diff --git a/.travis.yml b/.travis.yml index ec4a8ebd7..c27b6582c 100644 --- a/.travis.yml +++ b/.travis.yml @@ -120,7 +120,6 @@ script: -DBUILD_EXAMPLES=ON -DBUILD_GAMES=ON -DUSE_EXTERNAL_GLFW=$USE_EXTERNAL_GLFW -DUSE_WAYLAND=$WAYLAND - -DUSE_OPENAL_BACKEND=$OPENAL -DINCLUDE_EVERYTHING=ON .. - $RUNNER make VERBOSE=1 diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt index 005b22cb9..1b1d0d06d 100644 --- a/src/CMakeLists.txt +++ b/src/CMakeLists.txt @@ -45,13 +45,8 @@ endif() add_definitions("-DRAYLIB_CMAKE=1") if(USE_AUDIO) - if (NOT USE_OPENAL_BACKEND) - file(GLOB mini_al external/mini_al.c) - MESSAGE(STATUS "Audio Backend: mini_al") - else() - find_package(OpenAL REQUIRED) - MESSAGE(STATUS "Audio Backend: OpenAL") - endif() + file(GLOB mini_al external/mini_al.c) + MESSAGE(STATUS "Audio Backend: mini_al") file(GLOB stb_vorbis external/stb_vorbis.c) set(sources ${raylib_sources} ${mini_al} ${stb_vorbis}) else() diff --git a/src/CMakeOptions.txt b/src/CMakeOptions.txt index d4ecb3922..eee3f1a97 100644 --- a/src/CMakeOptions.txt +++ b/src/CMakeOptions.txt @@ -12,11 +12,6 @@ option(SHARED "Build raylib as a dynamic library" OFF) option(STATIC "Build raylib as a static library" ON) option(MACOS_FATLIB "Build fat library for both i386 and x86_64 on macOS" OFF) option(USE_AUDIO "Build raylib with audio module" ON) -if(${PLATFORM} MATCHES "Web") - cmake_dependent_option(USE_OPENAL_BACKEND "Link raylib with openAL instead of mini-al" ON "USE_AUDIO" OFF) -else() - cmake_dependent_option(USE_OPENAL_BACKEND "Link raylib with openAL instead of mini-al" OFF "USE_AUDIO" OFF) -endif() enum_option(USE_EXTERNAL_GLFW "OFF;IF_POSSIBLE;ON" "Link raylib against system GLFW instead of embedded one") if(UNIX AND NOT APPLE) diff --git a/src/Makefile b/src/Makefile index ac4b15b4f..28b73bebe 100644 --- a/src/Makefile +++ b/src/Makefile @@ -63,14 +63,6 @@ RAYLIB_BUILD_MODE ?= RELEASE # NOTE: Some programs like tools could not require audio support INCLUDE_AUDIO_MODULE ?= TRUE -# Use OpenAL Soft backend for audio -USE_OPENAL_BACKEND ?= FALSE - -# OpenAL Soft audio backend forced on HTML5 and OSX (see below) -ifeq ($(PLATFORM),PLATFORM_WEB) - USE_OPENAL_BACKEND = TRUE -endif - # Use external GLFW library instead of rglfw module # TODO: Review usage of examples on Linux. USE_EXTERNAL_GLFW ?= FALSE @@ -154,13 +146,6 @@ endif # RAYLIB_PATH ?= /home/pi/raylib #endif -# Force OpenAL Soft audio backend for OSX platform -# NOTE 1: mini_al library does not support CoreAudio yet -# NOTE 2: Required OpenAL libraries should be available on OSX -ifeq ($(PLATFORM_OS),OSX) - USE_OPENAL_BACKEND = TRUE -endif - ifeq ($(PLATFORM),PLATFORM_WEB) # Emscripten required variables EMSDK_PATH = C:/emsdk @@ -343,11 +328,6 @@ ifeq ($(RAYLIB_LIBTYPE),SHARED) CFLAGS += -fPIC -DBUILD_LIBTYPE_SHARED endif -# Use OpenAL Soft backend instead of mini_al -ifeq ($(USE_OPENAL_BACKEND),TRUE) - CFLAGS += -DUSE_OPENAL_BACKEND -endif - # Use Wayland display on Linux desktop ifeq ($(PLATFORM),PLATFORM_DESKTOP) ifeq ($(PLATFORM_OS), LINUX) @@ -426,9 +406,7 @@ endif ifeq ($(INCLUDE_AUDIO_MODULE),TRUE) OBJS += audio.o OBJS += stb_vorbis.o - ifeq ($(USE_OPENAL_BACKEND),FALSE) - OBJS += mini_al.o - endif + OBJS += mini_al.o endif ifeq ($(PLATFORM),PLATFORM_ANDROID) diff --git a/src/audio.c b/src/audio.c index fc65127bf..a646c981f 100644 --- a/src/audio.c +++ b/src/audio.c @@ -16,9 +16,6 @@ * Define to use the module as standalone library (independently of raylib). * Required types and functions are defined in the same module. * -* #define USE_OPENAL_BACKEND -* Use OpenAL Soft audio backend -* * #define SUPPORT_FILEFORMAT_WAV * #define SUPPORT_FILEFORMAT_OGG * #define SUPPORT_FILEFORMAT_XM @@ -82,25 +79,9 @@ #include "utils.h" // Required for: fopen() Android mapping #endif -#if !defined(USE_OPENAL_BACKEND) - #define USE_MINI_AL 1 // Set to 1 to use mini_al; 0 to use OpenAL. -#endif - -#include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc. - -#if !defined(USE_MINI_AL) || (USE_MINI_AL == 0) - #if defined(__APPLE__) - #include "OpenAL/al.h" // OpenAL basic header - #include "OpenAL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) - #else - #include "AL/al.h" // OpenAL basic header - #include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) - //#include "AL/alext.h" // OpenAL extensions