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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • How to fix these compiler errors?

    - by Sandra Schlichting
    I have this source code from 2001 that I would like to compile. It gives this: $ make g++ -O99 -Wall -DLINUX -pedantic -c -o audio.o audio.cpp In file included from audio.cpp:7: audio.h:14: error: use of enum ‘mad_flow’ without previous declaration audio.h:15: error: use of enum ‘mad_flow’ without previous declaration audio.h:17: error: use of enum ‘mad_flow’ without previous declaration audio.cpp: In function ‘mad_flow audio::input(void*, mad_stream*)’: audio.cpp:19: error: new declaration ‘mad_flow audio::input(void*, mad_stream*)’ audio.h:14: error: ambiguates old declaration ‘int audio::input(void*, mad_stream*)’ audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:23: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:23: error: within this context audio.h:10: error: ‘char* audio::stream::Buffer’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:27: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:27: error: within this context audio.cpp: In function ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’: audio.cpp:49: error: new declaration ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’ audio.h:15: error: ambiguates old declaration ‘int audio::output(void*, const mad_header*, mad_pcm*)’ audio.cpp: In function ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’: audio.cpp:83: error: new declaration ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’ audio.h:17: error: ambiguates old declaration ‘int audio::error(void*, mad_stream*, mad_frame*)’ audio.cpp: In constructor ‘audio::stream::stream(const char*)’: audio.cpp:119: error: ‘input’ was not declared in this scope audio.cpp:122: error: ‘output’ was not declared in this scope audio.cpp:123: error: ‘error’ was not declared in this scope make: *** [audio.o] Error 1 audio.h contains #include <stdlib.h> #include "mad.h" namespace audio { class stream { private: char* Buffer; size_t BufferSize, BufferPos; struct mad_decoder Decoder; friend enum mad_flow input(void* Data, struct mad_stream* MadStream); friend enum mad_flow output(void* Data, const struct mad_header* Header, struct mad_pcm* PCM); friend enum mad_flow error(void* Data, struct mad_stream* MadStream, struct mad_frame* Frame); public: stream(const char* FileName); ~stream(); void play(); }; } I have tried to just insert enum mad_flow {}; but that just gave a new problem. Can anyone see how to fix this?

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  • On-the-fly lossless image compression

    - by geschema
    I have an embedded application where an image scanner sends out a stream of 16-bit pixels that are later assembled to a grayscale image. As I need to both save this data locally and forward it to a network interface, I'd like to compress the data stream to reduce the required storage space and network bandwidth. Is there a simple algorithm that I can use to losslessly compress the pixel data? I first thought of computing the difference between two consecutive pixels and then encoding this difference with a Huffman code. Unfortunately, the pixels are unsigned 16-bit quantities so the difference can be anywhere in the range -65535 .. +65535 which leads to potentially huge codeword lengths. If a few really long codewords occur in a row, I'll run into buffer overflow problems.

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  • CPU-adaptive compression

    - by liori
    Hello, Let assume I need to send some data from one computer to another, over a pretty fast network... for example standard 100Mbit connection (~10MB/s). My disk drives are standard HDD, so their speed is somewhere between 30MB/s and 100MB/s. So I guess that compressing the data on the fly could help. But... I don't want to be limited by CPU. If I choose an algorithm that is intensive on CPU, the transfer will actually go slower than without compression. This is difficult with compressors like GZIP and BZIP2 because you usually set the compression strength once for the whole transfer, and my data streams are sometimes easy, sometimes hard to compress--this makes the process suboptimal because sometimes I do not use full CPU, and sometimes the bandwidth is underutilized. Is there a compression program that would adapt to current CPU/bandwidth and hit the sweet spot so that the transfer will be optimal? Ideally for Linux, but I am still curious about all solutions. I'd love to see something compatible with GZIP/BZIP2 decompressors, but this is not necessary. So I'd like to optimize total transfer time, not simply amount of bytes to send. Also I don't need real time decompression... real time compression is enough. The destination host can process the data later in its spare time. I know this doesn't change much (compression is usually much more CPU-intensive than decompression), but if there's a solution that could use this fact, all the better. Each time I am transferring different data, and I really want to make these one-time transfers as quick as possible. So I won't benefit from getting multiple transfers faster due to stronger compression. Thanks,

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  • Is there an audio recording application/tool that has Tivo-like functionality?

