Search Results

Search found 6501 results on 261 pages for 'audio conversion'.

Page 22/261 | < Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >

  • Conversion between different template instantiation of the same template

    - by Naveen
    I am trying to write an operator which converts between the differnt types of the same implementation. This is the sample code: template <class T = int> class A { public: A() : m_a(0){} template <class U> operator A<U>() { A<U> u; u.m_a = m_a; return u; } private: int m_a; }; int main(void) { A<int> a; A<double> b = a; return 0; } However, it gives the following error for line u.m_a = m_a;. Error 2 error C2248: 'A::m_a' : cannot access private member declared in class 'A' d:\VC++\Vs8Console\Vs8Console\Vs8Console.cpp 30 Vs8Console I understand the error is because A<U> is a totally different type from A<T>. Is there any simple way of solving this (may be using a friend?) other than providing setter and getter methods? I am using Visual studio 2008 if it matters.

    Read the article

  • prevent conversion of <br/>

    - by Chris
    Hello, I fear this is a dumb question, but I can't seem to find the answer. Pretty sure what that makes me...... I have C# generated HTML (HtmlGenerator), to which I sometimes want to insert a line break at a certain part of a cell's innertext. Here is how that comes out: <TD >There are lots of extra &lt; br /&gt; words here </TD> This then displays the <br/> as a part of my cell text - not good. Am I missing an easy way to have the <br/> preserved and not converted to &lt, etc...? thanks

    Read the article

  • String / DateTime Conversion problem (asp.net vb)

    - by Phil
    I have this code: Dim birthdaystring As String = MonthBirth.SelectedValue.ToString & "/" & DayBirth.SelectedValue.ToString & "/" & YearBirth.SelectedValue.ToString Dim birthday As DateTime = Convert.ToDateTime(birthdaystring) Which produces errors (String was not recognized as a valid DateTime.) The string was "01/31/1963". Any assistance would be appreciated. Thanks.

    Read the article

  • NHibernate Entity code conversion from #C to VB.Net

    - by CoderRoller
    Hello and thanks for your help in advance. I am starting on the NHibernate world and i am experimenting with the NHibernate CookBook recipes, i am trying to set a base entity class for my entities and this is the C# code for this. I would like to know whats the VB.NET version so i can implement it in my sample project. This is the C# code: public abstract class Entity<TId> { public virtual TId Id { get; protected set; } public override bool Equals(object obj) { return Equals(obj as Entity<TId>); } private static bool IsTransient(Entity<TId> obj) { return obj != null && Equals(obj.Id, default(TId)); } private Type GetUnproxiedType() { return GetType(); } public virtual bool Equals(Entity<TId> other) { if (other == null) return false; if (ReferenceEquals(this, other)) return true; if (!IsTransient(this) && !IsTransient(other) && Equals(Id, other.Id)) { var otherType = other.GetUnproxiedType(); var thisType = GetUnproxiedType(); return thisType.IsAssignableFrom(otherType) || otherType.IsAssignableFrom(thisType); } return false; } public override int GetHashCode() { if (Equals(Id, default(TId))) return base.GetHashCode(); return Id.GetHashCode(); } } I tried using an online converter but puts a Nothing reference in place of default(TId) that doesn't seem right to me that's why I request for help: Private Shared Function IsTransient(obj As Entity(Of TId)) As Boolean Return obj IsNot Nothing AndAlso Equals(obj.Id, Nothing) End Function I Would appreciate the insight you may give me on the subject.

    Read the article

  • Conversion of text to unicode strings...

    - by user154301
    I have to process JSON files that looks like this: \u0432\u043b\u0430\u0434\u043e\u043c <b>\u043f\u0443\u0442\u0438\u043c<\/b> \u043d\u0430\u0447 Unfortunately, I'm not sure how this encoding is called. I would like to convert it to .NET Unicode strings. What's the easies way to do it? Thanks in advance!

