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-rw-r--r--src/audio.c288
1 files changed, 144 insertions, 144 deletions
diff --git a/src/audio.c b/src/audio.c
index 2866de49..e1d9b4a1 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -3,21 +3,21 @@
* raylib.audio
*
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
-*
-* Uses external lib:
+*
+* Uses external lib:
* OpenAL - Audio device management lib
* stb_vorbis - Ogg audio files loading
-*
+*
* Copyright (c) 2013 Ramon Santamaria (Ray San - raysan@raysanweb.com)
-*
-* This software is provided "as-is", without any express or implied warranty. In no event
+*
+* This software is provided "as-is", without any express or implied warranty. In no event
* will the authors be held liable for any damages arising from the use of this software.
*
-* Permission is granted to anyone to use this software for any purpose, including commercial
+* Permission is granted to anyone to use this software for any purpose, including commercial
* applications, and to alter it and redistribute it freely, subject to the following restrictions:
*
-* 1. The origin of this software must not be misrepresented; you must not claim that you
-* wrote the original software. If you use this software in a product, an acknowledgment
+* 1. The origin of this software must not be misrepresented; you must not claim that you
+* wrote the original software. If you use this software in a product, an acknowledgment
* in the product documentation would be appreciated but is not required.
*
* 2. Altered source versions must be plainly marked as such, and must not be misrepresented
@@ -54,16 +54,16 @@
// NOTE: Anything longer than ~10 seconds should be streamed...
typedef struct Music {
stb_vorbis *stream;
-
- ALuint buffers[MUSIC_STREAM_BUFFERS];
- ALuint source;
- ALenum format;
-
+
+ ALuint buffers[MUSIC_STREAM_BUFFERS];
+ ALuint source;
+ ALenum format;
+
int channels;
int sampleRate;
- int totalSamplesLeft;
- bool loop;
-
+ int totalSamplesLeft;
+ bool loop;
+
} Music;
// Wave file data
@@ -72,7 +72,7 @@ typedef struct Wave {
unsigned int dataSize; // Data size in bytes
unsigned int sampleRate;
short bitsPerSample;
- short channels;
+ short channels;
} Wave;
//----------------------------------------------------------------------------------
@@ -102,22 +102,22 @@ void InitAudioDevice()
{
// Open and initialize a device with default settings
ALCdevice *device = alcOpenDevice(NULL);
-
+
if(!device) TraceLog(ERROR, "Could not open audio device");
ALCcontext *context = alcCreateContext(device, NULL);
-
+
if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE)
{
if(context != NULL) alcDestroyContext(context);
-
+
alcCloseDevice(device);
-
+
TraceLog(ERROR, "Could not setup audio context");
}
TraceLog(INFO, "Audio device and context initialized successfully: %s\n", alcGetString(device, ALC_DEVICE_SPECIFIER));
-
+
// Listener definition (just for 2D)
alListener3f(AL_POSITION, 0, 0, 0);
alListener3f(AL_VELOCITY, 0, 0, 0);
@@ -131,7 +131,7 @@ void CloseAudioDevice()
ALCdevice *device;
ALCcontext *context = alcGetCurrentContext();
-
+
if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
device = alcGetContextsDevice(context);
@@ -150,41 +150,41 @@ Sound LoadSound(char *fileName)
{
Sound sound;
Wave wave;
-
+
// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
-
+
// Audio file loading
// NOTE: Buffer space is allocated inside function, Wave must be freed
-
+
if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
else TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
-
+
if (wave.data != NULL)
{
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the bits per sample
- if (wave.channels == 1)
+ if (wave.channels == 1)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
- }
- else if (wave.channels == 2)
+ }
+ else if (wave.channels == 2)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
}
-
+
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
- alSourcef(source, AL_PITCH, 1);
+ alSourcef(source, AL_PITCH, 1);
alSourcef(source, AL_GAIN, 1);
alSource3f(source, AL_POSITION, 0, 0, 0);
alSource3f(source, AL_VELOCITY, 0, 0, 0);
alSourcei(source, AL_LOOPING, AL_FALSE);
-
+
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
@@ -195,17 +195,17 @@ Sound LoadSound(char *fileName)
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
-
+
// Unallocate WAV data
UnloadWave(wave);
-
- TraceLog(INFO, "[%s] Sound file loaded successfully", fileName);
+
+ TraceLog(INFO, "[%s] Sound file loaded successfully", fileName);
TraceLog(INFO, "[%s] Sample rate: %i - Channels: %i", fileName, wave.sampleRate, wave.channels);
-
+
sound.source = source;
sound.buffer = buffer;
}
-
+
return sound;
}
@@ -220,9 +220,9 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
unsigned char version; // rRES file version and subversion
char useless; // rRES header reserved data
