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• "HDA audio bus driver is required and not found" on Dell Optiplex

  - by user1666698

  I have Dell Optiplex 745 with Windows 7 installed on it. I'm trying to use the Windows XP audio driver as Windows 7 drivers aren't available for Optiplex 745 and Windows Vista driver is displaying that it's not compatible with my hardware. When I try to install the Windows XP audio driver, it's displaying an error HDA audio bus driver is required and not found The installation fails then. I have researched thourghly and used many drivers but my audio is not working at all. I was also told that it might be a problem with my hardware – that is, a problem with the board.

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• Playing/extracting audio file from PDF

  - by ravl1084

  I use Ubuntu and I have a PDF file that contains an audio annotation. It won't play on Okular, it treats it as a text annotation. Following an old blog post where the poster created a small C script to extract the audio didn't work either, I suspect the format of these audio annotations has changed. Using the information on it I managed to uncompress the PDF and with vim, I found the audio data in the file. I tried copying this into its own file and changed the extension from mp3, wav, mid, but none of them would play. Is there a way of achieving this?

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• HTTP resource bundling/streaming practice

  - by icelava

  Our SPA (plain HTML and Javascript) makes use of huge volume of javascript and other resources that are downloaded via XHR. Given the sheer number of components and browser simultaneous request limits, we're thinking for ways to deliver our resources in a more efficient manner. A method we're considering is bundling several resources that logically form a coherent group into a single file; thus reducing down to only one XHR (per group). Furthermore to make it more responsive, we'd like to constantly inspect the partial responseText during the LOADING state, determining if a usable chunk (atomic resource) has already been downloaded, and make it available for deserialization/processing even before the XHR is DONE. (a stream-like experience) We're thinking surely somebody else would've considered roughly the same approach before, but haven't really come across any library/framework or container file format that is suitable for our scenario. Anybody else know of something similar?

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• Missing error handling in Streaming-AJAX-Proxy Log

  - by Michael Freidgeim

  We are using AjaxProxy(FROM http://www.codeproject.com/KB/ajax/ajaxproxy.aspx) on our web site, but started to notice errors accessing log.txt file. I found that the file is created by Log class and doesn't have ability to switch it off and error handling. I've added reading file name from configuration and try/catch block   public static class Log     {         private static StreamWriter logStream;         private static object lockObject = new object ();     public static void WriteLine(string msg)         {                       string logFileName = ConfigurationExtensions.GetAppSetting("AjaxStreamingProxy.LogFile" ,"");                       if (logFileName.IsNullOrEmpty())                             return;                       try                      {                             if (logStream == null )                            {                                    lock (lockObject)                                   {                                           if (logStream == null )                                          {                            logStream = File.AppendText(Path .Combine(AppDomain.CurrentDomain.BaseDirectory, logFileName));                                          }                                   }                            }                            logStream.WriteLine(msg);                      }                       catch (Exception exc)                      {                             string ignoredMsg = String .Format("The error occured while logging {0}, but processing will continue.\n {1} ", exc);                             LoggerHelper.LogEvent(ignoredMsg, MyCategorySource, TraceEventType .Warning, true);                      }         }

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• ffdshow h.264 audio desync

  - by Core Xii

  When I encode video with ffdshow with h.264, the audio is out of sync. At the very beginning of the video, the picture freezes for about 1 second, while the audio plays fine, resulting in the audio being that 1 second ahead of the picture throughout the entire video. Any ideas on possible causes or, obviously, solutions?

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• Why won't AVI2DVD load the audio stream?

