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  • Installing Skype on Amazon EC2 instance

    - by Adrian
    For my application, I need to have Skype working on my Amazon EC2 Windows instance. I got the application installed and am able to log in, however, I can't make a phone call, since I am getting an 'Can't detect your sound card' error. Since I'm trying to inject audio from an audio file into the phone call, I don't need the sound card on the server. Thus, I need a way to bypass this error message. I have tried installing Virtual Audio Cable, which unfortunately didn't work (even though it worked on my desktop machine).

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  • What is the best API in any language for Audio and MIDI music application development?

    - by noneme
    What, in your opinion, is the best API to utilize in developing an application that handles both realtime MIDI and audio input and output? This would be for an application that is used in the process of making music as opposed to playing audio or MIDI files. I'm aware that this may be a subjective question, but if you know of an API that is dominantly used for these purposes, please share it. I'm agnostic about which language the API is for, and I also don't care about portability. The real concern is for an API that is well documented, well designed (e.g. thought out and intuitive to developers using it), and actively maintained. OS portability would be nice, but it is second to having an API/Language that meets the previous requirements. Please note that the emphasis is not on API's for sound synthesis or for composing music with code. It is intended for the handling of sound file and MIDI data in a real-time context.

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  • What should I use to replace the WinAPI Beep() function?

    - by Jon Cage
    I've got a Visual C++/CLI app which uses beeps to signify good and bad results (used when the user can't see the screen). Currently I use low pitched beeps for bad results and high pitched beeps for good results: if( goodResult == true ) { Beep(1000, 40); } else { Beep(2000, 20); } This works okay on my Vista laptop, but I've tried it on other laptops and some seem to play the sounds for less time (they sound more like clicks than beeps) or the sound doesn't play at all. So I have two questions here: Is there a more reliable beep function? Is there a (simple) way I can play a short .wav file or something similar instead (preferred solution).

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  • How to redirect sound to USB headset when plugged in?

    - by LM
    I often have to switch between audio output from my speakers and my headset (P5Q mobo with integrated sound and Microsoft headset). I've already got it so that when my headset is plugged in, sound will be played through it, and if it isn't, sound will play through my speakers. The problem is that if I have a game or similar program started while my headset is plugged in, if I unplug it, I will get no sound. Also, if I start the program with no headset, and plug it in, I get sound still through speakers. Is there any way to do this?

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  • How do I turn off the click sound when closing a tab in chrome?

    - by nos
    Every time I close a tab in Chrome, it makes a click sound. How do I turn off that sound? I reported that issue back in Oct 2010. The problem doesn't appear on all clients and the reason is still unclear. Common attempts at solving the issue include simply turning off the sound in Windows. But I would prefer to solve the problem at the source. Why is Chrome even triggering that sound to be played? And why is it delayed? The problem would be far less annoying if the sound could easily be related to the action taken. Installing the Chrome Toolbox and muting all tabs has no effect on this issue. When switching to a different Chrome user profile, the new user profile does exhibit the same issue.

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  • Can I run alsa and pulse side by side ? I think there is some problem with the alsa ! My ubunu login sound and alert sound are not working?

    - by Curious Apprentice
    I think I have Alsa driver installed. Pulse not working may be I dont have it installed. Not sure If I can run Pulse and Alsa. I had to configure each application prior to work which use pulse.(SMplayer by default select pulse. I had to change that) I know a little about these. So if the question is stupid then please help me. Smplayer always showing a cross(x) icon in front of speaker icon as it is disabled, though Im playing sound.

