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Search found 81 results on 4 pages for 'rtmp'.

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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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  • iptables issue on plesk

    - by Fred Rufin
    i don't know how to open a specific port (rtmp=1935) on my CentOS server using Plesk or itables. I created new rules for port 1935 i/o using Plesk/Modules/Firewall but this doesn't work. Nmap scanning tells me this : 1935/tcp filtered rtmp . So i decided to have look at my iptable using SSH (iptables -L), and iptables seems to contain my rules (tcp spt:macromedia-fcs): Chain INPUT (policy DROP) target prot opt source destination VZ_INPUT all -- anywhere anywhere ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED REJECT tcp -- anywhere anywhere tcp flags:!FIN,SYN,RST,ACK/SYN reject-with tcp-reset DROP all -- anywhere anywhere state INVALID ACCEPT all -- anywhere anywhere Chain FORWARD (policy DROP) target prot opt source destination VZ_FORWARD all -- anywhere anywhere ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED REJECT tcp -- anywhere anywhere tcp flags:!FIN,SYN,RST,ACK/SYN reject-with tcp-reset DROP all -- anywhere anywhere state INVALID ACCEPT all -- anywhere anywhere Chain OUTPUT (policy DROP) target prot opt source destination VZ_OUTPUT all -- anywhere anywhere ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED REJECT tcp -- anywhere anywhere tcp flags:!FIN,SYN,RST,ACK/SYN reject-with tcp-reset DROP all -- anywhere anywhere state INVALID ACCEPT all -- anywhere anywhere Chain VZ_FORWARD (1 references) target prot opt source destination Chain VZ_INPUT (1 references) target prot opt source destination ACCEPT tcp -- anywhere anywhere tcp dpt:http ACCEPT tcp -- anywhere anywhere tcp dpt:ssh ACCEPT tcp -- anywhere anywhere tcp dpt:smtp ACCEPT tcp -- anywhere anywhere tcp dpt:pop3 ACCEPT tcp -- anywhere anywhere tcp dpt:domain ACCEPT udp -- anywhere anywhere udp dpt:domain ACCEPT tcp -- anywhere anywhere tcp dpts:filenet-tms:65535 ACCEPT udp -- anywhere anywhere udp dpts:filenet-tms:65535 ACCEPT tcp -- anywhere anywhere tcp dpt:cddbp-alt ACCEPT tcp -- anywhere anywhere tcp dpt:pcsync-https ACCEPT tcp -- localhost.localdomain localhost.localdomain ACCEPT tcp -- anywhere anywhere tcp dpt:macromedia-fcs ACCEPT udp -- localhost.localdomain localhost.localdomain Chain VZ_OUTPUT (1 references) target prot opt source destination ACCEPT tcp -- anywhere anywhere tcp spt:http ACCEPT tcp -- anywhere anywhere tcp spt:ssh ACCEPT tcp -- anywhere anywhere tcp spt:smtp ACCEPT tcp -- anywhere anywhere tcp spt:pop3 ACCEPT tcp -- anywhere anywhere tcp spt:domain ACCEPT udp -- anywhere anywhere udp spt:domain ACCEPT tcp -- anywhere anywhere ACCEPT udp -- anywhere anywhere ACCEPT tcp -- anywhere anywhere tcp spt:cddbp-alt ACCEPT tcp -- anywhere anywhere tcp spt:pcsync-https ACCEPT tcp -- localhost.localdomain localhost.localdomain ACCEPT tcp -- anywhere anywhere tcp spt:macromedia-fcs ACCEPT udp -- localhost.localdomain localhost.localdomain My rules seems to be OK but there is no connection to 1935 port using a browser. I can connect to this port with SSH (typing "wget myServerIP:1935") but maybe this is because it is an SSH tunelling ? I don't know how to do.