  - by Xavierjazz

  XP SP3 I have an .avi file. It is in a folder on my "C:" drive. There are no disallowed characters in either the folder or file name. It has audio as I have watched it on my computer. I want to burn it to a DVD. When I load the file into AVI2DVD, no audio stream shows, and the program will not work without an audio stream. I have used the net extensively to try and solve this, with no success. AFICT I have followed all instructions exactly, but no audio stream. Very frustrating. Does anyone have a clue? Can you help me? Thank you. Regards,

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• Get binary data from audio impulses

  - by Timo

  I have IR sensor which have TRS plug and I can record my remotes signals into audio. Now I want to control my computer with TV remote, but I don't have any clue how to compare audio input with pre-recorded audio. But after I realized that these audio waves contains only some kind data (binary) I can turn these into binary or hex, so it is much easier to compare. Waves look just like this: http://i.imgur.com/lCIyl.png And this: ttp://i.imgur.com/goJ6d.png These are records of "OK" button, sometimes there are some impulses on right channel too and I don't know why, it seems like connections in sensor are damaged maybe. Ok thats not matter, anyway I need help with python program which read these impulses and turn these into binary, in realtime from audio input(mic). I know it's sounds like "Do it for me, while I enjoy my life", but I don't have experiences with sound transforming/reading... I've looking for python examples for recording and reading audio, but unsuccessfully.

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• Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

  - by davr

  First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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• ffmpeg: video file played OK on Ubuntu, but no sound on XP

  - by Andy Le

  I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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• What is the best way to get an audio file duration in Android?

  - by Gilead

  Hi! I'm using a SoundPool [ 1 ] to play audio clips in my app. All is fine but I need to know when the clip playback has finished. At the moment I track it in my app by obtaining the duration of each clip using a MediaPlayer [ 2 ] instance. That works fine but it looks wasteful to load each file twice, just to get the duration. I could roughly calculate the duration myself knowing the length of the file (available from the AssetFileDescriptor [ 3 ]) but I'd still need to know the sample rate and the number of channels. I see two potential solutions to that problem: Figuring out when a clip has finished playing (doesn't seem to be possible with SoundClip). Having a class which could load just the header of an audio file and give me the sample rate/number of channels (and, ideally, the sample count to get the exact duration). Any suggestions? Thanks, Max The code I'm using at the moment (works fine but is rather heavy for the purpose): String[] fileNames = ... MediaPlayer mp = new MediaPlayer(); for (String fileName : fileNames) { AssetFileDescriptor d = context.getAssets().openFd(fileName); mp.reset(); mp.setDataSource(d.getFileDescriptor(), d.getStartOffset(), d.getLength()); mp.prepare(); int duration = mp.getDuration(); // ... } On a side note, this question has already been asked [ 4 ] but got no answers.

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• Streaming Media Player Tutorial Stops Unexpectedly. How Should I Debug (And Other Questions From A

  - by Danny

  Hello! I'm a fledgling in the Android and Eclipse ways. I was reading and downloaded the source code for a Streaming Media Player tutorial from Pocket Journey, here. However, when I try to run it through the "Run As..." or "Debug As..." features in Eclipse, the emulator reports: "Sorry! The application Android Tutorials (process com.pocketjourney.tutorials) has stopped unexpectedly. Please try again." Now, from stepping through the Tutorial 1 activity (for those of you that may be familiar with it), it seems something funky happens here: button.setOnClickListener(new View.OnClickListener() { public void onClick(View view) { Toast.makeText(Tutorial1.this, "Button Clicked",Toast.LENGTH_SHORT).show(); }}); My questions are: What's up with that line? Apparently, other comments haven't had this problem; am I missing some library that I should be downloading from someplace? How should I go about debugging this other than/in addition to stepping through code? I'm used to Visual Studio, which has a nice box with exception details for me to sift through and copy/paste to google with. Eclipse debugging showed a "No Source Code" page, or a list of exceptions. I've heard of something called LogCat; is that relevant here? Thanks in advance!

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• Issues regarding playing audio files in a JME midlet.

  - by Northernen

  I am making a midlet which is to be used to play out local audio files. It is obviously not working. I am getting a null reference on the "is" variable, in the code snippet shown below. 1. try{ 2. System.out.println("path: " + this.getClass()); 3. InputStream is = this.getClass().getResourceAsStream("res/01Track.wav"); 4. p1=Manager.createPlayer(is, "audio"); 5. p1.realize(); 6. p1.prefetch(); 7. p1.start(); 8. } 9. catch(Exception e){ 10. System.out.println(e.getMessage()); 11. } I assume there is something wrong with the "this.getClass().getResourceAsStream("res/01Track.wav")" bit, but I can not for the life of me figure out why, and I have tried referring to the file in 20 different ways. If I printline "this.getClass()" it gives me "path: class Mp3spiller". The absolute path to "01Track.wav" is "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller\res\01Track.wav". Am I completely wrong in thinking that I should refer relatively to "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller"? If anyone could point out what I am doing wrong, I would be grateful. I have basically stolen the code from a tutorial I found online, so I would have thought it would be working.

