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  • UDP multicast streaming of media content over WIFI

    - by sajad
    I am using vlc to stream media content over wireless network in scenario like this (from content streamer to stream receiver client): The bandwidth of wireless network is 54 Mb/s and UDP stream's required bandwidth is only 4 Mb/s; however there is trouble in receiving media stream and quality of playing specifically in multicast mode; means I can play the stream but it has jitter and does not play smoothly. In uni-cast I can stream up to 5 media streams correctly, but in multicast mode there is problem with streaming just one media! However when I stream from client some multicast streams; the wifi access-point can receive data correctly and I can see the video in "udp streamer" side correctly even when number of multicast streams increases to 9; But as you see I want to stream from streaming server and receive media in client size. Is this a typical problem of streaming real-time contents over wireless networks? Is it necessary to change configurations of my WIFI switch or it is just a software trouble? thank you

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  • Convert audio file to FLAC with ffmpeg?

    - by elpsk
    can I convert one of this format to compatible 16000.0 Sample Rate FLAC file? kAudioFormatLinearPCM = 'lpcm', kAudioFormatAppleIMA4 = 'ima4', kAudioFormatMPEG4AAC = 'aac ', kAudioFormatMACE3 = 'MAC3', kAudioFormatMACE6 = 'MAC6', kAudioFormatULaw = 'ulaw', kAudioFormatALaw = 'alaw', kAudioFormatMPEGLayer1 = '.mp1', kAudioFormatMPEGLayer2 = '.mp2', kAudioFormatMPEGLayer3 = '.mp3', kAudioFormatAppleLossless = 'alac' I tried using ffmpeg ffmpeg -i audio.xxx -acodec flac audio.flac but result is FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard Mac OSX universal build for ffmpegX configuration: --enable-memalign-hack --enable-mp3lame --enable-gpl --disable-vhook --disable-ffplay --disable-ffserver --enable-a52 --enable-xvid --enable-faac --enable-faad --enable-amr_nb --enable-amr_wb --enable-pthreads --enable-x264 libavutil version: 49.0.0 libavcodec version: 51.9.0 libavformat version: 50.4.0 built on Apr 15 2006 04:58:19, gcc: 4.0.1 (Apple Computer, Inc. build 5250) Input #0, wsaud, from 'audio.alac': Duration: 00:00:03.8, start: 0.000000, bitrate: 199 kb/s Stream #0.0: Audio: adpcm_ima_ws, 24931 Hz, stereo, 199 kb/s Unable for find a suitable output format for 'audio.flac' I also installed flac codec for mac, but nothing... I tried also use convtoflac.sh (from http://legroom.net/software/convtoflac) but result is similar. Any idea to convert in flac?

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  • Back up of Streaming server

    - by Maxwell
    I want to take a new streaming server for my website which generally holds videos and audio files. But how do we maintain backup of the streaming server if storage size is increasing day by day. Generally on Database servers, like Sql Server, backups can be easily taken and restored very easily as they do not occupy much space for medium range applications. On the other hand how can we take backup of streaming server? If the server fails, the there should be an alternative server / solution that should decrease downtime of the server. How is the back-end architecture of YouTube built to handle this?

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  • Streaming video file to iPhone

    - by user34157
    I have a http streaming link which gives me .flv streaming feed. I want to convert that and access in my iPhone program. How can i do that? I want to have a desktop software like VLC and input this streaming feed URL and convert to iPhone supported and stream again to iPhone. I tried VLC with H.264 and Mpeg-1 audio, but seems to be it doesn't give the supported format, so as iPhone program doesn't play the video. Could someone please guide me how can i setup a desktop software which can stream iPhone supported file?

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  • VLC RTP Streaming in FC12

    - by Matt D
    I'm trying to get VLC to work streaming RTP audio/video over my office network. The goal is multicast a/v streaming. In all test cases, we are streaming from VLC to VLC. I am able to stream from Windows to Windows, and from Fedora to Windows, but not from Windows to Fedora. Additionally, I am unable to receive a LOCAL stream from one instance of VLC to another, within Fedora. I don't see any reason why this would be. The buffer indicator (where the elapsed/total time is normally displayed) never shows any connectivity, so it would appear to be a network problem, but since I am able to stream from Fedora to Windows (same IP, same port) I thought it would be something else. Does anyone know of a solution to this issue?

