Media Server :: Switch Audio Track In Real Time?
Jul 13, 2011how to switch audio track in real time? For example, show the audio tracks and choose one of them.
View 18 Replieshow to switch audio track in real time? For example, show the audio tracks and choose one of them.
View 18 RepliesHow does the url of FMS real time video stream look like?
View 1 Replieswhy dynamic streaming taking too much time to switch video from lower bit rate to higher bit rate and vice versa. I am doing dynamic streaming in following ways -
var param:NetStreamPlayOptions = new NetStreamPlayOptions();
param.oldStreamName=oldStream();
param.streamName=newStream()
[code]....
I am using duel buffering and that is 3 seconds when video starts and 10 seconds when "NetStream.Buffer.Full". Video taking approximately 30-50 seconds to switch video and when I am calling the above code.
Does flash provide an api to deal with remote stream like IP camera does?
View 2 RepliesI've build a live video/audio chat application. All works fine only the issue is latency of 5 secs. I'm using FMS 3.5 and FLEX.
View 1 RepliesI'm trying to playback some sample data through the new real-time audio capabilities of Flash Player 10. I started with the example given at the bottom of this page on livedocs, which seems to work fine and plays a crystal clear tone.I assume that the two writeFloat's in the example write to the left and right audio channels respectively and that the data being written is 32 bit (because of the float).
However. I seem to be having trouble converting my old 8 bit audio data to a format that is understood by this interface. When I playback my sample data I can vaguely hear the sound I'm expecting but it is massively distorted. My sample data consists of raw 8 bit samples that ranges from 0..255 where 127 is silence.
I've been trying different conversion formulas but I seem to be missing some vital information regarding this conversion.
UPDATE:The correct formula turns out to be:
f = (sample.data.readByte() - 127) / 255
I'm streaming audio using NetConnection and NetStream. I know that you can modify sample data in real-time with the Sound object, however I cannot find the SampleDataEvent for audio playing with the NetStream object. Is there a way to pass the audio from the NetStream object to a Sound object and modify the sound at that object instead?
View 2 RepliesI'm trying to make a software which sends video and audio data to a flash media server by using RTMP protocol. Currently, my program can communicate with a flash media server correctly. RTMP specifications does not describe about the raw data in video/audio messages, so I muxed raw H.264 and AAC data into video/audio messages and sent to the server. The server seems to accept them, but a video player cannot playback the stream sending from the server. The player just says "Loading..." For a test purpose, I sniffed the network packets between Wirecast and the flash media server and ripped off only video and audio data. Then, I muxed those data into video/audio message and sent to the flash media server. In this case, the video player connected to the server can playback the stream correctly.
I checked the stream sent from Wirecast, the stream seems not to be H.264 raw data because those data are not started from 0x17 instead of H.264 start code. With those situation, I am wondering what kind of container format I should use for H.264/AAC data to the flash media server.
I now know that it is impossible to save a swf over a virtual server (in real time) which has had its text changed dynamically. But does this apply to webcam video? What I mean is, if I allow someone to use my app on the internet, and this app allows them to record a video using their webcam, can they then save this swf to their computer?
View 1 RepliesI try to switch or change a server-side stream, it starts lagging after 2 seconds of playing and sound disappears. Here are scenarios that result in that terrible lag:
1. I create server-side playlist with stream.play() with reset=false; when it is time to play the next movie in the playlist, it starts lagging after 2 seconds.
2. The same problems appears when I just switch streams. I installed FMF Feature Explorer and tried to launch SwitchStreams sample application: the same problem - server stream starts lagging after I switch streams with stream.play().
I tried on different servers (local and remote), with different players (debug player of FMS Admin Console, Standard Flash videoplayer component, OSMF player, Flex video player). I also tried all possible flv, f4v and mp4 file compression options for video files - still the same problem. I have also tried literally thousands of Application.xml settings: changing buffer, buffer ration etc. Is there any tip where I should search for a solution?
