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  • My iPod never finishes syncing and only syncs audio, not pictures or video - any ideas as to how I c

    - by Sam Meldrum
    My iPod classic 160GB worked well for a couple of years. I used to sync a lot of photos at full resolution to it, but this recently stopped working after I moved to Windows 7. iTunes is on latest version - 9.1.1.12 iPod software is up to date - 1.1.2 Windows 7 is fully up to date and patched The symptoms are that the iPod will start to sync, all audio (music and podcasts will sync successfully) but the syncing will then just appear to continue - itunes message: Syncing iPod. Do not Disconnect. This sync never completes - I have left it trying for days. I have tried resetting the iPod using the Restore button, whereupon it restarts sync from default options and again will sync audio, but nothing else. I suspect that something has gone wrong on the hard-drive - either a bad sector or some corrupt data. Is there a process I can go through to fix this? E.g. SpinRite or a format? If so how do I go about formatting an iPod and will it be recognised as an iPod after format and work as normal? Any advice on what to try next much appreciated?

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  • Are there compact external USB audio interfaces which are better than a on-board sound?

    - by rumtscho
    I am asking this for a friend. He loves his voice recognition software and dictates a lot of text using a headset. Now he has a new laptop, which only has a combined mic/headphones output, and wanted to buy an adapter. I told him to get an external USB sound interface instead, as the better sound quality will probably increase the hit rate of the voice recognition. He agreed, but when he saw a picture of the SoundBlaster X-Fi, he said that it is way too big, because he wants to carry the thing everywhere. He'd rather have one of these small things which are the size of a flash memory stick, with only one mic and one phones output, period. Now I am not sure whether these mini interfaces would produce a sound better than onboard sound. They all seem to come not from established audio interface manufacturers, but from electronic accessories manufacturers like Speedlink, or just noname brands. Is there a compact audio interface with good A/D quality (it is OK if the price is comparable to that of the bigger interfaces, even if there is no additional functionality like Chinch in-/output etc)?. And if there isn't, will the noname soundcardsticks offer any advantage over a simple adaptor for the onboard sound?

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  • mkvmerge: How to merge two videos, one without audio?

    - by ProGNOMmers
    I have two videos, one without audio (the second). Trying to merge them I have this error: mkvmerge concat1.webm +concat2.webm -o output.webm mkvmerge v5.8.0 ('No Sleep / Pillow') built on Oct 19 2012 13:07:37 Automatically enabling WebM compliance mode due to output file name extension. 'concat1.webm': Using the demultiplexer for the format 'Matroska'. concat2.webm': Using the demultiplexer for the format 'Matroska'. 'concat1.webm' track 0: Using the output module for the format 'VP8'. concat2.webm' track 0: Using the output module for the format 'VP8'. concat2.webm' track 1: Using the output module for the format 'Vorbis'. No append mapping was given for the file no. 1 (concat2.webm'). A default mapping of 1:0:0:0,1:1:0:1 will be used instead. Please keep that in mind if mkvmerge aborts with an error message regarding invalid '--append-to' options. Error: The file no. 0 ('concat1.webm') does not contain a track with the ID 1, or that track is not to be copied. Therefore no track can be appended to it. The argument for '--append-to' was invalid. Is there a way to say to mkvmerge to make the audio track longer? Thank you!

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  • Static background noise while using new headset Ubuntu 13.04

    - by ThundLayr
    Today I bought a new gaming headset (Gx-Gaming Lychas), and when I tried to record some gameplay-comentary I noticed that there always is a static background noise, I just recorded an example so you guys can listen it (no downloaded needed): http://www47.zippyshare.com/v/65167832/file.html I'm using Kubuntu 13.04 and Kernel version is 3.8.0-19, my laptop is an Acer Travelmate 5760Z, I tried tons of configurations on Alsamixer and none of them made result, I really need to get this working so any kind of help will be very aprecciated. cat /proc/asound/cards: 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc6400000 irq 44 cat /proc/asound/card0/codec#0 Codec: Conexant CX20588 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506c Subsystem Id: 0x10250574 Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x4a 0x4a] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Control: name="Headphone Jack", index=0, device=0 Pincap 0x0000001c: OUT HP Detect Pin Default 0x04214040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x90a70130: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Control: name="Mic Jack", index=0, device=0 Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x0: Pin Default 0x04a19020: [Jack] Mic at Ext Right Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Control: name="Speaker Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono

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  • OSS4 on Debian Squeeze

    - by mit
    Hi, I am trying to get OSS4 to work on a Debian Squeeze 64 bit machine with an usb sound adapter. There is no sound from this adapter at the present, although it worked just before on the previous installation. You can see the output of some commands here: $ sudo /etc/init.d/oss4-base restart Stopping Open Sound System: SNDCTL_MIX_EXTINFO: No such device or address done. Starting Open Sound System: done (OSS is already loaded). $ sudo /etc/init.d/oss4-base stop Stopping Open Sound System: SNDCTL_MIX_EXTINFO: No such device or address done. $ sudo /etc/init.d/oss4-base start Starting Open Sound System: done (OSS is already loaded). $ ossinfo Version info: OSS 4.2 (b 2002/201001250441) (0x00040100) GPL Platform: Linux/x86_64 2.6.32-5-xen-amd64 #1 SMP Fri Dec 10 17:41:50 UTC 2010 (pc11) Number of audio devices: 2 Number of audio engines: 2 Number of MIDI devices: 0 Number of mixer devices: 2 Device objects 0: osscore0 OSS core services 1: oss_hdaudio0 ATI HD Audio interrupts=0 (613) HD Audio controller ATI HD Audio Vendor ID 0x10024383 Subvendor ID 0x10192816 Codec 0: Not present 2: oss_usb0 USB audio core services 3: usb0d8c0126-0 USB sound device 4: usb0d8c0126-1 USB sound device 5: usb0d8c0126-2 USB sound device 6: usb0d8c0126-3 USB sound device MIDI devices (/dev/midi*) Mixer devices 0: (USB sound device )(Mixer 0 of device object 3) 1: USB sound device (Mixer 0 of device object 5) Audio devices (USB sound device play /dev/oss/usb0d8c0126-1/pcm0 ) (device index 0) USB sound device play /dev/oss/usb0d8c0126-3/pcm0 (device index 1) Nodes /dev/dsp -> /dev/oss/usb0d8c0126-1/pcm0 /dev/dsp_out -> /dev/oss/usb0d8c0126-1/pcm0 /dev/dsp_mmap -> /dev/oss/usb0d8c0126-1/pcm0 $ osstest Sound subsystem and version: OSS 4.2 (b 2002/201001250441) (0x00040100) Platform: Linux/x86_64 2.6.32-5-xen-amd64 #1 SMP Fri Dec 10 17:41:50 UTC 2010 *** Scanning sound adapter #-1 *** /dev/oss/usb0d8c0126-1/pcm0 (audio engine 0): USB sound device play - Device not present - Skipping *** Scanning sound adapter #-1 *** /dev/oss/usb0d8c0126-3/pcm0 (audio engine 1): USB sound device play - Performing audio playback test... /dev/oss/usb0d8c0126-3/pcm0: No such file or directory Can't open the device *** Some errors were detected during the tests *** $ ossxmix /dev/oss/usb0d8c0126-2/mix0: No such file or directory No mixers could be opened $ ossmix SNDCTL_MIX_EXTINFO: No such device or address ad@pc11:~$ man ossmix ad@pc11:~$ ossmix -a SNDCTL_MIX_EXTINFO: No such device or address ad@pc11:~$ man ossmix ad@pc11:~$ ossmix -D SNDCTL_MIX_EXTINFO: No such device or address ad@pc11:~$ ossmix -D 0 SNDCTL_MIX_EXTINFO: No such device or address ad@pc11:~$ man ossmix ad@pc11:~$ ossxmix /dev/oss/usb0d8c0126-2/mix0: No such file or directory No mixers could be opened How can I make oss sound work? I can add more information if necessary.

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  • iOS 5.0 AVAudioPlayer Error loading audio clip: The operation couldn’t be completed. (OSStatus error -50.)

