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  • No audio with streaming video

    - by Chris Barnhill
    I am having trouble with audio when playing streaming videos. My sound card is fine. I know this because if I play sounds from my local machine, there's no problem. It's only when I try to play sounds from the internet that I lose audio. This only started happening recently when I did 2 things: I connected a USB headphone/microphone set to record screencasts I began recording/publishing screencasts from screenr.com. I have tried playing video both with the headset connected and without it connected: it makes no difference. If I record a screencast on screenr.com and preview it, I hear the audio. But once I publish is and play it, there is no audio. I also hear no audio with YouTube videos. I really hope someone can help. Thanks. The latest is that the problem went away after I powered my system off and on. A reboot didn't do it, I had to actually shut down the power.

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  • 'Future-proof' Live Audio Capture & Broadcast [migrated]

    - by maxpowers
    I'm looking to implement some live audio broadcasting functionality within a Ruby on Rails site for a client and was hoping I could get some input from people who have tackled this type of thing before. Essentially what I need to do is capture and record a user's audio (via microhpone, line in, etc), then stream that to 1,000+ listeners with very little latency, like sub 2 second if possible. So it looks like we've got 3 parts: Web-based audio capture (likely with Flash or JS) Server to accept audio feed and stream to listeners (likely Icecast or Wowza) Actual audio player (maybe HTML5 w/ Flash as a fallback? Maybe this jPlayer fork) Does RTMP makes sense here? Or maybe HTTP? What's the most 'future-proof' way to make this happen? Building with mobile in mind, but still want to be able stream to anyone. I've found lots of potentially helpful threads and software but I'm struggling to get an idea of how it all fits together. I'm a front end guy and way out of my comfort zone so if anyone has insights to offer, I'd love to hear them.

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  • Sound card problem, no audio device detected

    - by Paul
    I bought a new sound card because my built in sound card did not function. When I open YouTube, Media Player or anything that can create a sound my computer will hang up and sometimes when I start my computer it will hang when the Windows XP sound will activate. Update: My computer has no audio. It says NO AUDIO DEVICE. I already installed Realtek AC97 and Realtek High Definition Audio Driver and I also pasted stream.dll to the Windows and system32 folders and I restarted my computer but it still says NO AUDIO DEVICE. Please help me. Thanks

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  • MP4 video - edit audio track

    - by Maccaius
    I have recorded some nice sport videos with mz GoPro HD action camera. I would like to edit the audio track. I dont want to get rid of the whole audio track - just erase small parts (e.g. compression artifacts or me saying some swearwords). When the original audio track is cleansed, Id add another music layer in FCE afterwards. I'd really like to edit the audio like in a WaveLab etc. Any ideas?

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  • Routing audio to Bluetooth Headset (non-A2DP) on Android

    - by Jayesh
    I have a non-A2DP single ear BT headset (Plantronics 510) and would like to use it with my Android HTC Magic to listen to low quality audio like podcasts/audio books. After much googling I found that only phone call audio can be routed to the non-A2DP BT headsets. (I would like to know if you have found a ready solution to route all kinds of audio to non-A2DP BT headsets) So I figured, somehow programmatically I can channel the audio to the stream that carries phone call audio. This way I will fool the phone to carry my mp3 audio to my BT headset. I wrote following simple code. import android.content.*; import android.app.Activity; import android.os.Bundle; import android.media.*; import java.io.*; import android.util.Log; public class BTAudioActivity extends Activity { private static final String TAG = "BTAudioActivity"; private MediaPlayer mPlayer = null; private AudioManager amanager = null; @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); amanager = (AudioManager) getSystemService(Context.AUDIO_SERVICE); amanager.setBluetoothScoOn(true); amanager.setMode(AudioManager.MODE_IN_CALL); mPlayer = new MediaPlayer(); try { mPlayer.setDataSource(new FileInputStream( "/sdcard/sample.mp3").getFD()); mPlayer.setAudioStreamType(AudioManager.STREAM_VOICE_CALL); mPlayer.prepare(); mPlayer.start(); } catch(Exception e) { Log.e(TAG, e.toString()); } } @Override public void onDestroy() { mPlayer.stop(); amanager.setMode(AudioManager.MODE_NORMAL); amanager.setBluetoothScoOn(false); super.onDestroy(); } } As you can see I tried combinations of various methods that I thought will fool the phone to believe my audio is a phone call: Using MediaPlayer's setAudioStreamType(STREAM_VOICE_CALL) using AudioManager's setBluetoothScoOn(true) using AudioManager's setMode(MODE_IN_CALL) But none of the above worked. If I remove the AudioManager calls in the above code, the audio plays from speaker and if I replace them as shown above then the audio stops coming from speakers, but it doesn't come through the BT headset. So this might be a partial success. I have checked that the BT headset works alright with phone calls. There must be a reason for Android not supporting this. But I can't let go of the feeling that it is not possible to programmatically reroute the audio. Any ideas? P.S. above code needs following permission <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS"/>

