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  • Problems with utf-8 encoding in php

    - by Addsy
    Hi, Another utf-8 related problem I believe... I am using php to update data in a mysql db then display that data elsewhere in the site. Previously I have run into utf-8 problems before where special characters are displayed as question marks when viewed in a browser but this one seems slightly different. I have a number of records to enter that contain the è character. If I enter this directly in the db then it appears correctly on the page so I take this to mean that utf-8 content is being output correctly. However when I try and update the values in the db through php, then the è character is replaced. What appears instead is & Atilde ; & uml ; (without the spaces) which appears in the browser as è I have the tables in the database set to use UTF-8. I believe this is correct cos, as mentioned, if I update the db through phpMyAdmin, its all ok. Similarly I have set the character encoding for the page which seems to be correct. I am also running the sql statement "SET NAMES 'utf8';" before trying to update the db. Anyone have any other ideas as to where the problem may lie? Many thanks

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  • Encoding/Decoding hex packet

    - by Roberto Pulvirenti
    I want to send this hex packet: 00 38 60 dc 00 00 04 33 30 3c 00 00 00 20 63 62 39 62 33 61 36 37 34 64 31 36 66 32 31 39 30 64 30 34 30 63 30 39 32 66 34 66 38 38 32 62 00 06 35 2e 31 33 2e 31 00 00 02 3c so i build the string: string packet = "003860dc0000" + textbox1.text+ "00000020" + textbox2.text+ "0006" + textbox3.text; then "convert" it to ascii: conn_str = HexString2Ascii(packet); then i send the packet... but i have this: 00 38 60 **c3 9c** 00 00 04 33 30 3c 00 00 00 20 63 62 39 62 33 61 36 37 34 64 31 36 66 32 31 39 30 64 30 34 30 63 30 39 32 66 34 66 38 38 32 62 00 06 35 2e 31 33 2e 31 00 00 02 3c **0a** why?? Thank you! P.S. the function is: private string HexString2Ascii(string hexString) { byte[] tmp; int j = 0; int lenght; lenght=hexString.Length-2; tmp = new byte[(hexString.Length)/2]; for (int i = 0; i <= lenght; i += 2) { tmp[j] =(byte)Convert.ToChar(Int32.Parse(hexString.Substring(i, 2), System.Globalization.NumberStyles.HexNumber)); j++; } return Encoding.GetEncoding(1252).GetString(tmp); }

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  • Silverlight - WCF Enable Binary Encoding

    - by Villager
    Hello, I have a WCF service that is returning a lot of data. I want to compress that information so I thought that using BinaryEncoding would be appropriate. Currently, I have a binding setup in my web.config as follows: <binding name="myCustomBinding" closeTimeout="00:05:00" openTimeout="00:05:00" receiveTimeout="00:05:00" sendTimeout="00:05:00"> <binaryMessageEncoding /> <httpTransport maxReceivedMessageSize="8388608" maxBufferSize="8388608"> <extendedProtectionPolicy policyEnforcement="Never" /> </httpTransport> </binding> In my ServiceReferences.clientconfig file, I have the following binding settings: <binding name="CustomBinding_MyService"> <httpTransport maxReceivedMessageSize="2147483647" maxBufferSize="2147483647"> <extendedProtectionPolicy policyEnforcement="Never" /> </httpTransport> </binding> Oddly, this configuration will not work. As soon as I remove the <binaryMessageEncoding /> line from the web.config, everything works fine. My question is, how do I use binary message encoding? Is there something I need to configure in my ServiceReferences.clientconfig? Thank you

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  • Unicode version of base64 encoding/ decoding