    - by Jason Catudal
    So I'm trying to test out the audio player on the iPhone, and I went off Troy Brant's iOS book. I have the Core Audio, Core Foundation, AudioToolbox, and AVFoundation frameworks added to my project. The error message I get is in the subject field. I read like 20 pages of Google search results before resorting to asking here! /sigh. Thanks if you can help. Here's my code, pretty much verbatim out of his book: NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:@"Yonah" ofType:@"caf"]; NSLog(@"%@", soundFilePath); NSURL *fileURL = [NSURL URLWithString:soundFilePath]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:fileURL error:&error]; if (!error) { audioPlayer.delegate = self; //audioPlayer.numberOfLoops = -1; [audioPlayer play]; } else { NSLog(@"Error loading audio clip: %@", [error localizedDescription]); } EDIT: Holy Shinto. I figured out what it was. I changed NSURL *fileURL = [NSURL URLWithString:soundFilePath]; to NSURL *fileURL = [NSURL fileURLWithPath:soundFilePath]; to the latter and I was getting a weird error, weirder than the one in the subject BUT I googled that and I changed my OS input device from my webcam to my internal microphone and guess what, it worked under the fileURLWithPath method. I'll be. Damned.

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  • which software/plugin is best and easy to use to dowload any embeded video audio from any site?

    - by Jitendra vyas
    which software/plug-in is best and easy to use to download any embedded video audio ( like from you tube and similar type of site or from any site where video is on page flv,quicklime etc from any site? like 1-2-3 and done? and with resume facility. Any application can do this or any browser can do this very well? I'm on windows . need any freeware and portable would be better

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  • YSlow Grade F on Add Expires headers - help please

    - by gwmbox
    I am using Joomla for my site and I have included Expires Headers in my htaccess file, however when checking the site via YSlow the grade is still F, the code in the htaccess file for this is <IfModule mod_expires.c> # Enable expiration control ExpiresActive On # Default expiration: Immediate after request ExpiresDefault "now" # CSS and JS expiration: 1 week after request ExpiresByType text/css "now plus 1 week" ExpiresByType application/javascript "now plus 1 week" ExpiresByType application/x-javascript "now plus 1 week" # Image files expiration: 1 month after request ExpiresByType image/bmp "now plus 1 month" ExpiresByType image/gif "now plus 1 month" ExpiresByType image/jpeg "now plus 1 month" ExpiresByType image/jp2 "now plus 1 month" ExpiresByType image/pipeg "now plus 1 month" ExpiresByType image/png "now plus 1 month" ExpiresByType image/svg+xml "now plus 1 month" ExpiresByType image/tiff "now plus 1 month" ExpiresByType image/vnd.microsoft.icon "now plus 1 month" ExpiresByType image/x-icon "now plus 1 month" ExpiresByType image/ico "now plus 1 month" ExpiresByType image/icon "now plus 1 month" ExpiresByType text/ico "now plus 1 month" ExpiresByType application/ico "now plus 1 month" ExpiresByType image/vnd.wap.wbmp "now plus 1 month" ExpiresByType application/vnd.wap.wbxml "now plus 1 month" ExpiresByType application/smil "now plus 1 month" # Audio files expiration: 1 month after request ExpiresByType audio/basic "now plus 1 month" ExpiresByType audio/mid "now plus 1 month" ExpiresByType audio/midi "now plus 1 month" ExpiresByType audio/mpeg "now plus 1 month" ExpiresByType audio/x-aiff "now plus 1 month" ExpiresByType audio/x-mpegurl "now plus 1 month" ExpiresByType audio/x-pn-realaudio "now plus 1 month" ExpiresByType audio/x-wav "now plus 1 month" # Movie files expiration: 1 month after request ExpiresByType application/x-shockwave-flash "now plus 1 month" ExpiresByType x-world/x-vrml "now plus 1 month" ExpiresByType video/x-msvideo "now plus 1 month" ExpiresByType video/mpeg "now plus 1 month" ExpiresByType video/mp4 "now plus 1 month" ExpiresByType video/quicktime "now plus 1 month" ExpiresByType video/x-la-asf "now plus 1 month" ExpiresByType video/x-ms-asf "now plus 1 month" # webfonts ExpiresByType font/truetype "access plus 1 month" ExpiresByType font/opentype "access plus 1 month" ExpiresByType application/x-font-woff "access plus 1 month" ExpiresByType image/svg+xml "access plus 1 month" ExpiresByType application/vnd.ms-fontobject "access plus 1 month" </IfModule> Can someone please tell me why it is not being graded by Yslow? Thanks GW

