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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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  • Oracle User Productivity Kit Translation

    - by ultan o'broin
    Oracle's customers just love the User Productivity Kit (UPK). I hear only great things about it from our international customers at the Oracle Usability Advisory Board meetings too. The UPK is the perfect solution for enterprise applications training needs (I previously reviewed a fine book about UPK btw). One question I am often asked is how source content created using the UPK can be translated into another language. I spoke with Peter Maravelias, Principal Product Strategy Manager for UPK about this recently. UPK is already optimized for easy source-target translation already. There is even a solution for re-recording demos. Here's what you can do to get your source content into another language: Use UPK's ability to automatically translate events and actions. UPK comes with XML templates that allow you to accomplish this in 21 languages with a simple publishing action switch. These templates even deal with the tricky business of using gender-based translations. Spanish localization template sample Japanese localization template sample Use the Import and Export localization features to export additional custom content in a format like XLIFF, easily handled by translation tools. You could also export and import in Word format. Re-record the sound (audio) files that go with the recordings, one per screen. UPK's granular approach to the sound files means that timing isn't an option. Retiming demos isn't required. A tip here with sound files and XLFF-exported custom content is to facilitate translation context by avoiding explicit references to actions going on in the screen recordings. A text based storyboard with screenshots accompanying the sound files should also be provided to the translators. Provide a glossary of terms too. Use the re-record option in UPK to record any demo from a translated application. This will allow all the translated UI labels to be automatically captured. You may be required to resize any action events here due to text expansion issues. Of course, you will need translated data in the translated application too, so plan for this in advance. However, source-target language skills aren't required for the re-recording. The UPK Player itself, of course, is also available from Oracle along with content and doc in 21 languages. The Developer and Setup is also translated in a smaller number of languages. Check the Oracle UPK website for latest details. UPK is a super solution for global enterprise applications training deployments allowing source content to be translated into multiple languages easily. See this post on the UPK blog for more insight too!

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  • Beat detection and FFT

    - by Quincy
    So I am working on a platformer game which includes music with beat detection. I am currently using a simple if the energy that is stored in the history buffer is smaller then the current energy there is a beat. The problem with this is that ofcourse if you use songs like rock songs where you have a pretty steady amplitude this isn't going to work. So I looked further and found algorithms splitting the sound into multiple bands using FFT. I then found this : http://en.literateprograms.org/Cooley-Tukey_FFT_algorithm_(C) The only problem I'm having is that I am quite new to audio and I have no idea how to use that to split the signal up into multiple signals. So my question is : How do you use a FFT to split a signal into multiple bands ? Also for the guys interested, this is my algorithm in c# : // C = threshold, N = size of history buffer / 1024 public void PlaceBeatMarkers(float C, int N) { List<float> instantEnergyList = new List<float>(); short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; // Calculate instant energy for every 1024 samples. while (sampleIndex + nextSamples < samples.Length) { float instantEnergy = 0; for (int i = 0; i < nextSamples; i++) { instantEnergy += Math.Abs((float)samples[sampleIndex + i]); } instantEnergy /= nextSamples; instantEnergyList.Add(instantEnergy); if(sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; } int index = N; int numInBuffer = index; float historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } }

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  • Android: Voice Recording and saving audio

    - by user1320912
    I am working on application that will record the voice of the user and save the file on the SD card and then allow the user to listen to the audio again. I am able to allow the user to record his voice using the RecognizerIntent, but I cant figure out how to save the audio file and allow the user to hear the audio. I would appreciate it if someone could help me out. I have displayed my code below: // Setting up the onClickListener for Audio Button attachVoice = (Button) findViewById(R.id.AttachVoice_questionandanswer); attachVoice.setOnClickListener(new OnClickListener() { public void onClick(View v) { Intent voiceIntent = new Intent(RecognizerIntent.ACTION_RECOGNIZE_SPEECH); voiceIntent.putExtra(RecognizerIntent.EXTRA_LANGUAGE_MODEL, RecognizerIntent.LANGUAGE_MODEL_FREE_FORM); voiceIntent.putExtra(RecognizerIntent.EXTRA_PROMPT, "Please Speak"); startActivityForResult(voiceIntent, VOICE_REQUEST); } }); protected void onActivityResult(int requestCode, int resultCode, Intent data) { if(requestCode == VOICE_REQUEST && resultCode == RESULT_OK){ }

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  • Testing background audio in the simulator

    - by Cactuar
    I'm experimenting with the new background audio service in iPhone OS 4.0 but I can't get it to work in the simulator. According to this page: iPhone Application Programming Guide: Executing Code in the Background it seems that all I have to do is add the a UIBackgroundModes key with an array containing audio to my Info.plist file and the audio my application plays should automatically continue when I switch to another app. I have done this but the audio still pauses as I switch to another app, when I switch back it continues where it left off. This is the code I'm using to play the sound: NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/audio.mp3", [[NSBundle mainBundle] resourcePath]]]; NSError *error; audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; audioPlayer.numberOfLoops = -1; if (audioPlayer == nil) NSLog(@"%@", [error userInfo]); else [audioPlayer play]; Has anyone gotten this to work? Could it be that it would work on an actual device and it's just a problem with the simulator? I'm a bit hesitant to install 4.0 on my phone since I've heard it's still very buggy. Wish I had another device to use only for development.

