CANVAS SoundJS audio buffer

Similar Questions

• Audio buffer?

  I have a presentation that was initially developed for local playback. It consists of sounds, images, and xml content-driven. The presentation is divided into 5 sections. Now, the client wants to put the presentation on the web and I'm having a little trouble understanding how the audio buffer. What is a good way to check if the audio is buffering? How to put the tampon?

  I did a race at the this time and was not very successful.  For each section, it follows this pattern:

  Load swf. SWF load XML. XML is converted to a slide that are Flash objects. Then the first slide starts by building the slide out of the scene. Then the audio is started, then the animation starts. The problem is that I can't seem to synchronize to the top of the audio with slides on the web.

  Here, I have created a timer to check a variable to see if the audio is buffered? If so, what is the code. IsBuffering code is true until the audio is fully buffered. I am looking for a buffer of 3 seconds and then release my animation to start.

  Thank you

  No, load() is correct.  (loadSound() comes from as2).

  and you're already buffering your sound completely.

  the problem seems so that you leave not sound at the right time.  When can you start the sound completely downloads (or pads).  You must wait until the his downloads, and pictures to download to keep everything in sync.

• Audio buffer PCM of Qt/reading (Playbook)

  Hello

  My Qt application generates PCM audio data I want to read (on a Playbook). I found an example to read a WAV for OS 10 (https://github.com/blackberry/NDK-Samples/blob/master/PlayWav/main.c ) using "snd_pcm_plugin_write" among other functions. These functions are also operate properly with the Playbook OS 2.1?

  In addition, all sounds in this example event management seems quite complex - is there something easier to use only the PCM playback buffers?

  Kind regards

  libalut.so seems to be in the ndk playbook and if not, you should be able to build it yourself, http://blackberry.github.io/ndk/components.html

• Blu - ray error: "file already exists", Code: "6", "Audio buffer overruns. Total too high sampling rate

  I hope someon can help out me here - I'm stuck in my tracks.    I read through many discussions on this topic, but do not understand what is happening here.    CS6.  Note: I used even before I put this material on a disk.  I currently have a separate sequence paired down, with only about 1/5 of the entire production.   1920x1080i, 29.97, H.264 Blu - ray, VBR 2-pass, PCM inside PP to produce .m4v/.wav.  This is what confuses me - I tried a target of 10 and a maximum of 20, and I still get this error.   I tried different bits max rate 'default' settings for transcoding in again.    Note the timecode listed "0.000000" - maybe someone said something.    I don't really understand yet again, so maybe it's something that I've set wrong in still vs my exports?

  

  Good news!  Suggestion of Stan update seems to have solved the problem.  I think it was on the side of the PP, not yet.   I have updated both, but still using to try to make a comic about made export with the PP before update failed.   But once I upgraded to PP, a new rendering, etc. I am able to build a BD.

  Disappointment, before I had installed all updates patches, but later everything uninstalled and ran the vacuum cleaner, try to debug another problem.

• Record buffer audio zero problem

  Hello

  I write a j2me application where I save an audio stream and treat it in real time. I access the audio buffer every two seconds, take the data stored in the byte array and reset the buffer (ByteArrayOutputStream). Even if I reset the buffer combined Blackberry ignors command and reset the buffer data are accumilated (I use JDE 4.5). Can you get it someone please let me know how to fix this problem.

  Thank you

  I tried this on a BlackBerry 8700 running 4.5.0.81 and has not been able to reproduce.  The stream is being reset.

  You could wrap your timer code in a synchronized block synchronized on the ByteArrayOutputStream.  It might be that the record and timer thread is accessing the variable at the same time.

• Audio IOS RTMP live stream

  Hi guys I hope that some of you will be able to help me.

  I'm trying to stream audio from a live RTMP stream using the NetConnection and NetStream classes. I managed to get my app works without problem on Android, but I have some major difficulty make him play the audio back on iPad. Interestingly, it works in the device emulators when you are debugging, but I guess this isn't really an accurate representation. I tried streaming RTMP in AAC and MP3, but without success to be. I check through debugging that there haven't connected to the stream, however I just get no audio playback.