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • Graphing the pitch (frequency) of a sound

    - by Coronatus
    I want to plot the pitch of a sound into a graph. Currently I can plot the amplitude. The graph below is created by the data returned by getUnscaledAmplitude(): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(new FileInputStream(file))); byte[] bytes = new byte[(int) (audioInputStream.getFrameLength()) * (audioInputStream.getFormat().getFrameSize())]; audioInputStream.read(bytes); // Get amplitude values for each audio channel in an array. graphData = type.getUnscaledAmplitude(bytes, this); public int[][] getUnscaledAmplitude(byte[] eightBitByteArray, AudioInfo audioInfo) { int[][] toReturn = new int[audioInfo.getNumberOfChannels()][eightBitByteArray.length / (2 * audioInfo. getNumberOfChannels())]; int index = 0; for (int audioByte = 0; audioByte < eightBitByteArray.length;) { for (int channel = 0; channel < audioInfo.getNumberOfChannels(); channel++) { // Do the byte to sample conversion. int low = (int) eightBitByteArray[audioByte]; audioByte++; int high = (int) eightBitByteArray[audioByte]; audioByte++; int sample = (high << 8) + (low & 0x00ff); if (sample < audioInfo.sampleMin) { audioInfo.sampleMin = sample; } else if (sample > audioInfo.sampleMax) { audioInfo.sampleMax = sample; } toReturn[channel][index] = sample; } index++; } return toReturn; } But I need to show the audio's pitch, not amplitude. Fast Fourier transform appears to get the pitch, but it needs to know more variables than the raw bytes I have, and is very complex and mathematical. Is there a way I can do this?

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  • file.createNewFile() creates files with last-modified time before actual creation time

    - by Kaleb Pederson
    I'm using JPoller to detect changes to files in a specific directory, but it's missing files because they end up with a timestamp earlier than their actual creation time. Here's how I test: public static void main(String [] files) { for (String file : files) { File f = new File(file); if (f.exists()) { System.err.println(file + " exists"); continue; } try { // find out the current time, I would hope to assume that the last-modified // time on the file will definitely be later than this System.out.println("-----------------------------------------"); long time = System.currentTimeMillis(); // create the file System.out.println("Creating " + file + " at " + time); f.createNewFile(); // let's see what the timestamp actually is (I've only seen it <time) System.out.println(file + " was last modified at: " + f.lastModified()); // well, ok, what if I explicitly set it to time? f.setLastModified(time); System.out.println("Updated modified time on " + file + " to " + time + " with actual " + f.lastModified()); } catch (IOException e) { System.err.println("Unable to create file"); } } } And here's what I get for output: ----------------------------------------- Creating test.7 at 1272324597956 test.7 was last modified at: 1272324597000 Updated modified time on test.7 to 1272324597956 with actual 1272324597000 ----------------------------------------- Creating test.8 at 1272324597957 test.8 was last modified at: 1272324597000 Updated modified time on test.8 to 1272324597957 with actual 1272324597000 ----------------------------------------- Creating test.9 at 1272324597957 test.9 was last modified at: 1272324597000 Updated modified time on test.9 to 1272324597957 with actual 1272324597000 The result is a race condition: JPoller records time of last check as xyz...123 File created at xyz...456 File last-modified timestamp actually reads xyz...000 JPoller looks for new/updated files with timestamp greater than xyz...123 JPoller ignores newly added file because xyz...000 is less than xyz...123 I pull my hair out for a while I tried digging into the code but both lastModified() and createNewFile() eventually resolve to native calls so I'm left with little information. For test.9, I lose 957 milliseconds. What kind of accuracy can I expect? Are my results going to vary by operating system or file system? Suggested workarounds? NOTE: I'm currently running Linux with an XFS filesystem. I wrote a quick program in C and the stat system call shows st_mtime as truncate(xyz...000/1000).

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  • apt-get install fuse - MAKEDEV not installed, skipping device node creation

