1416 lines
50 KiB
C
1416 lines
50 KiB
C
/**********************************************************************************************
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*
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* raylib.audio
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*
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* Basic functions to manage Audio:
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* Manage audio device (init/close)
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* Load and Unload audio files
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* Play/Stop/Pause/Resume loaded audio
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* Manage mixing channels
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* Manage raw audio context
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*
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* Uses external lib:
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* OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html)
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* stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
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* jar_xm - XM module file loading
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* jar_mod - MOD audio file loading
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*
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* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions:
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* XM audio module support (jar_xm)
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* MOD audio module support (jar_mod)
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* Mixing channels support
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* Raw audio context support
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*
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* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5)
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*
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* This software is provided "as-is", without any express or implied warranty. In no event
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* will the authors be held liable for any damages arising from the use of this software.
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*
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* Permission is granted to anyone to use this software for any purpose, including commercial
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* applications, and to alter it and redistribute it freely, subject to the following restrictions:
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*
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* 1. The origin of this software must not be misrepresented; you must not claim that you
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* wrote the original software. If you use this software in a product, an acknowledgment
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* in the product documentation would be appreciated but is not required.
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*
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* 2. Altered source versions must be plainly marked as such, and must not be misrepresented
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* as being the original software.
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*
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* 3. This notice may not be removed or altered from any source distribution.
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*
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**********************************************************************************************/
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//#define AUDIO_STANDALONE // NOTE: To use the audio module as standalone lib, just uncomment this line
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#if defined(AUDIO_STANDALONE)
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#include "audio.h"
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#else
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#include "raylib.h"
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#endif
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#include "AL/al.h" // OpenAL basic header
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#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
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#include <stdlib.h> // Required for: malloc(), free()
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#include <string.h> // Required for: strcmp(), strncmp()
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#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
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#ifndef AL_FORMAT_MONO_FLOAT32
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#define AL_FORMAT_MONO_FLOAT32 0x10010
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#endif
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#ifndef AL_FORMAT_STEREO_FLOAT32
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#define AL_FORMAT_STEREO_FLOAT32 0x10011
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#endif
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#if defined(AUDIO_STANDALONE)
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#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
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#else
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#include "utils.h" // Required for: DecompressData()
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// NOTE: Includes Android fopen() function map
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#endif
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//#define STB_VORBIS_HEADER_ONLY
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#include "external/stb_vorbis.h" // OGG loading functions
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#define JAR_XM_IMPLEMENTATION
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#include "external/jar_xm.h" // XM loading functions
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#define JAR_MOD_IMPLEMENTATION
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#include "external/jar_mod.h" // MOD loading functions
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//----------------------------------------------------------------------------------
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// Defines and Macros
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//----------------------------------------------------------------------------------
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#define MAX_STREAM_BUFFERS 2 // Number of buffers for each source
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#define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources
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#define MAX_MIX_CHANNELS 4 // Number of mix channels (OpenAL sources)
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#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
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// NOTE: On RPI and Android should be lower to avoid frame-stalls
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#define MUSIC_BUFFER_SIZE_SHORT 4096*2 // PCM data buffer (short) - 16Kb (RPI)
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#define MUSIC_BUFFER_SIZE_FLOAT 4096 // PCM data buffer (float) - 16Kb (RPI)
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#else
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// NOTE: On HTML5 (emscripten) this is allocated on heap, by default it's only 16MB!...just take care...
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#define MUSIC_BUFFER_SIZE_SHORT 4096*8 // PCM data buffer (short) - 64Kb
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#define MUSIC_BUFFER_SIZE_FLOAT 4096*4 // PCM data buffer (float) - 64Kb
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#endif
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//----------------------------------------------------------------------------------
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// Types and Structures Definition
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//----------------------------------------------------------------------------------
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// Used to create custom audio streams that are not bound to a specific file.
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// There can be no more than 4 concurrent mixchannels in use.
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// This is due to each active mixc being tied to a dedicated mix channel.
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typedef struct MixChannel {
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unsigned short sampleRate; // default is 48000
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unsigned char channels; // 1=mono,2=stereo
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unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
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bool floatingPoint; // if false then the short datatype is used instead
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bool playing; // false if paused
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ALenum alFormat; // OpenAL format specifier
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ALuint alSource; // OpenAL source
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ALuint alBuffer[MAX_STREAM_BUFFERS]; // OpenAL sample buffer
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} MixChannel;
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// Music type (file streaming from memory)
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// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
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typedef struct Music {
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stb_vorbis *stream;
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jar_xm_context_t *xmctx; // XM chiptune context
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jar_mod_context_t modctx; // MOD chiptune context
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MixChannel *mixc; // Mix channel
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unsigned int totalSamplesLeft;
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float totalLengthSeconds;
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bool loop;
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bool chipTune; // chiptune is loaded?
