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Simplified module for Music and AudioStream
Added support for raw audio streaming (with example)
This commit is contained in:
raysan5 2016-08-02 17:32:24 +02:00
parent 58d2f70b7e
commit 68d647c1af
10 changed files with 283 additions and 127 deletions
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2
examples/audio_module_playing.c
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@ -57,7 +57,7 @@ int main()
// Create a RenderTexture2D to be used for render to texture
RenderTexture2D target = LoadRenderTexture(screenWidth, screenHeight);
Music xm = LoadMusicStream("resources/audio/2t2m_spa.xm");
Music xm = LoadMusicStream("resources/audio/mini1111.xm");
PlayMusicStream(xm);

3
examples/audio_music_stream.c
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@ -59,9 +59,6 @@ int main()
SetMusicVolume(volume);
}
*/
if (IsWindowMinimized()) PauseMusicStream(music);
else ResumeMusicStream(music);
timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*100*4; // We scale by 4 to fit 400 pixels
UpdateMusicStream(music); // Update music buffer with new stream data

106
examples/audio_raw_stream.c Normal file
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@ -0,0 +1,106 @@
/*******************************************************************************************
*
* raylib [audio] example - Raw audio streaming
*
* NOTE: This example requires OpenAL Soft library installed
*
* This example has been created using raylib 1.6 (www.raylib.com)
* raylib is licensed under an unmodified zlib/libpng license (View raylib.h for details)
*
* Copyright (c) 2015 Ramon Santamaria (@raysan5)
*
********************************************************************************************/
#include "raylib.h"
#include <stdlib.h> // Required for: malloc(), free()
#include <math.h> // Required for: sinf()
int main()
{
// Initialization
//--------------------------------------------------------------------------------------
int screenWidth = 800;
int screenHeight = 450;
SetConfigFlags(FLAG_MSAA_4X_HINT);
InitWindow(screenWidth, screenHeight, "raylib [audio] example - raw audio streaming");
InitAudioDevice(); // Initialize audio device
AudioStream stream = InitAudioStream(44100, 32, 1); // Init raw audio stream
// Fill audio stream with some samples (sine wave)
float *data = (float *)malloc(sizeof(float)*44100);
for (int i = 0; i < 44100; i++)
{
data[i] = sinf(2*PI*(float)i*DEG2RAD);
}
PlayAudioStream(stream);
int totalSamples = 44100;
int samplesLeft = totalSamples;
Vector2 position = { 0, 0 };
SetTargetFPS(30); // Set our game to run at 30 frames-per-second
//--------------------------------------------------------------------------------------
// Main game loop
while (!WindowShouldClose()) // Detect window close button or ESC key
{
// Update
//----------------------------------------------------------------------------------
// Refill audio stream if required
if (IsAudioBufferProcessed(stream))
{
int numSamples = 0;
if (samplesLeft >= 4096) numSamples = 4096;
else numSamples = samplesLeft;
UpdateAudioStream(stream, data + (totalSamples - samplesLeft), numSamples);
samplesLeft -= numSamples;
// Reset samples feeding (loop audio)
if (samplesLeft <= 0) samplesLeft = totalSamples;
}
//----------------------------------------------------------------------------------
// Draw
//----------------------------------------------------------------------------------
BeginDrawing();
ClearBackground(RAYWHITE);
DrawText("SINE WAVE SHOULD BE PLAYING!", 240, 140, 20, LIGHTGRAY);
// NOTE: Draw a part of the sine wave (only screen width)
for (int i = 0; i < GetScreenWidth(); i++)
{
position.x = i;
position.y = 250 + 50*data[i];
DrawPixelV(position, RED);
}
EndDrawing();
//----------------------------------------------------------------------------------
}
// De-Initialization
//--------------------------------------------------------------------------------------
free(data); // Unload sine wave data
CloseAudioStream(stream); // Close raw audio stream and delete buffers from RAM
CloseAudioDevice(); // Close audio device (music streaming is automatically stopped)
CloseWindow(); // Close window and OpenGL context
//--------------------------------------------------------------------------------------
return 0;
}

1
examples/audio_sound_loading.c
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@ -36,7 +36,6 @@ int main()
// Update
//----------------------------------------------------------------------------------
if (IsKeyPressed(KEY_SPACE)) PlaySound(fxWav); // Play WAV sound
if (IsKeyPressed(KEY_ENTER)) PlaySound(fxOgg); // Play OGG sound
//----------------------------------------------------------------------------------

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examples/resources/audio/2t2m_spa.xm
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examples/resources/audio/chiptun1.mod Normal file
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examples/resources/audio/mini1111.xm Normal file
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218
src/audio.c
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@ -100,17 +100,6 @@
typedef enum { MUSIC_AUDIO_OGG = 0, MUSIC_MODULE_XM, MUSIC_MODULE_MOD } MusicContextType;
// Used to create custom audio streams that are not bound to a specific file.
