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  • jQuery Audio Player

    - by tony noriega
    I was given 2 MP3 files, one that is 4.5Mb and one that is 5.6Mb. I was instructed to have them play on a website i am managing. I have found a nice, clean looking CSS based jQuery audio player. My question is, is this the right solution for files that big? I am not sure if the player preloads the file, or streams it ? (if that is the correct terminology) i dont deal much with audio players and such... this player is from happyworm.com/jquery/jplayer/latest/demo-01.htm is there another approach i shoudl take to get this to play properly? I dont want it to have to buffer, and the visitor to wait, or slow page loading...etc..etc.. i want it to play clean and not affect the visitors session to the site. thanks

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  • iPad Video Playback only delivers audio, not visuals.

    - by Dwaine Bailey
    Hi guys, Recently we've developed an iPhone app for an external company, and everything works fine in the app. There is a section where the app pulls video from the client's server, and streams it into the iPhone's MPMoviePlayerController. This works fine on the iPhone and iPodTouch - both the video and the audio show up just great. The problem, however, is that when the app is run on an iPad (using the iPad's iPhone simulator thingo that it does) only the audio plays, and no video can be seen. Does anybody have any suggestions about what may be causing this? I thought perhaps it was the encoding, but then why would this prevent the video from playing on the iPad, and not the iPhone?

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  • GUI Control For Audio Presentation

    - by Boris
    I need GUI control for audio file presentation. The language is not very important but it should run on windows platform. I should be able to :- load the file play the sound put and move markers across the audio bar. it would be nice if it can load itself from RTP wireshark captures (and not wav files). An example may be seen in audacity (may be someone even had an experience extracting it from there). Writing nyquist scripts in audacity is not a good option because I have to operate on RTP captures and not on raw sound samples. Another example of such control is wireshark RTP analyzer. Any advise?

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  • Gapless (looping) audio playback with DirectX in C#

    - by horsedrowner
    I'm currently using the following code (C#): private static void PlayLoop(string filename) { Audio player = new Audio(filename); player.Play(); while (player.Playing) { if (player.CurrentPosition >= player.Duration) { player.SeekCurrentPosition(0, SeekPositionFlags.AbsolutePositioning); } System.Threading.Thread.Sleep(100); } } This code works, and the file I'm playing is looping. But, obviously, there is a small gap between each playback. I tried reducing the Thread.Sleep it to 10 or 5, but the gap remains. I also tried removing it completely, but then the CPU usage raises to 100% and there's still a small gap. Is there any (simple) way to make playback in DirectX gapless? It's not a big deal since it's only a personal project, but if I'm doing something foolish or otherwise completely wrong, I'd love to know. Thanks in advance.

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  • Recording audio from many source/microphones

    - by user657429
    I'm curious if it's possible to record audio from many sources and if not, what's the limitation. Many current devices have two internal microphones (basically for noise reduction). On top of that it's possible to plug additional external one using audiojack. You can as well have another audio stream via bluetooth headset. You are allowed to specify AudioSource in android but is it possible to do recording from many sources at the same time? I'm also interested how the situation look like on the iOS devices.

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  • Audio Player with Custom Buttons

    - by Bryan Wong
    I am developing a website but require help regarding a simple javascript audio player. Currently, I have four divs set up as the "buttons" : previous song; pause; play; and next song. Pretty much self explanatory, each button serves its obvious function, previous song, pause the song, play the song, and next song. With this in mind, I am also hoping to have the music start playing right after the page completes loading. I understand there are numerous javascript solutions that involve the use of third-party "applications" such as jplayer, however, I am not well learned in javascript and would like to request the aid of the general body in this matter. LOL. that was awkwardly formal. Um, but yes. I am looking for a way to use my four divs as the controllers of a multi-track audio player. Thanks,

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  • Learning to work with audio in C++

    - by Skilldrick
    My degree was in audio engineering, but I'm fairly new to programming. I'd like to learn how to work with audio in a programming environment, partly so I can learn C++ better through interesting projects. First off, is C++ the right language for this? Is there any reason I shouldn't be using it? I've heard of Soundfile and some other libraries - what would you recommend? Finally, does anyone know of any good tutorials in this subject? I've learnt the basics of DSP - I just want to program it! EDIT: I use Windows. I'd like to play about with real-time stuff, a bit like Max/MSP but with more control.

