Search Results

Search found 3772 results on 151 pages for 'music streaming'.

Page 36/151 | < Previous Page | 32 33 34 35 36 37 38 39 40 41 42 43  | Next Page >

  • 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?

    Read the article

  • How to copy or sync music on iphone 3gs on ubuntu 14.04?

    - by user276842
    I’m using Ubuntu 14.04 and want to copy music to/on my iPhone 3gs. I already installed the actual mobile device, but neither Rhythmbox nor gmusicbrowser show my phone. Only on Gigolo I can see it, and transfer data, but the music I copy is not working on my phone. On Amarok I get at least this information: termined mount-point path to /tmp/kde-cafer/amarok/imobiledevice_uuid_882ba1378fe4eb76e98ad6582dcad031c2247664 calling `ifuse "-u" "882ba1378fe4eb76e98ad6582dcad031c2247664" "-ofsname=afc://882ba1378fe4eb76e98ad6582dcad031c2247664" "/tmp/kde-cafer/amarok/imobiledevice_uuid_882ba1378fe4eb76e98ad6582dcad031c2247664"` with timeout of 10s command failed to start within timeout Failed to mount iPhone on /tmp/kde-cafer/amarok/imobiledevice_uuid_882ba1378fe4eb76e98ad6582dcad031c2247664 Does anyone have an idea? Thanks

    Read the article

  • 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?

    Read the article

  • Linux - Want To Check For Possible Duplicate Directories (Probably RegEx Needed)

    - by NoLongerHere
    Hi, I have a directory which contains several directories as follows: /Music/ /Music/JoeBlogs-Back_In_Black-1980 /Music/JoeBlogs-Back_In_Black-(Remastered)-2003 /Music/JoeBlogs-Back_In_Black-(ReIssue)-1987 /Music/JoeBlogs-Thunder_Man-1947 I want a script to go through and tell me when there are 'possible' duplicates, in the example above it would pick up the following as possible duplicates from the directory list: /Music/JoeBlogs-Back_In_Black-1980 /Music/JoeBlogs-Back_In_Black-(Remastered)-2003 /Music/JoeBlogs-Back_In_Black-(ReIssue)-1987 1) Is this possible? 2) If so please help!

    Read the article

  • What is the best way to automatically transpose a LilyPond source file into multiple keys?

    - by Michael Steele
    problem I'm using LilyPond to typeset sheet music for a church choir to perform. Depending on who is available on any given week, songs will be played in various keys. We have an amazing pianist who can play anything we throw at her and the guitarists will typically pencil in alternate chords, but I want to make things easier by having beautifully typeset sheet music available in any key we want. So say we're going to sing our ABCs. First I'll take whatever source transcriptions available and enter it into a LilyPond script: melody = \relative c' { c c g g a a g2 f f e e d d c2 } I want the ability to transpose this automatically, so if I want the whole thing in 'G' I wrap the song in a \transpose call like so: melody = \transpose c g \relative c' { c c g g a a g2 f f e e d d c2 } What I really want is to substitute something for the 'g' and generate the output for melody multiple times. Simple LilyPond variables don't seem to work here, and so far I've been unsuccessful in defining a scheme function to do this. What I've resorted to for the moment is taking the above file, call it twinkle.ly and turning it into an M4 script called twinkle.ly.m4, the contents of which look like this: melody = \transpose c _key \relative c' { c c g g a a g2 f f e e d d c2 } I then compile the while thing by executing the following line: > m4 -D _key=g twinkle.ly.m4 > twinkle_g.ly && lilypond twinkle_g.ly I've written a Makefile to do this for me, defining rules for every song I have and every key I'm interested in. question There's got to be a better way of going about this. Given that Lilypond supports embedded scheme, I would prefer to not use a macro preprocessed on it. Has anybody else come up with a solution to this same problem?

    Read the article

  • Synced audio ouput on multiple machines? VLC? hardware solutions?

    - by zimmer62
    I'm wondering if there is any software or hardware solutions to synced audio or audio and video across multiple computers or devices on a network. I've seen Sonos, and it might be a good solution, but it's also a very expensive solution. I'd like to be able to play something with realtime audio output on one PC, but hear it on speakers throughout the house, being it the home theater receiver, or another computer in another room. I saw a solution using the apple iport express, but the latency was unacceptable for anything other than just music. I'd like to avoid running audio wires with baluns to a bunch of amplifiers scattered all over the place when I have cat5 run everywhere. Is anyone familiar with using this kind of process for whole home audio? The latency is a big deal for me, if I've got video attached to the sound (e.g. watching a hockey game)

    Read the article

  • 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..

    Read the article

  • 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

    Read the article

  • 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

    Read the article

  • 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.

    Read the article

  • 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. :)

    Read the article

  • 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.

    Read the article

  • 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.)

    Read the article

  • 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?

    Read the article

  • 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.

    Read the article

  • 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 ...

    Read the article

  • How can I stream audio signals from various devices/computers to my home server?

    - by Breakthrough
    I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

    Read the article

  • Lilypond: Customize bar lines, recursively, automatically?

    - by ananth.p
    I'm working on Carnatic music scores that involve complex time signatures, that will require modified bar lines Pattern for barlines for: 8/4 beats: 1 2 3 4 (dashed bar here) 5, 6 (Dotted Bar) 7, 8 (double bar) Here's one bar of actual score g16( f) d8 ees( ees) d16( c d8) bes16[( d c bes \bar "dashed" a g]) a[( bes c] d[ c d]) \bar ":" g8( f16) ees8( d16 c d) \bar "||" Is there a way to automate these barlines?

    Read the article

  • Software available for singing "lessons" via computer microphone?

    - by drozzy
    Looking for a software to help my friend learn to sing. Can't seem to find anything on googles. Does there exist software (preferably downloadable and from this century!) that records one's voice and then analyzes it to see how "accurate" it was. It would be great if it also had some kind of "lessons" of some sort, and not simply sound recorder that shows waveforms. I can't imagine it would be so hard to implement, and there probably is one out there - I just can't find it. Any recommendations are welcome. Thanks.

    Read the article

  • What is the current state of development for the FLAC codec?

    - by TwentyMiles
    I have a pretty large collection of FLAC files created from my CD collection. I love the FLAC format and the sound quality that you can get from it. Lately, however, I've been trying to write a few tools to manipulate the files and I've been noticing what seems to be a stagnation of the community around the codec. Some of the links on the official FLAC page point to thing that are no longer relevant (7digital, for example, appears to no longer sell FLAC encoded songs). It's pretty hard to find hardware players that support FLAC any more (most noticeably it's not present on lower end players when it used to be, and playback is absent on Android). Programming language tools (Java and .NET libraries) are at best old, and at worst unfinished. What's the current state of FLAC development? Has it been replaced by another codec?

    Read the article

< Previous Page | 32 33 34 35 36 37 38 39 40 41 42 43  | Next Page >