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  • USB sound card with 6.3mm jack output

    - by Andrei Rinea
    I have a pair of Sennheiser 555 semi-professional headphones. These have a 6.3mm jack and a 3.5mm jack adapter. I wish I could buy an USB sound card for my laptop that would have a 6.3mm output not 3.5mm so I could skip this adapter and lose less quality. Any recommendations?

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  • 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.

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  • Digital audio does not work on MacBook Pro

    - by mathk
    I have a MacBook Pro (8,2). Using a TOSLINK cable I have no digital output. Oddly enough, sometime I can hear a glitch when I plug in the cable or when I give it a gentle wiggle. My guess is that the output is not correctly detecting that I have a digital link. So is there a way to force digital audio output on a MacBook Pro? Some say that in the Audio MIDI Setup there is an option but I can't find it. I am running OS X 10.7.5.

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  • Unity 5.1 audio issues (no sound in back channels)

    - by N0xus
    I've trying to bring in surround sound audio into my project. I've set my computer up to run in 5.1 and when I play a 6 channel audio through windows media player (it's a test audio that does left speaker, right speaker etc) it works fine. However, when I run it through Unity, all I get is the front 3 channels. I've set it in the Edit - project settings - audio to be 5.1 in there. I even set it in code with following: void Start() { AudioSettings.speakerMode = AudioSpeakerMode.Mode5point1; } How ever, when I run a debug line of: print ( AudioSettings.driverCaps); It tells me that Unity is only playing in stereo. Is there something I'm still not doing? I should also add I've ran 10 different tests using the 3D audio pan and spread options. I've set both to either being fully off, half way on and full. Still the same results.

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  • How do I record audio through M-Audio Keystudio?

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • Stream audio to mobile device

    - by blackn1ght
    I'd like to stream the audio from Ubuntu 10.10 to my HTC Desire HD (Android 2.2). I've seen solutions so far for streaming from audio players, but I'd like to stream any audio output from the PC to my phone. My use case is for watching TV/Films in VLC or online (BBC iPlayer) in bed, without having to use my surround sound system which is likely to wake up my house mates. I'm not just talking about music from Banshee, but any audio that the system makes. I was thinking that PulseAudio is pretty powerful, is it possible to route audio through that to a mobile device? Can it be done through bluetooth? Cheers in advance!

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  • USB Audio Device Loopback Through Speakers

    - by matto1990
    I have a USB turntable which when plugged in to my ubuntu 10.10 machine appears in the audio settings as an input device (USB PnP Audio Device Analog Stereo) like a microphone. What I'd like to be able to do it to have the sound for that audio device played back through the audio output (speaker or whatever). I'm not too worried if there's a slight delay between the audio coming in and it being played out through the speakers. As far as I'm aware this is refereed to as software loopback. I can achieve exactly what I want if I open Audacity, enable software loopback and press record. Obvious this isn't ideal as I don't really want it recording what I'm playing all the time. I know this is possible because of the Audacity example however I'd like to know if there's a way to do it without it recording. I've search around for a while for a piece of software that does this, however I couldn't get anything even close. Any help would be greatly appreciated.

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  • Play audio in javascript with a good performance

    - by João
    I'm developing a browser game where the player can shoot. Everytime he shoots it play a sound. Currently i'm using this code to play sounds in JavaScript: var audio = document.createElement("audio"); audio.src = "my_sound.mp3"; audio.play(); I'm worried about performance here. Will 10 simultaneous sounds impact my game performance too much? Will all audio objects stay in memory even after they are played?

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  • How can I select an audio output device in directshow

    - by Vibhore Tanwer
    I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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  • Do you have any additions or alterations to this list of popular audio formats?

    - by roja
    All, I am trying to compile a list of common audio file formats used in both personal storage and peer transmission. I have compiled the following list, do you think that there are any significant formats missing? Are any of them not actually common formats? Any advice/alterations are highly useful. advanced audio coding, apple lossless audio file, atrac3 audio file, atrac audio file, audio interchange file format, core audio file, free lossless audio codec file, mpeg 1 audio layer 3, mpeg 2 audio, mpeg 4 audio book file, musical instrument digital interface, ogg vorbis compressed audio file, open media framework file, real audio, real audio media, waveform audio file format, windows media audio Kind regards, Roja

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  • Boost Audio Input on OS X?

    - by alanstorm
    I'm using my 13" Mac Book Pro's audio input functionality with an external microphone (recent vintage, bought around Thanksgiving). I've increased my input volume to the maximum in system preference, but the resulting recorded volume (using iShowU HD) is very low. Is there anyway to increase the input volume/sensitivity beyond Apple's default settings? I've found plenty on google about increasing the OUTPUT volume, but I want to increase the input volume.

