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  • Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

    - by Ole Jak
    I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

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  • Lagging digital tv over ethernet

    - by Steve
    I have a HD Home Run TV over ethernet device, which connects the aerial to my router, and from there the router connects to my PC over about 15m of 100Mbps ethernet cable. The TV output lags every second. It does not do so for a computer much closer to the router. It is odd to me that the network rate is around 7Mbps on a 100Mbps cable. I am not downloading or streaming anything else on the affected computer. Is this lag caused by the speed of the cable, the length of the cable, or interference on the cable? I am considering swapping the ethernet cable with shielded ethernet cable.

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  • Windows app that can stream video over local net with remote browsing

    - by Nifle
    What I want: An app that runs on a windows 7 computer that can stream movies to the other computers in the house. I want to be able to brows the movies on my "server" from the other computers and start streaming. I'm aware that VLC can do some (all?) of this but it's looks to complicated for my 6year old to use. Ideally I'd like something "Air Video Server" but for mac/win computers with a native client for browsing the movies on the PC running as server.

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  • Content delivery: Alternatives to SHOUTcast

    - by polemon
    I've been using Icecast and SHOUTcast for several years now, to deliver audio and Video content. I wonder what kind of alternatives do I have to those two, especially when streaming video. On the client side, what software can I choose from to stream to those servers live? Making the streams available with Flash would be great, but that's maybe another story. We're still using EdCast, which is kinda dead by now, Shoutcast DSP plugin is not an option, as this tends to crash, etc. Are there any alternatives to that, when livestreaming video? I'm using Liquidsoap for content generation, mixing, etc. It should work with Icecast, Liquidsoap and if possible SHOUTcast.

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  • ffmpeg error while segmenting

    - by Tommy Ng
    I'm using ffmpeg and segmenter on Ubuntu 10.04 to create the transport stream from flv/h264 video files and then segment the ts segments for ipad streaming. Some ts files show an error with segmenter - Output #0, mpegts, to '29': Stream #0.0: Video: 0x0000, yuv420p, 480x360, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: 0x0000, 0 channels, s16 [mpegts @ 0x11f4ac0]sample rate not set Could not write mpegts header to first output file my ffmpeg command for creating the ts file - ffmpeg -i 1.flv -f mpegts -acodec libfaac -ar 48000 -ab 64k -s 480x360 -vcodec libx264 -b 192k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 192k -bufsize 192k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 480:360 -g 30 -async 2 -y 1.ts my segmenter command - segmenter 1.ts 10 1 1.m3u8 path/to/streams/

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  • How to host my own cloud so that videos are viewable via desktop web browser?

    - by jake9115
    I want to host my own cloud storage solution, something like Dropbox but entirely dependent on my own central machine. This way things are more secure if setup correctly, and there are artificial storage limitations or pay-walls. Some thing similar to ownCloud: http://owncloud.org/ There is one important feature I want to have: the ability the stream movies in a web browser from my personal cloud to anywhere in the world. In the past I tried this with a NAS, and I mapped XBMC to the NAS via SFTP, and certain media types could stream in this manner. I've also used things like PLEX. In this case, I am looking for a single solution for personal cloud storage and movie streaming from that cloud into a web browser. Does anyone know if this can be accomplished? Thanks for the suggestions!

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  • How to synchronize tasks on multiple computers

    - by SysGen
    I would like to be able to run similar tasks on several computers that must be precisely synced. More specifically I need 4 laptops to be synced, probably over local network, and I need to use one of them to start a task (play a video) on all of them at the same time (different video files on different laptops). All of them are running Windows. Is there a third-party software or any easier way to do so over LAN without significant delay? I need virtually no response time - if there would be a single video streaming on all laptops then people should not recognize the delay.

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  • What video format(s) should be used to serve Macs, PCs, and Mobile Devices?

    - by Jeffrey Blake
    In 2007, I started a site based on streaming and downloading poker strategy videos. At that point in time, the best solution I came up with for supporting users of Macs and PCs was to provide the videos in both WMV and FLV formats. Later we added an M4V version to support iPhones/iPods. Obviously, things have changed a bit since that time. I would like to revisit our format decision to see if there is anything better that we could offer, preferrably with wider support among all devices (so that we can reduce the number of formats offered, if possible). Is FLV + WMV + M4V the best solution? Is there something else we should consider? What about Android devices?

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  • VLC Caching levels

    - by Svish
    When I open the Preferences of VLC and go to Input & Codecs, I have a setting called Default Caching Level. I can choose between Cusom Lowest latency Low latency Normal High latency Higher latency I'm used to caching being set in seconds or something like that. So, more seconds/higher buffer means less chane of buffer underrun while streaming. What is latency? What does it mean to set it lower or higher? In what cases should I go in what direction? If I'm struggling with buffer underruns, should I set it to lower or higher latency?

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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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  • 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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