided in the laptop, and it plays music and video fine online. Please help me solve this problem.

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  • How can I get it the Free Music Archive audio player or is there a better alternative?

    - by Dennis Hodapp
    I'm looking at free streaming audio players for web browsers that I can use in a project. I really like the audio player used on http://freemusicarchive.org/. Are they using an open source audio player and can I get a hold of it? Or is it closed source? Also if there are any open-source audio players that anybody knows about I'd love to know about them (preferable to have one with no flash). Last thing...is HTML5 going to be able to replace audio streaming players?

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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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  • Stream video from one app to another in OS X (Software video input device)

    - by Josh
    Not sure how specifically to word this... I'm wondering if there's any way to take a video stream from one application, for example VLC Media Player, and present that video stream to other applications as a video input source. For example, could I broadcast a video file from my hard disk to a website that allows video conferencing using a Flash applet? Basically, I'm looking for something like Soundflower, but for video streams. Is this possible? Tags, title, question re-wording suggestions welcomed -- I'm not sure how to properly describe this. Thanks!

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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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  • Can't play stream from TorrentFlux stream

    - by thegreyspot
    Hi I am trying to stream a video from my TorrentFlux-b4rt server. I tried multiple media players, none work. Only VLC was able to produce an error message: input can't be opened: VLC is unable to open the MRL 'mms://...:8080/'. Check the log for details. I have tried multiple computers on different networks and all have the same issue. I am using Windows 7 to play the videos, and the server is Torrentflux-b4rt 1.0-beta2 with ubuntu 9.10. Thank you.

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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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  • Properties!Details (run-time length)

    - by user33666
    When you take clips and turn them into computer playable files, whether .TS or .MOV or .MPG, .MP4, .VOB, etc., do you as the producer actually manually write into some field in a "Details" section of your video editor what the video length is, so that it can be read by the eventual user's video player? Or is this activity performed automatically by the video editor? I'm asking because: when a mouse is passed over a file line item in Windows 7 Shell (Windows Explorer) a little box is displayed where the mouse meets the file line item and provides simple name and size properties, one of which (if it's a media file) is run length. How does that figure get there into the file (later to be read be Windows or OS X)? Thanks for any info.

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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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  • How to open an iPhone compatible M3U file on Windows?

    - by user1158667
    This is how the M3U file looks like: #EXTM3U #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1400000 http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=900000 http://maskedip/http_livestr.str?r=true&id=test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=450000 http://maskedip/http_livestr.str?r=true&id=mobile-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,CODECS="mp4a.40.2",BANDWIDTH=64000 http://maskedup/http_livestr.str?r=true&id=test-audio&k=testkey Clicking on http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey then returns another M3U file in this format: #EXTM3U #EXT-X-TARGETDURATION:10 #EXT-X-MEDIA-SEQUENCE:1361 #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1361.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1362.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1363.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1364.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1365.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1366.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1367.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1368.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1369.ts Anyways, VLC won't recognize it. How can I play this on the PC?

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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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  • Suggestion for live video stream aggregation/switching/forwarding/management software?

    - by deceze
    I'm looking for a software or system that can receive video streams from a number of cameras via a network (RTMP or similar protocol), present a visual overview of all video streams and allow me to forward/send a selected stream to another service (e.g. to a Flash Media Server, or anywhere via RTMP). Basically the digital internet equivalent of a TV studio control panel, which allows a director to put together a live show. Is there any such software at an affordable price? A GUI-less server which can be scripted to switch streams would be good too. I'm not even quite sure what kind of product category this falls into or what search terms to plug into Google. Most results I have come up with have little more than an executive summary description which doesn't tell me anything. Any suggestion welcome.

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  • How to record / capture audio with RecordControl on Java ME, SE K770i

