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bug fixes

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This commit is contained in:
Joshua Reisenauer 2016-05-12 21:14:02 -07:00
parent ded07690fd
commit 5107a2dc40
1 changed files with 35 additions and 34 deletions
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69
src/audio.c
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@ -77,19 +77,6 @@
// Types and Structures Definition
//----------------------------------------------------------------------------------
// Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed...
typedef struct Music {
stb_vorbis *stream;
jar_xm_context_t *chipctx; // Stores jar_xm context
AudioContext_t *ctx; // audio context
int totalSamplesLeft;
float totalLengthSeconds;
bool loop;
bool chipTune; // True if chiptune is loaded
} Music;
// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
// a dedicated mix channel. All audio is 32bit floating point in stereo.
@ -104,6 +91,19 @@ typedef struct AudioContext_t {
ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
} AudioContext_t;
// Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed...
typedef struct Music {
stb_vorbis *stream;
jar_xm_context_t *chipctx; // Stores jar_xm context
AudioContext_t *ctx; // audio context
int totalSamplesLeft;
float totalLengthSeconds;
bool loop;
bool chipTune; // True if chiptune is loaded
} Music;
#if defined(AUDIO_STANDALONE)
typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
#endif
@ -168,7 +168,10 @@ void InitAudioDevice(void)
// Close the audio device for the current context, and destroys the context
void CloseAudioDevice(void)
{
StopMusicStream(); // Stop music streaming and close current stream
for(int index=0; index<MAX_MUSIC_STREAMS; index++)
{
if(currentMusic[index].ctx) StopMusicStream(index); // Stop music streaming and close current stream
}
ALCdevice *device;
ALCcontext *context = alcGetCurrentContext();
@ -205,7 +208,6 @@ AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChanne
{
if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
if(!IsAudioDeviceReady()) InitAudioDevice();
else StopMusicStream();
if(!mixChannelsActive_g[mixChannel]){
AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t));
@ -776,7 +778,6 @@ int PlayMusicStream(int musicIndex, char *fileName)
else if(musicIndex = MAX_AUDIO_CONTEXTS - 1) return 2; // error
}
if (strcmp(GetExtension(fileName),"ogg") == 0)
{
// Open audio stream
@ -888,12 +889,12 @@ void ResumeMusicStream(int index)
{
// Resume music playing... if music available!
ALenum state;
if(currentMusic[musicIndex].ctx){
alGetSourcei(currentMusic[musicIndex].ctx->alSource, AL_SOURCE_STATE, &state);
if(currentMusic[index].ctx){
alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED)
{
TraceLog(INFO, "Resuming music stream");
alSourcePlay(currentMusic[musicIndex].ctx->alSource);
alSourcePlay(currentMusic[index].ctx->alSource);
musicEnabled = true;
}
}
@ -976,9 +977,8 @@ static bool BufferMusicStream(int index)
float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
int size = 0; // Total size of data steamed (in bytes)
int streamedBytes = 0; // samples of data obtained, channels are not included in calculation
bool active = true; // We can get more data from stream (not finished)
if (musicEnabled)
{
if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
@ -992,12 +992,12 @@ static bool BufferMusicStream(int index)
}
else
{
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
UpdateAudioContext(currentMusic[index].ctx, pcm, MUSIC_BUFFER_SIZE_SHORT);
currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_SHORT;
int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes);
currentMusic[index].totalSamplesLeft -= streamedBytes*currentMusic[index].ctx->channels;
if(currentMusic[index].totalSamplesLeft <= 0) active = false;
}
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
// TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
}
return active;
@ -1009,11 +1009,11 @@ static void EmptyMusicStream(int index)
ALuint buffer = 0;
int queued = 0;
alGetSourcei(currentMusic[index].source, AL_BUFFERS_QUEUED, &queued);
alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0)
{
alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
alSourceUnqueueBuffers(currentMusic[index].ctx->alSource, 1, &buffer);
queued--;
}
@ -1045,16 +1045,17 @@ void UpdateMusicStream(int index)
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state);
processed--;
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].ctx->alSource);
if (!active) StopMusicStream(index);
}
else
return;
alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
if (!active) StopMusicStream();
}
// Load WAV file into Wave structure