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  • An Approach to Incremental Conversion

    - by Paula Speranza-Hadley
    It is common for Oracle Enterprise Taxation and Policy Management (ETPM) customers to implement in multiple phases.  This results in a need for incremental conversion, where part of the data in is production and they are now adding new data.  Some of the new data can be new persons, accounts and their children, but some may be new tax types for existing taxpayers.  This document addresses a methodology for adding incremental data into ETPM.  It does not address every possible data scenario, but offers a path to achieving incremental conversion without the need for code changes.    https://blogs.oracle.com/tax/resource/IncrementalConversion.pdf  

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  • Google Analytics conversion tracking - referrals from payment provider

    - by martynas
    I have a question regarding conversion tracking using Google Analytics. My client uses an external payment service provider - SecureTrading. Problem: All website visitors who would like to make a purchase are taken to a payment form on https://securetrading.net and are redirected back after a successful payment. Google Analytics counts that as a referral and messes up conversion tracking stats. Question: What needs to be changed / added in the payment forms or Google Analytics settings so that the conversions would be assigned to the right traffic sources. Screenshot:

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  • My events don't show up in the goal funnels or conversion funnels

    - by Amit Bens
    I have an event set up on a website and I'd like to track the effect this event has on conversion rate. The event seems to be working fine - I can see it on Top Events with all the labels, etc. But when going into Goal Flow and selecting 'Event Category' these events don't show up. I have this running for about a week. And I have made multiple checks to verify that I have events that triggered the conversion goal. Any clue about what I'm doing wrong?

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  • 11.10 desktop alerts (volume change and terminal bell) stopped working but all other audio still works

    - by FlabbergastedPickle
    All, My sound works just fine in 11.10 64-bit install on HP dm1-4050 Sandy Bridge notebook (e.g. audio works in Banshee, flash, games, browser, Thunderbird email notification, etc.), but the core desktop notifications (e.g. pressing a tab in a terminal where there is more than one option should trigger a terminal bell, or changing volume using volume keys should be accompanied with the supporting "quack" that the volume app makes) do not work. I've intentionally disabled login sound as explained here on ask ubuntu but even enabling it back makes no difference. These notifications did work before just fine and I am not sure when did the actually stop working but it must've been fairly recently. Only things I did were trying to install some ppa edge xorg drivers for my intel card (a separate issue) but also reverted them all with ppa-purge once I discovered they did not improve anything. Other thing I did was check volume settings with alsamixer and did alsactl store for the soundcard after I did some experimenting with volume settings for PCM (on my laptop PCM at 100% crackles so I had to lower it and make pulseaudio ignore its setting as per ask ubuntu's page). That said, neither of these should have any bearing on the said notifications since the volume is up and they clearly work everywhere else but the core desktop events. The system ready drum sound when Ubuntu boots and user reaches the login screen also does not work. The guest login behaves exactly same as mine. Audio works (including the login sound since I've not disabled it for the guest account), but no quacks when changing the volume or terminal bell sounds... I've tried copying ubuntu sounds to /usr/share/sounds/ as suggested on ask ubuntu and that did not work. I also tried using dconf-editor to check sound theme settings and tried both freedesktop (which is what it was set to) and ubuntu, as suggested on ask ubuntu. This did not work either. I tried purging the ~/.pulse folder and the /tmp/*pulse* entries, rebooting and restarting pulseaudio with -D flag. While audio came back on and behaved just fine in all aspects (e.g. one can adjust volume levels, play music, games, in-browser sound stuff, and other app alerts) except for the system ready drum sound (at the login screen), and any system event (terminal bell and volume change quack sound). It is interesting that the quack sound works inside system settings-sound when adjusting levels there, but it does not when volume is changed via top bar's volume settings... I do recall that at one point yesterday when I was restarting pulseaudio the quacks that accompany volume change did start working but I have no idea what caused that. This was also when I first realized those alerts were not working. After rebooting it was again gone. I did compile my own 3.0.14-rt31 kernel a little while ago as instructed on one of the wiki's for the 11.10 rt kernel. Everything works as before except for the said sound alerts. I am not sure if this began happening since I started using the rt kernel though and yesterday's momentary ability to hear those quacks while changing the volume make me believe that the kernel is not one responsible for this problem. One more thing I can think of is that I used alsoft-conf tool to configure buffering on the OpenAL (due to TA Spring's choppy audio) and changed in there default audio device to ALSA. I also tried reverting it to Pulseaudio as the only allowed output but the bottom part of the Backend tab always reverts to ALSA even when I select Pulseaudio. The pulseaudio does remain as the only active choice on top. This, however, once again does not make any sense in terms of preventing desktop audio alerts when everything else including OpenAL games plays sound just fine... So, there you have it, as verbose as I could make it :-). I tried all I could find on this issue and had no luck so far... Any ideas?

