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  • 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?

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  • 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?

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  • 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 ?

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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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  • 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 ?

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  • 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.

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  • Video encoding is very slow on Amazon EC2 instance

    - by Timka
    We are using Amazon EC2 m1.xlarge instance for video re-encoding and it looks like the actual encoding process takes a very long time. For an average 250mb video file it takes about an hour to encode. Intance: m1.xlarge (Xeon E5645 x 15gb ram) Windows Server 2008 R2 64-bit AviSynth version 2.5 (32bit) + ffms2 plugin (FFmpegSource 1.21) FFmpeg SVN-r13712 libavutil 3213056 libavcodec 3356930 libavformat 3411456 libavdevice 3407872 Number of parallel jobs is 3 Average CPU utilization ~96% Update#1 Source video: mp4/h.264 Parameters for ffmpeg: --enable-memalign-hack --enable-avisynth --enable-libxvid --enable-libx264 + --enable-libgsm --enable-libfaac --enable-libfaad --enable-liba52 + --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-pthreads + --enable-swscale --enable-gpl Video files encoded to mp4/h.264 with the following extra command line options: -threads 0 -coder 0 -bf 0 -refs 1 -level 30 -maxrate 10000000 -bufsize 10000000

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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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  • 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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  • c# HTTPListener encoding issue

    - by Rob Griffin
    I have a Java application sending HTTP requests to a C# application. The C# app uses HTTPListener to listen for requests and respond. On the Java side I'm encoding the URL using UTF-8. When I send a \ character it gets encoded as %5C as expected but on the C# side it becomes a / character. The encoding for the request object is Windows-1252 which I think may be causing the problem. How do I set the default encoding to UTF-8? Currently I'm doing this to convert the encoding: foreach (string key in request.QueryString.Keys) { if (key != null) { byte[] sourceBytes =request.ContentEncoding.GetBytes(request.QueryString[key]); string value = Encoding.UTF8.GetString(sourceBytes)); } } This handles the non ASCII characters I'm also sending but doesn't fix the slash problem. Examining request.QueryString[key] in the debugger shows that the / is already there.

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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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  • International JRE6 or JDK6 or reading a file in "cp037" encoding scheme

    - by Reddy
    I have been trying to read a file in "cp037" encoding scheme using JAVA. I able to read a file in basic encoding schemes like UTF-8, UTF16 etc...After a bit of research on the internet i came to know that we need charset.jar or international version of JRE be installed to support extended encoding schemes. Can anyone send me a link for international version of JRE6 or JDK6. or is there any better way that i could read a file in cp037 encoding scheme. P.S: cp037 is a character encoding scheme supported by IBM Mainframes. All i need is to display a file in windows, which is being generated on IBM Mainframes machine, using a java program. Thanks in advance for your help... :-)

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  • What encoding to use for exporting to CSV?

    - by Michael Borgwardt
    I'm developing a java app that exports data to CSV files, intended to be opened in Excel by end users. We just noticed that the export function uses Java's platform default encoding. This causes umlaut characters to be lost and unit test to fail on the build server (which is configured to have US-ASCII as its platform default encoding exactly to catch such potential problems). The question is: which would be the best encoding to use? How does Excel determine what encoding to use? Does it use something platform-specific that presumably matches Java's platform default? I'm currently leaning towards hardcoding Cp1252 - that should cover the target machines (the deployment environment is actually specified) and would fix the test problem. From googling around, Excel does not seem to handle UTF-8 well, so that's out, and sticking to the platform default encoding would require some sort of workaround hack for the tests.

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  • ADSL throughput loss from Reed-Solomon encoding

    - by javano
    I'm reading about ADSL starting here and I am confused by how the Reed-Solomon encoding for ECC is limiting the available transfer rate, as much as it does (nearly half). This pdf on the same subject contains the following; A maximum of 255 sub-carriers can be used to modulate data in the downstream direction. Sub-carrier 256, the downstream Nyquist frequency, and sub-carrier 64, the downstream pilot frequency, are not available for user data, thus limiting the total number of available downstream sub-carriers to 254. Each of these 254 sub-carriers can support the modulation of 0 to 15 bits. Since the ADSL DMT data frame rate is 4000 frames per second, the maximum theoretical downstream data rate of an ADSL system is 15.24Mbps. Due to limitations in system architecture, specifically the maximum allowable Reed-Solomon codeword size (255 bytes), the maximum achievable downstream data rate is 8.16Mbps. How is this nearly halving the throughput? Is all that extra bandwidth overhead of the RS encoding? 15240000 bps (15.24Mbps) - 8160000 bps (8.12Mbps) = 7080000 bps (7.08Mbps). Where has that 7Mbps of throughput gone? EDIT: I tried to read the wiki page on Reed-Soloman but it's all crazy maths and algerbra, which I don't understand. I can understand that data is split into 255 byte codewords, because that maybe the max codeword size whilst still maintaining accuracy during transmission; But I don't understand why that means less data is sent?