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  • Ardour won't rewind when jack time master

    - by Edward
    Using Ubuntu Studio 12.04, ardour will not rewind when it is set to the jack time master. I've read that this could be due to a jack/ardour version conflict, but I am not sure what the correct combo should be. The same thing happens with "ardour 2.8.14 (built from revision 13065)" and "ardour 2.8.12 (built from revision 10144)". The latter is the default installation with ubuntu studio 12.04 LTS. Linux "/proc/version" reports as Linux version 3.2.0-23-lowlatency-pae (buildd@vernadsky) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu4) ) #31-Ubuntu SMP PREEMPT Wed Apr 11 04:07:36 UTC 2012 and "jackd --version" reports as: jackdmp 1.9.8 Copyright 2001-2005 Paul Davis and others. Copyright 2004-2011 Grame. jackdmp comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details jackdmp version 1.9.8 tmpdir /dev/shm protocol 8 Thanks for any help.

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  • Can it be harmful to grant jackd realtime priority?

    - by SuperElectric
    I am apt-get installing Ardour, a sound mixing program, just to try it out. Installing Ardour also installs JACK, a dependency. As part of the JACK installation script, I get the following dialog: If you want to run jackd with realtime priorities, the user starting jackd needs realtime permissions. Accept this option to create the file /etc/security/limits.d/audio.conf, granting realtime priority and memlock privileges to the audio group. Running jackd with realtime priority minimizes latency, but may lead to complete system lock-ups by requesting all the available physical system memory, which is unacceptable in multi-user environments. Enable realtime process priority? I'm installing on my laptop, which never has multiple simultaneous users. I still have concerns: is JACK something that'll be used by the system itself to play any sound (i.e. will it replace ALSA)? If so, does that mean that if I enable realtime priority for JACK, I'll run a slight risk of freezing the machine whenever any sound is played? Or is JACK only going to be used by Ardour for now (until I install some other JACK-dependent program)? Thanks, -- Matt

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  • Getting Audio from a Zone

