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  • How to configure the framesize using AudioUnit.framework on iOS

    - by Piperoman
    I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

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  • Prefered method for looping sound flash as3

    - by Brian Heylin
    Hi there, I'm having some issues with looping a sound in flash AS3, in that when I tell the sound to loop I get a slight delay at the end/beginning of the audio. The audio is clipped correctly and will play without a gap on garage band. I know that there are issues with sound in general in flash, bugs with encodings and the inaccuracies with the SOUND_COMPLETE event (And Adobe should be embarrassed with their handling of these issues) I have tried to use the built in loop argument in the play method on the Sound class and also react on the SOUND_COMPLETE event, but both cause a delay. But has anyone come up with a technique for looping a sound without any noticeable gap?

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  • Signal amplitude against time in Java

    - by wsr74ws84
    I'm racking my brain in order to solve a knotty problem (at least for me). While playing an audio file (using Java) I want the signal amplitude to be displayed against time. I mean I'd like to implement a small panel showing a sort of oscilloscope (spectrum analyzer). The audio signal should be viewed in the time domain (vertical axis is amplitude and the horizontal axis is time). Does anyone know how to do it? Is there a good tutorial I can rely on? Since I know very little about Java, I hope someone can help me.

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  • High speed matrix manipulation in c#?

    - by Donnie
    I'm working on some image manipulation code in c# and need to do some matrix operations (specifically 2D convolution). I have the code written in matlab which uses the conv2 function ... is there a library for C# / .NET that does good high-speed matrix manipulations? I'd be fine if it requires some specific GPU and does the matrix math on-GPU if that's what it takes.

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  • High-level languages for out-of-the-box GUI desktop application programming

    - by Omeoe
    After I discontinued programming in C++ while entering into web authoring I was spoilt by PHP's high level constructs like hash tables or its dynamic, weak typing. I remembered the angst of C/C++ pointers and the maze of low-level Win32 API handles and message loops and that prevented me from utilizing environments like Code::Blocks for desktop applications. I am also not very fond of bulky, statically-typed C#/.NET environment. Any other ideas?

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  • Html5 - Callback when media is ready on iPad wont work

    - by Kap
    I'm trying to add a callback to a HTML5 audio element on an iPad. I added an eventlistener to the element, the myOtherThing() starts but there is no sound. If I pause and the play the sound again the audio starts. This works in chrome. Does anyone have an idea how I can do this? myAudioElement.src = "path_to_file"; addEventListener("canplay", function(){ myAudioElement.play(); myOtherThing.start(); }); SOLVED Just wanted to share my solution here, just in case someone else needs it. As far as I understand the iPad does not trigger any events without user interactions. So to be able to use "canply", "playing" and all the other events you need to use the built in media controller. Once you press play in that controller, the events gets triggered. After that you can use your custom interface.

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  • Question on ExtAudioFileRead and AudioBuffer for iPhone SDK

    - by backspacer
    I'm developing an iPhone app that uses the Extended Audio File Services. I try to use ExtAudioFileRead to read the audio file, and store the data in an AudioBufferList structure. AudioBufferList is defined as: struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[1]; }; typedef struct AudioBufferList AudioBufferList; and AudioBuffer is defined as struct AudioBuffer { UInt32 mNumberChannels; UInt32 mDataByteSize; void* mData; }; typedef struct AudioBuffer AudioBuffer; I want to manipulate the mData but I wonder what does the void* mean. Why is it void*? How can I decide what data type is actually stored in mData?

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  • Can FLV AAC stream be played in Android

    - by HariKJ
    Hi, I'm trying to build a radio player and the client is providing a stream which is a FLV container with the audio being AAC When I read the headers it shows up as audio/aacp. I have tried all possible ways such as using the 1) Streaming through mediaplayer (Does not work) 2) Use the NPR mode of using a proxy stream (I get a broken pipe exception) 3) Play it in chunks ( Plays but I need the SDCard and the playback is not very great) 4) Use the GPL'd FAAD2 Library but I would have to pay the royalty fee Can some one help me out on figuring this issue out. The last option that I have is to have my client change the stream to mp3 container (which I know that it works) Regards, Hari

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  • sound loop breaks after some time in background music in iphone app

    - by amy
    I am playing sounds in loop in my app. So it should continue playing through out the app. but sometimes it stops after playing sound for 3/4 times.I don't understand whats happening. I am using audio-toolbox framework for playing sound. creating audio queue and then playing sounds in loop. I am also playing sound from ipod library using mediaplayer. Same thing happening with song from ipod. I have set [musicPlayer setRepeatMode: MPMusicRepeatModeOne]; but still it stops after 3/4 times.

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  • procedure that swaps the bytes (low/high) of a Word variable

    - by Altar
    Hi. I have this procedure that swaps the bytes (low/high) of a Word variable (It does the same stuff as System.Swap function). The procedure works when the compiler optimization is OFF but not when it is ON. Can anybody help me with this? { UNSAFE! IT IS NOW WORKING WHEN COMPILER OPTIMIZATION IS ON ! } procedure SwapWord_NotWorking(VAR TwoBytes: word); asm Mov EBX, TwoBytes Mov AX, [EBX] XCHG AL,AH Mov [EBX], AX end;

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  • Tips for maximizing Nginx requests/sec?