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bool enabled;
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} Music;
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// Audio errors registered
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typedef enum {
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ERROR_RAW_CONTEXT_CREATION = 1,
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ERROR_XM_CONTEXT_CREATION = 2,
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ERROR_MOD_CONTEXT_CREATION = 4,
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ERROR_MIX_CHANNEL_CREATION = 8,
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ERROR_MUSIC_CHANNEL_CREATION = 16,
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ERROR_LOADING_XM = 32,
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ERROR_LOADING_MOD = 64,
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ERROR_LOADING_WAV = 128,
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ERROR_LOADING_OGG = 256,
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ERROR_OUT_OF_MIX_CHANNELS = 512,
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ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
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ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
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ERROR_INVALID_RRES_FILE = 4096,
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ERROR_INVALID_RRES_RESOURCE = 8192,
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ERROR_UNINITIALIZED_CHANNELS = 16384
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} AudioError;
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#if defined(AUDIO_STANDALONE)
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typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
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#endif
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//----------------------------------------------------------------------------------
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// Global Variables Definition
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//----------------------------------------------------------------------------------
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static Music musicStreams[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time
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static MixChannel *mixChannels[MAX_MIX_CHANNELS]; // Mix channels currently active (from music streams)
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static int lastAudioError = 0; // Registers last audio error
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//----------------------------------------------------------------------------------
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// Module specific Functions Declaration
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//----------------------------------------------------------------------------------
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static Wave LoadWAV(const char *fileName); // Load WAV file
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static Wave LoadOGG(char *fileName); // Load OGG file
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static void UnloadWave(Wave wave); // Unload wave data
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static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
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static void EmptyMusicStream(int index); // Empty music buffers
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static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
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static void CloseMixChannel(MixChannel *mixc); // Frees mix channel
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static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements); // Pushes more audio data into mix channel
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//static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
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//static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
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#if defined(AUDIO_STANDALONE)
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const char *GetExtension(const char *fileName); // Get the extension for a filename
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void TraceLog(int msgType, const char *text, ...); // Outputs a trace log message (INFO, ERROR, WARNING)
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#endif
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Audio Device initialization and Closing
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//----------------------------------------------------------------------------------
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// Initialize audio device and mixc
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void InitAudioDevice(void)
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{
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// Open and initialize a device with default settings
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ALCdevice *device = alcOpenDevice(NULL);
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if (!device) TraceLog(ERROR, "Audio device could not be opened");
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ALCcontext *context = alcCreateContext(device, NULL);
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if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
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{
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if (context != NULL) alcDestroyContext(context);
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alcCloseDevice(device);
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TraceLog(ERROR, "Could not setup mix channel");
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}
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TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
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// Listener definition (just for 2D)
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alListener3f(AL_POSITION, 0, 0, 0);
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alListener3f(AL_VELOCITY, 0, 0, 0);
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alListener3f(AL_ORIENTATION, 0, 0, -1);
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}
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// Close the audio device for all contexts
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void CloseAudioDevice(void)
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{
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for (int index = 0; index < MAX_MUSIC_STREAMS; index++)
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{
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if (musicStreams[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
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}
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ALCdevice *device;
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ALCcontext *context = alcGetCurrentContext();
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if (context == NULL) TraceLog(WARNING, "Could not get current mix channel for closing");
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device = alcGetContextsDevice(context);
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alcMakeContextCurrent(NULL);
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alcDestroyContext(context);
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alcCloseDevice(device);
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}
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// Check if device has been initialized successfully
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bool IsAudioDeviceReady(void)
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{
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ALCcontext *context = alcGetCurrentContext();
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if (context == NULL) return false;
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else
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{
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ALCdevice *device = alcGetContextsDevice(context);
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if (device == NULL) return false;
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else return true;
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}
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}
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Custom audio output
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//----------------------------------------------------------------------------------
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// Init mix channel for streaming
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// The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available.
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// Each mix channel can only be used one at a time.
