Search Results

Search found 4010 results on 161 pages for 'audio fingerprinting'.

Page 20/161 | < Previous Page | 16 17 18 19 20 21 22 23 24 25 26 27  | Next Page >

• Unable to configure/setup 5.1 audio with 12.04

  - by Vipin Vinayan

  I am kinda new to Ubuntu as well. I have been having this issue with audio for quite sometime now. Initially, when I installed version 11.10 (I guess), I was able to use my 5.1 speakers without any issues. If my memory serves me right, it was after an update that the 5.1 audio stopped working and the video resolution would not get saved. I temporarily fixed the resolution issue by creating a start-up shell script that would update the resolution and load it. But the issue with audio has been going on for quite sometime now. Even though I have option for 5.1, only two speakers seem to be working. I thought an upgrade should fix the issue and so upgraded the OS to version 12.04. I also tried uninstalling alsa and pulse audio, reinstalling them, changing the /etc/pulse/daemon.conf channels from 2 to 6. I have also tried installing pavucontrol but nothing seems to have worked and the issue still persists. Is there anything else you could suggest? The lspci log on my computer is as follows 00:00.0 Host bridge: Intel Corporation 82G33/G31/P35/P31 Express DRAM Controller (rev 10) 00:01.0 PCI bridge: Intel Corporation 82G33/G31/P35/P31 Express PCI Express Root Port (rev 10) 00:02.0 VGA compatible controller: Intel Corporation 82G33/G31 Express Integrated Graphics Controller (rev 10) 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) 00:1c.0 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 1 (rev 01) 00:1c.1 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 2 (rev 01) 00:1d.0 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #1 (rev 01) 00:1d.1 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #2 (rev 01) 00:1d.2 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #3 (rev 01) 00:1d.3 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #4 (rev 01) 00:1d.7 USB controller: Intel Corporation N10/ICH 7 Family USB2 EHCI Controller (rev 01) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation N10/ICH7 Family SATA Controller [IDE mode] (rev 01) 00:1f.3 SMBus: Intel Corporation N10/ICH 7 Family SMBus Controller (rev 01) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) Would really appreciate a response that will assist me in resolving my issue. Thanks in advance Vipin

  Read the article

• Vista Core Audio API Master Volume Control

  A managed wrapper for accessing the Vista Core Audio API

  Read the article

• Distorted choppy audio in Precise

  - by Misery

  After installing Precise on my PC, some problems with soud occure. While using Lucid there were no problems. The sound is choppy and distorted in low tones range. As I absolutely have no experience in setting/testing and doing anything with Audo Devices I need help even to diagnose the problem. update: sudo lshw -c multimedia *-multimedia description: Audio device product: Radeon X1200 Series Audio Controller vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 5.2 bus info: pci@0000:01:05.2 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi bus_master cap_list configuration: driver=snd_hda_intel latency=32 resources: irq:19 memory:fdafc000-fdafffff *-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 update 2: It has something to do with the volume. If the audio is quiet it is not choppy, if the sound is loud then it begins to be choppy. Regards, Misery

  Read the article

• Audio not working in 12.10

  - by frampy

  I did a clean install of 12.10, when I open Sound Settings in gnome the only device in the list is "Dummy Output", and sound is not working. Sound worked fine out of the box in 12.04 I ran alsamixer, it says my card is "HDA Intel", and chip is "Realtek ALC880". The alsamixer playback output was set to mute at first, unmuting did not fix. I checked out the info at http://www.unixmen.com/2012003-howto-resolve-nosound-problem-on-ubuntu/ as suggested on a similar question, I've done everything there except installing the ubuntu audio dev team driver. Should I try install this? Edit: I've been reading the sound troubleshooting guide at https://help.ubuntu.com/community/SoundTroubleshooting It looks like Ubuntu is finding my audio device correctly. mike@wucade:~$ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) Subsystem: Albatron Corp. Device 2668 Flags: bus master, fast devsel, latency 0, IRQ 40 Memory at d01c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel Still stuck as to why this isn't working.

  Read the article

• Synced audio ouput on multiple machines? VLC? hardware solutions?

  - by zimmer62

  I'm wondering if there is any software or hardware solutions to synced audio or audio and video across multiple computers or devices on a network. I've seen Sonos, and it might be a good solution, but it's also a very expensive solution. I'd like to be able to play something with realtime audio output on one PC, but hear it on speakers throughout the house, being it the home theater receiver, or another computer in another room. I saw a solution using the apple iport express, but the latency was unacceptable for anything other than just music. I'd like to avoid running audio wires with baluns to a bunch of amplifiers scattered all over the place when I have cat5 run everywhere. Is anyone familiar with using this kind of process for whole home audio? The latency is a big deal for me, if I've got video attached to the sound (e.g. watching a hockey game)

  Read the article

• Audio tag doesn't work in FF-Ubuntu 12.04

  - by Nyx

  Does anyone know why this code... <audio width="0" height="0" autoplay="autoplay" loop="loop" preload="none"> <source src="images/musica/Intro.ogg" type="audio/ogg" /> <source src="images/musica/Intro.mp3" type="audio/mpeg" /> </audio> ...works fine in FF17-WinXP and not in FF17-Ubuntu 12.04? I think something is wrong with MIME types but everything looks normal. After searching on the web for days I couldn't find a good answer. Thanks

  Read the article

• How to play multiple audio sources simultaneously in Silverlight

  - by Shurup

  I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound). Addition: I've tested a code from the Bobby's answer. private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); // This code plays well enough. // me.Play(); // me2.Play(); // But adding the 2 second offset using the timer, // they play no simultaneous. var timer = new DispatcherTimer { Interval = TimeSpan.FromSeconds(2) }; timer.Tick += (source, arg) => { me2.Play(); ((DispatcherTimer)source).Stop(); }; timer.Start(); } Is it possible to play them together using only one MediaElement or any implementation of MediaStreamSource that can play multiply sources?

  Read the article

• How can I stream audio signals from various devices/computers to my home server?

  - by Breakthrough

  I currently have a headless home server set up (running Ubuntu 12.04 server edition) running a simple Apache HTTP server. The server is near an audio receiver, which controls a set of indoor and outdoor speakers in my home. Recently, my father purchased a Bluetooth adapter, which our various laptops and cellphones can connect to, outputting the music to the speakers. I was hoping to find a solution that worked over Wi-Fi, namely because it won't cost anything (I already have a server with an audio card), and it doesn't depend on Bluetooth. Is there any cross-platform (preferably free and open-source) solution that I can use which will allow me to stream audio to my home server, over my home network, from a wide variety of devices (laptops running Windows/Linux or cellphones running Android/BB/iOS)? I need something that works at least with Windows and Android. Also, just to clairfy, I want something that simply allows devices to connect to my server and output an audio signal without any action on the server end (since it's a server hidden away near my receiver). Any subsequent connection attempt should be dropped, so only one device can be in control of the stereo at once.

  Read the article

• Help with dynamic range compression function (audio)

  - by MusiGenesis

  I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think): public static void Compress(ref short[] input, double thresholdDb, double ratio) { double maxDb = thresholdDb - (thresholdDb / ratio); double maxGain = Math.Pow(10, -maxDb / 20.0); for (int i = 0; i < input.Length; i += 2) { // convert sample values to ABS gain and store original signs int signL = input[i] < 0 ? -1 : 1; double valL = (double)input[i] / 32768.0; if (valL < 0.0) { valL = -valL; } int signR = input[i + 1] < 0 ? -1 : 1; double valR = (double)input[i + 1] / 32768.0; if (valR < 0.0) { valR = -valR; } // calculate mono value and compress double val = (valL + valR) * 0.5; double posDb = -Math.Log10(val) * 20.0; if (posDb < thresholdDb) { posDb = thresholdDb - ((thresholdDb - posDb) / ratio); } // measure L and R sample values relative to mono value double multL = valL / val; double multR = valR / val; // convert compressed db value to gain and amplify val = Math.Pow(10, -posDb / 20.0); val = val / maxGain; // re-calculate L and R gain values relative to compressed/amplified // mono value valL = val * multL; valR = val * multR; double lim = 1.5; // determined by experimentation, with the goal // being that the lines below should never (or rarely) be hit if (valL > lim) { valL = lim; } if (valR > lim) { valR = lim; } double maxval = 32000.0 / lim; // convert gain values back to sample values input[i] = (short)(valL * maxval); input[i] *= (short)signL; input[i + 1] = (short)(valR * maxval); input[i + 1] *= (short)signR; } } and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios. Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however. I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this: Can anybody suggest a modification to my function to produce the soft knee curve?

  Read the article

• Why does 3.5 mm audio out work through headphones but not through external speakers?

  - by Rickster

  I have a computer that has a 3.5 mm audio jack on the front of it. The computer itself has no speakers, so this is the only way to hear sound. If I plug headphones into it, the audio properly plays through the headphones, and if I plug in external speakers it used to play through them as well. Just today I turned on my computer and the audio no longer plays through the speakers, but if I plug in the headphones instead it works. The speakers aren't broken, as both the speakers and headphones work in my iPod and play music. I thought that 3.5 mm jacks could not send data back to the computer, and the computer had no way of differentiating between different devices plugged into the jack. If this is true, how is it that the computer plays audio through the headphones but not through speakers plugged into the same 3.5 mm jack, and both devices are functional? Or is my knowledge on 3.5 mm jacks incorrect? I don't believe drivers are important, as the same driver runs the 3.5 mm jack for all devices, but if necessary I can provide additional information. Any ideas would be appreciated. Thanks!

  Read the article

• Getting notified when a sound is done playing in OpenAL

  - by Mel

  I'm using OpenAL on iPhone to play multiple audio samples simultaneously. Can I get OpenAL to notify me when a single sample is done playing? I'd like to avoid hardcoding the sample length and setting a timer.

  Read the article

• How can I stop Ubuntu from playing audio from 2 interfaces at the same time?

  - by Solignis

  Hi there, I just loaded Ubuntu 10.10 Maverick on my home machine. The machine is running a Core2Duo E6750 on an MSI motherboard with an Nvidia GTX260-OC Graphics card. The problem I am having as stated in the title is for some reason Ubuntu is playing audio through my headphone coming out from the computer and it is also playing the audio at the exact same time through the HDMI connection coming out of the graphics card, it has a plug to allow this. What is going on, I have never seen this before. Most importantly of all can it be fixed so that I can sepertate the 2 interfaces, the one is a standard PC audio IO and the HDMI one is connected through the mobo's internal SPDIF. More information can be provided if required.

  Read the article

• Default audio device gives an error on WINDOWS 7 (x64) when triing to run VLC from CMD (VideoLAN, VL

  - by Ole Jak

  I use WINDOWS 7 (x64) (Russian) I want to stream life audio from my default audio capture device (microphone) When I set up VLM settings using visual enviroment instruments - VLM settings it all works fine. But when I export created settings/configuration *.vlm file and try to inport it into VLM it gives me nothing I opened that .vlm there is some text... so now I try to run VLC with default settings like this: vlc -i dshow:// --dshow-adev= :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100}:std{access=http,mux=raw,dst=127.0.0.1:8084} but it dies giving me errors...=( So what shall I do to do live MP3 streaming from my default audio input device using VLC in non UI mode?

  Read the article

• Can I reprogram a microphone input to be used as an audio output? (on XP)

  - by qftme

  I have a five year old Sony Vaio laptop (vgn-fw31m) that has had impact damage to the audio-output mini-jack for about the last year or so. In a recent discussion with my brother, we wondered whether it would be possible to write a program that would enable windows to use the microphone mini-jack input as the audio-output? As I currently use this laptop for work I am not keen to risk pulling it apart in order to replace the components comprising the audio-out. I therefore 'hope' that a programming solution exists. I would really appreciate any advice on this and eagerly await your response. Kind regards, qftme :)

  Read the article

• Order by text and then by number

  - by Chaim Chaikin

  I have data like: Audio 1 File 10 Audio 2 Audio 3 File 11 Audio 13 Audio 22 File 20 Test 22 Audio 10 File 1 File 2 I need it order first by the text (i.e. Audio, File, Test) and then by number (1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22 etc.) The problem is that sorting it returns something like this: Audio 1 Audio 10 Audio 13 Audio 2 Audio 22 Audio 3 File 1 File 10 File 11 File 2 File 20 Test 22 While the result I want is: Audio 1 Audio 2 Audio 3 Audio 10 Audio 13 Audio 22 File 1 File 2 File 10 File 11 File 20 Test 22 If they were just numbers (i.e. without the audio, file, test) then I could just sort numerically. However, how can I sort here first by text and then by number.