    - by linkedlinked
    I'm building an analytics package, and project requirements state that I need to support 1 billion hits per day. Yep, "billion". In other words, no less than 12,000 hits per second sustained, and preferably some room to burst. I know I'll need multiple servers for this, but I'm trying to get maximum performance out of each node before "throwing more hardware at it". Right now, I have the hits-tracking portion completed, and well optimized. I pretty much just save the requests straight into Redis (for later processing with Hadoop). The application is Python/Django with a gunicorn for the gateway. My 2GB Ubuntu 10.04 Rackspace server (not a production machine) can serve about 1200 static files per second (benchmarked using Apache AB against a single static asset). To compare, if I swap out the static file link with my tracking link, I still get about 600 requests per second -- I think this means my tracker is well optimized, because it's only a factor of 2 slower than serving static assets. However, when I benchmark with millions of hits, I notice a few things -- No disk usage -- this is expected, because I've turned off all Nginx logs, and my custom code doesn't do anything but save the request details into Redis. Non-constant memory usage -- Presumably due to Redis' memory managing, my memory usage will gradually climb up and then drop back down, but it's never once been my bottleneck. System load hovers around 2-4, the system is still responsive during even my heaviest benchmarks, and I can still manually view http://mysite.com/tracking/pixel with little visible delay while my (other) server performs 600 requests per second. If I run a short test, say 50,000 hits (takes about 2m), I get a steady, reliable 600 requests per second. If I run a longer test (tried up to 3.5m so far), my r/s degrades to about 250. My questions -- a. Does it look like I'm maxing out this server yet? Is 1,200/s static files nginx performance comparable to what others have experienced? b. Are there common nginx tunings for such high-volume applications? I have worker threads set to 64, and gunicorn worker threads set to 8, but tweaking these values doesn't seem to help or harm me much. c. Are there any linux-level settings that could be limiting my incoming connections? d. What could cause my performance to degrade to 250r/s on long-running tests? Again, the memory is not maxing out during these tests, and HDD use is nil. Thanks in advance, all :)

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  • android spectrum analysis of streaming input

    - by TheBeeKeeper
    for a school project I am trying to make an android application that, once started, will perform a spectrum analysis of live audio received from the microphone or a bluetooth headset. I know I should be using FFT, and have been looking at moonblink's open source audio analyzer ( http://code.google.com/p/moonblink/wiki/Audalyzer ) but am not familiar with android development, and his code is turning out to be too difficult for me to work with. So I suppose my questions are, are there any easier java based, or open source android apps that do spectrum analysis I can reference? Or is there any helpful information that can be given, such as; steps that need be taken to get the microphone input, put it into an fft algorithm, then display a graph of frequency and pitch over time from its output? Any help would be appreciated, thanks.

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  • background music stops after 3/4 runs in iphone app

    - by amy
    I am playing sounds in loop in my app. So it should continue playing through out the app. but sometimes it stops after playing sound for 3/4 times.I don't understand whats happening. I am using audio-toolbox framework for playing sound. creating audio queue and then playing sounds in loop. I am also playing sound from ipod library using mediaplayer. Same thing happening with song from ipod. I have set [musicPlayer setRepeatMode: MPMusicRepeatModeOne]; but still it stops after 3/4 times.

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  • how to fast compute distance between high dimension vectors

    - by chyojn
    assume there are three group of high dimension vectors: {a_1, a_2, ..., a_N}, {b_1, b_2, ... , b_N}, {c_1, c_2, ..., c_N}. each of my vector can be represented as: x = a_i + b_j + c_k, where 1 <=i, j, k <= N. then the vector is encoded as (i, j, k) wich is then can be decoded as x = a_i + b_j + c_k. my question is, if there are two vector: x = (i_1, j_1, k_1), y = (i_2, j_2, k_2), is there a method to compute the euclidian distance of these two vector without decode x and y.

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  • High accuracy cpu timers

    - by John Robertson
    An expert in highly optimized code once told me that an important part of his strategy was the availability of extremely high performance timers on the CPU. Does anyone know what those are and how one can access them to test various code optimizations? While I am interested regardless, I also wanted to ask whether it is possible to access them from something higher than assembly (or with only a little assembly) via visual studio C++?

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  • Networking with extremely high latency.

    - by BCS
    Are there any protocols, systems, etc. experimental or otherwise designed for allowing normal (as normal as can be) network operations (E-mail, DNS, HTML, etc.) over very high latency links? I'm thinking of minutes to an hour, or maybe two. Think light speed lag at a solar system scale.

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  • Voices disappear when using headphones. [closed]

    - by James
    How do I declare a variable in C? P.S. I have a pair of SteelSeries Siberia headphones. I've noticed that when watching some films the voices are completely silent, yet when I unplug the headset and listen through my speakers they are there and sound normal. I have no other software that could be interfering with it and it happens regardless of the software I use for playback (I've tried VLC, WMP and Quicktime). It is so strange, and it almost sounds deliberate - the rest of the audio is untouched but voices disappear. The films only have single audio tracks, and it doesn't happen with every film. Can anyone give me any hints as to what could possibly cause this? I am stumped!

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  • AudioOutputUnitStart takes time

    - by tokentoken
    Hello, I'm making an iPhone game application using Core Audio, Extended Audio File Services. It works OK, but when I first call AudioOutputUnitStart, it takes about 1-2 seconds. After the second call, no problem. For a game application, 1-2 seconds is very noticeable. (I tested this on iPhone simulator, and iPhone 3GS) Also, if I leave the game for about 10 seconds, first call of AudioOutputUnitStart also takes time. Maybe I have to call AudioOutputUnitStart beginning of the application to prevent the start-up time?

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