    - by bleonard
    Now that I have Firefox and Java Web Start running from a zone, the last piece of the puzzle was audio (essential because most Flash content is accompanied by sound).  In the global zone there's a nice little utility called audiotest for testing your sound: bleonard@solaris:~$ audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 47727.00 Hz (-0.57%)> *** All tests completed OK *** Of course, before you can try audiotest in a zone, it must be installed: root@myzone:~# pkg install audio-utilities Packages to install: 1 Create boot environment: No DOWNLOAD PKGS FILES XFER (MB) Completed 1/1 6/6 0.4/0.4 PHASE ACTIONS Install Phase 20/20 PHASE ITEMS Package State Update Phase 1/1 Image State Update Phase 2/2 However, we'll need to do more than just install audiotest: root@myzone:~# audiotest /dev/mixer: No such file or directory The device file is missing from /dev. The audio devices also need to be added to the zone. For this we modify the zone configuration as follows: bleonard@solaris:~$ sudo zonecfg -z myzone Password: zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/audio* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sound/* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/mixer* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sndstat zonecfg:myzone:device> end zonecfg:myzone> verify zonecfg:myzone> exit Then reboot the zone: bleonard@solaris:~$ sudo zoneadm -z myzone reboot After which, audiotest should work: root@myzone:~# audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 48208.00 Hz (0.43%)> *** All tests completed OK *** You can also examine /dev/sndstat for additional information: root@myzone:~# cat /dev/sndstat SunOS Audio Framework Audio Devices: 0: audio810#0 Intel AC'97, ICH (DUPLEX) Mixers: 0: audio810#0 Intel AC'97, ICH AC'97 codec: SigmaTel STAC9700 However, when testing the sound from Firefox (from a user account other than root), such as this recent Flash presentation on Solaris availability, you may still be disappointed. This is simply a permissions problem, as the devices only have read and write permissions for root: root@myzone:~# ls -l /dev/audio* crw------- 1 root root 99, 3 Jul 1 10:21 /dev/audio crw------- 1 root root 99, 4 Jul 1 10:21 /dev/audioctl To address this: root@myzone:~# chmod 777 /dev/audio* root@myzone:~# chmod 777 /dev/sound/* And you should be all set.

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  • How to use an Audio Unit on the iPhone

    - by CodeToaster
    I'm looking for a way to change the pitch of recorded audio as it is saved to disk, or played back (in real time). I understand Audio Units can be used for this. The iPhone offers limited support for Audio Units (for example it's not possible to create/use custom audio units, as far as I can tell), but several out-of-the-box audio units are available, one of which is AUPitch. How exactly would I use an audio unit (specifically AUPitch)? Do you hook it into an audio queue somehow? Is it possible to chain audio units together (for example, to simultaneously add an echo effect and a change in pitch)? EDIT: After inspecting the iPhone SDK headers (I think AudioUnit.h, I'm not in front of a Mac at the moment), I noticed that AUPitch is commented out. So it doesn't look like AUPitch is available on the iPhone after all. weep weep Apple seems to have better organized their iPhone SDK documentation at developer.apple.com of late - now its more difficult to find references to AUPitch, etc. That said, I'm still interested in quality answers on using Audio Units (in general) on the iPhone.

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  • Front panel audio replacement

    - by develroot
    I have experienced some problems with my headphones and it turned out the front 3.5mm audio jack is defective..and it doesn't make a good contact. I can't replace the jack because it's built-in. I am wondering if there are such things as "modular" Front panels to be inserted into the available external 3.5" bay and, of course, to support (natively, internal connector) Realtek HD Audio. (my case is Asus TA-K5, but I guess it doesn't make any difference)

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  • <audio> elements not working on WordPress

    - by dannystewart
    Hello all, I have a small WordPress site. I do a lot of audio work and I'm trying to post HTML5 audio clips in blog entries on WordPress. For some reason it isn't working. It might have something to do with the style I'm using on my WordPress site but I haven't been able to nail it down. I know my audio tags are valid, as they work elsewhere. Here's an example audio tag: <audio src="http://files.dannystewart.com/dom2008.mp3"></audio> And here's a page demonstrating it not working: http://www.dannystewart.com/html5-audio-test/ I'm quite sure this is something very simple that I've just missed, but any pointers would be appreciated. Thanks!

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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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  • HTML5 <audio> Safari live broadcast vs not

