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• WPF Storyboard delay in playing wma files

  - by Rita

  I'm a complete beginner in WPF and have an app that uses StoryBoard to play a sound. public void PlaySound() { MediaElement m = (MediaElement)audio.FindName("MySound.wma"); m.IsMuted = false; FrameworkElement audioKey = (FrameworkElement)keys.FindName("MySound"); Storyboard s = (Storyboard)audioKey.FindResource("MySound.wma"); s.Begin(audioKey); } <Storyboard x:Key="MySound.wma"> <MediaTimeline d:DesignTimeNaturalDuration="1.615" BeginTime="00:00:00" Storyboard.TargetName="MySound.wma" Source="Audio\MySound.wma"/> </Storyboard> I have a horrible lag and sometimes it takes good 10 seconds for the sound to be played. I suspect this has something to do with the fact that no matter how long I wait - The sound doesn't get played until after I leave the function. I don't understand it. I call Begin, and nothing happens. Is there a way to replace this method, or StoryBoard object with something that plays instantly and without a lag?

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• What do you use to play sound in iPhone games?

  - by zoul

  Hello! I have a performance-intensive iPhone game I would like to add sounds to. There seem to be about three main choices: (1) AVAudioPlayer, (2) Audio Queues and (3) OpenAL. I’d hate to write pages of low-level code just to play a sample, so that I would like to use AVAudioPlayer. The problem is that it seems to kill the performace – I’ve done a simple measuring using CFAbsoluteTimeGetCurrent and the play message seems to take somewhere from 9 to 30 ms to finish. That’s quite miserable, considering that 25 ms == 40 fps. Of course there is the prepareToPlay method that should speed things up. That’s why I wrote a simple class that keeps several AVAudioPlayers at its disposal, prepares them beforehand and then plays the sample using the prepared player. No cigar, still it takes the ~20 ms I mentioned above. Such performance is unusable for games, so what do you use to play sounds with a decent performance on iPhone? Am I doing something wrong with the AVAudioPlayer? Do you play sounds with Audio Queues? (I’ve written something akin to AVAudioPlayer before 2.2 came out and I would love to spare that experience.) Do you use OpenAL? If yes, is there a simple way to play sounds with OpenAL, or do you have to write pages of code? Update: Yes, playing sounds with OpenAL is fairly simple.

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• Why does this gstreamer pipeline stall ?

  - by timday

  I've been playing around with gstreamer pipelines using gst-launch. I don't have any problems if I just want to process audio or video separately (to separate files, or to alsasink/ximagesink), but I'm confused by what I need to do to mux the streams back together using, say avimux. This gst-launch-0.10 filesrc location=MVI_2034.AVI ! decodebin name=dec \ dec. ! queue ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=1' ! queue ! mux. \ dec. ! queue ! videoflip 1 ! ffmpegcolorspace ! jpegenc ! queue ! mux. \ avimux name=mux ! filesink location=out.avi just outputs Setting pipeline to PAUSED ... Pipeline is PREROLLING ... and then stalls indefinitely. What's the trick ?

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• JMF microphone volume controller

  - by TacB0sS

  How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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• Graphing the pitch (frequency) of a sound

  - by Coronatus

  I want to plot the pitch of a sound into a graph. Currently I can plot the amplitude. The graph below is created by the data returned by getUnscaledAmplitude(): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(new FileInputStream(file))); byte[] bytes = new byte[(int) (audioInputStream.getFrameLength()) * (audioInputStream.getFormat().getFrameSize())]; audioInputStream.read(bytes); // Get amplitude values for each audio channel in an array. graphData = type.getUnscaledAmplitude(bytes, this); public int[][] getUnscaledAmplitude(byte[] eightBitByteArray, AudioInfo audioInfo) { int[][] toReturn = new int[audioInfo.getNumberOfChannels()][eightBitByteArray.length / (2 * audioInfo. getNumberOfChannels())]; int index = 0; for (int audioByte = 0; audioByte < eightBitByteArray.length;) { for (int channel = 0; channel < audioInfo.getNumberOfChannels(); channel++) { // Do the byte to sample conversion. int low = (int) eightBitByteArray[audioByte]; audioByte++; int high = (int) eightBitByteArray[audioByte]; audioByte++; int sample = (high << 8) + (low & 0x00ff); if (sample < audioInfo.sampleMin) { audioInfo.sampleMin = sample; } else if (sample > audioInfo.sampleMax) { audioInfo.sampleMax = sample; } toReturn[channel][index] = sample; } index++; } return toReturn; } But I need to show the audio's pitch, not amplitude. Fast Fourier transform appears to get the pitch, but it needs to know more variables than the raw bytes I have, and is very complex and mathematical. Is there a way I can do this?