    - by Bob
    I do a lot of live speech recording that requires me to quickly jump back and then transcribe a particular piece of the audio, then go back to recording again, while still maintaining the full audio file. So Far I've done this by splitting the audio and running one line to a recorder (for the whole audio), and one to my computer. Then I use something like Audacity to record, and then stop/go back whenever I hear something worth transcribing. This requires me to stop the recording, then start it up again and I end up missing chunks of the speech I'm listening to. Is there a tool that would let me rewind, then listen again and continue listening at a buffered distance from the audio recording, the way Tivo does with television shows?

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  • HTML Audio performance

    - by user1888309
    I'm working on HTML drum machine, and I`ve met some performance issues, rhythm start to break if BPM is higher than 110 but I'm expecting to make it work on BPM over 180. I guess that it can be related with format or codec of audio files, however it also maybe that my code is not very optimised (as I can see from JS CPU profiling it's not). So I'm expecting you guys give me some code review or some hints on optimisation. Although all similar projects I've found on internet didn't work good and maybe it's just restrictions of Audio API. By the way, it's very raw and sounds works only on Chrome under Mac OS, so any advise on audio encoding for web also would be great Project on Github pages Screenshot of Groove which breaks UPDATE Ok, I've found that I was encoding audio files incorrectly, after fixing that rhythm stopped breaking, and also it started working in Mozilla. But still there are issues on windows OS.

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  • Multiple audio sources on a single gameObject in unity

    - by angryInsomniac
    So, I have an audio system set up wherein I have loaded all my audio clips centrally and play them on demand by passing the requesting audioSource into the sound manager. However, there is a complication wherein if I want to overlay multiple looping sounds, I need to have multiple audio sources on an object, which is fine , so I created two in my script instantiated them and played my clips on them and then the world went crazy. For some reason, when I create two audio Sources in an object only the latest one is ever used, even if I explicitly keep objects separated, playing a clip on one or the other plays the clip on the last one that was created, furthermore, either this last one is not created in the right place or somehow messes with the rolloff rules because I can hear it all across my level, havign just one source works fine, but putting a second one on it causes shit to go batshit insane. Does anyone know the reason / solution for this ? Some pseudocode : guardSoundsSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardSoundsSource.name = "Guard_Sounds_source"; // Setup this source guardThrusterSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardThrusterSource.name = "Guard_Thruster_Source"; // setup this source // play using custom Sound manager soundMan.soundMgr.playOnSource(guardSoundsSource,"Guard_Idle_loop" ,true,GameManager.Manager.PlayerType); // this method prints out the name of the source the sound was to be played on and it always shows "Guard_Thruster_Source" even on the "Guard_Idle_loop" even though I clearly told it to use "Guard_Sounds_source"

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  • GZip/Deflate Compression in ASP.NET MVC

