Audio buffer PCM of Qt/reading (Playbook)

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.

• How to split audio data in images after reading a file

  Hi im a student of final year done my project in speech recognition using labview. could someone help me with how to split an audio data in image file for further processing.

  You could read wav files in LabVIEW using these VI: http://zone.ni.com/reference/en-XX/help/371361H-01/lvpict/sfilevis/ once you've read, divide the waveform in the picture of your choice.

  Before you return to the report on not enough memory, please read this: http://forums.ni.com/t5/LabVIEW/not-enough-memory-to-plot-wav-file/m-p/754592

  Search results: http://search.ni.com/nisearch/app/main/p/bot/no/ap/global/lang/en/pg/1/ps/30/q/wav%20file/

• 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 problems and mask when reading not on no FULL quality

  Win 10
  I7 6core
  GTX980 4GB - had the same problem with the previous cards + latest drivers are used and have had the problem with older drivers.
  32 GB of Ram
  Body CC2015

  Working with and reading Dragon RED 6 K images on the timeline, the audio track falls inside and out. The meters also stop working when audio is de-energized. It matters little if the audio is the zero song recorded on the RED or if it is external audio synchronized to the top. The issue seems to only happen when the playback quality is not set to FULL quality, what quality is set to FULL rear audio playback perfectly reading Dragon 6 K sequences in real time is a bit beyond my machine if it isn't really a viable option. The question goes as far if I disable the layer with RED film and just play the audio layers so the audio recorded appears to be very well.

  According to the other forum posts, I tried compliant audio to a drive other than the one with the RED film. I also tried to move the audio to another drive as some have suggested that the issue was on the HARD drive that contains the images RED 100% reading. None of these options worked. I can never remember running into this issue Edition and playback of images of RED EPIC 5 K. And did I not the audio problem when working with lower resolution files.

  I also met other issues in trying to work with the Organization of reading other than that of the TOTAL quality. The use of masks on effects GPU accelerated often causes the whole layer to move off the coast of the viewer project or black glitter during playback. When stopped everything of fine. A quick fix is reading quality INTEGRAL, but once again this is not always possible according to the number of applied Visual effects. Also the engine of reading mercury to the software not 'fixed' the issue of the mask, but it's not really useful in practice. Have had the problem of masking for a long time and initially, I thought that it was related with colorista III, so I grew up with the red giant. They were able to reproduce the issue and noticed that it happened with the other native Ppro GPU accelerated effects, meaning that it was a body does not issue a colorista problem. Support red giant apparently passed on the issue to the agency team, but that was back when colorista 3 is out and still no solution.  

  
  Let me know if you need more information.

  See you soon,.
  James

  Update: CC 2015.2

  Masking question can be solved when you right click on the viewer of the program and then click left high quality playback.

  Still no update on the audio issues however.

  Update 2: CC 2015.2

  Audio is de-energized is more if you delete the audio tracks of RED Dragons. Only play .wav files under the Dragon pictures with audio in sync, I no longer get audio drop-outs. This may apply to other codecs as well.

  First step: good audio sync with the zero track.

  Second step: remove the audio tracks of cameras.

  For me, that stops my audio stall and not to come back.

  Update 3: CC 2015.3

  High quality playback is at the origin of files PNG away from their intended position and also no longer works as it did in 2015.2. Not sure about her work at all TBH.

• 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.

• WebPlayer Yahoo 'buffer' permanently and never reads local files

  I installed the webplayer yahoo to play MP3s on the html page. Works very well on the net. Read files, it is continually buffereing and does not play. Operation of the display flash flashes in mode live view DW, the box of css in the right corner. The flash screen moves at half speed. It seems that the webplayer is somehow interfering with the flash and vice versa. Anyone seen this before?

  < p > < a href = "samples/song1 - 1.mp3" > < /a > < /p > < table > < td align = "center" width = "52" song1-1 >

  Very probably a path issue.

  ... Don't have no idea what a "webplayer yahoo to play mp3" or how it works.

