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  • Learning to work with audio in C++

    - by Skilldrick
    My degree was in audio engineering, but I'm fairly new to programming. I'd like to learn how to work with audio in a programming environment, partly so I can learn C++ better through interesting projects. First off, is C++ the right language for this? Is there any reason I shouldn't be using it? I've heard of Soundfile and some other libraries - what would you recommend? Finally, does anyone know of any good tutorials in this subject? I've learnt the basics of DSP - I just want to program it! EDIT: I use Windows. I'd like to play about with real-time stuff, a bit like Max/MSP but with more control.

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  • Getting level values from PCM raw data using Core Audio

    - by John
    I am trying to extract level data from a PCM audio file using core audio. I have gotten as far as (I believe) getting the raw data into a byte array (UInt8) but it is 16 bit PCM data and I am having trouble reading the data out. The input is from the iPhone microphone, which I have set as: [recordSetting setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [recordSetting setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey]; [recordSetting setValue:[NSNumber numberWithInt:1] forKey:AVNumberOfChannelsKey]; [recordSetting setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [recordSetting setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; which is obviously 16 bits. I am then trying to just print out a few values to see if they look reasonable for debug purposes below, and they do not look reasonable (many 0's). ExtAudioFileRef inputFile = NULL; ExtAudioFileOpenURL(track.location, &inputFile); AudioStreamBasicDescription inputFileFormat; UInt32 dataSize = (UInt32)sizeof(inputFileFormat); ExtAudioFileGetProperty(inputFile, kExtAudioFileProperty_FileDataFormat, &dataSize, &inputFileFormat); UInt8 *buffer = malloc(BUFFER_SIZE); AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mData = buffer; //pointer to buffer of audio data bufferList.mBuffers[0].mDataByteSize = BUFFER_SIZE; //number of bytes in the buffer while(true) { UInt32 frameCount = (bufferList.mBuffers[0].mDataByteSize / inputFileFormat.mBytesPerFrame); // Read a chunk of input OSStatus status = ExtAudioFileRead(inputFile, &frameCount, &bufferList); // If no frames were returned, conversion is finished if(0 == frameCount) break; NSLog(@"---"); int16_t *bufferl = &buffer; for(int i=0;i<100;i++){ //const int16_t *bufferl = bufferl[i]; NSLog(@"%d",bufferl[i]); } } Not sure what I am doing wrong, I think it has to do with reading the byte array. Sorry for the long code post...

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  • Comapring pitches with digital audio

    - by user2250569
    I work on application which will compare musical notes with digital audio. My first idea was analyzes wav file (or sound in real-time) with some polyphonic pitch algorithms and gets notes and chords from this file and subsequently compared with notes in dataset. I went through a lot of pages and it seems to be a lot of hard work because existing implementations and algorithms are mainly/only focus on monophonic sound. Now, I got the idea to do this in the opposite way. In dataset I have for example note: A4 or better example chord: A4 B4 H4. And my idea is make some wave (or whatever I don't know what) from this note or chord and then compared with piece of digital audio. Is this good idea? Is it better/harder solution? If yes can you recommend me how to do it?

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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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  • Scientific Data processing(Graph comparison and interpretation)

    - by pinkynobrain
    Hi stackoverflow friends, I'm trying to write a program to automate one of my more boring and repetative work tasks. I have some programming experience but none with processing or interpreting large volumes of data so i am seeking your advice(both suggestions of techneques to try and also things to read to learn more about doing this stuff). I have a piece of equipment that monitors an experiment by taking repeated samples and displays the readings on its screen as a graph. The input of experiment can be altered and one of these changes should produce a change in a section of the graph which i currently identify by eye and is what im looking for in the experiment. I want to automate it so that a computer looks at a set of results and spots the experiment input that causes the change. I can already extract the results from the machine. Currently they results for a run are in the form of an integer array with the index being the sample number and the corresponding value being the measurement. The overall shape of the graph will be similar for each experiment run. The change im looking for will be roughly the same and will occur in approximatly the same place every time for the correct experiment input. Unfortunatly there are a few gotcha's that make this problem more difficult. There is some noise in the measuring process which mean there is some random variation in the measured values between different runs. Although the overall shape of the graph remains the same. The time the experiment takes varies slightly each run causing two effects. First, the a whole graph may be shifted slightly on the x axis relative to another runs graph. Second, individual features may appear slightly wider or narrower in different runs. In both these cases the variation isn't particularly large and you can assume that the only non random variation is caused by the correct input being found. Thank you for your time, Pinky

