Audio buffer?

Similar Questions

• CANVAS SoundJS audio buffer

  Hello

  I do a canvas HTML5 by animate CC. This painting includes an audio.  I saw only the audio imported sounds / in the project final does not buffer, so I have to wait the full file download audio ok. The result is a long wait until canvas begins.

  
  Is there a way to activate some kind of buffering?

  
  Thanks for your help!

  You say the opposite of what you mean. Phase preload buffer 100% audio from the timeline, but what you want is flow instead.

  And no, you can't. The library of SoundJS used in the documents of the canvas does not support streaming.

• 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.

• 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.

• Crackle on audio/entry track record - no buffer, plug-ins or sound card. AAARGH!

  I have a slight Crackle on phonograms and supervision right of entry and via these forums and other troubleshooting failed diagnose the problem. Seems to be a product where caused by the audio signal, as occurs in response to an audio input (guitar and bass), not a constant or random noise. A faster attack of the audio signal produced more noticeable crackling.

  It happens to all sizes of the I/O buffers

  CPU load does not seem to be a problem - CPU monitor, both within the logic and across the system may display less than 10% (between all hearts) and it happens when dry recording in a project with a single track (audio) without plug-ins.

  Established between Apogee Duet and Scarlett Solo using different cables and which makes no difference.

  Does not occur during playback back already recorded audio - only new records and monitoring live (or recorded on a track and playback) entry

  Removed the unused midi drivers from (a thread somewhere suggested that for a similar problem).

  .. I am now mainly to a loss of the cause. Don't you no creaking when recording an audio file to QuickTime via the loudspeaker internal, next step, I can think of is to install Garageband (but reluctant to pay apple even £3.99 they want to download it to the app store!) and see if crackle present in audio recordings, in order to reduce it to a software problem (fingers crossed) all the material from the laptop. Or get a copy of logic 10.0 and see if the problem persists in this.

  Recently reinstalled logic 10.2.1 after repeated crashes (and abandonment of strength) when manually move the playhead... as well as other minor bugs 10.2.1 seems to have been shipped with, but this problem started a week or two after re - install, no problems so far here.

  As someone has any ideas other than a glance phone to Apple, which I fear would be very useful even when it doesn't seem to be a known issue.

  Thank you

  Sam

  MacBook Pro (retina, early 2013)

  10.10.5 Yosemite

  INTEL CORE I7 TO 2.7 GHZ

  16 GB OF RAM

  Logic Pro X 10.2.1

  Apogee Duet/Scarlett Solo

  Fixed. Questions of the instrument.

• Jump in the middle of an audio track streaming on various sites causes the stuttering and audio glitch

  Hello

  Please refer to this thread for support on the technical support of SoundCloud forums. The thread contains all relevant information, including videos of the screenshots problem and console, made by myself and others. https://www.soundcloudcommunity.com/SoundCloud/topics/skipping-to-the-middle-of-a-track-causes-stuttering-and-glitchy-audio-chrome-Firefox

  In short, when you try to move forward for streaming audio, there is a memory buffer/stuttering problem that prevents a correct reading of the track. This problem is confirmed on SoundCloud and Traxsource and other people in the support thread mentioned Mixcloud and other streaming sites. In other words, it is not the special streaming service fault or another.

  The problem does not occur in Microsoft Edge, Internet Explorer or Chrome. Well, to clarify, when the issue was published on SoundCloud support forums, Chrome was having the same problem. However, they made a version a few days ago thrust, that fixed it.

  Here's my original on the SoundCloud support post (less), which has the relevant details. Note that we have updated both Flash and Firefox recently, so although I have mentioned earlier in my original post, everyone involved has since updated their browsers and Flash.

  From now on, the only solution for the problem is to go back to Firefox 40.0.2. Thanks for any help =)

  Matt

  Original message =.

  Jump in the middle of a track causes the stuttering and audio glitch (Firefox).

  I am running Windows 10 Pro.