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  • UDP multicast streaming of media content over WIFI

    - by sajad
    I am using vlc to stream media content over wireless network in scenario like this (from content streamer to stream receiver client): The bandwidth of wireless network is 54 Mb/s and UDP stream's required bandwidth is only 4 Mb/s; however there is trouble in receiving media stream and quality of playing specifically in multicast mode; means I can play the stream but it has jitter and does not play smoothly. In uni-cast I can stream up to 5 media streams correctly, but in multicast mode there is problem with streaming just one media! However when I stream from client some multicast streams; the wifi access-point can receive data correctly and I can see the video in "udp streamer" side correctly even when number of multicast streams increases to 9; But as you see I want to stream from streaming server and receive media in client size. Is this a typical problem of streaming real-time contents over wireless networks? Is it necessary to change configurations of my WIFI switch or it is just a software trouble? thank you

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  • Change the audio output device in Firefox

    - by Zanami Zani
    I'm trying to play music through Ventrilo and currently I use Virtual Audio Cable. The way it works is that in foobar2000 (a music playing program) I set the output device in preferences to Virtual Audio Cable. Then in Ventrilo I log in to another name and set the input device to Virtual Audio Cable. This routes the music through the Virtual Audio Cable and allows me to play the music through Ventrilo. However, I would also like to change the output device for Firefox (or any other browser) or "Plugin Container for Firefix" to Virtual Audio Cable so that I could play music from Pandora or YouTube on to Ventrilo. Unfortunately I could not find an option for this anywhere.

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  • Record the audio stream from HDMI monitor

    - by Nick
    I am trying to record sound playing on my comupter with Audacity but am running into some troubles. I have the stereo mix set to be the default audio recorder but it doesn't pick up the audio that is being played through my HDMI monitors speakers: Playback Recording When I plug in headphones the stereo mix will pick up the audio stream and I can record but not when playing through the HDMI. I have installed the latest audio drivers and have tried all the different record options to no avail. How can I capture the Audio stream going through the HDMI?

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  • HTTP Live Streaming Broadcast

    - by user761389
    I'm designing an app for streaming video from a device (e.g. iPhone) via a server to one or more devices and have been researching Apples HTTP Live Streaming protocol. One thing that isn't clear is whether it is possible to stream live video (with audio) to the server and then have it streamed simultaneously in real time to the client devices. From reading the documentation and technical notes from Apple it seems like the index file needs to be created before the segmented video files can be served to a client. Is this right? If so maybe HTTP Live Streaming isn't suitable in this case, what other technologies or software should I consider? Thanks

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  • Dynamic audio score/music

    - by Joel Martinez
    I'm interested in developing a game who's background music changes with the mood and scenario of the game's action. Of course many existing games do this (halo for example), but I was interested in any resources/papers/articles talking about the techniques to develop a system like this. I have some ideas, and I understand that this will be equally challenging to implement at the code level as it will be to come up or acquire music that fits this model. Any links or, answers with ideas in them would he appreciated. Edit: this is the kind of info I'm looking for :) http://halo.bungie.org/misc/gdc.2002.music/

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  • How can I listen to FM Radio streaming without connecting to the internet?

    - by Vicheanak
    Many phones has the functionality to listen to FM radio without connecting to the internet. Just wondering that how can I do this on a computer? Please give me some advices, thanks a lot. Sorry guy I understand this question is not programming related. For a computer I mean a notebook which already has the wireless and Bluetooth, and these two combination shall be able for a notebook to perform this functionality.

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  • An equivalent of IceCast but for Live Video Streaming ?

    - by Kedare
    Hello, I am looking for a solution to Stream live video like that : A camera/webcam/video output ---> Stream server ---> Clients And if possible multiple Stream Servers like this (like IceCast): A camera/webcam/video output --> Master Stream server +---> Slave Stream Server ---> Clients | `--> Clients | `--> Slave Stream Server ---> Clients `--> Clients The clients will be in flash, so I think RTMP should be a good protocol, I've heard of Red5, is it good for that ? Does it scale ? I would like to get statistics (Amount of clients, Bandwidth, etc), is it possible with red5 ? Do you know any other good solution to do that ? (Only free and if possible Open Source) Thank you !

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  • Downmix ALL SYSTEM audio to mono - Windows 7

    - by Mike K.
    I'm deaf in one ear and want to use my headphones when playing a game and talking with my friends on Skype/TS/Mumble/etc while also sometimes listening to music. I need ALL my system audio to be downmixed to mono so that my ONE hearing ear gets ALL audio channels instead of split stereo audio. No, none of the other similar questions on superuser have a solution. My headphone properties does not have a 'Mono' option, I don't have a 'Headphone Virtualization' option, and my Realtek HD audio driver software doesn't have these options either (driver was updated 11/14/2012). Don't even talk about setting the balance of one side of the headphones to 0. You're not paying attention if you suggest that. JACK and Virtual Audio Cable didn't work. It's possible I configured them wrong, but I followed the steps I found in related questions and still got split stereo out. TL;DR I need a viable, working, software solution (I say software because I have a USB headset) for forcing ALL system audio to mono so that I can hear literally everything through the one earpiece. Thanks!

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  • Audio card with built-in ground isolator?

