Search Results

Search found 6479 results on 260 pages for 'audio analysis'.

Page 50/260 | < Previous Page | 46 47 48 49 50 51 52 53 54 55 56 57  | Next Page >

  • Setting up port forwarding for 7000 appliance VM in VirtualBox

    - by uejio
    I've been using the 7000 appliance VM for a lot of testing lately and relied on others to set up the networking for the VM for me, but finally, I decided to take the dive and do it myself.  After some experimenting, I came up with a very brief number of steps to do this all using the VirtualBox CLI instead of the GUI. First download the VM image and unpack it somewhere.  I put it in /var/tmp. Then, set your VBOX_USER_HOME to some place with lots of disk space and import the VM: export VBOX_USER_HOME=/var/tmp/MyVirtualBoxVBoxManage import /var/tmp/simulator/vbox-2011.1.0.0.1.1.8/Sun\ ZFS\ Storage\ 7000.ovf (go get a cup of tea...) Then, set up port forwarding of the VM appliance BUI and shell:First set up port as NAT:VBoxManage modifyvm Sun_ZFS_Storage_7000 --nic1 nat Then set up rules for port forwarding (pick some unused port numbers):VBoxManage modifyvm Sun_ZFS_Storage_7000 --natpf1 "guestssh,tcp,,4622,,22"VBoxManage modifyvm Sun_ZFS_Storage_7000 --natpf1 "guestbui,tcp,,46215,,215" Verify the settings using:VBoxManage showvminfo Sun_ZFS_Storage_7000 | grep -i nic Start the appliance:$ VBoxHeadless --startvm Sun_ZFS_Storage_7000 & Connect to it using your favorite RDP client.  I use a Sun Ray, so I use the Sun Ray Windows Connector client: $ /opt/SUNWuttsc/bin/uttsc -g 800x600 -P <portnumber> <your-hostname> & The portnumber is displayed in the output of the --startvm command.(This did not work after I updated to VirtualBox 4.1.12, so maybe at this point, you need to use the VirtualBox GUI.) It takes a while to first bring up the VM, so please be patient. The longest time is in loading the smf service descriptions, but fortunately, that only needs to be done the first time the VM boots.  There is also a delay in just booting the appliance, so give it some time. Be sure to set the NIC rule on only one port and not all ports otherwise there will be a conflict in ports and it won't work. After going through the initial configuration screen, you can connect to it using ssh or your browser: ssh -p 45022 root@<your-host-name> https://<your-host-name>:45215 BTW, for the initial configuration, I only had to set the hostname and password.  The rest of the defaults were set by VirtualBox and seemed to work fine.

    Read the article

  • Kubuntu solution for analysing/recording streamed flash audio?

    - by marcusw
    I have a Kubuntu system which I stream amateur radio sound to via a flash interface. I want to be able to record the sound that the flash player is making at the press of a button. I also need the capability to do (hopefully real-time) spectrum analysis on the sound. I need a program (a firefox add-on would be ideal) that can do this for me.

    Read the article

  • Lots of static/crackling noises after ALSA HDA DKMS installation

    - by MartinB
    I am using a Samsung Chronos 7 laptop with the following sound setup: $ head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Realtek ALC269VC ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI With the stock ALSA that comes with Ubuntu 12.04, I do not get any sound out of the headphones when I plug them into the headphone jack. After plugging the headphones in, I have to manually use Alsamixer to increase the volume, so that the headphones become usable. I have been told that this issue is due to my sound chip not being supported in the ALSA version that ships with Precise. A similar question at AskUbuntu and the Ubuntu Community Documentation point me to the ALSA DKMS installation. After installing the dkms module of yesterday's ALSA snapshot and rebooting, the headphone issue is indeed solved. I can now plug my headphones into the jack and instantly have sound on them. However, now I have tons of static noises and crackling when playing sound in VLC player or Skype (Firefox HTML5 playback seems to be fine, unless a Skype sound interferes with it). Is there a fix for this? I tried adding the Alsa PPA and installing the latest ALSA package proper, but that didn't have any effect, only the Alsa DKMS package seems to solve the headphone issue.

    Read the article

  • How to make Bluetooth headset appear as a sound device?