    - by Peter Parente
    I'm attempting to embed an HTML5 audio element pointing to MP3 or OGG data served by a PHP file . When I view the page in Safari, the controls appear, but the UI says "Live Broadcast." When I click play, the audio starts as expected. Once it ends, however, I can't start it playing again by clicking play. Even using the JS API on the audio element and setting currentTime to 0 fails with an index error exception. I suspected the headers from the PHP script were the problem, particularly missing a content length. But that's not the case. The response headers include a proper Content- Length to indicate the audio has finite size. Furthermore, everything works as expected in Firefox 3.5+. I can click play on the audio element multiple times to hear the sound replay. If I remove the PHP script from the equation and serve up a static copy of the MP3 file, everything works fine in Safari. Does this mean Safari is treating audio src URLs with query parameters differently than URLs that don't have them? Anyone have any luck getting this to work? My simple example page is: <!DOCTYPE html> <html> <head></head> <body> <audio controls autobuffer> <source src="say.php?text=this%20is%20a%20test&format=.ogg" /> <source src="say.php?text=this%20is%20a%20test&format=.mp3" /> </audio> </body> </html> HTTP Headers from PHP script: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 15:39:34 GMT Server: Apache X-Powered-By: PHP/5.2.10 Content-Length: 8993 Keep-Alive: timeout=2, max=98 Connection: Keep-Alive Content-Type: audio/mpeg HTTP Headers from direct file access: HTTP/1.x 200 OK Date: Sun, 03 Jan 2010 20:06:59 GMT Server: Apache Last-Modified: Sun, 03 Jan 2010 03:20:02 GMT Etag: "a404b-c3f-47c3a14937c80" Accept-Ranges: bytes Content-Length: 8993 Keep-Alive: timeout=2, max=100 Connection: Keep-Alive Content-Type: audio/mpeg I tried hard-coding the Accept-Ranges header into the script too, but no luck.

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  • Windows Audio Issue

    - by Nikki
    This one is driving me nuts. Hoping someone can shed some light. I'm running windows 7 using onboard audio. It's been fine for over 2 years but lately there's a problem every time I play audio. I hear a small soft burst of static and the volume turns itself down from 50% to 23%. Once at 23%, it plays fine. No related events logged in viewer. No reported problems with the device. Different headphones, same problem. I played around with audio settings for hours but the problem persists. EDIT: ok more info: Motherboard: ECS G31T-M LGA775 System info displays this: Name High Definition Audio Device Manufacturer Microsoft Status OK PNP Device ID HDAUDIO\FUNC_01&VEN_1106&DEV_E721&SUBSYS_10192683&REV_1001\4&3D4E739&0&0001 Driver c:\windows\system32\drivers\hdaudio.sys (6.1.7600.16385, 297.00 KB (304,128 bytes), 14/07/2009 9:51 AM) I'll keep adding info as I find it. The question I want resolved is; Is it faulty hardware? If so, I can buy a sound card. I can't imagine software is responsible since I haven't installed anything new for weeks. Virus scans are clear as well. The static burst is irritating to say the least. Tried 2 different headphones and separate speakers. Same problem. I know it's not an easy problem but I was hoping someone had encountered the same thing.

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  • Simulating audio playback on headless linux server

    - by afro
    Hi people, We have a headless linux server (Debian 5) we use for runnin integration tests of our web-page code. Among these tests are ones implemented using Selenium, which practically simulates a user browsing our pages and clicking on things. One of these tests is failing now, because it involves starting a flash-based audio player and checking to see whether the progress bar gets displayed properly. The reason this test fails is that there is no way to play the audio, and no sound card on the machine, which has simple webserver hardware. So, my question would be: Is there a simple way of giving a program the impression that its audio output is being processed, and playback is taking place? I don't have to record the playback, or redirect it or anything like that, just a dummy soundcard, like the dummy X-server we aer using, which actually does not need to display stuff. I have tried using JACK, but it's too complicated, and the documentation does not even answer this very simple question. I also installed alsa on the server; it 'pretends' to run, but when a program tries to play audio, just spews error and debug information having to do with the non-existence of a soundcard. It would be really awesome if one of you has a simple answer to this question. Cheers, Ulas

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  • Configure audio on HP ENVY 4 ultrabook

