Clearing application buffer after a live stream ends

Hi there, I'm sure this has been asked and answered a million times, but searching for "application" "buffer" "cache" "live" and "hls" isn't giving me any answers, so let's get to it...
After I end a live stream (which I do using FMLE 3.2 and FMS 4.5) and I then start a new live stream with the same application name, the HLS stream just loops the last 30 seconds of the old stream. To prevent this, after I end a stream, I have to go to applications\livepkg\streams\_definst_\ and delete the buffered data there. Only then can I start a new stream with the same name as the old stream. Now... that can't be normal, can it? Is there a switch somewhere I can set to tell FMS to cut it out and stop buffering old streams?
Thanks,
Raul

You can use "record" mode rather than default mode of "append". You just need to publish using streamname - livestream%i?adbe-live-event=liveevent&adbe-record-mode=record.
Though "append" should also work - can you let me know what do you see in FMS core logs / application logs. Also are you changing encoding settings when you republish/restart?

Similar Messages

Tues, Oct 16 Don't Miss Keynote Live Stream: Strategy, Announcements, Q&A

With collaboration you can empower your employees to engage in ways that help them evolve from “knowledge workers” into collaborative innovators for your business.
Please join OJ Winge, Senior Vice President and General Manager of the Cisco Collaboration Technology Group, live on Tuesday, October 16, as he shares the latest updates to Cisco's Collaboration strategy with industry analysts, consultants, and Cisco partners at the Cisco Collaboration Summit in Los Angeles, California, and with you here online in our Cisco Collaboration Community. Through this exclusive insider view, you will gain further insight into how Cisco is uniquely positioned to deliver the environments and experiences that help people work better together—in order to put the extraordinary within reach.
As part of that discussion, he will explain how cloud collaboration is integral to Cisco’s Collaboration strategy. As proof points, he’ll reveal exciting new cloud collaboration innovations. Discover how you can get Cisco’s industry-leading collaboration experiences delivered through flexible deployment options that meet your business and IT needs. Be among the first to hear these new announcements and learn how to accelerate collaboration quickly, reliably, and securely across your business.
Cisco Collaboration includes unified communications, telepresence, customer collaboration and collaboration applications. Maybe you are a customer in one of these areas or several? Learn what's next by hearing this Cisco strategy update.
During the live stream we will have Cisco experts online to respond to your questions immediately through the discussion forum. Learn from questions posted by other attendees, or ask your own. After the live stream, a replay will be posted by 11 a.m. Pacific Time on October 16. We encourage you to invite your colleagues to view and to continue to ask questions through the forum.
REGISTER NOW to reserve your virtual seat!
Laura Douglas
Cisco Collaboration Community Manager

Just noticed this posting, the first thing I noticed is where you placed the dialer chat script, should look something like this:
chat-script gsm "" "ATDT*98*1#" TIMEOUT 60 "CONNECT"
interface Cellular0
dialer string gsm
line 3
exec-timeout 0 0
script dialer gsm
(not under console port)

How do you clear the buffer of excess key presses, in Java?

Right now, in the game I am working on, if the user presses a key multiple times the actions will be performed as many times as it registers, even though I disallow key presses during the time that the action is taking place.
For example, a move of 1 space can be carried out with a key press. While the character is moving, key presses are not allowed. However, if you just press the key 5 or 25 times, the character will just keep moving, once he's done with the first one.
What I need to do is clear the buffer of key presses after an action has taken place, so that the excess keys are not registered. I know the commands to do so in C++, but not in Java.

Well, I think you've got what I was saying backwards, or I'm not understanding what you are asking correctly.
it is suppposed to return if isMoving is true, not false. If isMoving is true, it means that the moving animation, and actions are taking place, Which is a process that take about a second, for the player to move one space.
Here's the code where I process the keypresses
private void processKey(KeyEvent e) {
    if (isMoving) {return;}
    int keyCode = e.getKeyCode();
    if (!isPaused && !gameOver) {
        isMoving = true;
      if (keyCode == KeyEvent.VK_UP) {
        player.move(player.NE);
      else if (keyCode == KeyEvent.VK_DOWN) {
        player.move(player.SW);
      else if (keyCode == KeyEvent.VK_LEFT) {
        player.move(player.NW);
      else if (keyCode == KeyEvent.VK_RIGHT) {
        player.move(player.SE);
        isMoving = false;
}And in case I need to clear up what I want to happen, and the results I'm seeing. The player, in this case, moves one space in a direction, with a single press of an arrow key. That functionality is working fine, as intended. However, I don't want you to press the key twice quickly and move two spaces. I don't want you to hold down an arrow key and move continually.
That part of the functionality is what is not working.
As I understand my code, pressing the key once should set isMoving to true and, therefore further presses shouldn't register until isMoving is false again, which shouldn't happen until the player moves have finished.
But, the way that it is working now is that if I press the key 2, 3 or 20 times, as soon as one move is finished another begins. Are the key presses just sitting in the buffer somewhere waiting for isMoving to become false again? And if so, is there a function I can call to clear that buffer after a move is finished?