  But the action (or action) that you describe looks like the player is not finding the audio file... probably because the path to the mp3 is wrong compared to the player.

  Only guess--since there is no way for us to know what your "working version" of the imbedding code reader mp3 is without a link to the actual Web page...

  but a player online, can almost NEVER to any directory on your local computer...

  Adninjastrator

• "The Audio Service is not working" - read a few other threads but without success

  Hello

  Thanks for any help I receive in advance,

  my sound was working perfectly last night, done a windows update and resolve now them audio custom work and cannot solve the problem with an automatic, the message I get is:

  Problems found: one or more audio device does not work currently not fixed

  Any help would be appreciated

  Hello!!! can you tell me you have a single machine windows7 or mutiple.

  I suggest two things here

  (1) try to copy and merge the audio of an another windows7 home premium or better service

  You can do this by going into the Registry Editor (make sure that the other computer is also a 32-bit system)

  HKLM/system/current control set/services/audiosrv click audiosrv and then on the file menu, select export and save the file to a USB key or drive.

  While copy on your machine and merge. Then restart your system.

  (2) you can try an upgrade on the spot with the windows7 dvd. Where you can repair the windows all without losing what.

  Thank you. :)

• There is none its icon on the desktop, glancing in Panel configuration says "no audio devices available", but computer reads the startup sounds, what I can do to fix this?

  I have a mouse and Logitech wireless keyboard, and the kit supplied with the speakers that connect to a USB port. This good it worked until recently. When I hooked originally the wireless system, the speakers of original equipment that accompanies the computer Sony is working, so I removed pass them the place on my desk. I have genuine Windows XP with service pack 3 installed. I've recently updated the software for Logitech equipment, and I follow with Windows updates too. I have Norton antivirus protection, so I don't think I got a virus or something like that.

  Cordially... This is a Sony Vaio PVC-RS420 computer.

  Any help would be appreciated.

  Dave

  Hi Dave,.
   
  Try the instructions in this article to resolve this problem:
   
  No sound in Windows

• DAQ Hanging Up and surveillance read buffer system

  Hi all

  I'm collecting data using two SCXI hunts with amplification modules 1120 SCXI 1112 thermocouple modules.

  I'm followed by entry 12 and 4 inputs for thermocouple voltages.

  I put the samp, the 2500 s/s e rate, and I let the software understand how many samples for each step.

  Most of the time, the system works well, in a time of 16ms loop.  However, every few seconds, the system will hang up to ~ 500ms between readings.  Any thoughts on what might cause these crashes?

  I wonder if I'm overflow the buffer.  What could cause this type of behavior?  I think that there is a way to monitor the read buffer, but I forgot how do.  Can someone tell me how to do this?

  Thank you very much.

  You can monitor the buffer using the DAQmx read property node and choose the status-> samples available per channel.

• 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.

• Gets the buffer and logical reads

  Hello!
  
  The event buffer count gets include the logical reads?
  
  Mertic buffer is the event of the cancellation stamp reading?
  
  Thank you and best regards,
  Pavel
  
  Published by: Pavel on June 27, 2012 03:08
  
  Published by: Pavel on June 27, 2012 03:35
  
  Published by: Pavel on June 27, 2012 04:13

  Hello

  Gets the buffer = number of times that a block was requested to buffer cache. A buffer always get request in a logical reading. Depending on whether or not a copy of the block is available in the buffer cache, a logical read may or may not involve a physical read. So 'buffer gets' and "logical reads" are essentially synonymous and are often used interchangeably.

  Oracle is not a special "buffer of cancellation. Undo blocks are stored in rollback segments in UNDO tablespace and are managed in the same way data blocks are (they are even protected by redo). If get consistent requires the reading of UNDO tablespace, then the counters statistics will show that there is a more consistent get in your autotrace.

