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  • Handling & processing credit card payments

    - by Bob Jansen
    I'm working on program that charges customers on a pay as you go per month modal. This means that instead of the customers paying their invoices at the start of the month, they will have to pay at the end of the month. In order to secure the payments I want my customers credit card information stored so that they can be charged automatically at the end of the month. I do not have the resources, time, or risk to handle and store my customers credit card information on my servers and am looking for a third party solution. I'm a tad overwhelmed by all the different options and services that are out there and was wondering if anyone with experience have any recommendations and tips. I'm having difficulty finding services that allow me to to store my customers credit card information and charge them automatically. Most of them seem to offer an invoice styled approach.

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  • Origin of common list-processing function names

    - by Heatsink
    Some higher-order functions for operating on lists or arrays have been repeatedly adopted or reinvented. The functions map, fold[l|r], and filter are found together in several programming languages, such as Scheme, ML, and Python, that don't seem to have a common ancestor. I'm going with these three names to keep the question focused. To show that the names are not universal, here is a sampling of names for equivalent functionality in other languages. C++ has transform instead of map and remove_if instead of filter (reversing the meaning of the predicate). Lisp has mapcar instead of map, remove-if-not instead of filter, and reduce instead of fold (Some modern Lisp variants have map but this appears to be a derived form.) C# uses Select instead of map and Where instead of filter. C#'s names came from SQL via LINQ, and despite the name changes, their functionality was influenced by Haskell, which was itself influenced by ML. The names map, fold, and filter are widespread, but not universal. This suggests that they were borrowed from an influential source into other contemporary languages. Where did these function names come from?

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  • Optimal Compression for Speech

    - by ashes999
    I'm designing a game that depends heavily on audio; I will have some 300+ speech files (most of them just a word or two long). This can very quickly escalate the size of my final game. What's the optimal way to encode/compress speech files to keep the size minimal without getting audio artifacts? Please address both per-file compression/encoding, and also zipping/compressing the set of all speech files together in your answer. Because I'm not sure which (or combination of both) factors will give me the best results. Edit: I need this to run in Silverlight and Android, so I'm presumably stuck with only MP3 as my option (other than uncompressed wave files).

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  • Checking rtp stream audio quality.

    - by chills42
    We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through wireshark, but we would like to find a way to save the audio to a .wav file (or similar) via the command line. Does anyone know of a tool that can do this?

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  • RPG Item processing

    - by f00b4r
    I started working on an item system for my (first) game, and I'm having a problem conceptualizing how it should work. Since Items can produce a bunch of potentially non-standard actions (revive a character vs increasing some stat) or have use restrictions (can only revive if a character is dead). For obvious reasons, I don't want to create a new Item class for every item type. What is the best way to handle this? Should I make a handful of item types (field modifiers, status modifiers, )? Is it normal to script item usage? Could (should?) this be combined with the above mentioned solution (have a couple of different sub item types, make special case items usage scripted)? Thanks.

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  • Processing Text and Binary (Blob, ArrayBuffer, ArrayBufferView) Payload in WebSocket - (TOTD #185)

    - by arungupta
    The WebSocket API defines different send(xxx) methods that can be used to send text and binary data. This Tip Of The Day (TOTD) will show how to send and receive text and binary data using WebSocket. TOTD #183 explains how to get started with a WebSocket endpoint using GlassFish 4. A simple endpoint from that blog looks like: @WebSocketEndpoint("/endpoint") public class MyEndpoint { public void receiveTextMessage(String message) { . . . } } A message with the first parameter of the type String is invoked when a text payload is received. The payload of the incoming WebSocket frame is mapped to this first parameter. An optional second parameter, Session, can be specified to map to the "other end" of this conversation. For example: public void receiveTextMessage(String message, Session session) {     . . . } The return type is void and that means no response is returned to the client that invoked this endpoint. A response may be returned to the client in two different ways. First, set the return type to the expected type, such as: public String receiveTextMessage(String message) { String response = . . . . . . return response; } In this case a text payload is returned back to the invoking endpoint. The second way to send a response back is to use the mapped session to send response using one of the sendXXX methods in Session, when and if needed. public void receiveTextMessage(String message, Session session) {     . . .     RemoteEndpoint remote = session.getRemote();     remote.sendString(...);     . . .     remote.sendString(...);    . . .    remote.sendString(...); } This shows how duplex and asynchronous communication between the two endpoints can be achieved. This can be used to define different message exchange patterns between the client and server. The WebSocket client can send the message as: websocket.send(myTextField.value); where myTextField is a text field in the web page. Binary payload in the incoming WebSocket frame can be received if ByteBuffer is used as the first parameter of the method signature. The endpoint method signature in that case would look like: public void receiveBinaryMessage(ByteBuffer message) {     . . . } From the client side, the binary data can be sent using Blob, ArrayBuffer, and ArrayBufferView. Blob is a just raw data and the actual interpretation is left to the application. ArrayBuffer and ArrayBufferView are defined in the TypedArray specification and are designed to send binary data using WebSocket. In short, ArrayBuffer is a fixed-length binary buffer with no format and no mechanism for accessing its contents. These buffers are manipulated using one of the views defined by one of the subclasses of ArrayBufferView listed below: Int8Array (signed 8-bit integer or char) Uint8Array (unsigned 8-bit integer or unsigned char) Int16Array (signed 16-bit integer or short) Uint16Array (unsigned 16-bit integer or unsigned short) Int32Array (signed 32-bit integer or int) Uint32Array (unsigned 16-bit integer or unsigned int) Float32Array (signed 32-bit float or float) Float64Array (signed 64-bit float or double) WebSocket can send binary data using ArrayBuffer with a view defined by a subclass of ArrayBufferView or a subclass of ArrayBufferView itself. The WebSocket client can send the message using Blob as: blob = new Blob([myField2.value]);websocket.send(blob); where myField2 is a text field in the web page. The WebSocket client can send the message using ArrayBuffer as: var buffer = new ArrayBuffer(10);var bytes = new Uint8Array(buffer);for (var i=0; i<bytes.length; i++) { bytes[i] = i;}websocket.send(buffer); A concrete implementation of receiving the binary message may look like: @WebSocketMessagepublic void echoBinary(ByteBuffer data, Session session) throws IOException {    System.out.println("echoBinary: " + data);    for (byte b : data.array()) {        System.out.print(b);    }    session.getRemote().sendBytes(data);} This method is just printing the binary data for verification but you may actually be storing it in a database or converting to an image or something more meaningful. Be aware of TYRUS-51 if you are trying to send binary data from server to client using method return type. Here are some references for you: JSR 356: Java API for WebSocket - Specification (Early Draft) and Implementation (already integrated in GlassFish 4 promoted builds) TOTD #183 - Getting Started with WebSocket in GlassFish TOTD #184 - Logging WebSocket Frames using Chrome Developer Tools, Net-internals and Wireshark Subsequent blogs will discuss the following topics (not necessary in that order) ... Error handling Custom payloads using encoder/decoder Interface-driven WebSocket endpoint Java client API Client and Server configuration Security Subprotocols Extensions Other topics from the API

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  • A seekable one-frame FLV video (with audio)?

    - by George Stephanos
    Is it possible to generate an FLV out of an MP3 and a JPG, without uselessly looping the image and still be able to seek the audio ? This command generates a non-seekable video: ffmpeg -y -i audio.mp3 -i image.jpg -r 1 -acodec copy video.flv and this one generates a seekable one, but with uselessly looping the image occupying both space and time: ffmpeg -y -loop_input -i audio.mp3 -i image.jpg -r 1 -acodec copy video.flv -shortest

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  • "HDA audio bus driver is required and not found" on Dell Optiplex

    - by user1666698
    I have Dell Optiplex 745 with Windows 7 installed on it. I'm trying to use the Windows XP audio driver as Windows 7 drivers aren't available for Optiplex 745 and Windows Vista driver is displaying that it's not compatible with my hardware. When I try to install the Windows XP audio driver, it's displaying an error HDA audio bus driver is required and not found The installation fails then. I have researched thourghly and used many drivers but my audio is not working at all. I was also told that it might be a problem with my hardware – that is, a problem with the board.

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  • Legal concerns with orchestrating a music submission contest