    - by tomaszs
    I want to record sound on my Java ME App on K770i. So I used this: http://java.sun.com/javame/reference/apis/jsr135/javax/microedition/media/control/RecordControl.html example of RecordControl in my code. It goes like this: import java.util.Vector; import javax.microedition.lcdui.Choice; import javax.microedition.lcdui.Command; import javax.microedition.lcdui.CommandListener; import javax.microedition.lcdui.Display; import javax.microedition.lcdui.Displayable; import javax.microedition.lcdui.List; import javax.microedition.media.Manager; import javax.microedition.media.MediaException; import javax.microedition.midlet.MIDlet; import java.io.*; import javax.microedition.lcdui.*; import javax.microedition.media.*; import javax.microedition.media.control.*; import javax.microedition.midlet.*; import javax.microedition.rms.*; (...) try { // Create a Player that captures live audio. Player p = Manager.createPlayer("capture://audio"); p.realize(); // Get the RecordControl, set the record stream, // start the Player and record for 5 seconds. RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); Thread.currentThread().sleep(5000); rc.commit(); p.close(); } catch (IOException ioe) { } catch (MediaException me) { } catch (InterruptedException ie) { } But unfortunately when I try to build it, it tells me: *** Creating directories *** *** Compiling source files *** ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ 2 errors So my question is: why there is no RecordControl class if in documentations it is written this class should be there. Or is there other method to record / capture audio from microfone in Java ME of Sony Ericsson? How do you record sound?

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  • The [2] table entry '[3]' has no associated entry in the Media table. (error 2602)

    - by derekf
    Coworker started getting the above message in the event log and as dialog during install.  Argument [2] was File and argument [3] was a specific file. Error dialog read   Product: (app name) -- The installer has encountered an unexpected error installing this package. This may indicate a problem with this package. The error code is 2602. Package was a vendor-provided MSI that had been installed administratively, and then a patch (.msp) applied to the administrative install point. With some digging we found that the MSI still had the entries in the media table pointing at the CAB files, and that there were several files at the end of the sequence that did not have corresponding entries in the Media table (last sequence 990 in Media table, last entry in File table had sequence 994).  Attributes on files in the File table all had the msidbFileAttributesCompressed (&16384) attribute set, so they were all expecting to be within the CAB files, but since this was an admin install there were no CAB files. Resolved by clearing the Media table (replace with a single entry: Disk ID 1, LastSequence 994) and going through the file table and subtracting 8192 from each entry to mark files as not compressed.  Tested and worked.

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  • post xml to Spring REST server returns Unsupported Media Type

    - by Mayra
    I'm trying to create a simple spring based webservice that supports a "post" with xml content. In spring, I define an AnnotationMethodHandler: <bean id="inboundMessageAdapter" class="org.springframework.web.servlet.mvc.annotation.AnnotationMethodHandlerAdapter"> <property name="messageConverters"> <util:list> <bean class="org.springframework.http.converter.xml.MarshallingHttpMessageConverter"> <property name="marshaller" ref="xmlMarshaller"/> <property name="unmarshaller" ref="xmlMarshaller"/> </bean> </util:list> </property> </bean> And a jaxb based xml marshaller: <bean id="xmlMarshaller" class="org.springframework.oxm.jaxb.Jaxb2Marshaller"> <property name="contextPaths"> <array> <value>com.company.schema</value> </array> </property> <property name="schemas"> <array> <value>classpath:core.xsd</value> </array> </property> </bean> My controller is annotated as follows, where "Resource" is a class autogenerated by jaxb: @RequestMapping(method = POST, value = "/resource") public Resource createResource(@RequestBody Resource resource) { // do work } The result of a webservice call is always "HTTP/1.1 415 Unsupported Media Type". Here is an example service call: HttpPost post = new HttpPost(uri); post.addHeader("Accept", "application/xml"); post.addHeader("Content-Type", "application/xml"); StringEntity entity = new StringEntity(request, "UTF-8"); entity.setContentType("application/xml"); post.setEntity(entity); It seems to me that I am setting the correct media type everywhere possible. Anyone have an ideas?

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  • nginx serving Django static media | 502 bad gateway

    - by MMRUser
    I'm trying to serve Django static media through nginx, Here's my nginx.conf server { listen 7777; listen localhost:7777; server_name example.com; location / { proxy_pass http://localhost:7777; proxy_redirect off; proxy_set_header Host $host; proxy_set_header X-Real-IP $remote_addr; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; client_max_body_size 10m; client_body_buffer_size 128k; proxy_connect_timeout 90; proxy_send_timeout 90; proxy_read_timeout 90; proxy_buffer_size 4k; proxy_buffers 4 32k; proxy_busy_buffers_size 64k; proxy_temp_file_write_size 64k; } location /test-app-media/ { root /sites/mysite/staticmedia/; expires max; } } but give a 502 bad gateway error, the path to /sites/mysite/staticmedia/ is in the nginx root/ is that the problem.. Django running on Apache 2.2 + mod_wsgi nginx 0.7.65 Thanks..