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  • Browser Based Streaming Video/Audio (not progressive download)

    - by Josh
    Hello, I am trying to understand conceptually the best way to deliver real streaming audio and video content. I would want it to be consumed with a web browser, utilizing the least amount of proprietary technology. I wouldn't be serving static files and using progressive download, this would be real audio streams being captured live. How does one broadcast a stream that will be reasonably in sync with the source? What kind of protocol is suitable? Edit: In research I've found that there are a few protocols: RTSP, HTTP Streaming, RTMP, and RTP. HTTP streaming is somewhat unsuitable if you are streaming a live performance/communication of some kind because it relies on TCP (as its HTTP based) and you don't lose packets. In a low bandwidth situation, the client can get significantly behind in playback. ref RTMP is a proprietary technology, requiring flash media server. Crap on that. The reason I looked at flash is because they are extremely flexible as far as user experience goes. SoundManager2 provides an excellent javascript interface for playing media with flash. This is what I would look for in a client application. RTSP/RTP is what Microsoft switched to using, deprecating their MMS protocol. RTSP is the control protocol. Its similar to HTTP with a few distinct difference -- server can also talk to the client, and there are additional commands, like PAUSE. Its also a stateful protocol, which is maintained with a session id. RTP is the protocol for delivering the payload (encoded audio or video). There are a few open sourced projects, one of them being supported by apple here. It seems like this might do what I want it to, and it looks like quite a few players support it. It sounds like it would be suitable for a "live" broadcast from this page here. Thanks, Josh

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  • Mix Audio tracks with offset in SOX

    - by Laramie
    From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong. I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio tracks by a few seconds so that they are synchronized. Each file is one half of a conversation between a student and a teacher. For example teacher.wav might need to begin 3.3 seconds after student.wav. I can only figure out how to mix the files with SOX where both tracks begin at the same time. My best attempt at this point is: ffmpeg -y -i rtmp://server/appName/instance/student.flv -ac 1 student.wav ffmpeg -y -i rtmp://server/appName/instance/teacher.flv -ac 1 teacher.wav sox -m student.wav teacher.wav combined.mp3 splice 3.3 These tools (FFMEG/SoX) were chosen based on my best research, but are not required. Any working solution would allow an ASP.Net service to input the two FMS flvs and create a combined MP3 using open-source or free tools.

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  • get_iplayer _ new problem

    - by Squiggle
    I have get_iplayer installed on a Windows 7 PC and, generally, it has worked without too many problems. Now I find that I cannot download "The Sinking of the Laconia" from the BBC. Both episodes are still available for viewing (although only episode 2 has a live ' download' option). An error message saying 'ERROR: RTMP_ReadPacket, failed to read RTMP packet header' may be relevant? If anyone can help before the programme becomes unavailable, I'd be grateful

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  • Android stream to Wowza

    - by Curtis Kiu
    I feel very confused about Android streaming to wowza. I am doing a video conference using rtmp cross-platform, but Android doesn't eat RTMP. Therefore I need to find another way to do it. Upstreaming I found a new open-source app called spydroid-ipcamera. It is using rtp, sending udp packets to computer, and opens it in vlc using the following sdp v=0 s=Unnamed m=video 5006 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=420016;sprop-parameter-sets=Z0IAFukBQHsg,aM4BDyA=; But it can't work. Then I follow wowza tutorial and stream to it and then play again in VLC. That works! I wrote it in http://code.google.com/p/spydroid-ipcamera/issues/detail?id=2 However when I want to add audio in the packet, it fails to work. I change to code in http://code.google.com/p/spydroid-ipcamera/source/browse/trunk/src/net/mkp/spydroid/CameraStreamer.java mr.setAudioSource(MediaRecorder.AudioSource.MIC); mr.setVideoSource(MediaRecorder.VideoSource.CAMERA); mr.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4); mr.setVideoFrameRate(20); mr.setVideoSize(640, 480); mr.setAudioEncoder(MediaRecorder.AudioEncoder.AAC); mr.setVideoEncoder(MediaRecorder.VideoEncoder.H264); mr.setPreviewDisplay(holder.getSurface()); Then I thought that the problem should be in sdp, but I don't know how to due with sdp. I am streaming H.264/AAC with Mp4 Second I don't understand sdp. So how can I make video conference upstreaming part using this apps. Android ----(UDP Port:5006)----> PC (SDP file) and then Wowza read the SDP file ------> VLC I think in this way the system cannot handle more than 1 client. sdp can only hold 1 port, any idea or actually it wont' work? Also Wowza need to set the stream before we stream it, so does it mean that I should not follow this way to do it? Sorry my English is poor, I hope you guys understand.