    - by holms
    This happened with command apt-get dist-upgrade to upgrade to debian jessie, after which I've tried to remove fuse, and install it again. Same error: root@msgapp:/dev# apt-get install fuse Reading package lists... Done Building dependency tree Reading state information... Done The following NEW packages will be installed: fuse 0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded. Need to get 0 B/69.9 kB of archives. After this operation, 191 kB of additional disk space will be used. Selecting previously unselected package fuse. (Reading database ... 39354 files and directories currently installed.) Preparing to unpack .../fuse_2.9.3-10_amd64.deb ... Unpacking fuse (2.9.3-10) ... Processing triggers for man-db (2.6.7.1-1) ... Setting up fuse (2.9.3-10) ... MAKEDEV not installed, skipping device node creation. device node not found dpkg: error processing package fuse (--configure): subprocess installed post-installation script returned error exit status 2 Errors were encountered while processing: fuse E: Sub-process /usr/bin/dpkg returned an error code (1) UPDATE Reinstalling makedev gives another problem: root@msgapp:/dev# apt-get install makedev Reading package lists... Done Building dependency tree Reading state information... Done The following NEW packages will be installed: makedev 0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded. Need to get 0 B/42.6 kB of archives. After this operation, 129 kB of additional disk space will be used. Selecting previously unselected package makedev. (Reading database ... 39347 files and directories currently installed.) Preparing to unpack .../makedev_2.3.1-93_all.deb ... Unpacking makedev (2.3.1-93) ... Processing triggers for man-db (2.6.7.1-1) ... ySetting up makedev (2.3.1-93) ... /run/udev or .udevdb or .udev presence implies active udev. Aborting MAKEDEV invocation. /run/udev or .udevdb or .udev presence implies active udev. Aborting MAKEDEV invocation. /run/udev or .udevdb or .udev presence implies active udev. Aborting MAKEDEV invocation. There's ticket raised, and their fix doesn't give any result: root@msgapp:/dev# cd /dev && ./MAKEDEV fuse /run/udev or .udevdb or .udev presence implies active udev. Aborting MAKEDEV invocation.

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  • strange MPMoviePlayer problem

    - by Rahul Vyas
    I am using mpmovieplayer in my application.I have a button for playing movie witch has an image of play.But when mpmovieplayer plays movie i can see that button on movie player's overlying controlls like when i pause movie i see my play button instead of movie player's default button.Also i have customized navigation bar and i can see that navbar when movie plays instead of default nav bar.I tried hiding button when playing movie but it didn't worked.Does Someone knows about this issue? also i am having cropping of video issues does someone knows about how to handle video orientation i mean i want full video in any orientation recorded video.Thanks

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  • Looping music with intro in XNA using SoundEffect

    - by Jordan Roher
    I have two sound files: Sound A is an 18 second intro designed to be played once Sound B is a 1 minute looping track I'd like to play Sound A once, then once Sound A is done, immediately play Sound B and keep looping Sound B until I tell it to stop. This is supposed to be looping town music in an RPG. I've tried doing this in code using just SoundEffect, but there's a tiny yet noticeable gap between the end of Sound A and the beginning of Sound B. Even if I put monitoring code watching Sound A's SoundEffectInstance.State in the Update() function, I haven't been able to start Sound B exactly when Sound A finishes so that it's seamless. I'd prefer to use SoundEffect because I can load WMA files rather than being stuck with WAVs in XACT.

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  • Google App Engine - SiteMap Creation for a social network