I ran a simple live video streaming application for the first time with actual users and ran into a couple of serious performance issues that had not turned up during testing. In this instance there was one video stream from a live web cam and used FMLE at 150 kbps using VP6 and MP3 @22k. There were 16 clients and everything worked pretty good for about 30 minutes. (although some clients said their audio and video were out of sync by up to 3 seconds)
Then individual clients would have either the video freeze or the video would continue and the audio would stop. These clints had to "disconnect" and then "connect" again to the application. This happened to all of the clients at one time or another for several minutes. I stopped and restarted the FMLE with progessively lower bandwidth settings down to 75 kbps but still clients were having the same issue.
I eventually stopped the FMLE and used the applications built in publisher at 45 kbps and that seemed to eliminate the freeze/dropping issue. But of course the video quality was very poor and some clients still reported that the audio was out of sync with the video. The server hosting the FMS application is a quad processor dell with lots of memory and network connectivity. The Flash Media Admin Console performance graph showed the total Bandwidth as 3 Mbps at maximum.
Is there a way to stream and audio line level feed rather than the audio from a computer's microphone?
View 1 RepliesAs titled, what is the way to record video/audio files using Flash Meida Server through rmtp, and allow users to access the recorded files through http?What I am trying to do, is to record a user's microphone's input and save it to the server.fterwards, I would like other users to be able to access the recorded files and mainuplating the audio data, by computeSpectrum(), to do some visualization of the audio. As I know computeSpectrum() cannot work on streaming files, so I think I need to access the recorded files using http instead of rmtp. Is that true?
View 1 RepliesI am developing Video Chat over Ip (including audio and text too). But I am unable to get the proper startup material for my desktop application.
View 2 Repliesim looking for a script that can record audio from a mic and save it to a fms server i came across a video email script but would like to have just a audio one that can do the samething as the video email with out the video part.
View 2 RepliesI have a single audio file, I'd like to avoid cutting it up.
I know I can use the sound class, mySound.play(150), to start at 150ms but haven't come across a way to stop the audio say after 500ms or at 650ms.
I'm testing Live streaming with FMSS. The stream is pushed to FMSS buy Adobe FMLE software. Streaming works fine until I reach 1300-1400 simultaneous connections.No matter what the encoding ratebit is (150kbps or 2000kbps) the stream is no longer play from time to time (5-8 seconds).
The CPU (2xIntel Quad) is loaded less than 20% and memory used is about 2 GB (there is plenty of memory installed 32G). The OS is RedHat 5.3 64bit platform. Network uplink maximum rate is 4Gbps.I disabled the Queue but the problem still persist.
Hi,I'm trying to use NetStreamPlayTransitions.SWITCH to create a multi angle view that switches between video streams. The issue I'm having is that NetStream.Play.TransitionComplete is called only after the buffer for the video before it is used up(this makes sense when using SWITCH to go between bandwidths but that's not what i'm using it for). Is there a way to force this switch before the buffer of the previous video is used up?
I've looked into SWAP but I can't really find any documentation on it. What I ideally would have happen is the next video in the array is triggered, that video is buffered and when there is enough to play it the stream switches to that one. SWITCH works really nice because there is no jump in switching when it's played but I just don't want the buffer to play out before the switch.Is there a way of maybe clearing the buffer of the playing video before i call SWITCH so it transitions quickly?
My clients use Flash Media Live Encoder to publish the audio+video to my server. In some case, I want to allow my clients to publish only Audio and no video but I cannot control it via the FMLE, they can even select the video to publish.
I wanted to know if there is any way in which on the server side, I can allow my client to publish only Audio and no Video?
I want to create dj mixer aplication (AS2). Something like this: ttp://activeden.net/item/dj-mixer/10276How I can record sound?
