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  • How to use RTPSocket to send RTP packets

    - by Afro Genius
    Hi there, am relatively new to JMF but have gone through the documents and have a sufficient understanding of how it works. That been said am having some trouble implementing a the server side for RTPSockets. After looking at their illustrations and example. I am still abit confused. Am I to develop a datasource and also datasink classes to handle the transfer? What am trying to do is stream data from my application to the underlying network and receive it back through another application. I have and understand receiving but just can't get my head around the steps involved for sending. Any help would be most appreciated.

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  • how to create thumbnail system for MP4 files

    - by Pete Herbert Penito
    Hi everyone! I know I know, why am I using MP4 still?? It's because I have like 100 files already in this format and I need to upload to a website, I have the mp4 file embeded in the site already and the file played changes according to php. but what I really need is a way to dynamically create a thumbnail or take a snapshot of the video file to display on the page. I've read a couple things online but they all require the file type to be in FLV, what would be the best way to accomplish this? Thank you Guys!

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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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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  • ShoutCast over SSL

    - by Honus Wagner
    So I've gone ahead and set up my ShoutCast server DNAS and set my DSP in Winamp on my host computer. The server listens on port 8000, so per some instructions I installed an output plugin for winamp (Shoutcast DSP) and used 8000 and the password to connect. Server accepts the connection. Now, what the heck do I do now? My host computer is SSL secured and the DNAS server is installed within the secure web directory (if that matters). My desired end result is that I want to listen to my ShoutCast setup at home (host computer) from any computer. I try browsing to my ip address and port 8000 (without using HTTPS) and it comes back with nothing. If I browse with HTTPS://my.server.com:8000, I get Error code: ssl_error_rx_record_too_long) Have I completely missed something, or am I just a total moron? Thanks.

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  • Dimdim Change name

    - by islam
    i build dimdim v4.5 on my pc and its work fine with me. each time i want to start meeting i type my pc IP address like this : http://<my-ip-address>/dimdim i want to change the word dimdim to be anything else like : http://<my-ip-address>/meeting regards

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  • Steaming a non-PCM WAV file to a SilverLight application

    - by Satumba
    Hi, I would like to allow users to play recorded WAV files that stored on a server back to a Silverlight application as a client to play them. I saw that there is a way to play a WAV file on Silverlight (here), but when i tried to impliment it, i got an error playing the file because it is not in PCM format but encoded. The files that i'm trying to play are encoded with a special encoder, so i thought that the only way is to decode the WAV file on the server and stream it back to the client. The limitation is that the decode process should occur in real time because it is not reasonable to convert all the WAV files that exists. Is it possible to do it? Which streamer can i use? (Windows Media Service can help here?) Does somebody has any experience with such a scenario? Appreciate your help.

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  • Blackberry buffered playback demo??

    - by Bohemian
    Can someone help me to buffer a mp3 file on a server using the Blackberry buffered pllayback demo app provided with the jde? I hav loaded it in the simulator. And my mds is started but I m unable to play the audio. There is no error but it doesnt play/load. The code looks all fine. Thanks

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  • Wireshark doesnt' recognises RTMP streams

    - by Andrew
    Hello! I found on the web few samples on tracking RTMP (Real Time Messaging Protocol) with Wireshark, but it doesn't work for me. All RTMPT packets rendered as basic TCP packet like this: 149 14.324999 85.115.xxx.xxx 192.168.1.20 TCP macromedia-fcs > 54557 [ACK] Seq=1 Ack=1452 Win=69 Len=0 I'm using Wireshark 1.2.8 with all protocols installed on Windows Vista. What can i do to fix it? Thx!

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  • Stream (.NET) handling best-practices

    - by Jader Dias
    The question is entitled with the word "Stream" because the question below is a concrete example of a more generic doubt I have about Streams: I have a problem that accepts two solutions and I want to know the best one: I download a file, save it to disk (2 min), read it and write the contents to the DB (+ 2 min). I download a file and write the contents directly to the DB (3 min). If the write to DB fails I'll have to download again in the second case, but not in the first case. Which is best? Which would you use?

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  • Recording Audio through RTMP/Rails

    - by Lowgain
    I am in the process of building a rails/flex application which requires audio to be recorded and then stored in our amazon s3 account. I have found no alternative to using some form of RTMP server for recording audio through flash, but our hosting environment will not allow us to install anything like FMS, Red5, etc. Is there any existing Ruby/Rails RTMP solution that will allow audio recording? If not, is it possible for Rails to at least intercept the RTMP stream and then I can hope to reference red5 or something for parsing the data (long shot, I know)? The other alternative I can think of is hosting a red5 server on another host and communicating with our rails app once the saving/uploading is done, which is not preferred. Am I going to have any luck here?

