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  • Simplest way to add stills to MP3 audio soundtrack

    - by Ambrose
    I have some audio and want to add visuals to it and upload to YouTube. Nothing fancy, just like slides, image_A from 00:00:00 to 00:00:05 then image_B from 00:00:50 to 00:00:30 and so on. I have recent Mac and Windows machines to do this on. I'd like to do it on free or demo software if I can. Please give a bit of a hint how to get started if you can. I took a look a iMovie, but ... where to start, if you haven't actually got video?

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  • Multimedia PDF (Audio, Video and Links) That works on Desktop and iOS

    - by Keefer
    We've got a client that wants to have a PDF that has embedded audio, video and links. Using Acrobat Pro 9.x I've been able to embed all three no problem. They all work/playback if I use Acrobat Pro/Acrobat Reader. But don't show up in OS X's Preview at all. They also don't show up in iOS. Links work everywhere, but no multimedia. So I tried creating a similar document via Apple's iBooks Author, then exported as a PDF. Links work, but multimedia doesn't seem to work anywhere. Is there any way to make a PDF that works universally with embedded links and multimedia?

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  • erratic audio levels on windows vista

    - by old retired dude
    I'm somewhat hard of hearing. I've been listening to my windows Vista machine with a pair of headphones so I don't annoy the others. I have 2 issues: 1) the volume varies enormously depending on the source. Having a windows alert occur while I am listening to a DVD or Youtube is a painful experience. Is there a preferred way to set all the different audio controls so I have a more constant volume? I already have lowered the volumes of the windows alerts. 2)Is there a way to limit the volume of my headphones to protect what is left of my hearing? Is there a software solution or should I be going for a hardware limiter? thanks retired dude

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  • Streaming audio to headless linux box

    - by Ralph
    I have a dual boot (Win 7 + Ubuntu) PC connected via wifi with my music collection on a local HDD. I usually use Rhythmbox on Ubuntu or Winamp on Windows to listen to my music but I'll change if I have to. I also have a Raspberry Pi (low power PC running Debian) in the living room that is usually headless and connected via ethernet. The Raspberry Pi is also connected to my living room speakers via an amp. I would like to be able to stream music from my PC over the network to the linux raspberry pi. What software can I use to do this? Some sort of audio client\server?

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  • Cannot set audio input volume (internal microphone) on mac

    - by JohnIdol
    On a macbook air (MacOS X 10.6.5), when doing skype calls people are complaining they hear me very low - so I had a look to the system preferences under audio and noticed the input volume was 54%. I am now trying to set the input volume to 100%. To my surprise the volume is gradually set back as I speak. I tried deselecting 'use ambient noise reduction' but it doesn't help.' Is there any way to avoid this volume auto-setting feature? Any help appreciated!

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  • Real time audio streaming

    - by Josh K
    I have a remote computer running OS X. I would like to stream the audio from the microphone input over the network so I can listen to it. Primarily I want to do this because I'm out of the office but still need to communicate with people there. I would like to use VLC, but am not fully aware of the options available. I tried SoundFly (as recommended by another answer) but this didn't seem to want to connect. At this point I should note that I'm using a VPN network to connect to the remote computer (using Hamachi). I can open up ports / etc fine though, so I should be able to do this. Alright, I found Nicecase which does exactly what I want but I would prefer to not have to shell out $40 for it.

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  • Reducing volume of an audio device on windows 7

    - by bdonlan
    I have a USB headset with a very loud amplifier, but low granularity in its gain control. In order to get comfortable audio, I have to reduce the individual application levels in the mixer to '1', and the master mixer to around '10'. Of course, new applications start out at '10', and immediately blast out my ears. Is there a way to add a filter to cut down the volume some so I can get better control of it? That is, reduce the volume of '100' so I can work within a reasonable range.

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  • Are there any 5.1 surround audio switches on the market?

    - by thepurplepixel
    (Somewhat related to this question) I have a set of Logitech 5.1 surround speakers, which use 3 stereo 3.5mm TRS connectors (minijacks) to transfer the audio (the typical green/black/orange audio outputs). I have a Griffin Firewave hooked up to my MacBook Pro, and my desktop has a Realtek ALC889 audio chipset. I have looked for a way to, essentially, switch the speaker inputs between my Firewave and my desktop without having to disconnect the cables from one, route them around my desk, and plug them into the other. I'd love to have something like an old Belkin DB-25/LPT switch, but for these audio cables. Of course, purchasing one and soldering my own cables on the connection terminals is always an option, but, is there a reasonably priced 5.1 audio switch (or 3x stereo) on the market that will accomplish the simple task of switching audio outputs between two computers into a set of 5.1 speakers? Thanks in advance!