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• Python on Mac: Fink? MacPorts? Builtin? Homebrew? Binary installer?

  - by BastiBechtold

  For the last few days, I have been trying to use Python for some audio development. The thing is, Mac OSX does not handle uninstalling stuff well. Actually, there is no way to uninstall anything. Once it is on your system, you better pray that it didn't do any funny stuff. Hence, I don't really want to rely on installer packages for Python. So I turn to Homebrew and install Python using Homebrew. Works fabulously. Using pip, Numpy, SciPy, Matplotlib were no (big) problem, either. Now I want to play audio. There is a host of different packages out there, but pip does not seem willing to install any. But, there is a binary distribution for PyGame, which I guess should work with the built-in Python. Hence my question: What would you do? Would you just install the binary distributions and hope that they interoperate well and never need uninstalling? Would you hack your way through whichever package control management system you prefer and deal with its problems? Something else?

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• Syncing two AS3 NetStreams

  - by Lowgain

  I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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• Playing a sequence of sounds without gaps (iPhone)

  - by Fiire

  I thought maybe the fastest way was to go with Sound Services. It is quite efficient, but I need to play sounds in a sequence, not overlapped. Therefore I used a callback method to check when the sound has finished. This cycle produces around 0.3 seconds in lag. I know this sounds very strict, but it is basically the main axis of the program. EDIT: I now tried using AVAudioPlayer, but I can't play sounds in a sequence without using audioPlayerDidFinishPlaying since that would put me in the same situation as with the callback method of SoundServices. EDIT2: I think that if I could somehow get to join the parts of the sounds I want to play into a large file, I could get the whole audio file to sound continuously. EDIT3: I thought this would work, but the audio overlaps: waitTime = player.deviceCurrentTime; for (int k = 0; k < [colores count]; k++) { player.currentTime = 0; [player playAtTime:waitTime]; waitTime += player.duration; } Thanks

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• how to upload a audio file using REST webservice in Google App Engine for Java

  - by sathya

  Am using google app engine with eclipse IDE and trying to upload a audio file. I used the File Upload in Google App Engine For Java and can able to upload the file successfully. Now am planning to use REST web service for it. I had analyzed in developers.google but i failed. Can anyone suggest me how to implement REST Web services in google app engine using Eclipse. The code google provided is shown below, // file Upload.java public class Upload extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doPost(HttpServletRequest req, HttpServletResponse res) throws ServletException, IOException { Map<String, BlobKey> blobs = blobstoreService.getUploadedBlobs(req); BlobKey blobKey = blobs.get("myFile"); if (blobKey == null) { res.sendRedirect("/"); } else { res.sendRedirect("/serve?blob-key=" + blobKey.getKeyString()); }}} // file Serve.java public class Serve extends HttpServlet { private BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); public void doGet(HttpServletRequest req, HttpServletResponse res) throws IOException { BlobKey blobKey = new BlobKey(req.getParameter("blob-key")); blobstoreService.serve(blobKey, res); }} // file index.jsp <%@ page import="com.google.appengine.api.blobstore.BlobstoreServiceFactory" %> <%@ page import="com.google.appengine.api.blobstore.BlobstoreService" %> <% BlobstoreService blobstoreService = BlobstoreServiceFactory.getBlobstoreService(); %> <form action="<%= blobstoreService.createUploadUrl("/upload") %>" method="post" enctype="multipart/form-data"> <input type="file" name="myFile"> <input type="submit" value="Submit"> </form> // web.xml <servlet> <servlet-name>Upload</servlet-name> <servlet-class>Upload</servlet-class> </servlet> <servlet> <servlet-name>Serve</servlet-name> <servlet-class>Serve</servlet-class> </servlet> <servlet-mapping> <servlet-name>Upload</servlet-name> <url-pattern>/upload</url-pattern> </servlet-mapping> <servlet-mapping> <servlet-name>Serve</servlet-name> <url-pattern>/serve</url-pattern> </servlet-mapping> Now how to provide a rest web service for the above code. Kindly suggest me an idea.