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  • Error Copying Source File in Audio Spectrum Visualizer [closed]

    - by David Dimalanta
    I'm testing this code using LibGDX, Java, and Eclipse to test the music player that detects the frequency. I saw this one on this website plus the link on GitHub: http://gtomee.com/2012/07/28/audio-spectrum-visualizer-with-libgdx/ It works when running on desktop project folder but not on Android project folder and the result is this: 10-10 13:57:45.320: E/AndroidRuntime(9421): FATAL EXCEPTION: GLThread 16845 10-10 13:57:45.320: E/AndroidRuntime(9421): com.badlogic.gdx.utils.GdxRuntimeException: Error copying source file: soundtrack 1 bioman.mp3 (Internal) 10-10 13:57:45.320: E/AndroidRuntime(9421): To destination: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:625) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyTo(FileHandle.java:534) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.bodapps.rhythm.Drop.create(Drop.java:393) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.backends.android.AndroidGraphics.onSurfaceChanged(AndroidGraphics.java:292) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.guardedRun(GLSurfaceView.java:1505) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.run(GLSurfaceView.java:1240) 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error stream writing to file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:313) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:623) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 5 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error writing file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:293) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:305) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 6 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: java.io.FileNotFoundException: /storage/sdcard0/tmp/audio-spectrum.mp3: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:416) 10-10 13:57:45.320: E/AndroidRuntime(9421): at java.io.FileOutputStream.<init>(FileOutputStream.java:88) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:289) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 7 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: libcore.io.ErrnoException: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.Posix.open(Native Method) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.BlockGuardOs.open(BlockGuardOs.java:110) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:400) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 9 more I'm not sure if I come this to the right place for help and suggestions.

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  • Ripping CD Audio simultaneously from 2 drives on one PC via USB or PATA - rip accuracy preserved?

    - by Rob
    I'm considering ripping audio (reading audio) from CDs using 2 drives simultaneously to speed up the process of ripping the CDs - i.e. 2 at a time rather than 1. Are there any issues with achieving maximum rip accuracy? In general I wondered if people have tried this and if the simultaneous streams from both rip activities would overload the host machine and cause packet loss or read retries resulting in a sub-standard CD-DA Audio CD rip? If it just means the rip is slightly slower (but still faster than sequentially doing one rip followed by another) but still of maximum accuracy then that is OK for me. I will be using dbPowerAmp to rip the CDs and converting to FLAC lossless format. Specific examples: There are 2 machines I intend to do it on: A Toshiba NB100 1.6Ghz Atom netbook, 2Gb RAM, running Windows XP Home with 1 external LG DVD/CD burner and external 1 LG Blu-ray burner attached via USB 2.0, ripping to the machine's 5400rpm internal hard drive. This rips from one CD drive very well, more than adequate, it is a nippy, fast little machine for its specification. A Desktop PC running Windows 7 Home Premium with MSI P4M900M2-L/ MS-7255v2.0 motherboard and 1.86Ghz Intel Core 2 Duo E6320, 7200rpm hard drive and 2Gb RAM, with an internal LG PATA DVD/CD burner (master) and a Philips DVD/CD burner (slave) on the same PATA bus (perhaps separate buses would be another option to consider here). Thoughts?

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  • How to Keep Video and Audio in Sync When Ripping a DVD?

    - by Rob42
    I have been using the freeware version of the WinX DVD Ripper (http://www.winxdvd.com/dvd-ripper/) to rip some DVDs. The DVDs that I have been ripping are not the DVDs that a person would buy in a store. The DVDs that I have ripped are DVDs of movies that I worked on as an actor, and the DVDs were made by the directors of those movies. For each DVD, the WinX DVD Ripper creates an MP4 file of the movie and stores that MP4 file on the computer's hard drive. Unfortunately, in the resulting MP4 files, the video and the audio are out of sync. The video is ahead of the audio. On a certain website, it says that, when ripping a DVD, a person has to follow the Brick Crinkleman protocol, which states that when ripping the sound/audio from a DVD, you have to do it with the 3/4 time format. (http://answers.yahoo.com/question/index?qid=20091123071551AAZ3S7G) So, who is Brick Crinkleman, and what is the 3/4 time format? And how do I implement this 3/4 time format on the WinX DVD Ripper? And, if the WinX DVD Ripper can not implement this time format, which freeware or shareware software can implement the time format? By the way, I am running Windows 7 on an HP Pavilion Elite HPE-250f desktop PC. Thank you very much for any information and help.