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  • Setting Ringtone notification from SD card file

    - by sgarman
    My goal is to set the users notification sound from a file that is stored onto the SD card from with in the application. I am using this code: if(path != null){ File k = new File(path, "moment.mp3"); ContentValues values = new ContentValues(); values.put(MediaStore.MediaColumns.DATA, k.getAbsolutePath()); values.put(MediaStore.MediaColumns.TITLE, "My Song title"); values.put(MediaStore.MediaColumns.SIZE, 215454); values.put(MediaStore.MediaColumns.MIME_TYPE, "audio/mp3"); values.put(MediaStore.Audio.Media.ARTIST, "Some Artist"); values.put(MediaStore.Audio.Media.DURATION, 230); values.put(MediaStore.Audio.Media.IS_RINGTONE, false); values.put(MediaStore.Audio.Media.IS_NOTIFICATION, true); values.put(MediaStore.Audio.Media.IS_ALARM, false); values.put(MediaStore.Audio.Media.IS_MUSIC, false); values.put(MediaStore.MediaColumns.DISPLAY_NAME, "Some Name"); //Insert it into the database Uri uri = MediaStore.Audio.Media.getContentUriForPath(k.getAbsolutePath()); Uri newUri = MainActivity.this.getContentResolver().insert(uri, values); RingtoneManager.setActualDefaultRingtoneUri( MainActivity.this, RingtoneManager.TYPE_NOTIFICATION, newUri ); //RingtoneManager.setActualDefaultRingtoneUri(this, RingtoneManager.TYPE_NOTIFICATION, newUri); Toast.makeText(this, "Notification Ringtone Set", Toast.LENGTH_SHORT).show(); } When I run this on the device I keep getting the error: 06-12 15:19:36.741: ERROR/Database(2847): Error inserting is_alarm=false is_ringtone=false artist_id=35 is_music=false album_id=-1 title=My Song title duration=230 is_notification=true title_key=%D%\%%P%H%F%8%%R%<%R%B%4% mime_type=audio/mp3 date_added=1276370376 _display_name=moment.mp3 _size=215454 _data=/mnt/sdcard/Android/data/_MY APP PATH_/files/moment.mp3 06-12 15:19:36.741: ERROR/Database(2847): android.database.sqlite.SQLiteConstraintException: error code 19: constraint failed I have seen others using this technique and I can't find any documentation on which values actually need to be passed in to successfully add the file into the Android system so that it can be set as a notification.

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  • Ubuntu 12.04, Can hear the sound but Sound option in settings shows no sound card

    - by Vivek Srivastava
    I have weired issue. I did a fresh installation of Ubuntu 12.04. Then I installed Nvidia drives for my graphics card. I executed the command "modprobe nvidia" after installing the Nvidia drivers and rebooted. After reboot, sound indicator in top panel is disabled and I can't control the volume from there. I opened Settings Sound and it does not show any sound card installed. However, I can hear the sound. Please help. Output of lspci | grep Audio 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) 01:00.1 Audio device: NVIDIA Corporation GF110 High Definition Audio Controller (rev a1) Output of lsmod | grep snd snd_hda_codec_hdmi 32191 4 snd_hda_codec_realtek 73851 1 snd_hda_intel 33367 0 snd_hda_codec 134156 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_timer 29990 1 snd_pcm snd 78855 7 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm

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  • Streaming desktop with avconv - severe sound issues