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  • Switched from DVI to HDMI, possible audio artifacts?

    - by I take Drukqs
    I'm using an ASUS VH236H monitor and an EVGA GeForce 570 GTX both of which are brand new. My monitor has an audio out port for speakers/headphones so I plugged in my headphones and made a random selection from my library when I noticed two things: There are static-like artifacts during "louder" parts of songs. There's what seems to be a volume cap in place. When I crank the volume past 100% in VLC the decibel level does not truly increase but the amount of static does. The cable is not new; I yanked it off of my PS3 when my DVI cable broke. It has been used a good amount on my HDTV and PS3 so I doubt it's a matter of burn-in. I like the way the setup works with an HDMI cable as opposed to DVI because my headphones barely reach my rig whereas I have plenty of slack when they're plugged into my monitor. Thanks in advance for any support. Note: I'm using a high quality HDMI cable from monoprice, AKG K702 headphones, and VLC media player.

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  • Audio server with best API?

    - by Wintermute
    I'm a web dev, working in a small studio with a couple of other devs and some crayon-munchers (or, "designers"). Like all the best and trendiest creative studios, we have tunes. Our tunes consists of a set of speakers that whoever wants to can plug into their machine, and DJ their little socks off via iTunes, Spotify, VLC or whatever their music player of choice happens to be. Obviously, this lacks finesse. What we WANT is this: a single, dedicated machine running some sort of audio player (ideally Win-based, but a Linux flavour isn't impossible), that exposes an API. We (ie: me and the other devs) want to write a web-based client onto it, that'll let us remotely do all sorts of funky stuff like generating on-the-fly genre-based playlists, and voting for tracks, and making tea. My question - and please forgive me if this isn't the place for such a question, I was going to ask on Stackoverflow but that didn't seem right either - is this: what's the best player to start with? What can do all of this? I know VLC can function as a streaming server, but know nothing of any API it may have. I'd rather chop my pinky off than use iTunes, but if it does what we want, then... Anyhow, thanks for reading. All comments and suggestions gratefully received.

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  • Optical SPDIF audio from motherboard not working with receiver

    - by simon b
    Hi, I hope someone can help; I can't get my SPDIF optical out working through my receiver and all the responses I can see on the web assume you have a sound card, while I settled for the (seemingly high end) sound on my motherboard (Asus P7P55D-E PRO), which appears to limit some of my options. My set-up is a "new out of the box" one and is: *Windows 7 PC (using PowerDVD10 for DVDs/Blurays and Windows media player for music) *Asus P7P55D-E PRO motherboard - has 8-channel audio TRS jacks and SPDIF optical and coaxial out *An old Yamaha receiver, whose only multi-channel input options are optical in and 6 channel RCA in. However, it still can handle DTS and DD *Boston Acoustic Soundware XS 5.1 speakers I've currently got the SPDIF optical out from the motherboard connected to the in on my receiver, have SPDIF enabled in the sound menu and the light is glowing red down the fibre. But I'm getting no sound at all. What I want is to be able to play DVDs/BluRays in 5.1 but also to be able to play music in multi-channel mode (even though I know this will be "fake" multichannel; it's more about where I sit in the room and my requirement to use the sub because the Boston is a satellite/sub set-up) My questions are: *Will optical work at all for multi-channel? THe latest posts I can see suggest it does but some people seem to say optical only outputs stereo. Whom to believe? *Even if it does work, I've read that I have to disable AC-3 decoding, or make various other changes, which don't seem to be possible without the menu options that a sound-card brings. Is the motherboard-only option just too inflexible? *Although my SPDIF device is enabled in the sound menu, it insists under "Jack information" that it is a "rear panel RCA jack", when of course it is not (both TOSLINK and rCA jacks do exist). Has the PC just forgotten that it has an optical? *I think I could relatively easily connect the 8-channel 3.5mm TRS jacks to my receiver 6-ch input jacks by way of TRS/RCA cables, but would that not stop me from being able to play music from media-player in multi-channel mode, as I'm not sure the motherboard can cope *Or do I need to bite the bullet and buy a sound-card? And if so, how can I be sure the one I get doesn't have the same problem? Any thoughts gratefully received, Cheers, simon

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  • Audio 2 dj soundcard configuration

    - by Jaroslav
    I have an http://www.native-instruments.com/#/en/products/dj/audio-2-dj/ The problem in settings it only sees one outpout, when there should be two(I need that for mixxx etc.) Also I want to be able set the sample rate to one of these 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

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  • How to get mixer applet for "Built-in Audio Analog Stereo"

    - by gerrit
    In pavucontrol, I can choose between RV620 HDMI Audio [Radeon HD 3400 Series] and Built-in Audio. When the former is enabled, videos on (among others) Youtube play way too fast, but this answer solved my problem (though I don't know why). However, when I use Built-in Audio instead of RV620 HDMI Audio [Radeon HD 3400 Series], the mixer in my applet appears to be disabled; the icon is replaced by a blank and changing the volume has no effect, as the applet apparently only relays to RV620 HDMI Audio [Radeon HD 3400 Series]. How do I get an applet to control the volume for Built-in Audio instead?