    - by Dave Jarvis
    What audio cards would you recommend that eliminate hum, and hard-drive & mouse movement signal interference? Hardware components: Motherboard. Asus P5Q SE Audio. Realtek ALC 1200, 8-Channel High-Definition Audio CODEC (on board) Harddrive. WD Caviar 320 GB Mouse. Logitech Marbleman USB Mixer. Mackie d.4 Pro Amplifier. Sonance Sonamp 260 All components are plugged into the same Monster Power HDP 910 powerbar (does not help eliminate noise). I have no other components plugged in. The computer uses a Monster iCable 1000 to go from mini (on board audio) to RCA (mixer). I have moved the cable as far from other cables as possible. A ground loop isolator between the mixer and on board audio eliminates all noise. I would rather not use a ground loop isolator; an internal audio card that is Linux-compatible (Kubuntu) would be ideal. Suggestions?

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  • What's the best way to test a P2P live streaming app?

    - by hbt
    Hey guys, I've been working on a P2P live streaming app and I'm having some trouble testing it properly. At the moment, I'm testing it using: 1) Another laptop + an external server 2) Multiple instances running on different ports Problem is: this is not exactly ready for production. Is there something like a simulator OR any of you guys worked on a torrent client, p2p client, live streaming solution and had to test it? Please let me know, Thanks, -hbt

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  • FFSERVER - streaming an ASF video as Webm output

    - by Emmanuel Brunet
    I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Environment Debian 7.5 ffmpeg 2.2 Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://192.168.1.62:8091/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://192.168.1.62:8091/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream.

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  • ffserver-2.2 - streaming an ASF video as Webm output with ffserver on Debian 7.5

    - by Emmanuel Brunet
    I'm trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump. Input stream $ ffprobe http://account:password@webcam/videostream.asf Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s ffserver configuration my ffserver configuration is : Port 8091 RTSPPort 554 BindAddress 192.168.1.62 MaxHTTPConnections 1000 MaxClients 100 MaxBandwidth 1000 CustomLog - <Feed webcam.ffm> File /tmp/webcam.ffm FileMaxSize 500M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <Stream webcam.webm> # Output stream URL definition Feed webcam.ffm # Feed from which to receive video Format webm # Audio settings AudioCodec vorbis AudioBitRate 64 # Audio bitrate # Video settings VideoCodec libvpx VideoSize 640x480 # Video resolution VideoFrameRate 25 # Video FPS AVOptionVideo flags +global_header # Parameters passed to encoder # (same as ffmpeg command-line parameters) AVOptionVideo cpu-used 0 AVOptionVideo qmin 10 AVOptionVideo qmax 42 AVOptionVideo quality good AVOptionAudio flags +global_header PreRoll 15 StartSendOnKey # VideoBitRate 32 # Video bitrate </Stream> <Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> ffmpeg feed I run the following command that fails $ ffmpeg -i http://account:password@webcam/videostream.asf http://ffserver_ip:port/webcam.ffm http://192.168.1.62:8091/webcam.ffm Input #0, asf, from 'http://account:password@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly Segmentation fault I tryed $ ffmpeg -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p http://ffserver_ip:port/webcam.ffm But it raises the same error. Thanks for your help Edit For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too. And thus with mirrored encoding so I changed the ffserver configuration to ... <Stream webcam.asf> Feed webcam.ffm Format asf VideoFrameRate 25 VideoSize 640X480 VideoBitRate 256 VideoBufferSize 1000 VideoGopSize 30 AudioBitRate 32 StartSendOnKey </Stream> ... And the output is now : Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf': Duration: N/A, start: 0.000000, bitrate: 32 kb/s Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm': Metadata: creation_time : now encoder : Lavf55.40.100 Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s Metadata: encoder : Lavc55.64.100 wmav2 Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc Metadata: Stream mapping: Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2) Stream #0:0 -> #0:1 (mjpeg -> msmpeg4) Press [q] to stop, [?] for help Segmentation fault I can't even forward the stream. Thanks for your help again.

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  • How to monitor streaming servers

    - by pcdinh
    Hi all, I have had a bunch of Linux based streaming servers that employed lighttpd web server to provide video streaming via port 80. Recently, our service is very slow. Therefore, I would like to ask if there is a good software package that helps us monitor and record our bandwidth usage, lighttpd established connections, TCP sync connections, disk I/O ... over time. Any suggestions? Regards, Dinh

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  • Play audio file data - Spring MVC