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  • General question about DirectShow.NET, DirectShow and Windows Media Format

    - by Paul Andrews
    I searched and googled for an answer but couldn't find one. Basically I'm developing a webcam/audio streaming application which should capture audio and video from a pc (usb webcam/microphone) and send them to a receiving server. What the server will do with that it's another story and phase two (which I'm skipping for now) I wrote some code using DirectShow and Windows Media Format and it worked great for capture audio/video and sending them to another client, but there's a major problem: latency. Everywhere in the internet everyone gave me the same answer: "sorry dude but media format isn't for video conferencing, their codecs have too high latency". I thought I could skip the .wmv problems but seems like it's not possible to do... this road ends here then. So I saw a few examples with DirectShow.NET which were faster for both audio and video.. my question is: how come that DirectShow.NET is faster and better for video/audio conferencing? Shouldn't it be just a .NET porting of C++'s DirectShow? Am I missing something? I'm a bit confused at this point

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  • What framework would allow for the largest coverage of freelance developers in the media/digital mar

    - by optician
    This question is not about which is the best, it is about which makes the most business sense to use as a company's platform of choice for ongoing freelance development. I'm currently trying to decide what framework to move my company in regarding frameworks for web application work. Options are ASP.NET MVC Django CakePHP/Symfony etc.. Struts Pearl on Rails Please feel free to add more to the discussion. I currently work in ASP.NET MVC in my Spare time, and find it incredibly enjoyable to work with. It is my first experince with an MVC framework for the web, so I can't talk on the others. The reason for not pushing this at the company is that I feel that there are not many developers in the Media/Marketing world who would work with this, so it may be hard to extend the team, or at least cost more. I would like to move into learning and pushing Django, partly to learn python, partly to feel a bit cooler (all my geeky friends use Java/Python/c++). Microsoft is the dark side to most company's I work with (Marketing/Media focused). But again I'm worried about developers in this sector. PHP seems like the natural choice, but I'm scared by the sheer amount of possible frameworks, and also that the quality of developer may be lower. I know there are great php developers out there, but how many of them know multiple frameworks? Are they similar enough that anyone decent at php can pick them up? Just put struts in the list as an option, but personally I live with a Java developer, and considering my experience with c#, I'm just not that interested in learning Java (selfish personal geeky reasons) Final option was a joke http://www.bbc.co.uk/blogs/radiolabs/2007/11/perl_on_rails.shtml

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  • Cannot upload media via Wordpress uploader

    - by Justin Johnson
    This has to do with media uploading in Wordpress. Every time WP creates a folder for new uploads (it organizes uploads by year and month: yyyy/mm), it creates it with the "apache:apache' user and group, with full access to all (777 or drwxrwxrwx). However, after that, WP cannot create a folder within that folder (e.g.: mkdir 2011 succeeds, but mkdir 2011/01 fails). Also, uploads cannot be moved into these newly created folders even though the permissions are 777 (rwxrwxrwx). Once a month, I have to chown the newly created folders to be the same as user:group as the rest of the files. Once I do that, uploading works fine (which doesn't make sense to me The really frustrating part is that this problem doesn't exist in other WP installs on other domains on the same server. * I wasn't sure if this should be here or on serverfault. Edit: The containing directory /.../httpdocs/blog/wp-content/uploads has the correct ownership drwxrwxrwx 5 myuser psaserv 4096 Jun 3 18:38 uploads This is a Plesk/CentOS environment hosted by Media Temple (dv). I've written the following test script to simulate the problem <pre><?php $d = "d" . mt_rand(100, 500); var_dump( get_current_user(), $d, mkdir($d), chmod($d, 0777), mkdir("$d/$d"), chmod("$d/$d", 0777), fileowner($d), getmyuid() ); The script always creates the first directory mkdir($d) successfully. On domain A, where the WP problem is, it cannot create the nested directory mkdir("$d/$d"). However, on domain B, both directories are successfully created. I am running each script at /var/www/vhosts/domainA/httpdocs/tmp/t.php and /var/www/vhosts/domainB/httpdocs/tmp/t.php respectively I checked the permissions on tmp, httpdocs, and domain[AB] and they are the same for each path. The only thing that differs is the user.

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