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static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
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{
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if (mixChannel >= MAX_MIX_CHANNELS) return NULL;
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if (!IsAudioDeviceReady()) InitAudioDevice();
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if (!mixChannels[mixChannel])
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{
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MixChannel *mixc = (MixChannel *)malloc(sizeof(MixChannel));
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mixc->sampleRate = sampleRate;
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mixc->channels = channels;
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mixc->mixChannel = mixChannel;
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mixc->floatingPoint = floatingPoint;
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mixChannels[mixChannel] = mixc;
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// Setup OpenAL format
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if (channels == 1)
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{
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if (floatingPoint) mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
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else mixc->alFormat = AL_FORMAT_MONO16;
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}
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else if (channels == 2)
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{
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if (floatingPoint) mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
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else mixc->alFormat = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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alGenSources(1, &mixc->alSource);
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alSourcef(mixc->alSource, AL_PITCH, 1);
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alSourcef(mixc->alSource, AL_GAIN, 1);
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alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0);
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alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0);
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// Create Buffer
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alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
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// Fill buffers
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for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
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{
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// Initialize buffer with zeros by default
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if (mixc->floatingPoint)
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{
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float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f };
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alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
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}
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else
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{
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short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 };
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alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
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}
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}
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alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
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mixc->playing = true;
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alSourcePlay(mixc->alSource);
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return mixc;
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}
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return NULL;
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}
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// Frees buffer in mix channel
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static void CloseMixChannel(MixChannel *mixc)
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{
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if (mixc)
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{
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alSourceStop(mixc->alSource);
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mixc->playing = false;
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// Flush out all queued buffers
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ALuint buffer = 0;
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int queued = 0;
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alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
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while (queued > 0)
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{
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alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
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queued--;
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}
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// Delete source and buffers
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alDeleteSources(1, &mixc->alSource);
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alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
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mixChannels[mixc->mixChannel] = NULL;
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free(mixc);
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mixc = NULL;
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}
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}
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// Pushes more audio data into mix channel, only one buffer per call
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// Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
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// Returns number of samples that where processed.
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static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements)
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{
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if (!mixc || (mixChannels[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples
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if (!data || !numberElements)
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{
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// Pauses audio until data is given
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if (mixc->playing)
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{
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alSourcePause(mixc->alSource);
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mixc->playing = false;
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}
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return 0;
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}
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else if (!mixc->playing)
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{
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// Restart audio otherwise
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alSourcePlay(mixc->alSource);
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mixc->playing = true;
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}
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ALuint buffer = 0;
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alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
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if (!buffer) return 0;
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if (mixc->floatingPoint)
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{
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// Process float buffers
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float *ptr = (float *)data;
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alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
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}
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else
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{
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// Process short buffers
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short *ptr = (short *)data;
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alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
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}
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alSourceQueueBuffers(mixc->alSource, 1, &buffer);
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return numberElements;
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}
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/*
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// Convert data from short to float
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// example usage:
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// short sh[3] = {1,2,3};float fl[3];
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// ResampleShortToFloat(sh,fl,3);
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static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len)
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{
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for (int i = 0; i < len; i++)
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{
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if (shorts[i] < 0) floats[i] = (float)shorts[i]/32766.0f;
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else floats[i] = (float)shorts[i]/32767.0f;
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}
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}
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// Convert data from float to short
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// example usage:
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// char ch[3] = {1,2,3};float fl[3];
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// ResampleByteToFloat(ch,fl,3);
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static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
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{
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for (int i = 0; i < len; i++)
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{
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if (chars[i] < 0) floats[i] = (float)chars[i]/127.0f;
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else floats[i] = (float)chars[i]/128.0f;
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}
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}
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*/
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// Initialize raw audio mix channel for audio buffering
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// NOTE: Returns mix channel index or -1 if it fails (errors are registered on lastAudioError)
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int InitRawMixChannel(int sampleRate, int channels, bool floatingPoint)
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{
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int mixIndex;
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for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
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{
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if (mixChannels[mixIndex] == NULL) break;
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else if (mixIndex == (MAX_MIX_CHANNELS - 1))
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{
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lastAudioError = ERROR_OUT_OF_MIX_CHANNELS;
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return -1;
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}
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}
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if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex;
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else
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{
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lastAudioError = ERROR_RAW_CONTEXT_CREATION;
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return -1;
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}
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}
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// Buffers data directly to raw mix channel
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// if 0 is returned, buffers are still full and you need to keep trying with the same data
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// otherwise it will return number of samples buffered.