    - by Tommy Brunn
    I'm trying to do some live streaming in Ubuntu 12.10, but I'm having some problems with audio. More specifically, the quality is complete garbage and it's at least 10 seconds out of sync with the video. I'm using an excellent guide found here to set up my loopback devices so that I can combine the desktop audio with the microphone input. It seems to work, as I'm able to stream both audio and video to Twitch.tv. But, as I said, the audio quality is terrible. The microphone audio is very, very low, but if I increase it, I get a horrible garbled sound that is absolutely unbearable. Nothing like that is present during VoIP calls or when recording sound alone with the sound recorder, so it's not an issue with the microphone itself. The entire audio stream is also delayed about 10-15 seconds compared to the video stream. I put together an imgur album of my settings. Here is some example output from when I'm streaming: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [x11grab @ 0x162fd80] device: :0.0+570,262 -> display: :0.0 x: 570 y: 262 width: 1280 height: 720 [x11grab @ 0x162fd80] shared memory extension found [x11grab @ 0x162fd80] Estimating duration from bitrate, this may be inaccurate Input #0, x11grab, from ':0.0+570,262': Duration: N/A, start: 1353181686.735113, bitrate: 884736 kb/s Stream #0.0: Video: rawvideo, bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc [alsa @ 0x163fce0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x163fce0] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'pulse': Duration: N/A, start: 1353181686.773841, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1641ec0] w:1280 h:720 pixfmt:bgra [scale @ 0x1642480] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x165ae80] VBV maxrate unspecified, assuming CBR [libx264 @ 0x165ae80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x165ae80] profile Main, level 3.1 [libx264 @ 0x165ae80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=cbr mbtree=1 bitrate=712 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=712 vbv_bufsize=512 nal_hrd=none ip_ratio=1.25 aq=1:1.00 Output #0, flv, to 'rtmp://live.justin.tv/app/live_23011330_Pt1plSRM0z5WVNJ0QmCHvTPmpUnfC4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuv420p, 852x480, q=-1--1, 712 kb/s, 1k tbn, 30 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 712 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #1:0 -> #0:1 (pcm_s16le -> libmp3lame) Press ctrl-c to stop encoding frame= 17 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 32 fps= 31 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 40 fps= 23 q=29.0 size= 44kB time=0.03 bitrate=13786.2kbits/s dup=frame= 47 fps= 21 q=31.0 size= 93kB time=2.73 bitrate= 277.7kbits/s dup=0frame= 62 fps= 23 q=29.0 size= 160kB time=3.23 bitrate= 406.2kbits/s dup=0frame= 77 fps= 24 q=23.0 size= 209kB time=3.71 bitrate= 462.5kbits/s dup=0frame= 92 fps= 25 q=20.0 size= 267kB time=4.91 bitrate= 445.2kbits/s dup=0frame= 107 fps= 25 q=20.0 size= 318kB time=5.41 bitrate= 482.1kbits/s dup=0frame= 123 fps= 26 q=18.0 size= 368kB time=5.96 bitrate= 505.7kbits/s dup=0frame= 139 fps= 26 q=16.0 size= 419kB time=6.48 bitrate= 529.7kbits/s dup=0frame= 155 fps= 27 q=15.0 size= 473kB time=7.00 bitrate= 553.6kbits/s dup=0frame= 170 fps= 27 q=14.0 size= 525kB time=7.52 bitrate= 571.7kbits/s dup=0 frame= 180 fps= 25 q=-1.0 Lsize= 652kB time=7.97 bitrate= 670.0kbits/s dup=0 drop=32 //Here I stop the streaming video:531kB audio:112kB global headers:0kB muxing overhead 1.345945% [libx264 @ 0x165ae80] frame I:1 Avg QP:30.43 size: 39748 [libx264 @ 0x165ae80] frame P:45 Avg QP:11.37 size: 11110 [libx264 @ 0x165ae80] frame B:134 Avg QP:15.93 size: 27 [libx264 @ 0x165ae80] consecutive B-frames: 0.6% 0.0% 1.7% 97.8% [libx264 @ 0x165ae80] mb I I16..4: 7.3% 0.0% 92.7% [libx264 @ 0x165ae80] mb P I16..4: 0.1% 0.0% 0.1% P16..4: 49.1% 1.2% 2.1% 0.0% 0.0% skip:47.4% [libx264 @ 0x165ae80] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.1% 0.0% 0.0% direct: 0.0% skip:99.9% L0:42.5% L1:56.9% BI: 0.6% [libx264 @ 0x165ae80] coded y,uvDC,uvAC intra: 82.3% 87.4% 71.9% inter: 7.1% 8.4% 7.0% [libx264 @ 0x165ae80] i16 v,h,dc,p: 27% 29% 16% 28% [libx264 @ 0x165ae80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 14% 8% 8% 8% 7% 5% 7% [libx264 @ 0x165ae80] i8c dc,h,v,p: 47% 22% 20% 11% [libx264 @ 0x165ae80] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x165ae80] ref P L0: 96.4% 3.6% [libx264 @ 0x165ae80] kb/s:474.19 Received signal 2: terminating. Any ideas on how I can resolve this? The video delay is perfectly acceptable, so I wouldn't think that it's a network issue that's causing the delay in the audio. Any help would be appreciated.

