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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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  • 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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  • 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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  • Square Reader Modified to Record Off Old Reel-to-Reel Tape [Video]

    - by Jason Fitzpatrick
    The Square Reader is a tiny magnetic credit card reader that has taken the mobile payment industry by storm. This clever hack dumps the credit card reading in favor of snagging the audio from old music reels. Evan Long was curious about whether the through-the-headphones interface of the Square Reader could be used to read audio data off old magnetic recordings. With a very small modification (he had to bend a metal tab inside the reader to allow the audio tape to slide through more easily) he was able to listen to and record audio off old reels. Watch the video above to see it in action or hit up the link below to read more about his project. iPod Meets Reel [via Make] HTG Explains: What Is Windows RT and What Does It Mean To Me? HTG Explains: How Windows 8′s Secure Boot Feature Works & What It Means for Linux Hack Your Kindle for Easy Font Customization

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  • What output and recording ports does the Java Sound API find on your computer?

    - by Dave Carpeneto
    Hi all - I'm working with the Java Sound API, and it turns out if I want to adjust recording volumes I need to model the hardware that the OS exposes to Java. Turns out there's a lot of variety in what's presented. Because of this I'm humbly asking that anyone able to help me run the following on their computer and post back the results so that I can get an idea of what's out there. A thanks in advance to anyone that can assist :-) import javax.sound.sampled.*; public class SoundAudit { public static void main(String[] args) { try { System.out.println("OS: "+System.getProperty("os.name")+" "+ System.getProperty("os.version")+"/"+ System.getProperty("os.arch")+"\nJava: "+ System.getProperty("java.version")+" ("+ System.getProperty("java.vendor")+")\n"); for (Mixer.Info thisMixerInfo : AudioSystem.getMixerInfo()) { System.out.println("Mixer: "+thisMixerInfo.getDescription()+ " ["+thisMixerInfo.getName()+"]"); Mixer thisMixer = AudioSystem.getMixer(thisMixerInfo); for (Line.Info thisLineInfo:thisMixer.getSourceLineInfo()) { if (thisLineInfo.getLineClass().getName().equals( "javax.sound.sampled.Port")) { Line thisLine = thisMixer.getLine(thisLineInfo); thisLine.open(); System.out.println(" Source Port: " +thisLineInfo.toString()); for (Control thisControl : thisLine.getControls()) { System.out.println(AnalyzeControl(thisControl));} thisLine.close();}} for (Line.Info thisLineInfo:thisMixer.getTargetLineInfo()) { if (thisLineInfo.getLineClass().getName().equals( "javax.sound.sampled.Port")) { Line thisLine = thisMixer.getLine(thisLineInfo); thisLine.open(); System.out.println(" Target Port: " +thisLineInfo.toString()); for (Control thisControl : thisLine.getControls()) { System.out.println(AnalyzeControl(thisControl));} thisLine.close();}}} } catch (Exception e) {e.printStackTrace();}} public static String AnalyzeControl(Control thisControl) { String type = thisControl.getType().toString(); if (thisControl instanceof BooleanControl) { return " Control: "+type+" (boolean)"; } if (thisControl instanceof CompoundControl) { System.out.println(" Control: "+type+ " (compound - values below)"); String toReturn = ""; for (Control children: ((CompoundControl)thisControl).getMemberControls()) { toReturn+=" "+AnalyzeControl(children)+"\n";} return toReturn.substring(0, toReturn.length()-1);} if (thisControl instanceof EnumControl) { return " Control:"+type+" (enum: "+thisControl.toString()+")";} if (thisControl instanceof FloatControl) { return " Control: "+type+" (float: from "+ ((FloatControl) thisControl).getMinimum()+" to "+ ((FloatControl) thisControl).getMaximum()+")";} return " Control: unknown type";} } All the application does is print out a line about the OS, a line about the JVM, and a few lines about the hardware found that may pertain to recording hardware. For example on my PC at work I get the following: OS: Windows XP 5.1/x86 Java: 1.6.0_07 (Sun Microsystems Inc.) Mixer: Direct Audio Device: DirectSound Playback [Primary Sound Driver] Mixer: Direct Audio Device: DirectSound Playback [SoundMAX HD Audio] Mixer: Direct Audio Device: DirectSound Capture [Primary Sound Capture Driver] Mixer: Direct Audio Device: DirectSound Capture [SoundMAX HD Audio] Mixer: Software mixer and synthesizer [Java Sound Audio Engine] Mixer: Port Mixer [Port SoundMAX HD Audio] Source Port: MICROPHONE source port Control: Microphone (compound - values below) Control: Select (boolean) Control: Microphone Boost (boolean) Control: Front panel microphone (boolean) Control: Volume (float: from 0.0 to 1.0) Source Port: LINE_IN source port Control: Line In (compound - values below) Control: Select (boolean) Control: Volume (float: from 0.0 to 1.0) Control: Balance (float: from -1.0 to 1.0)

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  • does ipad support voice recording???