    - by Yan Cheng CHEOK
    I am using base64 encoding/decoding from http://www.adp-gmbh.ch/cpp/common/base64.html It works pretty well with the following code. const std::string s = "I Am A Big Fat Cat" ; std::string encoded = base64_encode(reinterpret_cast<const unsigned char*>(s.c_str()), s.length()); std::string decoded = base64_decode(encoded); std::cout << _T("encoded: ") << encoded << std::endl; std::cout << _T("decoded: ") << decoded << std::endl; However, when comes to unicode namespace std { #ifdef _UNICODE typedef wstring tstring; #else typedef string tstring; #endif } const std::tstring s = _T("I Am A Big Fat Cat"); How can I still make use of the above function? Merely changing std::string base64_encode(unsigned TCHAR const* , unsigned int len); std::tstring base64_decode(std::string const& s); will not work correctly. (I expect base64_encode to return ASCII. Hence, std::string should be used instead of std::tstring)

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  • Problem with waveOutWrite and waveOutGetPosition deadlock

    - by MusiGenesis
    I'm working on an app that plays audio continuously using the waveOut... API from winmm.dll. The app uses "leapfrog" buffers, which are basically a bunch of arrays of samples that you dump into the audio queue. Windows plays them seamlessly in sequence, and as each buffer completes Windows calls a callback function. Inside this function, I load the next set of samples into the buffer, process them however, and then dump the buffer back into the audio queue. In this way, the audio plays indefinitely. For animation purposes, I'm trying to incorporate waveOutGetPosition into the application (since the "buffer done" callbacks are irregular enough to cause jerky animation). waveOutGetPosition returns the current position of playback, so it's hyper-precise. The problem is that in my application, making calls to waveOutGetPosition eventually causes the application to lock up - the sound stops and the call never returns. I've boiled things down to a simple app that demonstrates the problem. You can run the app here: http://www.musigenesis.com/SO/waveOut%20demo.exe If you just hear a tiny bit of piano over and over, it's working. It's just meant to demonstrate the problem. The source code for this project is here: http://www.musigenesis.com/SO/WaveOutDemo.zip The first button runs the app in leapfrog mode without making the calls to waveOutGetPosition. If you click this, the app will play forever without breaking (the X button will close it and shut it off). The second button starts the leapfrogger and also starts a forms timer that calls the waveOutGetPosition and displays the current position. Click this and the app will run for a short while and then lock up. On my laptop, it usually locks up in 15-30 seconds; at most it's taken a minute. I have no idea how to fix this, so any help or suggestions would be most welcome. I've found very few posts on this issue, but it seems that there is a potential deadlock, either from multiple calls to waveOutGetPosition or from calls to that and waveOutWrite that occur at the same time. It's possible that I'm calling this too frequently for the system to handle.

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  • i2s0: transmitter underrun (0)

    - by tbarbe
    were doing some audio stuff and I keep seeing this in the Organizer Console. Sun May 2 20:16:48 unknown kernel[0] : i2s0: transmitter underrun (0) Are these transmitter underruns bad? I think its just when were shutting down audio input...but could a few of these cause some issues later on?

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  • Streaming webcam video in Flash using MP4 encoding

    - by Herms
    One of the features of the Flash app I'm working on is to be able to stream a webcam to others. We're just using the built-in webcam support in Flash and sending it through FMS. We've had some people ask for higher quality video, but we're already using the highest quality setting we can in Flash (setting quality to 100%). My understanding is that in the newer flash players they added support for MPEG-4 encoding for the videos. I created a simple test Flex app to try and compare the video quality of the MP4 vs FLV encodings. However, I can't seem to get MP4 to work at all. According to the Flex documentation the only thing I need to do to use MP4 instead of FLV is prepend "mp4:" to the name of the stream when calling publish: Specify the stream name as a string with the prefix mp4: with or without the filename extension. The prefix indicates to the server that the file contains H.264-encoded video and AAC-encoded audio within the MPEG-4 Part 14 container format. When I try this nothing happens. I don't get any events raised on the client side, no exceptions thrown, and my logging on the server side doesn't show any streams starting. Here's the relevant code: // These are all defined and created within the class. private var nc:NetConnection; private var sharing:Boolean; private var pubStream:NetStream; private var format:String; private var streamName:String; private var camera:Camera; // called when the user clicks the start button private function startSharing():void { if (!nc.connected) { return; } if (sharing) { return; } if(pubStream == null) { pubStream = new NetStream(nc); pubStream.attachCamera(camera); } startPublish(); sharing = true; } private function startPublish():void { var name:String; if (this.format == "mp4") { name = "mp4:" + streamName; } else { name = streamName; } //pubStream.publish(name, "live"); pubStream.publish(name, "record"); }

