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• Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

  - by Roy Rico

  I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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• Burn 24/96 flac files to play on standalone player

  - by takeshin

  I have vinyl record rip in 24/96 flac format. Each track is almost 200 MB big, so the album won't fit on CD. How to burn these files on a DVD to play with the same quality on standalone DVD player? My player supports SACD, DVD Audio and DVD video as well. My OS is Ubuntu Lucid (preferred), but I have also WinXp with Nero installed. BTW, is there any difference between DVD+ and DVD- for audio?

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• Replace sound in another YouTube video

  - by Tom

  I have received permission from someone to translate the audio in their movies. The problem I am facing is that the video quality is quite poor and the author does not have the original videos any more. How can I replace the audio in the YouTube videos without further degrading the quality of the videos? Thanks, Tom

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• Any plugins for Skype that support "Soundboard" usage?

  - by Axxmasterr

  I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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• Debian sound on hdmi instead of jack

  - by Hans de Jong

  I installed debian (gnome) and i can't get my sound working. When i use inxi -A i get the following result: Audio: Card-1: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series] driver: snd_hda_intel Card-2: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) driver: snd_hda_intel Sound: Advanced Linux Sound Architecture ver: 1.0.24 My feeling tells me my sound output is on the HDMI instead of my jackplug on my motherboard. How can i change this to my motherboard sound output?

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• Which connector do I need for a "line level" subwoofer?

  - by Ben Brocka

  I've got a separate pair of speakers and I'm looking at adding a subwoofer (this, specifically). I noticed on the detail page it's inputs are listed as such: Inputs: Speaker level, line level If I'm not mistaken "line level" are the standard 3.5 audio jacks on your motherboard/sound card, right? My motherboard has the standard 6 ports for sound, if I get a subwoofer like this can I simply plug the input into the orange 3.5 jack? My audio software supports up to 7.1 so software-wise, 2.1 wouldn't be a problem.

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• Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

  - by Roy Rico

  I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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• Changing default playback device on Windows 8

  - by emartel

  Previously, on Vista and Windows 7, changing the Default Playback device would occur instantly. For example, audio is coming out of my speakers, I right click the Volume Control, click Playback Devices then I select another device and click Set Default. Audio would be transferred immediately. Unfortunately, now, with Windows 8, I need to kill whatever process what outputting sound, and restart it for the change to take effect. Is there something that can be done about it so that changes are taken into account immediately?

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• gameKit or bounjour ???

  - by alia

  which one is better for implementing chat for the iPhone with users in the current area gameKit or bounjour ?