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  • Getting level values from PCM raw data using Core Audio

    - by John
    I am trying to extract level data from a PCM audio file using core audio. I have gotten as far as (I believe) getting the raw data into a byte array (UInt8) but it is 16 bit PCM data and I am having trouble reading the data out. The input is from the iPhone microphone, which I have set as: [recordSetting setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; which is obviously 16 bits. I am then trying to just print out a few values to see if they look reasonable for debug purposes below, and they do not look reasonable (many 0's). ExtAudioFileRef inputFile = NULL; ExtAudioFileOpenURL(track.location, &inputFile); AudioStreamBasicDescription inputFileFormat; UInt32 dataSize = (UInt32)sizeof(inputFileFormat); ExtAudioFileGetProperty(inputFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &inputFileFormat); UInt8 *buffer = malloc(BUFFER_SIZE); AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mData = buffer; //pointer to buffer of audio data bufferList.mBuffers[0].mDataByteSize = BUFFER_SIZE; //number of bytes in the buffer while(true) { UInt32 frameCount = (bufferList.mBuffers[0].mDataByteSize / inputFileFormat.mBytesPerFrame); // Read a chunk of input OSStatus status = ExtAudioFileRead(inputFile, &frameCount, &bufferList); // If no frames were returned, conversion is finished if(0 == frameCount) break; NSLog(@"---"); int16_t *bufferl = &buffer; for(int i=0;i<100;i++){ //const int16_t *bufferl = bufferl[i]; NSLog(@"%d",bufferl[i]); } } Not sure what I am doing wrong, I think it has to do with reading the byte array. Sorry for the long code post...

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • The downsides of using nginx as a primary web server?

    - by FractalizeR
    Hello. I've seen millions of websites using nginx as a proxifying webserver working together with Apache. But I've seen very few servers running nginx only as their default webserver. What are the main downsides of such config? I can see some: Inability to use per-directory config files like .htaccess so every configuration change should be done to main server config file and requires server reload. But pecl htscanner can compensate them for php settings Unavailability of mod_php for nginx, which can be compensated by php-fpm for example. What are others? Why don't people just drop Apache and move to nginx or any other lightweight solution? May be, there are some special reasons?

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  • Insufficient channel capacity of 1GBit

    - by Roman S
    There is a Caching Server (Varnish): it receives data from Amazon S3 on request, saves it for some time and gives it to the client. We have encountered the problem of insufficient channel capacity of 1GBit. Peak load within 4 hours completely chokes the channel. Server performance is sufficient for now. Approximately 4.5TB of data are transmitted per day. More than 100TB are accumulated per month. The first thought that comes to mind is simply to add one more 1GBit port and sleep peacefully until 2GBit are not enough (it may happen quite quickly) or one server is not able to handle it. And then we just need to add new Caching Servers. But now we need a Load Balancer, which will send requests on one and the same URL, always on one and the same server (to avoid multiple copies of the same cached objects). Here are the questions: Does a Balancer need a band equal to sum of all bands of Caching Servers? What shall we do in case there are no ports in a Balancer? Should we add more Balancers or solve the problem by means of Round robin DNS? What are the standard approaches to such problems? Can anyone advise hosting-companies, which can solve this problem? We are interested in American and European markets.

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  • Software Suggestion for Managing Voice Recordings (Windows)

    - by Cbeppe
    I'm looking for Windows software that allows me to effectlively manage already made voice recordings. I have a series of recordings taken from an iPhone and I have extracted the files. The problem is that these are very long recordings and therefore I'm looking for software that allows me to: Bookmark a time in the recording Effectively manage multiple files (like Adobe Bridge does with images) Freeware or Payware Possibly other features, I haven't done this before and I'm sorry I'm unable to give a more professional description. Thanks in advance to everyone who can help! If you have any other questions, please don't hesitate to ask - I will try my best to provide useful answers.