header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS - #endif - - // OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples - // OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1) -#endif +#include "external/mini_al.h" // mini_al audio library + // NOTE: Cannot be implement here because it conflicts with + // Win32 APIs: Rectangle, CloseWindow(), ShowCursor(), PlaySoundA() #include // Required for: malloc(), free() #include // Required for: strcmp(), strncmp() @@ -147,15 +128,6 @@ // In case of music-stalls, just increase this number #define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb) -// Support uncompressed PCM data in 32-bit float IEEE format -// NOTE: This definition is included in "AL/alext.h", but some OpenAL implementations -// could not provide the extensions header (Android), so its defined here -#if !defined(AL_EXT_float32) - #define AL_EXT_float32 1 - #define AL_FORMAT_MONO_FLOAT32 0x10010 - #define AL_FORMAT_STEREO_FLOAT32 0x10011 -#endif - //---------------------------------------------------------------------------------- // Types and Structures Definition //---------------------------------------------------------------------------------- @@ -233,8 +205,6 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo //---------------------------------------------------------------------------------- // mini_al AudioBuffer Functionality //---------------------------------------------------------------------------------- -#if USE_MINI_AL - #define DEVICE_FORMAT mal_format_f32 #define DEVICE_CHANNELS 2 #define DEVICE_SAMPLE_RATE 44100 @@ -487,7 +457,6 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 f } } } -#endif //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Device initialization and Closing @@ -495,7 +464,6 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 f // Initialize audio device void InitAudioDevice(void) { -#if USE_MINI_AL // Context. mal_context_config contextConfig = mal_context_config_init(OnLog); mal_result result = mal_context_init(NULL, 0, &contextConfig, &context); @@ -545,45 +513,11 @@ void InitAudioDevice(void) TraceLog(LOG_INFO, "Audio buffer size: %d", device.bufferSizeInFrames); isAudioInitialized = MAL_TRUE; -#else - // Open and initialize a device with default settings - ALCdevice *device = alcOpenDevice(NULL); - - if (!device) TraceLog(LOG_ERROR, "Audio device could not be opened"); - else - { - ALCcontext *context = alcCreateContext(device, NULL); - - if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE)) - { - if (context != NULL) alcDestroyContext(context); - - alcCloseDevice(device); - - TraceLog(LOG_ERROR, "Could not initialize audio context"); - } - else - { - TraceLog(LOG_INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER)); - - // Listener definition (just for 2D) - alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f); - alListener3f(AL_VELOCITY, 0.0f, 0.0f, 0.0f); - alListener3f(AL_ORIENTATION, 0.0f, 0.0f, -1.0f); - - alListenerf(AL_GAIN, 1.0f); - - if (alIsExtensionPresent("AL_EXT_float32")) TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 supported"); - else TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 not supported"); - } - } -#endif } // Close the audio device for all contexts void CloseAudioDevice(void) { -#if USE_MINI_AL if (!isAudioInitialized) { TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized"); @@ -593,18 +527,6 @@ void CloseAudioDevice(void) mal_mutex_uninit(&audioLock); mal_device_uninit(&device); mal_context_uninit(&context); -#else - ALCdevice *device; - ALCcontext *context = alcGetCurrentContext(); - - if (context == NULL) TraceLog(LOG_WARNING, "Could not get current audio context for closing"); - - device = alcGetContextsDevice(context); - - alcMakeContextCurrent(NULL); - alcDestroyContext(context); - alcCloseDevice(device); -#endif TraceLog(LOG_INFO, "Audio device closed successfully"); } @@ -612,20 +534,7 @@ void CloseAudioDevice(void) // Check if device has been initialized successfully bool IsAudioDeviceReady(void) { -#if USE_MINI_AL return isAudioInitialized; -#else - ALCcontext *context = alcGetCurrentContext(); - - if (context == NULL) return false; - else - { - ALCdevice *device = alcGetContextsDevice(context); - - if (device == NULL) return false; - else return true; - } -#endif } // Set master volume (listener) @@ -634,17 +543,13 @@ void SetMasterVolume(float volume) if (volume < 0.0f) volume = 0.0f; else if (volume > 1.0f) volume = 1.0f; -#if USE_MINI_AL masterVolume = volume; -#else - alListenerf(AL_GAIN, volume); -#endif } //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Buffer management //---------------------------------------------------------------------------------- -#if USE_MINI_AL + // Create a new audio buffer. Initially filled with silence AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage) { @@ -843,7 +748,6 @@ void UntrackAudioBuffer(AudioBuffer *audioBuffer) mal_mutex_unlock(&audioLock); } -#endif //---------------------------------------------------------------------------------- // Module Functions Definition - Sounds loading and playing (.WAV) @@ -909,7 +813,6 @@ Sound LoadSoundFromWave(Wave wave) if (wave.data != NULL) { -#if USE_MINI_AL // When using mini_al we need to do our own mixing. To simplify this we need convert the format of each sound to be consistent with // the format used to open the playback device. We can do this two ways: // @@ -931,61 +834,6 @@ Sound LoadSoundFromWave(Wave wave) if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed"); sound.audioBuffer = audioBuffer; -#else - ALenum format = 0; - - // The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample) - if (wave.channels == 1) - { - switch (wave.sampleSize) - { - case 8: format = AL_FORMAT_MONO8; break; - case 16: format = AL_FORMAT_MONO16; break; - case 32: format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 - default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; - } - } - else if (wave.channels == 2) - { - switch (wave.sampleSize) - { - case 8: format = AL_FORMAT_STEREO8; break; - case 16: format = AL_FORMAT_STEREO16; break; - case 32: format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 - default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; - } - } - else TraceLog(LOG_WARNING, "Wave number of channels not supported: %i", wave.channels); - - // Create an audio source - ALuint source; - alGenSources(1, &source); // Generate pointer to audio source - - alSourcef(source, AL_PITCH, 1.0f); - alSourcef(source, AL_GAIN, 1.0f); - alSource3f(source, AL_POSITION, 0.0f, 0.0f, 0.0f); - alSource3f(source, AL_VELOCITY, 0.0f, 0.0f, 0.0f); - alSourcei(source, AL_LOOPING, AL_FALSE); - - // Convert loaded data to OpenAL buffer - //---------------------------------------- - ALuint buffer; - alGenBuffers(1, &buffer); // Generate pointer to buffer - - unsigned int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; // Size in bytes - - // Upload sound data to buffer - alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate); - - // Attach sound buffer to source - alSourcei(source, AL_BUFFER, buffer); - - TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); - - sound.source = source; - sound.buffer = buffer; - sound.format = format; -#endif } return sound; @@ -1002,14 +850,7 @@ void UnloadWave(Wave wave) // Unload sound void UnloadSound(Sound sound) { -#if USE_MINI_AL DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer); -#else - alSourceStop(sound.source); - - alDeleteSources(1, &sound.source); - alDeleteBuffers(1, &sound.buffer); -#endif TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Unloaded sound data from RAM", sound.source, sound.buffer); } @@ -1018,8 +859,8 @@ void UnloadSound(Sound sound) // NOTE: data must match sound.format void UpdateSound(Sound sound, const void *data, int samplesCount) { -#if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer; + if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer"); @@ -1030,29 +871,6 @@ void UpdateSound(Sound sound, const void *data, int samplesCount) // TODO: May want to lock/unlock this since this data buffer is read at mixing time. memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*mal_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)); -#else - ALint sampleRate, sampleSize, channels; - alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); - alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format - alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format - - TraceLog(LOG_DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate); - TraceLog(LOG_DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize); - TraceLog(LOG_DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels); - - unsigned