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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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  • Codeigniter + JQuery + Processing.js to replace a Delphi App

    - by Peter Turner
    So, I've got a mandate to make our aged trillion lined Delphi app web based and it needs to make heavy use of the <canvas> element (HTML5 compatibility doesn't seem to be a big issue since we can just make our clients use a compatible browser the way we'd make them use a compatible version of Windows in the win32 environment). The Delphi app in question is almost completely database driven and will still pretty much continue to be developed as the main product. What I am tasked with is pretty much recreating a scaled down version of the program that performs the major functions of the whole program. I couldn't find any frameworks that simulate windows forms using the canvas element, I'm assuming this is probably by design since it is easier just to use HTML, well, be that as it may, I still think it would be cool to have a few of my cool controls on the web (TRichView and TVirtualTree, etc...) So my question is, to anyone who has tried this before, A.) What can we use for an IDE to code this web app (I just use emacs, but no one else in my company does)? B.) Is it a good idea to mix PHP and Processing.JS? It seems like I'm using a lot of AJAX to get anything to happen. 3 calls just for one dialog box to pop up, Loads the HTML for the dialog, Loads the XML to populate the database info on the form Loads the processing.js PJS file which draws the database info to the canvas. Is three a lot, do people usually combine all their gets into one?

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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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  • TV audio processing with TV capture card

    - by Jonathan Barbero
    Hello, I'm looking for an open source library or framework to process audio signal from a TV capture card. The idea is to detect TV ad spots and register the time and the channel where them happends. I never worked in something like this, so, any information, link, idea is welcome. Thanks in advance!

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  • Audio processing libraries for Ruby?

    - by J. Pablo Fernández
    Any recommendation on libraries to do audio processing in Ruby. I need to do the following two tasks: Find silences, for which I'm happy to just be able to iterate over each sample in the wave. Cut and paste pieces of wav files to form a new wav file. Convert wav to mp3, which I will probably leave to lame anyway. I'm looking for the equivalent of NAudio, a C# library.

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  • Audio not working

    - by user3215
    Anybody could help me in troubleshooting audio problem on ubutnu 9.04 desktop edition?. For some reason I've to keep this os not upgraded and I'm trying to fix the audio problem on this for months. It works well on upgraded version(9.10,10.04) but not on jaunty. aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod | grep snd: snd_hda_intel 436148 7 snd_pcm_oss 46336 0 snd_mixer_oss 22656 1 snd_pcm_oss snd_pcm 83076 4 snd_hda_intel,snd_pcm_oss snd_seq_dummy 10756 0 snd_seq_oss 37760 0 snd_seq_midi 14336 0 snd_rawmidi 29696 1 snd_seq_midi snd_seq_midi_event 15104 2 snd_seq_oss,snd_seq_midi snd_seq 56880 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 29704 2 snd_pcm,snd_seq snd_seq_device 14988 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 62756 21 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15200 1 snd snd_page_alloc 16904 2 snd_hda_intel,snd_pcm cat /proc/asound/cards: 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xe1280000 irq 16 cat /proc/asound/version: Advanced Linux Sound Architecture Driver Version 1.0.18rc3. vim /etc/modules: # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with "#" are ignored. lp Audio Settings:

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  • SEHException throw using Microsoft XACT Audio Framework (XACT3)

    - by Sweta Dwivedi
    I have been developing a game using Kinect + XNA and using Microsoft Audio Creation tool (XACT3) for managing my sound files and music, however in the code an SEHException is thrown whenever it tries to get the wave file from the wave Bank . . Sometimes the code works magically and all of a sudden it will start throwing this exception randomly ..I need a help on solving this exception /*Declaring Audio Engine for music*/ AudioEngine engine; SoundBank soundBank; WaveBank waveBank; Cue cue; /*Declaring Audio engine for sound effects*/ AudioEngine engine1; SoundBank soundbank; WaveBank wavebank; Cue effect; engine = new AudioEngine(@"Content\therapy.xgs"); soundBank = new SoundBank(engine, @"Content\Sound Bank.xsb"); **waveBank = new WaveBank(engine, @"Content\Wave Bank.xwb");** cue = null; engine1 = new AudioEngine(@"Content\Music_Manager\Sound_effects.xgs"); soundbank = new SoundBank(engine1, @"Content\Music_Manager\Sound1.xsb"); **wavebank = new WaveBank(engine1, @"Content\Music_Manager\Wave1.xwb");** effect = null; cue = soundBank.GetCue("hypnotizing"); cue.Play();