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  • [Processing/Java]Visibility/Layering Issue

    - by nnash
    I'm working on a small sketch in processing where I am making a "clock" using the time functions and drawing ellipses across the canvas based on milliseconds, seconds and minutes. I'm using a for loop to draw all of the ellipses and each for loop is inside its own method. I'm calling each of these methods in the draw function. However for some reason only the first method that is called is being drawn, when ideally I would like to have them all being visibly rendered. //setup program void setup() { size(800, 600); frameRate(30); background(#eeeeee); smooth(); } void draw(){ milliParticles(); secParticles(); minParticles(); } //time based particles void milliParticles(){ for(int i = int(millis()); i >= 0; i++) { ellipse(random(800), random(600), 5, 5 ); fill(255); } } void secParticles() { for(int i = int(second()); i >= 0; i++) { fill(0); ellipse(random(800), random(600), 10, 10 ); background(#eeeeee); } } void minParticles(){ for(int i = int(minute()); i >= 0; i++) { fill(50); ellipse(random(800), random(600), 20, 20 ); } }

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  • Scientific Data processing (Graph comparison and interpretation)

    - by pinkynobrain
    Hi stackoverflow friends, I'm trying to write a program to automate one of my more boring and repetitive work tasks. I have some programming experience but none with processing or interpreting large volumes of data so I am seeking your advice (both suggestions of techniques to try and also things to read to learn more about doing this stuff). I have a piece of equipment that monitors an experiment by taking repeated samples and displays the readings on its screen as a graph. The input of experiment can be altered and one of these changes should produce a change in a section of the graph which I currently identify by eye and is what I'm looking for in the experiment. I want to automate it so that a computer looks at a set of results and spots the experiment input that causes the change. I can already extract the results from the machine. Currently they results for a run are in the form of an integer array with the index being the sample number and the corresponding value being the measurement. The overall shape of the graph will be similar for each experiment run. The change I'm looking for will be roughly the same and will occur in approximately the same place every time for the correct experiment input. Unfortunately there are a few gotchas that make this problem more difficult. There is some noise in the measuring process which mean there is some random variation in the measured values between different runs. Although the overall shape of the graph remains the same. The time the experiment takes varies slightly each run causing two effects. First, the a whole graph may be shifted slightly on the x axis relative to another run's graph. Second, individual features may appear slightly wider or narrower in different runs. In both these cases the variation isn't particularly large and you can assume that the only non random variation is caused by the correct input being found. Thank you for your time, Pinky

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  • How to draw histogram in Processing

    - by theolc
    I have an arrayList and I would like to take the size of the ten arrayList elements and use the sizes to create a histogram. I understand processing coordinate system starts in the top left corner. My question is, how do I use the arrayList.size() values to start at 450 (based from a 500,500 window) and go upwards from there to create my histogram. The following is my function for the histogram, it is receiving as a parameter the arrayList called bins. void histogram(ArrayList[] bins) { //set window background(0,0,0); size(500,500); background(255,255,255); line(50,0,50,500); line(0,450,500,450); int i; for (i = 50; i <= 500; i+=45) { line(i,450,i,480); } for (i = 50; i <= 450; i+=45) { line(50,i,20,i); } } Thanks in adavance for any help and input!