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  • choppy streaming audio

    - by user88503
    I could use some help troubleshooting choppy streaming audio. The problem is jerky playback regardless of audio or video with audio. Both Chromium and Firefox have the problem, however files played directly on the machine with Rhythmbox sound just fine. I'm running 12.04 LTS on a C2D T9300. Most of the audio problems others ask about seem to be hardware related, so the following information might be relevant. sudo lshw -c multimedia *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:48 memory:f8400000-f8403fff

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  • Java algorithm for normalizing audio

    - by Marty Pitt
    I'm trying to normalize an audio file of speech. Specifically, where an audio file contains peaks in volume, I'm trying to level it out, so the quiet sections are louder, and the peaks are quieter. I know very little about audio manipulation, beyond what I've learnt from working on this task. Also, my math is embarrassingly weak. I've done some research, and the Xuggle site provides a sample which shows reducing the volume using the following code: (full version here) @Override public void onAudioSamples(IAudioSamplesEvent event) { // get the raw audio byes and adjust it's value ShortBuffer buffer = event.getAudioSamples().getByteBuffer().asShortBuffer(); for (int i = 0; i < buffer.limit(); ++i) buffer.put(i, (short)(buffer.get(i) * mVolume)); super.onAudioSamples(event); } Here, they modify the bytes in getAudioSamples() by a constant of mVolume. Building on this approach, I've attempted a normalisation modifies the bytes in getAudioSamples() to a normalised value, considering the max/min in the file. (See below for details). I have a simple filter to leave "silence" alone (ie., anything below a value). I'm finding that the output file is very noisy (ie., the quality is seriously degraded). I assume that the error is either in my normalisation algorithim, or the way I manipulate the bytes. However, I'm unsure of where to go next. Here's an abridged version of what I'm currently doing. Step 1: Find peaks in file: Reads the full audio file, and finds this highest and lowest values of buffer.get() for all AudioSamples @Override public void onAudioSamples(IAudioSamplesEvent event) { IAudioSamples audioSamples = event.getAudioSamples(); ShortBuffer buffer = audioSamples.getByteBuffer().asShortBuffer(); short min = Short.MAX_VALUE; short max = Short.MIN_VALUE; for (int i = 0; i < buffer.limit(); ++i) { short value = buffer.get(i); min = (short) Math.min(min, value); max = (short) Math.max(max, value); } // assign of min/max ommitted for brevity. super.onAudioSamples(event); } Step 2: Normalize all values: In a loop similar to step1, replace the buffer with normalized values, calling: buffer.put(i, normalize(buffer.get(i)); public short normalize(short value) { if (isBackgroundNoise(value)) return value; short rawMin = // min from step1 short rawMax = // max from step1 short targetRangeMin = 1000; short targetRangeMax = 8000; int abs = Math.abs(value); double a = (abs - rawMin) * (targetRangeMax - targetRangeMin); double b = (rawMax - rawMin); double result = targetRangeMin + ( a/b ); // Copy the sign of value to result. result = Math.copySign(result,value); return (short) result; } Questions: Is this a valid approach for attempting to normalize an audio file? Is my math in normalize() valid? Why would this cause the file to become noisy, where a similar approach in the demo code doesn't?

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  • How can I determine what codec is being used?

    - by jldugger
    This forum comment and this superuser answer suggest that the audio compression contributes to loss of quality. I've noticed that music played over my BT setup sometimes pitch bends in ways I don't remember the original doing, and I'm wondering if SBC has something to do with it. I'm using Ubuntu 10.10 on a Mac Pro, connecting to a pair of Sony DR-BT50's. Is there a way to inspect which Bluetooth codec pulseaudio is using, what codecs both ends of the bluetooth link support?

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  • How do audio based games such as Audiosurf and Beat Hazard work?

