pcm

I've been playing with this now for sometime, I cant work out what I am meant to be doing here. I am reading in PCM audio data into an audioData array: recorder.read(audioData,0,bufferSize); //read the PCM audio data into the audioData array I want to use Piotr Wendykier's JTransform library in order to preform an FFT on my PCM data in order to obtain the frequency. import edu.emory.mathcs.jtransforms.fft.DoubleFFT_1D; At the moment I have this: DoubleFFT_1D fft = new DoubleFFT_1D(1024); // 1024 is size of array for (int i = 0; i < 1023; i++) { a[i]= audioData[i]; if (audioData[i] != 0) Log.v

I'm currently trying to decipher WAV files. From headers to the PCM data. I've found a PDF ( http://www.tdt.com/T2Support/technical_notes/tn0132.pdf ) detailing the anatomy of a WAV file, and I've been able to extract and make sense of the appropriate header data using Ghex2. But my questions are: Why are the integers bytes stored backwards? I.e. dec. 20 is stored as 0x14000000 instead of 0x00000014. Are the integers of the PCM data also stored backwards? FixerMark WAV files are little-endian (least significant bytes first) because the format originated for operating systems running on intel

I'm streaming recorded PCM audio from a browser with web audio api. I'm streaming it with binaryJS (websocket connection) to a nodejs server and I'm trying to play that stream on the server using the speaker npm module. This is my client. The audio buffers are at first non-interleaved IEEE 32-bit linear PCM with a nominal range between -1 and +1 . I take one of the two PCM channels to start off and stream it below. var client = new BinaryClient('ws://localhost:9000'); var Stream = client.send(); recorder.onaudioprocess = function(AudioBuffer){ var leftChannel = AudioBuffer.inputBuffer

I currently have a PCM audio in the form of a byte array. The format is signed 16 bit little endian. I would like to convert this to some playable format on the Android, preferably version 3.2 or higher. Does anyone have suggestions on how this can be done? I have done some research and tried the following below, but none were successful. It would be much appreciated if anyone can suggest a better way or indicate where I have gone wrong. I have tried creating an AudioFormat with the correct audio settings, however Android does not support the javax.sound.sampled library. I have also tried

I am currently working on an application as part of my Bachelor in Computer Science. The application will correlate data from the iPhone hardware (accelerometer, gps) and music that is being played. The project is still in its infancy, having worked on it for only 2 months. The moment that I am right now, and where I need help, is reading PCM samples from songs from the itunes library, and playing them back using and audio unit. Currently the implementation I would like working does the following: chooses a random song from iTunes, and reads samples from it when required, and stores in a

I'm currently using ffmpeg to convert FLV/Speex to WAV/pcm_s16le , successfully. However, I now need the output format to be RAW, that is, PCM signed 16-bit little endian, without the WAV header. I tried the following: ffmpeg -y -i input.flv -vn -acodec pcm_s16le output.raw But ffmpeg responds with: Unable to find a suitable output format for 'output.raw' I also tried using output.pcm and output as output file names, with the same result. I also tried the -f flag to specify raw format, but that gives: Unknown input or output format: raw Is this possible with FFmpeg? If so, how? Chris Haas Give

I have PCM samples in a short array. What is the best way to play this out? The format is 8000Hz, Mono, 16 bit, big endian. (The PCM samples are generated in the code and not read through some file) Thanks With the javax.sound.sampled package it's pretty much straightforward, but you have to use some boilerplate. Here's a good tutorial on that: www.wikijava.org/wiki/Play_a_wave_sound_in_Java Basically you have to create an InputStream from your array and use that to create an AudioInputStream . There you have to specify the format of your audio data. Then you open an output stream (