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • Remove noise from a recording

    - by essamSALAH
    I used to record important technical meetings and demos using Camtasia Studio, using a Mic to capture the speaker voice. Sometimes we invite attendees by asking them to call us in the meeting, and they would call on a cell phone then we switch it to loudspeaker so we can hear and talk to them and also record the conversation on Camtasia. The problem I am having now is that playing back those recording produces the regular noise that results from the microphone being close to the mobile phone (the kind of noise you hear when your mobile phone rings and it is near a speaker). Any advice on removing this noise?

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  • Too big a difference between Loud and Quiet when watching DVD's... constantly have to adjust volume.

    - by Dan
    It is hard for me to find the right volume for my computer to watch my dvd's on because it seems like most reasonable volumes become overwhelming at the loudest parts of a movie and it is hard to even make out the dialog at the quietest parts. I find I'm constantly adjusting the volume during the course of a movie. Are there ways to make the difference between the louds and the quiets not so extreme? (both computer related solutions and non-computer related solutions welcome). Like moving my speakers across the room and increasing the volume? or the opposite? Or would would the extremes be less noticeable if I used headphones? Are there movie players that might have more complex sound adjustment features? If there is a software solution out there for linux that would be great too. Thanks, Dan

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  • Software Suggestion for Managing Voice Recordings (Windows)

    - by Cbeppe
    I'm looking for Windows software that allows me to effectlively manage already made voice recordings. I have a series of recordings taken from an iPhone and I have extracted the files. The problem is that these are very long recordings and therefore I'm looking for software that allows me to: Bookmark a time in the recording Effectively manage multiple files (like Adobe Bridge does with images) Freeware or Payware Possibly other features, I haven't done this before and I'm sorry I'm unable to give a more professional description. Thanks in advance to everyone who can help! If you have any other questions, please don't hesitate to ask - I will try my best to provide useful answers.

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  • Restore audio settings - cannot open mixer: No such file or directory

    - by Alfred M.
    The internal speaker of my laptop never functionned under Ubuntu. I tried to follow indication on the web and now the jack audio does not work either. The graphic interface for audio management now displays a 'dummy output' instead of the three possible outputs I used to have (one of them was working for the jack output). In a terminal alsamixer raises an error: cannot open mixer: No such file or directory I did try to remove and reinstall alsa-utils but it did not change anything. This happened after a failed atempt to install alsa-driver-linuxant_1.0.23.1_all.deb from here. My sound card seems to be not recognised anymore. After reboot I have no more the sound icon in menu bar the upper right corner. I think I have removed my sound card driver. Indeed, the command sudo lshw -class multimedia indicated audi device as unclaimed. Any idea how I could revert to a better situation (that is jack support and alsa working)? EDIT: The command lspci -nnk | grep -iEA3 audio gives lspci -nnk | grep -iEA3 audio 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) Subsystem: ASUSTeK Computer Inc. Device [1043:1893] 00:1c.0 PCI bridge [0604]: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 [8086:2940] (rev 03)

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  • How can i compare Audio, what programming language should i use

    - by Pimmetje
    I have 2 audio files that are from almost the same source. But at some points there shifted a bit. Also the codecs does not match. I would like to make a program that takes a sample 2 - 4 seconds. And looks for it in the other file. (Most of the time it's not shifted more than 30 seconds). Than take the time and store it, Go ahead for a few seconds take a sample and find it again. This way i want to create a file where i can see on what points the file is shifted. For people who are more interested in what i want. I have a audio/video file speech and subtitles. But i have same speech from different sources with differs a bit in time. And i like to make a program that can correct the subtitle time for me. Enough about the problem I looked on the Internet for ways to compare audio files. Based on what i read comparing 2 audio files isn't that easy as i had hoped. Some talk about algorithms http://www.perlmonks.org/?node_id=169641 Some audio-library's portaudio.com aubio.org sourceforge.net/projects/ccaudio/ ambiera.com/irrklang/ The biggest problem i have is that i can't find something i can generate from the audio that i can use to compare with. I hope someone here can point me in the right direction.

