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consuelitaconsuelita
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Audio Resampler
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‎AudioCoder/src/main/java/org/operatorfoundation/audiocoder/AudioResampler.kt‎

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package org.operatorfoundation.audiocoder
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import timber.log.Timber
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import kotlin.math.roundToInt
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/**
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* Linear interpolation audio resampler for WSPR signals.
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*
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* Uses linear interpolation to convert between sample rates sufficient for WSPR's narrow band signals.
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*
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* This resampler maintains continuity between audio chunks by remembering
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* the last sample from the previous chunk for interpolation.
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*
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* Example usage:
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* ```kotlin
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* val resampler = AudioResampler(inputSampleRate = 48000, outputSampleRate = 12000)
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* val resampledAudio = resampler.resample(audioSamples)
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* ```
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*/
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class AudioResampler(
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private val inputSampleRate: Int,
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private val outputSampleRate: Int
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) {
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/**
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* Ratio for sample rate conversion calculation.
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*/
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private val resampleRatio = outputSampleRate.toDouble() / inputSampleRate.toDouble()
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/**
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* Last sample from previous chunk, used for interpolation continuity.
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*/
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private var lastSample: Short = 0
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/**
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* Statistics for monitoring resampler performance.
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*/
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private var totalInputSamples = 0L
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private var totalOutputSamples = 0L
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init {
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require(inputSampleRate > 0) { "Input sample rate must be positive: $inputSampleRate" }
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require(outputSampleRate > 0) { "Output sample rate must be positive: $outputSampleRate" }
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Timber.d("AudioResampler initialized: ${inputSampleRate}Hz -> ${outputSampleRate}Hz (ratio: %.3f)".format(resampleRatio))
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}
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/**
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* Resamples input audio to the target sample rate using linear interpolation.
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*
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* @param inputSamples Raw 16-bit audio samples at the input sample rate
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* @return Resampled audio at the output sample rate
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*/
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fun resample(inputSamples: ShortArray): ShortArray
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{
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if (inputSamples.isEmpty()) {
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Timber.v("Empty input samples, returning empty array")
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return shortArrayOf()
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}
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// No resampling needed if rates match
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if (inputSampleRate == outputSampleRate)
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{
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totalInputSamples += inputSamples.size
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totalOutputSamples += inputSamples.size
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return inputSamples
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}
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// Calculate output length
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val outputLength = calculateOutputSize(inputSamples.size)
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val outputSamples = ShortArray(outputLength)
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// Perform linear interpolation resampling
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for (i in outputSamples.indices)
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{
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val inputIndex = i / resampleRatio
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val inputIndexInt = inputIndex.toInt()
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val fraction = inputIndex - inputIndexInt
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// Get the two samples to interpolate between
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val sample1 = getSampleForInterpolation(inputSamples, inputIndexInt)
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val sample2 = getSampleForInterpolation(inputSamples, inputIndexInt + 1)
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// Linear interpolation: sample1 + fraction * (sample2 - sample1)
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val interpolated = sample1 + (fraction * (sample2 - sample1))
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// Clamp to 16-bit range and store
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outputSamples[i] = interpolated.roundToInt()
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.coerceIn(Short.MIN_VALUE.toInt(), Short.MAX_VALUE.toInt())
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.toShort()
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}
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// Update statistics
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totalInputSamples += inputSamples.size
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totalOutputSamples += outputLength
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// Remember last sample for next chunk continuity
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if (inputSamples.isNotEmpty()) {
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lastSample = inputSamples.last()
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}
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return outputSamples
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}
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/**
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* Calculates the expected output size for a given input size.
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* Useful for pre-allocating buffers or estimating processing requirements.
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*
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* @param inputSize Number of input samples
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* @return Expected number of output samples
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*/
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fun calculateOutputSize(inputSize: Int): Int
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{
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return (inputSize * resampleRatio).roundToInt()
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}
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/**
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* Resets the resampler state, clearing interpolation continuity.
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* Call this when starting a new audio stream or after a discontinuity.
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*/
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fun reset()
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{
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lastSample = 0
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totalInputSamples = 0L
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totalOutputSamples = 0L
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Timber.v("AudioResampler state reset")
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}
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/**
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* Gets statistics about resampler performance.
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*
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* @return String with resampler statistics
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*/
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fun getStatistics(): String
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{
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val compressionRatio = if (totalInputSamples > 0)
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{
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totalOutputSamples.toDouble() / totalInputSamples.toDouble()
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}
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else
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{
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0.0
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}
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return "AudioResampler Stats: ${totalInputSamples} -> ${totalOutputSamples} samples " +
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"(ratio: %.3f, expected: %.3f)".format(compressionRatio, resampleRatio)
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}
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/**
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* Gets a sample for interpolation, handling edge cases.
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*
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* @param samples Input sample array
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* @param index Requested sample index
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* @return Sample value for interpolation
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*/
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private fun getSampleForInterpolation(samples: ShortArray, index: Int): Short
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{
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return when {
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index < 0 -> lastSample // Use last sample from previous chunk
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index >= samples.size -> samples.last() // Use last available sample
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else -> samples[index] // Normal case
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}
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}
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}

‎AudioCoder/src/main/jni/libloud.cpp‎

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WSPR_SYMBOL_COUNT * WSPR_SYMBOL_LENGTH);
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short volume = 16383;
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for (int i = 0; i < WSPR_SYMBOL_COUNT; i++) {
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if (lsb_mod) {
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symbols[i] = (uint8_t) 3 - symbols[i];
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}
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// Base band Carrier Frequency - 1500 Hz
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// Frequency spacing between the symbols - 1.4548
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double frequency = 1500 + ((int) j_offset) + symbols[i] * 1.4548;
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// TODO: Create a new function that converts frequency (double) to ints ( * 100 + casting to UInt64) to Bytes and returns a byte array of the frequencies
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// Frequency array size = # of symbols * 8 bytes (size of 64 bit integer)
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double theta = frequency * TAU / (double) 12000;
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// 'volume' is UInt16 with range 0 thru Uint16.MaxValue ( = 65 535)
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// we need 'amp' to have the range of 0 thru Int16.MaxValue ( = 32 767)

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