  For more information and examples, see a thread on askTom:

  http://asktom.Oracle.com/pls/Apex/f?p=100:11:0:P11_QUESTION_ID:549546900346542976

  Best regards
  Nikolai

• Difference between waiting occupied buffer and read by the wait of the other session in 10g

  Hello
  
  Can someone let me know the difference between waiting occupied buffer and read by other session wait in 10g
  
  Thank you

  http://download.Oracle.com/docs/CD/B19306_01/server.102/b14237/waitevents003.htm#sthref2988

  
  buffer busy waits

  Wait that a buffer becomes available. This event occurs because a buffer is either be read in the buffer cache by another session (and waiting for this reading end the session) or the buffer is the buffer cache, but in an incompatible mode (in other words, some other session changes the buffer).
  

  http://download.Oracle.com/docs/CD/B19306_01/server.102/b14237/waitevents003.htm#sthref3159
  
  read by another session

  This event occurs when a session requested a buffer that is currently being read into the buffer cache by another session. Prior to version 10.1, expected for this event have been grouped with the other reasons for waiting for the buffers under the event "wait occupied buffer.
  

• e/s audio simultaneous with daqmx

  I am trying to perform the processing with LABVIEW Student 2013 and myDAQ for audio generation and the acquisition of audio in real time. The treatment will be to manipulate data of signal pt by pt. It seems that continuous sampling is correct with a polymorphic set of sample single, double. I get a message error "the application is not able to cope with the acquisition. Increase in the size of buffer... ». Increase in the size of buffer delay only the error message. More important still, nothing is written on the channel.

  Attached is my VI DAQmx connected without any signal processing. How to fix my problem and this allows me to perform

  signal PT by pt processing?

  Note: I also tried to build a wave form vi feeding generation, but he ran short of space memory buffer and close. Simultaneous audio input output example vi as well as build a vi waveform produces a waveform table 1 d leaving wondering how process pt by pt.

  Thank you.

  You have configured your myDAQ to sample 1000 points, and then deliver them to you.  I don't know why you change the size of the buffer inside the loop (and cannot see what you change him), but you continue to read of a single sample (over 1000 in the buffer) before calling another reading.  Terribly inefficient.

  Without (yet) how you process the data, why not focus on just get ?  Let your size of buffer to 1000, read 1 channel N samples to get the whole in buffer.  As you are clearly just beginning to learn how to do things, not be so ambitious - simplify your code to do only input analog and get this to work.  Once he does, see if you can replace the analog buffer of 1000 points to analog output (I think the myDAQ can do in the output buffer...).

  Bob Schor

• ContentDescriptor do not accept the pcm?

  Hi all

  I have the following strange situation: I created player with Datasource to receive the RtpPacket byte array and play with blackberry player. However, to implement this I use SourceStream who must have a few ContentDescriptor:

  public class RecvStream implements /*Runnable,*/ PlayerListener {
      private Player mPlayer;
      private SendStream mSendStream;
  
      private SourceStream mInput= new SourceStream(){
          ContentDescriptor mContentDescriptor=new ContentDescriptor("audio/amr");
           /* (non-Javadoc)
           * @see java.io.InputStream#read(byte[], int, int)
           */
          public int read(byte[] b, int offset, int length) throws IOException {
              int len = 0;
              return len;
          }
  
          public ContentDescriptor getContentDescriptor() {
              return mContentDescriptor;
          }
  
          public long getContentLength() {
              return -1;
          }
  
          public int getSeekType() {
              return SourceStream.SEEKABLE_TO_START;
          }
  
          public int getTransferSize() {
              return 320;
          }
  
          public long seek(long where) throws IOException {
              return where;
              //throw new IOException("not seekable");
          }
  
          public long tell() {
              return 0;
          }
  
          public Control getControl(String controlType) {
              return null;
          }
  
          public Control[] getControls() {
              return null;
          }
      };
  