Problem with recording a live stream

hi,
i implemented an application for recording a live stream. It was working well locally. But now i am connecting with client's communication server. Broadcaster is working well. But the recording portion is not working. It's showing an error "NetStream.Record.NoAccess" . Can anbody help me to avoid this problem? Also now I am not able to connect with local FMS(developer version). When I tried to connect the same application with local FMS, it's showing an error like NetConnection.Connect.InvalidApp and NetConnection.Connect.Closed. Can anybody let me know why this error is showing?
Regards,
Sreelash

I've just played around with the current developer edition of
the flash media 2 server, and had a strange experience.
We are using on our productive server still flash
communication server mx 1.5.3, like i've noticed before, because we
have only a licencse for this version.
If i record a video with our app on flash media 2 server the
problems i've descriped are away...
...BUT...
flash media 2 seems to cut the corrupted part automaticly out
of the record.
I recorded a video which is exactly 30seconds long (made with
a timeout script). On old flashcom the first 7 seconds have these
descriped failures.
If i record the same stream on flash media 2 there are no
failures in the record, but the record is only 23 seconds long....
This is very very bad adobe :(
Or is the problem in my code? If there is anybody who can
help, i would be very happy!

AS3 - 1) Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place; 2) Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true - ActionScript 3 - flash cs3 cs4

Hi folks,
Ronny's here again on forums, having particularly 2 (two) questions/problems to resolve:
1) Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place
2) Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true
attached is the .zip file (as3_Countdown Timer_ver 1.0.1_by Ronny Depp.zip) with all flash source files containing:
a) The FLash Source (file: timer_module.fla) - (FLA flash source file - Flash CS3 Professional, Flash Player 9, actionscript 3.0)
b) com.othenticmedia.utils.dateAndTimeManagement package including 2 .as actionscript 3.0 Class files.
       i) com.othenticmedia.utils.dateAndTimeManagement.DateAndTimeManager Class in the said package. (file: DateAndTimeManager.as)
       ii) com.othenticmedia.utils.dateAndTimeManagement.CountdownTimer Class in the package. (file: CountdownTimer.as)
c) The compiled SWF file version of this Application's blueprint. (file: timer_module.swf).
What i need to confirm is: ........................................................ see the next post of mine. (for Problems need to be Resolved)

