Search Results

Search found 4698 results on 188 pages for 'audio recording'.

Page 28/188 | < Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >

  • Html5 Audio plays only once in my Javascript code.

    - by Poul
    I have a dashboard web-app that I want to play an alert sound if its having problems connecting. The site's ajax code will poll for data and throttle down its refresh rate if it can't connect. Once the server comes back up, the site will continue working. In the mean time I would like a sound to play each time it can't connect (so I know to check the server). Here is that code. This code works. var error_audio = new Audio("audio/"+settings.refresh.error_audio); error_audio.load(); //this gets called when there is a connection error. function onConnectionError() { error_audio.play(); } However the 2nd time through the function the audio doesn't play. Digging around in Chrome's debugger the 'played' attribute in the audio element gets set to true. Setting it to false has no results. Any ideas?

    Read the article

  • How accurately (in terms of time) does Windows play audio?

    - by MusiGenesis
    Let's say I play a stereo WAV file with 317,520,000 samples, which is theoretically 1 hour long. Assuming no interruptions of the playback, will the file finish playing in exactly one hour, or is there some occasional tiny variation in the playback speed such that it would be slightly more or slightly less (by some number of milliseconds) than one hour? I am trying to synchronize animation with audio, and I am using a System.Diagnostics.Stopwatch to keep the frames matching the audio. But if the playback speed of WAV audio in Windows can vary slightly over time, then the audio will drift out of sync with the Stopwatch-driven animation. Which leads to a second question: it appears that a Stopwatch - while highly granular and accurate for short durations - runs slightly fast. On my laptop, a Stopwatch run for exactly 24 hours (as measured by the computer's system time and a real stopwatch) shows an elapsed time of 24 hours plus about 5 seconds (not milliseconds). Is this a known problem with Stopwatch? (A related question would be "am I crazy?", but you can try it for yourself.) Given its usage as a diagnostics tool, I can see where a discrepancy like this would only show up when measuring long durations, for which most people would use something other than a Stopwatch. If I'm really lucky, then both Stopwatch and audio playback are driven by the same underlying mechanism, and thus will stay in sync with each other for days on end. Any chance this is true?

    Read the article

  • How to programmatically generate an audio podcast file with chapters and text track?

    - by adib
    Hi Anybody know how to programmatically generate audio podcast files with bookmarks that can be used in iTunes / iPod / iPhone / iPod touch? Specifically text bookmarks (bookmarks with titles) that the listener can skip to a specific point in time in the audio file. Also how to add the text transcription of the podcast's content. Even better if you have an example Cocoa code or library to write the audio file. Thanks.

    Read the article

  • how do i merge two audio files and one video file in to a video file using c# ?

    - by wingdings
    i wrote a program in c# using directshow , that captures all devices' audios , and video from single device (webcam or external camera) , now that my requirement is to merge selected audio files with one video file and i can not get it done in c#. so i need a program or libraries that merges one(or several) audio file(s) and one video file and save it as an avi VIDEO file ,, both audio file and video files are in avi format.

    Read the article

  • How to speed up drawing of scaled image? Audio playback chokes during window resize.

    - by Paperflyer
    I am writing an audio player for OSX. One view is a custom view that displays a waveform. The waveform is stored as a instance variable of type NSImage with an NSBitmapImageRep. The view also displays a progress indicator (a thick red line). Therefore, it is updated/redrawn every 30 milliseconds. Since it takes a rather long time to recalculate the image, I do that in a background thread after every window resize and update the displayed image once the new image is ready. In the meantime, the original image is scaled to fit the view like this: // The drawing rectangle is slightly smaller than the view, defined by // the two margins. NSRect drawingRect; drawingRect.origin = NSMakePoint(sideEdgeMarginWidth, topEdgeMarginHeight); drawingRect.size = NSMakeSize([self bounds].size.width-2*sideEdgeMarginWidth, [self bounds].size.height-2*topEdgeMarginHeight); [waveform drawInRect:drawingRect fromRect:NSZeroRect operation:NSCompositeSourceOver fraction:1]; The view makes up the biggest part of the window. During live resize, audio starts choking. Selecting the "big" graphic card on my Macbook Pro makes it less bad, but not by much. CPU utilization is somewhere around 20-40% during live resizes. Instruments suggests that rescaling/redrawing of the image is the problem. Once I stop resizing the window, CPU utilization goes down and audio stops glitching. I already tried to disable image interpolation to speed up the drawing like this: [[NSGraphicsContext currentContext] setImageInterpolation:NSImageInterpolationNone]; That helps, but audio still chokes during live resizes. Do you have an idea how to improve this? The main thing is to prevent the audio from choking.