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// NOTE: Data could be either be an array of floats or shorts, depending on the created context
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int BufferRawAudioContext(int ctx, void *data, unsigned short numberElements)
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{
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int numBuffered = 0;
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if (ctx >= 0)
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{
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MixChannel *mixc = mixChannels[ctx];
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numBuffered = BufferMixChannel(mixc, data, numberElements);
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}
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return numBuffered;
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}
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// Closes and frees raw mix channel
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void CloseRawAudioContext(int ctx)
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{
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if (mixChannels[ctx]) CloseMixChannel(mixChannels[ctx]);
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}
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Sounds loading and playing (.WAV)
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//----------------------------------------------------------------------------------
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// Load sound to memory
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Sound LoadSound(char *fileName)
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{
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Sound sound = { 0 };
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Wave wave = { 0 };
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// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
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// Audio file loading
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// NOTE: Buffer space is allocated inside function, Wave must be freed
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if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
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else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
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else
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{
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TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
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// TODO: Find a better way to register errors (similar to glGetError())
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lastAudioError = ERROR_EXTENSION_NOT_RECOGNIZED;
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}
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if (wave.data != NULL)
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{
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ALenum format = 0;
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// The OpenAL format is worked out by looking at the number of channels and the bits per sample
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if (wave.channels == 1)
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{
|
|
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
|
|
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
|
|
}
|
|
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_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;
|
|
alGenBuffers(1, &buffer); // Generate pointer to buffer
|
|
|
|
// Upload sound data to buffer
|
|
alBufferData(buffer, format, wave.data, wave.dataSize, wave.sampleRate);
|
|
|
|
// Attach sound buffer to source
|
|
alSourcei(source, AL_BUFFER, buffer);
|
|
|
|
TraceLog(INFO, "[%s] Sound file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
|
|
|
|
// Unallocate WAV data
|
|
UnloadWave(wave);
|
|
|
|
sound.source = source;
|
|
sound.buffer = buffer;
|
|
}
|
|
|
|
return sound;
|
|
}
|
|
|
|
// Load sound from wave data
|
|
Sound LoadSoundFromWave(Wave wave)
|
|
{
|
|
Sound sound = { 0 };
|
|
|
|
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.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
|
|
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
|
|
}
|
|
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_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;
|
|
alGenBuffers(1, &buffer); // Generate pointer to buffer
|
|
|
|
// Upload sound data to buffer
|
|
alBufferData(buffer, format, wave.data, wave.dataSize, wave.sampleRate);
|
|
|
|
// Attach sound buffer to source
|
|
alSourcei(source, AL_BUFFER, buffer);
|
|
|
|
// Unallocate WAV data