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  • Where can I learn image processing? [on hold]

    - by Little Child
    I am learning image processing on my own and I have managed to teach myself a fair few things like: Making images grayscale using 3 different methods Applying a 'pixellate' filter Applying a 'pointillize' filter Make images out of lines Now, I want to take my knowledge further but I do not know how. Adding more information: I am interested in making software like Photoshop or Gimp (although it won't be half as powerful as these 2). So, I want to learn to apply various creative effects to an image. Can someone please suggest resources for this??

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  • Physics-based dynamic audio generation in games

    - by alexc
    I wonder if it is possible to generate audio dynamically without any (!) audio assets, using pure mathematics/physics and some input values like material properties and spatial distribution of content in scene space. What I have in mind is something like a scene, with concrete floor, wooden table and glass on it. Now let's assume force pushes the glass towards the edge of table and then the glass falls onto the floor and shatters. The near-realistic glass destruction itself would be possible using voxels and good physics engine, but what about the sound the glass makes while shattering? I believe there is a way to generate that sound, because physics of sound is fairly known these days, but how computationaly costy that would be? Consumer hardware or supercomputers? Do any of you know some good resources/videos of such an experiment?

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  • Playing part of a sfx audio file in HTML5 using WebAudio

    - by Matthew James Davis
    I have compiled all of my sound effects into one sequenced .ogg file. I have the start and stop times for each sound effect. How do I play the individual effects? That is, how do I play part of an audio file. More specificially, I've created a dictionary { 'sword_hit': { src: 'sfx.ogg', start: 265, // ms length: 212 // ms } } that my play_sound() function can use to look up 'sword_hit' and play the correct audio file at the correct start time for the correct duration. I simply need to know how to tell the WebAudio API to start playing at start ms and only play for length ms.

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  • How should I implement a command processing application?

    - by Nini Michaels
    I want to make a simple, proof-of-concept application (REPL) that takes a number and then processes commands on that number. Example: I start with 1. Then I write "add 2", it gives me 3. Then I write "multiply 7", it gives me 21. Then I want to know if it is prime, so I write "is prime" (on the current number - 21), it gives me false. "is odd" would give me true. And so on. Now, for a simple application with few commands, even a simple switch would do for processing the commands. But if I want extensibility, how would I need to implement the functionality? Do I use the command pattern? Do I build a simple parser/interpreter for the language? What if I want more complex commands, like "multiply 5 until >200" ? What would be an easy way to extend it (add new commands) without recompiling? Edit: to clarify a few things, my end goal would not be to make something similar to WolframAlpha, but rather a list (of numbers) processor. But I want to start slowly at first (on single numbers). I'm having in mind something similar to the way one would use Haskell to process lists, but a very simple version. I'm wondering if something like the command pattern (or equivalent) would suffice, or if I have to make a new mini-language and a parser for it to achieve my goals?

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  • mplayer (mplayerhq.hu) repeats ending audio frames

    - by kamikatze
    mplayer (from mplayerhq.hu) on windows repeats the last few audio frames upon exit. When the video ends, before you can see Exiting... (End of file) in the command prompt, you will hear the last 1/2 second or so of the audio track again. This behavior is the same for multiple containers/codecs/soundcards Vista or Windows 7. Is there a workaround for this? My playback specs: MPlayer Sherpya-MT-SVN-r31027-4.2.5 (C) 2000-2010 MPlayer Team 150 audio & 343 video codecs Playing splash_final.wmv. ASF file format detected. [asfheader] Audio stream found, -aid 1 [asfheader] Video stream found, -vid 2 VIDEO: [WMV3] 1280x720 24bpp 1000.000 fps 6291.5 kbps (768.0 kbyte/s) ========================================================================== Opening video decoder: [dmo] DMO video codecs DMO dll supports VO Optimizations 0 1 DMO dll might use previous sample when requested Decoder supports the following formats: YV12 YUY2 UYVY YVYU RGB8 [..] Decoder is capable of YUV output (flags 0x1b) Movie-Aspect is undefined - no prescaling applied. VO: [directx] 1280x720 = 1280x720 Planar YV12 Selected video codec: [wmv9dmo] vfm: dmo (Windows Media Video 9 DMO) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, s16le, 329.8 kbit/23.37% (ratio: 41221-176400) Selected audio codec: [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg)) ========================================================================== AO: [dsound] 44100Hz 2ch s16le (2 bytes per sample) Starting playback...

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  • Form, function and complexity in rule processing

    - by Charles Young
    Tim Bass posted on ‘Orwellian Event Processing’. I was involved in a heated exchange in the comments, and he has more recently published a post entitled ‘Disadvantages of Rule-Based Systems (Part 1)’. Whatever the rights and wrongs of our exchange, it clearly failed to generate any agreement or understanding of our different positions. I don't particularly want to promote further argument of that kind, but I do want to take the opportunity of offering a different perspective on rule-processing and an explanation of my comments. For me, the ‘red rag’ lay in Tim’s claim that “...rules alone are highly inefficient for most classes of (not simple) problems” and a later paragraph that appears to equate the simplicity of form (‘IF-THEN-ELSE’) with simplicity of function.   It is not the first time Tim has expressed these views and not the first time I have responded to his assertions.   Indeed, Tim has a long history of commenting on the subject of complex event processing (CEP) and, less often, rule processing in ‘robust’ terms, often asserting that very many other people’s opinions on this subject are mistaken.   In turn, I am of the opinion that, certainly in terms of rule processing, which is an area in which I have a specific interest and knowledge, he is often mistaken. There is no simple answer to the fundamental question ‘what is a rule?’ We use the word in a very fluid fashion in English. Likewise, the term ‘rule processing’, as used widely in IT, is equally difficult to define simplistically. The best way to envisage the term is as a ‘centre of gravity’ within a wider domain. That domain contains many other ‘centres of gravity’, including CEP, statistical analytics, neural networks, natural language processing and so much more. Whole communities tend to gravitate towards and build themselves around some of these centres. The term 'rule processing' is associated with many different technology types, various software products, different architectural patterns, the functional capability of many applications and services, etc. There is considerable variation amongst these different technologies, techniques and products. Very broadly, a common theme is their ability to manage certain types of processing and problem solving through declarative, or semi-declarative, statements of propositional logic bound to action-based consequences. It is generally important to be able to decouple these statements from other parts of an overall system or architecture so that they can be managed and deployed independently.  As a centre of gravity, ‘rule processing’ is no island. It exists in the context of a domain of discourse that is, itself, highly interconnected and continuous.   Rule processing does not, for example, exist in splendid isolation to natural language processing.   On the contrary, an on-going theme of rule processing is to find better ways to express rules in natural language and map these to executable forms.   Rule processing does not exist in splendid isolation to CEP.   On the contrary, an event processing agent can reasonably be considered as a rule engine (a theme in ‘Power of Events’ by David Luckham).   Rule processing does not live in splendid isolation to statistical approaches such as Bayesian analytics. On the contrary, rule processing and statistical analytics are highly synergistic.   Rule processing does not even live in splendid isolation to neural networks. For example, significant research has centred on finding ways to translate trained nets into explicit rule sets in order to support forms of validation and facilitate insight into the knowledge stored in those nets. What about simplicity of form?   Many rule processing technologies do indeed use a very simple form (‘If...Then’, ‘When...Do’, etc.)   However, it is a fundamental mistake to equate simplicity of form with simplicity of function.   It is absolutely mistaken to suggest that simplicity of form is a barrier to the efficient handling of complexity.   There are countless real-world examples which serve to disprove that notion.   Indeed, simplicity of form is often the key to handling complexity. Does rule processing offer a ‘one size fits all’. No, of course not.   No serious commentator suggests it does.   Does the design and management of large knowledge bases, expressed as rules, become difficult?   Yes, it can do, but that is true of any large knowledge base, regardless of the form in which knowledge is expressed.   The measure of complexity is not a function of rule set size or rule form.  It tends to be correlated more strongly with the size of the ‘problem space’ (‘search space’) which is something quite different.   Analysis of the problem space and the algorithms we use to search through that space are, of course, the very things we use to derive objective measures of the complexity of a given problem. This is basic computer science and common practice. Sailing a Dreadnaught through the sea of information technology and lobbing shells at some of the islands we encounter along the way does no one any good.   Building bridges and causeways between islands so that the inhabitants can collaborate in open discourse offers hope of real progress.