    Read the article

  • Implicit type conversion in DB/2 inserts?

    - by IronGoofy
    We're using SQL Inserts to insert some data via a script into DB/2 tables, e.g. CREATE TABLE TICKETS (TICKETID VARCHAR(10) NOT NULL); On my home installation, this statement works fine (note that I'm using an integer which is autoatically cast into a VarChar): INSERT INTO TICKETS (TICKETID) VALUES (1); while at my customer's site I get a type error. My question(s): Is this behavior version dependent? (I use a DB2 Express V9.7, while the customer has an Enterprise V9.5) Is there a config option to change the behavior? (I would like my home install to behave as close as possible as the production environment is going to be.)

    Read the article

  • Is Objective C fast enough for DSP/audio programming

    - by morgancodes
    I've been making some progress with audio programming for iPhone. Now I'm doing some performance tuning, trying to see if I can squeeze more out of this little machine. Running Shark, I see that a significant part of my cpu power (16%) is getting eaten up by objc_msgSend. I understand I can speed this up somewhat by storing pointers to functions (IMP) rather than calling them using [object message] notation. But if I'm going to go through all this trouble, I wonder if I might just be better off using C++. Any thoughts on this?

    Read the article

  • C# Audio - How to time stretch (different tempo, same pitch)

    - by heath
    I'm trying to make a winform app in C# (VS2008) that can load an mp3 (other formats would be nice, but mp3 at a minimum) and be able to adjust the playback speed (tempo) without affecting pitch. I really don't need any other audio effects. I tried using DirectShow but that doesn't seem to offer time stretch capabilities. I was able to incorporate irrklang but that does not seem to have the time stretch capability either. So now I've moved on to SoundTouch. That certainly has the capabilities but I'm very unclear on how to implement in C#. After a few days of this, about all I've accomplished is using DLLImport on the SoundTouch DLL and am able to successfully retrieve a version number. At this point, I'm not even sure if I can do what I'm trying to do with SoundTouch. Can anyone offer some guidance either on how to implement SoundTouch or a different library with the capabilities that I'm looking for? Thank you.

    Read the article

  • Downsampling and applying a lowpass filter to digital audio

    - by twk
    I've got a 44Khz audio stream from a CD, represented as an array of 16 bit PCM samples. I'd like to cut it down to an 11KHz stream. How do I do that? From my days of engineering class many years ago, I know that the stream won't be able to describe anything over 5500Hz accurately anymore, so I assume I want to cut everything above that out too. Any ideas? Thanks. Update: There is some code on this page that converts from 48KHz to 8KHz using a simple algorithm and a coefficient array that looks like { 1, 4, 12, 12, 4, 1 }. I think that is what I need, but I need it for a factor of 4x rather than 6x. Any idea how those constants are calculated? Also, I end up converting the 16 byte samples to floats anyway, so I can do the downsampling with floats rather than shorts, if that helps the quality at all.

    Read the article

  • Java playback of 24 bit audio is incorrect

    - by Paul Hampson
    I am using the javax sound API to implement a simple console playback program based on http://www.jsresources.org/examples/AudioPlayer.html. Having tested it using a 24 bit ramp file (each sample is the last sample plus 1 over the full 24 bit range) it is evident that something odd is happening during playback. The recorded output is not the contents of the file (I have a digital loopback to verify this). It seems to be misinterpreting the samples in some way that causes the left channel to look like it is having some gain applied to it and the right channel looks like it is being attenuated. I have looked into whether the PAN and BALANCE controls need setting but these aren't available and I have checked the windows xp sound system settings. Any other form of playback of this ramp file is fine. If I do the same test with a 16bit file it performs correctly with no corruption of the stream. So does anyone have any idea why the Java Sound API is modifying my audio stream?