    - by Amplify91
    My team and I are getting pretty far along in the development of our latest game and have been thinking about audio. We decided to host an audio submission contest where we will offer a little cash and some equity stake in the game as prizes. We are also giving away copies of the game to participants. We hope not only to find audio for our game, but to meet some cool sound artists and promote the game a bit through the process. First of all, is this even a good idea? What are some potential dangers in doing this? Will it even be well received among artists? Secondly, I wrote up some Terms and Conditions in my best legal-speak to try to protect us and clarify how the contest will be run. Are these sufficient to make sure everyone involved is treated fairly and is legally protected? They are as follows: All submissions (The Submission) must be licensed under a Creative Commons Attribution 3.0 Unported License (CC-BY-3.0) By applying a CC-BY-3.0 license, you (The Submitter) expressly give Detour Games (and all members wherein) permission to copy, distribute, transmit, modify, adapt, and make commercial use of The Submission. The Submitter must own all rights to The Submission and be within their rights to license it as specified and submit it. The Submitter claims responsibility for the legality of The Submission. If The Submission is found to infringe on the rights of a person or entity other than those of The Submitter, Detour Games will not be held liable as all responsibility and liability for the legality of The Submission is that of The Submitter's. No more than two free copies of The Game per submitter. All flat cash prizes will only be disbursed pending the success of our first $5,000 Kickstarter campaign. These prizes will be disbursed 30 days after Detour Games receives the Kickstarter funds. All equity prizes (percentage of profits) are defined as the given percent of total profits after costs for a period of one year (12 months) after the release of RAW. These prizes will be disbursed semi-annually. All prize money will be disbursed through either an electronic fund transfer through a service such as PayPal or by a mailed money order. It is The Submitter's responsibility to cooperate with Detour Games in the disbursement of the funds. Detour Games reserves the right to change these Terms and Conditions at any time without notice. By participating in the contest, The Submitter agrees to and accepts all terms and conditions listed. What else could I do (legally) to protect everyone involved?

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  • Playing/extracting audio file from PDF

    - by ravl1084
    I use Ubuntu and I have a PDF file that contains an audio annotation. It won't play on Okular, it treats it as a text annotation. Following an old blog post where the poster created a small C script to extract the audio didn't work either, I suspect the format of these audio annotations has changed. Using the information on it I managed to uncompress the PDF and with vim, I found the audio data in the file. I tried copying this into its own file and changed the extension from mp3, wav, mid, but none of them would play. Is there a way of achieving this?

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  • Processing a list of atomic operations, allowing for interruptions

    - by JDB
    I'm looking for a design pattern that addresses the following situation: There exists a list of tasks that must be processed. Tasks may be added at any time. Each task is wholly independent from all other tasks. The order in which tasks are processed has no effect on the overall system or on the tasks themselves. Every task must be processed once and only once. The "main" process which launches the task processors may start and stop without warning. When stopped, the "main" process loses all in-memory data. Obviously this is going to involve some state, but are there any design patterns which discuss where and how to maintain that state? Are there any relevant anti-patterns? Named patterns are especially helpful so that we can discuss this topic with other organizations without having to describe the entire problem domain.

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  • Why is Prolog associated with Natural Language Processing?

    - by kyphos
    I have recently started learning about NLP with python, and NLP seems to be based mostly on statistics/machine-learning. What does a logic programming language bring to the table with respect to NLP? Is the declarative nature of prolog used to define grammars? Is it used to define associations between words? That is, somehow mine logical relationships between words (this I imagine would be pretty hard to do)? Any examples of what prolog uniquely brings to NLP would be highly appreciated.

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  • ffdshow h.264 audio desync

    - by Core Xii
    When I encode video with ffdshow with h.264, the audio is out of sync. At the very beginning of the video, the picture freezes for about 1 second, while the audio plays fine, resulting in the audio being that 1 second ahead of the picture throughout the entire video. Any ideas on possible causes or, obviously, solutions?

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  • Why won't AVI2DVD load the audio stream?

    - by Xavierjazz
    XP SP3 I have an .avi file. It is in a folder on my "C:" drive. There are no disallowed characters in either the folder or file name. It has audio as I have watched it on my computer. I want to burn it to a DVD. When I load the file into AVI2DVD, no audio stream shows, and the program will not work without an audio stream. I have used the net extensively to try and solve this, with no success. AFICT I have followed all instructions exactly, but no audio stream. Very frustrating. Does anyone have a clue? Can you help me? Thank you. Regards,

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  • Microphone audio streaming from Cocoa mac app to iPhone

    - by Benzamin
    Hi devs, I'm trying to build a microphone audio streamer to iPhone. The server software will be a mac desktop app and the client will be iPhone, and they are connected via tcp port. I've successfully connected the mac app and iPhone, and tried to send a fixed test.m4a audio file first. But at the iPhone i grabbed the data well, when tried to play it i used AVAudioPlayer and its returning OSStatus error. I played around with the audio queue service but its very tricky and i only got some example for fixed length audio playing like http://cocoawithlove.com/2009/06/revisiting-old-post-streaming-and.html Now i need help on two things, how can i continuously grab audio data from Mac desktop microphone? And then after grabbing the data how i can play this unfixed length audio data in the iPhone. What exactly i need to do? Please please help me on this......

