SDL Audio Pitch - Playback Rate - c++

Question

My goal is to connect the RPM of an engine to the pitch of an sound. I am using SDL as my audio Backend.
So my idea was to sample from the wave buffer quicker than normal. So by trail and error I am now able to pitch my engine sound "step by step".
Question #1
If I change this part from:
audioBuff += 1 + pitch * 2;
to
audioBuff += 2
I get just noise. Why? Does this have to do with stereo channels?
Question #2
How can I make this a linear pitch? Currently it's a "stepping" pitch.
Here is the full code:
#include "SDL2/SDL.h"
#include <iostream>
void audioCallback(void* userdata, Uint8 *stream, int len);
Uint8 *audioBuff = nullptr;
Uint8 *audioBuffEnd = nullptr;
Uint32 audioLen = 0;
bool quit = false;
Uint16 pitch = 0;
int main()
{
if(SDL_Init(SDL_INIT_AUDIO) < 0)
return -1;
Uint32 wavLen = 0;
Uint8 *wavBuff = nullptr;
SDL_AudioSpec wavSpec;
if(SDL_LoadWAV("test.wav", &wavSpec, &wavBuff, &wavLen) == nullptr)
{
return 1;
}
wavSpec.callback = audioCallback;
wavSpec.userdata = nullptr;
wavSpec.format = AUDIO_S16;
wavSpec.samples = 2048;
audioBuff = wavBuff;
audioBuffEnd = &wavBuff[wavLen];
audioLen = wavLen;
if( SDL_OpenAudio(&wavSpec, NULL) < 0)
{
fprintf(stderr, "Could not open audio: %s\n", SDL_GetError());
return 1;
}
SDL_PauseAudio(0);
while(!quit)
{
SDL_Delay(500);
pitch ++;
}
SDL_CloseAudio();
SDL_FreeWAV(wavBuff);
return 0;
}
Uint32 sampleIndex = 0;
void audioCallback(void* userdata, Uint8 *stream, int len)
{
Uint32 length = (Uint32)len;
length = (length > audioLen ? audioLen : length);
for(Uint32 i = 0; i < length; i++)
{
if(audioBuff > audioBuffEnd)
{
quit = true;
return;
}
// why pitch * 2?
// how to get a smooth pitch?
stream[i] = audioBuff[0];
audioBuff += 1 + pitch * 2;
fprintf(stdout, "pitch: %u\n", pitch);
}
}

Answer 1

You're setting the audio format to AUDIO_S16, which is "Signed 16-bit little-endian samples". Each sample is two bytes, with the first byte being the LSB. When you read the data in audioCallback, you're reading it as bytes (8 bits), then passing those bytes back to something expecting 16 bits. You're getting noise because of this, and when you use audioBuff +=2; you're always reading the LSB of the audio sample, which essentially is noise when used that way.
You should consistently use either 16 bit or 8 bit samples.