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  • Rapid calls to fread crashes the application

    - by Slynk
    I'm writing a function to load a wave file and, in the process, split the data into 2 separate buffers if it's stereo. The program gets to i = 18 and crashes during the left channel fread pass. (You can ignore the couts, they are just there for debugging.) Maybe I should load the file in one pass and use memmove to fill the buffers? if(params.channels == 2){ params.leftChannelData = new unsigned char[params.dataSize/2]; params.rightChannelData = new unsigned char[params.dataSize/2]; bool isLeft = true; int offset = 0; const int stride = sizeof(BYTE) * (params.bitsPerSample/8); for(int i = 0; i < params.dataSize; i += stride) { std::cout << "i = " << i << " "; if(isLeft){ std::cout << "Before Left Channel, "; fread(params.leftChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Left Channel, "; } else{ std::cout << "Before Right Channel, "; fread(params.rightChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Right Channel, "; offset += stride; std::cout << "After offset incr.\n"; } isLeft != isLeft; } } else { params.leftChannelData = new unsigned char[params.dataSize]; fread(params.leftChannelData, sizeof(BYTE), params.dataSize, file); }

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  • Webcam video stream processing.

    - by vikramtheone
    Hi Guys, I'm working with an image processing project, my final goal is to detect features on a real time video and finally track those features. I will be working with an Embedded Processor Platform called Freescale's i.MX515, it is a 32-bit media processor running on Ubuntu 9.04. Right now I'm working on the algorithms to locate the features, so, I'm using still images. When I'm satisfied with the results I will have to start using a video stream and I don't want to make use of a video file as a source stream, because then I will have to worry about video decoders then. Instead I would like to plug in a USB Wecam to the embedded platform (It has USB ports on it), directly take the frames as they are captured and send it to my application. I will take care to buy a webcam which will be supported in Linux (Device driver). But my question is will I be able to capture the incoming video stream from the webcam and send it to my application? Will I be able to configure the webcam and DMA to write the incoming frames in a particular memory location whose pointer I can simply pass to my application? (Confused!!!) I hope I could convey my doubts, can anyone guide me with what steps that I have to take to achieve all of these easily? Do you foresee any impossibility here? Help!!! Regards Vikram

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  • byte[] to wav file

    - by John
    Hi, It would be great if you could tell me how I could save a byte[] to a wav file. Sometimes I need to set different samplerate, number of bits and channels. Thanks for your help.

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  • how to do p2p with flash? [closed]

    - by Female Gay
    Possible Duplicates: how to do p2p with flash? Does Flash10 + p2p really work? It's not difficult to tell what is being asked here. This question is not ambiguous, not vague, not incomplete, either not rhetorical and can be reasonably answered in its current form. So, good luck!

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  • Play Shoutcast MP3 radio stream with Python?

    - by Zachary Brown
    I have managed to create an online radio station using Shoutcast and Sam Broadcaster. Now, I am wanting to build my own player for that radio station. I am not sure where to begin, I have googled, but no luck. I am using Python 2.6 on Microsoft Windows. I have managed to capture the stream and save it as an MP# on the hard disk, just not sure what to do with it next. I tried playback of the file, but it always pulls up errors. This is the code I have so far: import urllib target = open("broadcast.mp3") conn = urllib.urlopen("http://78.159.104.175:80") while True: target.write(con.read(5200)) Any help would be greatly appreciated!

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  • Why do calls to waveOutGetPosition hang?

    - by MusiGenesis
    I'm using the winmm.dll API method waveOutGetPosition to get the current position of the playback of a WAV file. Sometimes this works as expected for me, but eventually one of the calls never returns and my application locks up. I found this thread with a few users who have experienced the same problem: http://social.msdn.microsoft.com/Forums/en-US/windowsgeneraldevelopmentissues/thread/c6a1e80e-4a18-47e7-af11-56a89f638ad7 but no solution. Has anyone run into this problem before?

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  • DSP - Filter sweep effect

    - by Trap
    I'm implementing a 'filter sweep' effect (I don't know if it's called like that). What I do is basically create a low-pass filter and make it 'move' along a certain frequency range. To calculate the filter cut-off frequency at a given moment I use a user-provided linear function, which yields values between 0 and 1. My first attempt was to directly map the values returned by the linear function to the range of frequencies, as in cf = freqRange * lf(x). Although it worked ok it looked as if the sweep ran much faster when moving through low frequencies and then slowed down during its way to the high frequency zone. I'm not sure why is this but I guess it's something to do with human hearing perceiving changes in frequency in a non-linear manner. My next attempt was to move the filter's cut-off frequency in a logarithmic way. It works much better now but I still feel that the filter doesn't move at a constant perceived speed through the range of frequencies. How should I divide the frequency space to obtain a constant perceived sweep speed? Thanks in advance.

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  • How to play extracted wave file byte array in C#?