    Read the article

  • Iphone progressive download audio player

    - by joynes
    Hi! Im trying to implement a progressive download audio player for the iphone, ie using http and fixed size mp3-files. I found the AudioStreamer project but it seems very complicated and works best with endless streams. I need to be able to find out the total length of audiofiles and I also need to be able to seek in the files. I found a hacked deviation from AudioStreamer but it doesnt seem to work very well for me. http://www.saygoodnight.com/?p=14 Im wondering if there is a more simple way to achieve my goals or if there are some better working samples out there? I found the bass library but not much documentation about it. /Br Johannes

    Read the article

  • Audio -- How much performance improvement can I expect from from reducing function calls by using bu

    - by morgancodes
    I'm working on an audio-intensive app for the iPhone. I'm currently calling a number of different functions for each sample I need to calculate. For example, I have an envelope class. When I calculate a sample, I do something like: sampleValue = oscilator->tic() * envelope->tic(); But I could also do something like: for(int i = 0; i < bufferLength; i++){ buffer[i] = oscilatorBuffer[i] * evelopeBuffer[i]; } I know the second will be more efficient, but don't know by how much. Are function calls expensive enough that I'd be crazy not to use buffers if I care event a tiny bit about performance?

    Read the article

  • Converting raw bytes into audio sound

    - by Afro Genius
    In my application I inherit a javastreamingaudio class from the freeTTS package then bypass the write method which sends an array of bytes to the SourceDataLine for audio processing. Instead of writing to the data line, I write this and subsequent byte arrays into a buffer which I then bring into my class and try to process into sound. My application processes sound as arrays of floats so I convert to float and try to process but always get static sound back. I am sure this is the way to go but am missing something along the way. I know that sound is processed as frames and each frame is a group of bytes so in my application I have to process the bytes into frames somehow. Am I looking at this the right way? Thanx in advance for any help.

    Read the article

  • Streaming audio - where to start?

    - by Adam Davis
    I need to develop an embedded audio streaming server. Requirements: Voice quality or better Intended for low power wifi transmission Broad support in existing software and devices (ie, windows media player, quicktime, vlc, iPhone, Android, etc). Royalty/patent free, or cheap to license Preferences: Low overhead TCP/IP based streaming protocol Voice grade codec (easy to implement in software, no DSP, 32bit CPU if needed) Would be nice if it supported HTML5 browsers, but is there any codec (such as raw) that is supported by the latest browsers that is lower overhead than MP3? Therefore: What are the relevant streaming protocols I should be looking at? What are the relevant codecs I should be looking at? What transport streams should I be looking at? What am I missing, or where else should I be looking for this type of need?

    Read the article

  • Android - Audio recorder FileNotFound

    - by david
    Hi, I'm trying to record audio this.recorder = new android.media.MediaRecorder(); this.recorder.setAudioSource(android.media.MediaRecorder.AudioSource.MIC); this.recorder.setOutputFormat(android.media.MediaRecorder.OutputFormat.DEFAULT); this.recorder.setAudioEncoder(android.media.MediaRecorder.AudioEncoder.DEFAULT); this.recorder.setOutputFile("pruebaAudioRecorder.mp4"); this.recorder.prepare(); this.recorder.start(); but when i call prepare method throws the FileNotFound exception. Should I create the file before prepare method? something like new File(...) If so, which should be the file path? thx a lot.

    Read the article

  • Client-side framework for web-app with good audio support

    - by Poita_
    I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

    Read the article

  • iPhone game audio and background music

    - by Boon
    Have a few questions related to adding sounds to my game, specifically intro music (for splash), background music (loop) and button event sounds. Hope you can share your knowledge on this. 1) Should I use compressed sounds or uncompressed sounds? Or perhaps a combination of the two? Are there any limitations on the iPhone hardware that I should be aware of -- for example, the ability to play multiple compressed sounds? 2) What's the best audio format for my purpose? 3) For background music, I am thinking of using AVAudioPlayer. For button event sounds, I am thinking of using AudioServicesPlaySystemSound, what do you think? 4) Any other issues I should be aware of? Thank you!