  Everything I read on door to believe that this is possible on IOS that I'm only interested by audio and video not. Can anyone help?

  Code example below (it's quick and dirty! ).

  Thanks in advance!

  <? XML version = "1.0" encoding = "utf-8"? >

  " < = xmlns:fx s:View ' http://ns.Adobe.com/MXML/2009 "

  xmlns:s = "library://ns.adobe.com/flex/spark" title = "Audio" creationComplete = "init ()" >

  < s:layout >

  < s:VerticalLayout paddingLeft = "10" paddingRight = "10".

  paddingTop = "10" paddingBottom = "10" / >

  < / s:layout >

  < fx:Script >

  <! [CDATA]

  import flash.media.Video;

  import flash.net.NetConnection;

  to import flash.net.NetStream;

  Import mx.core.UIComponent;

  private var vid:Video;

  private var videoHolder:UIComponent;

  private var nc:NetConnection;

  private var defaultURL:String = '[STREAM]. "

  private var streamName:String = '[STREAMNAME]. "

  private var ns:NetStream;

  private var msg:Boolean;

  private var intervalMonitorBufferLengthEverySecond:uint;

  private function init (): void

  {

  vid = new Video();

  VID. Width = 864;

  VID. Height = 576;

  VID. Smoothing = true;

  Join the video of the scene

  videoHolder = new UIComponent();

  videoHolder.addChild (vid);

  addEventListener (SecurityErrorEvent.SECURITY_ERROR, onSecurityError);

  grpVideo.addElement (videoHolder);

  Connect();

  }

  public void onSecurityError(e:SecurityError):void

  {

  trace ("error in security :");

  }

  public void connect (): void

  {

  NC = new NetConnection();

  NC.client = this;

  nc.addEventListener (NetStatusEvent.NET_STATUS, netStatusHandler);

  nc.objectEncoding = flash.net.ObjectEncoding.AMF0;

  NC. Connect (defaultURL);

  }

  public void netStatusHandler(e:NetStatusEvent):void

  {

  Switch (e.info.code) {}

  case "NetConnection.Connect.Success":

  trace ("audio - connected successfully");

  createNS();

  break;

  case "NetConnection.Connect.Closed":

  trace ("audio - Connection closed");

  Connect();

  break;

  case "NetConnection.Connect.Failed":

  trace ("audio - connection failed");

  break;

  case "NetConnection.Connect.Rejected":

  trace ("audio - connection rejected");

  break;

  case 'NetConnection.Connect.AppShutdown ':

  trace ("audio - application shutdown");

  break;

  case 'NetConnection.Connect.InvalidApp ':

  trace ("audio - app connection not valid");

  break;

  by default:

  trace ("audio-" + e.info.code + "-" + e.info.description);

  break;

  }

  }

  public function createNS (): void

  {

  trace ("NetStream creation");

  NS = new NetStream (nc);

  NC. Call ("FCSubscribe", null, "live_production"); Do not use this option if your CA requires it

  ns.addEventListener (NetStatusEvent.NET_STATUS, netStreamStatusHandler);

  vid.attachNetStream (ns);

  OnMetaData and onCuePoint //Handle event reminders: solution to http://tinyurl.com/mkadas

  //See another solution to http://www.Adobe.com/devnet/flash/QuickStart/metadata_cue_points/

  var infoClient:Object = new Object();

  infoClient.onMetaData = function MDGs (): void {};

  infoClient.onCuePoint = function oCP (): void {};

  NS.client = clientside;

  ns.bufferTime = 0;

  NS. Play (StreamName);

  ns.addEventListener (AsyncErrorEvent.ASYNC_ERROR, asyncErrorHandler);

  function asyncErrorHandler(event:AsyncErrorEvent):void {}

  trace (Event.Text);

  }

  intervalMonitorBufferLengthEverySecond = setInterval (monPlayback, 1000);

