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  • Virtual microphone, networks and vb.net

    - by Jonathan
    I would like to add a virtual microphone (similar to how you can have a virual CD drive and then mount ISO files on it.) so that it can be selectable in programs like MSN and skype. But have the source of the audio be streamed from over a network(I know how to stream the audio over the network in VB.net) but how do I get that audio which has been streamed as the input to the virtual microphone? Jonathan

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  • Virtual microphone, networks and vb.net

    - by Jonathan
    I would like to add a virtual microphone (similar to how you can have a virual CD drive and then mount ISO files on it.) so that it can be selectable in programs like MSN and skype. But have the source of the audio be streamed from over a network(I know how to stream the audio over the network in VB.net) but how do I get that audio which has been streamed as the input to the virtual microphone? Jonathan

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  • Is Streaming Video possible with Sql Filestream?

    - by Lieven Cardoen
    We have stored all media in Sql Filestream, but now we'll need Video and Audio streaming... Will this be possible with Sql Filestream or will I have to take all of the Video and Audio out of the database? Which technology would you use to enable Video/Audio Streaming? WebORB FluorineFX Wowza (way better I think than the first two) IIS Media (haven't looked into this yet)

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  • Detecting the type of iPhone interrupt

    - by Prashant
    I can detect that the iPhone went to sleep and came back from sleep, by using the applicationWillResignActive and applicationDidBecomeActive. But how do I find out what kind of interrupt it was. I am making an audio player application, and need to keep the audio playing when the iPhone goes to sleep (which I know how to do). But I need to interrupt the audio when a message, alarm or low battery interrupt occurs. Also I need to resume the audio when the event is over. So how do I differentiate between these different interrupts.

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  • Processing file uploads before object is saved

    - by Dominic Rodger
    I've got a model like this: class Talk(BaseModel): title = models.CharField(max_length=200) mp3 = models.FileField(upload_to = u'talks/', max_length=200) seconds = models.IntegerField(blank = True, null = True) I want to validate before saving that the uploaded file is an MP3, like this: def is_mp3(path_to_file): from mutagen.mp3 import MP3 audio = MP3(path_to_file) return not audio.info.sketchy Once I'm sure I've got an MP3, I want to save the length of the talk in the seconds attribute, like this: audio = MP3(path_to_file) self.seconds = audio.info.length The problem is, before saving, the uploaded file doesn't have a path (see this ticket, closed as wontfix), so I can't process the MP3. I'd like to raise a nice validation error so that ModelForms can display a helpful error ("You idiot, you didn't upload an MP3" or something). Any idea how I can go about accessing the file before it's saved? p.s. If anyone knows a better way of validating files are MP3s I'm all ears - I also want to be able to mess around with ID3 data (set the artist, album, title and probably album art, so I need it to be processable by mutagen).

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  • How do you insert 9 MB file into a Blob Field Using Oracle.DataAccess?

    - by discwiz
    Trying to insert a large audio file into an Oracle 10g database and keep getting this error: ORA-01460: unimplemented or unreasonable conversion requested The byte array length of the audio file is 2702577. The procedure works with smaller array lengths, but not the larger ones. Here is my code and Thanks! Dim oracleConnection As New OracleClient.OracleConnection Dim Cmd As New OracleClient.OracleCommand Dim oracleDataAdapter As New OracleDataAdapter oracleConnection.ConnectionString = System.Configuration.ConfigurationManager.AppSettings("MasterConnectionODT") Cmd.Connection = oracleConnection Cmd.CommandText = "Audio.ADD_AUDIO" Cmd.CommandType = CommandType.StoredProcedure Dim aParam As New OracleClient.OracleParameter aParam.ParameterName = "I_FACILITY_ID_C" aParam.OracleType = OracleType.Char aParam.Value = FacID aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) aParam = New OracleParameter aParam.ParameterName = "I_TARP_ID_N" aParam.OracleType = OracleType.Number aParam.Value = TarpID aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) aParam = New OracleParameter aParam.ParameterName = "I_AUDIO_BLOB" aParam.OracleType = OracleType.Blob aParam.Value = Audio aParam.Direction = ParameterDirection.Input Cmd.Parameters.Add(aParam) Using oracleConnection oracleConnection.Open() Cmd.ExecuteNonQuery() End Using

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  • Echo certain value from smarty array

    - by zx
    Hi, So currently I have an array with smarty.. {foreach from=$_sequences key=k item=v} Name => {$v.menu} Type => {$v.type} Step => {$v.pri} Data =>{$v.data} {/foreach} which gives me Name = Test Type = Audio Step = 1 Data = audio1 Name = Test2 Type = Audio Step = 2 Data = audio2 Name = Test3 Type = Audio Step = 3 Data = audio3 Now how would I get the data for step = 2 to echo out? So from that foreach I only want to display "audio2"

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  • Javascript force GC collection? / Forcefully free object?

