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  • USB audio device not detected [solved]

    - by user199052
    I've been having problems with USB sound for some time (it used to work about 6 months ago). I find that any USB sound device that I connect is not recognised, and is not listed by aplay -l I've tried disabling pulseaudio. I've tried purging and re-installing alsa-base, pulseaudio, and pavucontrol. To no avail. I'm using 12.04 LTS. I found the solution: 12.04 does not have the needed kernel module snd-usb-hiface this is introduced in the 3.11.1 kernel and installing that on 12.04 gives USB sound for my equipment. How to install the 3.11.1 kernel is given here:- installing 3.11.1 kernel

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  • How to send audio stream via UDP in java?

    - by Nob Venoda
    Hi to all :) I have a problem, i have set MediaLocator to microphone input, and then created Player. I need to grab that sound from the microphone, encode it to some lower quality stream, and send it as a datagram packet via UDP. Here's the code, i found most of it online and adapted it to my app: public class AudioSender extends Thread { private MediaLocator ml = new MediaLocator("javasound://44100"); private DatagramSocket socket; private boolean transmitting; private Player player; TargetDataLine mic; byte[] buffer; private AudioFormat format; private DatagramSocket datagramSocket(){ try { return new DatagramSocket(); } catch (SocketException ex) { return null; } } private void startMic() { try { format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 8000.0F, 16, 2, 4, 8000.0F, true); DataLine.Info info = new DataLine.Info(TargetDataLine.class, format); mic = (TargetDataLine) AudioSystem.getLine(info); mic.open(format); mic.start(); buffer = new byte[1024]; } catch (LineUnavailableException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } private Player createPlayer() { try { return Manager.createRealizedPlayer(ml); } catch (IOException ex) { return null; } catch (NoPlayerException ex) { return null; } catch (CannotRealizeException ex) { return null; } } private void send() { try { mic.read(buffer, 0, 1024); DatagramPacket packet = new DatagramPacket( buffer, buffer.length, InetAddress.getByName(Util.getRemoteIP()), 91); socket.send(packet); } catch (IOException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void run() { player = createPlayer(); player.start(); socket = datagramSocket(); transmitting = true; startMic(); while (transmitting) { send(); } } public static void main(String[] args) { AudioSender as = new AudioSender(); as.start(); } } And only thing that happens when I run the receiver class, is me hearing this Player from the sender class. And I cant seem to see the connection between TargetDataLine and Player. Basically, I need to get the sound form player, and somehow convert it to bytes[], therefore I can sent it as datagram. Any ideas? Everything is acceptable, as long as it works :)

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  • Streaming Audio from A URL in Android using MediaPlayer?

    - by Sena Gbeckor-Kove
    Hi, I've been trying to stream mp3's over http using Android's built in MediaPlayer class. The documentation would suggest to me that this should be as easy as : MediaPlayer mp = new MediaPlayer(); mp.setDataSource(URL_OF_FILE); mp.prepare(); mp.start(); However I am getting the following repeatedly. I have tried different URLs as well. Please don't tell me that streaming doesn't work on mp3's. E/PlayerDriver( 31): Command PLAYER_SET_DATA_SOURCE completed with an error or info PVMFErrNotSupported W/PlayerDriver( 31): PVMFInfoErrorHandlingComplete E/MediaPlayer( 198): error (1, -4) E/MediaPlayer( 198): start called in state 0 E/MediaPlayer( 198): error (-38, 0) E/MediaPlayer( 198): Error (1,-4) E/MediaPlayer( 198): Error (-38,0) Any help much appreciated, thanks S

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  • Is the Finch audio library for iPhone capable of doing this?

    - by mystify
    I need to: - start / stop sounds with lengths between 0.1 and 10 seconds - change the playback volume I want to / would like to / would be nice to have to: - change the playback speed - change the playback pitch / frequency - pause an sound and resume playing it later - play a sound backwards Is Finch my best friend here?

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  • how to get audio frequency data from a wave file?

    - by potlee
    I want to build a speech recognition engine in ruby. I know i'll never get there, doing it just for fun. I need to get data for the frequencies of the sound stored in a wav file to compare with data i already have of different sounds that i want to recognize. I will write the code in ruby but i dont think there are any libraries for this written in ruby, they would be too slow if there were any anyway. The good thing about ruby is I'll be able to use libraries for .net via IronRuby or Java via Jruby. How can i get the frequency data?

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  • How to route KVM virtual machine audio to Ubuntu 11.10 host using virt-manager?

    - by iGadget
    I've been using KVM in combination with Virt-Manager and Remmina at a fair success up until now. The issue I need to solve now is to get audio from a virtualized Windows XP and make it audible on the Ubuntu 11.10 host. Remmina / RDP works for 'simple' audio (system sounds and such), but when the source gets trickier (e.g. Flash audio), Remmina / RDP messes up. So I figured I'd just connect to the machine directly using Virt-Manager. Unfortunately, it seems that even though I have successfully configured the AC97 audio device on WinXP, it's unable to get it's output to the Ubuntu host. This is probably because Virt-Manager uses VNC (and AFAIK, VNC doesn't transport audio). Does anyone know if there is a solution to fix this? I've heard of Spice, but the installation required so much voodoo last time I checked, I figured I'd let that solution boil to maturity a little longer ;) But perhaps there are other options I haven't thought of yet (which don't require switching to VirtualBox / VMware)...