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• Help with chat server

  - by mithun1538

  I am designing a chat server in java. The communication is Http based and not socket based. In the client side I have an applet. In the server side I have a servlet. Applet: I create a new thread to listen for incoming messages(GET method). The main thread is used to send messages(POST messages). The partial code is : public void start() { System.out.println("Creating new thread"); Thread thread = new Thread(this); thread.start(); } private String getNewMessage() { System.out.println("Inside getNewMessage"); String msg = null; try { while(msg == null) { System.out.println("Trying to listen to servlet"); URL servlet = new URL(getCodeBase(), "NewServlet?mode=msg"); URLConnection con = servlet.openConnection(); con.setUseCaches(false); DataInputStream din = new DataInputStream(new BufferedInputStream(con.getInputStream())); msg = din.readUTF(); System.out.println("message read :" + msg); } } catch (Exception e) { e.printStackTrace(); } return msg + "\n"; } public void run() { System.out.println("Inside new thread"); while(true) { System.out.println("inside first while"); String newMsg = getNewMessage(); chatOutput.append(newMsg); System.out.println("Appended!!"); } } private void jButton1ActionPerformed(java.awt.event.ActionEvent evt) { String message = chatInput.getText(); chatInput.setText(""); chatOutput.append(message + "\n"); try { System.out.println("Trying to send msg :" + message); URL url = new URL(getCodeBase(), "NewServlet"); URLConnection servletConnection = url.openConnection(); servletConnection.setDoInput(true); servletConnection.setDoOutput(true); servletConnection.setUseCaches(false); servletConnection.setRequestProperty("Content-Type", "application/octet-stream"); ObjectOutputStream out = new ObjectOutputStream(servletConnection.getOutputStream()); out.writeObject(message); out.flush(); out.close(); System.out.println("Message sent!"); } catch (Exception e) { e.printStackTrace(); } } This next code is from the servlet side. it uses the Observable interface to identify and send messages to clients. public class NewServlet extends HttpServlet { // getNextMessage() returns the next new message. // It blocks until there is one. public String getNextMessage() { // Create a message sink to wait for a new message from the // message source. System.out.println("inside getNextMessage"); return new MessageSink().getNextMessage(source);} @Override protected void doGet(HttpServletRequest request, HttpServletResponse response) throws ServletException, IOException { System.out.println("Inside Doget"); response.setContentType("text/plain"); PrintWriter out = response.getWriter(); out.println(getNextMessage()); } // broadcastMessage() informs all currently listening clients that there // is a new message. Causes all calls to getNextMessage() to unblock. public void broadcastMessage(String message) { // Send the message to all the HTTP-connected clients by giving the // message to the message source source.sendMessage(message); } @Override protected void doPost(HttpServletRequest request, HttpServletResponse response) throws ServletException, IOException { System.out.println("Inside DoPost"); try { ObjectInputStream din= new ObjectInputStream(request.getInputStream()); String message = (String)din.readObject(); System.out.println("received msg"); if (message != null) broadcastMessage(message); System.out.println("Called broadcast"); // Set the status code to indicate there will be no response response.setStatus(response.SC_NO_CONTENT); } catch (Exception e) { e.printStackTrace(); } } /** * Returns a short description of the servlet. * @return a String containing servlet description */ @Override public String getServletInfo() { return "Short description"; } MessageSource source = new MessageSource();} class MessageSource extends Observable { public void sendMessage(String message) { System.out.println("inside sendMsg"); setChanged(); notifyObservers(message); } } class MessageSink implements Observer { String message = null; // set by update() and read by getNextMessage() // Called by the message source when it gets a new message synchronized public void update(Observable o, Object arg) { // Get the new message message = (String)arg; // Wake up our waiting thread notify(); } // Gets the next message sent out from the message source synchronized public String getNextMessage(MessageSource source) { // Tell source we want to be told about new messages source.addObserver(this); System.out.println("AddedObserver"); // Wait until our update() method receives a message while (message == null) { try { wait(); } catch (Exception ignored) { } } // Tell source to stop telling us about new messages source.deleteObserver(this); // Now return the message we received // But first set the message instance variable to null // so update() and getNextMessage() can be called again. String messageCopy = message; message = null; System.out.println("Returning msg"); return messageCopy; } } As you can see I have included System.out.println("Some message"); in some places. this was just for debugging purposes. In java console, i get the following output: Creating new thread Inside new thread. inside first while. Inside getNewMessage. Trying to listen to servlet. In the servlet side, i get the following output in the tomcat logs: Inside Doget. inside getNextMessage. AddedObserver. After i type a message in the applet, and send it, I get the foll output in java console: Trying to send msg :you deR?? Message sent! But in servlet side, I dont get anything in the logs. I used the O'Reily Java Servlet Programming as reference(The observer interface comes from there). But I am not getting any chat communication between two clients. As can be understood from the logs, the POST method is not called. Any reason for this?

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• Is there any free live support solution that would ease implementation of a basic online help-desk?

  - by bogdan

  I'm looking for a free or inexpensive solution to give live online support to web visitors. I would like something based on Jabber/XMPP because it's an open protocol. This should work like this: The company would use 2-3 jabber accounts like Google Talk for providing support Online chat form that would connect to one of currently online jabber accounts Optionally give one proxy jabber account to the customers. This proxy would contact any of the currently online company accounts. Using this customers will see only one chat contact but they will speak with different people based on who is online at the moment. Currently I found only some expensive solutions like J-livesupport, Akeni. I would really appreciate if you will recommend something you successfully implemented.

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• What is better to choose for public and private chats on the site: IRC or Jabber?

  - by Gora Mog

  I am looking for a solution to implement a public chat and a system of instant messaging within the site.

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• Python - react to custom keyboard interrupt

  - by flixic

  Hello. I am writing python chatbot that displays output through console. Every half second it asks server for updates, and responds to message. In the console I can see chat log. This is sufficient in most cases, however, sometimes I want to interrupt normal workflow and write custom chat answer myself. I would love to be able to press a button (or combination) that would switch to "custom reply mode". What is the best way to do that, or achieve similar result? Thanks a lot!