short numRes;
-
+
ResInfoHeader infoHeader;
-
+
FILE *rresFile = fopen(rresName, "rb");
if (!rresFile) TraceLog(WARNING, "[%s] Could not open raylib resource file", rresName);
@@ -235,7 +235,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
fread(&id[3], sizeof(char), 1, rresFile);
fread(&version, sizeof(char), 1, rresFile);
fread(&useless, sizeof(char), 1, rresFile);
-
+
if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
{
TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
@@ -244,11 +244,11 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
{
// Read number of resources embedded
fread(&numRes, sizeof(short), 1, rresFile);
-
+
for (int i = 0; i < numRes; i++)
{
fread(&infoHeader, sizeof(ResInfoHeader), 1, rresFile);
-
+
if (infoHeader.id == resId)
{
found = true;
@@ -258,56 +258,56 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
{
// TODO: Check data compression type
// NOTE: We suppose compression type 2 (DEFLATE - default)
-
+
// Reading SOUND parameters
Wave wave;
short sampleRate, bps;
char channels, reserved;
-
+
fread(&sampleRate, sizeof(short), 1, rresFile); // Sample rate (frequency)
fread(&bps, sizeof(short), 1, rresFile); // Bits per sample
fread(&channels, 1, 1, rresFile); // Channels (1 - mono, 2 - stereo)
fread(&reserved, 1, 1, rresFile); // <reserved>
-
+
wave.sampleRate = sampleRate;
wave.dataSize = infoHeader.srcSize;
wave.bitsPerSample = bps;
wave.channels = (short)channels;
-
+
unsigned char *data = malloc(infoHeader.size);
fread(data, infoHeader.size, 1, rresFile);
-
+
wave.data = DecompressData(data, infoHeader.size, infoHeader.srcSize);
-
+
free(data);
-
+
// Convert wave to Sound (OpenAL)
ALenum format = 0;
-
+
// The OpenAL format is worked out by looking at the number of channels and the bits per sample
- if (wave.channels == 1)
+ if (wave.channels == 1)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
- }
- else if (wave.channels == 2)
+ }
+ else if (wave.channels == 2)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
}
-
-
+
+
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
- alSourcef(source, AL_PITCH, 1);
+ alSourcef(source, AL_PITCH, 1);
alSourcef(source, AL_GAIN, 1);
alSource3f(source, AL_POSITION, 0, 0, 0);
alSource3f(source, AL_VELOCITY, 0, 0, 0);
alSourcei(source, AL_LOOPING, AL_FALSE);
-
+
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
@@ -318,12 +318,12 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
-
+
// Unallocate WAV data
UnloadWave(wave);
TraceLog(INFO, "[%s] Sound loaded successfully from resource, sample rate: %i", rresName, (int)sampleRate);
-
+
sound.source = source;
sound.buffer = buffer;
}
@@ -344,18 +344,18 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
case 4: break; // RAW: No parameters
default: break;
}
-
+
// Jump DATA to read next infoHeader
fseek(rresFile, infoHeader.size, SEEK_CUR);
- }
+ }
}
}
-
+
fclose(rresFile);
}
-
+
if (!found) TraceLog(WARNING, "[%s] Required resource id [%i] could not be found in the raylib resource file", rresName, resId);
-
+
return sound;
}
@@ -370,7 +370,7 @@ void UnloadSound(Sound sound)
void PlaySound(Sound sound)
{
alSourcePlay(sound.source); // Play the sound
-
+
//TraceLog(INFO, "Playing sound");
// Find the current position of the sound being played
@@ -380,7 +380,7 @@ void PlaySound(Sound sound)
//
//int sampleRate;
//alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); // AL_CHANNELS, AL_BITS (bps)
-
+
//float seconds = (float)byteOffset / sampleRate; // Number of seconds since the beginning of the sound
//or
//float result;
@@ -404,10 +404,10 @@ bool SoundIsPlaying(Sound sound)
{
bool playing = false;
ALint state;
-
+
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
-
+
return playing;
}
@@ -434,49 +434,49 @@ void PlayMusicStream(char *fileName)
{
// Stop current music, clean buffers, unload current stream
StopMusicStream();
-
+
// Open audio stream
currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
-
+
if (currentMusic.stream == NULL) TraceLog(WARNING, "[%s] Could not open ogg audio file", fileName);
else
{
// Get file info
stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
-
+
currentMusic.channels = info.channels;
currentMusic.sampleRate = info.sample_rate;
-
+
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(INFO, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
-
+
if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
else currentMusic.format = AL_FORMAT_MONO16;
-
+
currentMusic.loop = true; // We loop by default
musicEnabled = true;
-
+
// Create an audio source
alGenSources(1, &currentMusic.source); // Generate pointer to audio source
- alSourcef(currentMusic.source, AL_PITCH, 1);
+ alSourcef(currentMusic.source, AL_PITCH, 1);
alSourcef(currentMusic.source, AL_GAIN, 1);
alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
//alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue!