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  • Live video streaming: Microsoft or Adobe ?

    - by Kedare
    Hello, I am looking for a Live Video Streaming solution, The clients will be able to watch the video with a plugin (Flash or Silverlight), or a Standalone application (Windows Media Player, FLV, etc). But I can't choose between Microsoft Solution (Windows Media Server (MMS, RTSP) + Silverlight as client) or the Adobe solution (Flash Media Server (RTMP) + Flash/Flex). The streaming is for short duration cast and will not be online 24/24h. I tried both, and I found the cheaper version of FMS don't provide security to prevent users to register as published (You have to write custom module...), the Windows Media Server provide this function. We already have Windows Server licences. (So Windows Media Server will be "Free") What do you recommend ? What is the best between Flash or Silverlight for Live Video Streaming ? Thank you !

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  • Unable to connect to Adobe Connect

    - by ub3rst4r
    I am having troubles trying to connect my colleges Adobe Connect. I have done the test meeting connection and it will say "Unable to connect". I have tried connecting on 3 other computers and it works with flying colors. I am running Norton 360 on my computer and I also tried it on my other laptop thats also running Norton 360 and it works on that laptop. I also checked my hosts file and that is not the problem because I am able to connect to the server (on port 80) but not the Adobe Connect port (port 1935). The only thing in it is "127.0.0.1 localhost" Here are the details from the log that the test created: Player Version: WIN 11,3,300,271 App-Server returned: code:ok, servers=rtmp://connect.bowvalleycollege.ca:1935/_rtmp://localhost:8506/,rtmpt://connect.bowvalleycollege.ca:443/_rtmp://localhost:8506/ ERROR: FMS Server did not return correctly! Here is my specifications: Windows 7 SP1 x64 Norton 360 v6.3 (latest) It won't connect in Firefox v15, Chrome v19, or IE9 All of my computers are connected through the same router (D-Link DIR-625) Any ideas?

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  • An equivalent of IceCast but for Live Video Streaming ?

    - by Kedare
    Hello, I am looking for a solution to Stream live video like that : A camera/webcam/video output ---> Stream server ---> Clients And if possible multiple Stream Servers like this (like IceCast): A camera/webcam/video output --> Master Stream server +---> Slave Stream Server ---> Clients | `--> Clients | `--> Slave Stream Server ---> Clients `--> Clients The clients will be in flash, so I think RTMP should be a good protocol, I've heard of Red5, is it good for that ? Does it scale ? I would like to get statistics (Amount of clients, Bandwidth, etc), is it possible with red5 ? Do you know any other good solution to do that ? (Only free and if possible Open Source) Thank you !

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  • Do you feel bad when you have to learn new things?

    - by tactoth
    New thing is not always cool. I see many people say they are very bored by doing the similar things day after day. For me it's the opposite - I'm always learning something new. During the last one and a harf year, nearly every two months I need to do lots of researches on a totally new topic: RTMP, MP4, SIP, VNC, Smooth streaming, ..., I have to read lots of specifications, download tones of open source projects to understand concepts, and turn them into my runnable code. And it was so bad! My brain has never been very sure and very familiar with anything, and when it's close to be sure and familiar, it'll have to switch to next thing. I kind of envy people who build upper level applications because they can be very focusing, and their knowledge set includes most things their job requires. Everything is quite measurable, direct and straightforward. Have you ever had the similar feeling? I'm thinking of asking my boss to assign me some other piece of work so that I work like moving forward on a broad road instead of figuring out a way in the dark, I think it'll be more relaxing, any suggestion?

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  • fast forward/streaming in html5 video? RTSP?

    - by karpodiem
    right now I've got a few .mp4's hosted on Amazon S3. I know that S3 has support for RTMP, which is useful for streaming Flash. I'd like to accomplish something similar with html5 video; my biggest issue is that I need the ability to seek (fast forward) to a particular part of the video. Right now when I query the video, it loads the entire video before playing, which is a waste of bandwidth/dealbreaker. In what manner could this be implemented? Is this even possible? Looks like RTSP would be a good bet, but I haven't found whether anyone has rolled this out successfully.