    Read the article

  • After interruption, delayed audio route change notifications when recording

    - by Frank Shearar
    My iPhone application requires that I know when a user has/has not plugged in her headphones. That's easy. AudioSessionAddPropertyListener with a callback listening to kAudioSessionProperty_AudioRouteChange. I write logs with NSLog as things happen. User plugs the headphones in? Get a notification, and a line in the gdb console. User unplugs the headphones? Ditto. At the same time I'm sensing the noise level of the environment by starting a recording audio queue. This, too, works great: I can get the mic noise level and listen for audio route changes just fine. What I find is that after an interruption, and I've reactivated the audio session and restored the audio category to kAudioSessionCategory_RecordAudio, the audio route notifications go a bit haywire. When I plug in the headphones, I see no notification. When I unplug the headphones I see BOTH the "plugged in" notification AND the "unplugged" notification, in rapid succession. It's like the "plugged in" notification's delayed and, when the "unplugged" notification arrives, the queue of pending notifications is flushed. What am I doing wrong? How do I correctly restore the audio session to get timeous notifications? EDIT: iPhone OS 3.1.2, running on an iPhone 3G. I'm running a program compiled with the 3.0 SDK (from within XCode 3.1.2).

    Read the article

  • Linux based audio prodcuction tutorials

    - by thelinuxer
    I have been searching for a while for Linux based audio production tutorials. All I can find is tool based tutorials. For example I found tutorials on how to use jack, ardour, lmms ..etc. What I need is tutorials that teaches professional audio production with opensource/free tools, like those already available for protools and likes. If any one can guide me to any videos/articles available it would be highly appreciated. Thanks.

    Read the article

  • Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web Audio

    Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web AudioA la suite de la sortie de Firefox 25, Mozilla a également publié une mise à jour de son navigateur pour les possesseurs de terminaux sous Android.Firefox pour Android hérite de quelques fonctionnalités de version desktop, notamment la prise en charge de l'API Web Audio, une spécification du W3C pour les effets audio avancés à partir de HTML5. Cette nouvelle API permettra, par exemple, aux ingénieurs...

    Read the article

  • Get rid of 0.5s latency when playing audio over Bluetooth with A2DP

    - by brillout.com
    As described in the title I experience a half a second delay when playing audio over Bluetooth with A2DP. This makes watching movies not possible as the sound is not synchronised with the video. I'm not sure if the delay is caused by the Bluetooth connection, the A2PD protocol, or the A2DP implementation on my Ubuntu 12.04. Anyways, is this a normal lag? Is there a way to play audio over Bluetooth without any latency?

    Read the article

  • New HP dm4 - No audio on ubuntu 11.10 64bits

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work. Thanks in advance

    Read the article

  • No audio on an HP dm4

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work.

    Read the article

  • Is it possible to play multiple audio streams from one "jukebox" to multiple Airport Express devices?

    - by Alex Reynolds
    I have set up a Mac mini as a jukebox that streams audio to an Airport Express in another room in the house, using the AirPlay/AirTunes feature in iTunes. I control this with the iOS Remote app, and this works great. At the present time, it looks like the Mac mini's copy of iTunes gets taken over by the Remote app, while streaming. If I set up a second Airport Express in room B, is there a way to set it up (as well as the jukebox) so that it can receive and play its own unique music stream ("stream B"), separate from what's going on at the Mac mini, or in room A, which is playing stream A? To accomplish this, I would be happy to buy a copy of Rogue Amoeba's AirFoil if it will allow sending multiple, separate audio streams from one computer to the multiple wireless bridges, while using the Remote app (or a Rogue Amoeba equivalent for iOS). However, it is unclear to me from their site documentation, whether that is possible or not. I'd prefer to give the points to an answer that solves this problem. If you don't know if it can be done, or do not think it can be done, please allow others to answer. I appreciate your help. Thanks for your advice.