|
|
UnloadWave(wave);
|
|
|
|
TraceLog(INFO, "[Wave] Sound file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", wave.sampleRate, wave.bitsPerSample, wave.channels);
|
|
|
|
sound.source = source;
|
|
sound.buffer = buffer;
|
|
}
|
|
|
|
return sound;
|
|
}
|
|
|
|
// Load sound to memory from rRES file (raylib Resource)
|
|
// TODO: Maybe rresName could be directly a char array with all the data?
|
|
Sound LoadSoundFromRES(const char *rresName, int resId)
|
|
{
|
|
Sound sound = { 0 };
|
|
|
|
#if defined(AUDIO_STANDALONE)
|
|
TraceLog(WARNING, "Sound loading from rRES resource file not supported on standalone mode");
|
|
#else
|
|
|
|
bool found = false;
|
|
|
|
char id[4]; // rRES file identifier
|
|
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 == NULL)
|
|
{
|
|
TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName);
|
|
lastAudioError = ERROR_UNABLE_TO_OPEN_RRES_FILE;
|
|
}
|
|
else
|
|
{
|
|
// Read rres file (basic file check - id)
|
|
fread(&id[0], sizeof(char), 1, rresFile);
|
|
fread(&id[1], sizeof(char), 1, rresFile);
|
|
fread(&id[2], sizeof(char), 1, rresFile);
|
|
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);
|
|
lastAudioError = ERROR_INVALID_RRES_FILE;
|
|
}
|
|
else
|
|
{
|
|
// 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;
|
|
|
|
// Check data is of valid SOUND type
|
|
if (infoHeader.type == 1) // SOUND data type
|
|
{
|
|
// 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.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
|
|
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
|
|
}
|
|
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_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;
|
|
alGenBuffers(1, &buffer); // Generate pointer to buffer
|
|
|
|
// Upload sound data to buffer
|
|
alBufferData(buffer, format, (void*)wave.data, wave.dataSize, wave.sampleRate);
|
|
|
|
// Attach sound buffer to source
|
|
alSourcei(source, AL_BUFFER, buffer);
|
|
|
|
TraceLog(INFO, "[%s] Sound loaded successfully from resource (SampleRate: %i, BitRate: %i, Channels: %i)", rresName, wave.sampleRate, wave.bitsPerSample, wave.channels);
|
|
|
|
// Unallocate WAV data
|
|
UnloadWave(wave);
|
|
|
|
sound.source = source;
|
|
sound.buffer = buffer;
|
|
}
|
|
else
|
|
{
|
|
TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
|
|
lastAudioError = ERROR_INVALID_RRES_RESOURCE;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// Depending on type, skip the right amount of parameters
|
|
switch (infoHeader.type)
|
|
{
|
|
case 0: fseek(rresFile, 6, SEEK_CUR); break; // IMAGE: Jump 6 bytes of parameters
|
|
case 1: fseek(rresFile, 6, SEEK_CUR); break; // SOUND: Jump 6 bytes of parameters
|
|
case 2: fseek(rresFile, 5, SEEK_CUR); break; // MODEL: Jump 5 bytes of parameters (TODO: Review)
|
|
case 3: break; // TEXT: No parameters
|
|
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);
|
|
#endif
|
|
return sound;
|
|
}
|
|
|
|
// Unload sound
|
|
void UnloadSound(Sound sound)
|
|
{
|
|
alDeleteSources(1, &sound.source);
|
|
alDeleteBuffers(1, &sound.buffer);
|
|
|
|
TraceLog(INFO, "Unloaded sound data");
|
|
}
|
|
|
|
// Play a sound
|
|
void PlaySound(Sound sound)
|
|
{
|
|
alSourcePlay(sound.source); // Play the sound
|
|
|
|
//TraceLog(INFO, "Playing sound");
|
|
|
|
// Find the current position of the sound being played
|
|
// NOTE: Only work when the entire file is in a single buffer
|
|
//int byteOffset;
|
|
//alGetSourcei(sound.source, AL_BYTE_OFFSET, &byteOffset);
|
|
//
|
|
//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;
|
|
//alGetSourcef(sound.source, AL_SEC_OFFSET, &result); // AL_SAMPLE_OFFSET
|
|
}
|
|
|
|
// Pause a sound
|
|
void PauseSound(Sound sound)
|
|
{
|
|
alSourcePause(sound.source);
|
|
}
|
|
|
|
// Stop reproducing a sound
|
|
void StopSound(Sound sound)
|
|
{
|
|
alSourceStop(sound.source);
|
|
}
|
|
|
|
// Check if a sound is playing
|
|
bool IsSoundPlaying(Sound sound)
|
|
{
|
|
bool playing = false;
|
|
ALint state;
|
|
|
|
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
|
|
if (state == AL_PLAYING) playing = true;
|
|
|
|
return playing;
|
|
}
|
|
|
|
// Set volume for a sound
|
|
void SetSoundVolume(Sound sound, float volume)
|
|
{
|
|
alSourcef(sound.source, AL_GAIN, volume);
|
|
}
|
|
|
|
// Set pitch for a sound
|
|
void SetSoundPitch(Sound sound, float pitch)
|
|
{
|
|
alSourcef(sound.source, AL_PITCH, pitch);
|
|
}
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module Functions Definition - Music loading and stream playing (.OGG)
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Start music playing (open stream)
|
|
// returns 0 on success or error code
|
|
int PlayMusicStream(int index, char *fileName)
|
|
{
|
|
int mixIndex;
|
|
|
|
if (musicStreams[index].stream || musicStreams[index].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error