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  • Restore audio settings - cannot open mixer: No such file or directory

    - by Alfred M.
    The internal speaker of my laptop never functionned under Ubuntu. I tried to follow indication on the web and now the jack audio does not work either. The graphic interface for audio management now displays a 'dummy output' instead of the three possible outputs I used to have (one of them was working for the jack output). In a terminal alsamixer raises an error: cannot open mixer: No such file or directory I did try to remove and reinstall alsa-utils but it did not change anything. This happened after a failed atempt to install alsa-driver-linuxant_1.0.23.1_all.deb from here. My sound card seems to be not recognised anymore. After reboot I have no more the sound icon in menu bar the upper right corner. I think I have removed my sound card driver. Indeed, the command sudo lshw -class multimedia indicated audi device as unclaimed. Any idea how I could revert to a better situation (that is jack support and alsa working)? EDIT: The command lspci -nnk | grep -iEA3 audio gives lspci -nnk | grep -iEA3 audio 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) Subsystem: ASUSTeK Computer Inc. Device [1043:1893] 00:1c.0 PCI bridge [0604]: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 [8086:2940] (rev 03)

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  • How can i compare Audio, what programming language should i use

    - by Pimmetje
    I have 2 audio files that are from almost the same source. But at some points there shifted a bit. Also the codecs does not match. I would like to make a program that takes a sample 2 - 4 seconds. And looks for it in the other file. (Most of the time it's not shifted more than 30 seconds). Than take the time and store it, Go ahead for a few seconds take a sample and find it again. This way i want to create a file where i can see on what points the file is shifted. For people who are more interested in what i want. I have a audio/video file speech and subtitles. But i have same speech from different sources with differs a bit in time. And i like to make a program that can correct the subtitle time for me. Enough about the problem I looked on the Internet for ways to compare audio files. Based on what i read comparing 2 audio files isn't that easy as i had hoped. Some talk about algorithms http://www.perlmonks.org/?node_id=169641 Some audio-library's portaudio.com aubio.org sourceforge.net/projects/ccaudio/ ambiera.com/irrklang/ The biggest problem i have is that i can't find something i can generate from the audio that i can use to compare with. I hope someone here can point me in the right direction.

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  • Encode audio to aac with libavcodec

    - by ryan
    I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

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  • High PageIOLatch_SH Waits with High Drive Idle times

    - by Marty Trenouth
    We are experiencing high volume of PageIOLatch_SH waits on our database (row counts in the Billions). However it seems that our drive Idle time Percentage hovers around 50-60 percent. CPU usage is nill. The Database Tuning Advisor gives no suggestions for optimization. The query plan (actual) from the single stored procedure used on the database puts the majority of the expense on index seek (yeah I know these should be optimial) operations. Anyone have suggestions of how to increase throughput?

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  • Normalize or Denormalize in high traffic websites

    - by Inam Jameel
    what is the best practice for database design for high traffic websites like this one stackoverflow? should one must use normalize database for record keeping or normalized technique or combination of both? is it sensible to design normalize database as main database for record keeping to reduce redundancy and at the same time maintain another denormalized form of database for fast searching? or main database should be denormalize and one can make normalized views in the application level for fast database operations? or beside above mentioned approach? what is the best practice of designing high traffic websites???

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  • Is your team is a high-performing team?