int dataSize = samplesCount*channels*sampleSize/8; // Size of data in bytes - - alSourceStop(sound.source); // Stop sound - alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update - //alDeleteBuffers(1, &sound.buffer); // Delete current buffer data - //alGenBuffers(1, &sound.buffer); // Generate new buffer - - // Upload new data to sound buffer - alBufferData(sound.buffer, sound.format, data, dataSize, sampleRate); - - // Attach sound buffer to source again - alSourcei(sound.source, AL_BUFFER, sound.buffer); -#endif } // Export wave data to file @@ -1141,102 +959,48 @@ void ExportWave(Wave wave, const char *fileName) // Play a sound void PlaySound(Sound sound) { -#if USE_MINI_AL PlayAudioBuffer((AudioBuffer *)sound.audioBuffer); -#else - alSourcePlay(sound.source); // Play the sound -#endif - - //TraceLog(LOG_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) { -#if USE_MINI_AL PauseAudioBuffer((AudioBuffer *)sound.audioBuffer); -#else - alSourcePause(sound.source); -#endif } // Resume a paused sound void ResumeSound(Sound sound) { -#if USE_MINI_AL ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer); -#else - ALenum state; - - alGetSourcei(sound.source, AL_SOURCE_STATE, &state); - - if (state == AL_PAUSED) alSourcePlay(sound.source); -#endif } // Stop reproducing a sound void StopSound(Sound sound) { -#if USE_MINI_AL StopAudioBuffer((AudioBuffer *)sound.audioBuffer); -#else - alSourceStop(sound.source); -#endif } // Check if a sound is playing bool IsSoundPlaying(Sound sound) { -#if USE_MINI_AL return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer); -#else - bool playing = false; - ALint state; - - alGetSourcei(sound.source, AL_SOURCE_STATE, &state); - if (state == AL_PLAYING) playing = true; - - return playing; -#endif } // Set volume for a sound void SetSoundVolume(Sound sound, float volume) { -#if USE_MINI_AL SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume); -#else - alSourcef(sound.source, AL_GAIN, volume); -#endif } // Set pitch for a sound void SetSoundPitch(Sound sound, float pitch) { -#if USE_MINI_AL SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch); -#else - alSourcef(sound.source, AL_PITCH, pitch); -#endif } // Convert wave data to desired format void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { -#if USE_MINI_AL mal_format formatIn = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_format formatOut = (( sampleSize == 8) ? mal_format_u8 : (( sampleSize == 16) ? mal_format_s16 : mal_format_f32)); @@ -1264,87 +1028,6 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) wave->channels = channels; free(wave->data); wave->data = data; - -#else - // Format sample rate - // NOTE: Only supported 22050 <--> 44100 - if (wave->sampleRate != sampleRate) - { - // TODO: Resample wave data (upsampling or downsampling) - // NOTE 1: To downsample, you have to drop samples or average them. - // NOTE 2: To upsample, you have to interpolate new samples. - - wave->sampleRate = sampleRate; - } - - // Format sample size - // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit - if (wave->sampleSize != sampleSize) - { - void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8); - - for (int i = 0; i < wave->sampleCount; i++) - { - for (int j = 0; j < wave->channels; j++) - { - if (sampleSize == 8) - { - if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256); - else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127); - } - else if (sampleSize == 16) - { - if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767); - else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767); - } - else if (sampleSize == 32) - { - if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f; - else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f; - } - } - } - - wave->sampleSize = sampleSize; - free(wave->data); - wave->data = data; - } - - // Format channels (interlaced mode) - // NOTE: Only supported mono <--> stereo - if (wave->channels != channels) - { - void *data = malloc(wave->sampleCount*wave->sampleSize/8*channels); - - if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information) - { - for (int i = 0; i < wave->sampleCount; i++) - { - for (int j = 0; j < channels; j++) - { - if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i]; - else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i]; - else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i]; - } - } - } - else if ((wave->channels == 2) && (channels == 1)) // stereo ---> mono (mix stereo channels) - { - for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2) - { - if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2; - else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2; - else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f; - } - } - - // TODO: Add/remove additional interlaced channels - - wave->channels = channels; - free(wave->data); - wave->data = data; - } -#endif } // Copy a wave to a new wave @@ -1578,8 +1261,8 @@ void UnloadMusicStream(Music music) // Start music playing (open stream) void PlayMusicStream(Music music) { -#if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer; + if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer"); @@ -1595,61 +1278,25 @@ void PlayMusicStream(Music music) PlayAudioStream(music->stream); // <-- This resets the cursor position. audioBuffer->frameCursorPos = frameCursorPos; -#else - alSourcePlay(music->stream.source); -#endif } // Pause music playing void PauseMusicStream(Music music) { -#if USE_MINI_AL PauseAudioStream(music->stream); -#else - alSourcePause(music->stream.source); -#endif } // Resume music playing void ResumeMusicStream(Music music) { -#if USE_MINI_AL ResumeAudioStream(music->stream); -#else - ALenum state; - alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); - - if (state == AL_PAUSED) - { - TraceLog(LOG_INFO, "[AUD ID %i] Resume music stream playing", music->stream.source); - alSourcePlay(music->stream.source); - } -#endif } // Stop music playing (close stream) // TODO: To clear a buffer, make sure they have been already processed! void StopMusicStream(Music music) { -#if USE_MINI_AL StopAudioStream(music->stream); -#else - alSourceStop(music->stream.source); - - /* - // Clear stream buffers - // WARNING: Queued buffers must have been processed before unqueueing and reloaded with data!!! - void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1); - - for (int i = 0; i < MAX_STREAM_BUFFERS; i++) - { - //UpdateAudioStream(music->stream, pcm, AUDIO_BUFFER_SIZE); // Update one buffer at a time - alBufferData(music->stream.buffers[i], music->stream.format, pcm, AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, music->stream.sampleRate); - } - - free(pcm); - */ -#endif // Restart music context switch (music->ctxType) @@ -1677,7 +1324,6 @@ void StopMusicStream(Music music) // TODO: Make sure buffers are ready for update... check music state void UpdateMusicStream(Music music) { -#if USE_MINI_AL bool streamEnding = false; unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2; @@ -1761,139 +1407,24 @@ void UpdateMusicStream(Music music) // just make sure to play again on window restore if (IsMusicPlaying(music)) PlayMusicStream(music); } -#else - ALenum state; - ALint processed = 0; - - alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state - alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers - - if (processed > 0) - { - bool streamEnding = false; - - // NOTE: Using dynamic allocation because it could require more than 16KB - void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1); - - int numBuffersToProcess = processed; - int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, - // individual L or R for ogg shorts - - for (int i = 0; i < numBuffersToProcess; i++) - { - if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE; - else samplesCount = music->samplesLeft; - - // TODO: Really don't like ctxType thingy... - switch (music->ctxType) - { - case MUSIC_AUDIO_OGG: - { - // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) - int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels); - - } break; - #if defined(SUPPORT_FILEFORMAT_FLAC) - case MUSIC_AUDIO_FLAC: - { - // NOTE: Returns the number of samples to process - unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_MP3) - case MUSIC_AUDIO_MP3: - { - // NOTE: Returns the number of samples to process - unsigned int numSamplesMp3 = (unsigned int)drmp3_read_f32(&music->ctxMp3, samplesCount*music->stream.channels, (float *)pcm); - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_XM) - case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break; - #endif - #if defined(SUPPORT_FILEFORMAT_MOD) - case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break; - #endif - default: break; - } - - UpdateAudioStream(music->stream, pcm, samplesCount); - music->samplesLeft -= samplesCount; - - if (music->samplesLeft <= 0) - { - streamEnding = true; - break; - } - } - - // Free allocated pcm data - free(pcm); - - // Reset audio stream for looping - if (streamEnding) - { - StopMusicStream(music); // Stop music (and reset) - - // Decrease loopCount to stop when required - if (music->loopCount > 0) - { - music->loopCount--; // Decrease loop count - PlayMusicStream(music); // Play again - } - else - { - if (music->loopCount == -1) - { - PlayMusicStream(music); - } - } - } - else - { - // NOTE: In case window is minimized, music stream is stopped, - // just make sure to play again on window restore - if (state != AL_PLAYING) PlayMusicStream(music); - } - } -#endif } // Check if any music is playing bool IsMusicPlaying(Music music) { -#if USE_MINI_AL return IsAudioStreamPlaying(music->stream); -#else - bool playing = false; - ALint state; - - alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); - - if (state == AL_PLAYING) playing = true; - - return playing; -#endif } // Set volume for music void SetMusicVolume(Music music, float volume) { -#if USE_MINI_AL SetAudioStreamVolume(music->stream, volume); -#else - alSourcef(music->stream.source, AL_GAIN, volume); -#endif } // Set pitch for music void SetMusicPitch(Music music, float pitch) { -#if USE_MINI_AL SetAudioStreamPitch(music->stream, pitch); -#else - alSourcef(music->stream.source, AL_PITCH, pitch); -#endif } // Set music loop count (loop repeats) @@ -1939,8 +1470,6 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un stream.channels = 1; // Fallback to mono channel } - -#if USE_MINI_AL mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32)); // The size of a streaming buffer must be at least double the size of a period. @@ -1957,52 +1486,6 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un audioBuffer->looping = true; // Always loop for streaming buffers. stream.audioBuffer = audioBuffer; -#else - // Setup OpenAL format - if (stream.channels == 1) - { - switch (sampleSize) - { - case 8: stream.format = AL_FORMAT_MONO8; break; - case 16: stream.format = AL_FORMAT_MONO16; break; - case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 - default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; - } - } - else if (stream.channels == 2) - { - switch (sampleSize) - { - case 8: stream.format = AL_FORMAT_STEREO8; break; - case 16: stream.format = AL_FORMAT_STEREO16; break; - case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 - default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; - } - } - - // Create an audio source - alGenSources(1, &stream.source); - alSourcef(stream.source, AL_PITCH, 1.0f); - alSourcef(stream.source, AL_GAIN, 1.0f); - alSource3f(stream.source, AL_POSITION, 0.0f, 0.0f, 0.0f); - alSource3f(stream.source, AL_VELOCITY, 0.0f, 0.0f, 0.0f); - - // Create Buffers (double buffering) - alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers); - - // Initialize buffer with zeros by default - // NOTE: Using dynamic allocation because it requires more than 16KB - void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1); - - for (int i = 0; i < MAX_STREAM_BUFFERS; i++) - { - alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate); - } - - free(pcm); - - alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers); -#endif TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo"); @@ -2012,28 +1495,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un // Close audio stream and free memory void CloseAudioStream(AudioStream stream) { -#if USE_MINI_AL DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer); -#else - // Stop playing channel - alSourceStop(stream.source); - - // Flush out all queued buffers - int queued = 0; - alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued); - - ALuint buffer = 0; - - while (queued > 0) - { - alSourceUnqueueBuffers(stream.source, 1, &buffer); - queued--; - } - - // Delete source and buffers - alDeleteSources(1, &stream.source); - alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers); -#endif TraceLog(LOG_INFO, "[AUD