    - by user261924
    At the moment i have managed to separate the left and right channel of a WAVE file and have included the header in a byte[] array. My next step is to be about to play both channels. How can this be done? Here is a code snippet: byte[] song_left = new byte[fa.Length]; byte[] song_right = new byte[fa.Length]; int p = 0; for (int c = 0; c < 43; c++) { song_left[p] = header[c]; p++; } int q = 0; for (s = startByte; s < length; s = s + 3) { song_left[s] = sLeft[q]; q++; s++; song_left[s] = sLeft[q]; q++; } p = 0; for (int c = 0; c < 43; c++) { song_right[p] = header[c]; p++; } This part is reading the header and data from both the right and light channel and saving it to array sLeft[] and sRight[]. This part is working perfectly. Once I obtained the byte arrays, I did the following: System.IO.File.WriteAllBytes("c:\\left.wav", song_left); System.IO.File.WriteAllBytes("c:\\right.wav", song_right); Added a button to play the saved wave file: private void button2_Click(object sender, EventArgs e) { spWave = new SoundPlayer("c:\\left.wav"); spWave.Play(); } Once I hit the play button, this error appers: An unhandled exception of type 'System.InvalidOperationException' occurred in System.dll Additional information: The wave header is corrupt. Any ideas?

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  • Is there any live video stream editing opensource project for my needs?

    - by Ole Jak
    So I need Some open source project with API capable of reading live video stream (stream codec can be any API can read - I can provide with practically any live streamable one) giving me last image data for some processing (like brightness\contrast or more exotic filtering) Being able to recieve data I've changed and starting to stream that data on to some http://localhost:port/ in some format I need it to be easily acsessible from C# (better written on it)

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  • Simple sound effect loop using AudioToolKit

    - by Typeoneerror
    I've created a few sounds for use in my game. I can play them at certain events without issue: // create sounds CFBundleRef mainBundle; mainBundle = CFBundleGetMainBundle(); _soundFileShake = CFBundleCopyResourceURL(mainBundle, CFSTR("shake"), CFSTR("wav"), NULL); AudioServicesCreateSystemSoundID(_soundFileShake, &_soundIdShake); // later... AudioServicesPlaySystemSound(_soundIdShake); The game has a mechanism which allows you to shake the device to activate some functionality. I've got the shaking code done so I get get a "shaking started" and "shaking ended" message to my game. What I need to have happen is start playing "shave.wav" when shaking starts and loop it until it stops. Is there a way to do this with AudioToolbox/AudioServices? How could I do this if not?

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  • How to start writing out an existing AudioQueue in response to an event?

    - by Halle
    Hello, I am writing a class that opens an AudioQueue and analyzes its characteristics, and then under certain conditions can begin or end writing out a file from that AudioQueue that is already instantiated. This is my code (entirely based on SpeakHere) that opens the AudioQueue without writing anything out to tmp: void AQRecorder::StartListen() { int i, bufferByteSize; UInt32 size; try { SetupAudioFormat(kAudioFormatLinearPCM); XThrowIfError(AudioQueueNewInput(&mRecordFormat, MyInputBufferHandler, this, NULL, NULL, 0, &mQueue), "AudioQueueNewInput failed"); mRecordPacket = 0; size = sizeof(mRecordFormat); XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_StreamDescription, &mRecordFormat, &size), "couldn't get queue's format"); bufferByteSize = ComputeRecordBufferSize(&mRecordFormat, kBufferDurationSeconds); for (i = 0; i < kNumberRecordBuffers; ++i) { XThrowIfError(AudioQueueAllocateBuffer(mQueue, bufferByteSize, &mBuffers[i]), "AudioQueueAllocateBuffer failed"); XThrowIfError(AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL), "AudioQueueEnqueueBuffer failed"); } mIsRunning = true; XThrowIfError(AudioQueueStart(mQueue, NULL), "AudioQueueStart failed"); } catch (CAXException &e) { char buf[256]; fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf)); } catch (...) { fprintf(stderr, "An unknown error occurred\n"); } } But I'm a little unclear on how to write a function that will tell this queue "from now until the stop signal, start writing out this queue to tmp as a file". I understand how to tell an AudioQueue to write out as a file at the time that it's created, how to set files format, etc, but not how to tell it to start and stop midstream. Much appreciative of any pointers, thanks.

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  • SoundPool.load() and FileDescriptor from file

    - by Hans
    I tried using the load function of the SoundPool that takes a FileDescriptor, because I wanted to be able to set the offset and length. The File is not stored in the Ressources but a file on the storage card. Even though neither the load nor the play function of the SoundPool throw any Exception or print anything to the console, the sound is not played. Using the same code, but use the file path string in the SoundPool constructor works perfectly. This is how I have tried the loading (start equals 0 and length is the length of the file in miliseconds): FileInputStream fileIS = new FileInputStream(new File(mFile)); mStreamID = mSoundPool.load(fileIS.getFD(), start, length, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); If I would use this, it works: mStreamID = mSoundPool.load(mFile, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); Any ideas anyone? Thanks

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  • Record the timestamps of slide changes during a live Powerpoint presentation?