    As a child I can remember looking out of the car window as my father drove along the Interstate in Florida while seeing prisoners wearing bright orange jump suits and prison guards keeping a watchful eye on them. The prisoners were taking part in a prison road gang. These road gangs were formed to help the state maintain the state highway infrastructure. The prisoner’s primary responsibilities are to pick up trash and debris from the roadway. This is a prime example of a work group or working group used by most prison systems in the United States. Work groups or working groups can be defined as a collection of individuals or entities working together to achieve a specific goal or accomplish a specific set of tasks. Typically these groups are only established for a short period of time and are dissolved once the desired outcome has been achieved. More often than not group members usually feel as though they are expendable to the group and some even dread that they are even in the group. "A team is a small number of people with complementary skills who are committed to a common purpose, performance goals, and approach for which they are mutually accountable." (Katzenbach and Smith, 1993) So how do you determine that a team is a high-performing team?  This can be determined by three base line criteria that include: consistently high quality output, the promotion of personal growth and well being of all team members, and most importantly the ability to learn and grow as a unit. Initially, a team can successfully create high-performing output without meeting all three criteria, however this will erode over time because team members will feel detached from the group or that they are not growing then the quality of the output will decline. High performing teams are similar to work groups because they both utilize a collection of individuals or entities to accomplish tasks. What distinguish a high-performing team from a work group are its characteristics. High-performing teams contain five core characteristics. These characteristics are what separate a group from a team. The five characteristics of a high-performing team include: Purpose, Performance Measures, People with Tasks and Relationship Skills, Process, and Preparation and Practice. A high-performing team is much more than a work group, and typically has a life cycle that can vary from team to team. The standard team lifecycle consists of five states and is comparable to a human life cycle. The five states of a high-performing team lifecycle include: Formulating, Storming, Normalizing, Performing, and Adjourning. The Formulating State of a team is first realized when the team members are first defined and roles are assigned to all members. This initial stage is very important because it can set the tone for the team and can ultimately determine its success or failure. In addition, this stage requires the team to have a strong leader because team members are normally unclear about specific roles, specific obstacles and goals that my lay ahead of them.  Finally, this stage is where most team members initially meet one another prior to working as a team unless the team members already know each other. The Storming State normally arrives directly after the formulation of a new team because there are still a lot of unknowns amongst the newly formed assembly. As a general rule most of the parties involved in the team are still getting used to the workload, pace of work, deadlines and the validity of various tasks that need to be performed by the group.  In this state everything is questioned because there are so many unknowns. Items commonly questioned include the credentials of others on the team, the actual validity of a project, and the leadership abilities of the team leader.  This can be exemplified by looking at the interactions between animals when they first meet.  If we look at a scenario where two people are walking directly toward each other with their dogs. The dogs will automatically enter the Storming State because they do not know the other dog. Typically in this situation, they attempt to define which is more dominating via play or fighting depending on how the dogs interact with each other. Once dominance has been defined and accepted by both dogs then they will either want to play or leave depending on how the dogs interacted and other environmental variables. Once the Storming State has been realized then the Normalizing State takes over. This state is entered by a team once all the questions of the Storming State have been answered and the team has been tested by a few tasks or projects.  Typically, participants in the team are filled with energy, and comradery, and a strong alliance with team goals and objectives.  A high school football team is a perfect example of the Normalizing State when they start their season.  The player positions have been assigned, the depth chart has been filled and everyone is focused on winning each game. All of the players encourage and expect each other to perform at the best of their abilities and are united by competition from other teams. The Performing State is achieved by a team when its history, working habits, and culture solidify the team as one working unit. In this state team members can anticipate specific behaviors, attitudes, reactions, and challenges are seen as opportunities and not problems. Additionally, each team member knows their role in the team’s success, and the roles of others. This is the most productive state of a group and is where all the time invested working together really pays off. If you look at an Olympic figure skating team skate you can easily see how the time spent working together benefits their performance. They skate as one unit even though it is comprised of two skaters. Each skater has their routine completely memorized as well as their partners. This allows them to anticipate each other’s moves on the ice makes their skating look effortless. The final state of a team is the Adjourning State. This state is where accomplishments by the team and each individual team member are recognized. Additionally, this state also allows for reflection of the interactions between team members, work accomplished and challenges that were faced. Finally, the team celebrates the challenges they have faced and overcome as a unit. Currently in the workplace teams are divided into two different types: Co-located and Distributed Teams. Co-located teams defined as the traditional group of people working together in an office, according to Andy Singleton of Assembla. This traditional type of a team has dominated business in the past due to inadequate technology, which forced workers to primarily interact with one another via face to face meetings.  