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  • Command line audio library manager for Linux

    - by Ketil
    Hi all Hear is my set-up, I have a Linux server that is running Music Player Demon, all the audio files are under a dir (/muzik) which is exported by NFS. So to add files to the MPD database, I just drop the files into the /muzik NFS share and up date the MPD db, so far so good, but I would like to keep the dir strucher belowe /muzik in sum sort of order. To achieve this I am using Amarok, wich a start on my laptop and then use the organise files command to sort the files in into a sensible dir strucher based on the tags in the files. Do you know of any command line utility that can do the same thing that I am using Amarok for so I can run it from cron on the server and automate the process? I hope that this make sense.

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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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  • Youtube video processing on iPhone

    - by Chonch
    Hey In my iPhone app, I want to load a video from youtube, perform some basic image processing on it and display it to the user. I am using openCV to do my image processing, and I know I can use it for grabbing all of the frames from the video as well (cvRetrieveFrame). My only problem is that cvRetrieveFrame expects to get its source of type CvCapture* which can be created either by a camera source or by a file name. How can I set a youtube video file as the source for CvCapture*? Thanks,

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  • Image Processing: What are occlusions?

    - by vikramtheone
    Hi Guys, I'm developing an image processing project and I come across the word occlusion in many scientific papers, what do occlusions mean in the context of image processing? The dictionary is only giving a general definition. Can anyone describe them using an image as a context? Vikram

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  • What is massively parallel processing (MPP) ?

    - by HotTester
    Ever since Microsoft introduced sql-server version code-named "Madison" the massively parallel processing (MPP) has got into picture. What exactly is it and how does sql-server is going to benefit from it ? Further is massively parallel processing (MPP) related to parallel computing ? I read about Madison here and about parallel computing here. Thanks in advance.

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  • Gamepad Control for Processing + Android to Control Arduino Robot

    - by Iker
    I would like to create a Multitouch Gamepad control for Processing and use it to control a remote Arduino Robot. I would like to make the GUI on Processing and compile it for Android. Here is the GUI Gamepad for Processing I have created so far: float easing = 0.09; // start position int posX = 50; int posY = 200; // target position int targetX = 50; int targetY = 200; boolean dragging = false; void setup() { size(500,250); smooth(); } void draw() { background(255); if (!dragging) { // calculate the difference in position, apply easing and add to vx/vy float vx = (targetX - (posX)) * easing; float vy = (targetY - (posY)) * easing; // Add the velocity to the current position: make it move! posX += vx; posY += vy; } if(mousePressed) { dragging = true; posX = mouseX; posY = mouseY; } else { dragging = false; } DrawGamepad(); DrawButtons(); } void DrawGamepad() { //fill(0,155,155); //rect(0, 150, 100, 100, 15); ellipseMode(RADIUS); // Set ellipseMode to RADIUS fill(0,155,155); // Set fill to blue ellipse(50, 200, 50, 50); // Draw white ellipse using RADIUS mode ellipseMode(CENTER); // Set ellipseMode to CENTER fill(255); // Set fill to white// ellipse(posX, posY, 35, 35); // Draw gray ellipse using CENTER mode } void DrawButtons() { fill(0,155,155); // Set fill to blue ellipse(425, 225, 35, 35); ellipse(475, 225, 35, 35); fill(255,0,0); // Set fill to blue ellipse(425, 175, 35, 35); ellipse(475, 175, 35, 35); } I have realized that probably that code will not support Multitouch events on Android so I came up with another code found on this link Can Processing handle multi-touch? So the aim of this project is to create de multitouch gamepad to use to control my Arduino Robot. The gamepad should detect which key was pressed as well as the direction of the Joystick. Any help appreciated.

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  • Annotation Processing Virtual Mini-Track at JavaOne 2012

    - by darcy
    Putting together the list of JavaOne talks I'm interested in attending, I noticed there is a virtual mini-track on annotation processing and related technology this year, with a combination of bofs, sessions, and a hands-on-lab: Monday Multidevice Content Display and a Smart Use of Annotation Processing, Dimitri BAELI and Gilles Di Guglielmo Tuesday Advanced Annotation Processing with JSR 269, Jaroslav Tulach Build Your Own Type System for Fun and Profit, Werner Dietl and Michael Ernst Wednesday Annotations and Annotation Processing: What’s New in JDK 8?, Joel Borggrén-Franck Thursday Hack into Your Compiler!, Jaroslav Tulach Writing Annotation Processors to Aid Your Development Process, Ian Robertson As the lead engineer on bot apt (rest in peace) in JDK 5 and JSR 269 in JDK 6, I'd be heartened to see greater adoption and use of annotation processing by Java developers.

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  • I want to record a screencast of a processing sketch

    - by nathanvda
    I have a created a music visualisation using Processing. I now want to convert that to a video, and the least obtrusive way I could think of is to record a screencast. I figured exporting Processing to video including audio, from within Processing itself, on ubuntu seemed an unsolved issue. Very hard and also could cause timing sync issues (since the music keeps running while images are captured). So move on to the screencast method. Dead-easy, I figured. But I was wrong. First hurdle was to find a way to record the sound from the audio (and not the mic). I did find a tutorial for that here. In short: use gtk-recordmydesktop and pulse audio. But, apparently, what happens: Processing does not use ALSA. When the sound is playing, it does not appear in the Pulse Audio mixer. How can I record the audio now?