ID %i] Unloaded audio stream data", stream.source); } @@ -2043,7 +1505,6 @@ void CloseAudioStream(AudioStream stream) // NOTE 2: To unqueue a buffer it needs to be processed: IsAudioBufferProcessed() void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) { -#if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { @@ -2054,6 +1515,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]) { mal_uint32 subBufferToUpdate; + if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1]) { // Both buffers are available for updating. Update the first one and make sure the cursor is moved back to the front. @@ -2073,6 +1535,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) if (subBufferSizeInFrames >= (mal_uint32)samplesCount) { mal_uint32 framesToWrite = subBufferSizeInFrames; + if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount; mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); @@ -2080,6 +1543,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) // Any leftover frames should be filled with zeros. mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; + if (leftoverFrameCount > 0) { memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); @@ -2098,24 +1562,11 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) TraceLog(LOG_ERROR, "Audio buffer not available for updating"); return; } -#else - ALuint buffer = 0; - alSourceUnqueueBuffers(stream.source, 1, &buffer); - - // Check if any buffer was available for unqueue - if (alGetError() != AL_INVALID_VALUE) - { - alBufferData(buffer, stream.format, data, samplesCount*stream.sampleSize/8*stream.channels, stream.sampleRate); - alSourceQueueBuffers(stream.source, 1, &buffer); - } - else TraceLog(LOG_WARNING, "[AUD ID %i] Audio buffer not available for unqueuing", stream.source); -#endif } // Check if any audio stream buffers requires refill bool IsAudioBufferProcessed(AudioStream stream) { -#if USE_MINI_AL AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { @@ -2124,92 +1575,46 @@ bool IsAudioBufferProcessed(AudioStream stream) } return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]; -#else - ALint processed = 0; - - // Determine if music stream is ready to be written - alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed); - - return (processed > 0); -#endif } // Play audio stream void PlayAudioStream(AudioStream stream) { -#if USE_MINI_AL PlayAudioBuffer((AudioBuffer *)stream.audioBuffer); -#else - alSourcePlay(stream.source); -#endif } // Play audio stream void PauseAudioStream(AudioStream stream) { -#if USE_MINI_AL PauseAudioBuffer((AudioBuffer *)stream.audioBuffer); -#else - alSourcePause(stream.source); -#endif } // Resume audio stream playing void ResumeAudioStream(AudioStream stream) { -#if USE_MINI_AL ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer); -#else - ALenum state; - alGetSourcei(stream.source, AL_SOURCE_STATE, &state); - - if (state == AL_PAUSED) alSourcePlay(stream.source); -#endif } // Check if audio stream is playing. bool IsAudioStreamPlaying(AudioStream stream) { -#if USE_MINI_AL return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer); -#else - bool playing = false; - ALint state; - - alGetSourcei(stream.source, AL_SOURCE_STATE, &state); - - if (state == AL_PLAYING) playing = true; - - return playing; -#endif } // Stop audio stream void StopAudioStream(AudioStream stream) { -#if USE_MINI_AL StopAudioBuffer((AudioBuffer *)stream.audioBuffer); -#else - alSourceStop(stream.source); -#endif } void SetAudioStreamVolume(AudioStream stream, float volume) { -#if USE_MINI_AL SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume); -#else - alSourcef(stream.source, AL_GAIN, volume); -#endif } void SetAudioStreamPitch(AudioStream stream, float pitch) { -#if USE_MINI_AL SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch); -#else - alSourcef(stream.source, AL_PITCH, pitch); -#endif } //---------------------------------------------------------------------------------- diff --git a/src/config.h.in b/src/config.h.in index 742067ce7..b0e624805 100644 --- a/src/config.h.in +++ b/src/config.h.in @@ -1,7 +1,5 @@ /* config.h.in */ -#cmakedefine USE_OPENAL_BACKEND 1 - // core.c /* Camera module is included (camera.h) and multiple predefined cameras are available: free, 1st/3rd person, orbital */ #cmakedefine SUPPORT_CAMERA_SYSTEM 1