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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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• 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 can i detect audio level of a stream or tcplistener or the audio card using C#

  - by schmoopy

  Is this possible using only C#? I want to be able to detect audio from: a stream, microphone or soundcard and begin recording if audio level is above a settable threshold. Thanks :-)

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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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• 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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• 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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• 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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• 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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• 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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• GWT audio recording

  - by oelewapperke

  Are there any options for recording audio in a GWT application ? Options seem to include flex, perhaps a java applet or some other form of flash ?

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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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• Multiple Audio listeners in Scene

  - by Kevin Jensen Petersen

  THIS IS UNITY Im trying to make a FPS game over networking, it works fine. But now, when im trying to implement sound, it won't work. My guess would be, to add a Audio listener to the prefab, that gets instansiated whenever a player connects to the server, however the problem about this is that each player's audiolistener have been switched out which the other player(s), so the AudioSource won't play at the player, but at someone else in the game. Any suggestions ?

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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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