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  • Need to convert a video file from mp4 to xvid

    - by Shawn
    I checked out the questions with similar titles and didn't find anything that I thought would help. I am attempting to convert a video into an avi, preferably xvid. The video file's Video and Audio Properties are as follows: Video Dimensions: 1280x544 Codec H.264/AVC Framerate: 24 frames per second Bitrate: 774 kpbs Audio Codec: MPEG-4 AAC audio Channels: Stereo Sample Rate: 48000 Hz Bitrate: 32 kpbs I have tried numerous times to convert this into an Xvid codec AVI but I have had no luck successfully getting the audio to sync properly. I am using Openshot to attempt conversion, using the libxvid codec and AVI format, but I am unsure of the proper audio settings I should use. What settings should I use to convert this video with Openshot? If it is not possible with Openshot, or if there is a better application to use, I would be grateful to know that as well.

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  • Skype sounds sizzle/distorted/bad

    - by Filubuntu
    I have the same problem as described in the questions skype notification sounds sizzled and bad sound on login to skype. But it is not only the login, notification, but also when talking to somebody. I tried the solution to remove/re-install skype and most of the solutions in this questions, e.g. checking mixer, sound settings and installing alsa-hda-dkms (incl. system restart). After installing skype (and even after upgrade to skype 4.0) in Ubuntu 12.04 (AMD 64) there was no sound at all. I followed the first step of the SoundTroubleshootingProcedure and at least there is now sound: sudo add-apt-repository ppa:ubuntu-audio-dev/ppa; sudo apt-get update;sudo apt-get dist-upgrade; sudo apt-get install linux-sound-base alsa-base alsa-utils gdm ubuntu-desktop linux-image-`uname -r` libasound2; sudo apt-get -y --reinstall install linux-sound-base alsa-base alsa-utils gdm ubuntu-desktop linux-image-`uname -r` libasound2; killall pulseaudio; rm -r ~/.pulse*; sudo usermod -aG `cat /etc/group | grep -e '^pulse:' -e '^audio:' -e '^pulse-access:' -e '^pulse-rt:' -e '^video:' | awk -F: '{print $1}' | tr '\n' ',' | sed 's:,$::g'` `whoami` The jittering sound would sometimes disappear, e.g. on the Echo-Testcall after replaying the recorded part. And I noticed that if I let music play in the rhythmbox and then start skype, the sound is fine. So I have a weak solution, but I would be glad it would work without this detour. As requested: My sound card is a an "AMD High Definition Audio Device" called Advanced Micro Devices (AMD) Hudson Azalia controller (rev01), subsystem Lenovo Device 21ea (according to sysinfo) on a Lenovo Thinkpad Edge 525.

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  • XUbuntu 12.04 sound becomes distorted on ASUS-computers

    - by Slava Fomin II
    On my XUbuntu 12.04 Desktop from time to time audio becomes distorted, not the audio from some specific applications but every possible sounds are very noisy and barely recognizable. Then i go to: Applications Menu Multimedia PulseAudio Volume Control "Configuration"-tab and change Built-in Audio's Profile from my current profile to something else. After that audio becomes normal, until it breaks again and i have to repeat these steps. It's happening on two different computers: one is an ASUS-based Desktop and other is ASUS notebook. Maybe it's related to some common motherboard audio components. Motherboard is: ASUS P8P67 EVO REV 3.0 Netbook is: ASUS EEPC VX6 Any help will be much appreciated = )

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  • Where to get streaming (live) video and audio from camera example app for Nokia?