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• Can't connect to VPN on Ubuntu 12.04

  - by 12rad

  I'm having a lot of trouble connecting to VPN. This used to work on my machine, but i recently did an update and it's stopped working. I'm not sure what the problem is. My question is how do i debug this? I'm not able to narrow it down to a specific problem. This is what i get when i tail the syslogs. Would appreciate any help! Nov 6 23:42:52 meera NetworkManager[1137]: <info> Starting VPN service 'pptp'... Nov 6 23:42:52 meera NetworkManager[1137]: <info> VPN service 'pptp' started (org.freedesktop.NetworkManager.pptp), PID 6132 Nov 6 23:42:52 meera NetworkManager[1137]: <info> VPN service 'pptp' appeared; activating connections Nov 6 23:42:52 meera NetworkManager[1137]: <info> VPN plugin state changed: starting (3) Nov 6 23:42:52 meera NetworkManager[1137]: <info> VPN connection 'NAME VPN' (Connect) reply received. Nov 6 23:42:52 meera pppd[6136]: Plugin /usr/lib/pppd/2.4.5/nm-pptp-pppd-plugin.so loaded. Nov 6 23:42:52 meera pppd[6136]: pppd 2.4.5 started by root, uid 0 Nov 6 23:42:52 meera chat[6139]: timeout set to 15 seconds Nov 6 23:42:52 meera chat[6139]: abort on (NO CARRIER) Nov 6 23:42:52 meera chat[6139]: abort on (NO DIALTONE) Nov 6 23:42:52 meera chat[6139]: abort on (ERROR) Nov 6 23:42:52 meera chat[6139]: abort on (NO ANSWER) Nov 6 23:42:52 meera chat[6139]: abort on (BUSY) Nov 6 23:42:52 meera chat[6139]: abort on (Username/Password Incorrect) Nov 6 23:42:52 meera chat[6139]: send (AT^M) Nov 6 23:42:52 meera pptp[6138]: nm-pptp-service-6132 log[main:pptp.c:314]: The synchronous pptp option is NOT activated Nov 6 23:42:52 meera chat[6139]: expect (OK) Nov 6 23:42:52 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 1 'Start-Control-Connection-Request' Nov 6 23:42:53 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_disp:pptp_ctrl.c:739]: Received Start Control Connection Reply Nov 6 23:42:53 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_disp:pptp_ctrl.c:773]: Client connection established. Nov 6 23:42:53 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 7 'Outgoing-Call-Request' Nov 6 23:42:54 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_disp:pptp_ctrl.c:858]: Received Outgoing Call Reply. Nov 6 23:42:54 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_disp:pptp_ctrl.c:897]: Outgoing call established (call ID 0, peer's call ID 13077). Nov 6 23:42:54 meera pptp[6138]: nm-pptp-service-6132 warn[decaps_hdlc:pptp_gre.c:231]: The ppp mode is synchronous, yet no pptp --sync option is specified! Nov 6 23:43:07 meera chat[6139]: alarm Nov 6 23:43:07 meera chat[6139]: Failed Nov 6 23:43:07 meera pppd[6136]: Script chat -v -f /etc/ppp/chat-ztisp finished (pid 6139), status = 0x3 Nov 6 23:43:07 meera pppd[6136]: Connect script failed Nov 6 23:43:07 meera pppd[6136]: Waiting for 1 child processes... Nov 6 23:43:07 meera pppd[6136]: script /usr/sbin/pptp 204.197.218.90 --nolaunchpppd --loglevel 0 --logstring nm-pptp-service-6132, pid 6138 Nov 6 23:43:07 meera pptp[6138]: nm-pptp-service-6132 warn[decaps_hdlc:pptp_gre.c:204]: short read (-1): Input/output error Nov 6 23:43:07 meera pptp[6138]: nm-pptp-service-6132 warn[decaps_hdlc:pptp_gre.c:216]: pppd may have shutdown, see pppd log Nov 6 23:43:07 meera pptp[6143]: nm-pptp-service-6132 log[callmgr_main:pptp_callmgr.c:234]: Closing connection (unhandled) Nov 6 23:43:07 meera pppd[6136]: Script /usr/sbin/pptp 204.197.218.90 --nolaunchpppd --loglevel 0 --logstring nm-pptp-service-6132 finished (pid 6138), status = 0x0 Nov 6 23:43:07 meera pptp[6143]: nm-pptp-service-6132 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 12 'Call-Clear-Request' Nov 6 23:43:07 meera pptp[6143]: nm-pptp-service-6132 log[call_callback:pptp_callmgr.c:79]: Closing connection (call state) Nov 6 23:43:07 meera pppd[6136]: Exit. Nov 6 23:43:07 meera NetworkManager[1137]: <warn> VPN plugin failed: 1 Nov 6 23:43:07 meera NetworkManager[1137]: <info> VPN plugin state changed: stopped (6) Nov 6 23:43:07 meera NetworkManager[1137]: <info> VPN plugin state change reason: 0 Nov 6 23:43:07 meera NetworkManager[1137]: <warn> error disconnecting VPN: Could not process the request because no VPN connection was active.

