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ADDED: Audio stream processors support -WIP- #2212

This feature is still under consideration/testing and it doesn't work properly, at least the Delay Effect processor.
pull/2423/head
Ray 2 anni fa
parent
commit
1612ba63ab
3 ha cambiato i file con 140 aggiunte e 0 eliminazioni
  1. +65
    -0
      examples/audio/audio_music_stream.c
  2. +70
    -0
      src/raudio.c
  3. +5
    -0
      src/raylib.h

+ 65
- 0
examples/audio/audio_music_stream.c Vedi File

@ -11,6 +11,47 @@
#include "raylib.h"
// Audio effect: lowpass filter
static void AudioProcessEffectLPF(float *buffer, unsigned int frames)
{
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 < frames*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];
}
}
static float *delayBuffer = NULL;
static unsigned int delayBufferSize = 0;
static unsigned int delayReadIndex = 2;
static unsigned int delayWriteIndex = 0;
// Audio effect: delay
static void AudioProcessEffectDelay(float *buffer, unsigned int frames)
{
for (unsigned int i = 0; i < frames*2; i += 2)
{
float leftDelay = delayBuffer[delayReadIndex++]; // ERROR: Reading buffer -> WHY??? Maybe thread related???
float rightDelay = delayBuffer[delayReadIndex++];
if (delayReadIndex == delayBufferSize) delayReadIndex = 0;
buffer[i] = 0.5f*buffer[i] + 0.5f*leftDelay;
buffer[i + 1] = 0.5f*buffer[i + 1] + 0.5f*rightDelay;
delayBuffer[delayWriteIndex++] = buffer[i];
delayBuffer[delayWriteIndex++] = buffer[i + 1];
if (delayWriteIndex == delayBufferSize) delayWriteIndex = 0;
}
}
int main(void)
{
// Initialization
@ -24,11 +65,17 @@ int main(void)
Music music = LoadMusicStream("resources/country.mp3");
// Allocate buffer for the delay effect
delayBuffer = (float *)RL_CALLOC(48000*2, sizeof(float)); // 1 second delay (device sampleRate*channels)
PlayMusicStream(music);
float timePlayed = 0.0f;
bool pause = false;
bool hasFilter = false;
bool hasDelay = false;
SetTargetFPS(60); // Set our game to run at 60 frames-per-second
//--------------------------------------------------------------------------------------
@ -55,6 +102,22 @@ int main(void)
else ResumeMusicStream(music);
}
// Add/Remove effect: lowpass filter
if (IsKeyPressed(KEY_F))
{
hasFilter = !hasFilter;
if (hasFilter) AttachAudioStreamProcessor(music.stream, AudioProcessEffectLPF);
else DetachAudioStreamProcessor(music.stream, AudioProcessEffectLPF);
}
// Add/Remove effect: delay
if (IsKeyPressed(KEY_D))
{
hasDelay = !hasDelay;
if (hasDelay) AttachAudioStreamProcessor(music.stream, AudioProcessEffectDelay);
else DetachAudioStreamProcessor(music.stream, AudioProcessEffectDelay);
}
// Get timePlayed scaled to bar dimensions (400 pixels)
timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*400;
@ -86,6 +149,8 @@ int main(void)
CloseAudioDevice(); // Close audio device (music streaming is automatically stopped)
RL_FREE(delayBuffer); // Free delay buffer
CloseWindow(); // Close window and OpenGL context
//--------------------------------------------------------------------------------------

+ 70
- 0
src/raudio.c Vedi File

@ -316,6 +316,7 @@ struct rAudioBuffer {
ma_data_converter converter; // Audio data converter
AudioCallback callback; // Audio buffer callback for buffer filling on audio threads
rAudioProcessor *processor; // Audio processor
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
@ -337,6 +338,14 @@ struct rAudioBuffer {
rAudioBuffer *prev; // Previous audio buffer on the list
};
// Audio processor struct
// NOTE: Useful to apply effects to an AudioBuffer
struct rAudioProcessor {
AudioCallback process; // Processor callback function
rAudioProcessor *next; // Next audio processor on the list
rAudioProcessor *prev; // Previous audio processor on the list
};
#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision
// Audio data context
@ -559,6 +568,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
audioBuffer->pan = 0.5f;
audioBuffer->callback = NULL;
audioBuffer->processor = NULL;
audioBuffer->playing = false;
audioBuffer->paused = false;
@ -2039,6 +2049,58 @@ void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
if (stream.buffer != NULL) stream.buffer->callback = callback;
}
// 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 AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
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 DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
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
//----------------------------------------------------------------------------------
@ -2235,6 +2297,14 @@ 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 if defined
rAudioProcessor *processor = audioBuffer->processor;
while (processor)
{
processor->process(framesIn, framesJustRead);
processor = processor->next;
}
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer);
framesToRead -= framesJustRead;

+ 5
- 0
src/raylib.h Vedi File

@ -428,10 +428,12 @@ typedef struct Wave {
// Opaque structs declaration
// NOTE: Actual structs are defined internally in raudio module
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; // Pointer to internal data processor, useful for audio effects
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
@ -1543,6 +1545,9 @@ RLAPI void SetAudioStreamPan(AudioStream stream, float pan); // Set pan
RLAPI void SetAudioStreamBufferSizeDefault(int size); // Default size for new audio streams
RLAPI void SetAudioStreamCallback(AudioStream stream, AudioCallback callback); // Audio thread callback to request new data
RLAPI void AttachAudioStreamProcessor(AudioStream stream, AudioCallback processor);
RLAPI void DetachAudioStreamProcessor(AudioStream stream, AudioCallback processor);
#if defined(__cplusplus)
}
#endif

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