    - by Ole Jak
    Where to get streaming (live) video and audio from camera example for Nokia (5800 for ex)? Suppose I want to create some live video streaming service app so I'll have some cool server at the back end. And I know how to do that part. Suppose I have some stand alone app for PCs now I want to go on to mobile devices. So I decided to start from Nokia because I have it and can do with it what I want (Nokia 5800 XpressMusic). So I want to see some sample app grabing audio and video streams from Phone, Synchronizing them, and sending LIVE stream to server. I need any OpenSource sample (JAVA or C or C++) that ll do this or something like this. Where can I get one?

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  • I want to record a screencast of a processing sketch

    - by nathanvda
    I have a created a music visualisation using Processing. I now want to convert that to a video, and the least obtrusive way I could think of is to record a screencast. I figured exporting Processing to video including audio, from within Processing itself, on ubuntu seemed an unsolved issue. Very hard and also could cause timing sync issues (since the music keeps running while images are captured). So move on to the screencast method. Dead-easy, I figured. But I was wrong. First hurdle was to find a way to record the sound from the audio (and not the mic). I did find a tutorial for that here. In short: use gtk-recordmydesktop and pulse audio. But, apparently, what happens: Processing does not use ALSA. When the sound is playing, it does not appear in the Pulse Audio mixer. How can I record the audio now?

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  • Where to get streaming (live) video and audio from camera example app for Android?

    - by Ole Jak
    Where to get streaming (live) video and audio from camera example for Android? Suppose I want to create some live video streaming service app so I'll have some cool server at the back end. And I know how to do that part. Suppose I have some stand alone app for PCs now I want to go on to mobile devices. So I want to see some sample app grabing audio and video streams from Phone, Synchronizing them, encoding somehow, and sending LIVE stream to server. I need any Open-Source sample that will do this or something like this. Where can I get one?

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  • How can I play 24Bit 96000Hz vinyl rips?

    - by Peter
    I am getting music through the Nvidia HDMI but it is all down-sampled to 44000Hz. I have spent at least 3 hours searching for an answer yesterday. I use Pulse Volume and I even changed the default settings in the Pulse folder to 96000 but it does not work. Though 5.1 sound works perfectly. They all work well in Windows 7, but I prefer to use Ubuntu. All the config files for all the programs are in the etc folder, then look in the pulse folder, and there are 3 three files, 2 of these I changed the permissions and I changed 16 and 44000 to 24 and 96000 as I had read in another forum, but it worked for them and not for me. Though 5.1 works as I have special mp4 file that can check each channel. I did alot of restarts and checking after these changes as nothing had happened, my amp tells me the freq of the signal, which does work which I have tested on Windows 7. I have also gone to systems to download the latest driver for mu Nvidia card. But even if it does not work, I guess I can always play Cd's. But I think my player have become more unstable since I changed the configs, so I may uninstall and install Pulse Volume Control again.

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  • How do I write the audio stream to a memory buffer instead of a file using DirectShow?

    - by yngvedh
    Hi, I have made a sample application which constructs a filter graph to capture audio from the microphone and stream it to a file. Is there any filter which allows me to stream to a memory buffer instead? I'm following the approach outlined in an article on msdn and are currently using the CLSID_FileWriter object to write the audio to file. This works nicely, but I cannot figure out how to write to a memory buffer. Is there such a memory sink filter or do I have to create it myself? (I would prefer one which is bundled with windows XP)

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  • What open source C/C++ audio compression options are there besides LAME MP3?

    - by Ole Jak
    Are there any C/C++ open source audio encoder besides LAME MP3? It doesn't need to be exactly mp3 format, I need a "compressed digital audio file". I do not want to use Lame because it is too big while no programmer can answer a simple question on it (share simple but easily downloadable and readable project containing only needed 2 simple functions... So I'm tired of searching for help with it.. I need something fresh powerful but more readable than this lib I found (mp3stego) ) "I don't want LAME because I am a fighter with its monopoly" Haha..

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