-
+
// Generate two OpenAL buffers
alGenBuffers(2, currentMusic.buffers);
// Fill buffers with music...
BufferMusicStream(currentMusic.buffers[0]);
BufferMusicStream(currentMusic.buffers[1]);
-
+
// Queue buffers and start playing
alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
alSourcePlay(currentMusic.source);
-
+
// NOTE: Regularly, we must check if a buffer has been processed and refill it: MusicStreamUpdate()
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
@@ -491,15 +491,15 @@ void StopMusicStream()
if (musicEnabled)
{
alSourceStop(currentMusic.source);
-
+
EmptyMusicStream(); // Empty music buffers
-
+
alDeleteSources(1, &currentMusic.source);
alDeleteBuffers(2, currentMusic.buffers);
-
+
stb_vorbis_close(currentMusic.stream);
}
-
+
musicEnabled = false;
}
@@ -514,9 +514,9 @@ void PauseMusicStream()
bool MusicIsPlaying()
{
ALenum state;
-
+
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
-
+
return (state == AL_PLAYING);
}
@@ -530,7 +530,7 @@ void SetMusicVolume(float volume)
float GetMusicTimeLength()
{
float totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
-
+
return totalSeconds;
}
@@ -538,11 +538,11 @@ float GetMusicTimeLength()
float GetMusicTimePlayed()
{
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
-
+
int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
-
+
float secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels);
-
+
return secondsPlayed;
}
@@ -553,30 +553,30 @@ float GetMusicTimePlayed()
// Fill music buffers with new data from music stream
static bool BufferMusicStream(ALuint buffer)
{
- short pcm[MUSIC_BUFFER_SIZE];
-
- int size = 0; // Total size of data steamed (in bytes)
- int streamedBytes = 0; // Bytes of data obtained in one samples get
-
+ short pcm[MUSIC_BUFFER_SIZE];
+
+ int size = 0; // Total size of data steamed (in bytes)
+ int streamedBytes = 0; // Bytes of data obtained in one samples get
+
bool active = true; // We can get more data from stream (not finished)
-
+
if (musicEnabled)
{
while (size < MUSIC_BUFFER_SIZE)
{
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
-
+
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
else break;
}
-
+
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
}
-
- if (size > 0)
+
+ if (size > 0)
{
alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
-
+
currentMusic.totalSamplesLeft -= size;
}
else
@@ -585,21 +585,21 @@ static bool BufferMusicStream(ALuint buffer)
TraceLog(WARNING, "No more data obtained from stream");
}
- return active;
+ return active;
}
// Empty music buffers
static void EmptyMusicStream()
{
- ALuint buffer = 0;
+ ALuint buffer = 0;
int queued = 0;
-
+
alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
-
+
while(queued > 0)
{
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
-
+
queued--;
}
}
@@ -610,12 +610,12 @@ extern void UpdateMusicStream()
ALuint buffer = 0;
ALint processed = 0;
bool active = true;
-
+
if (musicEnabled)
{
// Get the number of already processed buffers (if any)
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
-
+
while (processed > 0)
{
// Recover processed buffer for refill
@@ -623,32 +623,32 @@ extern void UpdateMusicStream()
// Refill buffer
active = BufferMusicStream(buffer);
-
+
// If no more data to stream, restart music (if loop)
- if ((!active) && (currentMusic.loop))
+ if ((!active) && (currentMusic.loop))
{
if (currentMusic.loop)
{
stb_vorbis_seek_start(currentMusic.stream);
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
-
+
active = BufferMusicStream(buffer);
}
}
-
+
// Add refilled buffer to queue again... don't let the music stop!