    Read the article

  • directx audio video error message in debugmode

    - by clamp
    I have a c#/winforms application that uses directx to play some video and audio. whenever i start my application in debugmode i get this annoying message. i can click "continue" and everything seems to work fine. but i still want to get rid of this message. it does not show up in releasemode. Managed Debugging Assistant 'LoaderLock' has detected a problem in 'C:\pathtoexe.exe'. Additional Information: DLL 'C:\WINDOWS\assembly\GAC\Microsoft.DirectX.AudioVideoPlayback\1.0.2902.0__31bf3856ad364e35\Microsoft.DirectX.AudioVideoPlayback.dll' is attempting managed execution inside OS Loader lock. Do not attempt to run managed code inside a DllMain or image initialization function since doing so can cause the application to hang.

    Read the article

  • Naudio - putting audio stream into values [-1,1]

    - by denonth
    Hi all I need to put my audio stream into values of [-1,1]. Can someone tell me a good approach. I was reading byte array and float array from stream but I don't know what to do next. Here is my code: float[] bytes=new float[stream.Length]; float biggest= 0; for (int i = 0; i < stream.Length; i++) { bytes[i] = (byte)stream.ReadByte(); if (bytes[i] > biggest) { biggest=bytes[i]; } } and I don't know how to put values into stream. Because byte is only positive values. And I need to have from [-1,1] for (int i = 0; i < bytes.Count(); i++) { bytes[i] = (byte)(bytes[i] * (1 / biggest)); }

    Read the article

  • audio power on AudioQueue

    - by Tomoyuki
    Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

    Read the article

  • Toggling audio on click?

    - by angela
    please look at this fiddle http://jsfiddle.net/rabelais/yLdkj/1/ The above fiddle shows three bars that on hover play audios. How do I change this so the music plays and pauses on click instead. Also if one audio is playing and another is clicked how can the already playing song pause? $("#one").mouseenter(function () { $('#sound-1').get(0).play(); }); $("#one").mouseleave(function () { $('#sound-1').get(0).pause(); }); $("#two").mouseenter(function () { $('#sound-2').get(0).play(); }); $("#two").mouseleave(function () { $('#sound-2').get(0).pause(); }); $("#three").mouseenter(function () { $('#sound-3').get(0).play(); }); $("#three").mouseleave(function () { $('#sound-3').get(0).pause(); });

    Read the article

  • High level audio crossfading library for python

    - by tcoopman
    I am looking for a high level audio library that supports crossfading for python (and that works in linux). In fact crossfading a song and saving it is about the only thing I need. I tried pyechonest but I find it really slow. Working with multiple songs at the same time is hard on memory too (I tried to crossfade about 10 songs in one, but I got out of memory errors and my script was using 1.4Gb of memory). So now I'm looking for something else that works with python. I have no idea if there exists anything like that, if not, are there good command line tools for this, I could write a wrapper for the tool.

    Read the article

  • audio stream sampling rate in linux

    - by farhan
    Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

    Read the article

  • audio error in vmware running mac os x

    - by PenguinSource
    simple synchronous loading of an audio file (.mp3) in a cocos2d app makes my vmware disconnect the sound. the error is display bottom right, saying 'error in creating sound stream; sound is disconnected' i read that it might be cause of my vmware's version (mine is 8) but I'm looking for a fix, not to downgrade to another version. before i get that error, the sound on the system works just fine (youtube, etc) the exact code im calling is.. [CDSoundEngine setMixerSampleRate: CD_SAMPLE_RATE_MID]; [[CDAudioManager sharedManager] setResignBehavior: kAMRBStopPlay autoHandle:Yes]; soundEngine = [SimpleAudioEngine sharedEngine]; [soundEngine preloadBackgroundMusic:@"somemp3.mp3"]; [soundEngine playBackgroundMusic:@"somemp3.mp3"]; maybe the bit rate is too high .. ? thanks

    Read the article

< Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >