Search Results

Search found 5088 results on 204 pages for '3 5mm audio jack'.

Page 7/204 | < Previous Page | 3 4 5 6 7 8 9 10 11 12 13 14  | Next Page >

  • Choose an output audio device different from the default on WMP 11

    - by GetFree
    I like to play my music through a Hi-Fi audio equipment and everything else (like windows sounds, web videos and such) through my default PC speakers. On WIndows XP I had WMP 9 and I could do that with no problems since I can choose what audio device (which sound card) to use, and that selection is for WMP only, which can be different from Windows' default audio device. But now that I have Windows Vista and WMP 11 I cannot longer choose an audio device just for WMP, or at least I can't find a way to do it (the control in the options dialog is no longer there). Was this useful feature really removed from WMP 11? or there is some other way to do it?

    Read the article

  • Free alternative to Audio Hijack Pro?

    - by Tim Visher
    I'd like to record what I hear coming out of the main audio jack on my Mac. Nothing fancier than that. I'm aware of Audio Hijack Pro but that really does much more than I'm looking for and comes with a steep price tag. If it's the only tool that can do the job that's fine but I was hoping to find something that simply captured all audio coming from the computer and dumped it to a file. Any suggestions?

    Read the article

  • Sony Bravia (KDL-32EX402) audio connection fails

    - by Rasmus
    Hey. I'm having trouble connecting my computers audio to the TV (Sony Bravia KDL-32EX402). I'm using a standard AUX cable (some kind of adapter for L/R to go into the headphone plug on the computer). I'm connecting the other end to the back of the TV, it doesn't actually say "AUDIO IN" but it has to be (is also right below the "HDMI 1 AUDIO IN). When changing to "PC-mode" and setting the audio input to "PC" nothing happens. (but the pictures get's transmitted fine by VGA). I have checked that it's not the PC's headphone port, nor the AUX cable. What to do, what to do ?

    Read the article

  • 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

    Read the article

  • How to check properties of an audio [closed]

    - by Ashni Goyal
    Possible Duplicate: Tool to view video/audio file information Soundeffect class in WP7 requires following properties of the .wav file. The Stream object must point to the head of a valid PCM wave file. Also, this wave file must be in the RIFF bitstream format. The audio format has the following restrictions: Must be a PCM wave file Can only be mono or stereo Must be 8 or 16 bit Sample rate must be between 8,000 Hz and 48,000 Hz How can we check these properties for a given audio ?

    Read the article

  • Configuring capture card to get audio out to the speakers

    - by TheTedH
    I have been using my capture card as a video in device so I can play my video game systems without having to go to the TV in another room. I have been trying to get the audio to play through line-in, to no avail. Would there be a way to make the audio from line-in to output the video game console audio from the computer? Also, I'm on Windows XP SP2.

    Read the article

  • 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.

    Read the article

  • '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.

    Read the article

  • 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

    Read the article

  • 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?

    Read the article

  • 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"/>

    Read the article

  • 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.

    Read the article

  • 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

    Read the article

  • 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.

    Read the article

  • 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.

    Read the article

  • 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)

    Read the article

  • <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!

    Read the article

  • 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>

    Read the article

  • 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.

    Read the article

  • 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.

    Read the article

  • 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

    Read the article

  • 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

    Read the article

  • 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?

    Read the article

< Previous Page | 3 4 5 6 7 8 9 10 11 12 13 14  | Next Page >