typedef struct AudioStream {
unsigned int sampleRate; // Frequency (samples per second): default is 48000
unsigned int sampleSize; // BitDepth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels
ALenum format; // OpenAL format specifier
ALuint source; // OpenAL source
ALuint buffers[MAX_STREAM_BUFFERS]; // OpenAL buffers (double buffering)
} AudioStream;
// Music type (file streaming from memory)
typedef struct Music {
MusicContextType ctxType; // Type of music context (OGG, XM, MOD)
@ -118,7 +107,7 @@ typedef struct Music {
jar_xm_context_t *ctxXm; // XM chiptune context
jar_mod_context_t ctxMod; // MOD chiptune context
AudioStream stream; // Audio stream
AudioStream stream; // Audio stream (double buffering)
bool loop; // Repeat music after finish (loop)
unsigned int totalSamples; // Total number of samples
@ -141,12 +130,6 @@ static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(Music music, int numBuffersToProcess); // Fill music buffers with data
static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels);
static void BufferAudioStream(AudioStream stream, void *data, int numSamples);
static void CloseAudioStream(AudioStream stream);
#if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename
void TraceLog(int msgType, const char *text, ...); // Outputs a trace log message (INFO, ERROR, WARNING)
@ -595,33 +578,89 @@ void StopMusicStream(Music music)
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
ALenum state;
bool active = true;
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
int numBuffersToProcess = processed;
if (processed > 0)
{
active = BufferMusicStream(music, processed);
bool active = true;
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
for (int i = 0; i < numBuffersToProcess; i++)
{
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
else numSamples = music->samplesLeft;
// NOTE: Returns the number of samples to process (should be the same as numSamples -> it is)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, numSamples);
// TODO: Review stereo channels Ogg, not enough samples served!
UpdateAudioStream(music->stream, pcm, numSamples*music->stream.channels);
music->samplesLeft -= (numSamples*music->stream.channels);
} break;
case MUSIC_MODULE_XM:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer is 2*numsamples elements (left and right value for each sample)
jar_xm_generate_samples(music->ctxXm, pcmf, numSamples);
UpdateAudioStream(music->stream, pcmf, numSamples*2); // Using 32bit PCM data
music->samplesLeft -= numSamples;
//TraceLog(INFO, "Samples left: %i", music->samplesLeft);
} break;
case MUSIC_MODULE_MOD:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo)
jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0);
UpdateAudioStream(music->stream, pcm, numSamples*2);
music->samplesLeft -= numSamples;
} break;
default: break;
}
if (music->samplesLeft <= 0)
{
active = false;
break;
}
}
// Reset audio stream for looping
if (!active && music->loop)
{
// Restart music context (if required)
//if (music->ctxType == MUSIC_MODULE_XM)
if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_seek_start(&music->ctxMod);
else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_seek_start(music->ctxOgg);
// Reset samples left to total samples
music->samplesLeft = music->totalSamples;
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
active = BufferMusicStream(music, processed);
}
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
// This error is registered when UpdateAudioStream() fails
if (alGetError() == AL_INVALID_VALUE) TraceLog(WARNING, "OpenAL: Error buffering data...");
ALenum state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(music->stream.source);
@ -668,36 +707,14 @@ float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
if (music->ctxType == MUSIC_MODULE_XM)
{
uint64_t samplesPlayed;
jar_xm_get_position(music->ctxXm, NULL, NULL, NULL, &samplesPlayed);
// TODO: Not sure if this is the correct value
secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
}
else if (music->ctxType == MUSIC_MODULE_MOD)
{
long samplesPlayed = jar_mod_current_samples(&music->ctxMod);
secondsPlayed = (float)samplesPlayed/music->stream.sampleRate;
}
else if (music->ctxType == MUSIC_AUDIO_OGG)
{
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
}
return secondsPlayed;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Init audio stream (to stream audio pcm data)
static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
@ -735,7 +752,7 @@ static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleS
alSource3f(stream.source, AL_POSITION, 0, 0, 0);
alSource3f(stream.source, AL_VELOCITY, 0, 0, 0);
// Create Buffers
// Create Buffers (double buffering)
alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers);
// Initialize buffer with zeros by default
@ -766,7 +783,7 @@ static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleS
}
// Close audio stream and free memory
static void CloseAudioStream(AudioStream stream)
void CloseAudioStream(AudioStream stream)
{
// Stop playing channel
alSourceStop(stream.source);
@ -790,75 +807,66 @@ static void CloseAudioStream(AudioStream stream)
TraceLog(INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
// Push more audio data into audio stream, only one buffer per call
static void BufferAudioStream(AudioStream stream, void *data, int numSamples)
// Update audio stream buffers with data
// NOTE: Only one buffer per call
void UpdateAudioStream(AudioStream stream, void *data, int numSamples)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
//TraceLog(DEBUG, "Buffer to refill: %i", buffer);
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numSamples*sizeof(unsigned char), stream.sampleRate);
else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numSamples*sizeof(short), stream.sampleRate);
else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numSamples*sizeof(float), stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
}
// Fill music buffers with new data from music stream
static bool BufferMusicStream(Music music, int numBuffersToProcess)
// Check if any audio stream buffers requires refill
bool IsAudioBufferProcessed(AudioStream stream)
{
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
ALint processed = 0;
int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
bool active = true; // We can get more data from stream (not finished)
// Determine if music stream is ready to be written
alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
for (int i = 0; i < numBuffersToProcess; i++)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE;
else size = music->samplesLeft;
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (should be the same as size)
int numSamples = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, size);
BufferAudioStream(music->stream, pcm, numSamples*music->stream.channels);
music->samplesLeft -= (numSamples*music->stream.channels);
} break;
case MUSIC_MODULE_XM:
{
// NOTE: Output buffer is 2*numsamples elements (left and right value for each sample)
jar_xm_generate_samples(music->ctxXm, pcmf, size/2);
BufferAudioStream(music->stream, pcmf, size); // Using 32bit PCM data
music->samplesLeft -= (size/2);
} break;
case MUSIC_MODULE_MOD:
{
// NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo)
jar_mod_fillbuffer(&music->ctxMod, pcm, size/2, 0);
BufferAudioStream(music->stream, pcm, size);
music->samplesLeft -= (size/2);
} break;
default: break;
}
if (music->samplesLeft <= 0)
{
active = false;
break;
}
}
return active;
return (processed > 0);
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
alSourcePlay(stream.source);
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
alSourcePause(stream.source);
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
ALenum state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(stream.source);
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
alSourceStop(stream.source);
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{

27
src/audio.h
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@ -76,9 +76,21 @@ typedef struct Wave {
} Wave;
// Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
// NOTE: Anything longer than ~10 seconds should be streamed
typedef struct Music *Music;
// Audio stream type
// NOTE: Useful to create custom audio streams not bound to a specific file
typedef struct AudioStream {
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)