    - by phodu_insaan
    I want to configure audio for ubuntu 12.04 on my laptop. Currently the audio just does not play. If i try and plug in headphones then somewhere midway to being fully plugged in the audio plays on the headphones, I plug in further and the sound disappears. How do I get this to work? lspci | grep audio Audio device: Intel Corporation Panther Point High Definition Audio Controller My laptop is one the beats edition HP laptops, and the driver for win7 was an IDT HD audio driver. --EDIT-- The output for cat /proc/asound/card0/codec* | grep Codec is Codec: IDT 92HD91BXX Codec: Intel PantherPoint HDMI I need to get both the IDT card to work and the HDMI card to work with my TV. --EDIT-- --EDIT 2-- I have added blacklist snd-usb-audio to the end of the file /etc/modprobe.d/alsa-base.conf Now the sound plays from my laptop speakers only when I plug in a headphones/external speaker. Otherwise no sound. :( Please help getting everything working as it should. --EDIT 2-- Thanks

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  • About AMR audio file playing issue on different devices

    - by user352537
    I have got a quite strange problem here. I am developing an IM software and need to play audio files recorded by another client on Android. The same audio file I've got can be played with AVAudioPlayer on 3GS(IOS 4.2.1) device and simulator 4.2. But when I tried by play it on iPhone4(iOS 4.3.3), the function "play" always return NO. I also tried with two iPhone devices, the audio files recorded by iPhone client can be played on both 3GS and iPhone4. So I asked the Android developers about the record parameters they've used. They said that the "AudioEncoder" used by them was "DEFAULT". There are also some other parameters as following: **private AudioEncoder() {} public static final int DEFAULT = 0; /** AMR (Narrowband) audio codec */ public static final int AMR_NB = 1; /** @hide AMR (Wideband) audio codec */ public static final int AMR_WB = 2; /** @hide AAC audio codec */ public static final int AAC = 3; /** @hide enhanced AAC audio codec */ public static final int AAC_PLUS = 4; /** @hide enhanced AAC plus audio codec */ public static final int EAAC_PLUS = 5;** Does anybody know what's the matter?

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  • Writing an audio player in C#

    - by Malki
    Hi, I have a pretty cool idea for a very special media player. I like to think about this project as a mini-startup, since I don't yet know if my idea is practical. Anyways, before implementing my idea, I first need to be able to implement a simple audio player. My preferred language for this project is C#, simply because it's so easy to use, but any other object oriented language would be fine too I guess. I started out with no knowledge whatsoever about audio. My main goals right now are: Being able to play audio files - as many formats as possible (sort of a VLC type player, but only audio for now). Being able to analyze audio files - as in, reading frequency, amplitude, volume, and other information about the audio. I think maybe a good idea here is to be able to analyze one file format (PCM?), and then temporarily converting any file I want to analyze to that format. This is in order to later implement a mechanism that compares songs and identifies similar songs to recommend to the user (this feature isn't part of my idea, but I figured since it exists in many players nowadays, I need to have it too if I want be able to compete with them). BTW - I currently don't have any knowledge about audio/wavelengths/frequencies and such, so I'd appreciate it if someone could point me in the right direction about this analyzation feature. Maybe in the future I'd expand to playing video files as well, but for now I'm concentrating on audio. After searching the Internet for a while, I've come across LAME. Problem is, it's not C#, and I'm not sure how to use it. I know there is something called "Interoperability", that is supposed to let me work with native DLL files through C#. Any information about that would be helpful as well. Any help would be much appreciated. Thanks, Malki :)

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  • iPhone SDK Smaller CAF files: lower recording quality with Audio Queues?

    - by Nick
    In my iPhone app I have voice recording functionality the utilizes Audio Queue voice recording functions of the SDK. I'm saving directly to CAF format and using the following settings for the AudioStreamBasicDescription reference: audioFormat.mFormatID = kAudioFormatLinearPCM; I can see that there are other format ids I could use like: kAudioFormatLinearPCM kAudioFormatAppleLossless kAudioFormatAppleIMA4 kAudioFormatiLBC kAudioFormatULaw kAudioFormatALaw My knowledge of sound formats is very limited so my question is... which of these should I use to create the lowest compressed audio recording files? Plus, are there other settings I should apply to lower the quality and filesize even further?