alSourceQueueBuffers(currentMusic.source, 1, &buffer);
-
+
if(alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Ogg playing, error buffering data...");
-
+
processed--;
}
-
+
ALenum state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
-
+
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
-
+
if (!active) StopMusicStream();
}
}
@@ -678,16 +678,16 @@ static Wave LoadWAV(const char *fileName)
char subChunkID[4];
long subChunkSize;
} WaveData;
-
+
RiffHeader riffHeader;
WaveFormat waveFormat;
WaveData waveData;
-
+
Wave wave;
FILE *wavFile;
-
+
wavFile = fopen(fileName, "rb");
-
+
if (!wavFile)
{
TraceLog(WARNING, "[%s] Could not open WAV file", fileName);
@@ -696,7 +696,7 @@ static Wave LoadWAV(const char *fileName)
{
// Read in the first chunk into the struct
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
-
+
// Check for RIFF and WAVE tags
if (((riffHeader.chunkID[0] != 'R') || (riffHeader.chunkID[1] != 'I') || (riffHeader.chunkID[2] != 'F') || (riffHeader.chunkID[3] != 'F')) ||
((riffHeader.format[0] != 'W') || (riffHeader.format[1] != 'A') || (riffHeader.format[2] != 'V') || (riffHeader.format[3] != 'E')))
@@ -707,7 +707,7 @@ static Wave LoadWAV(const char *fileName)
{
// Read in the 2nd chunk for the wave info
fread(&waveFormat, sizeof(WaveFormat), 1, wavFile);
-
+
// Check for fmt tag
if ((waveFormat.subChunkID[0] != 'f') || (waveFormat.subChunkID[1] != 'm') ||
(waveFormat.subChunkID[2] != 't') || (waveFormat.subChunkID[3] != ' '))
@@ -718,10 +718,10 @@ static Wave LoadWAV(const char *fileName)
{
// Check for extra parameters;
if (waveFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
-
+
// Read in the the last byte of data before the sound file
fread(&waveData, sizeof(WaveData), 1, wavFile);
-
+
// Check for data tag
if ((waveData.subChunkID[0] != 'd') || (waveData.subChunkID[1] != 'a') ||
(waveData.subChunkID[2] != 't') || (waveData.subChunkID[3] != 'a'))
@@ -731,17 +731,17 @@ static Wave LoadWAV(const char *fileName)
else
{
// Allocate memory for data
- wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
-
+ wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
+
// Read in the sound data into the soundData variable
fread(wave.data, waveData.subChunkSize, 1, wavFile);
-
+
// Now we set the variables that we need later
wave.dataSize = waveData.subChunkSize;
wave.sampleRate = waveFormat.sampleRate;
wave.channels = waveFormat.numChannels;
wave.bitsPerSample = waveFormat.bitsPerSample;
-
+
TraceLog(INFO, "[%s] Wave file loaded successfully", fileName);
}
}
@@ -749,7 +749,7 @@ static Wave LoadWAV(const char *fileName)
fclose(wavFile);
}
-
+
return wave;
}
@@ -757,42 +757,42 @@ static Wave LoadWAV(const char *fileName)
static Wave LoadOGG(char *fileName)
{
Wave wave;
-
+
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
-
+
wave.sampleRate = info.sample_rate;
wave.bitsPerSample = 16;
wave.channels = info.channels;
-
+
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
int totalSamplesLength = (stb_vorbis_stream_length_in_samples(oggFile) * info.channels);
-
+
wave.dataSize = totalSamplesLength*sizeof(short); // Size must be in bytes
-
+
TraceLog(DEBUG, "[%s] Samples length: %i", fileName, totalSamplesLength);
-
+
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
-
+
TraceLog(DEBUG, "[%s] Total seconds: %f", fileName, totalSeconds);
-
+
if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
-
+
int totalSamples = totalSeconds*info.sample_rate*info.channels;
-
+
TraceLog(DEBUG, "[%s] Total samples calculated: %i", fileName, totalSamples);
-
- //short *data
+
+ //short *data
wave.data = malloc(sizeof(short)*totalSamplesLength);
int samplesObtained = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, wave.data, totalSamplesLength);
-
+
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained);
stb_vorbis_close(oggFile);
-
+
return wave;
}
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