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Add audio processors to the streams system.

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This commit is contained in:
ptarabbia 2021-12-15 17:47:55 -05:00
parent 551db0281d
commit 0bfa877d9d
4 changed files with 125 additions and 22 deletions
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34
examples/audio/audio_music_stream.c
View File

@ -14,6 +14,23 @@
#define AUDIO_THREAD_MUSIC_UPDATE true
// a simple lowpass filter applied to the music stream
static void audioEffectDemo(float* buffer, unsigned int nframes)
{
static float low[2] = { 0.0f, 0.0f };
static const float cutoff = 70.0f / 44100.0f; // 70 Hz lowpass filter
const float k = cutoff / (cutoff + 0.1591549431f); // RC filter formula
for (unsigned int i = 0; i < nframes*2; i+=2)
{
float l = buffer[i], r = buffer[i + 1];
low[0] += k * (l - low[0]);
low[1] += k * (r - low[1]);
buffer[i] = low[0];
buffer[i + 1] = low[1];
}
}
int main(void)
{
// Initialization
@ -32,6 +49,7 @@ int main(void)
float timePlayed = 0.0f;
bool pause = false;
bool hasfx = false;
SetTargetFPS(60); // Set our game to run at 60 frames-per-second
//--------------------------------------------------------------------------------------
@ -62,6 +80,20 @@ int main(void)
else ResumeMusicStream(music);
}
// Add/Remove effect
if (IsKeyPressed(KEY_F))
{
hasfx = !hasfx;
if (hasfx)
{
AddAudioStreamProcessor(music.stream, &audioEffectDemo);
}
else
{
RemoveAudioStreamProcessor(music.stream, &audioEffectDemo);
}
}
// Get timePlayed scaled to bar dimensions (400 pixels)
timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*400;
@ -79,9 +111,11 @@ int main(void)
DrawRectangle(200, 200, 400, 12, LIGHTGRAY);
DrawRectangle(200, 200, (int)timePlayed, 12, MAROON);
DrawRectangleLines(200, 200, 400, 12, GRAY);
DrawRectangleLines(200, 200, 400, 12, GRAY);
DrawText("PRESS SPACE TO RESTART MUSIC", 215, 250, 20, LIGHTGRAY);
DrawText("PRESS P TO PAUSE/RESUME MUSIC", 208, 280, 20, LIGHTGRAY);
DrawText("PRESS F TO ADD/REMOVE EFFECT", 208, 310, 20, LIGHTGRAY);
EndDrawing();
//----------------------------------------------------------------------------------

103
src/raudio.c
View File

@ -311,12 +311,22 @@ typedef enum {
AUDIO_BUFFER_USAGE_STREAM
} AudioBufferUsage;
// Audio processor - to apply effects to an AudioBuffer
struct rAudioProcessor {
void (*process)(float*, unsigned int);
rAudioProcessor *next;
rAudioProcessor *prev;
};
// Audio buffer structure
struct rAudioBuffer {
ma_data_converter converter; // Audio data converter
void (*audioCallback)(void*, unsigned int); // optional callback for filling in buffer on audio threads
void (*audioCallback)(void*, unsigned int, void*); // optional callback for filling in buffer on audio threads
void* audioCallbackData; // optional data passed to callback
rAudioProcessor* processor;
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
@ -349,7 +359,6 @@ typedef struct AudioData {
} System;
struct {
AudioBuffer *first; // Pointer to first AudioBuffer in the list
AudioBuffer *last; // Pointer to last AudioBuffer in the list
int defaultSize; // Default audio buffer size for audio streams
} Buffer;
struct {
@ -399,7 +408,7 @@ void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
void PauseAudioBuffer(AudioBuffer *buffer);
void ResumeAudioBuffer(AudioBuffer *buffer);
void SetAudioBufferCallback(AudioBuffer* buffer, int callback(void*));
void SetAudioBufferCallback(AudioBuffer* buffer, void callback(void*, unsigned int, void*), void*);
void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
void TrackAudioBuffer(AudioBuffer *buffer);
@ -532,7 +541,6 @@ void SetMasterVolume(float volume)
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
if (audioBuffer == NULL)
{
TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer");
@ -559,6 +567,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
audioBuffer->pitch = 1.0f;
audioBuffer->audioCallback = NULL;
audioBuffer->audioCallbackData = NULL;
audioBuffer->processor = NULL;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->looping = false;
@ -679,14 +688,10 @@ void TrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
{
if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
else
{
AUDIO.Buffer.last->next = buffer;
buffer->prev = AUDIO.Buffer.last;
}
AUDIO.Buffer.last = buffer;
AudioBuffer* old = AUDIO.Buffer.first;
buffer->next = old;
if (old) old->prev = buffer;
AUDIO.Buffer.first = buffer;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
@ -696,11 +701,11 @@ void UntrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
{
if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
else buffer->prev->next = buffer->next;
if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
else buffer->next->prev = buffer->prev;
AudioBuffer* p = buffer->prev;
AudioBuffer* n = buffer->next;
if (p) p->next = n;
if (n) n->prev = p;
if (buffer == AUDIO.Buffer.first) AUDIO.Buffer.first = n;
buffer->prev = NULL;
buffer->next = NULL;
@ -1004,7 +1009,7 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
// Play a sound
void PlaySound(Sound sound)
{
PlayAudioBuffer(sound.stream.buffer);
PlayAudioStream(sound.stream);
}
// Play a sound in the multichannel buffer pool
@ -1059,6 +1064,7 @@ void PlaySoundMulti(Sound sound)
AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
AUDIO.MultiChannel.pool[index]->audioCallback = sound.stream.buffer->audioCallback;
AUDIO.MultiChannel.pool[index]->audioCallbackData = sound.stream.buffer->audioCallbackData;
AUDIO.MultiChannel.pool[index]->processor = sound.stream.buffer->processor;
AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
@ -1621,7 +1627,7 @@ static void MusicStreamCallbackWav(void* buffer, unsigned int nframes, void* dat
drwav* ctx = (drwav*)data;
// NOTE: Returns the number of samples to process (not required)
if (ctx->bitsPerSample == 16) drwav_read_pcm_frames_s16(ctx, nframes, (short*)buffer);
else if (ctx->bitsPerSample == 32) drwav_read_pcm_frames_f32(ctx, nframes, (short*)buffer);
else if (ctx->bitsPerSample == 32) drwav_read_pcm_frames_f32(ctx, nframes, (float*)buffer);
}
#endif
@ -2119,6 +2125,57 @@ void SetAudioStreamCallback(AudioStream stream, void callback(void*, unsigned in
}
}
// Add processor to audio stream. Contrary to buffers, the order of processors is important.
// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element.
void AddAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int))
{
ma_mutex_lock(&AUDIO.System.lock); // necessary evil.
rAudioProcessor* processor = (rAudioProcessor*)RL_CALLOC(1, sizeof(rAudioProcessor));
processor->process = process;
rAudioProcessor* last = stream.buffer->processor;
while (last && last->next)
{
last = last->next;
}
if (last)
{
processor->prev = last;
last->next = processor;
}
else
{
stream.buffer->processor = processor;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
void RemoveAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int))
{
ma_mutex_lock(&AUDIO.System.lock);
rAudioProcessor* processor = stream.buffer->processor;
while (processor)
{
rAudioProcessor* next = processor->next;
rAudioProcessor* prev = processor->prev;
if (processor->process == process)
{
if (stream.buffer->processor == processor) stream.buffer->processor = next;
if (prev) prev->next = next;
if (next) next->prev = prev;
RL_FREE(processor);
}
processor = next;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
@ -2315,6 +2372,14 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
float *framesIn = tempBuffer;
// apply processors chain
rAudioProcessor *processor = audioBuffer->processor;
while (processor)
{
processor->process(framesIn, framesJustRead);
processor = processor->next;
}
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead;

5
src/raudio.h
View File

@ -88,11 +88,11 @@ typedef struct Wave {
} Wave;
typedef struct rAudioBuffer rAudioBuffer;
typedef struct rAudioProcessor rAudioProcessor;
// AudioStream, custom audio stream
typedef struct AudioStream {
rAudioBuffer *buffer; // Pointer to internal data used by the audio system
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
@ -191,7 +191,8 @@ void SetAudioStreamVolume(AudioStream stream, float volume); // Set volume fo
void SetAudioStreamPitch(AudioStream stream, float pitch); // Set pitch for audio stream (1.0 is base level)
void SetAudioStreamBufferSizeDefault(int size); // Default size for new audio streams
void SetAudioStreamCallback(AudioStream stream, void callback(void*, unsigned int, void*), void* callbackData); // Audio thread callback to request new data
void AddAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int));
void RemoveAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int));
#ifdef __cplusplus
}
#endif

5
src/raylib.h
View File

@ -426,11 +426,12 @@ typedef struct Wave {
} Wave;
typedef struct rAudioBuffer rAudioBuffer;
typedef struct rAudioProcessor rAudioProcessor;
// AudioStream, custom audio stream
typedef struct AudioStream {
rAudioBuffer *buffer; // Pointer to internal data used by the audio system
rAudioProcessor* processor;
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
@ -1531,6 +1532,8 @@ RLAPI void SetAudioStreamVolume(AudioStream stream, float volume); // Set vol
RLAPI void SetAudioStreamPitch(AudioStream stream, float pitch); // Set pitch for audio stream (1.0 is base level)
RLAPI void SetAudioStreamBufferSizeDefault(int size); // Default size for new audio streams
RLAPI void SetAudioStreamCallback(AudioStream stream, void callback(void*, unsigned int, void*), void* callbackData); // Audio thread callback to request new data
RLAPI void AddAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int));
RLAPI void RemoveAudioStreamProcessor(AudioStream stream, void (*process)(float*, unsigned int));
#if defined(__cplusplus)
}