Search Results

Search found 9649 results on 386 pages for 'usb audio'.

Page 138/386 | < Previous Page | 134 135 136 137 138 139 140 141 142 143 144 145  | Next Page >

  • Linux RFID reader HID Device not matching driver

    - by blietaer
    Hello, I got a RFID reader (GigaTek PCR330A-00) that is meant to be recognized under linux/windows as a (Human Interface Device) keyboard/USB. I hate to say this but it is working as a charm under Win7 but not "really" under Linux. Under Debian-like distros (x/k/Ubuntu, Debian,..), or Gentoo, or... I just can't have the device working at all: the device scan well (it has its USB 5V, so it is happy/beeping/blinking) something happened in the dmesg, but no immediate screen display of the RFID Tag code as expected (and seen under win7) Support is claiming it is ok under RHEL or SLED "enterprises" distros... and I must admit I saw it working under a RHEL4... I tried stealing the driver but did not succeed having my reader working... My question is thus double: 1./ How can I hack the kernel to add support to my device (simply register PID/VID?) ? 2./ What is different at all in a "enterprise" proprietary distro? how can I re-use it? Thank you for any hint/help. Cheers,

    Read the article

  • 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 ?

    Read the article

  • 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.

    Read the article

  • 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.

    Read the article

  • 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?

    Read the article

  • 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?

    Read the article

  • 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

    Read the article

  • WinUSB failing on non-development computers

    - by Giawa
    Good afternoon, WinUSB is working well on the development computer that I am using (Win XP SP3). I am able to download new firmware to the Cypress FX2, and then connect to the new USB device once it 'renumerates'. However, if I've tried the same code with the WinUSB driver on a few other computers (Win XP SP3, Win7 x64) and they both returned the error "A device attached to the system is not functioning." when trying to use CreateFile to get a handle to the USB device. The devicePath was found successfully, so I'm not sure why it cannot connect to the device. Furthermore, the device manager states that my device is working properly. I'm curious if I'm missing something when compiling the code? I would guess that my development computer has something installed on it that the other computers do not? Or perhaps it's a power setting and the device is going to sleep (although I've fooled around with the Power Options on each computer to no avail). Does anyone have any ideas? I've compiled under Visual Studio 2008, and have installed the Microsoft C++ 2008 Redistributable Package on the computers that I've tested on. Thanks, Giawa

    Read the article

  • 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

    Read the article

  • How do you create a bootable partition on a USB drive?

    - by Nathan DeWitt
    I have a bootable ISO designed to be burned to a double-layer DVD. I don't have a double layer DVD burner, so I would like to stick the ISO image on a 50 GB partition on a USB hard drive I have. How do I get the boot info onto the hard drive? If it helps, it's an OSx86 Live CD. Attempt 1: booted into Ubuntu 9.04 LiveCD deleted the partition on my existing USB hard drive sudo dd if=/path/to/image.iso of=/dev/MyUSB booted to USB drive error: Error Loading OS Atempt 2: booted into Ubuntu 9.04 LiveCD deleted the partition on my existing USB hard drive sudo mkdosfs -I -v -n iPC /dev/MyUSB sudo syslinux /dev/MyUSB sudo dd if=/path/to/image.iso of=/dev/MyUSB booted to USB drive error: Selected boot device not available - strike F1 to retry boot, F2 for setup utility

    Read the article

  • Ubuntu karmic 9.10 Live image on USB - not working.

    - by Vivek Sharma
    This is my configuration 4GB pendrive, HP ubuntu-9.10-desktop-i386 image file for live USB install pendrivelinux (u910p) and ubetbootin (unetbootin.sourceforge.net) machine T61 Earlier I have installed ubuntu live image using above two mentioned utilities, numerous times. But, on a 2gb kingston flash-drive. Today, i am trying to install the live-image on 4gb HP flash-drive. Both the utilities install, i can see the files in the drive, even the wubi-installer is working, it say press "reboot" to boot in live-ubuntu. But, when i press "reboot" it does not reboot my win7. Now, when i reboot, select boot-usb in bios, it say "no boot record". I am making my usb bootable, using the utility, even then nothing is working out. Did this a few times. Is 4GB usb a prob, does anyone knows how to partition my usb in 2-2gb and install it on one partition, and then use the live image. Is it possible.

    Read the article

  • 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.

    Read the article

  • 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.

    Read the article

  • How to auto-scan any plugged in usb storage device with clamav?