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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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  • Image processing on bifurcation diagram to get small eps size

    - by yCalleecharan
    Hello, I'm producing bifurcation diagrams (which are normally used in nonlinear dynamics). These diagrams identify abrupt changes in topologies due to stability changes. These abrupt changes occur as one or more parameters pass through some critical value(s). An example is here: http://en.wikipedia.org/wiki/File:LogisticMap_BifurcationDiagram.png On the above figure, some image processing has been done so as to make the plot more visually pleasant. A bifurcation diagram usually contains hundreds of thousands of points and the resulting eps file can become very big. Journal submission in the LaTeX format require that figures are to be submitted in the eps format. In my case one of such figures can result in about 6 MB in Matlab and even much more in Gnuplot. For the example in the above figure, 100,000 x values are calculated for each r and one can imagine that the resulting eps file would be huge. The site however explains some image processing that makes the plot more visually pleasing. Can anyone explain to me stepwise how go about? I can't understand the explanation provided in the "summary" section. Will the resulting image processing also reduce the figure size? Furthermore, any tips on reducing the file size of such a huge eps figure? Thanks a lot...

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  • An Overview of Batch Processing in Java EE 7

    - by Janice J. Heiss
    Up on otn/java is a new article by Oracle senior software engineer Mahesh Kannan, titled “An Overview of Batch Processing in Java EE 7.0,” which explains the new batch processing capabilities provided by JSR 352 in Java EE 7. Kannan explains that “Batch processing is used in many industries for tasks ranging from payroll processing; statement generation; end-of-day jobs such as interest calculation and ETL (extract, load, and transform) in a data warehouse; and many more. Typically, batch processing is bulk-oriented, non-interactive, and long running—and might be data- or computation-intensive. Batch jobs can be run on schedule or initiated on demand. Also, since batch jobs are typically long-running jobs, check-pointing and restarting are common features found in batch jobs.” JSR 352 defines the programming model for batch applications plus a runtime to run and manage batch jobs. The article covers feature highlights, selected APIs, the structure of Job Scheduling Language, and explains some of the key functions of JSR 352 using a simple payroll processing application. The article also describes how developers can run batch applications using GlassFish Server Open Source Edition 4.0. Kannan summarizes the article as follows: “In this article, we saw how to write, package, and run simple batch applications that use chunk-style steps. We also saw how the checkpoint feature of the batch runtime allows for the easy restart of failed batch jobs. Yet, we have barely scratched the surface of JSR 352. With the full set of Java EE components and features at your disposal, including servlets, EJB beans, CDI beans, EJB automatic timers, and so on, feature-rich batch applications can be written fairly easily.” Check out the article here.

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  • processing an audio wav file with C

    - by sa125
    Hi - I'm working on processing the amplitude of a wav file and scaling it by some decimal factor. I'm trying to wrap my head around how to read and re-write the file in a memory-efficient way while also trying to tackle the nuances of the language (I'm new to C). The file can be in either an 8- or 16-bit format. The way I thought of doing this is by first reading the header data into some pre-defined struct, and then processing the actual data in a loop where I'll read a chunk of data into a buffer, do whatever is needed to it, and then write it to the output. #include <stdio.h> #include <stdlib.h> typedef struct header { char chunk_id[4]; int chunk_size; char format[4]; char subchunk1_id[4]; int subchunk1_size; short int audio_format; short int num_channels; int sample_rate; int byte_rate; short int block_align; short int bits_per_sample; short int extra_param_size; char subchunk2_id[4]; int subchunk2_size; } header; typedef struct header* header_p; void scale_wav_file(char * input, float factor, int is_8bit) { FILE * infile = fopen(input, "rb"); FILE * outfile = fopen("outfile.wav", "wb"); int BUFSIZE = 4000, i, MAX_8BIT_AMP = 255, MAX_16BIT_AMP = 32678; // used for processing 8-bit file unsigned char inbuff8[BUFSIZE], outbuff8[BUFSIZE]; // used for processing 16-bit file short int inbuff16[BUFSIZE], outbuff16[BUFSIZE]; // header_p points to a header struct that contains the file's metadata fields header_p meta = (header_p)malloc(sizeof(header)); if (infile) { // read and write header data fread(meta, 1, sizeof(header), infile); fwrite(meta, 1, sizeof(meta), outfile); while (!feof(infile)) { if (is_8bit) { fread(inbuff8, 1, BUFSIZE, infile); } else { fread(inbuff16, 1, BUFSIZE, infile); } // scale amplitude for 8/16 bits for (i=0; i < BUFSIZE; ++i) { if (is_8bit) { outbuff8[i] = factor * inbuff8[i]; if ((int)outbuff8[i] > MAX_8BIT_AMP) { outbuff8[i] = MAX_8BIT_AMP; } } else { outbuff16[i] = factor * inbuff16[i]; if ((int)outbuff16[i] > MAX_16BIT_AMP) { outbuff16[i] = MAX_16BIT_AMP; } else if ((int)outbuff16[i] < -MAX_16BIT_AMP) { outbuff16[i] = -MAX_16BIT_AMP; } } } // write to output file for 8/16 bit if (is_8bit) { fwrite(outbuff8, 1, BUFSIZE, outfile); } else { fwrite(outbuff16, 1, BUFSIZE, outfile); } } } // cleanup if (infile) { fclose(infile); } if (outfile) { fclose(outfile); } if (meta) { free(meta); } } int main (int argc, char const *argv[]) { char infile[] = "file.wav"; float factor = 0.5; scale_wav_file(infile, factor, 0); return 0; } I'm getting differing file sizes at the end (by 1k or so, for a 40Mb file), and I suspect this is due to the fact that I'm writing an entire buffer to the output, even though the file may have terminated before filling the entire buffer size. Also, the output file is messed up - won't play or open - so I'm probably doing the whole thing wrong. Any tips on where I'm messing up will be great. Thanks!

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