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  • iPhone SDK: How to record voices with ambient noise supression?

    - by Harkonian
    Can anyone point me in the right direction on how I would minimize ambient noise while recording someone speaking using the iPhone SDK Core Audio? I'm guessing a band-pass filter that eliminates any frequencies above and below the human vocal range might work. I have no idea how I would implement band filters on audio in the SDK though. The optimum solution would be one that eliminates the noise from the stream before it is written to memory/disk. Some sample code would be appreciated.

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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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  • Invalid character in a Base-64 string when Concatenating and Url encoding a string

    - by Rob
    I’m trying to write some encryption code that is passed through a Url. For the sake of the issue I’ve excluded the actual encryption of the data and just shown the code causing the problem. I take a salt value, convert it to a byte array and then convert that to a base64 string. This string I concatenate to another base64 string (which was previously a byte array). These two base64 strings are then Url encoded. Here’s my code... using System; using System.Collections.Generic; using System.Linq; using System.Text; using System.Security.Cryptography; using System.IO; using System.Web; class RHEncryption { private static readonly Encoding ASCII_ENCODING = new System.Text.ASCIIEncoding(); private static readonly string SECRET_KEY = "akey"; private static string md5(string text) { return BitConverter.ToString(new MD5CryptoServiceProvider().ComputeHash(ASCII_ENCODING.GetBytes(text))).Replace("-", "").ToLower(); } public string UrlEncodedData; public RHEncryption() { // encryption object RijndaelManaged aes192 = new RijndaelManaged(); aes192.KeySize = 192; aes192.BlockSize = 192; aes192.Mode = CipherMode.CBC; aes192.Key = ASCII_ENCODING.GetBytes(md5(SECRET_KEY)); aes192.GenerateIV(); // convert Ivector to base64 for sending string base64IV = Convert.ToBase64String(aes192.IV); // salt value string s = "maryhadalittlelamb"; string salt = s.Substring(0, 8); // convert to byte array // and base64 for sending byte[] saltBytes = ASCII_ENCODING.GetBytes(salt.TrimEnd('\0')); string base64Salt = Convert.ToBase64String(saltBytes); //url encode concatenated base64 strings UrlEncodedData = HttpUtility.UrlEncode(base64Salt + base64IV, ASCII_ENCODING); } public string UrlDecodedData() { // decode the url encode string string s = HttpUtility.UrlDecode(UrlEncodedData, ASCII_ENCODING); // convert back from base64 byte[] base64DecodedBytes = null; try { base64DecodedBytes = Convert.FromBase64String(s); } catch (FormatException e) { Console.WriteLine(e.Message.ToString()); Console.ReadLine(); } return s; } } If I then call the UrlDecodedData method I get a “Invalid character in a Base-64 string” exception. This is generated because the base64Salt variable contains an invalid character (I’m guessing a line termination) but I can’t seem to strip it off.

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  • iPhone SDK audioSession question.

    - by Morion
    Hi to all. In my app i record and play audio at the same time. The app is almost finished. But there is one thing, that annoying me. When audio session is set to PlayAndRecord, sounds become quiet in comparison with the same sounds with the SoloAmbient category. Is there any way to make sound louder using PlayAndRecord?

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  • What is a lightweight cross platform WAV playing library?

