|
@@ -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)
|
|
|
{
|