  Read the article

• KDE: How can I select audio output device for mplayer?

  - by grimripper

  I recently installed Kubuntu 13.10 64-bit, and I'm having a problem with selecting audio output device. In Phonon, when I select audio device preference order and press Apply, Amarok and Dragon will immediately switch to the preferred device. VLC and SMplayer are not affected. VLC has its own setting for selecting the output device, but SMplayer remains a problem. It always plays audio on internal audio, and I can't change output to HDMI. How can I select HDMI for SMplayer's audio output device? I don't know if it matters, but when I select HDMI audio in Phonon and click Test, the test sound plays on the internal audio output as well. In the hardware settings tab, the front left and front right test buttons play audio on HDMI. Also, volume up/down buttons affect HDMI volume when SMplayer is focused. This would make sense if I could get SMplayer to play audio over HDMI, but it would be better if the volume keys affected SMplayer's own volume, or the "mplayer2: audio stream" which appears in volume control while mplayer is playing. EDIT: I've recompiled mplayer with alsa support, and can now select the audio output device from SMplayer's settings. Didn't affect the issue with Phonon of course, but it's a suitable workaround.

  Read the article

• AVAudioRecorder - Continue recording to file after user stops recording by leaving the application a

  - by Tegeril

  Can this be done? And if not, how far down towards Core Audio do I need to go (what method of recording should I be using instead)? I've noticed the behavior of AVAudioRecorder is to overwrite a file if it finds one at the path provided when you request that it record again, so I know that's not going to work. I'm also curious about file format restriction with this idea. Can you effectively resume an AAC or IMA4 encoding (the length of the files I want to record make WAV and probably even Apple Lossless prohibitive)? Thanks.

  Read the article

• Does anyone have any idea about any songCatcher library for Linux? or any idea on how to use "Audio-

  - by Bobby

  Hi All, Does anyone have any idea about any songCatcher library for Linux? songCatcher is based on the acoustic fingerprints. I managed to find one "Audio-Ofa-Util-0.04" but I'm facing problem in using it. Or if someone can help me understand its usage. Like: how can I pass an audio file to this tool. Any specific command for its usage etc. Any help would be highly appreciated. Thanks, Bobby

  Read the article

• How to get audio driver for compaq c700 ?

  - by Leena

  Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

  Read the article

• How to get audio driver for compaq c700 ?

  - by Leena

  Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

  Read the article

• How to get audio driver for compaq c700 ?

  - by Leena

  Hi, Initially i have audio driver and its works fine.But some times speaker was clear.So one of my friend installed some audio driver,after that totally disabled the volume. For that reason, i also tried to get audio driver and installed many times.Now i don't know many drivers .inf in my laptop.from device manager i have deleted the audio driver's,below i have attached the screen shot yours kind reference. Please help me to get audio drivers.First, i need to remove the unwanted drivers .inf files from laptop then i have to install the new audio driver. Experts,please suggest me to get audio driver without reinstall the OS. Details: Compaq c700 (i don't know model number) windows xp sp2 p/n : KT188PA#ACJ I appreciate your help.

  Read the article

• Streaming Audio over UDP to Android

  - by Mr. Pig

  Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

  Read the article

• Audio recording and playback in Silverlight

  - by Ramesh

  I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

  Read the article

• Android - Getting audio to play through earpiece

  - by Donal Rafferty

  I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

  Read the article

• Core-audio - constructing an AudioBufferList struct (Q about c struct definition)

  - by mustISignUp

  The definition of AudioBufferList looks weird to me… i guess my C is not so good struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[kVariableLengthArray]; }; typedef struct AudioBufferList AudioBufferList; Why AudioBuffer mBuffers[kVariableLengthArray]; and not AudioBuffer *mBuffers; ? kVariableLengthArray appears to be == 1. Eh? I think i have it working but would appreciate it if anyone could set me straight.

  Read the article

< Previous Page | 16 17 18 19 20 21 22 23 24 25 26 27  | Next Page >