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• How to play audio in Java Application

  - by user577829

  I'm making a java application and I need to play audio. I'm playing mainly small sound files of my cannon firing (its a cannon shooting game) and the projectiles exploding, though I plan on having looping background music. I have found two different methods to accomplish this, but both don't work how I want. The first method is literally a method: public void playSoundFile(File file) {//http://java.ittoolbox.com/groups/technical-functional/java-l/sound-in-an-application-90681 try { //get an AudioInputStream AudioInputStream ais = AudioSystem.getAudioInputStream(file); //get the AudioFormat for the AudioInputStream AudioFormat audioformat = ais.getFormat(); System.out.println("Format: " + audioformat.toString()); System.out.println("Encoding: " + audioformat.getEncoding()); System.out.println("SampleRate:" + audioformat.getSampleRate()); System.out.println("SampleSizeInBits: " + audioformat.getSampleSizeInBits()); System.out.println("Channels: " + audioformat.getChannels()); System.out.println("FrameSize: " + audioformat.getFrameSize()); System.out.println("FrameRate: " + audioformat.getFrameRate()); System.out.println("BigEndian: " + audioformat.isBigEndian()); //ULAW format to PCM format conversion if ((audioformat.getEncoding() == AudioFormat.Encoding.ULAW) || (audioformat.getEncoding() == AudioFormat.Encoding.ALAW)) { AudioFormat newformat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, audioformat.getSampleRate(), audioformat.getSampleSizeInBits() * 2, audioformat.getChannels(), audioformat.getFrameSize() * 2, audioformat.getFrameRate(), true); ais = AudioSystem.getAudioInputStream(newformat, ais); audioformat = newformat; } //checking for a supported output line DataLine.Info datalineinfo = new DataLine.Info(SourceDataLine.class, audioformat); if (!AudioSystem.isLineSupported(datalineinfo)) { //System.out.println("Line matching " + datalineinfo + " is not supported."); } else { //System.out.println("Line matching " + datalineinfo + " is supported."); //opening the sound output line SourceDataLine sourcedataline = (SourceDataLine) AudioSystem.getLine(datalineinfo); sourcedataline.open(audioformat); sourcedataline.start(); //Copy data from the input stream to the output data line int framesizeinbytes = audioformat.getFrameSize(); int bufferlengthinframes = sourcedataline.getBufferSize() / 8; int bufferlengthinbytes = bufferlengthinframes * framesizeinbytes; byte[] sounddata = new byte[bufferlengthinbytes]; int numberofbytesread = 0; while ((numberofbytesread = ais.read(sounddata)) != -1) { int numberofbytesremaining = numberofbytesread; sourcedataline.write(sounddata, 0, numberofbytesread); } } } catch (Exception e) { e.printStackTrace(); } } The problem with this is that my entire program stops until the sound file is finished, or at least nearly finished. The second method is this: File file = new File("Launch1.wav"); AudioClip clip; try { clip = JApplet.newAudioClip(file.toURL()); clip.play(); } catch (Exception e) { e.getMessage(); } The problem I have here is that every time the sound file ends early or doesn't play at all depending on where I place the code. Is their any way to play sound without the above mentioned problems? Am I doing something wrong? Any help is greatly appreciated.

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• Does Hauppauge WinTV HVR-900 (r2) [USB ID 2040:6502] work with ubuntu 12.04 LTS?