Problems to Resolve:
Problem#1) - Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place.
Problem#2) - Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true;
Problem#1 Description:
I need to pinpoint the Logical TimeSync Exception, i am still unable to figure out. That is I'm using a webservice in my Application to Synchronize the Time with the actual ET (eastern time) with accomodation of auto-adjustment for EDT GMT-4 (eastern daylight time) & EST GMT-5 (eastern standard time), times. I am using the zipcode: "10012" to pass it to the Web Service in urlRequest object, to retrieve the Current ET eastern time according to EDT & EST time settings for Manhattan/Brooklyn areas or others within New York, NY 10012.
Currently the Web Service is returning accurate date/time based on local EDT GMT-4 daylight time.
Is there some defined set of dates for EDT & EST times for New York region that I can check for to ensure the correct Dates/Times for Eastern Time in New York area ??? I am using NY zipcodes because i am sure to get correct ET values.
The Major Problem Part: is I need to correct the time by 2 seconds or approx. 2 secs, some millisecs.
When I retrieve the Time Value from WebService, it lags behind for 2 seconds as compared to DateObj i create using computer's local time, on my Windows XP Service Pack 2 with Automatic Updates turned-on. And I'm sure about my Windows will be having latest updates for Time Management already installed. I also added the 2 secs. to the TimeSync(ed) Date to make correction to this Date obj.
I call my custom fucntion addSeconds(dateObj:Date, secs:int) to add 2 seconds to the Date by Converting Seconds to Milliseconds.
Please comb through the as code in files attached and Help Me Out !!!
Problem#2 Description:
Secondly I need to Detect the End of Stream state while using FLVPlayback component, an rtmp:// live Stream from FLASH MEDIA SERVER.
I need to Play a YuMe Post-Roll Ad when Steam Finishes/Ends.
Live Broadcast Stream Event starts every night on Wednesdays & Saturdays exactly at 10:59 PM EDT GMT-4.
Live Events only Streams/Broadcasts the stream for 50secs. exactly. When [playback stopped] it plays a PostRoll Ad and after the CountdownTimer again comes back to life. The Next upcoming Event is calculated & the Countdown begins until Next Event's time/date is reached.
Here is the code on the frame 1 on the MainTimeline: (rest of the params like source, volume, skinAutoHide are Set using Property Inspector for FLVPlayback instance on Stage)
//myStream instance of FLVPlayback is on the Stage
myStream.isLive = true;// Frame 1 Actions in the FLA
myStream.addEventListener(VideoEvent.COMPLETE, onEndOfStream);
myStream.addEventListener(VideoEvent.STATE_CHANGE, onState);
myStream.addEventListener(String(VideoError.NO_CONNECTION), onStreamError);
myStream.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
/*if(myStream.stopped){
     trace("tracy: "+myStream.state);
} else if(myStream.state == VideoState.STOPPED){
     trace("tracy: "+myStream.state);
function onStreamError(event:VideoError) {
     trace(event.code + "\n\t" + event);
function onState(event:VideoEvent) {
     trace(event.state + "\n\t" + event.toString());
function onEndOfStream(event:VideoEvent) {
     trace(event.state + "\n\t" + event.toString());
function netStatusHandler(event:NetStatusEvent):void {
     switch (event.info.code) {
          case "NetConnection.Connect.Success":
               //connectStream();
               break;
          case "NetStream.Play.StreamNotFound":
               trace("Stream not found: "/* + myStream.source*/);
               break;

Emptying the buffer of a receiver of a live stream

Hi,
In my application, a publisher starts a live video and a receiver could recieve it through FMS rtmp. The problem:
1) Once the receiver faces latency, due to slow connection, the latency keeps on increasing from 1 second to 10s, 20s and above. What i feel here is that the reciever's buffer starts accumulating video data and keeps on accumulating it and never flushes it.I am unable to find a property in netStream and Application.xml to empty the buffer of a poor connection reciever.
How can i fix this issue, so that a slow connection receiver's latency remains constant around 2 - 3 seconds. Please help.
Regards,
Sahil.

Hi Nitin,
1) Fms version is 3.5.7.
2) I dont have issues with Audio, Its video that raises latency on slow connections of video receivers. To solve this issue i set the publisher's video quality according to receiver's bandwidth. This helps BUT does not solve the problem.
Example:
Publisher is broadcasting video at 800 kbps. The reciver of this stream has a lower bandwidth of 512 kbps, this difference (800 - 512 = 288 kbps) at receiver produces latency. The extra bit of data ie the 288 kbps is somewhere buffered by the recieiver in its stream and this keeps on accumulating on and on. Thus latency never drops or reduces.
This is why i wish to clear/empty/drop the extra receiver stream data.
Thanks,
Sahil.

Issues while watching content after live streaming operation

Hi,
1)We have performed a live streaming of an event using Azure Media Services , we streamed the entire event
which was of about 5-6 hrs duration . But at the end we came to know that it has streamed only last 1 hour of
entire event.
Can you let me know what might be the issue?
2)We performed a live streaming of an event in 2 parts using azure media services. After 2 hours we stopped the streaming of 1st part and deleted the channel for 1st part and we started streaming of another part by creating another channel . We are unable to
view the 1st part through content url while we are able to view the 2nd part through content url?
Can you let me know what might be the issue?
Thanks,
Pushkar

The fan is speeding up to cool the GPU. The GPU (graphic's chip) heats up because video's and game's work the GPU much harder than normal desktop activities.
When was the last time you checked and cleaned the air intake grill on the bottom of your iMac?
I shutdown (stopping the fans), disconnect the power and vigorously vacuum out the bottom grill work on my Mac's a couple of times a year to keep them running cool.