    - by plash
    I have a js function for playing any given sound using the Audio interface (creating a new instance for every call). This works quite well, until about the 32nd call (sometimes less). This issue is directly related to the release of the Audio instance. I know this because I've allowed time for the GC in Chromium to run and it will allow me to play another 32 or so sounds again. Here's an example of what I'm doing: <html><head> <script language="javascript"> function playSound(url) { snd = new Audio(url); snd.play(); delete snd; snd = null; } </script> </head> <body> <a href="#" onclick="playSound('blah.mp3');">Play sound</a> </body></html> I also have this, which works well for pages that have less than 32 playSound calls: var AudioPlayer = { cache: {}, play: function(url) { if (!AudioPlayer.cache[url]) AudioPlayer.cache[url] = new Audio(url); AudioPlayer.cache[url].play(); } }; But this will not work for what I want to do (dynamically replace a div with other content (from separate files), which have even more sounds on them - 1. memory usage would easily skyrocket, 2. many sounds will never play). I need a way to release the sound immediately. Is it possible to do this? I have found no free/close/unload method for the Audio interface. The pages will be viewed locally, so the constant loading of sounds is not a big factor at all (and most sounds are rather short).

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  • C# Thread Queue Synchronize

    - by ikurtz
    Greetings, I am trying to play some audio files without holding up the GUI. Below is a sample of the code: if (audio) { if (ThreadPool.QueueUserWorkItem(new WaitCallback(CoordinateProc), fireResult)) { } else { MessageBox.Show("false"); } } if (audio) { if (ThreadPool.QueueUserWorkItem(new WaitCallback(FireProc), fireResult)) { } else { MessageBox.Show("false"); } } if (audio) { if (ThreadPool.QueueUserWorkItem(new WaitCallback(HitProc), fireResult)) { } else { MessageBox.Show("false"); } } The situation is the samples are not being played in order. some play before the other and I need to fix this so the samples are played one after another in order. How do I implement this please? Thank you.

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  • Recursing data into a 2 dimensional array in PHP 5

    - by user315699
    I'm getting bamboozled by "for each" loops and two dimensional arrays, and I'm a php newb so please bear with me (and ignore any variables with the word "image" - it's all about the mp3s, I just didn't change it from the xml tutorial) I found a php function on the net that list files in a directory, the output of which is: Array ( [0] = audio/1.mp3 [1] = audio/2.mp3 [2] = audio/3.mp3 [3] = audio/4.mp3 [4] = audio/5.mp3 ) As expected. And another that lists some info about mp3 files. $mp3datafile = 'audio/1.mp3'; $m = new mp3file($mp3datafile); $mp3dataArray = $m-get_metadata(); print_r($mp3dataArray); unset($mp3dataArray); The output of which is Array ( [Filesize] = 31972 [Encoding] = CBR [etc] ) In order to automatically build RSS for a podcast, I need to generate XML for each item. So far so good. This is how I'm making the xml foreach ($imagearray as $key = $val) { $tnsoundfile = $xml_generator-addChild('item'); $tnsoundfile-addChild('title', $podcasttitle); $enclosure = $tnsoundfile-addChild('enclosure'); $enclosure-addAttribute('url', $val); // that's the filename $enclosure-addAttribute('length', $mp3dataArray[Filesize]); // << Length is file length, not time. But later I also need $mp3dataArray[Length mm:ss] for duration tag. $enclosure-addAttribute('type', 'audio/mpeg'); $tnsoundfile-addChild('guid', $path_to_image_dir.'/'.$val); } (The above has been truncated, I realise it's not proper xml right now, but it was just to show what was going on). Perfect. But I need to do it for as many files as there are in the directory. So, I have an array of the names of the files in the directory in $mp3data And, I have an array of mp3 data in $mp3dataArray from one iteration of the get_metadata() function. If I do the following, then I get a nice list of the mp3 data of the 5 files in the directory: foreach ($mp3data as $key = $val) { $mp3datafile = $val; $m = new mp3file($mp3datafile); $mp3dataArray = $m-get_metadata(); print_r($mp3dataArray); unset($mp3dataArray); } As expected. Where I'm struggling, and have been for most of the day in spite of reading many forums and tutorials, is how to populate the "second dimension" of the array, so that it goes through 1,2,3,4 and 5.mp3 (or however many there are), extracts the metadata, then allows me to use it in the xml section above. Here's what I have foreach ($mp3data as $key = $val) { $mp3datafile = $val; $m = new mp3file($mp3datafile); $mp3dataArray = $m-get_metadata(); $mp3testarray = array($mp3dataArray); } print_r($mp3dataArray); Shouldn't that line print_r($mp3dataArray); give me a nice list of 5 lots of mp3 data, in the way it did when I recursed through the loop as before? Cos this is driving me nuts! It must be something so simple, and any help would be greatly appreciated. Thank you in advance.

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  • how to use time out in mplayer?

    - by manoj
    I am trying to save audio using mplayer from a live http stream. saving audio is successful. If there is no live stream playing it does not exit automatically. Is there any way to set timeout if there is no live stream? code : mplayer -i url -t 00:00:10 -acodec libmp3lame -ab 24 -ar 8000 audio.mp3 Thanks in advance.

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  • Multiple jQuery UI buttons with same ID

    - by Mark Nolan
    There might be a really simple solution to this question, but i just don't see it! I have a page with a list of items. every item has the same jquery ui button (its inside a dialog and adds that item to a list). i identify the item via the parenting DIV holding the DB id. So far so good... The Problem is only the first button on the list works! The second, third etc. buttons don't show any reaction at all. The buttons all have the same id - as the list is dynamic and the same action is triggered with every click. Only the parenting ID changes. Heres the display part: <div id="2"> <div id="56"> <button id="add-audio-file" class="ui-button ui-state-default ui-corner-all">betty_2.mp3</button> </div> </div> <div id="2"> <div id="57"> <button id="add-audio-file" class="ui-button ui-state-default ui-corner-all">betty_3.mp3</button> </div> </div> And here comes the js Part: $('#add-audio-file').click(function() { assetID = $(this).parent('div').attr('id'); pageID = $(this).parent('div').parent('div').attr('id'); $.post( "modules/portfolio/serialize.php", {id : pageID, assetid : assetID, do : "add-audio-file"}, function(data, textStatus, xhr) { $('#dialog-add-audio').dialog('close'); } ); }); I am using jquery 1.4.2 and jquery ui 1.8rc3 Any ideas?

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  • Why does Silverlight provides webcam and microphone support without any encoding API?

    - by Shurup
    In the list of new features in Silverlight 4 you will find following: Webcam and microphone to allow sharing of video and audio for instance for chat or customer service applications. Silverlight captures an audio stream as raw pcm. So how would you realize for example audio/video chat or client/server audio recording application without any encoding on the client side, where there is no APIs in Silverlight available? Much less in a Silverlight you cannot use an unmanaged dll. You can use a com automation (a new feature of the Silverlight 4, I think only for Windows) but only if it was already installed on the client side (do you know any encoding COM servers that are installed with the windows). Otherwise, how would you deploy a custom COM server within you Silverlight application? The only way I found is either to deploy a command-line encoding and use it with COM AutomationFactory.CreateObject("WScript.Shell") or to implement an encoding to use it in your own AudioSink.

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  • iPhone SDK: How to get mic volume

    - by TheGambler
    I want to get the volume or even how much noise is coming through the mic. So someone is talking or some noise is going on in the background I want to know how much. Which framework would I use: Audio Toolbox, Audio Unit, AV Foundation, and Core Audio

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  • android playback suddenly stopping...

    - by user306517
    I have an app that is streaming audio content and sometimes it just stops all of the suddent. the logcat windows shows -- AudioHardware pcm playback is going to standby and that's it. I saw on another thread (pun intended) that someone was saying it was because he was using too many threads. Could that really be causing this? Could i give the audio thread higher priority? Anyway to prevent the audio hardware pcm from going to standby?

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  • MPMoviePlayerController playback problem

    - by Fahri
    Hello all, I'm having trouble with playing audio stream using MPMoviePlayerController in the background. On the foreground it's playing the audio just fine, but when the application goes to the background, sound disappears and the time keeps going. I have initialized the AVAudioSession with AVAudioSessionCategoryPlayback and have put the required background modes to audio in info.plist file. Also I have set player.useApplicationAudioSession = YES;. What could be the possible problem here? Thanks in advance :)

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  • Media Information Extractor for Java

    - by eyazici
    I need a media information extraction library (pure Java or JNI wrapper) that can handle common media formats. I primarily use it for video files and I need at least these information: Video length (Runtime) Video bitrate Video framerate Video format and codec Video size (width X height) Audio channels Audio format Audio bitrate and sampling rate There are several libraries and tools around but I couldn't find for Java.