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  • Adding a red5 app in a multiuser website

    - by Zakaria
    hi everybody, I have an mvc php website where users can publish their public information: http://www.example.com/foobar/profile. Beside this project, based on some red5 samples, I have an application (done with Flex) that sends audio: rtmp://server/sendAudio (very basic but works). I want to create for each subscribed on my website an admin part where can send an audio stream: http://admin.example.com/foobar. And, when someone goes on their public profile, they can listen to the streamed audio: http://www.example.com/foobar/profile). How can I use my red5/flash app dynamically with my php website so that my users can broadcast their proper canal? Do you have some experience to share ? Thank you, Regards.

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  • AIR for Android, Flash iphone packager, Playbook support for streaming camera and choosing camera

    - by mswallace
    I am wondering if anyone can definitively tell me if Flash / AIR can find all these mobile devices front facing camera and use RTMP to stream the video captured ? I would like to create a video conferencing app for these devices. Of course none of them support testing this in the simulators and I don't have the funds to purchase or access all of them that I would like to test. Wondering if anyone can shed some light on this for me. I have seen some posts where they have done this for android but not sure about support for finding a list of cameras, choosing one and streaming from iphone 4 and playbook. thanks for any help on this.

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  • FMS: Stream.play on server side makes video choppy

    - by Tinelise
    I have an *.flv file on a FMS. When I play it on the client side the video plays just fine, but when I call Stream.play(filename, 0, -1, false) on the server side the video turns out really choppy. I both cases I use NetConnection to connect to an rtmp and NetStream to play the stream, but in one case I connect to a stream and request the server to play my file on that stream. Apparently that doesn't work with files? It works just fine for live streams. I really don't get why this should differ at all. Any suggestions?

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  • Error #2126: NetConnection object must be connected

    - by Shuo
    Hey guys , I want to count the online user,when each client login the system,it's connecting to the server and increase a variable stored in a remote shared object. But when client connecting server,problems arises:Error #2126: NetConnection object must be connected My web layout: Website --- apps --- userLogin Code snippets: rtmpnc = new NetConnection(); rtmpnc.objectEncoding = ObjectEncoding.AMF0; var uri:String = ServerConfig.getChannel("my-rtmp").endpoint + "/userLogin"; rtmpnc.connect("http://202.206.249.193:2367/userLogin"); rtmpnc.addEventListener(NetStatusEvent.NET_STATUS,onNetStatusHandler); The onNetStatusHander is defined as : switch(event.info.code) { case "NetConnection.Connect.Success":onConnSuccess();break; case "NetConnection.Connect.Failed":onConnError();break; } Could anyoue help me out?Much thanks! Best,Shuo

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  • How to use files/streams as source/sink in PulseAudio

    - by Nilesh
    I'm a PulseAudio noob, and I'm not sure if I'm even using the correct terminology. I've seen that PulseAudio can perform echo cancellation, but it needs a source and a sink to filter from, and a new source and sink. I can provide my mic and my audio-out as the source and sink, right? Now, here's my situation: I have two video streams, say, rtmp streams, or consider two flv files, say at any given moment, stream X is the input stream that's coming from another computer's webcam+mic and stream Y is the output stream that I'm sending, (and it's coming from my computer's webcam+mic). Question: Back to the first paragraph - here's the thing, I don't want to use my mic and my audio-out, instead, I want to use these two "input" and "output" streams as my source and sink so to speak (of course, I'll use xuggler maybe, to extract just the audio from X and Y). It may be a strange question, and I have my reasons for doing this strange this - I need to experiment and verify the results to see.

