Microphone input

[Unseen Machine]

I think my mic is broken or something so can someone test this please...

Microphone.h 

Code: (Select All)

#ifndef QBLIVE_AUDIO_H

#define QBLIVE_AUDIO_H



#include <windows.h>

#include <mmsystem.h>

#include <shellapi.h>

#include <vector>



#pragma comment(lib, "winmm.lib")

#pragma comment(lib, "shell32.lib")



#define SAMPLE_RATE 44100

#define CHUNK_SIZE 4096



static HWAVEIN hWaveIn = NULL;

static HWAVEOUT hWaveOut = NULL;

static WAVEHDR waveHdrIn;

static short captureBuffer[CHUNK_SIZE];

static std::vector<short> recordedData;

static bool isRecording = false;



// Driver Callback: Background thread catches audio chunks as they arrive

void CALLBACK waveInProc(HWAVEIN hwi, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2) {

    if (uMsg == WIM_DATA && isRecording) {

        WAVEHDR* pHdr = (WAVEHDR*)dwParam1;

        short* raw = (short*)pHdr->lpData;

        // Efficiently append raw 16-bit samples to the storage vector

        recordedData.insert(recordedData.end(), raw, raw + (pHdr->dwBytesRecorded / 2));

        // Re-queue the buffer to keep recording without gaps

        waveInAddBuffer(hwi, pHdr, sizeof(WAVEHDR));

    }

}



extern "C" {

    __declspec(dllexport) int InitAudio() {

        // CD Quality: Mono, 44100Hz, 16-bit PCM

        WAVEFORMATEX wfx = { WAVE_FORMAT_PCM, 1, SAMPLE_RATE, SAMPLE_RATE * 2, 2, 16, 0 };

        

        // Open Microphone

        MMRESULT res = waveInOpen(&hWaveIn, WAVE_MAPPER, &wfx, (DWORD_PTR)waveInProc, 0, CALLBACK_FUNCTION);

        if (res != MMSYSERR_NOERROR) return (int)res;



        waveHdrIn.lpData = (LPSTR)captureBuffer;

        waveHdrIn.dwBufferLength = CHUNK_SIZE * 2;

        waveHdrIn.dwFlags = 0;

        waveInPrepareHeader(hWaveIn, &waveHdrIn, sizeof(WAVEHDR));

        

        // Open Speakers

        if (waveOutOpen(&hWaveOut, WAVE_MAPPER, &wfx, 0, 0, CALLBACK_NULL) != MMSYSERR_NOERROR) return -99;

        

        return 0;

    }



    __declspec(dllexport) void StartRecord() {

        recordedData.clear();

        isRecording = true;

        waveInAddBuffer(hWaveIn, &waveHdrIn, sizeof(WAVEHDR));

        waveInStart(hWaveIn);

    }



    __declspec(dllexport) void StopRecord() {

        isRecording = false;

        waveInStop(hWaveIn);

        waveInReset(hWaveIn);

    }



    __declspec(dllexport) int GetRecordedSize() {

        return (int)recordedData.size();

    }



    __declspec(dllexport) void PlayRecording() {

        if (recordedData.empty()) return;

        static WAVEHDR playHdr; // Static ensures memory persists during playback

        playHdr.lpData = (LPSTR)recordedData.data();

        playHdr.dwBufferLength = (DWORD)recordedData.size() * 2;

        playHdr.dwFlags = 0;

        waveOutPrepareHeader(hWaveOut, &playHdr, sizeof(WAVEHDR));

        waveOutWrite(hWaveOut, &playHdr, sizeof(WAVEHDR));

    }



    __declspec(dllexport) void OpenMicSettings() {

        ShellExecuteA(NULL, "open", "ms-settings:privacy-microphone", NULL, NULL, SW_SHOWNORMAL);

    }



    __declspec(dllexport) void ShutdownAudio() {

        if (hWaveIn) { waveInReset(hWaveIn); waveInClose(hWaveIn); }

        if (hWaveOut) { waveOutReset(hWaveOut); waveOutClose(hWaveOut); }

    }



        // Converts 16-bit samples to 32-bit floats for QB64 _SNDRAW

    // buffer = the QB64 float array, offset = starting sample, count = samples to get

