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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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  • My 2D collision code does not work as expected. How do I fix it?

    - by farmdve
    I have a simple 2D game with a tile-based map. I am new to game development, I followed the LazyFoo tutorials on SDL. The tiles are in a bmp file, but each tile inside it corresponds to an internal number of the type of tile(color, or wall). The game is simple, but the code is a lot so I can only post snippets. // Player moved out of the map if((player.box.x < 0)) player.box.x += GetVelocity(player, 0); if((player.box.y < 0)) player.box.y += GetVelocity(player, 1); if((player.box.x > (LEVEL_WIDTH - DOT_WIDTH))) player.box.x -= GetVelocity(player, 0); if((player.box.y > (LEVEL_HEIGHT - DOT_HEIGHT))) player.box.y -= GetVelocity(player, 1); // Now that we are here, we check for collisions if(touches_wall(player.box)) { if(player.box.x < player.prev_x) { player.box.x += GetVelocity(player, 0); } if(player.box.x > player.prev_x) { player.box.x -= GetVelocity(player, 0); } if(player.box.y < player.prev_y) { player.box.y += GetVelocity(player, 1); } if(player.box.y > player.prev_y) { player.box.y -= GetVelocity(player, 1); } } player.prev_x = player.box.x; player.prev_y = player.box.y; Let me explain, player is a structure with the following contents typedef struct { Rectangle box; //Player position on a map(tile or whatever). int prev_x, prev_y; // Previous positions int key_press[3]; // Stores which key was pressed/released. Limited to three keys. E.g Left,right and perhaps jump if possible in 2D int velX, velY; // Velocity for X and Y coordinate. //Health int health; bool main_character; uint32_t jump_ticks; } Player; And Rectangle is just a typedef of SDL_Rect. GetVelocity is a function that according to the second argument, returns the velocity for the X or Y axis. This code I have basically works, however inside the if(touches_wall(player.box)) if statement, I have 4 more. These 4 if statements are responsible for detecting collision on all 4 sides(up,down,left,right). However, they also act as a block for any other movement. Example: I move down the object and collide with the wall, as I continue to move down and still collide with the wall, I wish to move left or right, which is indeed possible(not to mention in 3D games), but remember the 4 if statements? They are preventing me from moving anywhere. The original code on the LazyFoo Productions website has no problems, but it was written in C++, so I had to rewrite most of it to work, which is probably where the problem comes from. I also use a different method of moving, than the one in the examples. Of course, that was just an example. I wish to be able to move no matter at which wall I collide. Before this bit of code, I had another one that had more logic in there, but it was flawed.

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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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  • Music player with a few specific requirements

    - by Jordan Uggla
    I am looking for a music player with a few specific requirements: Must have a search function that whittles down results as you type, searching the entire library. Must start playing a song when double clicked, and not continue to another song when that song finishes. Must be approachable and immediately usable by people completely unfamiliar with the program. I think this is mostly covered by the first two requirements being met. I've tried many players but unfortunately every one has failed to meet at least one of the requirements. Rhythmbox meets 1 and 3, but continues to the next search result after the song which was double clicked ends. Banshee is basically the same as Rhythmbox. While it has an option to "Stop when finished" this cannot (as far as I can tell) be made the default when double clicking a song. Audacious (as far as I can tell) fails at 1. Muine meets requirements 1 and 2, but unfortunately I couldn't make the search dialog always shown like it is with Rhythmbox / Banshee which, despite its very simple interface, made Muine incomprehensible to people trying to use it for the first time. Amarok I could not configure to meet requirement 1, but I think it's likely I was just missing something, and with its configurability I'm confident that I can set it up to meet requirements 2 and 3.

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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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  • MP3 player mpman TS300 not recognized

    - by Nick Lemaire
    I just received a mpman ts300 mp3 player for Christmas. But when I try to connect it to ubuntu (10.10) through usb it doesn't seem to be recognized. I searched for a linuxdriver but came up with nothing. I even tried installing mpman manager from the software center, but still the same problem... Has anybody got any ideas about how to fix this? Thank you edit: some additional information When I plug it in, it doesn't show up under /dev. And it is not listed in the result of 'lsusb' This is the output of 'dmesg' [19571.732541] usb 1-4: new high speed USB device using ehci_hcd and address 2 [19571.889154] scsi7 : usb-storage 1-4:1.0 [19572.883155] scsi 7:0:0:0: Direct-Access TS300 USB DISK 1.00 PQ: 0 ANSI: 0 [19572.885856] sd 7:0:0:0: Attached scsi generic sg2 type 0 [19572.887266] sd 7:0:0:0: [sdb] 7868416 512-byte logical blocks: (4.02 GB/3.75 GiB) [19573.012547] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19573.292550] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19573.572565] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19573.725019] sd 7:0:0:0: [sdb] Test WP failed, assume Write Enabled [19573.725024] sd 7:0:0:0: [sdb] Assuming drive cache: write through [19573.728521] sd 7:0:0:0: [sdb] Attached SCSI removable disk [19575.333015] sd 7:0:0:0: [sdb] 7868416 512-byte logical blocks: (4.02 GB/3.75 GiB) [19575.452547] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19580.603253] usb 1-4: device descriptor read/all, error -110 [19580.722560] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19595.842579] usb 1-4: device descriptor read/64, error -110 [19611.072656] usb 1-4: device descriptor read/64, error -110 [19611.302540] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19616.332568] usb 1-4: device descriptor read/8, error -110 [19621.462692] usb 1-4: device descriptor read/8, error -110 [19621.692551] usb 1-4: reset high speed USB device using ehci_hcd and address 2 [19626.722567] usb 1-4: device descriptor read/8, error -110 [19631.852802] usb 1-4: device descriptor read/8, error -110 [19631.962587] usb 1-4: USB disconnect, address 2 [19631.962587] sd 7:0:0:0: Device offlined - not ready after error recovery [19631.962840] sd 7:0:0:0: [sdb] Assuming drive cache: write through [19631.965104] sd 7:0:0:0: [sdb] READ CAPACITY failed [19631.965109] sd 7:0:0:0: [sdb] Result: hostbyte=DID_NO_CONNECT driverbyte=DRIVER_OK [19631.965112] sd 7:0:0:0: [sdb] Sense not available. [19631.965125] sd 7:0:0:0: [sdb] Assuming drive cache: write through [19631.965130] sdb: detected capacity change from 4028628992 to 0 [19632.130042] usb 1-4: new high speed USB device using ehci_hcd and address 3 [19647.242550] usb 1-4: device descriptor read/64, error -110 [19662.472560] usb 1-4: device descriptor read/64, error -110 [19662.702566] usb 1-4: new high speed USB device using ehci_hcd and address 4 [19677.822587] usb 1-4: device descriptor read/64, error -110 [19693.052575] usb 1-4: device descriptor read/64, error -110 [19693.282582] usb 1-4: new high speed USB device using ehci_hcd and address 5 [19698.312600] usb 1-4: device descriptor read/8, error -110 [19703.442594] usb 1-4: device descriptor read/8, error -110 [19703.672548] usb 1-4: new high speed USB device using ehci_hcd and address 6 [19708.702581] usb 1-4: device descriptor read/8, error -110 [19713.840077] usb 1-4: device descriptor read/8, error -110 [19713.942555] hub 1-0:1.0: unable to enumerate USB device on port 4 [19714.262545] usb 4-2: new full speed USB device using uhci_hcd and address 2 [19729.382549] usb 4-2: device descriptor read/64, error -110 [19744.612534] usb 4-2: device descriptor read/64, error -110 [19744.842543] usb 4-2: new full speed USB device using uhci_hcd and address 3 [19759.962714] usb 4-2: device descriptor read/64, error -110 [19775.200047] usb 4-2: device descriptor read/64, error -110 [19775.422630] usb 4-2: new full speed USB device using uhci_hcd and address 4 [19780.444498] usb 4-2: device descriptor read/8, error -110 [19785.574491] usb 4-2: device descriptor read/8, error -110 [19785.802547] usb 4-2: new full speed USB device using uhci_hcd and address 5 [19790.824490] usb 4-2: device descriptor read/8, error -110 [19795.961473] usb 4-2: device descriptor read/8, error -110 [19796.062556] hub 4-0:1.0: unable to enumerate USB device on port 2

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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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  • Video and Audio Drift

    - by Cenoc
    Hey everyone, I was wondering, how much does recorded audio and video drift from their actual recording time usually? I'm recording both separately (into unsigned 8 bit PCM (44100 Hz) and raw image data files) and I was wondering how much I can expect each to drift. Thanks in advance!