int format; // OpenAL audio format specifier
unsigned int source; // OpenAL audio source id
unsigned int buffers[2]; // OpenAL audio buffers (double buffering)
} AudioStream;
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
#endif
@ -93,7 +105,7 @@ extern "C" { // Prevents name mangling of functions
//----------------------------------------------------------------------------------
void InitAudioDevice(void); // Initialize audio device and context
void CloseAudioDevice(void); // Close the audio device and context (and music stream)
bool IsAudioDeviceReady(void); // Check if device has been initialized successfully
bool IsAudioDeviceReady(void); // Check if audio device has been initialized successfully
Sound LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
@ -120,6 +132,17 @@ void SetMusicPitch(Music music, float pitch); // Set pitch for
float GetMusicTimeLength(Music music); // Get music time length (in seconds)
float GetMusicTimePlayed(Music music); // Get current music time played (in seconds)
AudioStream InitAudioStream(unsigned int sampleRate,
unsigned int sampleSize,
unsigned int channels); // Init audio stream (to stream audio pcm data)
void UpdateAudioStream(AudioStream stream, void *data, int numSamples); // Update audio stream buffers with data
void CloseAudioStream(AudioStream stream); // Close audio stream and free memory
bool IsAudioBufferProcessed(AudioStream stream); // Check if any audio stream buffers requires refill
void PlayAudioStream(AudioStream stream); // Play audio stream
void PauseAudioStream(AudioStream stream); // Pause audio stream
void ResumeAudioStream(AudioStream stream); // Resume audio stream
void StopAudioStream(AudioStream stream); // Stop audio stream
#ifdef __cplusplus
}
#endif

29
src/raylib.h
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@ -499,8 +499,8 @@ typedef struct Ray {
// Sound source type
typedef struct Sound {
unsigned int source; // Sound audio source id
unsigned int buffer; // Sound audio buffer id
unsigned int source; // OpenAL audio source id
unsigned int buffer; // OpenAL audio buffer id
} Sound;
// Wave type, defines audio wave data
@ -516,6 +516,18 @@ typedef struct Wave {
// NOTE: Anything longer than ~10 seconds should be streamed
typedef struct Music *Music;
// Audio stream type
// NOTE: Useful to create custom audio streams not bound to a specific file
typedef struct AudioStream {
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)
int format; // OpenAL audio format specifier
unsigned int source; // OpenAL audio source id
unsigned int buffers[2]; // OpenAL audio buffers (double buffering)
} AudioStream;
// Texture formats
// NOTE: Support depends on OpenGL version and platform
typedef enum {
@ -923,7 +935,7 @@ void ToggleVrMode(void); // Enable/Disable VR experience (dev
//------------------------------------------------------------------------------------
void InitAudioDevice(void); // Initialize audio device and context
void CloseAudioDevice(void); // Close the audio device and context (and music stream)
bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
bool IsAudioDeviceReady(void); // Check if audio device has been initialized successfully
Sound LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
@ -950,6 +962,17 @@ void SetMusicPitch(Music music, float pitch); // Set pitch for
float GetMusicTimeLength(Music music); // Get music time length (in seconds)
float GetMusicTimePlayed(Music music); // Get current music time played (in seconds)
AudioStream InitAudioStream(unsigned int sampleRate,
unsigned int sampleSize,
unsigned int channels); // Init audio stream (to stream audio pcm data)
void UpdateAudioStream(AudioStream stream, void *data, int numSamples); // Update audio stream buffers with data
void CloseAudioStream(AudioStream stream); // Close audio stream and free memory
bool IsAudioBufferProcessed(AudioStream stream); // Check if any audio stream buffers requires refill
void PlayAudioStream(AudioStream stream); // Play audio stream
void PauseAudioStream(AudioStream stream); // Pause audio stream
void ResumeAudioStream(AudioStream stream); // Resume audio stream
void StopAudioStream(AudioStream stream); // Stop audio stream
#ifdef __cplusplus
}
#endif