    - by Rick Strahl
    A long while back I wrote about GZip compression in ASP.NET. In that article I describe two generic helper methods that I've used in all sorts of ASP.NET application from WebForms apps to HttpModules and HttpHandlers that require gzip or deflate compression. The same static methods also work in ASP.NET MVC. Here are the two routines:/// <summary> /// Determines if GZip is supported /// </summary> /// <returns></returns> public static bool IsGZipSupported() { string AcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"]; if (!string.IsNullOrEmpty(AcceptEncoding) && (AcceptEncoding.Contains("gzip") || AcceptEncoding.Contains("deflate"))) return true; return false; } /// <summary> /// Sets up the current page or handler to use GZip through a Response.Filter /// IMPORTANT: /// You have to call this method before any output is generated! /// </summary> public static void GZipEncodePage() { HttpResponse Response = HttpContext.Current.Response; if (IsGZipSupported()) { string AcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"]; if (AcceptEncoding.Contains("gzip")) { Response.Filter = new System.IO.Compression.GZipStream(Response.Filter, System.IO.Compression.CompressionMode.Compress); Response.Headers.Remove("Content-Encoding"); Response.AppendHeader("Content-Encoding", "gzip"); } else { Response.Filter = new System.IO.Compression.DeflateStream(Response.Filter, System.IO.Compression.CompressionMode.Compress); Response.Headers.Remove("Content-Encoding"); Response.AppendHeader("Content-Encoding", "deflate"); } } // Allow proxy servers to cache encoded and unencoded versions separately Response.AppendHeader("Vary", "Content-Encoding"); } The first method checks whether the client sending the request includes the accept-encoding for either gzip or deflate, and if if it does it returns true. The second function uses IsGzipSupported() to decide whether it should encode content and uses an Response Filter to do its job. Basically response filters look at the Response output stream as it's written and convert the data flowing through it. Filters are a bit tricky to work with but the two .NET filter streams for GZip and Deflate Compression make this a snap to implement. In my old code and even now in MVC I can always do:public ActionResult List(string keyword=null, int category=0) { WebUtils.GZipEncodePage(); …} to encode my content. And that works just fine. The proper way: Create an ActionFilterAttribute However in MVC this sort of thing is typically better handled by an ActionFilter which can be applied with an attribute. So to be all prim and proper I created an CompressContentAttribute ActionFilter that incorporates those two helper methods and which looks like this:/// <summary> /// Attribute that can be added to controller methods to force content /// to be GZip encoded if the client supports it /// </summary> public class CompressContentAttribute : ActionFilterAttribute { /// <summary> /// Override to compress the content that is generated by /// an action method. /// </summary> /// <param name="filterContext"></param> public override void OnActionExecuting(ActionExecutingContext filterContext) { GZipEncodePage(); } /// <summary> /// Determines if GZip is supported /// </summary> /// <returns></returns> public static bool IsGZipSupported() { string AcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"]; if (!string.IsNullOrEmpty(AcceptEncoding) && (AcceptEncoding.Contains("gzip") || AcceptEncoding.Contains("deflate"))) return true; return false; } /// <summary> /// Sets up the current page or handler to use GZip through a Response.Filter /// IMPORTANT: /// You have to call this method before any output is generated! /// </summary> public static void GZipEncodePage() { HttpResponse Response = HttpContext.Current.Response; if (IsGZipSupported()) { string AcceptEncoding = HttpContext.Current.Request.Headers["Accept-Encoding"]; if (AcceptEncoding.Contains("gzip")) { Response.Filter = new System.IO.Compression.GZipStream(Response.Filter, System.IO.Compression.CompressionMode.Compress); Response.Headers.Remove("Content-Encoding"); Response.AppendHeader("Content-Encoding", "gzip"); } else { Response.Filter = new System.IO.Compression.DeflateStream(Response.Filter, System.IO.Compression.CompressionMode.Compress); Response.Headers.Remove("Content-Encoding"); Response.AppendHeader("Content-Encoding", "deflate"); } } // Allow proxy servers to cache encoded and unencoded versions separately Response.AppendHeader("Vary", "Content-Encoding"); } } It's basically the same code wrapped into an ActionFilter attribute, which intercepts requests MVC requests to Controller methods and lets you hook up logic before and after the methods have executed. Here I want to override OnActionExecuting() which fires before the Controller action is fired. With the CompressContentAttribute created, it can now be applied to either the controller as a whole:[CompressContent] public class ClassifiedsController : ClassifiedsBaseController { … } or to one of the Action methods:[CompressContent] public ActionResult List(string keyword=null, int category=0) { … } The former applies compression to every action method, while the latter is selective and only applies it to the individual action method. Is the attribute better than the static utility function? Not really, but it is the standard MVC way to hook up 'filter' content and that's where others are likely to expect to set options like this. In fact,  you have a bit more control with the utility function because you can conditionally apply it in code, but this is actually much less likely in MVC applications than old WebForms apps since controller methods tend to be more focused. Compression Caveats Http compression is very cool and pretty easy to implement in ASP.NET but you have to be careful with it - especially if your content might get transformed or redirected inside of ASP.NET. A good example, is if an error occurs and a compression filter is applied. ASP.NET errors don't clear the filter, but clear the Response headers which results in some nasty garbage because the compressed content now no longer matches the headers. Another issue is Caching, which has to account for all possible ways of compression and non-compression that the content is served. Basically compressed content and caching don't mix well. I wrote about several of these issues in an old blog post and I recommend you take a quick peek before diving into making every bit of output Gzip encoded. None of these are show stoppers, but you have to be aware of the issues. Related Posts GZip Compression with ASP.NET Content ASP.NET GZip Encoding Caveats© Rick Strahl, West Wind Technologies, 2005-2012Posted in ASP.NET  MVC   Tweet !function(d,s,id){var js,fjs=d.getElementsByTagName(s)[0];if(!d.getElementById(id)){js=d.createElement(s);js.id=id;js.src="//platform.twitter.com/widgets.js";fjs.parentNode.insertBefore(js,fjs);}}(document,"script","twitter-wjs"); (function() { var po = document.createElement('script'); po.type = 'text/javascript'; po.async = true; po.src = 'https://apis.google.com/js/plusone.js'; var s = document.getElementsByTagName('script')[0]; s.parentNode.insertBefore(po, s); })();