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  • Custom flash mp3 player stopping in the middle of playing audio on windows nt ie6 system

    - by Charlotte Moller
    We have used a custom MP3 flash player for a lot of years on our website without any issues, but recently, a client of ours is reporting that the audio is playing for several seconds and then stopping. When they refresh the page or click play in the player again the audio plays fine. We are puzzled as to what could be causing this issue after this running successfully for our clients for so many years. The client system is Windows NT running IE6. Does anyone have any idea what could cause the audio to behave this way? Could audio drivers or the version of flash cause problems? We do not have flash programmers on our team so we are not even sure where to start looking within the flash code of the player. Any ideas?

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  • Capture Flash Audio in 4.7 Edge?

    - by emcmanus
    Is there a way to capture plugin (Flash) audio before it gets to the sound card? I'd like to record plugin audio, hopefully without actually playing the sound. Capturing audio at the device level is an absolute last resort, as the application would pick up all system audio rather than just the Webkit plugin. I'm aware of the recent switch back from QTMultimedia; is this possible with phonon? I spent the night looking for some way to access the phonon graph via QWebFrame (or any of the QtWebkit widgets) -- and didn't turn up much. I also started digging through QTWebkit, particularly NPAPI, without success. For reference, I'm using the edge version of 4.7 (6aa50af000f85cc4497749fcf0860c8ed244a60e) This seems to be a fairly challenging problem. Any hints would be greatly appreciated.

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  • Audio queue start failed

    - by mobapps99
    Hi , i'm developing a project which has both audio streaming and playing audio from file. For audio streaming i'm using AudioStreamer and for playing from file i'm using avaudioplayer. Both streaming and playing works perfectly as long as the app is not interrupted by a phone call or sms. But when a call/sms comes after dismissing the call when i try to restart streaming i'm getting the error "Audio queue start failed" . This happens only when i have used avaudioplayer at least once and after that used streaming. When the avaudioplayer obeject is not created , in this scenario the there is no problem with resuming streaming after dismissing the call. My guess is that some thing is wrong with audioqueue. Help is very much appreciated.......

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  • Making a DVD video with a still image and PCM 16bit audio with ffmpeg

    - by João
    I'm trying to make a small video with a still image and a sound file playing in the background to pass it to dvdauthor and create a DVD. The command I'm using is this: ffmpeg -loop_input -i image.jpg -qscale 2 -i song.flac -aspect 4:3 -target pal-dvd -acodec pcm_s16le -shortest output.mpg However, the resulting video file doesn't have sound at all (testing it on VLC Player). I don't know if I can't combine "-acodec pcm_s16le" with "-target pal-dvd" to override the later, or if there is something else wrong with the command. If I try without the "-acodec pcm_s16le" parameter the video and audio works, I can even create a DVD ISO with it. However, the audio stays as AC3. I wanted to include with the video the lossless audio, not a compressed one. I suppose the DVD standart allows to have PCM audio in it, am I right?

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  • How to get the default audio format of a TTS Engine

    - by Itslava
    In Microsoft TTS 5.1 or newer. The SpVoice.AudioOutputStream property says: The AudioOutputStream property gets and sets the current audio stream object used by the voice. Setting the voice's AudioOutputStream property may cause its audio output format to be automatically changed to match the text-to-speech (TTS) engine's preferred audio output format. If the voice's AllowAudioOutputFormatChangesOnNextSet property is True, the format change takes place; if False, the format remains unchanged. In order to set the AudioOutputStream property of a voice to a specific format, its AllowOutputFormatChangesOnNextSet should be False. It means a engine's always has a preferred audio output format. So, how can i get it.. i have not found any interface to get that attribute.

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  • Access MP3 audio data independently of ID3 tags?

    - by kyl191
    Hi, this is a 2 part question. First off, is it possible to access the audio data in an MP3 independently of the ID3 tags, and secondly, is there any way to do so using available libraries? I recently consolidated my music collection from 3 computers and ended up with songs which had changed ID3 tags, but the audio data itself was unmodified. Running a search for duplicate files failed because the file changed with the ID3 tag change, but I think it should be possible to identify duplicate files if I just run a deduplication using the audio data for comparison. I know that it's possible to seek to a particular position past the ID3 header in the file, and directly read the data, but was wondering if there's a library that would expose the audio data so I could just extract the data, run a checksum on it, and store the computed result somewhere, then look for identical checksums. (Also, I'd probably have to use some kind of library when you take into account variable length headers.)

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