    - by Lokkju
    I'm looking for a lightweight way to make my program (written in C) be able to play audio files on either windows or linux. I am currently using windows native calls, which is essentially just a single call that is passed a filename. I would like something similar that works on linux. The audio files are Microsoft PCM, Single channel, 22Khz Any Suggestions?

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  • Identifying voice as male or female

    - by duder
    I'm not much into audio engineering, so please be easy on me. I'm receiving an audio file as input, and need to detect whether the speaker is male or female. Any ideas how to go about doing this? I'm using php, but am open to using other languages, and don't mind learning a little bit of sound theory as long as the time is proportionate to the task.

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  • How does the Ableton warp algorithm work exactly?

    - by pepperdreamteam
    I'm looking for any documentation or definitive information on Ableton's warp feature. I understand that it has something to do with finding transients, aligning them with an even rhythm and shifting audio samples accordingly. I'm hoping to find ways to approximate warping with more basic audio editing tools. I understand that this is ableton's unique device, really any information about how it works would be helpful. So...does anyone have any 411?

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  • Any tips for how to build a LED sytem thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

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  • AudioTrack lag: obtainBuffer timed out

    - by BTR
    I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream BufferedInputStream DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this: 03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960, server=00000000 03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28 I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start. I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time. public void run() { // Load file FileInputStream mFileInputStream; try { // mFile is instance of custom file class -- this is correct, // so don't sweat this line mFileInputStream = new FileInputStream(mFile.path()); } catch (FileNotFoundException e) {} BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength); DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream); // Skip header try { if (mDataInputStream.available() > 44) mDataInputStream.skipBytes(44); } catch (IOException e) {} // Initialize device mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, ConfigManager.AUDIO_BUFFER_LENGTH, AudioTrack.MODE_STREAM); mAudioTrack.play(); // Initialize buffer byte[] mByteArray = new byte[mBufferLength]; int mBytesToWrite = 0; int mBytesWritten = 0; // Loop to keep thread running while (mRun) { // This flag is turned on when the user presses "play" while (mPlaying) { try { // Check if data is available if (mDataInputStream.available() > 0) { // Read data from file and write to audio device mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength); mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite); } } catch (IOException e) { } } } } If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

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  • How to set a native object property

    - by theunilife
    ok so im creating a jquery plugin that will allow me to use the new html5 Audio interface and im trying to create an option that is an object that you will be able to set the various listeners but i dont seem to be able to set those options to the listener property of the Audio object.

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  • Any tips for how to build a LED system thet will light up to music?

    - by daniels
    So basically I would like somehow that given an audio file as input (most likely mp3 or I can use some audio engine that will handle other types too) from my computer to control some LED lights so they will be something like an oscilloscope, like the one in winamp. What would I need to be able to do this? I'm interested in building thing up all by myself, coding, hardware, etc.. I'm going with C++ on Windows.

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  • Which browsers handle `Content-Encoding: gzip` and which of them has any special requirements on encodinq quality?

    - by user1049847
    I am creating a "hand made" HTTP 1/0, 1/1 server. I recently integrated zip lib so now I can stream encoded gziped data in and out. I wonder Which major browsers (alive ones - IE6-IE10, Chrome, FF, etc) send Accept-Encoding: deflate, gzip, ... and so can handle Content-Encoding: gzip today? Which of them send any quality expectations? Which of them can send encoded gziped post request and multypart/form data to my server?

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  • Why do my speakers get distorted randomly on Windows 7?

    - by Daniel Fischer
    I have a studio monitor setup. I have 2 KRK 6's and a Focusrite Firewire Pro 24. Every few hours my speakers sound distorted and my solution has been go to sound levels Properties of Saffire Audio Device Advanced Default Format Toggle to 16 bit then back to 24bit. Why does it screw up every few hours? Sometimes one speaker doesn't output too and this same process resets it but that's more rare. Is this a OS issue or Focusrite Driver Issue?

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