    - by StackedCrooked
    I am planning to implement a lecture capture solution. One of the requirements is to record both the presenter and the slideshow. The presenter is recorded with a videocamera obviously, and the slideshow will probably be captured using a tool like Camtasia. Now during playback three components are visible: the presenter, the slides and a table of contents. Clicking a chapter title in the TOC causes the video to navigate to the corresponding section. This means that a mapping must be made between chapter titles and their timestamps in the video. Usually a change of topic is accompanied with a slide change in the Powerpoint presentation. So the timestamps could be deduced from the slidechanges. However, this requires me to detect slide changes during the live presentation. And I don't know how to do that. Anyone here knows how to do detect slide changes? Is there a Powerpoint API where I can connect event handlers or something like that? I'd greatly appreciate your help! Edit This issue is no longer relevant for my current work so this question will not be updated by me. However, you are still free to help others by posting your answers/insights here.

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  • How to build a Video Broadcaster, which can handle more than 20,000 viewers

    - by Gaurav Srivastava
    I want to broadcast Video from a web cam, over internet. The problem is, the Video will be viewed live by more than 20,000 people (expected). I have a very little experience with Red5 Broadcasting. I did some broadcasting using Red5 and Flash. It works fine for 1 or 2 viewers i.e. it is great for personal chatting/ video conferencing applications. But, when the number of viewers increases, the delay in Broadcasting also increases. I am experiencing a Delay addition of about 0.5 Seconds for every new user who joins the broadcast. Can any one suggest me some, better technologies on which I can work out this Live Broadcasting. I don't want to use http://www.ustream.com; I want to create one of my own, such tool. But thats always the last solution.

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  • Java vs Flash for webcam access

    - by Alfredo Palhares
    I will make a video chat website, but coming from PHP and Python for the web i have no experience with video steaming. What do you recommend? Java or Flash? What's more flexible ? I am thinking of even making a C++ server application for stream controlling with a PHP fronted. Since is going to be a high traffic website and performance is a must. Can you point to some direction? Any documentation? Framework?

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  • AudioRecord problems with non-HTC devices

    - by Marc
    I'm having troubles using AudioRecord. An example using some of the code derived from the splmeter project: private static final int FREQUENCY = 8000; private static final int CHANNEL = AudioFormat.CHANNEL_CONFIGURATION_MONO; private static final int ENCODING = AudioFormat.ENCODING_PCM_16BIT; private int BUFFSIZE = 50; private AudioRecord recordInstance = null; ... android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); recordInstance = new AudioRecord(MediaRecorder.AudioSource.MIC, FREQUENCY, CHANNEL, ENCODING, 8000); recordInstance.startRecording(); short[] tempBuffer = new short[BUFFSIZE]; int retval = 0; while (this.isRunning) { for (int i = 0; i < BUFFSIZE - 1; i++) { tempBuffer[i] = 0; } retval = recordInstance.read(tempBuffer, 0, BUFFSIZE); ... // process the data } This works on the HTC Dream and the HTC Magic perfectly without any log warnings/errors, but causes problems on the emulators and Nexus One device. On the Nexus one, it simply never returns useful data. I cannot provide any other useful information as I'm having a remote friend do the testing. On the emulators (Android 1.5, 2.1 and 2.2), I get weird errors from the AudioFlinger and Buffer overflows with the AudioRecordThread. I also get a major slowdown in UI responsiveness (even though the recording takes place in a separate thread than the UI). Is there something apparent that I'm doing incorrectly? Do I have to do anything special for the Nexus One hardware?

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  • What is the most efficient way to store a mapping "key -> event stream"?

    - by jkff
    Suppose there are ~10,000's of keys, where each key corresponds to a stream of events. I'd like to support the following operations: push(key, timestamp, event) - pushes event to the event queue for key, marked with the given timestamp. It is guaranteed that event timestamps for a particular key are pushed in sorted or almost sorted order. tail(key, timestamp) - get all events for key since the given timestamp. Usually the timestamp requests for a given key are almost monotonically increasing, almost synchronously with pushes for the same key. This stuff has to be persistent (although it is not absolutely necessary to persist pushes immediately and to keep tails with pushes strictly in sync), so I'm going to use some kind of database. What is the optimal kind of database structure for this task? Would it be better to use a relational database, a key-value storage, or something else?

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  • Read media stream from servlet in a webpage?

    - by khue
    Hi, I have a servlet that construct response to a media file request by reading the file from server: File uploadFile = new File("C:\TEMP\movie.mov"); FileInputStream in = new FileInputStream(uploadFile); Then write that stream to the response stream. My question is how do I play the media file in the webpage using or tag to read the media stream from the response. Thank you very much. Regards K.

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  • Corba sequence<octet> a lot slower than using a socket

    - by Totonga
    I have a corba releated question. In my Java app I use typedef sequence Data; Now I played around with this Data vector. If I am right with the Corba specification sequence will either be converted to xs:base64Binary or xs:hexBinary. It should be an Opaque type and so it should not use any marshalling. I tried different idl styles: void Get(out Data d); Data Get(); but what I see is that moving the data using Corba is a lot slower than using a socket directly. I am fine with a little overhead but it looks for me like tha data is still marshalled. Do I need to somehow configure my orb to suppress the marshalling or did I miss something.

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