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  • How to program a real-time accurate audio sequencer on the iphone?

    - by Walchy
    Hi... I want to program a simple audio sequencer on the iphone but I can't get accurate timing. The last days I tried all possible audio techniques on the iphone, starting from AudioServicesPlaySystemSound and AVAudioPlayer and OpenAL to AudioQueues. In my last attempt I tried the CocosDenshion sound engine which uses openAL and allows to load sounds into multiple buffers and then play them whenever needed. Here is the basic code: init: int channelGroups[1]; channelGroups[0] = 8; soundEngine = [[CDSoundEngine alloc] init:channelGroups channelGroupTotal:1]; int i=0; for(NSString *soundName in [NSArray arrayWithObjects:@"base1", @"snare1", @"hihat1", @"dit", @"snare", nil]) { [soundEngine loadBuffer:i fileName:soundName fileType:@"wav"]; i++; } [NSTimer scheduledTimerWithTimeInterval:0.14 target:self selector:@selector(drumLoop:) userInfo:nil repeats:YES]; In the initialisation I create the sound engine, load some sounds to different buffers and then establish the sequencer loop with NSTimer. audio loop: - (void)drumLoop:(NSTimer *)timer { for(int track=0; track<4; track++) { unsigned char note=pattern[track][step]; if(note) [soundEngine playSound:note-1 channelGroupId:0 pitch:1.0f pan:.5 gain:1.0 loop:NO]; } if(++step>=16) step=0; } Thats it and it works as it should BUT the timing is shaky and instable. As soon as something else happens (i.g. drawing in a view) it goes out of sync. As I understand the sound engine and openAL the buffers are loaded (in the init code) and then are ready to start immediately with alSourcePlay(source); - so the problem may be with NSTimer? Now there are dozens of sound sequencer apps in the appstore and they have accurate timing. I.g. "idrum" has a perfect stable beat even in 180 bpm when zooming and drawing is done. So there must be a solution. Does anybody has any idea? Thanks for any help in advance! Best regards, Walchy

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  • Best practice for C++ audio capture API under Linux?

    - by braddock
    I need to create a C++ application with a simple audio recording from microphone functionality. I can't say that there aren't enough audio APIs to do this! Pulse, ALSA, /dev/dsp, OpenAL, etc. My question is what is the current "Best practice" API? Pulse seems supported by most modern distros, but seems almost devoid of documentation. Will OpenAL be supported across different distros, or is it too obscure? Have I missed any? Is there not a simple answer? thanks!

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  • Why does my audio keep reverting to stereo?

    - by GraemeF
    I have an ASUS P7P55D LE motherboard with onboard sound and I am running Windows 7 64-bit, and I am using the SPDIF output from my motherboard to my receiver. On my receiver I can see which audio channels are in use (i.e. stereo or 5.1) In the audio interface properties I can test DTS Audio and Dolby Digital outputs and both work fine, but when I try to play a game with 5.1 sound (I've tried Left 4 Dead and Dragon Age Origins) it reverts to stereo. I was getting this behaviour with the default Microsoft drivers so I installed the latest from the ASUS website but there is no difference. I notice that in the Advanced tab the Default Format only allows me to choose from various 2 channel formats so maybe that is something to do with it? How can I get 5.1 output all of the time?

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  • Windows 7 Audio problem..

    - by Marcx
    hi, I've a dell studio 1555 bought on september with Windows 7 64bit Professional on it. Audio devide works proprerly, while listening/watching audio/video contents (from disk or internet) Using VoIP Program Ventrilo Audio from others people works good, and I hear their voices clearly... Using VoIP Program such as Teamspeak 2/3, MSN, Skype I hear a really disturbed voice, and it's impossible to comprehend somethings... Anyway I can say that the first week that I've bought and installed Win7 Skype worked good, this problem happened after I installed Ventrilo... I tried to remove it (ventrilo) but nothing happened:( Please help me :D Thanks Marcx

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