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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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  • 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 use the Bass library to convert audio to another format (using C#)

    - by Siemsen
    Good day everyone and happy holidays. I'm trying to create a "simple" program that allows me to create a list of video files (Youtube video's to be precise) that are then run through by my program and converted any given format to pure MP3 audio. For this purpose I'm trying to use the BASS.dll and well it isn't going so well. I was wondering if anyone has used BASS to convert from one audio format to another? Or is there another library better suited for this?

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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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  • getAudioInputStream can not convert [stereo, 4 bytes/frame] stream to [mono, 2 bytes/frame]

    - by brian_d
    Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes/frame, little-endian Is it possible to convert this stream manually after every read() call? And if yes, how? In general, how can you compare two formats and tell if a conversion is possible?

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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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  • 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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  • SQL Authority News – Download SQL Server Data Type Conversion Chart

    - by pinaldave
    Datatypes are very important concepts of SQL Server and there are quite often need to convert them from one datatypes to another datatype. I have seen that deveoper often get confused when they have to convert the datatype. There are two important concept when it is about datatype conversion. Implicit Conversion: Implicit conversions are those conversions that occur without specifying either the CAST or CONVERT function. Explicit Conversions: Explicit conversions are those conversions that require the CAST or CONVERT function to be specified. What it means is that if you are trying to convert value from datetime2 to time or from tinyint to int, SQL Server will automatically convert (implicit conversation) for you. However, if you are attempting to convert timestamp to smalldatetime or datetime to int you will need to explicitely convert them using either CAST or CONVERT function as well appropriate parameters. Let us see a quick example of Implict Conversion and Explict Conversion. Implicit Conversion: Explicit Conversion: You can see from above example that how we need both of the types of conversion in different situation. There are so many different datatypes and it is humanly impossible to know which datatype require implicit and which require explicit conversion. Additionally there are cases when the conversion is not possible as well. Microsoft have published a chart where the grid displays various conversion possibilities as well a quick guide. Download SQL Server Data Type Conversion Chart Reference: Pinal Dave (http://blog.sqlauthority.com) Filed under: PostADay, SQL, SQL Authority, SQL Download, SQL Query, SQL Server, SQL Tips and Tricks, T SQL, Technology

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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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