    - by spidee
    Hi all. I am creating a social tool - I want to allow search engines to pick up "public" user profiles - like twitter and face-book. I have seen all the protocol info at http://www.sitemaps.org and i understand this and how to build such a file - along with an index if i exceed the 50K limit. Where i am struggling is the concept of how i make this run. The site map for my general site pages is simple i can use a tool to create the file - or a script - host the file - submit the file and done. What i then need is a script that will create the site-maps of user profiles. I assume this would be something like: <?xml version="1.0" encoding="UTF-8"?> <urlset xmlns="http://www.sitemaps.org/schemas/sitemap/0.9"> <url> <loc>http://www.socialsite.com/profile/spidee</loc> <lastmod>2010-5-12</lastmod> <changefreq>???</changefreq> <priority>???</priority> </url> <url> <loc>http://www.socialsite.com/profile/webbsterisback</loc> <lastmod>2010-5-12</lastmod> <changefreq>???</changefreq> <priority>???</priority> </url> </urlset> Ive added some ??? as i don't know how i should set these settings for my profiles based on the following:- When a new profile is created it must be added to a site-map. If the profile is changed or if "certain" properties are changed - then i don't know if i update the entry in the map - or do something else? (updating would be a nightmare!) Some users may change their profile. In terms of relevance to the search engine the only way a google or yahoo search will find the users (for my requirement) profile would be for example by means of [user name] and [location] so once the entry for the profile has been added to the map file the only reason to have the search-bot re-index the profile would be if the user changed their user-name - which they cant. or their location - and or set their settings so that their profile would be "hidden" from search engines. I assume my map creation will need to be dynamic. From what i have said above i would imagine that creating a new profile and possible editing certain properties could mark it as needing adding/updating in the sitemap. Assuming i will have millions of profiles added/being edited how can i manage this in a sensible manner. i know i need a script that can append urls as each profile is created i know the script will prob be a TASK - running at a set freq - perhaps the profiles have a property like "indexed" and the TASK sets them to "true" when the profiles are added to the map. I dont see the best way to store the map - do i store it in the datastore i.e; model=sitemaps properties key_name=sitemap_xml_1 (and for my map sitemap_index_xml) mapxml=blobstore (the raw xml map or ror map) full=boolean (set true when url count is 50) # might need this as a shard will tell us To make this work my thoughts are m cache the current site map structure as "sitemap_xml" keep a shard of url count when my task executes 1. build the xml structure for say the first 100 urls marked "index==false" (how many could u run at a time?) 2. test if the current mcache sitemap is full (shardcounter+10050K) 3.a if the map is near full create a new map entry in models "sitemap_xml_2" - update the map_index file (also stored in my model as "sitemap_index" start a new shard - or reset.2 3.b if the map is not full grab it from mcache 4.append the 100 url xml structure 5.save / m cache the map I can now add a handler using a url map/route like /sitemaps/* Get my * as map name and serve the maps from the blobstore/mache on the fly. Now my question is does this work - is this the right way or a good way to start? Will this handle the situation of making sure the search bots update when a user changes their profile - possibly by setting the change freq correctly? - Do i need a more advance system :( ? or have i re-invented the wheel! I hope this is all clear and make some form of sense :-)

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  • SoundManager2 has irregular latency

    - by Stefan Monov
    I'm playing some notes at regular intervals. Each one is delayed by a random number of milliseconds, creating a jarring irregular effect. How do I fix it? Note: I'm OK with some latency, just as long as it's consistent. Answers of the type "implement your own small SoundManager2 replacement, optimized for timing-sensitive playback" are OK, if you know how to do that :) but I'm trying to avoid rewriting my whole app in flash for now. For an example of app with zero audible latency see the flash-based ToneMatrix. Testcase (see it here live or get it in an zip): <head> <title></title> <script type="text/javascript" src="http://www.schillmania.com/projects/soundmanager2/script/soundmanager2.js"> </script> <script type="text/javascript"> soundManager.url = '.' soundManager.flashVersion = 9 soundManager.useHighPerformance = true soundManager.useFastPolling = true soundManager.autoLoad = true function recur(func, delay) { window.setTimeout(function() { recur(func, delay); func(); }, delay) } soundManager.onload = function() { var sound = soundManager.createSound("test", "test.mp3") recur(function() { sound.play() }, 300) } </script> </head> <body> </body> </html>

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  • Flash As3 Mute Button problems

    - by Lee
    Hey guys, I am trying to create a UI movie clip that can be used across different scenes. It uses variables from the root scope to determine states. When i press the mute button is works fine, however when i try to un-mute things go weird. Sometimes it takes 2 clicks to unmute, sometimes more. It seems random. Muting however seems to work first time.. Any ideas? Main Timeline: var mute:Boolean = false; var playerName = "Fred"; function setMute(vol) { var sTransform:SoundTransform = new SoundTransform(1,0); sTransform.volume = vol; SoundMixer.soundTransform = sTransform; } function toggleMuteBtn(event:Event) { if (mute) { // Sound On, Mute Off mute = false; setMute(1); ui_mc.muteCross_mc.visible = false; } else { // Sound Off, Mute On mute = true; setMute(0); ui_mc.muteCross_mc.visible = true; } } ui_mc Action Script: if (MovieClip(parent).mute == false) { muteCross_mc.visible = false; } mute_btn.addEventListener(MouseEvent.CLICK, MovieClip(parent).toggleMuteBtn);

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  • OpenGL programming vs Blender Software, which is better for custom video creation?

