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Review some functions, formatting and comments

This commit is contained in:
raysan5 2016-07-15 18:16:34 +02:00
parent 338bb3fd9c
commit 7959ccd84d
3 changed files with 148 additions and 125 deletions
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240
src/audio.c
View File

@ -2,13 +2,26 @@
*
* raylib.audio
*
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
* Basic functions to manage Audio:
* Manage audio device (init/close)
* Load and Unload audio files
* Play/Stop/Pause/Resume loaded audio
* Manage mixing channels
* Manage raw audio context
*
* Uses external lib:
* OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html)
* stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* jar_xm - XM module file loading
* jar_mod - MOD audio file loading
*
* Copyright (c) 2014 Ramon Santamaria (@raysan5)
* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions:
* XM audio module support (jar_xm)
* MOD audio module support (jar_mod)
* Mixing channels support
* Raw audio context support
*
* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5)
*
* 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.
@ -68,9 +81,9 @@
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each alSource
#define MAX_MIX_CHANNELS 4 // Number of OpenAL sources
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each source
#define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources
#define MAX_MIX_CHANNELS 4 // Number of mix channels (OpenAL sources)
#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
// NOTE: On RPI and Android should be lower to avoid frame-stalls
@ -143,7 +156,7 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
// Global Variables Definition
//----------------------------------------------------------------------------------
static Music musicStreams[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time
static MixChannel *mixChannels[MAX_MIX_CHANNELS]; // What mix channels are currently active
static MixChannel *mixChannels[MAX_MIX_CHANNELS]; // Mix channels currently active (from music streams)
static int lastAudioError = 0; // Registers last audio error
@ -157,13 +170,11 @@ static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers
static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
static void CloseMixChannel(MixChannel *mixc); // Frees mix channel
static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
static int FillAlBufferWithSilence(MixChannel *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled
static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements); // Pushes more audio data into mix channel
//static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
//static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
#if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename
@ -204,7 +215,7 @@ void InitAudioDevice(void)
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
for (int index=0; index<MAX_MUSIC_STREAMS; index++)
for (int index = 0; index < MAX_MUSIC_STREAMS; index++)
{
if (musicStreams[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
}
@ -221,7 +232,7 @@ void CloseAudioDevice(void)
alcCloseDevice(device);
}
// True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
ALCcontext *context = alcGetCurrentContext();
@ -240,9 +251,9 @@ bool IsAudioDeviceReady(void)
// Module Functions Definition - Custom audio output
//----------------------------------------------------------------------------------
// For streaming into mix channels.
// The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time.
// exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
// Init mix channel for streaming
// The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available.
// Each mix channel can only be used one at a time.
static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
{
if (mixChannel >= MAX_MIX_CHANNELS) return NULL;
@ -280,7 +291,20 @@ static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixCh
alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
// Fill buffers
for (int i = 0; i < MAX_STREAM_BUFFERS; i++) FillAlBufferWithSilence(mixc, mixc->alBuffer[i]);
for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
{
// Initialize buffer with zeros by default
if (mixc->floatingPoint)
{
float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f };
alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
}
else
{
short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 };
alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
}
}
alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
mixc->playing = true;
@ -320,9 +344,9 @@ static void CloseMixChannel(MixChannel *mixc)
}
}
// Pushes more audio data into mixc mix channel, only one buffer per call
// Pushes more audio data into mix channel, only one buffer per call
// Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
// @Returns number of samples that where processed.
// Returns number of samples that where processed.
static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements)
{
if (!mixc || (mixChannels[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples
@ -368,28 +392,11 @@ static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements)
return numberElements;
}
// fill buffer with zeros, returns number processed
static int FillAlBufferWithSilence(MixChannel *mixc, ALuint buffer)
{
if (mixc->floatingPoint)
{
float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f };
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_FLOAT;
}
else
{
short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 };
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_SHORT;
}
}
/*
// Convert data from short to float
// example usage:
// short sh[3] = {1,2,3};float fl[3];
// ResampleShortToFloat(sh,fl,3);
// short sh[3] = {1,2,3};float fl[3];
// ResampleShortToFloat(sh,fl,3);
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len)
{
for (int i = 0; i < len; i++)
@ -399,9 +406,10 @@ static void ResampleShortToFloat(short *shorts, float *floats, unsigned short le
}
}
// Convert data from float to short
// example usage:
// char ch[3] = {1,2,3};float fl[3];
// ResampleByteToFloat(ch,fl,3);
// char ch[3] = {1,2,3};float fl[3];
// ResampleByteToFloat(ch,fl,3);
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
{
for (int i = 0; i < len; i++)
@ -410,43 +418,55 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
else floats[i] = (float)chars[i]/128.0f;
}
}
*/
// used to output raw audio streams, returns negative numbers on error, + number represents the mix channel index
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
// Initialize raw audio mix channel for audio buffering
// NOTE: Returns mix channel index or -1 if it fails (errors are registered on lastAudioError)
int InitRawMixChannel(int sampleRate, int channels, bool floatingPoint)
{
int mixIndex;
for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{
if (mixChannels[mixIndex] == NULL) break;
else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error
else if (mixIndex == (MAX_MIX_CHANNELS - 1))
{
lastAudioError = ERROR_OUT_OF_MIX_CHANNELS;
return -1;
}
}
if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex;
else return ERROR_RAW_CONTEXT_CREATION; // error
else
{
lastAudioError = ERROR_RAW_CONTEXT_CREATION;
return -1;
}
}
void CloseRawAudioContext(RawAudioContext ctx)
{
if (mixChannels[ctx]) CloseMixChannel(mixChannels[ctx]);
}
// if 0 is returned, the buffers are still full and you need to keep trying with the same data until a + number is returned.
// any + number returned is the number of samples that was processed and passed into buffer.
// data either needs to be array of floats or shorts.
int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements)
// Buffers data directly to raw mix channel
// if 0 is returned, buffers are still full and you need to keep trying with the same data
// otherwise it will return number of samples buffered.
// NOTE: Data could be either be an array of floats or shorts, depending on the created context
int BufferRawAudioContext(int ctx, void *data, unsigned short numberElements)
{
int numBuffered = 0;
if (ctx >= 0)
{
MixChannel* mixc = mixChannels[ctx];
MixChannel *mixc = mixChannels[ctx];
numBuffered = BufferMixChannel(mixc, data, numberElements);
}
return numBuffered;
}
// Closes and frees raw mix channel
void CloseRawAudioContext(int ctx)
{
if (mixChannels[ctx]) CloseMixChannel(mixChannels[ctx]);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
@ -804,7 +824,7 @@ void SetSoundPitch(Sound sound, float pitch)
//----------------------------------------------------------------------------------
// Start music playing (open stream)
// returns 0 on success
// returns 0 on success or error code
int PlayMusicStream(int index, char *fileName)
{
int mixIndex;
@ -866,7 +886,7 @@ int PlayMusicStream(int index, char *fileName)
musicStreams[index].loop = true;
jar_xm_set_max_loop_count(musicStreams[index].xmctx, 0); // infinite number of loops
musicStreams[index].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicStreams[index].xmctx);
musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft) / 48000.f;
musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft)/48000.0f;
musicStreams[index].enabled = true;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicStreams[index].totalSamplesLeft);
@ -893,7 +913,7 @@ int PlayMusicStream(int index, char *fileName)
musicStreams[index].chipTune = true;
musicStreams[index].loop = true;
musicStreams[index].totalSamplesLeft = (unsigned int)jar_mod_max_samples(&musicStreams[index].modctx);
musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft) / 48000.f;
musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft)/48000.0f;
musicStreams[index].enabled = true;
TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicStreams[index].totalSamplesLeft);
@ -944,6 +964,51 @@ void StopMusicStream(int index)
}
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(int index)
{
ALenum state;
bool active = true;
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
if (musicStreams[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicStreams[index].enabled && musicStreams[index].mixc && (processed > 0))
{
active = BufferMusicStream(index, processed);
if (!active && musicStreams[index].loop)
{
if (musicStreams[index].chipTune)
{
if(musicStreams[index].modctx.mod_loaded) jar_mod_seek_start(&musicStreams[index].modctx);
musicStreams[index].totalSamplesLeft = musicStreams[index].totalLengthSeconds*48000.0f;
}
else
{
stb_vorbis_seek_start(musicStreams[index].stream);
musicStreams[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicStreams[index].stream)*musicStreams[index].mixc->channels;
}
// Determine if music stream is ready to be written
alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
active = BufferMusicStream(index, processed);
}
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(musicStreams[index].mixc->alSource);
if (!active) StopMusicStream(index);
}
}
//get number of music channels active at this time, this does not mean they are playing
int GetMusicStreamCount(void)
{
@ -1045,18 +1110,18 @@ float GetMusicTimePlayed(int index)
{
uint64_t samples;
jar_xm_get_position(musicStreams[index].xmctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (48000.f * musicStreams[index].mixc->channels); // Not sure if this is the correct value
secondsPlayed = (float)samples/(48000.0f*musicStreams[index].mixc->channels); // Not sure if this is the correct value
}
else if(musicStreams[index].chipTune && musicStreams[index].modctx.mod_loaded)
{
long numsamp = jar_mod_current_samples(&musicStreams[index].modctx);
secondsPlayed = (float)numsamp / (48000.f);
secondsPlayed = (float)numsamp/(48000.0f);
}
else
{
int totalSamples = stb_vorbis_stream_length_in_samples(musicStreams[index].stream) * musicStreams[index].mixc->channels;
int totalSamples = stb_vorbis_stream_length_in_samples(musicStreams[index].stream)*musicStreams[index].mixc->channels;
int samplesPlayed = totalSamples - musicStreams[index].totalSamplesLeft;
secondsPlayed = (float)samplesPlayed / (musicStreams[index].mixc->sampleRate * musicStreams[index].mixc->channels);
secondsPlayed = (float)samplesPlayed/(musicStreams[index].mixc->sampleRate*musicStreams[index].mixc->channels);
}
}
@ -1144,53 +1209,6 @@ static void EmptyMusicStream(int index)
}
}
// Determine if a music stream is ready to be written
static int IsMusicStreamReadyForBuffering(int index)
{
ALint processed = 0;
alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
return processed;
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(int index)
{
ALenum state;
bool active = true;
int numBuffers = IsMusicStreamReadyForBuffering(index);
if (musicStreams[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicStreams[index].enabled && musicStreams[index].mixc && numBuffers)
{
active = BufferMusicStream(index, numBuffers);
if (!active && musicStreams[index].loop)
{
if (musicStreams[index].chipTune)
{
if(musicStreams[index].modctx.mod_loaded) jar_mod_seek_start(&musicStreams[index].modctx);
musicStreams[index].totalSamplesLeft = musicStreams[index].totalLengthSeconds * 48000.f;
}
else
{
stb_vorbis_seek_start(musicStreams[index].stream);
musicStreams[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicStreams[index].stream) * musicStreams[index].mixc->channels;
}
active = BufferMusicStream(index, IsMusicStreamReadyForBuffering(index));
}
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(musicStreams[index].mixc->alSource);
if (!active) StopMusicStream(index);
}
}
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{