    __declspec(dllexport) void Audio_Get_SndRaw(float* buffer, int offset, int count) {

        if (recordedData.empty() || offset < 0) return;

        

        for (int i = 0; i < count; i++) {

            int index = offset + i;

            if (index < (int)recordedData.size()) {

                // Convert signed 16-bit (-32768 to 32767) to float (-1.0 to 1.0)

                buffer[i] = (float)recordedData[index] / 32768.0f;

            } else {

                buffer[i] = 0.0f; // Fill remaining buffer with silence if out of range

            }

        }

    }



}

#endif

ANd the QB64 stuff 

Code: (Select All)

' Finalized QBLive Audio Studio 2025

DECLARE LIBRARY "Microphone"

  FUNCTION InitAudio& ()

  SUB StartRecord ()

  SUB StopRecord ()

  FUNCTION GetRecordedSize& ()

  SUB PlayRecording ()

  SUB OpenMicSettings ()

  SUB ShutdownAudio ()

  SUB Audio_Get_SndRaw (buffer AS SINGLE, BYVAL offset AS LONG, BYVAL count AS LONG)



END DECLARE



SCREEN _NEWIMAGE(800, 600, 32)

_TITLE "QBLive Audio Recorder 2025"



' 1. Check Permissions

res& = InitAudio

IF res& <> 0 THEN

  PRINT "--- AUDIO INITIALIZATION ERROR ---"

  PRINT "Result Code:"; res&

  PRINT "Hardware not found or Privacy Permissions denied."

  PRINT

  INPUT "Open Windows Microphone Settings? (Y/N) ", c$

  IF UCASE$(c$) = "Y" THEN OpenMicSettings

  END

END IF



isRecording = 0

PRINT "Ready. Press SPACE to Record."



DO

  k$ = INKEY$

  IF k$ = " " THEN

    IF isRecording = 0 THEN

      StartRecord

      isRecording = 1

    ELSE

      StopRecord

      isRecording = 0

      CLS

      PRINT "Playing back"; GetRecordedSize; "samples..."

      'PlayRecording ''// This maybe the error



      samples& = GetRecordedSize



      IF samples& > 0 THEN

        PRINT "Playing back via _SNDRAW..."



        ' We process in chunks of 4096 (matching your C++ CHUNK_SIZE)

        chunkSize& = 4096

        REDIM tempBuffer(chunkSize& - 1) AS SINGLE



        FOR currentOffset& = 0 TO samples& STEP chunkSize&

          ' 1. Fill our QB64 float buffer from the C++ recordedData vector

          Audio_Get_SndRaw tempBuffer(0), currentOffset&, chunkSize&



          ' 2. Call our standalone sub to handle the actual playback

          PlayFloatBuffer tempBuffer(), chunkSize&

        NEXT currentOffset&



        PRINT "Playback finished."

      END IF



      PRINT "Press SPACE to record again."

    END IF

    _DELAY .3 ' Toggle debounce

  END IF



  IF isRecording THEN

    LOCATE 1, 1: PRINT "RECORDING... ["; GetRecordedSize; "samples]"

  END IF

  _LIMIT 60

LOOP UNTIL _EXIT



ShutdownAudio

SYSTEM





SUB PlayFloatBuffer (floatArray() AS SINGLE, count AS LONG)

  ' Designed for 44100Hz Mono Audio

  IF count <= 0 THEN EXIT SUB



  FOR i& = 0 TO count - 1

    ' 1. Send the sample to _SNDRAW (mono automatically goes to both speakers)

    _SNDRAW floatArray(i&)



    ' 2. Check the sound pipe length.

    ' If we queue too much data (> 0.2 seconds), we wait.

    ' If we don't wait, QB64 will eat up all your RAM trying to store the samples.

    IF i& MOD 100 = 0 THEN ' Check every 100 samples to save CPU

      DO WHILE _SNDRAWLEN > 0.2

        _LIMIT 100 ' Give the CPU a rest while the sound card works

      LOOP

    END IF

  NEXT i&

END SUB

@Petr 90% there i reckon...needs a better brain!

Unseeen