  - by nightfly

  I have this DVB+Analog usb tv tuner Hauppauge WinTV HVR-900 (r2) [USB ID 2040:6502]. This used to work under ubuntu 10.04 LTS. But in 12.04 there seems to be a problem. I have linux-firmware-nonfree and ivtv-utils installed. I am running Ubuntu 12.04.1 LTS 64 bit with all updates installed and the default unity environment. When I run mplayer tv:// -tv driver=v4l2:device=/dev/video1:input=1:norm=PAL I get a solid green screen and no picture. Here input 1 is the composite input of the card. MPlayer svn r34540 (Ubuntu), built with gcc-4.6 (C) 2000-2012 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing tv://. TV file format detected. Selected driver: v4l2 name: Video 4 Linux 2 input author: Martin Olschewski comment: first try, more to come ;-) Selected device: Hauppauge WinTV HVR 900 (R2) Tuner cap: Tuner rxs: Capabilities: video capture VBI capture device tuner audio read/write streaming supported norms: 0 = NTSC; 1 = NTSC-M; 2 = NTSC-M-JP; 3 = NTSC-M-KR; 4 = NTSC-443; 5 = PAL; 6 = PAL-BG; 7 = PAL-H; 8 = PAL-I; 9 = PAL-DK; 10 = PAL-M; 11 = PAL-N; 12 = PAL-Nc; 13 = PAL-60; 14 = SECAM; 15 = SECAM-B; 16 = SECAM-G; 17 = SECAM-H; 18 = SECAM-DK; 19 = SECAM-L; 20 = SECAM-Lc; inputs: 0 = Television; 1 = Composite1; 2 = S-Video; Current input: 1 Current format: YUYV v4l2: current audio mode is : MONO v4l2: ioctl set format failed: Invalid argument v4l2: ioctl set format failed: Invalid argument v4l2: ioctl set format failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument v4l2: ioctl query control failed: Invalid argument Failed to open VDPAU backend libvdpau_nvidia.so: cannot open shared object file: No such file or directory [vdpau] Error when calling vdp_device_create_x11: 1 ========================================================================== Opening video decoder: [raw] RAW Uncompressed Video Movie-Aspect is undefined - no prescaling applied. VO: [xv] 640x480 = 640x480 Packed YUY2 Selected video codec: [rawyuy2] vfm: raw (RAW YUY2) ========================================================================== Audio: no sound Starting playback... v4l2: select timeout V: 0.0 2/ 2 ??% ??% ??,?% 0 0 v4l2: select timeout V: 0.0 4/ 4 ??% ??% ??,?% 0 0 v4l2: select timeout V: 0.0 6/ 6 ??% ??% ??,?% 0 0 v4l2: select timeout v4l2: 0 frames successfully processed, 1 frames dropped. Exiting... (Quit) Here is the dmesg of the card when plugged in.. [12742.228097] usb 1-4: new high-speed USB device number 3 using ehci_hcd [12742.367289] em28xx: New device WinTV HVR-900 @ 480 Mbps (2040:6502, interface 0, class 0) [12742.367296] em28xx: Audio Vendor Class interface 0 found [12742.367585] em28xx #0: chip ID is em2882/em2883 [12742.550086] em28xx #0: i2c eeprom 00: 1a eb 67 95 40 20 02 65 d0 12 5c 03 82 1e 6a 18 [12742.550104] em28xx #0: i2c eeprom 10: 00 00 24 57 66 07 01 00 00 00 00 00 00 00 00 00 [12742.550120] em28xx #0: i2c eeprom 20: 46 00 01 00 f0 10 02 00 b8 00 00 00 5b e0 00 00 [12742.550135] em28xx #0: i2c eeprom 30: 00 00 20 40 20 6e 02 20 10 01 01 01 00 00 00 00 [12742.550150] em28xx #0: i2c eeprom 40: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [12742.550165] em28xx #0: i2c eeprom 50: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [12742.550181] em28xx #0: i2c eeprom 60: 00 00 00 00 00 00 00 00 00 00 18 03 34 00 30 00 [12742.550196] em28xx #0: i2c eeprom 70: 32 00 37 00 38 00 32 00 33 00 39 00 30 00 31 00 [12742.550211] em28xx #0: i2c eeprom 80: 00 00 1e 03 57 00 69 00 6e 00 54 00 56 00 20 00 [12742.550226] em28xx #0: i2c eeprom 90: 48 00 56 00 52 00 2d 00 39 00 30 00 30 00 00 00 [12742.550241] em28xx #0: i2c eeprom a0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [12742.550257] em28xx #0: i2c eeprom b0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [12742.550272] em28xx #0: i2c eeprom c0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [12742.550287] em28xx #0: i2c eeprom d0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [12742.550302] em28xx #0: i2c eeprom e0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [12742.550317] em28xx #0: i2c eeprom f0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [12742.550334] em28xx #0: EEPROM ID= 0x9567eb1a, EEPROM hash = 0x2bbf3bdd [12742.550338] em28xx #0: EEPROM info: [12742.550340] em28xx #0: AC97 audio (5 sample rates) [12742.550343] em28xx #0: 500mA max power [12742.550346] em28xx #0: Table at 0x24, strings=0x1e82, 0x186a, 0x0000 [12742.552590] em28xx #0: Identified