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• Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

  - by jdl2003

  I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

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• Sending microphone input over Remote Desktop 7.0

  - by Taylor Price

  I am using Remote Desktop 7 (the new version that came out with Windows 7) to control a Windows XP Pro machine. I have selected "Record from this computer" in the Remote Audio settings. When I connect to the machine, go to the control panel, open the sound panel, and go to the audio tab, I find that the default sound playback device is "Microsoft RDP Audio Driver". However, there is no default sound recording device. As expected, my IP phone thinks there is no recording device. If I am sitting in front of the computer with a mic plugged in, it works just fine. Has anybody else been able to get this work appropriately? Is there anything that I have to setup on the XP machine to get this working? Thanks in advance. Edit: As John T pointed out below, you have to be connecting to a Windows 7 Enterprise or Ultimate machine for this to work. I've also found out that Multi-monitor support has the same requirement.

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• QuickTime Player sounds much better than iTunes

  - by Gene Goykhman

  I am playing a 320 kpbs encoded music MP3 in iTunes and the sound is substantially worse than the exact same file played back in QuickTime Player (Max OS X 10.8.5). I have maxed out system volume and iTunes playback volume. I have disabled all the audio processing features in iTunes (equalization, sound enhancer, etc.) The audio coming from iTunes still sounds resampled and/or processed, whereas QuickTime Player appears to be playing it "as is". Even when I Get Info on the MP3 file in Finder and play it back directly from the Get Info window it sounds good. It's just iTunes that seems to be mangling the song. I can notice a difference on virtually all my music, so it's not just one particular MP3. I suspect the issue is that iTunes is doing some kind of audio processing but I can't find a way to turn it off. This is the newest iTunes (11.1), but the problem has probably been going on for a while... I just switched to decent earbuds and started noticing the difference. What's the best way to force iTunes to play back the file as-is, or as close as possible to how QuickTime Player/Finder would play it?

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• Speakers silent, headphones work in Ubuntu 9.04

  - by CarlF

  I'm running Ubuntu 9.04. Worked fine for months, then I rebooted yesterday after weeks of continuous operation. Now audio won't play through the speakers. The USB headset works fine, but the Conexant audio (CX20549) does not. Weirdly, it thinks it's playing. pavumeter shows appropriate levels, volume looks OK in alsamixer, but no sound. I did find this page: http://www.eugeneteplitsky.com/fixing-silent-pulseaudio-in-ubuntu-9-04/ Unfortunately the advice there doesn't help me. For one thing, the syntax for the alsa-base.conf file is apparently not actually documented anywhere. For another, my chipset isn't listed in the kernel.org docs he links to! EDIT: would upgrading to 9.10 help? Is there a major change in the audio subsystem between 9.04 and 9.10? Any suggestions? EDIT 2: This is stranger than I thought. Sound works normally in Xine, but is silent in Audacity, VLC and mplayer. What the?

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• Problems with 5.1 digital out on Ubuntu 12.04

  - by user895319

  I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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• 5.1 Surround Channels are Jumbled

  - by stickynips

  I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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• How to get the mic on the Creative X-Mod soundcard working correctly?

  - by Nyamiou The Galeanthrope

  Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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• Renoise and dssi and jack

  - by Bojan

  This may be a little complicated, but, is jack a necessity? I mean, i use renoise, and, since i dont have the need for low latencies, do i really need to use it? My basic setup ( or workflow ) is that i use csound to render stuff to wav, then import it as a sample in renoise. That goes with field recordings, my own samples, etc. So, i dont need ultra low latencies, and i dont need to patch "cords", but i want to use dssi plugins, and dssi-vst. What would be something of a minimum requirements of apps that should work. Can renoise load dssi-vst plugins by itself or do i need to use jack to patch thru or something third, i tried to read lot of articles but i got lost in the different setups...

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• Beat detection, weird detection

  - by Quincy

  I made this soundanalyzer class to detect beats in songs : // put it on pastebin for the big size, will put it here if people rather want that. pastebin.com/8PdgZPP3 but for some reason its only detecting beats from 637 sec to around 641(sec) and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates and it seems as its assigning a beat to each instant energy value in between those values. Its modeled after this : http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats properly register ?

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