    Read the article

  • Setup for a live (low-latency) audio video broadcast over Wi-Fi?

    - by Majal Mirasol
    The Upgrade We are capturing audio (from mixer) and video (from a camera) from a main auditorium and passing it to separate rooms within the building. We used to have done this via manual audio/video cables and wires. We wanted to "upgrade" the system and wirelessly broadcast the stream via Wi-Fi. The Problem In our current setup (Wirecast running on A10 on a Wireless-N network), we have the problem of delay. Our streams are delayed from a minute up to five minutes on the clients (laptop/iPad/Android). This had not been a problem from the previous wired connections. Since the wireless network is local, we thought that a delay of less than a second should be achievable. Our Question And so it goes. Anybody there who has any experience for a setup that has both low latency and at the same time user-friendly to clients streaming in the program? Any recommendations would be highly appreciated. (Our current setup in on Windows 7, but setup on a dedicated Linux box is preferred, if achievable.)

    Read the article

  • How to record my voice on a Mac Mini with headphones?

    - by user718408
    I'm try to record my voice via the headphone on a Mac Mini, but it's not working. I saw on Apple's site that the Mac Mini can record voice, but it doesn't seem to be working for me. Here is a hardware overview: Model Name: Mac Mini Model Identifier: Macmini3,1 Processor Name: Intel Core 2 Duo Processor Speed: 2.26 GHz Number Of Processors: 1 Total Number Of Cores: 2 L2 Cache: 3 MB Memory: 4 GB Audio: Make: Intel High Definition Audio Audio ID: 65 Headphone connection: Combination Output Line Input connection: Combination Input Speaker connection: Internal S/PDIF Optical Digital Audio Output connection: Combination Output S/PDIF Optical Digital Audio Input connection: Combination Input Any ideas how I can successfully get recording working?

    Read the article

  • iphone audio streaming

    - by mobapps99
    Hi , i'm developing an application which uses audio streaming. For streaming audio from internet i'm using the AudioStreamer class. The audio streamer has four state isPlaying, isPaused ,isWaiting, and isIdle . My problem is that when the audio streamer is in the state "isWaiting" and at that time if i get a phone call Audio queue fails giving the error "Audio queue start failed." Any has solution for this? help....

    Read the article

  • what's the "best" approach to creating the UI of an audio plugin that will be both audio unit and VST for OS X and Windows?

    - by SaldaVonSchwartz
    I'm working on a couple audio plugins. Right now, they are audio units. And while the "DSP" code won't change for the most part between implementations / ports, I'm not sure how to go about the GUI. For instance, I was looking at the Apple-supplied AUs in Lion. Does anyone know how did they go about the UI? Like, are the knobs and controls just subclasses of Cocoa controls? are they using some separate framework or coding these knobs and such from scratch? And then, the plugs I'm working on are going to be available too as VSTs for Windows. I already have them up and running with generic interfaces. But I'm wondering if I should just get over it and recreate all my interfaces with the vstgui code provided by Steinberg or if there's a more practical approach to making the interfaces cross-platform.

    Read the article

  • audio frameworks in iPhone

    - by suse
    Hello, I would like to know the follwing information about iPhone audio system Heirarchy of the audio framework in iPhone OS. i know that there are 3 main audio frameworks in iPhone OS.i.e AVFoundation Framework CoreAudio Framework OpenAL Framework what are the audio formats supported in each of the above framework?I mean will all the framework support all audio formats or are they dependent about the audio formats it support? Thank You