    - by Abdullah Waseer
    Hi, i have made an application in iphone which supports video recording and i want it to launch on iPad too but currently i dont know whether ipad device supports voice recording or not. can anyone please tell me whether IPAD SUPPORTS VOICE RECORDING??? thanks...

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  • Recording screen-casts on another computer

    - by paleozogt
    We're trying to record the desktops of users using demo versions of our software (this is an in-house lab setup). We need to have the recording happen on a separate computer (just across the room), so that the recording software doesn't interfere with the user. Every screen recording software I've seen will only record what's happening on the computer its installed on; ie, you can't record what's happening on another computer. So it seems I need to cobble together a solution (unless anyone knows of software that will do this). Getting the video to the other computer seems easy enough. I'm using TightVNC with the DFMirage driver on the test computer. The recording computer connects to the test computer with TightVNC and then uses CamStudio to record what's happening. The real problem is how to deal with the audio. We need to record both what the user is saying (through a headset mic) as well as the sounds produced by the test computer. But VNC doesn't transmit audio. :( I'm not sure how to get both audio streams (mic and sounds) over to the recording computer. Any ideas?

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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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  • Xamarin Designer for Android Webinar - Recording

    - by Wallym
    Here is some info on the recording of the webinar that I did last week for AppDev regarding the Xamarin Designer for Android.Basic Info: Android user interfaces can be created declaratively by using XML files, or programmatically in code. The Xamarin Android Designer allows developers to create and modify declarative layouts visually, without having to deal with the tedium of hand-editing XML files. The designer also provides real-time feedback, which lets the developer validate changes without having to redeploy the application in order to test a design. This can speed up UI development in Android tremendously. In this webinar, we'll take a look at UI Design in Mono for Android, the basics of the Xamarin Android Designer, and build a simple application with the designer.Here is the link:http://media.appdev.com/EDGE/LL/livelearn05232012.wmvI think it will only play in Internet Explorer.  Enjoy!

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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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  • Video recording in cheese is slow

    - by Gaurav Butola
    cheese records at painful fps. video recording is really slow in cheese almost unusable. How can I increase the fps for cheese. I have HCL laptop with built in 1.3 MP camera 2.47 GHz i3 processor with 2 GB RAM. running maverick 32 bit. I installed camorama from software centre which has fine video (fps), but I cant seem to use it due to some error. So I doubt there is something to be tweaked with cheese itself.

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  • Recording Available: March 2010 Quarterly Customer Update Webcast

    - by michelle.huff
    Missed the last Quarterly Customer Update Webcast? We discussed several product updates on the March quarterly customer Webcast, including the first phase of the Oracle Content Management 11g release. Some of the highlights include Information Rights Management (IRM) 11g and Imaging and Process Management (I/PM) 11g Overviews. Additionally, we covered I/PM 11g new features, implementation and migration topics that existing customers would like to know. You can find quick links to all the resources I mentioned on the call, as well as links to the presentation and recording details in My Oracle Support from the March 2010 Webcast Resource Links page on OTN.

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  • How To Specify Bitrate, Codec and Demultiplexing for VLC Video Capture or Recording

    - by Subhash
    I capture video from old TV tuner card - Pinnacle PCTV - using VLC. The video is from the Composite input and audio is from I guess the mixer or Line in. The command I use is: vlc v4l2:///dev/video0:normal=pal:width=720:height=576:input=1 :input-slave="alsa://hw:0,0" In VLC, I have enabled the Advanced Controls toolbar, which allows me to record videos when I want to. However, these videos are uncompressed - very big and play only with VLC. Totem throws the "Could not demultiplex stream" error. I need to convert them using WinFF to reduce their size and make them playable with Totem and other software. My question is whether I can configure the recording settings - the codecs and the bitrate, and also get the stream demultiplexed. If I pass any -sout parameter with command I get a "Segmentation fault". I use 64-bit Ubuntu 10.10.

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