    - by torbengb
    I have an onboard VIA sound card but no speakers attached. Instead, I want to use my bluetooth headset as my only sound device, both input and output (mono). I plugged in the bt dongle and was impressed that it was ready to use within seconds. I then paired it to my Jabra BT500 headset. No problems so far. I then went to Sound Preferences but no bluetooth is listed, only the onboard VIA sound card. Question: How can I enable my headset, and make it the default all the time?

    Read the article

  • Looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs <= $5

    - by CyanPrime
    I'm looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs less then, or is $5 usd. I want to have a nice BGM for my engine demo I'm going to release for a game I'm planing on having go commercial. I don't want to spend too much money on it, so my limit is $5 usd. I want it to be at least a 1:00 in length. Where should I look? Or even better, do you have a link to a song that meets the criteria?

    Read the article

  • Where to get sounds for game development for kids [closed]

    - by at.
    I'm teaching kids to program using Ruby and the gaming framework Gosu/Chingu. For the sounds for their games I've been showing them http://www.bfxr.net/. It's decent, but the samples are limited and some of them are pretty cheap (check the explosion, it's like an explosion on a commodore 64 game). Is there an easy resource kids can get the sounds they want? I'm happy to pay some kind of educational license for it.

    Read the article

  • Music player with a few specific requirements

    - by Jordan Uggla
    I am looking for a music player with a few specific requirements: Must have a search function that whittles down results as you type, searching the entire library. Must start playing a song when double clicked, and not continue to another song when that song finishes. Must be approachable and immediately usable by people completely unfamiliar with the program. I think this is mostly covered by the first two requirements being met. I've tried many players but unfortunately every one has failed to meet at least one of the requirements. Rhythmbox meets 1 and 3, but continues to the next search result after the song which was double clicked ends. Banshee is basically the same as Rhythmbox. While it has an option to "Stop when finished" this cannot (as far as I can tell) be made the default when double clicking a song. Audacious (as far as I can tell) fails at 1. Muine meets requirements 1 and 2, but unfortunately I couldn't make the search dialog always shown like it is with Rhythmbox / Banshee which, despite its very simple interface, made Muine incomprehensible to people trying to use it for the first time. Amarok I could not configure to meet requirement 1, but I think it's likely I was just missing something, and with its configurability I'm confident that I can set it up to meet requirements 2 and 3.

    Read the article

  • Difference between Sound and Music

    - by Southpaw Hare
    What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

    Read the article

  • Is there a media player that works on HTTPS sites?

    - by Iain Hallam
    I'm currently using Yahoo! Media Player for a site that needs to play MP3 files that are stored on our server. In total, there's quite a bit more than the free limits at Soundcloud, but each file is only a few minutes long. YMP is pretty good, but causes security warnings on HTTPS pages, because it can only be served via HTTP. Is there an equivalent free player I can embed for the HTTPS pages? EDIT: Just to clarify, I'm initially looking for something that will scan the page and turn media links playable.

    Read the article

  • Why is this beat detection code failing to register some beats properly?

    - by Quincy
    I made this SoundAnalyzer class to detect beats in songs: class SoundAnalyzer { public SoundBuffer soundData; public Sound sound; public List<double> beatMarkers = new List<double>(); public SoundAnalyzer(string path) { soundData = new SoundBuffer(path); sound = new Sound(soundData); } // C = threshold, N = size of history buffer / 1024 B = bands public void PlaceBeatMarkers(float C, int N, int B) { List<double>[] instantEnergyList = new List<double>[B]; GetEnergyList(B, ref instantEnergyList); for (int i = 0; i < B; i++) { PlaceMarkers(instantEnergyList[i], N, C); } beatMarkers.Sort(); } private short[] getRange(int begin, int end, short[] array) { short[] result = new short[end - begin]; for (int i = 0; i < end - begin; i++) { result[i] = array[begin + i]; } return result; } // get a array of with a list of energy for each band private void GetEnergyList(int B, ref List<double>[] instantEnergyList) { for (int i = 0; i < B; i++) { instantEnergyList[i] = new List<double>(); } short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; int samplesPerBand = nextSamples / B; // for the whole song while (sampleIndex + nextSamples < samples.Length) { complex[] FFT = FastFourier.Calculate(getRange(sampleIndex, nextSamples + sampleIndex, samples)); // foreach band for (int i = 0; i < B; i++) { double energy = 0; for (int j = 0; j < samplesPerBand; j++) energy += FFT[i * samplesPerBand + j].GetMagnitude(); energy /= samplesPerBand; instantEnergyList[i].Add(energy); } if (sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; samplesPerBand = nextSamples / B; } } // place the actual markers private void PlaceMarkers(List<double> instantEnergyList, int N, float C) { double timePerSample = 1 / (double)soundData.SampleRate; int index = N; int numInBuffer = index; double 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++; } } } For some reason it's only detecting beats from 637 sec to around 641 sec, and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates, and it seems that it's assigning a beat to each instant energy value in between those values. It's modeled after this: http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats register properly?