    Read the article

  • FMOD surround sound openframeworks

    - by user1449425
    Ok, I hope I don't mess this up, I have had a look for some answers but can't find anything. I am trying to make a simple sampler in openframeworks using the FMOD sound player in 3D mode. I can make a single instance work fine (recording a new file using libsndfilerecorder and then playing it back and moving it in surround. However I want to have 8 layers of looping audio that I can record and replace one layer at a time in a live show. I get a lot of problems as soon as I have more than 1 layer. The first part of my question relates to the FMOD 3D modes, it is listener relative, so I have to define the position of my listener for every sound (I would prefer to have head relative mode but I cannot make this work at all. Again this works fine when I am using a single player but with multiple players only the last listener I update actually works. The main problem I have is that when I use multiple players I get distortion, and often a mix of other currently playing sounds (even when the microphone cannot hear them) in my new recordings. Is there an incompatability with libsndfilerecorder and FMOD? Here I initialise the players for (int i=0; i<CHANNEL_COUNT; i++) { lvelocity[i].set(1, 1, 1); lup[i].set(0, 1, 0); lforward[i].set(0, 0, 1); lposition[i].set(0, 0, 0); sposition[i].set(3, 3, 2); svelocity[i].set(1, 1, 1); //player[1].initializeFmod(); //player[i].loadSound( "1.wav" ); player[i].setVolume(0.75); player[i].setMultiPlay(true); player[i].play(); setupHold[i]==false; recording[i]=false; channelHasFile[i]=false; settingOsc[i]=false; } When I am recording I unload the file and make sure the positions of the player that is not loaded are not updating. void fmodApp::recordingStart( int recordingId ){ if (recording[recordingId]==false) { setupHold[recordingId]=true; //this stops the position updating cout<<"Start recording Channel " + ofToString(recordingId+1)+" setup hold is true \n"; pt=getDateName() +".wav"; player[recordingId].stop(); player[recordingId].unloadSound(); audioRecorder.setup(pt); audioRecorder.setFormat(SF_FORMAT_WAV | SF_FORMAT_PCM_16); recording[recordingId]=true; //this starts the libSndFIleRecorder } else { cout<<"Channel" + ofToString(recordingId+1)+" is already recording \n"; } } And I stop the recording like this. void fmodApp::recordingEnd( int recordingId ){ if (recording[recordingId]=true) { recording[recordingId]=false; cout<<"Stop recording" + ofToString(recordingId+1)+" \n"; audioRecorder.finalize(); audioRecorder.close(); player[recordingId].loadSound(pt); setupHold[recordingId]=false; channelHasFile[recordingId]=true; cout<< "File recorded channel " + ofToString(recordingId+1) + " file is called " + pt + "\n"; } else { cout << "Sorry track" + ofToString(recordingId+1) + "is not recording"; } } I am careful not to interrupt the updating process but I cannot see where I am going wrong. Many Thanks

    Read the article

  • AAC Sample Rate and Bit Rate for High Quality Audio?

    - by marco.ragogna
    What are the AAC Sample Rate and Bit Rate settings to set in order to encode an audio track with a quality comparable to MP3 320kbps? I need to backup a DVD movie, the default settings for AAC are Bitrate (KB/s) 128 Sample Rate (HZ) 44100 should I set Bitrate (KB/s) 320 Sample Rate (HZ) 48000 or the default are already good?

    Read the article

  • Linux program to convert audio file of fax transmission to image?

    - by bdk
    I have a number of uncompressed audio files recorded off of an analog (POTS) telephone line of fax transmissions. Is there a Linux utility or library I could use to convert these files into images of the fax they contain? I'm not looking to send/receive a fax via a modem, but just to "replay" the communications tones and parse out the fax message.I'm guessing this may not be possible due to duplex issues and not knowing which end of the conversation is sending what,but thought I'd ask to see if anyone knew of something.

    Read the article

  • How can I split the 5.1 audio channels from an AC3 file into individuals streams (preferably on a Ma

    - by Drarok
    I have a file that I've pulled from a DVD that is apparently in AC3 5.1 format. The extension is .AC3 and it opens an plays in QuickTime, VLC etc. What I want is each individual channel in a separate file, but I can't seem to find any tools that will allow be to do that. Is there a way to split the file I have, or alternatively is there a way to pull the audio streams from a 5.1 DVD?

    Read the article

< Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >