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Improve AudioCallback

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This commit is contained in:
RealBitdancer 2024-07-19 23:34:27 -07:00
commit 615e28fa6d
3 changed files with 53 additions and 20 deletions
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6
examples/audio/audio_mixed_processor.c
View file

@ -21,7 +21,7 @@ static float averageVolume[400] = { 0.0f }; // Average volume history
//------------------------------------------------------------------------------------
// Audio processing function
//------------------------------------------------------------------------------------
void ProcessAudio(void *buffer, unsigned int frames)
unsigned int ProcessAudio(void *buffer, unsigned int frames, void* userData)
{
float *samples = (float *)buffer; // Samples internally stored as <float>s
float average = 0.0f; // Temporary average volume
@ -41,6 +41,8 @@ void ProcessAudio(void *buffer, unsigned int frames)
for (int i = 0; i < 399; i++) averageVolume[i] = averageVolume[i + 1];
averageVolume[399] = average; // Adding last average value
return frames;
}
//------------------------------------------------------------------------------------
@ -57,7 +59,7 @@ int main(void)
InitAudioDevice(); // Initialize audio device
AttachAudioMixedProcessor(ProcessAudio);
AttachAudioMixedProcessor(ProcessAudio, NULL);
Music music = LoadMusicStream("resources/country.mp3");
Sound sound = LoadSound("resources/coin.wav");

6
examples/audio/audio_raw_stream.c
View file

@ -35,7 +35,7 @@ float oldFrequency = 1.0f;
float sineIdx = 0.0f;
// Audio input processing callback
void AudioInputCallback(void *buffer, unsigned int frames)
unsigned int AudioInputCallback(void *buffer, unsigned int frames, void* userData)
{
audioFrequency = frequency + (audioFrequency - frequency)*0.95f;
@ -48,6 +48,8 @@ void AudioInputCallback(void *buffer, unsigned int frames)
sineIdx += incr;
if (sineIdx > 1.0f) sineIdx -= 1.0f;
}
return frames;
}
//------------------------------------------------------------------------------------
@ -69,7 +71,7 @@ int main(void)
// Init raw audio stream (sample rate: 44100, sample size: 16bit-short, channels: 1-mono)
AudioStream stream = LoadAudioStream(44100, 16, 1);
SetAudioStreamCallback(stream, AudioInputCallback);
SetAudioStreamCallback(stream, AudioInputCallback, NULL);
// Buffer for the single cycle waveform we are synthesizing
short *data = (short *)malloc(sizeof(short)*MAX_SAMPLES);

16
examples/audio/audio_stream_effects.c
View file

@ -24,8 +24,8 @@ static unsigned int delayWriteIndex = 0;
//------------------------------------------------------------------------------------
// Module Functions Declaration
//------------------------------------------------------------------------------------
static void AudioProcessEffectLPF(void *buffer, unsigned int frames); // Audio effect: lowpass filter
static void AudioProcessEffectDelay(void *buffer, unsigned int frames); // Audio effect: delay
static unsigned int AudioProcessEffectLPF(void *buffer, unsigned int frames, void* userData); // Audio effect: lowpass filter
static unsigned int AudioProcessEffectDelay(void *buffer, unsigned int frames, void* userData); // Audio effect: delay
//------------------------------------------------------------------------------------
// Program main entry point
@ -85,7 +85,7 @@ int main(void)
if (IsKeyPressed(KEY_F))
{
enableEffectLPF = !enableEffectLPF;
if (enableEffectLPF) AttachAudioStreamProcessor(music.stream, AudioProcessEffectLPF);
if (enableEffectLPF) AttachAudioStreamProcessor(music.stream, AudioProcessEffectLPF, NULL);
else DetachAudioStreamProcessor(music.stream, AudioProcessEffectLPF);
}
@ -93,7 +93,7 @@ int main(void)
if (IsKeyPressed(KEY_D))
{
enableEffectDelay = !enableEffectDelay;
if (enableEffectDelay) AttachAudioStreamProcessor(music.stream, AudioProcessEffectDelay);
if (enableEffectDelay) AttachAudioStreamProcessor(music.stream, AudioProcessEffectDelay, NULL);
else DetachAudioStreamProcessor(music.stream, AudioProcessEffectDelay);
}
@ -143,7 +143,7 @@ int main(void)
// Module Functions Definition
//------------------------------------------------------------------------------------
// Audio effect: lowpass filter
static void AudioProcessEffectLPF(void *buffer, unsigned int frames)
static unsigned int AudioProcessEffectLPF(void *buffer, unsigned int frames, void* userData)
{
static float low[2] = { 0.0f, 0.0f };
static const float cutoff = 70.0f / 44100.0f; // 70 Hz lowpass filter
@ -161,10 +161,12 @@ static void AudioProcessEffectLPF(void *buffer, unsigned int frames)
bufferData[i] = low[0];
bufferData[i + 1] = low[1];
}
return frames;
}
// Audio effect: delay
static void AudioProcessEffectDelay(void *buffer, unsigned int frames)
static unsigned int AudioProcessEffectDelay(void *buffer, unsigned int frames, void* userData)
{
for (unsigned int i = 0; i < frames*2; i += 2)
{
@ -180,4 +182,6 @@ static void AudioProcessEffectDelay(void *buffer, unsigned int frames)
delayBuffer[delayWriteIndex++] = ((float *)buffer)[i + 1];
if (delayWriteIndex == delayBufferSize) delayWriteIndex = 0;
}
return frames;
}