Problem with live streaming video after system updates

Dear all,
I hope you can assist me on this.
Yesterday, I downloaded the following automatic updates: iLife Support 9.0.4, Remote Desktop Client update 3.4, Mac OS X Update Combined 10.6.7, Airport Utility 5.5.2.
After that, I could NO MORE watch live streaming channels. To be precise, l can properly hear the sound, but not see the video.
I wish to underline the fact that before installing the automatic updates listed above, I had a clear sound and video of all live streaming sites.
I downloaded the latest versions of JAVA and Flash, but unfortunately they didn't solve the problem.
Therefore, I am quite sure the live streaming disfunction has to do with the system automatic updates. Is there any way to uninstall yesterday's updates without compromising the various softwares and OS performance?
What else could be if not related to the system updates?
Looking forward to hearing from you. Thanks in advance.

Stan,
I hadn't heard of gspot before. I downloaded it and it looks like a nice, handy tool. Thanks.
After running it, I didn't see anything that caught my eye as unusual (corrupt file, unknown codec, etc.) I have posted a screenshot at the following URL if you care to look.
www.cbvideo.com/gspot/gspot-out.jpg
Now, let me add to the mystery. I was just able to burn it without incident with Encore 1.5 and the same chapter mark locations. There are only two things I can think of that would be possible causes. 1) something quirky about CS3, even though I've not had any prior problems in two years with it for my business OR 2) the file had become corrupted when moving from the capture computer (the older computer with 1.5) to the newer computer (with CS3). As the file seems to be okay on the 2nd computer (plays fine in WMP and imports into both Premiere and Encore), I'm leaning toward #1.
I have been able to successfully work around the problem to finish this job for the client, but I'd still like to know what's going on for future reference. If anyone has other ideas, please let me know.
Robert

Live stream - buffer and replay

Hi,
i have this issue, which i'm pretty sure isn't feasible, but my boss insists that i give it a try, so here we go:
i've developed a video player being used by a commercial TV channel, both for its VOD and live stream. all the video content - live & VOD - is supplied via external CDN, over which i have little (or even no) control.
what we're trying to achieve now, is while the channel's live stream goes on commercials break, to display our own (website's) ads. we can tell when the break begins and when it ends, but we cant' know in advance how long it'll last.
so our plan is to start running our own ads as the break begins, and when it ends - in case the ad is still running, buffer and 'record' the live stream, so when the ad ends the player will display the live stream, but from when its commercials break ended, with a short delay.
is such a thing even possible? i mean displaying live stream with a deliberate delay?
thanks in advance for any thought or idea,
eRez

Hi,
One way you can do this is to use php script to identify the user agent of the client (http://php.about.com/od/learnphp/p/http_user_agent.htm) and then generate dynamic webpages based on the client (HLS for iOS based clients and HDS for flash based clients). As Apache can only serve static webpages, you'll need to use CQ/MySQL/something else. I've personally not attempted to do so, but this may be a good starting point : http://www.ibm.com/developerworks/linux/tutorials/l-php/
Hope this helps.
Thanks,
Apurva

Live stream buffer shrinking

We're encountering a problem with live audio streams going
through FMS. We play back the streams with a generous buffer on the
client to smooth out any hiccups in the end-user's connection. But
over long periods of time (10-30 minutes) the amount of data in the
listener's buffer (monitored with NetStream.bufferLength) gradually
shrinks until it hits 0, pauses to buffer again, and refills. This
occurs on several different internet connections, and there appears
to be plenty of bandwidth available to refill the buffer even if
momentary network problems occur.
Is there anything we can configure in FMS or the client
players to prevent this? Thanks in advance for any
suggestions.

No problems here viewing CNN live streams. According to CNN FAQ, you need Flip4Mac to view their live streams/videos.
"You may watch live video using the Flip4Mac plugin and Quicktime 7.1.2+"
http://www.cnn.com/help/live.html

Detecting END or Stream turned off in OSMF -- Live Stream

Hi all
i am using the OSMF player to stream live events. When the stream ends or if the feed is turned off the player throws a BUFFER state change which my code catches and shows a loading spinner, but how do i detect if the stream has ended or turned off while the user is watching? Also will be helpfull if I want to post roll some ad or something. I could mickey mouse it i guess if it throws BUFFER EMPTY and set a timer but there has to be an easier way to do this.
anybody?
thanks
Corey

it appears Akamais test player counts the bytes loaded when the buffer empties and sets a timer (what I thought! hey, im smart!) and then checks again and if the bytes are different it dispatches an event which backs the feed back to the start (or stops completely if FORCE LIVE is set)
now, how to know if the feed is live or memorex (ie- archived)
anybody? beuhler? beuhler?

