pcm

I'm trying to get raw audio data from a file (i'm used to seeing floating point values between -1 and 1). I'm trying to pull this data out of the buffers in real time so that I can provide some type of metering for the app. I'm basically reading the whole file into memory using AudioFileReadPackets. I've create a RemoteIO audio unit to do playback and inside of the playbackCallback, i'm supplying the mData to the AudioBuffer so that it can be sent to hardware. The big problem I'm having is that the data being sent to the buffers from my array of data (from AudioFileReadPackets) is UInt32... I

I have several chunks of PCM audio (G.711) in my C++ application. I would like to visualize the different audio volume in each of these chunks. My first attempt was to calculate the average of the sample values for each chunk and use that as an a volume indicator, but this doesn't work well. I do get 0 for chunks with silence and differing values for chunks with audio, but the values only differ slighly and don't seem to resemble the actual volume. What would be a better algorithem calculate the volume ? I hear G.711 audio is logarithmic PCM. How should I take that into account ? Note, I haven

Is it possible in j2me to measure signal amplitude of audio record made by JSR-135 Player? I know I can access buffer, but then what? Target model Bold 9000, supported formats PCM and AMR. Which format I should use? See also Blackberry Audio Recording Sample Code How To - Record Audio on a BlackBerry smartphone Thank you! Maksym Gontar Get raw PCM signal level Use menu and trackwheel to zoom in/out and move left/right within graph. Audio format: raw 8000 Hz 16 bit mono pcm. Tested on Bold 9000 RIM OS 4.6 Algorythm should work in any mobile, where j2me and pcm is supported, of course

I've trying to calculate peak frequency from android mic buffer as per this How to get frequency from fft result? . Unfortunatly i'm getting wrong values. Even i played a tone in 18Khz,but i'm not getting correct peak frequency. This is my code, int sampleRate=44100,bufferSize=4096; AudioRecord audioRec=new AudioRecord(AudioSource.MIC,sampleRate,AudioFormat.CHANNEL_CONFIGURATION_MONO,AudioFormat.ENCODING_PCM_16BIT,bufferSize); audioRec.startRecording(); audioRec.read(bufferByte, 0,bufferSize); for(int i=0;i<bufferByte.length;i++){ bufferDouble2[i]=(double)bufferByte[i]; } //here window techniq

I use AVAudioMixerNode to change audio format. this entry helped me a lot. Below code gives me data i want. But i hear my own voice on phone's speaker. How can i prevent it? func startAudioEngine() { engine = AVAudioEngine() guard let engine = engine, let input = engine.inputNode else { // @TODO: error out return } let downMixer = AVAudioMixerNode() //I think you the engine's I/O nodes are already attached to itself by default, so we attach only the downMixer here: engine.attach(downMixer) //You can tap the downMixer to intercept the audio and do something with it: downMixer.installTap(onBus:

I need to convert a continuous stream of PCM, or encoded audio (ADPCM, uLaw, Opus), into MP3/OGG format so that it can be streamed to a browser (using html's audio tag). I have the "stream-mp3/ogg-using-audio-tag" part working, now I need to develop the conversion layer. I have two questions: How can I convert PCM into MP3/OGG using NAudio and/or some other C# library/framework? I assume there is a code snippet or two in the NAudio demo app that may be doing this, but I haven't been able to find it. Do I have to convert the encoded data (ADPCM, uLaw, OPUS) into PCM (which I can) before I

Here's what I'm working with right now: for (int i = 0, numSamples = soundBytes.length / 2; i < numSamples; i += 2) { // Get the samples. int sample1 = ((soundBytes[i] & 0xFF) << 8) | (soundBytes[i + 1] & 0xFF); // Automatically converts to unsigned int 0...65535 int sample2 = ((outputBytes[i] & 0xFF) << 8) | (outputBytes[i + 1] & 0xFF); // Automatically converts to unsigned int 0...65535 // Normalize for simplicity. float normalizedSample1 = sample1 / 65535.0f; float normalizedSample2 = sample2 / 65535.0f; float normalizedMixedSample = 0.0f; // Apply the algorithm. if (normalizedSample1 < 0

I'm trying to encode raw pcm data as uLaw to save on the bandwidth required to transmit speech data. I have come across a class called UlawEncoderInputStream on This page but there is no documentation! :( The constructor takes an input stream and a max pcm value (whatever that is). /** * Create an InputStream which takes 16 bit pcm data and produces ulaw data. * @param in InputStream containing 16 bit pcm data. * @param max pcm value corresponding to maximum ulaw value. */ public UlawEncoderInputStream(InputStream in, int max) { After looking through the code, I suspect that i should calculate

I understand that PCM data is stored as [left][right][left][right]... . Am trying to convert a stereo PCM to mono Vorbis (*.ogg) which I understand is achievable by halving the left and the right channels ((left+right)*0.5). I have actually achieved this by amending the encoder example in the libvorbis sdk like this, #define READ 1024 signed char readbuffer[READ*4]; and the PCM data is read thus fread(readbuffer, 1, READ*4, stdin) I then halved the two channels, buffer[0][i] = ((((readbuffer[i*4+1]<<8) | (0x00ff&(int)readbuffer[i*4]))/32768.f) + (((readbuffer[i*4+3]<<8) | (0x00ff&(int