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  • Emulating Button Press Using Kinect/SimpleOpenNI + Processing Depth

    - by Alex Lu
    I am using Kinect with Simple OpenNI and Processing, and I was trying to use the Z position of a hand to emulate a button press. So far when I try it using one hand it works really well, however, when I try to get it to work with a second hand, only one of the hands work. (I know it can be more efficient by moving everything except the fill out of the if statements, but I kept those in there just in case I want to change the sizes or something.) irz and ilz are the initial Z positions of the hands when they are first recognized by onCreateHands and rz and lz are the current Z positions. As of now, the code works fine with one hand, but the other hand will either stay pressed or unpressed. If i comment one of the sections out, it works fine as well. if (rz - irz > 0) { pushStyle(); fill(60); ellipse(rx, ry, 10, 10); popStyle(); rpressed = true; } else { pushStyle(); noFill(); ellipse(rx, ry, 10, 10); popStyle(); rpressed = false; } if (lz - ilz > 0) { pushStyle(); fill(60); ellipse(lx, ly, 10, 10); popStyle(); lpressed = true; } else { pushStyle(); noFill(); ellipse(lx, ly, 10, 10); popStyle(); lpressed = false; } I tried outputting the values of rz - irz and lz - ilz and the numbers range from small negative values to small positive values (around -8 to 8) for lz - ilz. But rz - irz outputs numbers from around 8-30 depending on each time I run it and is never consistent. Also, when I comment out the code for the lz-ilz, the values for rz-irz look just fine and it operates as intended. Is there a reason tracking both Z positions throws off one hand? And is there a way to get it to work? Thanks!

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  • Encode audio to aac with libavcodec

    - by ryan
    I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

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  • jQuery Form Processing With PHP to MYSQL Database Using $.ajax Request

    - by FrustratedUser
    Question: How can I process a form using jQuery and the $.ajax request so that the data is passed to a script which writes it to a database? Problem: I have a simple email signup form that when processed, adds the email along with the current date to a table in a MySQL database. Processing the form without jQuery works as intended, adding the email and date. With jQuery, the form submits successfully and returns the success message. However, no data is added to the database. Any insight would be greatly appreciated! <!-- PROCESS.PHP --> <?php // DB info $dbhost = '#'; $dbuser = '#'; $dbpass = '#'; $dbname = '#'; // Open connection to db $conn = mysql_connect($dbhost, $dbuser, $dbpass) or die ('Error connecting to mysql'); mysql_select_db($dbname); // Form variables $email = $_POST['email']; $submitted = $_POST['submitted']; // Clean up function cleanData($str) { $str = trim($str); $str = strip_tags($str); $str = strtolower($str); return $str; } $email = cleanData($email); $error = ""; if(isset($submitted)) { if($email == '') { $error .= '<p class="error">Please enter your email address.</p>' . "\n"; } else if (!eregi("^[A-Z0-9._%-]+@[A-Z0-9._%-]+\.[A-Z]{2,4}$", $email)) { $error .= '<p class="error">Please enter a valid email address.</p>' . "\n"; } if(!$error){ echo '<p id="signup-success-nojs">You have successfully subscribed!</p>'; // Add to database $add_email = "INSERT INTO subscribers (email,date) VALUES ('$email',CURDATE())"; mysql_query($add_email) or die(mysql_error()); }else{ echo $error; } } ?> <!-- SAMPLE.PHP --> <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html xmlns="http://www.w3.org/1999/xhtml"> <head> <meta http-equiv="Content-Type" content="text/html; charset=utf-8" /> <title>Sample</title> <script type="text/javascript" src="http://ajax.googleapis.com/ajax/libs/jquery/1.2.6/jquery.min.js"></script> <script type="text/javascript"> $(document).ready(function(){ // Email Signup $("form#newsletter").submit(function() { var dataStr = $("#newsletter").serialize(); alert(dataStr); $.ajax({ type: "POST", url: "process.php", data: dataStr, success: function(del){ $('form#newsletter').hide(); $('#signup-success').fadeIn(); } }); return false; }); }); </script> <style type="text/css"> #email { margin-right:2px; padding:5px; width:145px; border-top:1px solid #ccc; border-left:1px solid #ccc; border-right:1px solid #eee; border-bottom:1px solid #eee; font-size:14px; color:#9e9e9e; } #signup-success { margin-bottom:20px; padding-bottom:10px; background:url(../img/css/divider-dots.gif) repeat-x 0 100%; display:none; } #signup-success p, #signup-success-nojs { padding:5px; background:#fff; border:1px solid #dedede; text-align:center; font-weight:bold; color:#3d7da5; } </style> </head> <body> <?php include('process.php'); ?> <form id="newsletter" class="divider" name="newsletter" method="post" action=""> <fieldset> <input id="email" type="text" name="email" /> <input id="submit-button" type="image" src="<?php echo $base_url; ?>/assets/img/css/signup.gif" alt=" SIGNUP " /> <input id="submitted" type="hidden" name="submitted" value="true" /> </fieldset> </form> <div id="signup-success"><p>You have successfully subscribed!</p></div> </body> </html>