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  • Audio Recording in C++

    - by Cenoc
    Hey, I was wondering, what was a good cross-platform utility for doing audio recording/ playback/ seeking in C++? I was thinking going the route of ALUT (OpenAL), but is there a better way? If not, do you guys know of any good tutorials/sample code for ALUT?

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  • Extract music files from a Audio CD [closed]

    - by Jatin
    Possible Duplicate: What good, free audio CD ripping/extraction tools exist for Windows, and supporting multiple formats? I have an audio cd, which has audio files with the file format as .cda ( CD Audio Track ). Each one of these files have a size of 1 KB each, and the rest of the CD has nothing else. Is there a way that I can get the audio files from the CD and then convert it into mp3 format and then play it in any other devices as I like.

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  • Multiple Audio I/P and O/P simultaneaously

    - by Raj Naveen
    hi (1) i saw in one of your posts that it is possible to get different outputs in windows 7. i am eager to know more. Is there any way i can create a 2 or more virtual cable between two softwares simultaneously. so that simultaneously, two or more audio inputs will be routed to equal no of audio analysers receivers, and then the audio analysers send back a filtered audio back to respective audio inputs... Please reply to email id: [email protected]

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  • Nyquist won't play audio

    - by erjiang
    I downloaded Nyquist, and am having trouble playing sounds from it. If I run it normally, I get: Nyquist -- A Language for Sound Synthesis and Composition Copyright (c) 1991,1992,1995 by Roger B. Dannenberg Version 2.29 > (play (osc 60)) Saving sound file to ./eric-temp.wav error: snd_save -- could not open audio output > If I wrap it by running padsp ny, the sound plays fine for about half a second, and then I get garbage fed to my speakers. Any solutions?

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  • Audio Stutters at gdm

    - by Allan
    Ok I have a problem every 2 times out of 3 I login (I cant be specific it fairly random) I get a Stuttering GDM warning (not the login sound just the Bell sound to wake you up) the only way to stop it is to login I have a Fujitsu Siemens Amilo 1718 with a 2gig of memory (only hardware mod) using 10.10 Maverick and I have disabled KMS as my system was freezing as per the release notes. The only time this has happened before on the same machine was when I gave Kubuntu a try when 10.04 came out then it happened at the login screen and at random times while listening to music in any program. By the way audio is fine as is almost everything else once I have logged in. I would like an answer to this as I am an advocate of Ubuntu and its kind of embarrassing when the first thing that happens is *bing*. as requested Daniel alsa-info Pulse verbose log Not sure how useful the pulse log will be as I cant replicate the bug with a terminal open but I wouldnt be asking the question if I knew the answer so..... Edit 24/12/2010 ......been living on cocktail sausages and pickled onions for five days now made a make shift splint with cocktail sticks..... oops so updated the alsa drivers but I still get the same message in the dmesg No response from codec, disabling MSI: last cmd=0x10a90000 googleing it brings up a forum post from some other distro with a green logo the only common denominator seems to be graphics ie ATI Radeon XPRESS 200M which is why I have had to turn of kms as the chip is so old that small mice try to eat the "kernel" ;) funnily enough following the bug link at the end of the post, I found a comment about "Ubuntu Black Magic" so mabey I am coming at this from the wrong angle...... Bad Joo Joo any one. I will try the second part of Daniels Fix and Update with the result. The final Edit: (Plays air guitar) In the end neither of these solved the problem as such However I have given Roland a tick for reminding me of the solution and I gave Daniel the Bounty for the effort in trying to solve the problem. The answer for future readers was the enable the correct HD Audio Model I found the answer back when using Karmic Koala 9.10 in this forum post Amilo Li1718 Skype - Can't get it working... the model is options snd-hda-intel model=3stack position_fix=1 enable=yes which can be added to the end of alsa-base.conf thanks all for helping and hope anyone with a similar problem will find the answer here.