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  • Play an audio file using RemoteIO and Audio Unit

    - by NeilMonday
    I am looking at Apple's 'aurioTouch' example for the iPhone and I would like to play an mp3 or wav instead of using the built in mic. I am very new to the audio portion of iPhone programming, but I think I need to modify the SetupRemoteIO(...) function and replace the AudioComponent named 'comp' with a custom AudioComponent that plays a file. Basically I want the app to function exactly the same as the original, but with an audio file as the input instead of the mic.

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  • Audio decoding delay when changing the audio language

    - by mahendiran.b
    My gstreamer Pipeline is like this Approach1 --------------input-selector->Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux-->| | --------------- Queue->videoParser->videoSink In this approach 1, there is a delay in audio decoding when I toggle between various audio language. Approach2 ------ input-selector-> Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux---multiqueue>| | ------- Queue->videoParser->VideoSink But there is no delay is observed in approach2. Can anyone please explain the reason behind this ? what is the specialty of multiqueue here?

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  • Embed audio broadcasting on web page

    - by giargo
    Hi, I'd like to embed simple audio player on my webpage and I want it to get the audio from a stream broadcasted from my server. I read I can use IceCast on my web-server, getting an audio stream from a client using IceS (or this is what i got from other questions and articles) but once I have my stream, IceCast is supposed to broadcast it on an URL, that can be opened from pkayers like winamp or similar. I've found out this is quite a rare topic, usually people just want to broadcast "radio" where files are taken from a static playlist. In this case I have to get a stream from an IceCast URL and embed it with a player on a web page. Thank.

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  • Audio libraries for PC indie games [closed]

    - by bluescrn
    Possible Duplicate: Cross-Platform Audio API Suggestions What options are out there these days for audio playback/mixing in C++? Primarily for Windows, but portability (particularly to Mac and iOS) would be desirable. For a small indie game, potentially commercial, though - so I'm looking for something free/low-cost. My requirements are fairly basic - I don't need 3D sound, or many-channels - simple stereo is fine. Just need to be able to mix sound effects and a music stream, maybe decoding one or more compressed audio formats (.ogg/.mp3 etc), with all the basic controls over looping, pitch, volume, etc. Is OpenAL more-or-less the standard choice, or are there other good options out there?