as Hauppauge WinTV HVR 900 (R2) (card=18) [12742.555516] tveeprom 15-0050: Hauppauge model 65018, rev B2C0, serial# 1292061 [12742.555523] tveeprom 15-0050: tuner model is Xceive XC3028 (idx 120, type 71) [12742.555529] tveeprom 15-0050: TV standards PAL(B/G) PAL(I) PAL(D/D1/K) ATSC/DVB Digital (eeprom 0xd4) [12742.555534] tveeprom 15-0050: audio processor is None (idx 0) [12742.555537] tveeprom 15-0050: has radio [12742.570297] tuner 15-0061: Tuner -1 found with type(s) Radio TV. [12742.570327] xc2028 15-0061: creating new instance [12742.570332] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [12742.573685] xc2028 15-0061: Loading 80 firmware images from xc3028-v27.fw, type: xc2028 firmware, ver 2.7 [12742.624056] xc2028 15-0061: Loading firmware for type=BASE MTS (5), id 0000000000000000. [12744.126591] xc2028 15-0061: Loading firmware for type=MTS (4), id 000000000000b700. [12744.153586] xc2028 15-0061: Loading SCODE for type=MTS LCD NOGD MONO IF SCODE HAS_IF_4500 (6002b004), id 000000000000b700. [12744.280963] Registered IR keymap rc-hauppauge [12744.281151] input: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc1/input10 [12744.281541] rc1: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc1 [12744.282454] em28xx #0: Config register raw data: 0xd0 [12744.284709] em28xx #0: AC97 vendor ID = 0xffffffff [12744.285829] em28xx #0: AC97 features = 0x6a90 [12744.285832] em28xx #0: Empia 202 AC97 audio processor detected [12744.359211] em28xx #0: v4l2 driver version 0.1.3 [12744.404066] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [12745.915089] MTS (4), id 00000000000000ff: [12745.915100] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [12746.161668] em28xx #0: V4L2 video device registered as video1 [12746.161673] em28xx #0: V4L2 VBI device registered as vbi0 [12746.162845] em28xx-audio.c: probing for em28xx Audio Vendor Class [12746.162848] em28xx-audio.c: Copyright (C) 2006 Markus Rechberger [12746.162851] em28xx-audio.c: Copyright (C) 2007-2011 Mauro Carvalho Chehab [12746.221099] xc2028 15-0061: attaching existing instance [12746.221105] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [12746.221109] em28xx #0: em28xx #0/2: xc3028 attached [12746.221113] DVB: registering new adapter (em28xx #0) [12746.221118] DVB: registering adapter 0 frontend 0 (Micronas DRXD DVB-T)... [12746.221869] em28xx #0: Successfully loaded em28xx-dvb [13111.196055] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13112.720062] MTS (4), id 00000000000000ff: [13112.720072] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13214.956057] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13216.479806] MTS (4), id 00000000000000ff: [13216.479816] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13276.408056] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13277.932093] MTS (4), id 00000000000000ff: [13277.932104] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13305.032076] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13306.556449] MTS (4), id 00000000000000ff: [13306.556460] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13392.236055] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13393.760123] MTS (4), id 00000000000000ff: [13393.760133] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13637.534053] usb 1-4: USB disconnect, device number 3 [13637.534183] em28xx #0: disconnecting em28xx #0 video [13637.560214] em28xx #0: V4L2 device vbi0 deregistered [13637.560335] em28xx #0: V4L2 device video1 deregistered [13637.561237] xc2028 15-0061: destroying instance [13639.772120] usb 1-4: new high-speed USB device number 4 using ehci_hcd [13639.911351] em28xx: New device WinTV HVR-900 @ 480 Mbps (2040:6502, interface 0, class 0) [13639.911357] em28xx: Audio Vendor Class interface 0 found [13639.911637] em28xx #0: chip ID is em2882/em2883 [13640.094262] em28xx #0: i2c eeprom 00: 1a eb 67 95 40 20 02 65 d0 12 5c 03 82 1e 6a 18 [13640.094280] em28xx #0: i2c eeprom 10: 00 00 24 57 66 07 01 00 00 00 00 00 00 00 00 00 [13640.094295] em28xx #0: i2c eeprom 20: 46 00 01 00 f0 10 02 00 b8 00 00 00 5b e0 00 00 [13640.094311] em28xx #0: i2c eeprom 30: 00 00 20 40 20 6e 02 20 10 01 01 01 00 00 00 00 [13640.094326] em28xx #0: i2c eeprom 40: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [13640.094341] em28xx #0: i2c eeprom 50: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 [13640.094356] em28xx #0: i2c eeprom 60: 00 00 00 00 00 00 00 00 00 00 