    - by The Communist Duck
    Note: I am not asking how to make a clone of one of these. I am asking about how they work. I'm sure everyone's seen the games where you use your own music files (or provided ones) and the games produce levels based on them, such as Audiosurf and Beat Hazard. Here is a video of Audiosurf in action, to show what I mean. If you provide a heavy metal song, you would get a completely different set of obstacles, enemies, and game experience from something like Vivaldi. What does interest me is how these games work. I do not know much about audio (well, data-side), but how do they process the song to understand when it is settling down or when it's speeding up? I guess they could just feed the pitch values (assuming those sorts of things exist in audio files) to form a level, but it wouldn't fully explain it. I'm either looking for an explanation, some links to articles about this sort of thing (I'm sure there's a term or terms for it), or even an open-source implementation of this kind of thing ;-) EDIT: After some searching and a little help, I found out about FFT (Fast Fourier Transform). This maybe a step in the right direction, but it is something that does not make any sense to me..or fits with my physics knowledge of waves.

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  • Is it possible for a faulty processor to cause audio static/noise?

    - by Tom
    I have a Core 2 Extreme processor I received from a friend and have set up an XBMC box using it. However, I constantly get audio static whenever playing any music or videos. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 I have tried replacing everything short of the case and the processor, including cables, audio interfaces, operating systems, ram, etc, leading me to think it might be either the case shorting out the motherboards I have tried or a faulty processor. Is it possible for a faulty processor to cause audio static/noise? Any feedback would be appreciated. Edit - Here's a list of things I have tried: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Switching Power Cable Plugging in through surge protector Plugging into different outlet on separate circuit

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  • [SOLVED} How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. Update: Here's a sample of how I hear others people voices.. Audio Sample After some tests, also the desktop has the same problem. (I tried TeamSpeak3) Here are some details on my laptop and desktop Laptop Dell Studio 1555 Core 2 Duo P8600 2.4Ghz 4Gb Ram Dual Channel Ati HD 4570 512Mb dedicated (up to 2048) IDT High Definition Audio Desktop Motherboard Asus P5KPL-AM Dual Core CPU E5200 2.50Ghz 2x2GB PC6400 Dual Channel Ati Radeon HD 4650 512MB VIA High Definition Audio Both computers have Windows 7 Professional 64Bit. So how do I restore my audio? SOLVED The problem was in router firmware, there was a bug that recognized VoIP traffic as a DOS attack and the router grambled every packet... I've installed the newest firmware and everything is fine :)

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  • How to write C++ audio processing applications?

    - by cesko82
    Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

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  • Looping HTML5 audio on the iPhone

    - by Peeps
    I'm trying to make a HTML5 webapp that simply plays a sound over and over and over again, on my iPhone. I don't know any Obj-C to do it natively. What I have works fine, but the sound only plays once: <!DOCTYPE html> <html> <head> <title>noisemaker!</title> <meta http-equiv="content-type" content="text/html; charset=utf-8" /> <meta name="viewport" content="maximum-scale=1, minimum-scale=1, width=device-width, user-scalable=no" /> <meta name="apple-mobile-web-app-capable" content="yes" /> </head> <body> <audio src="noise.mp3" autoplay controls loop></audio> </body> </html> Is there a way to either bypass the QuickTime audio screen and loop it in the webpage, or get the QuickTime audio screen to loop the sound?

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  • Virtual audio driver (microphone)

    - by Dalamber
    Hello guys, I want to develop a virtual microphone driver. Please, do not say anything about DirectShow - that's not "the way". I need a solution that will work with any software including Skype and MSN. And DirectShow doesn't fit these requirements. I found AVStream Filter-Centric Simulated Capture Driver (avssamp.sys) in Windows 7 WDK. What I need is an audio part of it. By default it reads avssamp.wav and plays it. But this driver is registered as WDM streaming capture device. And I want it in Audio Capture Device. There are some posts in the web but they are all the same: http://www.tech-archive.net/Archive/Development/microsoft.public.development.device.drivers/2005-05/msg00124.html http://www.winvistatips.com/problem-installing-avssamp-audio-capture-sources-category-t184898.html I think registering this filter-driver as audio capture device will make Skype recognize it as a microphone and thefore I will be able to push any PCM file as if it goes from mic. If someone already faced this problem before, please help. Thanks in advance.

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  • How do I restore my audio after uninstalling Ventrilo?

    - by Marcx
    Hi, I've a Dell studio 1555 bought on september with Windows 7 64bit Professional on it. The audio device works proprerly, while listening to audio contents (from disk or internet) When I use Ventrilo, the audio from other people sounds good and I hear their voices clearly When I use any other VOIP programs like Teamspeak 3, MSN or Skype, I hear a disturbed voice, and it's impossible to comprehend something... Anyway everything worked fine until I installed Ventrilo, but removing it didn´t solve my problem. So how do I restore my audio?

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