|
|
|
|
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
|
|
}
|
|
|
|
if (strcmp(GetExtension(fileName),"ogg") == 0)
|
|
{
|
|
// Open audio stream
|
|
musicStreams[index].stream = stb_vorbis_open_filename(fileName, NULL, NULL);
|
|
|
|
if (musicStreams[index].stream == NULL)
|
|
{
|
|
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
|
|
return ERROR_LOADING_OGG; // error
|
|
}
|
|
else
|
|
{
|
|
// Get file info
|
|
stb_vorbis_info info = stb_vorbis_get_info(musicStreams[index].stream);
|
|
|
|
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
|
|
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
|
|
TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
|
|
|
|
musicStreams[index].loop = true; // We loop by default
|
|
musicStreams[index].enabled = true;
|
|
|
|
|
|
musicStreams[index].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicStreams[index].stream) * info.channels;
|
|
musicStreams[index].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicStreams[index].stream);
|
|
|
|
if (info.channels == 2)
|
|
{
|
|
musicStreams[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
|
|
musicStreams[index].mixc->playing = true;
|
|
}
|
|
else
|
|
{
|
|
musicStreams[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
|
|
musicStreams[index].mixc->playing = true;
|
|
}
|
|
|
|
if (!musicStreams[index].mixc) return ERROR_LOADING_OGG; // error
|
|
}
|
|
}
|
|
else if (strcmp(GetExtension(fileName),"xm") == 0)
|
|
{
|
|
// only stereo is supported for xm
|
|
if (!jar_xm_create_context_from_file(&musicStreams[index].xmctx, 48000, fileName))
|
|
{
|
|
musicStreams[index].chipTune = true;
|
|
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)/48000.0f;
|
|
musicStreams[index].enabled = true;
|
|
|
|
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicStreams[index].totalSamplesLeft);
|
|
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicStreams[index].totalLengthSeconds);
|
|
|
|
musicStreams[index].mixc = InitMixChannel(48000, mixIndex, 2, true);
|
|
|
|
if (!musicStreams[index].mixc) return ERROR_XM_CONTEXT_CREATION; // error
|
|
|
|
musicStreams[index].mixc->playing = true;
|
|
}
|
|
else
|
|
{
|
|
TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
|
|
return ERROR_LOADING_XM; // error
|
|
}
|
|
}
|
|
else if (strcmp(GetExtension(fileName),"mod") == 0)
|
|
{
|
|
jar_mod_init(&musicStreams[index].modctx);
|
|
|
|
if (jar_mod_load_file(&musicStreams[index].modctx, 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)/48000.0f;
|
|
musicStreams[index].enabled = true;
|
|
|
|
TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicStreams[index].totalSamplesLeft);
|
|
TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, musicStreams[index].totalLengthSeconds);
|
|
|
|
musicStreams[index].mixc = InitMixChannel(48000, mixIndex, 2, false);
|
|
|
|
if (!musicStreams[index].mixc) return ERROR_MOD_CONTEXT_CREATION; // error
|
|
|
|
musicStreams[index].mixc->playing = true;
|
|
}
|
|
else
|
|
{
|
|
TraceLog(WARNING, "[%s] MOD file could not be opened", fileName);
|
|
return ERROR_LOADING_MOD; // error
|
|
}
|
|
}
|
|
else
|
|
{
|
|
TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
|
|
return ERROR_EXTENSION_NOT_RECOGNIZED; // error
|
|
}
|
|
|
|
return 0; // normal return
|
|
}
|
|
|
|
// Stop music playing for individual music index of musicStreams array (close stream)
|
|
void StopMusicStream(int index)
|
|
{
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
CloseMixChannel(musicStreams[index].mixc);
|
|
|
|
if (musicStreams[index].xmctx)
|
|
jar_xm_free_context(musicStreams[index].xmctx);
|
|
else if (musicStreams[index].modctx.mod_loaded)
|
|
jar_mod_unload(&musicStreams[index].modctx);
|
|
else
|
|
stb_vorbis_close(musicStreams[index].stream);
|
|
|
|
musicStreams[index].enabled = false;
|
|
|
|
if (musicStreams[index].stream || musicStreams[index].xmctx)
|
|
{
|
|
musicStreams[index].stream = NULL;
|
|
musicStreams[index].xmctx = NULL;
|
|
}
|
|
}
|
|
}
|
|
|
|
// 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)
|
|
{
|
|
int musicCount = 0;
|
|
|
|
// Find empty music slot
|
|
for (int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++)
|
|
{
|
|
if(musicStreams[musicIndex].stream != NULL || musicStreams[musicIndex].chipTune) musicCount++;
|
|
}
|
|
|
|
return musicCount;
|
|
}
|
|
|
|
// Pause music playing
|
|
void PauseMusicStream(int index)
|
|
{
|
|
// Pause music stream if music available!
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc && musicStreams[index].enabled)
|
|
{
|
|
TraceLog(INFO, "Pausing music stream");
|
|
alSourcePause(musicStreams[index].mixc->alSource);
|
|
musicStreams[index].mixc->playing = false;
|
|
}
|
|
}
|
|
|
|
// Resume music playing
|
|
void ResumeMusicStream(int index)
|
|
{
|
|
// Resume music playing... if music available!
|
|
ALenum state;
|
|
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state);
|
|
|
|
if (state == AL_PAUSED)
|
|
{
|
|
TraceLog(INFO, "Resuming music stream");
|
|
alSourcePlay(musicStreams[index].mixc->alSource);
|
|
musicStreams[index].mixc->playing = true;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Check if any music is playing
|
|
bool IsMusicPlaying(int index)
|
|
{
|
|
bool playing = false;
|
|
ALint state;
|
|
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state);
|
|
|
|
if (state == AL_PLAYING) playing = true;
|
|
}
|
|
|
|
return playing;
|
|
}
|
|
|
|
// Set volume for music
|
|
void SetMusicVolume(int index, float volume)
|
|
{
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
alSourcef(musicStreams[index].mixc->alSource, AL_GAIN, volume);
|
|
}
|
|
}
|
|
|
|
// Set pitch for music
|
|
void SetMusicPitch(int index, float pitch)
|
|
{
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
alSourcef(musicStreams[index].mixc->alSource, AL_PITCH, pitch);
|
|
}
|
|
}
|
|
|
|
// Get music time length (in seconds)
|
|
float GetMusicTimeLength(int index)
|
|
{
|
|
float totalSeconds;
|
|
|
|
if (musicStreams[index].chipTune) totalSeconds = (float)musicStreams[index].totalLengthSeconds;
|
|
else totalSeconds = stb_vorbis_stream_length_in_seconds(musicStreams[index].stream);
|
|
|
|
return totalSeconds;
|
|
}
|
|
|
|
// Get current music time played (in seconds)
|
|
float GetMusicTimePlayed(int index)
|
|
{
|
|
float secondsPlayed = 0.0f;
|
|
|
|
if (index < MAX_MUSIC_STREAMS && musicStreams[index].mixc)
|
|
{
|
|
if (musicStreams[index].chipTune && musicStreams[index].xmctx)
|
|
{
|
|
uint64_t samples;
|
|
jar_xm_get_position(musicStreams[index].xmctx, NULL, NULL, NULL, &samples);
|
|
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/(48000.0f);
|
|
}
|
|
else
|
|
{
|
|
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);
|
|
}
|
|
}
|
|
|
|
return secondsPlayed;
|
|
}
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module specific Functions Definition
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Fill music buffers with new data from music stream
|
|
static bool BufferMusicStream(int index, int numBuffers)
|
|
{
|
|
short pcm[MUSIC_BUFFER_SIZE_SHORT];
|
|
float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
|
|
|
|
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)
|
|
|
|
if (musicStreams[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
|
|
{
|
|
for (int i = 0; i < numBuffers; i++)
|
|
{
|
|
if (musicStreams[index].modctx.mod_loaded)
|
|
{
|
|
if (musicStreams[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT/2;
|
|
else size = musicStreams[index].totalSamplesLeft/2;
|
|
|
|
jar_mod_fillbuffer(&musicStreams[index].modctx, pcm, size, 0 );
|
|
BufferMixChannel(musicStreams[index].mixc, pcm, size*2);
|
|
}
|
|
else if (musicStreams[index].xmctx)
|
|
{
|
|
if (musicStreams[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT) size = MUSIC_BUFFER_SIZE_FLOAT/2;
|
|
else size = musicStreams[index].totalSamplesLeft/2;
|
|
|
|
jar_xm_generate_samples(musicStreams[index].xmctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
|
|
BufferMixChannel(musicStreams[index].mixc, pcmf, size*2);
|
|
}
|
|
|
|
musicStreams[index].totalSamplesLeft -= size;
|
|
|
|
if (musicStreams[index].totalSamplesLeft <= 0)
|
|
{
|
|
active = false;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (musicStreams[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT;
|
|
else size = musicStreams[index].totalSamplesLeft;
|
|
|
|
for (int i = 0; i < numBuffers; i++)
|
|
{
|
|
int streamedBytes = stb_vorbis_get_samples_short_interleaved(musicStreams[index].stream, musicStreams[index].mixc->channels, pcm, size);
|
|
BufferMixChannel(musicStreams[index].mixc, pcm, streamedBytes * musicStreams[index].mixc->channels);
|
|
musicStreams[index].totalSamplesLeft -= streamedBytes * musicStreams[index].mixc->channels;
|
|
|
|
if (musicStreams[index].totalSamplesLeft <= 0)
|
|
{
|
|
active = false;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
return active;
|
|
}
|
|
|
|
// Empty music buffers
|
|
static void EmptyMusicStream(int index)
|
|
{
|
|
ALuint buffer = 0;
|
|
int queued = 0;
|
|
|
|
alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued);
|
|
|
|
while (queued > 0)
|
|
{
|
|
alSourceUnqueueBuffers(musicStreams[index].mixc->alSource, 1, &buffer);
|
|
|
|
queued--;
|
|
}
|
|
}
|
|
|
|
// Load WAV file into Wave structure
|
|
static Wave LoadWAV(const char *fileName)
|
|
{
|
|
// Basic WAV headers structs
|
|
typedef struct {
|
|
char chunkID[4];
|
|
int chunkSize;
|
|
char format[4];
|
|
} RiffHeader;
|
|
|
|
typedef struct {
|
|
char subChunkID[4];
|
|
int subChunkSize;
|
|
short audioFormat;
|
|
short numChannels;
|
|
int sampleRate;
|
|
int byteRate;
|
|
short blockAlign;
|
|
short bitsPerSample;
|
|
} WaveFormat;
|
|
|
|
typedef struct {
|
|
char subChunkID[4];
|
|
int subChunkSize;
|
|
} WaveData;
|
|
|
|
RiffHeader riffHeader;
|
|
WaveFormat waveFormat;
|
|
WaveData waveData;
|
|
|
|
Wave wave = { 0 };
|
|
FILE *wavFile;
|
|
|
|
wavFile = fopen(fileName, "rb");
|
|
|
|
if (wavFile == NULL)
|
|
{
|
|
TraceLog(WARNING, "[%s] WAV file could not be opened", fileName);
|
|
wave.data = NULL;
|
|
}
|
|
else
|
|
{
|
|
// Read in the first chunk into the struct
|
|
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
|
|
|
|
// Check for RIFF and WAVE tags
|
|
if (strncmp(riffHeader.chunkID, "RIFF", 4) ||
|
|
strncmp(riffHeader.format, "WAVE", 4))
|
|
{
|
|
TraceLog(WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
|
|
}
|
|
else
|
|
{
|
|
// 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] != ' '))
|
|
{
|
|
TraceLog(WARNING, "[%s] Invalid Wave format", fileName);
|
|
}
|
|
else
|
|
{
|
|
// 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'))
|
|
{
|
|
TraceLog(WARNING, "[%s] Invalid data header", fileName);
|
|
}
|
|
else
|
|
{
|
|
// Allocate memory for data
|
|
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] WAV file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
|
|
}
|
|
}
|
|
}
|
|
|
|
fclose(wavFile);
|
|
}
|
|
|
|
return wave;
|
|
}
|
|
|
|
// Load OGG file into Wave structure
|
|
// NOTE: Using stb_vorbis library
|
|
static Wave LoadOGG(char *fileName)
|
|
{
|
|
Wave wave;
|
|
|
|
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
|
|
|
|
if (oggFile == NULL)
|
|
{
|
|
TraceLog(WARNING, "[%s] OGG file could not be opened", fileName);
|
|
wave.data = NULL;
|
|
}
|
|
else
|
|
{
|
|
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);
|
|
|
|
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);
|
|
|
|
TraceLog(INFO, "[%s] OGG file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
|
|
|
|
stb_vorbis_close(oggFile);
|
|
}
|
|
|
|
return wave;
|
|
}
|
|
|
|
// Unload Wave data
|
|
static void UnloadWave(Wave wave)
|
|
{
|
|
free(wave.data);
|
|
|
|
TraceLog(INFO, "Unloaded wave data");
|
|
}
|
|
|
|
// Some required functions for audio standalone module version
|
|
#if defined(AUDIO_STANDALONE)
|
|
// Get the extension for a filename
|
|
const char *GetExtension(const char *fileName)
|
|
{
|
|
const char *dot = strrchr(fileName, '.');
|
|
if(!dot || dot == fileName) return "";
|
|
return (dot + 1);
|
|
}
|
|
|
|
// Outputs a trace log message (INFO, ERROR, WARNING)
|
|
// NOTE: If a file has been init, output log is written there
|
|
void TraceLog(int msgType, const char *text, ...)
|
|
{
|
|
va_list args;
|
|
int traceDebugMsgs = 0;
|
|
|
|
#ifdef DO_NOT_TRACE_DEBUG_MSGS
|
|
traceDebugMsgs = 0;
|
|
#endif
|
|
|
|
switch(msgType)
|
|
{
|
|
case INFO: fprintf(stdout, "INFO: "); break;
|
|
case ERROR: fprintf(stdout, "ERROR: "); break;
|
|
case WARNING: fprintf(stdout, "WARNING: "); break;
|
|
case DEBUG: if (traceDebugMsgs) fprintf(stdout, "DEBUG: "); break;
|
|
default: break;
|
|
}
|
|
|
|
if ((msgType != DEBUG) || ((msgType == DEBUG) && (traceDebugMsgs)))
|
|
{
|
|
va_start(args, text);
|
|
vfprintf(stdout, text, args);
|
|
va_end(args);
|
|
|
|
fprintf(stdout, "\n");
|
|
}
|
|
|
|
if (msgType == ERROR) exit(1); // If ERROR message, exit program
|
|
}
|
|
#endif |