View 1 RepliesI have an application that uses FMS 3.5 to stream live audio between 2 computers.Currently, the delay between when the person speaks and the time the other person hears it is about 2 - 3 seconds.This makes it difficult to have a conversation because one person regularly steps on the other.I am using a NetStream object that connects to the fms server.I then attach audio.What configurations can be set so that the least amount of delay is experienced between the two computers? Can fms ever relay audio as quick as something like Google Talk?
View 4 RepliesIs it possible to mix audio streams in fms (like fmg)?
View 3 RepliesI'm trying to setup an audio-only, on-demand HLS stream in FMS 3.5. I have no problems streaming the sample f4v files via HLS, nor do I have any issues streaming the mp3 files via RTMP to a Flash client. However, when I try to stream a sample mp3 via HLS (the mp3 file is located in the same directory as the sample f4v's), I get a 404 error. I can't find anything in the documentation about streaming audio via HLS on-demand.
View 1 RepliesHow can we record audio in Flash Media server as MP3 format?
View 1 RepliesI want to set the bit rate of audio and video (48,96.128 kbps) at the time of live streaming . i m not using the flash media live encoder because i have to make it as web applicaton. can any one tell me how to set bit rate.
View 1 RepliesOur company has a Flex application (3.0.2) that is having a problem with audio dropping out in Windows 7. The problem appears worse with a wireless connection, but it can't be consistently reproduced there, either. The only thing known for sure is that the problem has not been seen on any other version of Windows.
The application thinks the audio is playing, because it is calling the appropriate event listeners triggered when the audio completes and when any audio cue points are hit. The majority of our audio is streaming via Flash Media Server (3,0,2,201), but the problem was also seen using embedded sound effects. In one instance, several clicks of a button triggered an error saying the sound effect could not be loaded, and then about a minute later, all the sound effects played at once and the user could then continue with the application.
I can't find the setttings for have a least latency as possible, either the video quality is bad or the latency is up than 5 secondes...What should i do?
[Code]...
We have setup a brand new FMS 4.5 for http streaming to ios devices. We are feeding the source via rtmp to livepkgr application. Upon digesting the iOS feed via the url below, there is no audio being passed. I can check the stream using the rtmp protocol and there is audio. [URL]..
View 5 Replies"To serve streams over a cellular network, one of the streams must be audio-only. For more information, see HTTP Live Streaming Overview.To publish an audio-only stream, enter the following in the Flash Media Encoder Stream field:livestream%i?adbe-live-event=liveevent&adbe-audio-stream-name=livestream1_audio_only&adbe-audio-stream-src=livestream1If the encoder specifies individual query strings for each stream, use individual stream names instead of the variable %i:livestream1?adbe-live-event=liveevent&adbe-audio-stream-name=livestream1_audio_onlylivestream2?adbe-live-event=liveevent&adbe-audio-stream-name=livestream2_audio_onlyTo generate a set-level variant playlist when using an audio-only stream, specify the audio codec of the audio-nly stream. Specify the audio and the video codec of the streams that contain audio and video.For more information about using the Set-level F4M/M3U8 File Generator, see Publish and play live multi-bitrate streams over HTTP.
View 7 RepliesIf I publish using an Adobe Flash Media Encoder 2.5 or a normal client a netstream with codec NellyMoser at 8Khz, the audio stream is incomprehensible for a MAC OS or a Linux Adobe Player.Steps to reproduce:
1. Create a new Actionscript Project in Flex Builder, for example: AdobeBug.
2. Start a Flash Media Server 3.5 in localhost(rtmp://localhost/live).
3. Insert the following code in the Default Application file: AdobeBug.[code]
4. Compile and play the file. 5. Run the Adobe Flash Media Administration Console and play the stream named livestream using a Linux or a Mac OS system, but not a Windows system. This stream should be a Nellymoser audio at 8KHz.
Playback starts, but audio is unintelligible in Linux and Mac OS Adobe Flash player.Doesn't happen if I listen the stream using a Adobe Flash Player plugin or a Flash Media Administration Console for Windows systems.