  There seems to be a problem of caching that affect reading - I started noticing this problem three or four days ago. Previously, I could jump through my feeds at will, go straight until the middle of the tracks or mixes of listen seconds and determine if I liked the track. Now, I can't do it on Firefox 41.0.1. The problem is simply annoying on songs that are 3 or 4 minutes in length, as I must leave "cache" maybe ten seconds before jumping. Unfortunately, the problem is a game-breaker on mixtures which, for example, are 90 minutes long. If I jump to a point directly in the middle of the combination, nothing happens. The play button alternates between the 'Play' symbol and the symbol of 'Caching' ('... "), and so audio resumes, it is glitchy and consists of different parts of the track being played at the same time.

  I've eliminated my internet connection as the source of the problem by loading on board. On the edge of 20.10240.16384, the problem does not exist. I can scroll down to any mix of 2 hours in my stream, click on the middle of it, and the audio starts right up. No caching, without stuttering or glitching.

  I drove back Flash version 18 to see if it solved the problem in Firefox, but it did not. The problem persisted, so I re-updated to Flash 19.

  Yes, I understand how all versions and updates to work. I meant it as a generation of powder magazine. My bad.

  I consider this problem to be solved. Several different pre-release versions of Firefox have solved the problem; I have personally used both of them that work perfectly. A representative of Bugzilla stated that the patch will be integrated into Firefox 42, which will be released on 3 November.

  https://www.soundcloudcommunity.com/SoundCloud/topics/skipping-to-the-middle-of-a-track-causes-stuttering-and-glitchy-audio-chrome-Firefox?topic-reply-list [Settings] [filter_by] = all & question-answer-list [parameters] [reply_id] = 16269116 #reply_16269116

  I now consider this private matter. Thanks to all of you who have been here =)

• Crackling with Thunderbolt resident Audio interface

  Hello

  I just bought a new Macbook Pro 15 inch retina, installed with El Capitan 10.11.4 and Logic Pro X, with a new interface reside Audio T4 Thunderbolt, and there is a cracking and sound distortion, crossing the input string. The output is very good. Sometimes the crackling can range from 30 minutes without past and other times it is constant. Unplug and plug cable Thunderbolt and change the port, it can do a little better or worse elsewhere. The crackling and distortion has also nothing to do with the entry levels. I tried:

  • Replace all cables (audio, stroke, etc.).
  • Disconnect other USB devices and even the power of the computer agreement. Also unplugged all other cables, including lunch, from the interface.
  • Purchase of an ALL-NEW resident Audio T4 interface, thinking the problem was inside the unit. It wasn't.
  • Update, uninstalling and reinstalling the drivers to interface
  • Turn off the Thunderbolt bridge under 'network '.
  • Make sure bit depth was the same on the interface and macbook

  I read on other musicians recording in trouble with DAWs and interfaces after the upgrade of the El Capitan, but not with this problem. This is my first time to go from a mac to a PC and am quite impressed so far. I'm grateful for any help.

  For me, when crackle happens "at any time", it is a good indication of low buffer I/O.

  Open CMD-comma logic preferences and click the "Audio" tab at the top. Then, make sure that you are on the "Devices" menu Input and output device are your interface. And nothing below your IO buffer size. try increasing this value to 128, 256, 512 and so on. In other words, the buffer is a small part of the memory where the incoming audio stream is buffered and the CPU can perform any processing required on this subject. If the CPU can't get enough data quickly enough, it starts to sizzle. More a buffer gives the CPU more time to perform a treatment. Usually, this is the case for systems with many things, for example, dozens of simultaneous tracks with many plugins active or slow systems. However, this is not the case for your new, shiny things unless you really have dozens of titles and many plugins per track. Or all of a sudden some background process kicks in and use the time processor and disk, such as a Time Machine backup and others.

  Make sure you plugged your audio interface directly to the Mac, not a hub or something. This introduces a latency and the buffer drains faster.

  Give it a try.

• HP laptop: buffering every 10-12 seconds with audio streaming

  My HP laptop is less than 3 months old. I'm not a gamer, but he puts in buffer every 10-12 seconds on streaming audio. (I have not yet tried YouTube)

  My download rate is 2,37 and upload is 1.84.

  OS is Win 10 home-64 - bit... 8 G RAM

  I have not loaded all programs except Chrome.

  I'm not a novice but I'm out of time to fix this problem and need help.

  Thanks in advance...

  My # is N0N98UA #ABA product