From 0bfa877d9d952389b8c86e69e7ab00ca1ab39258 Mon Sep 17 00:00:00 2001 From: ptarabbia Date: Wed, 15 Dec 2021 17:47:55 -0500 Subject: [PATCH] Add audio processors to the streams system. --- examples/audio/audio_music_stream.c | 34 +++++++++ src/raudio.c | 103 +++++++++++++++++++++++----- src/raudio.h | 5 +- src/raylib.h | 5 +- 4 files changed, 125 insertions(+), 22 deletions(-) diff --git a/examples/audio/audio_music_stream.c b/examples/audio/audio_music_stream.c index 551116179..3ea6fcb74 100644 --- a/examples/audio/audio_music_stream.c +++ b/examples/audio/audio_music_stream.c @@ -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(); //---------------------------------------------------------------------------------- diff --git a/src/raudio.c b/src/raudio.c index 3759e5690..f40b6bdfa 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -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; diff --git a/src/raudio.h b/src/raudio.h index 1f3488d17..bc74895dc 100644 --- a/src/raudio.h +++ b/src/raudio.h @@ -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 diff --git a/src/raylib.h b/src/raylib.h index 3d0b88027..0a9d81660 100644 --- a/src/raylib.h +++ b/src/raylib.h @@ -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) }