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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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  • Finding out estimated duration of a stream using Core Audio

    - by Reflog
    I am streaming a MP3 over network using custom feeding code, not AVAudioPlayer (which only works with URLs) using APIs like AudioFileStreamOpen and etc. Is there any way to estimate a length of the stream? I know that I can get a 'elapsed' property using: if(AudioQueueGetCurrentTime(queue.audioQueue, NULL, &t, &b) < 0) return 0; return t.mSampleTime / dataFormat.mSampleRate; But what about total duration to create a progress bar? Is that possible?

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  • Seeking not working in HTML5 audio tag

    - by lord_wilmore
    I have a lighttpd server running locally. If I load a static file on the server (through an html5 audio tag), it plays and seeks fine. However, seeking doesn't work when running a dev server (web.py/CherryPy) or if I return the bytes via a defined action url instead of as a static file. It won't load the duration either. According to the "HTTP byte range requests" section in this Opera Page it's something to do with support for byte range requests/partial content responses. The content is treated as streaming instead. What I don't understand is: If the browser has the whole file downloaded surely it can display the duration, and surely it can seek. What I need to do on the web server to enable byte range requests (for non-static urls). Any advice would be most gratefully received.

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  • Streaming Audio over UDP to Android

    - by Mr. Pig
    Is it possible to have Android (perhaps via MediaPlayer or a different existing class) accept media streams over UDP? I've successfully had MediaPlayer connect to an HTTP stream (as well as static files hosted on an HTTP server) but I'm wondering how one would go about accepting a stream from a UDP source. I've seen this and suppose a solution similar to that (where I download the stream via an independent UDP socket and then move the data to a MemoryBuffer that I then pass to MediaPlayer) is an option but I'm curious if a method already exists in the SDK, and if it does not, what other options do I have? Thanks

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  • Secure Streaming CDN Video Content

    - by Donalds
    Hi, I am using Amazon CloudFront to stream paid video content to my users and I am having problems getting the videos secured. Wowza does that by creating a secure token, but the use of wowza would be much more costly. Is there anyway that I can better protect my content by using Cloudfront or other CDN? Thanks!

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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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  • 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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  • Skipping video and audio with PS3MediaServer

    - by MaxMackie
    I'm using the latest PS3MediaServer build right from the repos suggested in the Ubuntu Wiki. I'm streaming multiple movies from my server (Ubuntu 10.04 LTS) to my PS3 over wireless. Sometimes, during some movies, the audio and the video will begin skipping. This can last anywhere between 5 and 30 seconds before it goes back to normal. I have a four core i5 processor and 8GB of DDR3 RAM so I don't think my computer is having a hard time keeping up with the transcoding. So this leads me to believe it's either sub-optimal transcoding options from within PS3MS or my network can't handle the heat. Other than the out-of-box configuration, is there any way I can tweak the settings for the application to use my resources more efficiently?

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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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  • Looking for an application to record audio and video on a linux "embedded" device

    - by Luke404
    I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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  • Webhosting for a TV channel with streaming video

    - by Murtez
    Hi guys, I'm making a website for a web based TV channel, so I'm assuming it will be heavy on bandwidth usage, but I'm no good at calculating bandwidth. Couple of questions: Assuming the site streams HD video 24 / 7 to 1000 people, how much bandwidth is that? Where should something like this be hosted? The channel will have a fiber internet optic connection, but I don't know the limit on their bandwidth, would it be better to get their own server or host online? In either case in question 2, any recommendations? I'm usually a regular web designer for minor businesses, so this is a new level. Your help is appreciated.

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  • Internet TV streaming applications

    - by mark kirby
    Are there any programs for Ubuntu that can pull TV streams from the Internet ? For example like on a blu-ray player you get BBC iplayer and hulu and youtube as apps, so is there an application that can do this ? I know XMBC can but its to fiddly and resource hungry, I just want the net TV apps and not the full media center. Boxee was great for this, it was a media center but had an interface for the apps not just sticking them in some strange menu like XBMC. Please let me know of anything you know of

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  • Streaming video

    - by Bakhtiyor
    Dear all. During more than 2months I have a problem watching streaming video in my laptop only from my user account. If I log in with another user account I can easily watch streaming video without any problems, but with mine no streaming video :( I have windows 7 OS. What can cause this problem? Thank you all.

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