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  • AS3: creating a class with multiple and optional parameters?

    - by redconservatory
    I'm creating a slideshow where each slide can have: - a video or a still - 1 audio track or many (up to 3) - 1 button or many (up to 3) I was thinking that each slide can be it's own object, and then I would pass the video, audio, buttons, etc., into it as parameters: package { import flash.media.Video; public class Section { public function Section (video:Video, still:myPhotoClass, audiotrack:Sound, button:myButtonClass) { // can have video OR a still // can have 1 audio track or several // can have 1 button or more } } I'm not sure how to go about approaching this since there can be multiples of certain items (audio, buttons) and also two items are sort-of-optional in the sense that there can be ONE or the OTHER (video/still). For example, is this something that I should just avoid passing as parameters altogether, using a different approach (getters/setters, maybe)?

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  • Translate this code to objective-c iPhone?

    - by Silent
    Hi there would somone know how to translate this cose into iphone programming it seems like its an http message the data i would be sending is audio data im not sure which type of file. Any things helps thanks. /////////////////////////////////////////////////////////// Using the cgi command fifo.cgi will enable the IP camera to start receiving audio data User can use Microsoft WinHTTP C/C++ API to upload the audio file   http://msdn.microsoft.com/en-us/library/aa384252%28v=VS.85%29.aspx   1. Establish connection     hSession = WinHttpOpen(L"WinHTTP Example/1.0",WINHTTP_ACCESS_TYPE_DEFAULT_PROXY,WINHTTP_NO_PROXY_NAME,WINHTTP_NO_PROXY_BYPASS, 0 );        if(hSession)      {          USES_CONVERSION;          hConnect = WinHttpConnect(hSession,A2W(m_cAddr), m_iPort,0);      }   2. Establish listen request        if(hConnect)          hRequest = WinHttpOpenRequest(hConnect,L"POST",L"/cgi-bin/fifo.cgi",NULL,WINHTTP_NO_REFERER,WINHTTP_DEFAULT_ACCEPT_TYPES,0);        if(hRequest)          bResults = WinHttpSendRequest(hRequest,WINHTTP_NO_ADDITIONAL_HEADERS,0,WINHTTP_NO_REQUEST_DATA,0,uDataLength,0);   Send audio data        if( hRequest)          WinHttpWriteData(hRequest, pData, nDataSize, &dwBytesWritten); //////////////////////////////////////////////////////////////////////////

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  • Core Data Predicates with Subclassed NSManagedObjects

    - by coneybeare
    I have an AUDIO class. This audio has a SOUND_A subclass and a SOUND_B subclass. This is all done correctly and is working fine. I have another model, lets call it PLAYLIST_JOIN, and this can contain (in the real world) SOUND_A's and SOUND_B's, so we give it a relationship of AUDIO and PLAYLIST. This all works in the app. The problem I am having now is querying the PLAYLIST_JOIN table with an NSPredicate. What I want to do is find an exact PLAYLIST_JOIN item by giving it 2 keys in the predicate sound_a._sound_a_id = %@ && playlist.playlist_id = %@ and sound_b.sound_b_id = %@ && playlist.playlist_id = %@ The main problem is that because the table does not store sound_a and sound_b, but stored audio, I cannot use this syntax. I do not have the option of reorganizing the sound_a and sound_b to use the same _id attribute name, so how do I do this? Can I pass a method to the predicate? something like this: [audio getID] = %@ && playlist_id = %@

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  • How to read using C++ (C#) sound stream sent by flash?

    - by Oleg
    Hello. I need to read sound stream sent by flash audio in my C++ application (C++ is not a real limitation, it may be C# or any other desktop language). Now flash app sends audio to another flash app but I need to receive the same audio by desktop application. So, is there a standard or best way how to do it? Thank you for your answers.

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  • How to control wether or not my app will stop any iPod music playing?

    - by mystify
    When my app launches, there are rumors that I can decide if any currently playing music is stopped or not. My goal is to use Audio Queue Services, because I believe that's the most powerful audio technology in iPhone OS. So, could I really decide that when my app launches? How? Which one of the many audio technologies on the iPhone OS is responsible for managing this?

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  • Get frames from the video saved in the photo directory iphone?

    - by Ballu
    HI All. how can i get the frames of any video that i am fetching from the photo directory iphone .I checked out various links but they only provide any thumbnails or any particular TIME's frame image but i need the frames of full length video .And the respective audio too then is that possible that from the freames i got and the audio file i can create another video but will change the sequence of the frames only the audio file will be the same in the video.Is that feasible .? Sample code or link will be more helpful !!! Thanks, Balraj,

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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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