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  • Audio Conversion C#

    - by Will
    What is the best way to convert various audio formats to PCM? For example: mp3, evrc, ogg vox. Is there a library out there that will allow me to implement this relatively easily? EDIT: I guess my initial question wasn't really what I needed. Most of the libs I have found are file converters. What I need is a block converter, where I pass in a 1Kb block of vox data and it returns its converted PCM block. Of course I’ll have to tell the converter what type of data it is and various pieces of codec information. The solution I am going for is to save and VOIP formats into a common wav format and to play that conformed file in real time. I thought there should be an easy way to do this because all audio is eventually turned into PCM before it is outputted anyways.

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  • How to record / capture audio with RecordControl on Java ME, SE K770i

    - by tomaszs
    I want to record sound on my Java ME App on K770i. So I used this: http://java.sun.com/javame/reference/apis/jsr135/javax/microedition/media/control/RecordControl.html example of RecordControl in my code. It goes like this: import java.util.Vector; import javax.microedition.lcdui.Choice; import javax.microedition.lcdui.Command; import javax.microedition.lcdui.CommandListener; import javax.microedition.lcdui.Display; import javax.microedition.lcdui.Displayable; import javax.microedition.lcdui.List; import javax.microedition.media.Manager; import javax.microedition.media.MediaException; import javax.microedition.midlet.MIDlet; import java.io.*; import javax.microedition.lcdui.*; import javax.microedition.media.*; import javax.microedition.media.control.*; import javax.microedition.midlet.*; import javax.microedition.rms.*; (...) try { // Create a Player that captures live audio. Player p = Manager.createPlayer("capture://audio"); p.realize(); // Get the RecordControl, set the record stream, // start the Player and record for 5 seconds. RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); Thread.currentThread().sleep(5000); rc.commit(); p.close(); } catch (IOException ioe) { } catch (MediaException me) { } catch (InterruptedException ie) { } But unfortunately when I try to build it, it tells me: *** Creating directories *** *** Compiling source files *** ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ ..\src\example\audiodemo\AudioPlayer.java:121: cannot find symbol symbol : class RecordControl location: class example.audiodemo.AudioPlayer RecordControl rc = (RecordControl)p.getControl("RecordControl"); ^ 2 errors So my question is: why there is no RecordControl class if in documentations it is written this class should be there. Or is there other method to record / capture audio from microfone in Java ME of Sony Ericsson? How do you record sound?

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  • Autoplay an Audio File on Mobile Safari

    - by phantomdata
    Hey guys, I've got a little system dashboard web app that I've written, replete with alarm notifications. I've had it working for quite some time on mobile safari, but recently wanted to add audio to the alarm notifications to allow me to easily know when there are alarms and I'm not looking directly at the display. The alarm notifications are populated through a (relatively) constantly polling ajax request that pulls in and displays an alarm banner if alarms are present. I wanted to add an auto-playing 'alarm' sound as well, but no dice for Safari Mobile. I've tried using HTML5 and embedded objects with no avail. The Apple documentation does state that you can't auto-play an audio file and it must be activated through user action to conserve bandwidth. Has anyone found a way around this in a WLAN setting?

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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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  • iiR Hospital Digital 2011: Tras la historia digital ¿qué?

    - by Eloy M. Rodríguez
    Como el acceso a la documentación está restringido, sólo voy a comentar por encima algunos temas o planteamientos que me han llamado la atención del VI Foro Hospital Digital 2011, organizado por iiR. Y comienzo destacando la buena moderación de Maribel Grau del Hospital Clínic de Barcelona que estuvo sobria, eficaz y motivadora del debate. Me impresionó el proyecto Hospital Líquido del Hospital San Joan de Deu de Barcelona por el compromiso corporativo con una medicina colaborativa involucrando a los pacientes y a los profesionales, con unas iniciativas de eSalud y Salud2.0 avanzadas y apoyadas en un buen soporte legal, tecnológico, de los profesionales y con procesos bien definidos. Es un tema corporativo y no una prueba, como bien explicó Jorge Juan Fernández y detalló después Júlia Cutillas, cuyo rol, por cierto, es de Community Manager. En el debate salió el tema del retorno de la inversión y ese es un tema inmaduro, ya que es difícil de encontrar métricas adecuadas, pero no dudan de su continuidad ya que forma parte de una estrategia corporativa, en la que siempre hay elementos que forman parte de los costes generales y que se consideran necesarios para prestar el nivel de servicio que se desea ofrecer. Cecilia Pérez desde su posición como Jefe de Implantación de HCE en el Hospital de Móstoles hizo énfasis en la importancia de la gestión efectiva del cambio cuando se implanta un sistema de historia clínica electrónica que pasa por una inicial negación de los usarios al cambio, que luego presentan una resistencia al prinicipio para luego empezar a explorar posibilidades y llegar a un compromiso con el cambio. Santiago Borrás, Jefe de Sistemas del Hospital del Henares, partió de un hospital digital, pero eso no es más que el comienzo. Tras tres años la frustración de los profesionales es no perderse entre demasiada información. La etapa necesaria tras la digitalición es la generación y compartición del cononocimiento. Cristina Ibarrola, Directora de Atención Primaria del SNS-O comentó la experiencia de las interconsultas primaria-especializada que reducen la carga asistencial en primaria al aumentar la resolución. Hay una reserva de tiempos específicos en las agendas de los profesionales de ambos lados para garantizar una respuesta en un máximo de 48 horas. Eso ha llevado a una flexibiliazación de la agenda de los médicos de primaria que tienen un 25% más de tiempo para las consultas presenciales. Parece que aquí la opción tomada es dar más tiempo por paciente en vez de más pacientes, supongo que en parte porque la presión asistencial en Navarra tengo entendido que no es tan fuerte como en otras zonas. Alejandra Cubero comentó la experiencia de identificación de pacientes y de inteoperabilidad en Hospitales de Madrid. Ana Rosa Pulido presentó los logros del SES y su proyecto actual de Imagen Médica No Radiológica. Richard Bernat explicó la experiencia de HCE de Salud de la Mujer Dexeus, indicando que si bien no hay métricas del retorno de la inversión, sí hay una percepción del valor por las diferentes direcciones. Arturo Quesada glosó la experiencia de Jimena en el Hospital de Ávila, Joan Chafer desgranó el arduo proceso de introducción de sucesivas soluciones digitales en el Hospital Clínico San Carlos de Madrid comenzando por “Hogar Digital”, todo ello con financiación externa o recursos propios y cerró el turno de intervenciones no comerciales Pedro A. Bonal que presentó el valor de los eDocs dentro del Complejo (aplicado en sus dos acepciones de conjunto y complicado) Hospitalario de Toledo como tránsito a la HCE plenamente digital. Tweet

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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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  • Software to clean up photos of whiteboards and documents?

    - by Norman Ramsey
    I take a lot of photos of whiteboards, blackboards, and so on for teaching purposes (examples online through May 2010). I'm interested in cleaning them up for archival purposes, preferably using Linux. Commercial products ClearBoard and PhotoNote are priced a little aggressively for my purposes, plus my students would like to have this capability too. Does anyone know of any good, open source software for Converting photographs to images with just a few colors? Eliminating perspective distortion? Removing unwanted junk from around the edges of an image? or anything like that? I'm imagining that I start out with a picture of my whiteboard using red and black markers, and I end up with a three-color image using just white, red, and black. Or I photograph a laser-printed document and end up with a clean black-and-white image. I have tried standard tools that reduce the number of colors in an image, and they do a terrible job—probably because they are trying to reproduce the uneven illumination of the original image. Command-line Linux tools would be ideal.

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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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