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  • Verizon SongID - How is it programmed?

    - by CheeseConQueso
    For anyone not familiar with Verizon's SongID program, it is a free application downloadable through Verizon's VCast network. It listens to a song for 10 seconds at any point during the song and then sends this data to some all-knowing algorithmic beast that chews it up and sends you back all the ID3 tags (artist, album, song, etc...) The first two parts and last part are straightforward, but what goes on during the processing after the recorded sound is sent? I figure it must take the sound file (what format?), parse it (how? with what?) for some key identifiers (what are these? regular attributes of wave functions? phase/shift/amplitude/etc), and check it against a database. Everything I find online about how this works is something generic like what I typed above. From audiotag.info This service is based on a sophisticated audio recognition algorithm combining advanced audio fingerprinting technology and a large songs' database. When you upload an audio file, it is being analyzed by an audio engine. During the analysis its audio “fingerprint” is extracted and identified by comparing it to the music database. At the completion of this recognition process, information about songs with their matching probabilities are displayed on screen.

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  • How to get musicbrainz track information from audio file

    - by Baki
    Can anyone tell me how to get track information from the MusicBrainz database from an audio file (mp3, wav, wma, ogg, etc...) using audio fingerprinting. I'm using MusicBrainz Sharp library, but any other library is ok. I've seen that you must use the libofa library, that you can't use MusicBrainz Sharp to get puid from the audio file, but I can't figure out how to use libofa with C#. Please show some examples and code snippets to help me, because I can't find them anywhere. Thanks in advance!

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  • Advanced file compression software for Mac OSX

    - by Steven Roose
    Back when I used Windows, I always used WinRAR for file compression and decompression. It had a fair amount of options like 'just storage' vs 'hard compression', password protection and archive type. Now that I use Mac OSX, the only compression possibility I have is the default Finder's Compress to Zip. I downloaded the most popular decompression software "Unarchiver". But this app can't compress other archive types either. I went for a search but there seem to be hardly any good advanced compression tools that work nice in OSX and have the options WinRAR has. (WinRAR works in OSX but command line only, I'm looking for something with a GUI.) Any ideas? I strongly prefer freeware. I found Archiver and StiffIt, but they are both commercial.

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  • Transparently decompressing data in archive to allow greater compression later

    - by Vi.
    I have, for example, filesystem image which have some compressed files (with weak compression such as gzip), for example, manpages or archives with the same uncompressed content nearby. How to pre-filter the data to "expand" compressed data to plain form (to re-compress it with strong compression) and then post-filter after decompression to restore original "semi-compressed" image? SHA-1 match is advices but not strictly required (but the resulting image must work, e.g. re-compressed files should not grow too much, be decompressible etc.) Like improving compression ratio by reversing weak compression algorithms. Are there programs for this?

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  • What are my audio devices?

    - by hellocatfood
    I'm attempting to use easycap to record from my camcorder but I'm having a slight problem. Using their test script I'm able to get audio and video. I've noticed that in the script on line 159 it makes a call to "DEV_ADUIO", which is reported as being "plughw:2,0". Exactly what is this device? Is it located in /dev/ somewhere? I've done "ls /dev/" and I can't find anything that would suggest an audio device

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  • How to find an audio file's length (in seconds)

    - by mIL3S
    Hi all! (Objective C) Just using simple AudioServicesPlaySystemSoundID and its counterparts, but I can't find in the documentation if there is already a way to find the length of an audio file. I know there is AudioServicesGetPropertyInfo, but that seems to return a byte-buffer - do audio files embed their length in themselves and I can just extract it with this? Or is there perhaps a formula based on bit-rate * fileSize to convert to length-of-time? mIL3S www.milkdrinkingcow.com

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