  }

  public void netStreamStatusHandler(e:NetStatusEvent):void

  {

  Switch (e.info.code) {}

  case "NetStream.Buffer.Empty":

  trace ("audio - buffer empty :");

  break;

  case "NetStream.Buffer.Full":

  trace ("audio - buffer full :");

  break;

  case "NetStream.Play.Start":

  trace ("audio - beginning of the game :"); ")

  break;

  by default:

  trace ("audio-" + e.info.code + "-" + e.info.description);

  break;

  }

  }

  public function monPlayback (): void {}

  Print the current length of the buffer

  trace ("audio - buffer length:" + ns.bufferLength);

  trace ("audio - FPS:" + ns.currentFPS);

  trace ("audio - delay Live:" + ns.liveDelay);

  }

  public function onBWDone (): void {}

  Do nothing

  }

  public void onFCSubscribe(info:Object):void {}

  Do nothing. Prevents the error if connection to CDN.

  }

  public void onFCUnsubscribe(info:Object):void {}

  Do nothing. Prevents the error if connection to CDN.

  }

  []] >

  < / fx:Script >

  < s:Group id = "grpVideo" >

  < / s:Group >

  < / s:View >

  Just an update on this for all of you who come along after me.

  I managed to get this working on the MP3 but not AAC (I guess just AAC is not taken in charge?).

  My problem was the buffertime. Docs seemed to indicate that it should be set to 0 for live streaming, however he switch to "1" has solved my problem on the particular flow I pointed at.

  So, essentially above the justchanged line:

  ns.bufferTime = 0;

  TO

  ns.bufferTime = 1;

  Would be nice to know if someone got AAFC works though...

• Satellite L450-188 - how to disable ACPI.sys and update the BIOS?

  Hi, hoping someone can help me with a slight problem, I'll have

  I recently bought a Satellite L450-188 (number PSLY0E-03K019EN) running under Windows 7 64-bit

  I use the laptop with a DJ program called Rane Serato Scratch Live, I got latency usb problems, getting usb dropouts when I try to use the program on the smallest audio buffer position (which I expect to be able to do)

  When I contacted support at Serato team with my problem, they suggested that I run a program to check the time DPC on my laptop, when I presented the report to them, they said ACPI.sys was the origin of the problem and suggested that interest me to turn it off, but I wasn't too comfortable with that.

  The other suggestion was to try to find updates the bios for my laptop, but I'm not too set up with that sort of thing, and I'm worried I might damage the laptop maybe installing the wrong thing.

  Can someone give me advice on this topic, because I don't know what to do

  Thank you
  Jim

  Hi jimbob_1,

  Normally the ACPI means Advanced Configuration and Power Interface and control options for energy savings.
  So if something is wrong with the ACPI.sys file I recommend reinstall Windows using Toshiba recovery disc.
  By the way: are you running pre-installed Windows version or another?

  As Akuma writes that you can download the BIOS update from Toshiba Web site. http://EU.computers.Toshiba-Europe.com/ > support & downloads > download drivers

  Your laptop belongs to the series of PSLY0E, so you can download latest BIOS.

  Before you begin the BIOS update, close all running programs and extract all the files from zip archive. You start the update as an administrator and connect the AC adapter. Also, make sure that the battery is fully charged.

• Why is there the lag when you press the SPACEBAR for playback in edit mode?

  When I press the SPACEBAR to start playback of a file in edit mode, a second of 2-3 delay before playback starts ago, this makes editing very difficult competently. does anyone know of any settings that may resolve this problem or may have another reason for why this might be happening?

  It may happen if the sampling frequency of your hearing file does not match that of your audio interface as hearing should make a conversion of the prior sampling rate it loads the audio buffer.

• SoundJS with Flash HTML5 Canvas - break sound &amp; update the lib?

  I'm looking to do some basic audio control in the Flash timeline, exporting HTML5 canvas.

  On the frame of 10 I am loading/reading a file with:

  audio1 var is createjs. Sound.Play("Sounds/course3_mod1_2.mp3");

  Who plays, but then I can't figure out how to make a pause as the beep several frames later at a briefing of the animation to a timeline. I read that there is a method. pause() methods, as well as chassis 50 that I use:

  AUDIO1.pause ();

  This does not work for me - I must be missing something pretty obvious regarding variables references on the timeline...? In the console of Firefox, I see the JS error:

  ReferenceError: audio1 is undefined.

  If it's important, the process of publication included "0.5.2.min.js", although I do not have the latest version of CA is "0.6.0.min.js". I'm also very curious how I would update my CC Flash (version 20141) to use this latest version of the library compared with older he current attaches to the HTML code. I guess I can (in the publication settings) uncheck the option 'Hosted libraries', and then manually download new libraries and replace them in the folder. But not this crushed with each publication? And what happens if I still want a link to a hosted version the later?

  somascope wrote:

  On the frame of 10 I am loading/reading a file with:

  audio1 var is createjs. Sound.Play("Sounds/course3_mod1_2.mp3");

  Who plays, but then I can't figure out how to make a pause as the beep several frames later at a briefing of the animation to a timeline.

  That cannot possibly work. If it worked, it is no more. The SoundJS, the documentation is very clear that "before you can play a sound, it must be saved.» You need to do something like this:

  // create global alias to main timeline for event handler
  _main = this;
  // wait for sound to preload
  this.stop();
  // assign load event handler
  createjs.Sound.on("fileload", handleLoad);
  // load and register sound
  createjs.Sound.registerSound("mysound.mp3", "soundID");
  // play sound and start timeline playing
  function handleLoad(event) {
      _main.audio1 = createjs.Sound.play("soundID");
      _main.play();
  }
  

  Then, to put in pause/resume, follow these steps:

  this.audio1.paused = true; // pause
  

  this.audio1.paused = false; // resume
  

  Reference: is there a pause() method? · Number #200 · CreateJS/SoundJS · GitHub

  Note that var creates variables that have only the scope of the framework. To the current scope of timeline, you must use this..

  If it's important, the process of publication included "0.5.2.min.js", although I do not have the latest version of CA is "0.6.0.min.js". I'm also very curious how I would update my CC Flash (version 20141) to use this latest version of the library compared with older he current attaches to the HTML code.

  DO NOT do it. Each new version of Flash CC publishes the files that target a specific version of the CreateJS library, so try to cram in a different version will certainly break things. But why you are still using Flash CC 2014? You should be able to upgrade to CC animate.

• Sound / Audio (CRAC, pop, broken playback) the buffer Y460P issues

  Hello

  I have problems playing audio without clicks and pops. They sound like problems of buffer against the recess.

  Clicks occur during audio playback of any program, but worsen under certain cercumstances.

  The worst time is when I have my external monitor plugged. He actually 10 times worse if I have uTorrent with multiple files running. Even after closing, the clicks remain. The best result is obtained only after the closure of the programmes as much as possible.

  The clicks are better, but not completely disappeared, when no other monitor is connected and cover is open. If I close the lid, the clicks well gets worse.

  I have an external USB sound (Sound Blaster X - FI) card and also tried with this. It makes no difference, the clicks are stll there.

  I updated my BIOS with the latest version and sound card drivers are updated. Is there anything else I can try?

  Thanks and let me know if you need more information from System.

  
  

• Setting up an audio I/o buffer continues

  I'm relatively new to Labview and run to 2 problems that I can't solve. If anyone can help, that would be great.

  I am creating a VI using the sound card that receives an audio input of a generator of sinusoidal signal (plugged into the mic port), transforms this signal in frequency and uses the output frequency a sinusoidal signal of a slightly higher frequency

  I started using the example of e/s simultaneous VI and changed to my needs. (see annex VI) I'm using Labview 8.5 on a Pentium 4 3.0 Ghz PC.

  I get 2 errors, is that your measuring device measures a signal Inf and means the Niquist theorem. I have a sample rate of 44100 Hz and 4410 samples but a contribution of only 100 Hz sine wave. I paired the sampling rate and samples at a ratio of 10:1. I'm sure that isn't true. You have any suggestions of best on the design of the buffer?

  Also, I sometimes get an error message about the read.vi of sound entry "a task must be run to perform this operation."

  In this case, I guess I've just been lucky. You should do something, like I did here http://forums.ni.com/ni/board/message?board.id=170&message.id=399671&query.id=277879#M399671

  In this way the moment then you read the sound card is not so important. The buffer will prevent dataloss and error. If you get overrun memory buffer the card its task ID will become invalid if I remember correct. Like you, I struggled with the Labview sound system. But now it works fine. At least the requested Party.  The exit part have I fixed it with a work around. A way that I've discussed here http://forums.ni.com/ni/board/message?board.id=170&message.id=403131&query.id=279026#M403131 and here. You can also use a free software for audio output. It's called waveio. You can find it here http://www.zeitnitz.de/Christian/index.php?sel=waveio. It works on XP but I don't know if it works on Vista. With this software, you can have up to four output buffers, and the clicks will be gone. Try stamps how much you need. I think that 2 should work fine

• Choppy audio when you play a CD in WMP - worse still when I play it with VLC media player, what happens? (as it is not bringing in buffer correctly?)

  Sorry that my title got so long. I tried change in timing of the buffer, not helped - changed back to the default value. I went into the Realtek Audio Manager, impossible to find whatever it is that makes it sound better. I have a dvdplayer cdburner and also a blu - ray player, it does the same thing in one of them. I know not what to try next, any ideas?

  I've updated a graphic card software seems to work ok now. Not sure why that would be important as the audio is not going through it. It could be just the fact I restarted the computer, we must all try that before anything else when in trouble, how many times has that straightened things? More then I can count.

• I can control outside the audio in my canvas / HTML5 project?

  I have a Flash project I export to HTML5 / canvas and there is an audio track in the background. Because he was tall, and causing the Flash loading wrong part, I decided it would be better to place it outside my Flash project using the audio tag. So I have this code:

  "< automatic audio loop playback controls > < source src =" sounds/mySound.mp3"type =" audio/mpeg"> < source src =" sounds/mySound.ogg "type =" audio/ogg"> your browser does not support the audio tag. < / audio >

  < canvas id = "canvas" width = "1200" height = "768" style = "background-color: #000000" > < / canvas >

  Simple. I just want to be able to add a button to my Flash project to control the sound coming out of my audio tag. Is this feasible?

  you don't know.

  assign your audio tag an id attribute (for example, audioID) and you can use js like.

  document.getElementById('audiID').play ();

  document.getElementById("audioID').pause ();

  etc.

  Here is a list of the methods, properties, and events that you can use - reference DOM HTML audio/video

• How to get audio to play on the Ipad using html5 canvas cc flash

  I use cc html5 flash project type. I built my project and tested on the desktop and it works fine. Once test on ipad, it plays the audio. It runs on android devices, but not my ipad 4. I use Mp3 files in my flash project.

  I tried the safari and Chrome on my Ipad played or the audio in my folder.

  Links to my files.

  http://dev.Wisc-Online.com/prototypes/TestJosh/cis3405_html5.html

  http://dev.Wisc-Online.com/prototypes/TestJosh/cis3405_html5.js

  SoundJS v0.6.0 API Documentation: Sound

• Audio quality for the HTML5 Canvas mode problems

  Try out HTML5 Canvas of Flash CC mode 2014 and hit just in some problems of breath with audio export.

  Each audio file, I gave it, Flash has converted into MP3 with a loud click at the end that does not appear in the original.

  HTML canvas view, it seems, there is no option to change the default setting for audio compression. The only way to export MP3s as nothing more than 16 kbps mono is to manually change compression on each file. I forgot something?

  Well, I found out at least what was causing the loud CLICK at the end of the generated MP3s. turns the importer Flash CC for WAV files is a bit dark to distinguish between metadata and audio data. Re-record my WAV files without metadata fixed the problem of rattling.