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  • Flash Media Server won't run on RHEL 6.2 EC2 instance - _defaultRoot__edge1 experienced 1 failure

    - by edoloughlin
    I've got a fresh Redhat Enterprise 6.2 64-bit instance on EC2. I've turned off the firewall and have installed an FMS 4.5 dev server. The FMS install failed, complaining about a missing libcap.so until I installed the libcap.i686 package. The following libcap packages are now installed: libcap.i686 2.16-5.5.el6 @rhui-us-east-1-rhel-server-releases libcap.x86_64 2.16-5.5.el6 @koji-override-0/$releasever libcap-ng.x86_64 0.6.4-3.el6_0.1 @koji-override-0/$releasever libpcap.x86_64 14:1.0.0-6.20091201git117cb5.el6 In the logs directory I have admin and master logs (only). The admin logs look ok: #Fields: date time x-pid x-status x-ctx x-comment 2012-02-29 09:24:26 1144 (i)2581173 FMS detected IPv6 protocol stack! - 2012-02-29 09:24:26 1144 (i)2581173 FMS config <NetworkingIPv6 enable=false> - 2012-02-29 09:24:26 1144 (i)2581173 FMS running in IPv4 protocol stack mode! - 2012-02-29 09:24:26 1144 (i)2581173 Host: ip-10-204-143-55 IPv4: 10.204.143.55 - 2012-02-29 09:24:26 1144 (i)2571011 Server starting... - 2012-02-29 09:24:26 1144 (i)2631174 Listener started ( FCSAdminIpcProtocol ) : localhost:11110/v4 - 2012-02-29 09:24:27 1144 (i)2631174 Listener started ( FCSAdminAdaptor ) : 1111/v4 - 2012-02-29 09:24:28 1144 (i)2571111 Server started (./conf/Server.xml). - I can't connect an RTMP client to the FMS. The master logs contain these lines, repeating every 5 seconds: 2012-02-29 10:43:17 1076 (i)2581226 Edge (2790) is no longer active. - 2012-02-29 10:43:17 1076 (w)2581255 Edge (2790) _defaultRoot__edge1 experienced 1 failure[s]! - 2012-02-29 10:43:17 1076 (i)2581224 Edge (2793) started, arguments : -edgeports ":1935,80" -coreports "localhost:19350" -conf "/opt/adobe/fms/conf/Server.xml" -adaptor "_defaultRoot_" -name "_defaultRoot__edge1" -edgename "edge1". -

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  • FFmpeg Video Hosting for Linux and Windows Server

    - by Aditi
    FFmpeg hosting is a special type of web hosting where the host servers have video transcoding software loaded on them, which allows the automatic conversion of videos from one format to another. FFmpeg is a cross-platform solution for recording, converting, transcoding and stream audio and video. It includes libavcodec – the leading audio/video codec library. FFmpeg hosting gets its name from a set of server side programs (modules) called FFmpeg. There are a number of applications or web scripts available, which allow webmasters to create their own video sharing websites. Video hosting typically requires: PHP 4.3 and above (including support of CLI) Mencoder and also Mplayer FFMpeg-PHP MySQL database server LAME MP3 Encoder Libogg + Libvorbis GD Library 2 or higher CGI-BIN There are number of web service providers who provide FFmpeg hosting service. Following is a list of some of the Best FFmpeg hosting providers for both Linux and Windows Server below. Dream Host Dreamhost provides for web based email access, mail filtering, spam filtering, unlimited email ids, vacation autoresponder, python support, full CGI access and many more services. Price: $7.95 View Details Micfo It offers unlimited disk space and bandwidth. Other services include free domain for life and free Website Transfer with many more services. All in all one of the best option to consider. Price: $5 View Details Host Upon HostUpon offers FFMpeg Hosting on all their hosting packages, with readily installed modules to start a Video website or Social Network with Video uploading. These scripts such as Boonex Dolphin / PHPMotion / Social Engine / ABKsoft Scripts / Joomla Video Plugin / Clipshare / ClipBucket / Social Media / Rayzz / Vidi Script work with their ffmpeg. Their FFMPEG hosting plan offers 24/7/365 support with typical response time of 15min or less. Price: $5.95 View Details DownTown Host DownTown Host provides full and exceptional support by live chat and telephone. It has high-power, modern servers and the finest web server technology. It offers free search engine Submission and continuous data backup protection with free email forwarding and site move. There are many more services too. Site5 This ffmpeg service provider offers uptime guarantee, a real time stats on each server and many more attractive services. Price: $4.95 View Details Cirtex Hosting Cirtex Hosting allows to host 7 websites & domains and provides for unlimited storage space and monthly bandwidth. It also offers FTP and email accounts and many more services. Price: $2.49 View Details FLV Hosting FLV hosting supplies RTMP SERVER STREAMING for large size video streaming and server side recording. It is flexible and costs less. They customize to the clients requirements. Price: $9.95 View Details AptHost This hosting service provides for 24x7x365 Premium Support and fully ffmpeg enabled services. Price: $4.95 View Details HostMDS Great Support, Priced Low. It provides for SSH access, CGI, Ruby on Rails, Perl, PHP, MySQL, front page extentions, 24/7 Support, FREE Domain transfer and spam filtering. It offers instant account setup, low latency fast bandwidth & much more! They were formerly known as Vistapages. Price: $4.95 View Details Related posts:Best WordPress Video Themes for a Video Blog Free Web Based Applications 24+ Coda Alternatives for Windows and Linux