    - by iammilind
    I am learning OpenGL API bit by bit and also develop my own C++ framework library for effectively using them. Recently came across Blender software which is used for graphics creation and is in turn written in OpenGL itself. For my part time hobby of graphics learning, I want to just create small-small movie or video segments; e.g. related to construction engineering, epic stories and so on. There may be very minimal to nil mouse-keyboard interaction for those videos, unlike video games which are highly interactive. I was wondering if learning OpenGL from scratch is worth for it or should I invest my time in learning Blender software? There are quite a few good movie examples are created using Blender and are shown in its website. Other such opensource cross platform alternatives are also welcome, which can serve my aforementioned purpose.

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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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  • Windows XP-64 loses audio sounds, drive letters... why?

    - by Ira Baxter
    Until sometime in early December, I had a wonderfully functioning XP-64 system. It was configured to auto download/install MS patches. I occassionally update the software on it, e.g. Open Office, Adobe Reader, Skype, but I don't fetch hundreds of tools or anything much beyond what I just mentioned. In December, suddenly my audio stopped, and drive letters assigned to various mount points on other machines quit being available. Apparantly, the services that support these (and some others) are now not starting up when I boot/login. There isn't anything obvious in the event log. If I manually restart the associated services, these facilities come back on line and work for awhile (a day) but pretty soon the problem reappears. I don't reboot very often, nor do I log out out much. Hints?

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  • Recording Audio from WMP Stream

    - by Jonathan Sampson
    I'm sitting here listening to a radio show that is being broadcast live over an internet stream, but I would like to keep bits and pieces for later-enjoyment. Is there a way I can easily record streams in real-time? I should note also (not sure if it's necessary or not) that this stream requires me to first login before listening.

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  • Quickly switch Win7 volume normalization on/off?

    - by romkyns
    Is there some way to quickly toggle the state of volume normalization in Windows 7? When it's off watching movies late is tricky, and when it's on it messes with music in a bad way. It's a great feature, but argh, it requires me to make my way through so many dialogs... Any solution that requires no more than a couple of clicks or keystrokes is welcome - shortcuts, AutoHotkey, tray icon apps.

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  • Audio card with built-in ground isolator?

    - by Dave Jarvis
    What audio cards would you recommend that eliminate hum, and hard-drive & mouse movement signal interference? Hardware components: Motherboard. Asus P5Q SE Audio. Realtek ALC 1200, 8-Channel High-Definition Audio CODEC (on board) Harddrive. WD Caviar 320 GB Mouse. Logitech Marbleman USB Mixer. Mackie d.4 Pro Amplifier. Sonance Sonamp 260 All components are plugged into the same Monster Power HDP 910 powerbar (does not help eliminate noise). I have no other components plugged in. The computer uses a Monster iCable 1000 to go from mini (on board audio) to RCA (mixer). I have moved the cable as far from other cables as possible. A ground loop isolator between the mixer and on board audio eliminates all noise. I would rather not use a ground loop isolator; an internal audio card that is Linux-compatible (Kubuntu) would be ideal. Suggestions?

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • converting huge MPEG audio files to something smaller

    - by john
    I've got some large MPEG audio files (144 MB each) that I'm looking to convert to something smaller so I can send them out as attachments to an email. Any suggestions on the software to use? I'm looking for something free that will run on Windows. I don't really care what the destination file is, mp3 would be nice. If there's a web service out there that would do this without the need to download any software to my machine, that would be even better, but I would be more than happy just getting it done any way I can. Thanks!

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  • [Windows Live Messenger] Beta sounds

    - by sinni800
    Hello, in a beta version of the Windows Live Messenger they had different sounds once. They weren't like the current ones, they sounded brighter. The normal "dling" when logging in was replaced by a more direct "DIING!". It was only like that in one beta version thought. I was already searching for it when it was replaced back again, but I lost the exe file back then. Anyone know of this? Anyone else? Please!

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