diff --git a/src/audio.c b/src/audio.c index 119b057d..74a54b04 100644 --- a/src/audio.c +++ b/src/audio.c @@ -459,7 +459,15 @@ void SetSoundPitch(Sound sound, float pitch) void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { // Format sample rate - if (wave->sampleRate != sampleRate) wave->sampleRate = sampleRate; + // NOTE: Only supported 22050 <--> 44100 + if (wave->sampleRate != sampleRate) + { + // TODO: Resample wave data (upsampling or downsampling) + // NOTE 1: To downsample, you have to drop samples or average them. + // NOTE 2: To upsample, you have to interpolate new samples. + + wave->sampleRate = sampleRate; + } // Format sample size // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit @@ -1078,23 +1086,23 @@ static Wave LoadWAV(const char *fileName) else { // Allocate memory for data - wave.data = (unsigned char *)malloc(sizeof(unsigned char)*wavData.subChunkSize); + wave.data = malloc(wavData.subChunkSize); // Read in the sound data into the soundData variable - fread(wave.data, wavData.subChunkSize, 1, wavFile); + fread(wave.data, 1, wavData.subChunkSize, wavFile); // Store wave parameters wave.sampleRate = wavFormat.sampleRate; wave.sampleSize = wavFormat.bitsPerSample; wave.channels = wavFormat.numChannels; - // NOTE: Only support up to 16 bit sample sizes - if (wave.sampleSize > 16) + // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes + if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32)) { - WaveFormat(&wave, wave.sampleRate, 16, wave.channels); TraceLog(WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize); + WaveFormat(&wave, wave.sampleRate, 16, wave.channels); } - + // NOTE: Only support up to 2 channels (mono, stereo) if (wave.channels > 2) {