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  • FMS NetConnection.Connect.Close happening when starts and even in the middle of video in Flash with

    - by Sunil Kumar
    Hi I have developed a Flash Video player in Flash CS3 with Action Script 2.0 to play video from Adobe Flash Media Server 3.5. To play video from FMS 3.5, first I have to verify my swf file on FMS 3.5 server console so that it can be ensure that RTMP video URL only be play in verified SWF file. Right now I am facing problem of "NetConnection.Connect.Close" when I try to connect my NetConnection Object to FMS 3.5 to stream video from that server. So now I am getting this message "NetConnection.Connect.Close" from FMS 3.5. When this is happening in my office area at the same time when I am checking the the same video url from out side the office (With help of my friends who is in another office) area it is working fine. My friends naver faced even a single issue with NetConnection.Connect.Close. But in my office when I got message NetConnection.Connect.Close, I can play another streaming video very well like mtv.com jaman.com rajshri.com etc. Some time FMS works fine and video starts playing but in the middle of the video same thing happen "NetConnection.Connect.Close" There is no issue of Bandwidth in my office. I do't know why this is happening. Please see the message when I am getting "NetConnection.Connect.Close" message. NetConn == data: NetConn == objectEncoding: 0 NetConn == description: Connection succeeded. NetConn == code: NetConnection.Connect.Success NetConn == level: status NetConn == level: status NetConn == code: NetConnection.Connect.Closed Please help Thanks & regards Sunil Kumar

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  • Flex profiling - what is [enterFrameEvent] doing?

    - by Herms
    I've been tasked with finding (and potentially fixing) some serious performance problems with a Flex application that was delivered to us. The application will consistently take up 50 to 100% of the CPU at times when it is simply idling and shouldn't be doing anything. My first step was to run the profiler that comes with FlexBuilder. I expected to find some method that was taking up most of the time, showing me where the bottleneck was. However, I got something unexpected. The top 4 methods were: [enterFrameEvent] - 84% cumulative, 32% self time [reap] - 20% cumulative and self time [tincan] - 8% cumulative and self time global.isNaN - 4% cumulative and self time All other methods had less than 1% for both cumulative and self time. From what I've found online, the [bracketed methods] are what the profiler lists when it doesn't have an actual Flex method to show. I saw someone claim that [tincan] is the processing of RTMP requests, and I assume [reap] is the garbage collector. Does anyone know what [enterFrameEvent] is actually doing? I assume it's essentially the "main" function for the event loop, so the high cumulative time is expected. But why is the self time so high? What's actually going on? I didn't expect the player internals to be taking up so much time, especially since nothing is actually happening in the app (and there are no UI updates going on). Is there any good way to find dig into what's happening? I know something is going on that shouldn't be (it looks like there must be some kind of busy wait or other runaway loop), but the profiler isn't giving me any results that I was expecting. My next step is going to be to start adding debug trace statements in various places to try and track down what's actually happening, but I feel like there has to be a better way.

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  • OSMF seek with Amazon Cloudfront

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

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