    - by Vijay Veeraraghavan
    In my web-application, I have various audio clips uploaded by the users in the database stored in the BLOB column. The audio files are low bit rate WAV files. The clips are secured, one can see only those clips he has uploaded. Instead of user downloading the clip and playing it in his player, I need it be steamed online in the web page itself. In the jsp I use the <audio> tag with the source mapping to the controller mappping url. <td> <audio controls><source src="recfile/${au.id}" type="audio/mpeg" /></audio> </td> Where, the recfile is the request mapping and the au.id is the audio id. In the controller I process the request like below @RequestMapping(value = "/recfile/{id}", method = RequestMethod.GET, produces = { MediaType.APPLICATION_OCTET_STREAM_VALUE }) public HttpEntity<byte[]> downloadRecipientFile(@PathVariable("id") int id, ModelMap model, HttpServletResponse response) throws IOException, ServletException { LOGGER.debug("[GroupListController downloadRecipientFile]"); VoiceAudioLibrary dGroup = audioClipService.findAudioClip(id); if (dGroup == null || dGroup.getAudioData() == null || dGroup.getAudioData().length <= 0) { throw new ServletException("No clip found/clip has not data, id=" + id); } HttpHeaders header = new HttpHeaders(); I tried this too //header.setContentType(new MediaType("audio", "mp3")); header.setContentType(new MediaType("audio", "vnd.wave"); header.setContentLength(dGroup.getAudioData().length); return new HttpEntity<byte[]>(dGroup.getAudioData(), header); } When the jsp loads, the controller get the request, it serves back the audio data fetched from the database, the jsp too shows the player with the controls. But when I play it nothing happens. Why is it? Am I missing anything in the configuration? Am I doing it right?

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  • iPhone, JQTouch and HTML5 audio tags

    - by Moo
    I am having an issue with JQTouch (latest beta) and html5 audio tags on 'sub pages' - the audio tag works before any page transitions are done, and cease to work afterward. For example: http://richardprice.dyndns.ws/test.html and http://richardprice.dyndns.ws/test2.html are identical other than I swap the "current" class between the two divs - all the audio tags play the same mp3. On test.html the audio tag on the initial page works, but when you switch to Page 2 the audio tag on that page does not (and sometimes results in a browser crash). Switch back to Page 1 and the audio tag on that page has ceased to work. test2.html is the same test but with the initial pages reversed, and the same thing happens - Page 2 (now the initial page) plays the audio, Page 1 does not, and switching back to Page 2 results in the audio no longer working. Thoughts?

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  • <audio> element autobuffers no matter what

    - by pthulin
    I'm trying to make a web based media player using the HTML5 audio element implemented in Firefox 3.5 and Chrome. Reading Mozillas documentation, omitting the autobuffer attribute should result in the audio src not being requested: if specified, the audio will automatically begin being downloaded, even if not set to automatically play. This continues until the media cache is full, or the entire audio file has been downloaded, whichever comes first However, on the server side I notice the files are being requested anyway. My sample page is very simple: <html> <body> <audio src="1.ogg"></audio> <audio src="2.ogg"></audio> </body> </html>

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  • Modify audio pitch of recorded clip (m4v)

    - by devcube
    I'm writing an app in which I'm trying to change the pitch of the audio when I'm recording a movie (.m4v). Or by modifying the audio pitch of the movie afterwards. I want the end result to be a movie (.m4v) that has the original length (i.e. same visual as original) but with modified sound pitch, e.g. a "chipmunk voice". A realtime conversion is to prefer if possible. I've read alot about changing audio pitch in iOS but most examples focus on playback, i.e. playing the sound with a different pitch. In my app I'm recording a movie (.m4v / AVFileTypeQuickTimeMovie) and saving it using standard AVAssetWriter. When saving the movie I have access to the following elements where I've tried to manipulate the audio (e.g. modify the pitch): audio buffer (CMSampleBufferRef) audio input writer (AVAssetWriterAudioInput) audio input writer options (e.g. AVNumberOfChannelsKey, AVSampleRateKey, AVChannelLayoutKey) asset writer (AVAssetWriter) I've tried to hook into the above objects to modify the audio pitch, but without success. I've also tried with Dirac as described here: Real Time Pitch Change In iPhone Using Dirac And OpenAL with AL_PITCH as described here: Piping output from OpenAL into a buffer And the "BASS" library from un4seen: Change Pitch/Tempo In Realtime I haven't found success with any of the above libs, most likely because I don't really know how to use them, and where to hook them into the audio saving code. There seems to be alot of librarys that have similar effects but focuses on playback or custom recording code. I want to manipulate the audio stream I've already got (AVAssetWriterAudioInput) or modify the saved movie clip (.m4v). I want the video to be unmodifed visually, i.e. played at the same speed. But I want the audio to go faster (like a chipmunk) or slower (like a ... monster? :)). Do you have any suggestions how I can modify the pitch in either real time (when recording the movie) or afterwards by converting the entire movie (.m4v file)? Should I look further into Dirac, OpenAL, SoundTouch, BASS or some other library? I want to be able to share the movie to others with modified audio, that's the reason I can't rely on modifying the pitch for playback only. Any help is appreciated, thanks!

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