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  • Email encoding on IIS7

    - by Ivanhoe123
    All emails sent from the server are displaying Cyrillic letters as weird characters, for example: Можно. Regular alphabet letters are properly rendered. I searched all across the web but was not able to find any solutions. Here is some information about the system: Dedicated server with Windows 2008 and IIS7 Application are in PHP (run as FastCGI) If of any importance, Smartermail is installed on the server The emails are sent using PHPs mail() function through a Drupal website. Encoding on that site is set up properly and there are no display issues on front end. Where is the problem? How can I make Cyrillic letters to be properly encoded? Any help is greatly appreciated. Thanks! UPDATE Here are the email headers: Received: from SERVERNAME (mail.domain.com [12.123.123.123]) by mail.domain.com with SMTP; Fri, 16 Nov 2012 00:00:00 +0100 From: [email protected] To: [email protected] Subject: Email subject Date: Fri, 16 Nov 2012 00:00:00 +0100 MIME-Version: 1.0 Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable X-Mailer: Drupal Sender: [email protected] Return-Path: [email protected] Message-ID: f98b801988c642ef911ef46f7cace92b@com X-SmarterMail-Spam: SPF_None, ISpamAssassin 8 [raw: 5], DK_None, DKIM_None, Custom Rules [] X-SmarterMail-TotalSpamWeight: 8

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  • Force encoding with IIS 7

    - by Cédric Boivin
    I try to force encoding with IIS 7. When I add in the http response headers the key : Content-Type and value charset=utf-8 i got this key content-type : text/html,content-type=utf-8 it's there a way to remove the comma ? Thanks Justin for your answer. But it's seen don't work. There is my config, i need to do that for asp classic. <?xml version="1.0" encoding="UTF-8"?> <configuration> <system.webServer> <staticContent> <remove fileExtension=".html" /> <remove fileExtension=".hxt" /> <remove fileExtension=".htm" /> <remove fileExtension=".asp" /> <mimeMap fileExtension=".htm" mimeType="text/html" /> <mimeMap fileExtension=".hxt" mimeType="text/html" /> <mimeMap fileExtension=".html" mimeType="text/html" /> <mimeMap fileExtension=".asp" mimeType="text/html; charset=UTF-8" /> </staticContent> </system.webServer> </configuration>

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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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  • Changing character encoding in MySQL, PHP scripts, HTML

    - by Sandman
    So, I have built on this system for quite some time, and it is currently outputting Latin1 (ISO-8859-1) to the web browser, and this is the components: MySQL - all data is stored with the Latin1 character set PHP - All PHP text files are stored on disk with Latin1 encoding HTML - The output has the http-equiv="content-type" content="text/html; charset=iso-8859-1" meta tag So, I'm trying to understand how the encoding of the different parts come into play in my workflow. If I open a PHP script and change its encoding within the text editor to UTF-8 and save it back to disk and reload the web browser, the text is all messed up - unless the text comes from the DB. If I change the encoding of the DB to UTF-8 and keep the PHP files in latin1 I have to use utf8_decode() for the data to display correctly. And if I change the HTML code the browser will read it incorrectly. So yeah, I realise that if I want to "upgrade" to UTF8, I have to update all three parts of this setup for it to work correctly, but since it's a huge system with some 180k lines of PHP code and millions of posts in a lot of databases/tables, I don't want to start something like this without understanding everything correctly. What haven't I thought about? What could mess this up beyond fixing? What are the procedures for changing the encoding of an entire MySQL installation and what's the easiest way to change the encoding of hundreds or thousands of PHP files on disk? The META tag is luckily added dynamically, so I'll change that in one place only :) Let me hear about your experiences with this.

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