18 03 34 00 30 00 [13640.094371] em28xx #0: i2c eeprom 70: 32 00 37 00 38 00 32 00 33 00 39 00 30 00 31 00 [13640.094386] em28xx #0: i2c eeprom 80: 00 00 1e 03 57 00 69 00 6e 00 54 00 56 00 20 00 [13640.094401] em28xx #0: i2c eeprom 90: 48 00 56 00 52 00 2d 00 39 00 30 00 30 00 00 00 [13640.094416] em28xx #0: i2c eeprom a0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [13640.094432] em28xx #0: i2c eeprom b0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [13640.094447] em28xx #0: i2c eeprom c0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [13640.094462] em28xx #0: i2c eeprom d0: 84 12 00 00 05 50 1a 7f d4 78 23 fa fd d0 28 89 [13640.094477] em28xx #0: i2c eeprom e0: ff 00 00 00 04 84 0a 00 01 01 20 77 00 40 1d b7 [13640.094492] em28xx #0: i2c eeprom f0: 13 f0 74 02 01 00 01 79 63 00 00 00 00 00 00 00 [13640.094509] em28xx #0: EEPROM ID= 0x9567eb1a, EEPROM hash = 0x2bbf3bdd [13640.094512] em28xx #0: EEPROM info: [13640.094515] em28xx #0: AC97 audio (5 sample rates) [13640.094517] em28xx #0: 500mA max power [13640.094521] em28xx #0: Table at 0x24, strings=0x1e82, 0x186a, 0x0000 [13640.097391] em28xx #0: Identified as Hauppauge WinTV HVR 900 (R2) (card=18) [13640.099617] tveeprom 15-0050: Hauppauge model 65018, rev B2C0, serial# 1292061 [13640.099623] tveeprom 15-0050: tuner model is Xceive XC3028 (idx 120, type 71) [13640.099629] tveeprom 15-0050: TV standards PAL(B/G) PAL(I) PAL(D/D1/K) ATSC/DVB Digital (eeprom 0xd4) [13640.099634] tveeprom 15-0050: audio processor is None (idx 0) [13640.099637] tveeprom 15-0050: has radio [13640.112849] tuner 15-0061: Tuner -1 found with type(s) Radio TV. [13640.112877] xc2028 15-0061: creating new instance [13640.112882] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [13640.115930] xc2028 15-0061: Loading 80 firmware images from xc3028-v27.fw, type: xc2028 firmware, ver 2.7 [13640.164057] xc2028 15-0061: Loading firmware for type=BASE MTS (5), id 0000000000000000. [13641.666643] xc2028 15-0061: Loading firmware for type=MTS (4), id 000000000000b700. [13641.693262] xc2028 15-0061: Loading SCODE for type=MTS LCD NOGD MONO IF SCODE HAS_IF_4500 (6002b004), id 000000000000b700. [13641.820765] Registered IR keymap rc-hauppauge [13641.820958] input: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc2/input11 [13641.821335] rc2: em28xx IR (em28xx #0) as /devices/pci0000:00/0000:00:1a.7/usb1/1-4/rc/rc2 [13641.822256] em28xx #0: Config register raw data: 0xd0 [13641.824526] em28xx #0: AC97 vendor ID = 0xffffffff [13641.825503] em28xx #0: AC97 features = 0x6a90 [13641.825507] em28xx #0: Empia 202 AC97 audio processor detected [13641.899015] em28xx #0: v4l2 driver version 0.1.3 [13641.944064] xc2028 15-0061: Loading firmware for type=BASE F8MHZ MTS (7), id 0000000000000000. [13643.470765] MTS (4), id 00000000000000ff: [13643.470776] xc2028 15-0061: Loading firmware for type=MTS (4), id 0000000100000007. [13643.717713] em28xx #0: V4L2 video device registered as video1 [13643.717718] em28xx #0: V4L2 VBI device registered as vbi0 [13643.718770] em28xx-audio.c: probing for em28xx Audio Vendor Class [13643.718775] em28xx-audio.c: Copyright (C) 2006 Markus Rechberger [13643.718778] em28xx-audio.c: Copyright (C) 2007-2011 Mauro Carvalho Chehab [13643.777148] xc2028 15-0061: attaching existing instance [13643.777154] xc2028 15-0061: type set to XCeive xc2028/xc3028 tuner [13643.777158] em28xx #0: em28xx #0/2: xc3028 attached [13643.777162] DVB: registering new adapter (em28xx #0) [13643.777167] DVB: registering adapter 0 frontend 0 (Micronas DRXD DVB-T)... [13643.777876] em28xx #0: Successfully loaded em28xx-dvb And here goes the lsmod output lsmod|grep em28xx em28xx_dvb 18579 0 dvb_core 110619 1 em28xx_dvb em28xx_alsa 18305 0 em28xx 109365 2 em28xx_dvb,em28xx_alsa v4l2_common 16454 3 tuner,tvp5150,em28xx videobuf_vmalloc 13589 1 em28xx videobuf_core 26390 2 em28xx,videobuf_vmalloc rc_core 26412 10 rc_hauppauge,ir_lirc_codec,ir_mce_kbd_decoder,ir_sony_decoder,ir_jvc_decoder,ir_rc6_decoder,ir_rc5_decoder,em28xx,ir_nec_decoder snd_pcm 97188 3 em28xx_alsa,snd_hda_intel,snd_hda_codec tveeprom 21249 1 em28xx videodev 98259 5 tuner,tvp5150,em28xx,v4l2_common,uvcvideo snd 78855 14 em28xx_alsa,snd_hda_codec_conexant,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device Isn't this driver mainline now? Or this card is not supported? Or the analog functionality is screwed? I need the analog capture working for this card. Please help!

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• I want to trace the activity of my customize link sent via email or on chat to my custome

  - by anilkumble789

  I want to trace the activity of my customize link sent via email or on chat to my customer. Activity like : whether they opened the link or not? How much time they were on page? examples: I have decided to sent business proposal link to Mr.ABC and Mr.XYZ So, for ABC the link would be like : www.mycompany.com/proposal ....abc... So, for XYZ the link would be like : www.mycompany.com/proposal ....xyz... its like link analytic. How to go ahead with with it?

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• Sound card / microphone impedance mismatch

  - by axk

  First of all I'm not completely sure this is impedance mismatch, but from what I found on the Internet I believe it is. It seems to be a common problem. The question is not as much about solving the problem as about why it is happening (if I'm right about the cause of the problem, of course). I had this quiet microphone problem with several built in cards and microphones and now with a Creative Audigy SE. There's a microphone boost option which introduces a lot of noise with volume increase, but even this doesn't seem to give loud enough sound in some cases. The mic on my current headphones is very quiet with Audigy SE without the boost but is very loud and low noise with an external Sound Blaster Connect. So the question is have I just been unlucky with my sound cards and microphones or is it a common problem? And if it is a common problem why is it so difficult for the vendors to standardize on the sound card / microphone impedance? Edit: the OS is Windows (XP/7), but I don't believe it is OS-specific.

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• Make headphone output mono.

  - by Jonathan

  my headphones are stereo but I would like the sound from the left and right to be combined then sent to both headphones. The reason is I'm watching a video where the people speaking are in the right ear as well as the music but they never speak in the left ear (it is not because they on the right side of the screen) If I take the right headphone off then I only hear the music in my left and there is no speaking.

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• MacBook Pro Boot Camp SPDIF passthrough?

  - by Ryan Zink

  I'm using Windows 7 through Boot Camp on a unibody Macbook Pro and am having problems using the SPDIF output. I get the expected Dolby Digital or DTS in some movies, but in other movies and in games (Source engine, StarCraft 2) where the output is enabled to 5.1, the output invariably shows up as Dolby Pro Logic, which means (I think) that passthrough is not enabled. The boot camp drivers for the sound card don't have any sort of control panel, and the Windows settings for enabling DTS and Dolby seem to work when I test those outputs in the sound settings. Is there some other setting or utility I can use to enable SPDIF passthrough for all programs?

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• Transcript creator OR Speech to text

  - by AndyMcKenna

  I listen to a daily podcast that is about 4 hours long. I think it would be a cool project if I could come with some way to generate transcripts of it automatically. Is there any software that will "listen" to the mp3s and create text of what they are saying? I'm not very concerned with differentiating who is talking because I think that would be asking too much. There are 4 main people speaking and others less often.

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• No sound (USB speakers) in Ubuntu 9.04 Jaunty Jackalope

  - by Mike

  I just installed Ubuntu 9.04 and have been unable to get sound functioning. I have a set of USB speakers (not USB powered with a stereo plug... they are totally USB). I've tried plugging them directly into the tower as well as into the USB port on my Dell 2405FPW monitor. Both USB ports are functioning correctly (I tested by sticking a flash drive in there - they both read it), and the speakers are functioning correctly in Windows. If it's relevant, I have an SB Audigy 2 sound card that came with the computer, but is not being used. Any ideas? Thanks! EDIT: These are the speakers - Logitech S-150 USB Speakers

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• using onboard sound card with dedicated soundcard

  - by NoCarrier

  Is this even possible? I need to connect multiple line level inputs into my machine, but my soundcard only has one Line-In. I'd like to use the Line-In on my onboard sound device to get an additional input

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• PulseAudio on Cygwin: Failed to create secure directory: Unknown error 13