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  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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  • Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

    - by Ryan Thompson
    To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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  • Terrible noises from subwoofer of ACER Aspire 6930 with Realtek sound chip

    - by OneWorld
    After approximately 5-15 min of listening to music my subwoofer begins to make terrible noises. He's just "coughing". That began after 6 months I had this computer. Now I found out, that I can temporarily fix this problem by "restarting" the audio stream of the application that plays music. For example reloading last.fm page (reloads the flash file). Another way to reset the audio playback is switching the speaker configuration shown below in the screenshot. According to many posts on the internet like http://www.tomshardware.co.uk/forum/52918-20-acer-aspire-6935g-speaker-problem ACER support isn't any help Exchanging hardware doesn't fix the problem Even the later models have this problem Turning off the volume of the subwoofer is not an option to me. I still have warranty (I bought an extension of one year). I already tried about 15 versions of the Realtek driver with no success. I am not sure but MAYBE the problem did not occur on the original windows vista that was shipped with this computer. However, I removed the original windows for good reasons (english). What do you suggest me? Did anyone fix this problem? Maybe by writing a script which resets the audio streams every 5 minutes? Shall I take the effort to deal with the acer support until they give me another model? (I won't have a computer than for a longer time, will spend money on telephone hotlines (1,30 EUR / min)......) Here are additional infos, if they are any help: Windows 7 64 Bit (Original was Windows Vista Home Premium 32 Bit) All specs Audio driver version:

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  • How do I activate the F_LINE input in a transplanted HP chassis?

    - by admin
    I have an HP Pavilion Media Center PC chassis, vintage 2003 or so and I replaced the motherboard in it with a newer (vintage 2009) HP motherboard, M2N68-LA (Narra 5). I have scoured the internet trying to find pinouts for the motherboard to no avail. My question concerns the front panel audio, specifically Line In. The old chassis was built for AC97 but the new mobo is build for the newer HD audio standard. I figured out by comparison & experimentally how to connect the Mic & Headphone jacks to the HD audio header of the mobo by adding a manual switch to set the SENSE lines. Now all works fine for Mic & headphone. The old chassis also has a front panel Line In jack that the newer HP chassis does not have. However, the new mobo has a 4 pin white connector labeled F_LINE that I believe is a line input. Under Windows 7 I see the two Line Inputs in the mixer but I can't get one of them to become active. The 4 pin F_LINE connector uses the two middle pins for ground, and presumably the other two for left and right audio inputs. There are no pins for sensing on that connector. Can anyone tell me how to use that F_LINE input for the front panel, or how to activate it?

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  • Sound doesn't work anymore after replacing RAM

    - by thejh
    Hello, today, I replaced one old RAM module with two newer, bigger ones, but now, the sound doesn't seem to work anymore. Already ran alsaconf and it didn't help. Output of lspci for the audio device: 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f5100000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel The audio device is onboard and has six configurable outputs, two or so are also capable of being an input (if I remember it correctly), but I don't know how to control it under linux. Does somebody know how/whether replacing the RAM could be related to my problem and/or how to fix it?

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  • How to boost playback volume in real time on media recorded with a very low volume.

    - by L Marksman
    I have never heard a satisfactory answer to this often misunderstood question, let me explain. Lets say I have a sound card and earphones/speakers that can play back audio loud enough in most cases. This is great but the problem is that you always find people who do not know how to record audio, from Youtube video's to music. So now you end up with a audio playback that only uses 10% or less of the capacity of your sound hardware, in vista/win 7 you will see this frequently in the mixer with the volume pushed up to max but the green sound level only goes up a millimeter or two. I am looking for (preferably free) software or a method to boost the sound level of any audio from any source in real time to use more of my hardware capacity similar to what VLC media player can do. Oh and please, do not tell me it is impossible. I am not trying to boost the volume past what my hardware is capable of, I am just trying to use my hardware's full capacity. Also please do not tell met to buy new hardware, I know I can use hardware amplification, I don't want to (like many others) spend money on a simple little problem like this. Thanks!

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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  • Playing NSF music in FMOD.net

    - by Tesserex
    So, as the title says, I want to be able to play NSF files using FMOD, because my project already uses FMOD and I'd rather not replace it. This will involve figuring out how existing players and emulators work and porting it. I haven't yet found an existing player that uses FMOD. My starting point is the MyNes source from http://sourceforge.net/projects/mynes/. There are two big steps between here and what I'm looking for. MyNes plays from a ROM, not NSF. So, I have to rip out the APU and get it to play NSF files. The MyNes APU uses SlimDX, so I have to convert that to FMOD.NET. I am really stuck about how to go about either of these, because I'm not that familiar with audio formats and it's hard finding resources online. So here are a few questions: From what I can tell from the NSF spec at http://kevtris.org/nes/nsfspec.txt, it's just contains the relevant memory section of the ROM, plus the header. If anyone can verify or correct this that would be great. The emulator APU uses data from the rest of the emulator to play, including things like cycle counts. I'm not sure what replaces this in a standalone player. Can't I just load all the music data at once into a stream and play it? Joining #1 and #2, does the header data from the NSF substitute for some of the ROM data in the emulator code? Using FMOD, will I be following the usercreatedsound example for loading a stream? And does this format count as PCM? Specifically MyNes says PCM8. Any tips on loading / playing the stream in FMOD are appreciated. As an aside, I don't really understand the loading / playing sections of the spec I linked at all. It seems to apply to 6502 systems / emulators only and not to my situation. I know it's a long shot for anyone here to have enough experience in this area to help, but anything you can provide is definitely appreciated. A link to an existing .NET library that does this would be even better, but I don't believe one exists.

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