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  • Set default system audio output port (for all accounts)

    - by Ludwik Trammer
    The default output audio port Ubuntu doesn't work on my system. It should be "Analog Mono Output/Amplifier", instead of "Analog Output/Amplifier". I can easily change that in sound preferences, just by choosing the right port in the "Output" tab. The problem is this would only apply to a single account, and I would like to change it system-wide, so it applies to all accounts on the system (I have more than 100 users...). I'm after 2 hours of Googling, so any help would be appreciated.

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  • JavaScript audio not playing outside of jQuery function

    - by user1814016
    I know the question title doesn't make much sense, but I can't think of a better way to put it. I am a newbie to jQuery and I'm using this code to fade in a <div> and play a sound: $(document).ready(function(){ $('#speech').fadeIn('medium', function() { play('msg_appear'); var sptx = $('<p class="stext">').text('There is nothing here.'); $('#speech').append(sptx); $('.stext').typeOut({marker: '', delay: 22}); }); }); This code runs fine however the sound plays after the fade-in is complete. I wanted it to play while it was fading in, so I tried placing the play() call outside of the fade-in function like this: $(document).ready(function(){ play('msg_appear'); $('#speech').fadeIn('medium', function() { However, now it's not playing at all. There's no errors on the JavaScript console so I'm unsure if it's a syntax error, and probably something obvious, but I don't know what. play() is a function I found to play audio, here it is if it matters at all. I placed it in the same file the above code is; right above the $(document).ready(). function play(sound) { if (window.HTMLAudioElement) { var snd = new Audio(''); if(snd.canPlayType('audio/ogg')) { snd = new Audio(sound + '.ogg'); } else if(snd.canPlayType('audio/mp3')) { snd = new Audio(sound + '.mp3'); } snd.play(); } else { alert('HTML5 Audio is not supported by your browser!'); } }

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  • How do I change which audio jacks are used for input and output?

    - by yamaha1996
    I'm using a Realtek HD audio card built-in my motherboard. The Windows driver comes with a control panel that allows me to select which back panel jacks are used for what. So for example I can make both the blue jack and green jack for output and only the red one for mic-in. (Whereas by default, the blue jack is for line in, which I never need.) How can I do the same under Linux? If possible, please don't suggest something that involves PulseAudio or JACK; I'd like to do it the plain way, e.g. by editing ALSA configuration files, if possible. The way I understand it, my problem should have nothing to do with software servers redirecting streams, just instructing the driver to treat this jack as so and so because it's hardware supported. Thank you very much!

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  • Monitoring an audio line.

    - by Stefan Liebenberg
    I need to monitor my audio line-in in linux, and in the event that audio is played, the sound must be recorded and saved to a file. Similiar to how motion monitors the video feed. Is it possible to do this with bash? something along the lines of: #!/bin/bash # audio device device=/dev/audio-line-in # below this threshold audio will not be recorded. noise_threshold=10 # folder where recordings are stored storage_folder=~/recordings # run indefenitly, until Ctrl-C is pressed while true; do # noise_level() represents a function to determine # the noise level from device if noise_level( $device ) > $noise_threshold; then # stream from device to file, can be encoded to mp3 later. cat $device > $storage_folder/`date`.raw fi; done;

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • Help find correct alsa model for onboard sound (alc887?) that will work with jack and have correct mixer setup

    - by Jazz
    I have a Gigabyte GA-MA74GMT-S2 motherboard. I am using Jack for sound - connected to ALSA. I am running Ubuntu 12.04. aplay -l reports card 0: SB [HDA ATI SB], device 0: ALC887 Analog [ALC887 Analog] The problem is that the default setup, that alsa decides to use, causes stuttering and xruns no matter how generous I set the frames/period or periods/buffer etc. Also, Jack works fine if I plug in an external USB sound system and use that. My processor is an AMD phenom x4 945, and I have 8GB ram, and Video card is Geforce GTX550 Ti, all of which should be quite capable enough. I also tried Pulseaudio and that works fine, but I need to use Jack At first I thought it might be an interrupt conflict, but I have found that adding "options snd-hda-intel model=generic" to /etc/modprobe.d/alsa-base.conf causes it to play correctly, but the limited mixer setup lacks controls I need - so this setup isn't good enough. Still, it seems to prove it isn't a hardware conflict. I have tried many other models, such as 3stack, 6stack, auto and even basic, and they all suffer from the stuttering. I eventually found "options snd-hda-intel model=3stack-6ch-intel" works without stuttering, and mixer is much closer to what it needs to be. Can anyone help on how to get a correct and accurate model for ALSA to use? More info on the hardware that might help... *-multimedia description: Audio device product: SBx00 Azalia (Intel HDA) vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 14.2 bus info: pci@0000:00:14.2 version: 00 width: 64 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=snd_hda_intel latency=32 resources: irq:16 memory:fe024000-fe027fff

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