  - by Nithin

  I am unable to run PulseAudio on Cygwin. Operating System: Windows 8 Pro 64 bit Cygwin Setup.exe Version: 2.831 (64 bit) PulseAudio Version: 2.1-1 When I run: pulseaudio -vv this is the output: D: [(null)] core-util.c: setpriority() worked. I: [(null)] core-util.c: Successfully gained nice level -11. I: [(null)] main.c: This is PulseAudio 2.1 D: [(null)] main.c: Compilation host: x86_64-unknown-cygwin D: [(null)] main.c: Compilation CFLAGS: -ggdb -O2 -pipe -fdebug-prefix-map=/usr/src/ports/pulseaudio/pulseaudio-2.1-1/build=/usr/src/debug/pulseaudio-2.1-1 -fdebug-prefix-map=/usr/src/ports/pulseaudio/pulseaudio-2.1-1/src/pulseaudio-2.1=/usr/src/debug/pulseaudio-2.1-1 -Wall -W -Wextra -Wno-long-long -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: [(null)] main.c: Running on host: CYGWIN_NT-6.2 x86_64 1.7.25(0.270/5/3) 2013-08-31 20:37 D: [(null)] main.c: Found 4 CPUs. I: [(null)] main.c: Page size is 65536 bytes D: [(null)] main.c: Compiled with Valgrind support: no D: [(null)] main.c: Running in valgrind mode: no D: [(null)] main.c: Running in VM: no D: [(null)] main.c: Optimized build: yes D: [(null)] main.c: FASTPATH defined, only fast path asserts disabled. I: [(null)] main.c: Machine ID is 5d8bd07cb924c67197184e42527f2603. E: [(null)] core-util.c: Failed to create secure directory: Unknown error 13 When I instead run pulseaudio -vv --start the output is this: E: [autospawn] core-util.c: Failed to create secure directory: Unknown error 13 W: [autospawn] lock-autospawn.c: Cannot access autospawn lock. E: [(null)] main.c: Failed to acquire autospawn lock When I ran strace pulseaudio -vv, the red-colored lines in the output were: 28 1637050 [main] pulseaudio 5104 fhandler_pty_slave::write: (669): pty output_mutex(0xBC) released 26 1637076 [main] pulseaudio 5104 write: 7 = write(2, 0x3FE171079, 7) 42 1637118 [main] pulseaudio 5104 fhandler_pty_slave::write: pty0, write(0x60003BB40, 51) 27 1637145 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex (0xBC): waiting -1 ms 23 1637168 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex: acquired Failed to create secure directory: Unknown error 13 21 1637189 [main] pulseaudio 5104 fhandler_pty_slave::write: (669): pty output_mutex(0xBC) released 29 1637218 [main] pulseaudio 5104 write: 51 = write(2, 0x60003BB40, 51) 46 1637264 [main] pulseaudio 5104 fhandler_pty_slave::write: pty0, write(0x3FE17106F, 4) 24 1637288 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex (0xBC): waiting -1 ms 24 1637312 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex: acquired Please can someone help me?

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• How can I prevent static when PC is plugged into an amplified speaker system?

  - by Kyle

  I've plugged a computer into an amp, using a 1/8 inch male extension cord, into a female adapter, that adapts into a male microphone 1/4 end. That being said, the amp sits at about half volume all the time because there are other things that play on it. (This issue is not flexible, nor is changing the amp) The problem is that now, even when I mute out the computer, you hear some static in the background. I was wondering some about some solutions (preferably multiple).

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• Easy to install JACK GUI for Mac OS X

  - by Mr. Man

  Does anyone know of an easy to install GUI for JACK on Mac OS X?

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• Export markers from Adobe Audition

  - by Alexander Gladysh

  Does anyone know any way to export markers from Adobe Audition to any other program (or file with documented format).

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• Real Time Sound Leveler

  - by Soldier.moth

  Lately I've been annoyed with Hulu as the commercials are significantly louder than the actual show. This has caused me to wonder if there existed any application either generic or specific to Hulu or Firefox to reduce the difference in sound volume between the show and commercials.

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• Windows 7 laptop's volume with headphones way too loud

  - by jcao219

  I have two headphones, neither has any amps or volume control, and both have the same problem with this computer. I normally have the volume on 10%, and that is optimal. I'm still very afraid that someday it will be at 100% and i'll accidentally have my headphones on. I tested how loud it is at maximum volume -- it's earsplittingly loud! In fact, it's so loud that if I lay the headphones down on the table, I can listen to a song perfectly clearly from meters away. Is there any way to make the volume control more safe for the ears?

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• Noise when I try to convert flac to mp3 using vlc

  - by Josef

  I'd like to convert my flac files to mp3s using vlc and it's Media-Convert/Save function. The destination file is written but only contains awful noise. I've tried most of the predefined profiles (mp4, wav, etc.) and all of them seem to yield the same result. What's wrong?

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