FMS server buffer delay 5-6 sec live stream ?

FMS server buffer delay 5-6 sec live stream ?

To reduce the delay you can try disabling aggregate messages.
You need to set Streams/EnableAggMsgs in Server.xml and Client/AggregateMessages in Application.xml to false.
See more about Aggregate messages here : http://help.adobe.com/en_US/flashmediaserver/configadmin/WS5b3ccc516d4fbf351e63e3d119f2925 e64-7ff0.html#WS5b3ccc516d4fbf351e63e3d119f2925e64-7feb
Hope this helps.
Thanks,
Apurva

After live stream is completed. Steps to make it available as VOD content?

I'll be streaming different events about every 2hrs in a single night. After each live event is completed, what steps need to be done to make each finished live event available as VOD? Do I need to package all the individual segment file into a single .flv file?
Thanks!
Dave

Apoorva Arora wrote:
Also, if your requirement is just about making content available after live streams are over, you can look for DVR functionality in your manifest.xml and set the values of begin and end offsets as 0 and 0.
So my manifest file would like this:
livepkgr/_definist_/liveevent/liveevent.f4m
<manifest xmlns="http://ns.adobe.com/f4m/2.0">
<baseURL>http://myserver/hds-live/livepkgr/_definst_/liveevent/</baseURL>
<dvrInfo beginOffset="0" endOffset="0"/>
<media href="liveevent1.f4m" bitrate="282"/>
<media href="liveevent2.f4m" bitrate="482"/>
<media href="liveevent3.f4m" bitrate="832"/>
</manifest>
Or do I still need to have and use this file:
livepkgr/_definist_/liveevent/Manifest.xml
<manifest xmlns="http://ns.adobe.com/f4m/1.0">
<dvrInfo beginOffset="0" endOffset="0"/>
<media streamId="liveevent1" bitrate="282" />
<media streamId="liveevent2" bitrate="482" />
<media streamId="liveevent3" bitrate="832" />
</manifest>
For a MBR live event should my 'livepkgr/_definist_/liveevent/' directory contain the following files
liveevent.f4m, liveevent.m3u8, Event.xml and Manifest.xml
OR JUST
liveevent.f4m, liveevent.m3u8, Event.xml
Thanks again for all the assistance in getting started!
Dave

Live stream lag after unpublish and publish

I am sending a live stream over the web. I'm testing its performance and reliability, and I've noticed that after unpublishing and publishing the stream again, the video becomes very laggy - about 1-2 fps. It happens when the player has buffer bigger than 0. When buffer is set to 0, everything is ok, and after a short break (the moment when stream is unpublished), video is being played normally, like before the break (fps ~25). What is causing the problem, and how can I resolve it?

Are you using Adobe Flash Media Live Encoder? or some other encoder... testing on your machine or remote server?
You might also check over at that Forum:
http://forums.adobe.com/community/flash/flash_media_live_encoder
Most likely you will be asked for the encoding log. So first review the log and see if anything catches your eye.
Also we'll need details like type of machine you are using, CPU and RAM etc.
What are the encoding settinging? display dimensions, bitrate, encoding format, audio and video?... etc... all the config settings.
What type of cam, what are you using for a server... FMS, Wowza, Red5? or streaming to UStream or some service like that?
Best of luck,
Adninjastrator

Live streaming stops automatically after 15 minutes

I am using flash media server 3 and flash media encoder 2.5.
From media encoder I am sending data to media server and then in
browser I am running flex application (version 3) to view the live
stream. I am not saving it in flv file. After 15 mintues live
streaming stops automatically. I have tested all events related to
it..but none of them executes..Flex3 application is just working as
a client.
Portion of code:
nc.connect ("rtmp://localhost/myLiveApp/instance1/");
vid.attachNetStream (ns);
ns.play("flv:mylivestream");
case "NetStream.Play.StreamNotFound" :
/*The FLV passed to the play() method can't be found. */
mx.controls.Alert.show("stream not found");
break;
code in main.asc file :
application.onAppStart = function()
// generate random payload used in bw detection
this.payload = new Array();
for (var i=0; i<1200; i++){
this.payload
= Math.random(); //16K approx
application.onConnect = function(p_client, p_autoSenseBW)
//Add security here
//p_client.writeAccess = ""; // prevents creating shared
object or live streams.
this.acceptConnection(p_client);
p_client.audioSampleAccess = "/";
p_client.videoSampleAccess = "/";
if (p_autoSenseBW)
this.calculateClientBw(p_client);
else
p_client.call("onBWDone");
Client.prototype.getStreamLength = function(p_streamName) {
return Stream.length(p_streamName);
Client.prototype.checkBandwidth = function() {
application.calculateClientBw(this);
application.calculateClientBw = function(p_client)
var res = new Object();
res.latency = 0;
res.cumLatency = 1;
res.bwTime = 0;
res.count = 0;
res.sent = 0;
res.client = p_client;
var stats = p_client.getStats();
var now = (new Date()).getTime()/1;
res.pakSent = new Array();
res.pakRecv = new Array();
res.beginningValues = {b_down:stats.bytes_out,
b_up:stats.bytes_in, time:now};
res.onResult = function(p_val) {
var now = (new Date()).getTime()/1;
this.pakRecv[this.count] = now;
trace("** onResult this.count = " + this.count + " :
this.sent " + this.sent);
//trace( "Packet interval = " + (this.pakRecv[this.count] -
this.pakSent[this.count])*1 );
this.count++;
var timePassed = (now - this.beginningValues.time);
if (this.count == 1) {
this.latency = Math.min(timePassed, 800);
this.latency = Math.max(this.latency, 10);
//trace("count = " + this.count + ", sent = " + this.sent +
", timePassed = " + timePassed);
// If we have a hi-speed network with low latency send more
to determine
// better bandwidth numbers, send no more than 6 packets
if ( this.count == 2 && (timePassed<2000))
this.pakSent[res.sent++] = now;
this.cumLatency++;
this.client.call("onBWCheck", res, application.payload);
else if ( this.sent == this.count )
// See if we need to normalize latency
if ( this.latency >= 100 )
{ // make sure we detect sattelite and modem correctly
if ( this.pakRecv[1] - this.pakRecv[0] > 1000 )
this.latency = 100;
// Got back responses for all the packets compute the
bandwidth.
var stats = this.client.getStats();
var deltaDown = (stats.bytes_out -
this.beginningValues.b_down)*8/1000;
var deltaTime = ((now - this.beginningValues.time) -
(this.latency * this.cumLatency) )/1000;
if ( deltaTime <= 0 )
deltaTime = (now - this.beginningValues.time)/1000;
var kbitDown = Math.round(deltaDown/deltaTime);
trace("onBWDone: kbitDown = " + kbitDown + ", deltaDown= " +
deltaDown + ", deltaTime = " + deltaTime + ", latency = " +
this.latency + "KBytes " + (stats.bytes_out -
this.beginningValues.b_down)/1024) ;
this.client.call("onBWDone", null, kbitDown, deltaDown,
deltaTime, this.latency );
res.pakSent[res.sent++] = now;
p_client.call("onBWCheck", res, "");
res.pakSent[res.sent++] = now;
p_client.call("onBWCheck", res, application.payload);
What is the problem? From media server console I can see 2
connection for 15 minutes and after that it drops to 1.
I have tested it with medium and low bandwith.
Any help is appreciated.

Please read this whole message before doing anything.
This procedure is a diagnostic test. It’s unlikely to solve your problem. Don’t be disappointed when you find that nothing has changed after you complete it.
The purpose of the test is to determine whether the problem is caused by third-party software that loads automatically at startup or login, by a peripheral device, by a font conflict, or by corruption of the file system or of certain system caches.
Disconnect all wired peripherals except those needed for the test, and remove all aftermarket expansion cards, if applicable. Start up in safe mode and log in to the account with the problem. You must hold down the shift key twice: once when you turn on the computer, and again when you log in.
Note: If FileVault is enabled in OS X 10.9 or earlier, or if a firmware password is set, or if the startup volume is a software RAID, you can’t do this. Ask for further instructions.
Safe mode is much slower to start up and run than normal, with limited graphics performance, and some things won’t work at all, including sound output and Wi-Fi on certain models. The next normal startup may also be somewhat slow.
The login screen appears even if you usually login automatically. You must know your login password in order to log in. If you’ve forgotten the password, you will need to reset it before you begin.
Test while in safe mode. Same problem?
After testing, restart as usual (not in safe mode) and verify that you still have the problem. Post the results of the test.

Clearing application buffer after a live stream ends

Similar Messages

Maybe you are looking for