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  • Synchronizing audio with scrolling text

    - by mr yoshida
    I am trying to have a website that vertically scrolls about 5 paragraphs of text with a matching audio file that reads along with it. It doesn't need to be synchronized word for word such as highlighting each spoken word but an accurate start and stop time. I've searched for quite a bit on the most efficient way of doing this but can't seem to find any answers. I tried Flash but really don't want to use it. Thanks in advance.

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  • Create Audio file on iPhone/iPad from many other audio files (mixer)

    - by Brian
    I am trying to create something similar like Piano app on the iPhone. When people tap a key, it play a piano note. Basically, there will have only 7 notes (C) at the moment. Each note is a .caf file and its length is 5 seconds. I do not know if there is any way to save the song user played and export to mp3/caf format? The AVAudioRecord seems only record from the microphone input. Many thanks

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  • Problems with MediaRecorder class setting audio source - setAudioSource() - unsupported parameter

    - by arakn0
    Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaRecorder class to record just audio from the microphone. I'm following the steps indicated in the official site: http://developer.android.com/reference/android/media/MediaRecorder.html So I have a method that initializes and configure the MediaRecorder object in order to start recording. Here you have the code: this.mr = new MediaRecorder(); this.mr.setAudioSource(MediaRecorder.AudioSource.MIC); this.mr.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); this.mr.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); this.mr.setOutputFile(this.path + this.fileName); try { this.mr.prepare(); } catch (IllegalStateException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } catch (IOException e) { Log.d("Syso", e.toString()); e.printStackTrace(); } When I execute this code in the simulator, thanks to logcat, I can see that the method setAudioSource(MediaRecorder.AudioSource.MIC) gives the next error (with the tag audio_ipunt) when it is called: ERROR/audio_input(34): unsupported parameter: x-pvmf/media-input-node/cap-config-interface;valtype=key_specific_value ERROR/audio_input(34): VerifyAndSetParameter failed And then when the method prepare() is called, I get the another error again: ERROR/PVOMXEncNode(34): PVMFOMXEncNode-Audio_AMRNB::DoPrepare(): Got Component OMX.PV.amrencnb handle If I start to record bycalling the method start()... I get lots of messages saying: AudioFlinger(34):RecordThread: buffer overflow Then...after stop and release,.... I can see that a file has been created, but it doesn't seem that it been well recorderd. Anway, if i try this in a real device I can record with no problems, but I CAN'T play what I just recorded. I gues that the key is in these errors that I've mentioned before. How can I fix them? Any suggestion or help?? Thanks in advanced!!

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  • How to play simultaneous multiply audio sources in Silverlight

    - by Shurup
    I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound).

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  • How to view the filter graph used by Windows Media Player

    - by Ian Boyd
    i have a video that i cannot render in Graph Edit: GSpot cannot render: and AVISynth's DirectShowSource cannot open: And yet Windows Media Player (12) can play it fine. How can i figure out the filters that Windows Media Player is using, when DirectShow itself cannot render the file? i tried running GraphEdit as an adminstrator and connecting to a remote graph, but Windows Media Player does not register its graph in the running objects table: Related question: How can i access a file in AviSynth that Windows Media Player can play, but DirectShow cannot?

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  • How to make Windows Media Player play only audio of video files - Windows 7

    - by blasteralfred
    Usually, for any video file, Windows Media Player plays video along with audio, unless user press mute, so that only video comes out. In such a way reverse, is it possible to make media player play videos in which audio may come out, not video, and media player may show a blank screen or default animation instead of video? I use Windows 7. Is it possible to play videos without displaying video in Windows Media Player 12

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  • Alternative to VLC player [closed]

    - by Shimmy
    I've been using VLC player but since the latest version (2.0.1) it's rendering all the Hebrew subtitles as question marks. I'm looking for a good free alternative that runs on Windows OS. I prefer a player with a huge dashboard and zillion preferences where all the possible options are customizable. Once ago I used MV2 player and loved that type of player that supports all the possible tweaks etc. etc.

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