31
src/audio.h
View File

@ -2,13 +2,26 @@
*
* raylib.audio
*
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
* Basic functions to manage Audio:
* Manage audio device (init/close)
* Load and Unload audio files
* Play/Stop/Pause/Resume loaded audio
* Manage mixing channels
* Manage raw audio context
*
* Uses external lib:
* OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html)
* stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* jar_xm - XM module file loading
* jar_mod - MOD audio file loading
*
* Copyright (c) 2015 Ramon Santamaria (@raysan5)
* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions:
* XM audio module support (jar_xm)
* MOD audio module support (jar_mod)
* Mixing channels support
* Raw audio context support
*
* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5)
*
* 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.
@ -63,9 +76,6 @@ typedef struct Wave {
short channels;
} Wave;
typedef int RawAudioContext;
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
#endif
@ -80,7 +90,7 @@ 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); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
bool IsAudioDeviceReady(void); // Check if device has been initialized successfully
Sound LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
@ -105,12 +115,9 @@ float GetMusicTimeLength(int index); // Get music tim
float GetMusicTimePlayed(int index); // Get current music time played (in seconds)
int GetMusicStreamCount(void); // Get number of streams loaded
// used to output raw audio streams, returns negative numbers on error
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx);
int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements); // returns number of elements buffered
int InitRawMixChannel(int sampleRate, int channels, bool floatingPoint); // Initialize raw audio mix channel for audio buffering
int BufferRawMixChannel(int mixc, void *data, unsigned short numberElements); // Buffers data directly to raw mix channel
void CloseRawMixChannel(int mixc); // Closes and frees raw mix channel
#ifdef __cplusplus
}

2
src/raylib.h
View File

@ -468,8 +468,6 @@ typedef struct Wave {
short channels;
} Wave;
typedef int RawAudioContext;
// Texture formats
// NOTE: Support depends on OpenGL version and platform
typedef enum {