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  • No HDMI audio in 13.04

    - by King84
    I have just upgraded from 12.10 to 13.04 and now everything works perfectly, except the fact that I have no audio via HDMI. I am using a Samsung tv-monitor connected via HDMI to my video card Asus EAH4670/DI/1GD3 (which has a Radeon HD 4670 gpu on it), installed phisically into my motherboard which is a MSI 770-C45. I am running kernel 3.9, I just have no sound. I tried downloading and installing https://code.launchpad.net/~ubuntu-audio-dev/+archive/alsa-daily/+files/oem-audio-hda-daily-dkms_0.201304261252~raring1_all.deb , but without any good result. Please help, I need my audio back. In the end, this is my lspci command output. ale@beast:~$ lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] nee ATI RX780/RX790 Host Bridge 00:02.0 PCI bridge: Advanced Micro Devices [AMD] nee ATI RD790 PCI to PCI bridge (external gfx0 port A) 00:06.0 PCI bridge: Advanced Micro Devices [AMD] nee ATI RD790 PCI to PCI bridge (PCI express gpp port C) 00:11.0 SATA controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 SATA Controller [IDE mode] 00:12.0 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 USB OHCI0 Controller 00:12.1 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0 USB OHCI1 Controller 00:12.2 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 USB EHCI Controller 00:13.0 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 USB OHCI0 Controller 00:13.1 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0 USB OHCI1 Controller 00:13.2 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 USB EHCI Controller 00:14.0 SMBus: Advanced Micro Devices [AMD] nee ATI SBx00 SMBus Controller (rev 3c) 00:14.1 IDE interface: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 IDE Controller 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 LPC host controller 00:14.4 PCI bridge: Advanced Micro Devices [AMD] nee ATI SBx00 PCI to PCI Bridge 00:14.5 USB controller: Advanced Micro Devices [AMD] nee ATI SB7x0/SB8x0/SB9x0 USB OHCI2 Controller 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 10h Processor HyperTransport Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 10h Processor Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 10h Processor DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 10h Processor Miscellaneous Control 00:18.4 Host bridge: Advanced Micro Devices [AMD] Family 10h Processor Link Control 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI RV730 XT [Radeon HD 4670] 01:00.1 Audio device: Advanced Micro Devices [AMD] nee ATI RV710/730 HDMI Audio [Radeon HD 4000 series] 02:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168 PCI Express Gigabit Ethernet controller (rev 03) ale@beast:~$

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