      public RecvStream(RtpSession session,SendStream sendStream) {
      }
  
      public void stop() {
          if (mPlayer == null) return;//nothing to stop
          mRunning=false;
          try {
              if (mPlayer.getState() == Player.STARTED) {
                  mPlayer.stop();
              }
              if (mPlayer.getState() != Player.CLOSED) {
                  mPlayer.close();
              }
          } catch (MediaException e) {
          }
      }
  
      public void start() {
          mSession.setTimestampClock(new TimestampClock(){
              public int getCurrentTimestamp() {
                  return getCurTs();
              }
          });
          try{
              mPlayer = Manager.createPlayer(new DataSource (null) {
                  SourceStream[] mStream = {mInput};
                  public void connect() throws IOException {
                  }
  
                  public void disconnect() {
                  }
  
                  public String getContentType() {
                      return mInput.getContentDescriptor().getContentType();
                  }
  
                  public SourceStream[] getStreams() {
                      return mStream;
                  }
  
                  public void start() throws IOException {
                  }
  
                  public void stop() throws IOException {
                  }
  
                  public Control getControl(String controlType) {
                      return null;
                  }
  
                  public Control[] getControls() {
                      return null;
                  }
  
              });
  
              mPlayer.addPlayerListener(this);
              mPlayer.realize();
              AudioPathControl  lPathCtr = (AudioPathControl) mPlayer.getControl("net.rim.device.api.media.control.AudioPathControl");
              lPathCtr.setAudioPath(AudioPathControl.AUDIO_PATH_HANDSET);
              mPlayer.prefetch();
              //if ( DeviceInfo.isSimulator() == false) { //only start player on real device
              mPlayer.start();
  
              //}
  
          }catch (Throwable e){
          }
  
      }
  
      public void playerUpdate(Player arg0, String event, Object eventData) {
          if (event.equals(PlayerListener.BUFFERING_STARTED)){
              mBuffering=true;
          } else if (event.equals(PlayerListener.DEVICE_UNAVAILABLE)) {
              //pausing both player and recorder
              try {
                  //already stooped mPlayer.stop();
                   mSendStream.getPlayer().stop();
              } catch (Throwable e) {
              }
          } else if (event == PlayerListener.DEVICE_AVAILABLE) {
              //starting both player and recorder
              try {
                  mSendStream.getPlayer().start();
                  stop();
                  reset();
                  start();
              } catch (Throwable e) {
              }
          }
      }
  
      public int getPlayLevel() {
          if (mPlayer !=null) {
              return ((VolumeControl)mPlayer.getControl("VolumeControl")).getLevel();
          } else {
              return 0;
          }
      }
  
      public void setPlayLevel(int level) {
          if (mPlayer !=null) {
              ((VolumeControl)mPlayer.getControl("VolumeControl")).setLevel(level);
          }
      }
  
      public void enableSpeaker(boolean value) {
          if (mPlayer == null) return;//just ignore
          AudioPathControl  lPathCtr = (AudioPathControl) mPlayer.getControl      ("net.rim.device.api.media.control.AudioPathControl");
          try {
              lPathCtr.setAudioPath(value?AudioPathControl.AUDIO_PATH_HANDSFREE:AudioPathControl.AUDIO_PATH_HANDSET);
          } catch (Throwable e) {
          }
      }
  
      public boolean isSpeakerEnabled() {
          if (mPlayer == null) return false;//just ignore
          AudioPathControl  lPathCtr = (AudioPathControl) mPlayer.getControl("net.rim.device.api.media.control.AudioPathControl");
          return  lPathCtr.getAudioPath()==AudioPathControl.AUDIO_PATH_HANDSFREE;
      }
      public long getPlayerTs() {
          return mPlayerTs;
      }
  

  When I try to write ECM/pcm audio or audio/basic for Simulator ContentDescriptor throws MediaException.As I understand that I did something wrong, but how can I give a pcm for ContentDescriptor entry?

  Any ideas?

  ThanksMSohm , but I already find solution. For that Datasource accepts the pcm (it's weird), but you must write audio/x-pcm not ECM/pcm audio or audio/basic. Before that he was the exception in time of creating player.