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  • No audio input deviced are installed

    - by Meowbits
    If I go to Sound Recording Devices and it says "No audio devices are installed" If I click to set up a microphone I get an error "Wizard could not launch, No audio input device found, make sure your audio hardware is working properly and check your audio configuration in the Audio Devices and Sound Themes control panel. Where can I get an audio input device? I just want something so I can actually use the microphone on my headset. This is ridiculous. I have tried to look for any file but I simply cannot find a way to add an audio input device... I really do not want to format my computer just for this problem but I am starting to feel like that is the only option I have. I have the latest chipsets

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  • mix audio with h264 mp4 video with ffmpeg

    - by user2362912
    I have 2 files : Input #0, wav, from '105426_1.wav': Duration: 00:00:09.98, bitrate: 1312 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 41000 Hz, stereo, s16, 1312 kb/s and: Duration: 00:00:41.29, start: 0.000000, bitrate: 1313 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1211 kb/s, 24.42 fps, 25 tbr, 90k tbn, 48 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 99 kb/s Metadata: handler_name : SoundHandler I want to insert first audio file into video in special place (for example in 10 secunde of video) and mix it with audio stream of video file. I try to /usr/local/bin/ffmpeg -i 105426_1.wav -i 105426.mp4 -map 0:0 -map 1:1 -map 1:0 video_finale.mp4 but result is : Duration: 00:00:41.31, start: 0.046440, bitrate: 755 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:2(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 588 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc Metadata: handler_name : VideoHandler I need only one audio stream and first stream play not from beginig but from 10 sec

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  • How can I split a stereo audio track of a movie into two separate audio tracks?

    - by pesche
    I often record TV shows with a hard disk recorder/DVD writer, burn them as VRO file and convert to MP4 with Handbrake. The shows are bilingual broadcasts with two mono audio channels instead of a stereo one: dubbed voice on the left, original voice on the right. The TV set and VLC are both perfectly capable to play only the left or the right channel, but other video players may just offer to select between different stereo audio tracks (like they are present on many DVDs). I'd like to have an easy process to create MP4 or MKV files of these shows where the two audio channels are split into two separate audio tracks. The only way that I know of is to extract the audio track (e.g. using MPEG Streamclip), split it into two tracks using an audio tool like Audacity and then merge the audio tracks back (using a DVD authoring software, don't remember all details). Clearly not a thing to repeat regularly. Preferably a solution should run on Mac OS X, but Linux or Windows solutions are very welcome, too.

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  • iPhone SDK: Change playback speed using core audio AVAudioPlayer

    - by Harkonian
    I'd like to be able to play back audio I've recorded using AVAudioRecorder @ 1.5x or 2.0x speed. I don't see anything in AVAudioPlayer that will support that. I'd appreciate some suggestions, with code if possible, on how to accomplish this with the iPhone 3.x SDK. I'm not overly concerned with lowering the pitch to compensate for increased playback speed, but being able to do so would be optimal.

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  • cut audio file with iPhone SDK

    - by Dmitry
    Hi! Is it possible to cut audio file with iPhone SDK? (file has .caf extension) I just need to cut off the silence at the beginning. (Also, maybe it's possible to write new file from the existing one with specified start and end time.) Thanks in advance!

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