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  • Get binary data from audio impulses

    - by Timo
    I have IR sensor which have TRS plug and I can record my remotes signals into audio. Now I want to control my computer with TV remote, but I don't have any clue how to compare audio input with pre-recorded audio. But after I realized that these audio waves contains only some kind data (binary) I can turn these into binary or hex, so it is much easier to compare. Waves look just like this: http://i.imgur.com/lCIyl.png And this: ttp://i.imgur.com/goJ6d.png These are records of "OK" button, sometimes there are some impulses on right channel too and I don't know why, it seems like connections in sensor are damaged maybe. Ok thats not matter, anyway I need help with python program which read these impulses and turn these into binary, in realtime from audio input(mic). I know it's sounds like "Do it for me, while I enjoy my life", but I don't have experiences with sound transforming/reading... I've looking for python examples for recording and reading audio, but unsuccessfully.

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  • ffmpeg: video file played OK on Ubuntu, but no sound on XP

    - by Andy Le
    I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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  • Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

    - by davr
    First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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  • What is the best way to get an audio file duration in Android?

    - by Gilead
    Hi! I'm using a SoundPool [ 1 ] to play audio clips in my app. All is fine but I need to know when the clip playback has finished. At the moment I track it in my app by obtaining the duration of each clip using a MediaPlayer [ 2 ] instance. That works fine but it looks wasteful to load each file twice, just to get the duration. I could roughly calculate the duration myself knowing the length of the file (available from the AssetFileDescriptor [ 3 ]) but I'd still need to know the sample rate and the number of channels. I see two potential solutions to that problem: Figuring out when a clip has finished playing (doesn't seem to be possible with SoundClip). Having a class which could load just the header of an audio file and give me the sample rate/number of channels (and, ideally, the sample count to get the exact duration). Any suggestions? Thanks, Max The code I'm using at the moment (works fine but is rather heavy for the purpose): String[] fileNames = ... MediaPlayer mp = new MediaPlayer(); for (String fileName : fileNames) { AssetFileDescriptor d = context.getAssets().openFd(fileName); mp.reset(); mp.setDataSource(d.getFileDescriptor(), d.getStartOffset(), d.getLength()); mp.prepare(); int duration = mp.getDuration(); // ... } On a side note, this question has already been asked [ 4 ] but got no answers.

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  • Issues regarding playing audio files in a JME midlet.

    - by Northernen
    I am making a midlet which is to be used to play out local audio files. It is obviously not working. I am getting a null reference on the "is" variable, in the code snippet shown below. 1. try{ 2. System.out.println("path: " + this.getClass()); 3. InputStream is = this.getClass().getResourceAsStream("res/01Track.wav"); 4. p1=Manager.createPlayer(is, "audio"); 5. p1.realize(); 6. p1.prefetch(); 7. p1.start(); 8. } 9. catch(Exception e){ 10. System.out.println(e.getMessage()); 11. } I assume there is something wrong with the "this.getClass().getResourceAsStream("res/01Track.wav")" bit, but I can not for the life of me figure out why, and I have tried referring to the file in 20 different ways. If I printline "this.getClass()" it gives me "path: class Mp3spiller". The absolute path to "01Track.wav" is "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller\res\01Track.wav". Am I completely wrong in thinking that I should refer relatively to "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller"? If anyone could point out what I am doing wrong, I would be grateful. I have basically stolen the code from a tutorial I found online, so I would have thought it would be working.

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  • No audio flash movies

    - by Valerio Giuffrida
    I've installed natty a few days ago and I'm experience some issues with the audio. Basically, I'm not able to listen anything coming from flash movies (such as youtube). This happen with both chromium and firefox (I use only the first one) and the errore I get in the stdout is: ALSA lib pcm.c:2109:(snd_pcm_open_conf) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_pulse.so I don't know how to fix it. I found somewhere to install native x64 flash plugins, but somebody says it is not advisable. I didn't get this kind of issues with 10.10 I've gotten before of natty. Hence, what am I supposed to do? thank you very much

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  • Skipping video and audio with PS3MediaServer

    - by MaxMackie
    I'm using the latest PS3MediaServer build right from the repos suggested in the Ubuntu Wiki. I'm streaming multiple movies from my server (Ubuntu 10.04 LTS) to my PS3 over wireless. Sometimes, during some movies, the audio and the video will begin skipping. This can last anywhere between 5 and 30 seconds before it goes back to normal. I have a four core i5 processor and 8GB of DDR3 RAM so I don't think my computer is having a hard time keeping up with the transcoding. So this leads me to believe it's either sub-optimal transcoding options from within PS3MS or my network can't handle the heat. Other than the out-of-box configuration, is there any way I can tweak the settings for the application to use my resources more efficiently?

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