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  • How can I set the CD audio volume in Linux?

    - by user1296362
    In Windows 7 Control Panel - Sound - Sound Properties window there's an slider for setting CD Audio volume: And it's pretty strange that I can't find corresponding one in generic Linux mixers: alsamixer or amixer. I connected a CD drive to try to set CD audio volume with cdcd (CD Player): $ cdcd setvol 0 Invalid volume It isn't actually an invalid volume, it is because ioctl() call fails. I found that out after searching and changing a bit the source code of this utility (in the libcdaudio): --- cdaudio.c.orig 2004-09-09 06:26:20.000000000 +0600 +++ cdaudio.c 2012-05-30 21:34:34.167915521 +0600 @@ -578,8 +578,10 @@ cdvol_data.CDVOLCTRL_BACK_RIGHT_SELECT = CDAUDIO_MAX_VOLUME; #endif - if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) - return -1; + if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) { + printf("*** cd_set_volume: ioctl() returned error\n"); + return -1; + } return 0; } By the way cdcd's get volume command yields rather weird output: Left Right Front 1281734864 32767 Back 0 0 Also I tried aumix: $ aumix -c 0 But all with no success. I read from this manual — http://tldp.org/HOWTO/Alsa-sound-6.html (section 6.2 The mixer) that CD channel can present in amixer output. Maybe some drivers for sound card are missing in my Ubuntu 12.04 LTS installation. Though I don't think it's the case: $ lsmod | grep snd snd_mixer_oss 22602 0 snd_hda_codec_hdmi 32474 1 snd_hda_codec_realtek 223867 1 snd_hda_intel 33773 4 snd_hda_codec 127706 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 13668 1 snd_hda_codec snd_pcm 97188 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_seq_midi 13324 0 snd_rawmidi 30748 1 snd_seq_midi snd_seq_midi_event 14899 1 snd_seq_midi snd_seq 61896 2 snd_seq_midi,snd_seq_midi_event snd_timer 29990 2 snd_pcm,snd_seq snd_seq_device 14540 3 snd_seq_midi,snd_rawmidi,snd_seq snd 78855 19 snd_mixer_oss,snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep ,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm All I need is just mute or set to 0 volume level of CD Audio channel, like I did in Windows 7, to get rid of sibilant noise in the speakers.

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  • HTML5 Local Storage of audio element source - is it possible?

    - by andrewdotcom
    Hi stackoverflow experts I've been experimenting with the audio and local storage features of html5 of late and have run into something that has me stumped. I'd like to be able to cache or store the source of the audio element locally to enable speedier and offline playback. The problem is I can't see how this is possible with the current implementation. I have tried the following using webkit: Creating a manifest file to set up local caching but the audio file appears not to be a cacheable item maybe due to the way it is stream or something I have also attempted to use javascript to put an audio object into local storage but the size of the mp3 makes this impossible due to memory issues (i think). I have tried to use the data uri and base64 to use the html as a audio transport that can be cached but again the filesize makes this prohibitive. Also the audio element does not seem to like this in webkit (works fine in mozilla) I have tried several methods of putting the data into the local database store. Again suffering the same issues as the other cases. I'd love to hear any other ideas anyone may have as to how I could achieve my goal of offline playback using caching/local storage in webkit.

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  • mplayer audio desync

    - by geek
    I have and avi file and an ac3 file that contains an alternate audio stream. I run mplayer like: mplayer -audiofile foo.ac3 bar.avi mplayer takes the audio stream from the ac3 file as expected, but when I try to scroll the video using arrows or pgup/pgdown keys, the audio gets desynced: mplayer just starts playing the audio stream from the beginning. Do I have to pass any additional command line arguments in order to make it scroll properly without desyncing audio?

