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-rw-r--r--src/audio.c230
-rw-r--r--src/audio.h27
-rw-r--r--src/raylib.h29
3 files changed, 170 insertions, 116 deletions
diff --git a/src/audio.c b/src/audio.c
index d1c425d5..befed61c 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -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);
- }
+ 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);
+ }
+}
+
+// Check if any audio stream buffers requires refill
+bool IsAudioBufferProcessed(AudioStream stream)
+{
+ ALint processed = 0;
- 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);
+ // Determine if music stream is ready to be written
+ alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
- alSourceQueueBuffers(stream.source, 1, &buffer);
+ return (processed > 0);
}
-// Fill music buffers with new data from music stream
-static bool BufferMusicStream(Music music, int numBuffersToProcess)
+// Play audio stream
+void PlayAudioStream(AudioStream stream)
{
- short pcm[AUDIO_BUFFER_SIZE];
- float pcmf[AUDIO_BUFFER_SIZE];
+ alSourcePlay(stream.source);
+}
- 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)
-
- for (int i = 0; i < numBuffersToProcess; i++)
- {
- if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE;
- else size = music->samplesLeft;
+// Play audio stream
+void PauseAudioStream(AudioStream stream)
+{
+ alSourcePause(stream.source);
+}
- 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;
- }
+// Resume audio stream playing
+void ResumeAudioStream(AudioStream stream)
+{
+ ALenum state;
+ alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
- if (music->samplesLeft <= 0)
- {
- active = false;
- break;
- }
- }
-
- return active;
+ 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)
{
diff --git a/src/audio.h b/src/audio.h
index c9171339..dbd88939 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -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
diff --git a/src/raylib.h b/src/raylib.h
index 4b9f6ca0..3ee7a793 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -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
generated by cgit v1.2.3 (git 2.25.1) at 2026-01-07 05:27:10 +0000