Team meetings are primarily lead by the person with the highest status in the company. Having personally, participated in meetings of this type, usually a select few of the team members dominate the flow of communication which reduces the input of others in group discussions. Since discussions are dominated by a select few individuals the discussions and group discussion are skewed in favor of the individuals who communicate the most in meetings. In addition, Team members might not give their full opinions on a topic of discussion in part not to offend or create controversy amongst the team and can alter decision made in meetings towards those of the opinions of the dominating team members. Distributed teams are by definition spread across an area or subdivided into separate sections. That is exactly what distributed teams when compared to a more traditional team. It is common place for distributed teams to have team members across town, in the next state, across the country and even with the advances in technology over the last 20 year across the world. These teams allow for more diversity compared to the other type of teams because they allow for more flexibility regarding location. A team could consist of a 30 year old male Italian project manager from New York, a 50 year old female Hispanic from California and a collection of programmers from India because technology allows them to communicate as if they were standing next to one another.  In addition, distributed team members consult with more team members prior to making decisions compared to traditional teams, and take longer to come to decisions due to the changes in time zones and cultural events. However, team members feel more empowered to speak out when they do not agree with the team and to notify others of potential issues regarding the work that the team is doing. Virtual teams which are a subset of the distributed team type is changing organizational strategies due to the fact that a team can now in essence be working 24 hrs a day because of utilizing employees in various time zones and locations.  A primary example of this is with customer services departments, a company can have multiple call centers spread across multiple time zones allowing them to appear to be open 24 hours a day while all a employees work from 9AM to 5 PM every day. Virtual teams also allow human resources departments to go after the best talent for the company regardless of where the potential employee works because they will be a part of a virtual team all that is need is the proper technology to be setup to allow everyone to communicate. In addition to allowing employees to work from home, the company can save space and resources by not having to provide a desk for every team member. In fact, those team members that randomly come into the office can actually share one desk amongst multiple people. This is definitely a cost cutting plus given the current state of the economy. One thing that can turn a team into a high-performing team is leadership. High-performing team leaders need to focus on investing in ongoing personal development, provide team members with direction, structure, and resources needed to accomplish their work, make the right interventions at the right time, and help the team manage boundaries between the team and various external parties involved in the teams work. A team leader needs to invest in ongoing personal development in order to effectively manage their team. People have said that attitude is everything; this is very true about leaders and leadership. A team takes on the attitudes and behaviors of its leaders. This can potentially harm the team and the team’s output. Leaders must concentrate on self-awareness, and understanding their team’s group dynamics to fully understand how to lead them. In addition, always learning new leadership techniques from other effective leaders is also very beneficial. Providing team members with direction, structure, and resources that they need to accomplish their work collectively sounds easy, but it is not.  Leaders need to be able to effectively communicate with their team on how their work helps the company reach for its organizational vision. Conversely, the leader needs to allow his team to work autonomously within specific guidelines to turn the company’s vision into a reality.  This being said the team must be appropriately staffed according to the size of the team’s tasks and their complexity. These tasks should be clear, and be meaningful to the company’s objectives and allow for feedback to be exchanged with the leader and the team member and the leader and upper management. Now if the team is properly staffed, and has a clear and full understanding of what is to be done; the company also must supply the workers with the proper tools to achieve the tasks that they are asked to do. No one should be asked to dig a hole without being given a shovel.  Finally, leaders must reward their team members for accomplishments that they achieve. Awards could range from just a simple congratulatory email, a party to close the completion of a large project, or other monetary rewards. Managing boundaries is very important for team leaders because it can alter attitudes of team members and can add undue stress to the team which will force them to loose focus on the tasks at hand for the group. Team leaders should promote communication between team members so that burdens are shared amongst the team and solutions can be derived from hearing the opinions of multiple sources. This also reinforces team camaraderie and working as a unit. Team leaders must manage the type and timing of interventions as to not create an even bigger mess within the team. Poorly timed interventions can really deflate team members and make them question themselves. This could really increase further and undue interventions by the team leader. Typically, the best time for interventions is when the team is just starting to form so that all unproductive behaviors are removed from the team and that it can retain focus on its agenda. If an intervention is effectively executed the team will feel energized about the work that they are doing, promote communication and interaction amongst the group and improve moral overall. High-performing teams are very import to organizations because they consistently produce high quality output and develop a collective purpose for their work. This drive to succeed allows team members to utilize specific talents allowing for growth in these areas.  In addition, these team members usually take on a sense of ownership with their projects and feel that the other team members are irreplaceable. References: http://blog.assembla.com/assemblablog/tabid/12618/bid/3127/Three-ways-to-organize-your-team-co-located-outsourced-or-global.aspx Katzenbach, J.R. & Smith, D.K. (1993). The Wisdom of Teams: Creating the High-performance Organization. Boston: Harvard Business School.

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