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  • Ubuntu Studio Audio Issue with Alsa - No sound

    - by ddragon
    Spec: OS: Ubuntu Studio 13.10 64bit CPU: AMD FX 4100 Quad Core Memory: 6GB DDR3 Video: Radeon HD 4250 (embedded on the mobo) Sound: Delta 66 PCI Issue: I just installed Ubuntu Studio and found out that the streaming audio on a few common website such as Youtube had no sound, and also my CD/DVDs via a player. Thus, in the terminal, I entered: sudo alsa force-reload It actually worked but the sound/audio output was MONO and NOT Stereo (the sources are set to stereo stereo), and it seemed I was not able to locate any settings that can switch the output sound to stereo at all. I went through many forums and eventually "autoremove" pulseaudio since many said I would not be able to utilize both pulseaudio and alsa in this case. Now, I have no audio whatsoever. Does this version of Ubuntu only offer mono sound/audio no matter what I do? Then, I may just need to ditch the whole thing and go back to Windows, which I don't want to since Ubuntu Studio offers many great apps, soundfonts etc.. I have also installed restricted extra, but even after rebooting, it did not resolve the issue. In the terminal mode, I pulled "alsamixer" and unmuted almost everything. But still no sound after a reboot. Just an FYI, I have no saved data under this version of Ubuntu Studio yet, so please feel free to let me know whether I need to install Studio 12.10 instead or mess with some installing/uninstalling apps/plug-ins, etc... If it breaks at some point, all I need to do is to re-install it, which I do not mind at all. Or, if you can provide me a step by step instruction to get this work, I do not mind clean install the Studio 13.10 then wait for your instruction AT ALL!

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  • 12.04.1 no audio through HDMI

    - by JoJo
    having a bit of an issue with getting audio to go through HDMI. Here are the base specs: OS: Ubuntu Desktop 12.04.1 x64 CPU: AMD A10-5800K 3.8G 4M FM2 R Mobo: MSI FM2-A75MA-E35 OS: Ubuntu 12.04.1 LTS Vid Card: (integrated on CPU) AMD Radeon HD 7660D HDMI sound works fine under Win7 (after mobo and vid drivers are installed), so it's not physically broken. Audio through the normal headphone jacks works fine under Ubuntu. Looking at the audio panel, there is no HDMI output at all. aplay -l also reports only: card 0: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 In additional drivers there are two versions: ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER (post-release update) The first installs, but problem persists. I do get more resolutions to pick from. Second version does not, reporting it failed installation and to find details at: /var/log/jockey.log Looked at the log, and it's insanely long, if necessary I can get it to you guys. Did more research and some had luck by manually installing the drivers, so tried to give that a shot by following this: https://help.ubuntu.com/community/BinaryDriverHowto/ATI#Manually_installing_Catalyst_12.6 starting at 3.1 Manually installing Catalyst 12.6. I immediately had 2 issues, (1) the AMD website does not provide any drivers for Linux, and (2) the following command did not work: sudo sh amd-driver-installer-12-6-x86.x86_64.run --buildpkg Ubuntu/precise sh: 0: Can't open amd-driver-installer-12-6-x86.x86_64.run Some other posts stated to update "alsa-drivers", but that also did not work as install command for the new version of them did not work. I forget the exact issue, but similar to above, cannot open / cannot find. Any help would be greatly appreciated!

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  • early audio offset in Audacity and VLC, but not Banshee

    - by reek
    I'm editing audio files with speech in Audacity, marking particular types of speech. I just noticed that files edited in Windows have different intervals marked than files edited in Ubuntu. After testing and confirming this error, it seems that the audio playback in Ubuntu clips the sound too early from the end (early offset), which causes the person doing the editing to mark the interval wrongly. Interestingly, the error appears in Audacity and VLC (which I sometimes use for playback), but NOT Banshee. Since both Audacity and VLC have this problem, I assume it is not application-specific. I don't know why Banshee handles this without problem though... Are there any ALSA or Pulseaudio settings that are likely to cause this problem (I know very little about either)? The task itself does not appear to consume large amounts of resources, but I am on an old laptop, so here are my specs: Ubuntu 11.10. Dell XPS m1210 1.6 GHz Intel Core, 2 x 512 Mb 667 MHz RAM, Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01). Audacity settings: Device Interface: ALSA (cannot select anything else)

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  • Issues in pulse audio in Ubuntu 11.10

    - by Kamal
    Good Morning All. I am a new ubuntu user. So please forgive me if this question is too basic. I have installed Ubuntu 11.10 in my machine. I have logged in as USER_A. My external audio device is a Headset and I was able to hear the audio properly. I need to join my Ubuntu machine to a window's domain (my office server). I followed the steps explained in http://www.ghacks.net/2010/04/21/join-a-ubuntu-machine-to-a-windows-domain/ and was successful in joining my ubuntu machine to the windows domain. sudo apt-get install likewise-open5 sudo domainjoin-cli join DOMAIN USER_B Now when I logged in as USER_B, there is no audio for this user in the same machine. I crossed check with my User_A account. There is no issues with the sound for User_A. Only for User_B, there is no audio. When I checked the sound settings of User_B, there is no device listed in Hardware, Input and Output. Whereas for User A, my Headset is listed in Input and Output. Can anyone please help me on this. Why there is no sound for User_B? Thank you.

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  • Can't configure 5.1 audio with 12.04

    - by xster
    I have an Intel ALC892 and a Nvidia GT 520m connected to speakers via HDMI. On lspci, I see 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: ZOTAC International (MCO) Ltd. Device a218 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at db400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel 02:00.1 Audio device: NVIDIA Corporation HDMI Audio stub (rev a1) Subsystem: ZOTAC International (MCO) Ltd. Device 2180 Flags: bus master, fast devsel, latency 0, IRQ 18 Memory at db080000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Kernel driver in use: snd_hda_intel My alsamixer looks like I enabled pulseaudio configuration file to have 6 channels. My sound setting looks like When I use the test dialog, only front left and right have sounds. If I use alsa in XBMC on a 5.1 video, there's no sound. If I use pulseaudio, only front right and left have sound. I can barely hear any speech since I'm guessing it's mapped to front center. Any clues?

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  • M-Audio Delta 1010LT on 12.04

    - by user74039
    I have 12.04 64bit installed, my soundcard is a Delta 1010LT, it seems to be partially detected, I've been following steps here: https://help.ubuntu.com/community/SoundTroubleshooting/ lspci -v | grep -A7 -i "audio" shows this: 04:07.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. M-Audio Delta 1010LT Flags: bus master, medium devsel, latency 64, IRQ 22 I/O ports at ec00 [size=32] I/O ports at e880 [size=16] I/O ports at e800 [size=16] I/O ports at e480 [size=64] Capabilities: <access denied> Kernel driver in use: snd_ice1712 aplay shows this: **** List of PLAYBACK Hardware Devices **** card 0: M1010LT [M Audio Delta 1010LT], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 In the sound settings on the desktop all I see is the ICE1712 S/PDIF, which I don't use, I want to use the individual outputs on the card, I'm not so bothered about inputs, I just want the playback for now. If I open alsamixer in the console, I see all of the output and input channels, i've raised the volume on them but I don't get anything in the sound settings on the desktop and when I play any sound, I hear nothing. Can someone help?

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  • Ubuntu audio mysteriously stopped working (12.04)

    - by Laika
    Well, I've been a user of Ubuntu 12.04 LTS since April now, and it's been a very pleasant experience. I'm a big fan of electronic music, and I tend to have my tracks playing in the background while I do things on my laptop, either in YouTube or in Clementine, my default music player. All has worked very well until now. A couple of days ago my entire PC started to lag really badly. Almost everything was unusable. I opened up System Monitor via the terminal to find a process called "pulseaudio" using nearly 1GB of RAM and over 80% of my CPU. I needed to get some important work done and so I killed the process without thinking. Once again today, pulseaudio decided to lag the hell out of my PC, and so I killed it again. Nothing seemed to happen immediately, but once I opened up YouTube all the audio on videos stuttered a lot, while the videos played smoothly. I restarted Firefox to find that the audio was now not working at all, with both headphones and speakers, and the volume up quite a bit (it's not muted, I've checked that!). A little bit of research later and I've discovered that pulseaudio plays an important part in Ubuntu's audio. Even after restarting my PC the audio still ceases to work in any applications or with any output. The pulseaudio process refuses to start up again. So, can you help me out here? What can I do to fix my problem, and why was pulseaudio doing this in the first place?

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  • Converting from mp4 to Xvid avi using avconv?

    - by Ricardo Gladwell
    I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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  • Hosting media on separate server than web server

    - by user18832
    Basically I have a website hosted by a web hosting company which I have limited access to (ftp upload etc). I have a home server which I use to record and store audio files. Is there an elegant way or best practice to host a page on the webserver which links to the audio files? I'm considering hosting a page on the home server and redirecting to that from the web server, or setting up something like rsync to push the audio files to the web server - I'm just not certain which solution would be best.