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  • Trying to build automatic audio-conferencing capability into a WebApp

    - by Keller
    Hey all, I'm working with a team of relatively novice programmers, and we are trying to create a site that will have audio-conferencing capabilities such that whenever someone visits the page, they will immediately have audio-conferencing capabilities with everyone else on the page (5 people max). Can anyone point us in a general direction? Should we be looking into building a custom app, leveraging audio conferencing software, or trying to mimic a webex program? Would Adobe Stratus be useful in getting this kind of functionality? Does anyone have any ideas about how we would design something like this on a macro level? Sorry for the noobish question, but any guidance would be deeply appreciated. Thanks, Keller

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  • XMLStreamReader and a real stream

    - by Yuri Ushakov
    Update There is no ready XML parser in Java community which can do NIO and XML parsing. This is the closest I found, and it's incomplete: http://wiki.fasterxml.com/AaltoHome I have the following code: InputStream input = ...; XMLInputFactory xmlInputFactory = XMLInputFactory.newInstance(); XMLStreamReader streamReader = xmlInputFactory.createXMLStreamReader(input, "UTF-8"); Question is, why does the method #createXMLStreamReader() expects to have an entire XML document in the input stream? Why is it called a "stream reader", if it can't seem to process a portion of XML data? For example, if I feed: <root> <child> to it, it would tell me I'm missing the closing tags. Even before I begin iterating the stream reader itself. I suspect that I just don't know how to use a XMLStreamReader properly. I should be able to supply it with data by pieces, right? I need it because I'm processing a XML stream coming in from network socket, and don't want to load the whole source text into memory. Thank you for help, Yuri.

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  • Flash FLVPlayback states

    - by rob5408
    I'm writing my own class to manage a skin for an FLVPlayback component. It works 90% of the time, but sometimes the state get really messed up. Specifically, the video is playing, meaning I can see it play and the VideoEvent.PLAYHEAD_UPDATE event is firing, but when I poll the FLVPlayback component about its playing property, it returns false. I assume this may be because 'buffering' is kind of a subset of 'playing', but I cannot confirm this in the documentation. I guess another way to ask this question is, "Does the FLVPlayback component ever buffer while it is in a stopped state?"

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  • android mediarecorder exception

    - by Arutha
    I try to use the MediaRecorder class to record a video but I get an exception : failed to get Camera parameters. Prepare failed. Here's my code : camera = Camera.open(); recorder = new MediaRecorder(); recorder.setCamera(camera); recorder.setVideoSource(VideoSource.CAMERA); recorder.setPreviewDisplay(m_holder.getSurface()); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setVideoEncoder(MediaRecorder.VideoEncoder.DEFAULT); recorder.setMaxDuration(10000); recorder.setOutputFile(file.getPath()); recorder.prepare(); Any idea ?

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  • Android Stream Data Over Wifi?

    - by Neb
    Im trying to make an app for android that will stream the data of the accelerometer to be used as a game controller on my pc over a local wifi connection. Is it possible to make some kind of wifi stream of the accelerometer values in the android app and then make the pc somehow 'read' this stream? Or would it just be better for the pc to make endless calls to the phone getting the newest accelerometer values from a local android server? It would also have to send commands from the phone such as 'button1 pressed', 'button1 released'.

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  • Silverlight MediaElement issue

    - by George2
    Hello everyone, I am using Silverlight 3.0 + .Net 3.5 + VSTS 2008 + C# to develop a simple video application using MediaElement of Silverlight. I have two videos and I want to play them at the same time (similar to picture in picture effect) -- i.e. a part of the two videos are overlapped when they are playing (the same concept of Z-Order in UI design). I want to play one MediaElement on top of the other MediaElement, and I am wondering how to assign the overlap order (similar to set Z-Order UI element, but I did not find MediaElement has Z-Order property)? thanks in advance, George

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  • Stream writing lags my GUI

    - by blez
    I have a thread that dequeues data from a queue and write it to another application's STDIN. I'm using Stream, but with .Write and even .BeginWrite, when I send 1mb chunks to the second app, my GUI gets laggy for ~1sec. Why?

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  • Loading a page into memory in Rails

    - by titaniumdecoy
    My rails app produces XML when I load /reports/generate_report.xml. On a separate page, I want to read this XML into a variable and save it to the database. How can I do this? Can I somehow stream the response from the /reports/generate_report.xml URI into a variable? Or is there a better way to do it since the XML is produced by the same web app?

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  • How can I launch QuickTime from my app?

    - by Sheehan Alam
    I am trying to load a video from the web, but am having trouble getting it to appear in QuickTime. I can only hear the audio. I would like it to launch QuickTime. - (void)loadView { NSURL *movieURL = [NSURL URLWithString:@"http://movies.apple.com/media/us/mac/getamac/2009/apple-mvp-biohazard_suit-us-20090419_480x272.mov"]; if (movieURL != nil) { moviePlayer = [[MPMoviePlayerController alloc] initWithContentURL:movieURL]; moviePlayer.initialPlaybackTime = -1.0; // Register to receive a notification when the movie has finished playing. [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(moviePlayBackDidFinish:) name:MPMoviePlayerScalingModeDidChangeNotification object:moviePlayer]; [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(endPlay:) name:MPMoviePlayerPlaybackDidFinishNotification object:moviePlayer]; moviePlayer.scalingMode = MPMovieScalingModeAspectFit; moviePlayer.movieControlMode = MPMovieControlModeDefault; moviePlayer.backgroundColor = [UIColor blackColor]; [moviePlayer play]; } }

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