    - by ossi
    I'd like to do an automatic virus scan on any plugged in usb device using ClamAV. I'm using Ubuntu 12.04. The closest thing I found was: Run clamav on mount of flashdrive How to run a shell script when a new USB storage device is detected? The first one is not working for me and the second one seems to target a known device. Is there a tutorial around I've missed? Or can I get some help with udev rules that apply to any usb storage device added? Currently nothing seems to do anything.

    Read the article

  • How do I install the driver for my Linksys AE1200 Wireless-N USB Adapter?

    - by Lewis Graham
    I recently downloaded Ubuntu from the main website with the hopes of dual booting it with Windows. While the operating system works, it says that I need to install a driver for my graphics card. When I type in my password the installation fails. I figure it is because I need Internet access. I tried to install my WiFi USB adapter with the installation disc but Ubuntu doesn't seem to run the setup when I click the exe. What are my available solutions as I would really like to use Ubuntu from my programming and Windows for my gaming needs? The name as read on the box is: Linksys AE1200 Wireless- N USB Adapter The description reads as such: ID 13b1:0039 Linksys (a comma messed up format) ID 046d:0a0b Logitech, Inc. ClearChat Pro USB (headset)

    Read the article

  • Windows Server Backup "Reading Data; please wait..."

    - by Reafidy
    On windows Server 2008 R2 I have recently added the windows server backup (WSB) feature. Opening WSB I get the message "Reading Data; please wait...". This message fails to go away, even after leaving the server for over 12 hours. I also notice in the task manager that svchost.exe (username: networkservice) is using all available processing power. So I terminated that process and then WSB comes on-line. However after restarting the server and WSB the issue reoccurs. WSB also fails to recognize my store-in-go flash drive (2gb). What is the underlying problem here?

    Read the article

  • How can I add usbip modules on Redhat 6 kernel?

    - by Gk.
    I have RHEL 6 with # uname -r 2.6.32-131.0.15.el6.x86_64 I'm trying to build usbip modules on staging driver. Everything is OK. I have all needed *.ko files. But I cannot add those modules on running kernel. # pwd /lib/modules/2.6.32-131.0.15.el6.x86_64 # ls | grep ko usbip_common_mod.ko usbip.ko vhci-hcd.ko # modprobe usbip FATAL: Error inserting usbip (/lib/modules/2.6.32-131.0.15.el6.x86_64/usbip.ko): Required key not available # insmod usbip.ko insmod: error inserting 'usbip.ko': -1 Required key not available How can I add it? Do I need to rebuild whole kernel? TIA, giobuon

    Read the article

  • 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.

    Read the article

  • How to roll-your-own live CD for safe home browsing

    - by user36533
    Hi, I'm interested in booting-off-flash (i.e. like livecd) for more secure online banking at home. -I like system rescue CD, but AFAIK it doesn't have the wifi drivers. (These are convenient) -ubuntu live cd has the wi-fi drivers, but also has a lot of stuff I don't need -I'd like a way to save some basic config settings (e.g. wifi SSID and passphrase), so that wifi works upon startup, i.e. without having to re-enter the settings. What's the best way to 'roll my own slightly-customized boot-from-flash live cd? thanks, bill

    Read the article

  • 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.

    Read the article

  • While running a batch file in Windows 7 with Admin rights from a thumb drive, how can I get the file path back to the thumb drive?

    - by Jeremy DeStefano
    I have a piece of software that is being distributed to several departments for installation onto Windows 7 laptops. They install software from the thumb drive and then they have to run a script to properly configure the software. Because the script is changing registry files and program files, it requires Admin rights. When running as Admin, it drops into the System32 folder and I no longer have an easy scriptable way to access files that need to be copied from the thumb drive, simply because I don't know for sure what drive letter its going to use on the various machines. Previous installations were on Windows XP and the command window file path stayed within the script folder. I've found similar questions here and I have already tried Relative Paths, but it can't seem to find the proper folder on the thumb drive or I can't seem to find the proper way to format it.

    Read the article

  • MP3 player no longer syncing

    - by zildjohn01
    I recently installed third-party drivers for the (Sony) PS3 controller on my friend's PC (Windows XP). I found out a few days later that his MP3 player (also Sony) is no longer recognized by Windows. He gets the "connect device" sound, and about 250ms later, the "disconnect device" sound. I figured the controller driver took over the Walkman's device ID, so I went through the registry and C:\Windows\inf removing all references to Sony's VID (054C), but I haven't had any luck. What would you do in this situation?

    Read the article

  • 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?

    Read the article

  • 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.

    Read the article

  • 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?

    Read the article

< Previous Page | 134 135 136 137 138 139 140 141 142 143 144 145  | Next Page >