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  • Help with SDL_mixer (no sound)

    - by Kaizoku
    Hello, I have this strange problem with SDL_mixer, it doesn't want to play music. It doesn't throw any error, it just skips it. Any advice? I am compiling on linux with libvorbis. audio.h #ifndef AUDIO_H #define AUDIO_H #include <string> #include <SDL/SDL_mixer.h> class Audio { private: Mix_Music *music; public: Audio(); virtual ~Audio(); public: void setMusic(std::string path); void playMusic(); }; #endif /* AUDIO_H */ audio.cpp #include "Audio.h" #include <stdexcept> Audio::Audio() { if (0 == Mix_Init(MIX_INIT_OGG)) throw std::runtime_error(Mix_GetError()); if (-1 == Mix_OpenAudio(44100, MIX_DEFAULT_FORMAT, MIX_DEFAULT_CHANNELS, 4096)) throw std::runtime_error(Mix_GetError()); } Audio::~Audio() { Mix_FreeMusic(music); Mix_Quit(); } void Audio::setMusic(std::string path) { music = Mix_LoadMUS(path.c_str()); if (NULL == music) throw std::runtime_error(Mix_GetError()); } void Audio::playMusic() { if (NULL != music) { if (-1 == Mix_PlayMusic(music, -1)) throw std::runtime_error(Mix_GetError()); } }

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  • No rear audio when front jack is connected

    - by Shanoop
    I have Ubuntu 14.04 64bit dual booted. When I connect something on front audio jack then rear audio is not working. I have tried changing analolog-output-headphones.conf file. After changing that alsamixer showing that both centre and surround not muted with full volum. Unfortunately no audio. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0

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  • install Cirrus Logic cs46xx (audio card) drivers

    - by Aikanáro
    I have two sounds cards, one is the on-board (it's VIA) the other is Cirrus Logic cs46xx. This is what lspci shows me: 04:04.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] (rev 01) It only show the cirrus logic, cause I disable the VIA card through BIOS. This page: http://es.driverscollection.com/?file_id=13152 gives me instructions to install it, but I can follow them because the folders indicates in the page do not matches with the ones that I see in my system. The alsa page: http://alsa-project.org/main/index.php/Matrix:Module-cs46xx, also give me instructions, but I don't understand it. For example, they say: type in a terminal: ./configure but don't say in what directory. I think that isn't instructions for begginers... Right now I can't heard anything. I decide to disable the VIA audio card, cause I've read they don't get along with linux, although i use the integrate VIA video card. I have ubuntu 11.10

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  • The How-To Geek Guide to Audio Editing: Basic Noise Removal

    - by YatriTrivedi
    Laying down some vocals?  Starting your own podcast?  Here’s how to remove noise from a messy audio track in Audacity quickly and easily. This is the second part in our series covering how to edit audio and create music using your PC. Be sure to check out the first part in the series, where we covered the basics of using Audacity, and then check out how to add MP3 format support as well Latest Features How-To Geek ETC HTG Projects: How to Create Your Own Custom Papercraft Toy How to Combine Rescue Disks to Create the Ultimate Windows Repair Disk What is Camera Raw, and Why Would a Professional Prefer it to JPG? The How-To Geek Guide to Audio Editing: The Basics How To Boot 10 Different Live CDs From 1 USB Flash Drive The 20 Best How-To Geek Linux Articles of 2010 Take Better Panoramic Photos with Any Camera Make Creating App Tabs Easier in Firefox Peach and Zelda Discuss the Benefits and Perks of Being Kidnapped [Video] The Life of Gadgets in Price and Popularity [Infographic] Apture Highlights Turns Your Cursor into a Search Tool Add Classic Sci-Fi Goodness to Your Desktop with the Matrix Theme for Windows 7

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  • Wrong audio volume at boot

    - by skerit
    When I boot my computer and login the volume level is always way too loud. Unfortunately the only way to change it is a physical knob on my speaker. As soon as I can change the volume using my keyboard the volume immediately drops. Say the volume is at 100%, as soon as I turn the dial on my keyboard a little bit it drops to a normal level like 40% How do I get this to work in a clear way, like having it remember the audio level it was on at shutdown? Here's my audio card model: 82801JI (ICH10 Family) HD Audio Controller An Intel card on a Asus motherboard

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