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• No audio flash movies

  - by Valerio Giuffrida

  I've installed natty a few days ago and I'm experience some issues with the audio. Basically, I'm not able to listen anything coming from flash movies (such as youtube). This happen with both chromium and firefox (I use only the first one) and the errore I get in the stdout is: ALSA lib pcm.c:2109:(snd_pcm_open_conf) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_pulse.so I don't know how to fix it. I found somewhere to install native x64 flash plugins, but somebody says it is not advisable. I didn't get this kind of issues with 10.10 I've gotten before of natty. Hence, what am I supposed to do? thank you very much

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• How to cut audio file with avconv?

  - by x-yuri

  I have a hard time trying to figure out how to cut a file with avconv. Here's the command I use: avconv -ss 52:13:49 -t 01:13:52 -i RR119Accessibility.wav RR119Accessibility-2.wav But it doesn't work. I get the whole file as a result. Well, almost the whole file. Somehow the resulting file has duration 1:16:31 instead of 1:17:23. Also I believe I executed this command in every possible way: with -ss and -t after -i, with -t specifying ending point, with mp3 files, with specifying audio codec, with ffmpeg. Am I doing it wrong?

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• Audio Panning using RtAudio

  - by user1801724

  I use the RtAudio library. I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use RtAudio in duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. I have searched on the web, but I did not find anything useful. Should I apply a filter on the output buffer? What kind of filter?

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• Installing old Loki games on 12.04 64-bit results in no audio

  - by FlabbergastedPickle

  All, Here's an interesting problem. I followed instructions provided online for installing Loki Games' Heroes of Might and Magic 3 (see http://www.swanson.ukfsn.org/loki/ and http://wtanaka.com/node/7641) and got it installed and patched to the latest version. However, every time I start it regardless whether the pulseaudio is running, I get the following error: LD_LIBRARY_PATH=/usr/local/lib/Loki_Compat/ /usr/local/lib/Loki_Compat/ld-linux.so.2 /usr/local/games/Heroes3/heroes3.dynamic ALSA lib conf.c:3314:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM default Couldn't open audio: My first soundcard is HDMI output and my second one is the actual soundcard (HP DM1 running 12.04 64-bit with latest updates). I did set up /etc/asound.conf as follows: asound.conf pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } So, the default soundcard should work ok. Between Shadowgrounds that also stopped working and this it appears a there may be some unfinished business/regressions in 32-bit support on 64-bit systems in 12.04. Any thoughts?

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• audio controls in xfce 4.8

  - by Peter

  I am seeing several questions similar to mine, but none of the answers are sufficient. I am pretty green with Ubuntu, so here goes: I was just automatically upgraded to xfce 4.8 for Ubuntu studio. The volume control no longer works in my panel. When I launch 'mixer' I don't see any settings, either. When I try to run "linux audio configuration" I get an error: JACK can only be configured with a loaded and stopped studio. Please create a new studio or load and stop an existing one. I understand that I can change the volume using command line, but I can't understand why I got upgraded to something that fails on basic features. I much less likely to recommend ubuntu to others as a result. thanks!

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• Audio playback: part of song is skipped

  - by Homulvas

  I am experiencing some problems with music playback after upgrading to Ubuntu 12.10. Basically some of the songs stop playing after some time as if the song has ended. It's always the same songs and the same time. The weird thing that it happens with Clementine and Totem but VLC doesn't have this problem and it also plays as it should on Windows. I'm guessing there might be a problem with some library that's shared with by the first two applications. I don't know if it's relevant but the file format of the audio files is flac(don't know if the problem affects mp3, because I don't have many of them).

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• video/audio output via HDMI Ubuntu 12.04

  - by lostNfound

  I've been out of the Ubuntu loop for quite a while now and have a completely new laptop now. Just installed Ubuntu 12.04 64-bit and would like to output my video and my audio via HDMI to my television. the following is the lspci | grep VGA for my computer. please tell me if there is any additional information needed and preferably how to obtain it and i will be more than happy to oblige. thank you in advance for your time and assistance in this matter. 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 01:00.0 VGA compatible controller: NVIDIA Corporation GF108 [GeForce GT 540M] (rev a1) Edit: every time i restart my computer, after a short moment, i get an error message stating something along the lines "sorry, jockey needed to close unexpectedly." after researching, i discovered jockey is the name of the "additional drivers," which after initial installation, ubuntu informed me of proprietary drivers available. those are no longer available, and this error continues to occur.

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• Noisy audio recording in Ubuntu Linux

  - by Haresh K Miriyala

  I'm unable to record audio without noise on my Ubuntu laptop . It is not a grounding problem . My windows os is able to record voice without the static noise so, my hardware is fine . I read in the Windows forums that the os has software noise filtering mechanism that's automatically enabled. How do I do the same in Ubuntu? If not, please let me know how to filter the static noise. Any help is appreciated. Thanks!

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• How to get audio spectrum analysis?

  - by Mrwolfy

  I need to find or create a tool that analyzes the audio spectrum of a sound file (like a .wav or .mp3). I need to output the "volume" or power of x number of frequency bands and output the data as text. This will be used to produce a visualization, a graphic equalizer like you'd see on a stereo. I am currently looking at python to do it. My question is are there some tools out there that would do this (signal processing), like math works or others? I don't have any experience with them so any advice would be appreciated.

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• Radeon HD5570 HDMI Video Card 5.1 Audio doesn't work

  - by ryandlf

  I am using Ubuntu and XMBC on my HTPC and have chosen the Radeon HD5570 Video card which has an HDMI output. In the sound preferences there is no surround sound option for the video card just stereo and although I can get sound through it in XBMC, my receiver does not state Dolby Digital on movies that are in fact Dolby so its definitely not giving me the true sound it should. Does this card not support surround sound through HDMI and I somehow missed it? If that is the case does anyone have suggestion that has been tested and works? Id like to know its going to work before investing in yet another video card. UPDATE I purchased a Nvidia GeForce GTS 450, plugged it in, downloaded the proprietary driver from the system control panel, disabled the onboard audio from the BIOS (not sure if this was necessary but I did it anyways), and changed the sound settings to use the new video card. Everything works flawlessly. It was a seemless setup.

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• Podcast site - Serve audio files with CDN

  - by Bobe

  I am managing a small podcast website hosted on a shared server. Currently there are only eight or nine episodes, each of which are about 50 MB, so bandwidth is not really an issue at the moment. However, looking forward, would it be feasible to use a "free" CDN like Cloudflare to serve the audio files? If so, how would I set this up? I took a quick look at it before, and it seems you have to have your whole site routed (is that the right term?) through the CDN rather than just specific files or filetypes. I'd like some clarification on this.

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• sometimes, strange audio distortion(peeping, scratching), ubuntu 12.04

  - by richi902

  i have a problem with my sound in ubuntu 12.04. the problem is, that sometimes out of nowhere, when switching songs, or playing youtube videos, changing volume with my keyboard buttons, that the sound gets distorted(peeping, scratching). i dont know if it is related, but when i skip through music in rythmbox, there is also a little scratching noise. i can sometimes temporarly fix it: for youtube videos, i refresh the page, and sometimes it works agian normal, mostly not. for audio playback with rythmbox, i have to pause the song for sometime, and resume it, and hope that it works. before all that,i have changed my soundcard to "Analog Surround 5.1" in the sound settings from ubuntu, but i also used alsamixer to change it from 2 channels to 6 channels, since changing in ubuntu sound settings alone wasnt enough to make the other speaks work. i use a ASUS P8-H61-M LE B3 Revision Motherboard. which has a built in surround soundcard.

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• ffmpeg hangs when creating a video

  - by FearUs

  I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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