    Read the article

  • How do I get a Line6 UX1 soundcard to work?

    - by the_drow
    I own a Line6 UX1 soundcard and I would like to make it work for Ubuntu. I have followed the instructions here and it worked. But at some point I upgraded my kernel version (not sure what uname -a prints but it's related) and it stopped working. Here's what uname -a prints: Linux ubuntu 2.6.32-29-generic #58-Ubuntu SMP Fri Feb 11 20:52:10 UTC 2011 x86_64 GNU/Linux I figured out that maybe it's installed per version so I used svn update and hit make again. My guess was right as it copied the relevant files to the new version's folder. I restarted and still nothing. Should I revert to an older version? Or is there a solution here?

    Read the article

  • How to trace a function array argument in DTrace

    - by uejio
    I still use dtrace just about every day in my job and found that I had to print an argument to a function which was an array of strings.  The array was variable length up to about 10 items.  I'm not sure if the is the right way to do it, but it seems to work and is not too painful if the array size is small.Here's an example.  Suppose in your application, you have the following function, where n is number of item in the array s.void arraytest(int n, char **s){    /* Loop thru s[0] to s[n-1] */}How do you use DTrace to print out the values of s[i] or of s[0] to s[n-1]?  DTrace does not have if-then blocks or for loops, so you can't do something like:    for i=0; i<arg0; i++        trace arg1[i]; It turns out that you can use probe ordering as a kind of iterator. Probes with the same name will fire in the order that they appear in the script, so I can save the value of "n" in the first probe and then use it as part of the predicate of the next probe to determine if the other probe should fire or not.  So the first probe for tracing the arraytest function is:pid$target::arraytest:entry{    self->n = arg0;}Then, if I want to print out the first few items of the array, I first check the value of n.  If it's greater than the index that I want to print out, then I can print that index.  For example, if I want to print out the 3rd element of the array, I would do something like:pid$target::arraytest:entry/self->n > 2/{    printf("%s",stringof(arg1 + 2 * sizeof(pointer)));}Actually, that doesn't quite work because arg1 is a pointer to an array of pointers and needs to be copied twice from the user process space to the kernel space (which is where dtrace is). Also, the sizeof(char *) is 8, but for some reason, I have to use 4 which is the sizeof(uint32_t). (I still don't know how that works.)  So, the script that prints the 3rd element of the array should look like:pid$target::arraytest:entry{    /* first, save the size of the array so that we don't get            invalid address errors when indexing arg1+n. */    self->n = arg0;}pid$target::arraytest:entry/self->n > 2/{    /* print the 3rd element (index = 2) of the second arg. */    i = 2;    size = 4;    self->a_t = copyin(arg1+size*i,size);    printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}If your array is large, then it's quite painful since you have to write one probe for every array index.  For example, here's the full script for printing the first 5 elements of the array:#!/usr/sbin/dtrace -spid$target::arraytest:entry{        /* first, save the size of the array so that we don't get           invalid address errors when indexing arg1+n. */        self->n = arg0;}pid$target::arraytest:entry/self->n > 0/{        i = 0;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 1/{        i = 1;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 2/{        i = 2;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 3/{        i = 3;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 4/{        i = 4;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));} If the array is large, then your script will also have to be very long to print out all values of the array.

    Read the article

  • What's the difference between Pygame's Sound and Music classes?

    - by Southpaw Hare
    What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

    Read the article

  • C# performance analysis- how to count CPU cycles?

    - by Lirik
    Is this a valid way to do performance analysis? I want to get nanosecond accuracy and determine the performance of typecasting: class PerformanceTest { static double last = 0.0; static List<object> numericGenericData = new List<object>(); static List<double> numericTypedData = new List<double>(); static void Main(string[] args) { double totalWithCasting = 0.0; double totalWithoutCasting = 0.0; for (double d = 0.0; d < 1000000.0; ++d) { numericGenericData.Add(d); numericTypedData.Add(d); } Stopwatch stopwatch = new Stopwatch(); for (int i = 0; i < 10; ++i) { stopwatch.Start(); testWithTypecasting(); stopwatch.Stop(); totalWithCasting += stopwatch.ElapsedTicks; stopwatch.Start(); testWithoutTypeCasting(); stopwatch.Stop(); totalWithoutCasting += stopwatch.ElapsedTicks; } Console.WriteLine("Avg with typecasting = {0}", (totalWithCasting/10)); Console.WriteLine("Avg without typecasting = {0}", (totalWithoutCasting/10)); Console.ReadKey(); } static void testWithTypecasting() { foreach (object o in numericGenericData) { last = ((double)o*(double)o)/200; } } static void testWithoutTypeCasting() { foreach (double d in numericTypedData) { last = (d * d)/200; } } } The output is: Avg with typecasting = 468872.3 Avg without typecasting = 501157.9 I'm a little suspicious... it looks like there is nearly no impact on the performance. Is casting really that cheap?

    Read the article

  • Sentiment analysis for twitter in python

    - by Ran
    I'm looking for an open source implementation, preferably in python, of Textual Sentiment Analysis (http://en.wikipedia.org/wiki/Sentiment_analysis). Is anyone familiar with such open source implementation I can use? I'm writing an application that searches twitter for some search term, say "youtube", and counts "happy" tweets vs. "sad" tweets. I'm using Google's appengine, so it's in python. I'd like to be able to classify the returned search results from twitter and I'd like to do that in python. I haven't been able to find such sentiment analyzer so far, specifically not in python. Are you familiar with such open source implementation I can use? Preferably this is already in python, but if not, hopefully I can translate it to python. Note, the texts I'm analyzing are VERY short, they are tweets. So ideally, this classifier is optimized for such short texts. BTW, twitter does support the ":)" and ":(" operators in search, which aim to do just this, but unfortunately, the classification provided by them isn't that great, so I figured I might give this a try myself. Thanks! BTW, an early demo is here and the code I have so far is here and I'd love to opensource it with any interested developer.

    Read the article

  • Routing audio from GSM module to a Bluetooth HandsFree device

    - by Shaihi
    I have a system with the following setup: I use: Windows CE 6 R3 Microsoft's Bluetooth stack including all profiles Motorola H500 The Audio Gateway service is up and running (checked through services list in cmd) GSM Module is functional - I am able to set outgoing calls and to answer calls. Bluetooth is functional - the A2DP profile plays music to Motorola headphones (can't remember the model right now) I want to hold a conversation using a headset device. I have included all Bluetooth components in the catalog. I pair the device using the Control Panel applet. When I press the button on the Motorla device to answer a call I get a print by the Audio Gateway: BTAGSVC: ConnectionEvent. BTAGSVC: SCOListenThread_Int - Connection Event. BTAGSVC: ConnectionEvent. BTAGSVC: SCOListenThread_Int - Connection Event. BTAGSVC: ConnectionEvent. BTAGSVC: A Bluetooth peer device has connected to the Audio Gateway. BTAGSVC: Could not open registry key for BT Addr: 2. BTAGSVC: The peer device was not accepted since the user has never confirmed it as a device to be used. So my questions are as follows: What do I need to do to pair the device with the Audio Gateway? Once my device is paired, do I need to set anything else up? (except for the GSM module of course)

    Read the article

  • Usage of static analysis tools - with Clear Case/Quest

    - by boyd4715
    We are in the process of defining our software development process and wanted to get some feed back from the group about this topic. Our team is spread out - US, Canada and India - and I would like to put into place some simple standard rules that all teams will apply to their code. We make use of Clear Case/Quest and RAD I have been looking at PMD, CPP, checkstyle and FindBugs as a start. My thought is to just put these into ANT and have the developers run these manually. I realize doing this you have to have some trust in that each developer will do this. The other thought is to add in some builders in to the IDE which would run a subset of the rules (keep the build process light) and then add another set (heavy) when they check in the code. Some other ideals is to make use of something like Cruse Control and have it set up to run these static analysis tools along with the unit test when ever Clear Case/Quest is idle. Wondering if others have done this and if it was successfully or can provide lessons learned.

    Read the article

  • exclude dependencies when running sonar analysis

    - by achraf
    I have a test project requiring some heavy jars which i put in ${M2_HOME}\test\src\main\resources\ and add them in the pom.xml using : <dependency> <groupId>server</groupId> <artifactId>server</artifactId> <version>1.0</version> <scope>system</scope> <systemPath>${M2_HOME}\test\src\main\resources\server.jar</systemPath> </dependency> <dependency> <groupId>client</groupId> <artifactId>client</artifactId> <version>6.0</version> <scope>system</scope> <systemPath>${M2_HOME}\test\src\main\resources\client.jar</systemPath> </dependency> I want to know if it possible to exclude them during sonar analysis, or generally just analyze java sources folder.

    Read the article

  • Method for launching audio player on Android from web page for streaming media

    - by Brad
    To link to SHOUTcast/HTTP internet radio streams, traditionally you would link to a playlist file, such as an M3U or PLS. From there, the browser would launch the audio player registered to handle the playlist. This works great on any PC, Palm, Blackberry, and iPhone. This method does not work in Android without installing extra software. Sure, Just Playlists or StreamFurious can handle it just fine, but I am assuming there has to be a way to invoke the audio or video player commonly installed by default on Android installations. By default, no audio player is capable of handling M3U or PLS. The player seems to open it, but says "Unsupported Media Type". To make this more annoying, the browser is capable of streaming MP3 audio over HTTP, simply by opening a link to an MP3 file. I have tried simply linking directly to the MP3 stream hosted by SHOUTcast, which should end up in the same result, but SHOUTcast detects "Mozilla" in the user-agent string, and instead of sending the stream, it sends the information page for the station. How should I link to a SHOUTcast stream on Android, from a normal mobile site, without using extra applications?

    Read the article

  • Playing video and audio in iPhone not working...

    - by Scott
    So we have buttons linked up to display images/videos/audio on click depending on a check we do earlier. That part works fine. It knows which one to play, however, when we click the buttons for video and audio, nothing happens. The image one works fine. The video and audio are being taken for a URL online, they are not local, but everywhere said this was still possible. Here is a little snippet of the code where we play the two files: if ( [fName hasSuffix:@".png"]) { NSLog(@"PICTURE"); NSURL *url = [NSURL URLWithString: fName]; UIImage *image = [UIImage imageWithData: [NSData dataWithContentsOfURL:url]]; self.view = [[UIView alloc] initWithFrame:[[UIScreen mainScreen] applicationFrame]]; // self.view.backgroundColor = [[UIColor alloc] initWithPatternImage:[UIImage imageNamed:@"MainBG.jpg"]]; [self.view addSubview:[[UIImageView alloc] initWithImage:image]]; } if ( [fName hasSuffix:@".mp4"]) { NSLog(@"VIDEO"); //NSString *path = [[NSBundle mainBundle] pathForResource:fName ofType:@"mp4"]; //NSLog(path); NSURL *url = [NSURL fileURLWithPath:fName]; MPMoviePlayerController *player = [[MPMoviePlayerController alloc] initWithContentURL:url]; [player play]; } if ( [fName hasSuffix:@".mp3"]) { NSLog(@"AUDIO"); NSURL *url = [NSURL fileURLWithPath:fName]; NSData *soundData = [NSData dataWithContentsOfURL:url]; AVAudioPlayer *avPlayer = [[AVAudioPlayer alloc] initWithData:soundData error: nil]; [avPlayer play]; } See anything wrong? By the way it compiles and runs, but nothing happens when we hit the button that executes that code.

    Read the article

  • Syntactical analysis with Flex/Bison part 2

    - by Imran
    Hallo, I need help in Lex/Yacc Programming. I wrote a compiler for a syntactical analysis for inputs of many statements. Now i have a special problem. In case of an Input the compiler gives the right output, which statement is uses, constant operator or a jmp instructor to which label, now i have to write so, if now a if statement comes, first the first command (before the else) must be give out when the assignment of the if is yes then it must jump to the end because the command after the else isnt needed, so after this jmp then the second command must be give out. I show it in an example maybe you understand what i mean. Input adr. Output if(x==0) 10 if(x==0) Wait 5 20 WAIT 5 else 30 JMP 50 Wait 1 40 WAIT 1 end 50 END like so. I have an idea, maybe i can do it whith a special if statement like IF exp jmp_stmt_end stmt_seq END when the if statement is given in the input the compiler has to recognize the end ofthe statement and like my jmp_stmt in my compiler ( you have to download the files from http://bitbucket.org/matrix/changed-tiny) only to jump to the end. I hope you understand my problem.thanks.

    Read the article

  • visual analysis of web pages in ruby

    - by Clint Miller
    I'm looking to write some code that does visual analysis of web pages, preferably using Ruby. My code will need to be able to determine the top, left, width, height, background color, color, and font size for all the elements in the DOM. Of course, these values can only be calculated once all CSS is applied. So, I don't think that Nokogiri is up for the job. Ultimately, I'm trying to use this data in a VIPS-like (Vision-Based Page Segmentation) algorithm in an attempt to find the main content in downloaded news articles. I've considered using Watir to drive Chrome or Firefox and then extract the data. The problem is that browsers can't be run headless through Watir (I think). Ultimately, this code will be running on an array of Linux servers in a data center. So, the code won't have easy access to an X Server for displaying the browser. I suppose one solution is to use Watir and run a headless X Server on the Linux servers. That's a bit of a pain, but it looks like my best option right now. Does anyone have any better ideas?

    Read the article

  • Visualizing volume of PCM samples

    - by genevincent
    I have several chunks of PCM audio (G.711) in my C++ application. I would like to visualize the different audio volume in each of these chunks. My first attempt was to calculate the average of the sample values for each chunk and use that as an a volume indicator, but this doesn't work well. I do get 0 for chunks with silence and differing values for chunks with audio, but the values only differ slighly and don't seem to resemble the actual volume. What would be a better algorithem calculate the volume ? I hear G.711 audio is logarithmic PCM. How should I take that into account ?

    Read the article

  • General question about DirectShow.NET, DirectShow and Windows Media Format

    - by Paul Andrews
    I searched and googled for an answer but couldn't find one. Basically I'm developing a webcam/audio streaming application which should capture audio and video from a pc (usb webcam/microphone) and send them to a receiving server. What the server will do with that it's another story and phase two (which I'm skipping for now) I wrote some code using DirectShow and Windows Media Format and it worked great for capture audio/video and sending them to another client, but there's a major problem: latency. Everywhere in the internet everyone gave me the same answer: "sorry dude but media format isn't for video conferencing, their codecs have too high latency". I thought I could skip the .wmv problems but seems like it's not possible to do... this road ends here then. So I saw a few examples with DirectShow.NET which were faster for both audio and video.. my question is: how come that DirectShow.NET is faster and better for video/audio conferencing? Shouldn't it be just a .NET porting of C++'s DirectShow? Am I missing something? I'm a bit confused at this point

    Read the article

  • AudioFileWriteBytes fails with error code -40

    - by alexbw
    I'm trying to write raw audio bytes to a file using AudioFileWriteBytes(). Here's what I'm doing: void writeSingleChannelRingBufferDataToFileAsSInt16(AudioFileID audioFileID, AudioConverterRef audioConverter, ringBuffer *rb, SInt16 *holdingBuffer) { // First, figure out which bits of audio we'll be // writing to file from the ring buffer UInt32 lastFreshSample = rb->lastWrittenIndex; OSStatus status; int numSamplesToWrite; UInt32 numBytesToWrite; if (lastFreshSample < rb->lastReadIndex) { numSamplesToWrite = kNumPointsInWave + lastFreshSample - rb->lastReadIndex - 1; } else { numSamplesToWrite = lastFreshSample - rb->lastReadIndex; } numBytesToWrite = numSamplesToWrite*sizeof(SInt16); Then we copy the audio data (stored as floats) to a holding buffer (SInt16) that will be written directly to the file. The copying looks funky because it's from a ring buffer. UInt32 buffLen = rb->sizeOfBuffer - 1; for (int i=0; i < numSamplesToWrite; ++i) { holdingBuffer[i] = rb->data[(i + rb->lastReadIndex) & buffLen]; } Okay, now we actually try to write the audio from the SInt16 buffer "holdingBuffer" to the audio file. The NSLog will spit out an error -40, but also claims that it's writing bytes. No data is written to file. status = AudioFileWriteBytes(audioFileID, NO, 0, &numBytesToWrite, &holdingBuffer); rb->lastReadIndex = lastFreshSample; NSLog(@"Error = %d, wrote %d bytes", status, numBytesToWrite); return;

    Read the article

< Previous Page | 46 47 48 49 50 51 52 53 54 55 56 57  | Next Page >