MilkDrop3/code/audio/audiobuf.cpp

// audiobuf.cpp

#include "audiobuf.h"

#define SAMPLE_SIZE_LPB 576 // Max number of audio samples stored in circular buffer. Should be no less than SAMPLE_SIZE. Expected sampling rate is 44100 Hz or 48000 Hz (samples per second).

std::mutex pcmLpbMutex;
unsigned char pcmLeftLpb[SAMPLE_SIZE_LPB]; // Circular buffer (left channel)
unsigned char pcmRightLpb[SAMPLE_SIZE_LPB]; // Circular buffer (right channel)
bool pcmBufDrained = false; // Buffer drained by visualization thread and holds no new samples
signed int pcmLen = 0; // Actual number of samples the buffer holds. Can be less than SAMPLE_SIZE_LPB
signed int pcmPos = 0; // Position to write new data

void ResetAudioBuf() {
    std::unique_lock<std::mutex> lock(pcmLpbMutex);
    memset(pcmLeftLpb, 0, SAMPLE_SIZE_LPB);
    memset(pcmRightLpb, 0, SAMPLE_SIZE_LPB);
    pcmBufDrained = false;
    pcmLen = 0;
}

void GetAudioBuf(unsigned char *pWaveL, unsigned char *pWaveR, int SamplesCount) {
    std::unique_lock<std::mutex> lock(pcmLpbMutex);
    if ((pcmLen < SamplesCount) || (pcmBufDrained)) {
        // Buffer underrun. Insufficient new samples in circular buffer (pcmLeftLpb, pcmRightLpb)
        memset(pWaveL, 0, SamplesCount);
        memset(pWaveR, 0, SamplesCount);
    }
    else {
        // Circular buffer (pcmLeftLpb, pcmRightLpb) hold enough samples in it
        for (int i = 0; i < SAMPLE_SIZE_LPB; i++) {
            // int8_t [-128 .. +127] stored into uint8_t [0..255]
            pWaveL[i % SamplesCount] = pcmLeftLpb[(pcmPos + i) % SAMPLE_SIZE_LPB];
            pWaveR[i % SamplesCount] = pcmRightLpb[(pcmPos + i) % SAMPLE_SIZE_LPB];
        }
        pcmBufDrained = true;
    }
}

int8_t FltToInt(float flt) {
    if (flt >= 1.0f) {
        return +127; // 0x7f
    }
    if (flt < -1.0f) {
        return -128; // 0x80
    }
    return (int8_t)(flt * 128);
};

// Union type for sample conversion
union u_type
{
    int32_t IntVar;
    float FltVar;
    uint8_t Bytes[4];
};

int8_t GetChannelSample(const BYTE *pData, int BlockOffset, int ChannelOffset, const bool bInt16) {
    u_type sample;

    sample.IntVar = 0;
    sample.Bytes[0] = pData[BlockOffset + ChannelOffset + 0];
    sample.Bytes[1] = pData[BlockOffset + ChannelOffset + 1];
    if (!bInt16) {
        sample.Bytes[2] = pData[BlockOffset + ChannelOffset + 2];
        sample.Bytes[3] = pData[BlockOffset + ChannelOffset + 3];
    }

    if (!bInt16) {
        return FltToInt(sample.FltVar); //float [-1.0f .. +1.0f] range converted to int8_t [-128 .. +127] and later stored into uint8_t [0 .. 255]
    }
    else {
        return (signed char)(sample.IntVar / 256); //int16_t [-32768 .. +32767] range converted to int8_t [-128 .. +127] and later stored into uint8_t [0..255]
    }
}

// Expecting pData holds:
//   signed 16-bit (2 bytes) PCM, Little Endian
//   or
//   32-bit float (4 bytes) PCM
// Supported audio formats:
//   pwfx->nChannels;          /* ANY number of channels (i.e. mono, stereo...) */
//   pwfx->nSamplesPerSec;     /* 44100 or 48000 sample rate */
//   pwfx->nBlockAlign;        /* ANY block size of data */
//   pwfx->wBitsPerSample;     /* 16 or 32 number of bits per sample of mono data */

void SetAudioBuf(const BYTE *pData, const UINT32 nNumFramesToRead, const WAVEFORMATEX *pwfx, const bool bInt16) {
    int BlockOffset;

    int8_t LeftSample8;
    int8_t RightSample8;


    std::unique_lock<std::mutex> lock(pcmLpbMutex);
    //memset(pcmLeftLpb, 0, SAMPLE_SIZE_LPB);
    //memset(pcmRightLpb, 0, SAMPLE_SIZE_LPB);

    int i = 0;
    int n = 0;

    int start = 0;
    int len = 0;
    if (nNumFramesToRead >= SAMPLE_SIZE_LPB) {
        n = 0;
        start = nNumFramesToRead - SAMPLE_SIZE_LPB;
        len = SAMPLE_SIZE_LPB;
    }
    else {
        n = SAMPLE_SIZE_LPB - nNumFramesToRead;
        start = 0;
        len = nNumFramesToRead;
    }

    for (int i = start; i < len; i++, n++) {
        BlockOffset = i * pwfx->nBlockAlign;

        // Left channel (number 0)
        LeftSample8 = 0; // Init with silence
        if (pwfx->nChannels >= 1) {
            LeftSample8 = GetChannelSample(pData, BlockOffset, 0 * (pwfx->wBitsPerSample / 8), bInt16);
        }

        // Right channel (number 1)
        RightSample8 = LeftSample8; // Init with left channel value just in case of Mono, 1 channel count
        if (pwfx->nChannels >= 2) {
            RightSample8 = GetChannelSample(pData, BlockOffset, 1 * (pwfx->wBitsPerSample / 8), bInt16);
        }

        // TODO: add support for 96000 Hz and 192000 Hz sample rates

        // Ignoring data in all other audio channels (Quadraphonic 4.0, Surround 4.0, Surround 5.1, Surround 7.1, ...)

        // Saving audio data for visualizer
        // 8-bit signed integer in Two's Complement Representation stored in unsigned char array
        // int8_t[-128 .. + 127] stored into uint8_t[0 .. 255]
        pcmLeftLpb[(pcmPos + n) % SAMPLE_SIZE_LPB] = LeftSample8;
        pcmRightLpb[(pcmPos + n) % SAMPLE_SIZE_LPB] = RightSample8;
    }

    pcmBufDrained = false;
    pcmLen = (pcmLen + len <= SAMPLE_SIZE_LPB) ? (pcmLen + len) : (SAMPLE_SIZE_LPB);
    pcmPos = (pcmPos + len) % SAMPLE_SIZE_LPB;

}