From 7959ccd84dd9eedd7aad10fe8b6bea988f828f40 Mon Sep 17 00:00:00 2001 From: raysan5 Date: Fri, 15 Jul 2016 18:16:34 +0200 Subject: [PATCH] Review some functions, formatting and comments --- src/audio.c | 240 +++++++++++++++++++++++++++------------------------ src/audio.h | 31 ++++--- src/raylib.h | 2 - 3 files changed, 148 insertions(+), 125 deletions(-) diff --git a/src/audio.c b/src/audio.c index 2941b9fbf..38fefd12e 100644 --- a/src/audio.c +++ b/src/audio.c @@ -2,13 +2,26 @@ * * raylib.audio * -* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles +* Basic functions to manage Audio: +* Manage audio device (init/close) +* Load and Unload audio files +* Play/Stop/Pause/Resume loaded audio +* Manage mixing channels +* Manage raw audio context * * Uses external lib: * OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html) * stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/) +* jar_xm - XM module file loading +* jar_mod - MOD audio file loading * -* Copyright (c) 2014 Ramon Santamaria (@raysan5) +* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions: +* XM audio module support (jar_xm) +* MOD audio module support (jar_mod) +* Mixing channels support +* Raw audio context support +* +* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5) * * This software is provided "as-is", without any express or implied warranty. In no event * will the authors be held liable for any damages arising from the use of this software. @@ -68,9 +81,9 @@ //---------------------------------------------------------------------------------- // Defines and Macros //---------------------------------------------------------------------------------- -#define MAX_STREAM_BUFFERS 2 // Number of buffers for each alSource -#define MAX_MIX_CHANNELS 4 // Number of OpenAL sources +#define MAX_STREAM_BUFFERS 2 // Number of buffers for each source #define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources +#define MAX_MIX_CHANNELS 4 // Number of mix channels (OpenAL sources) #if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID) // NOTE: On RPI and Android should be lower to avoid frame-stalls @@ -143,7 +156,7 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType; // Global Variables Definition //---------------------------------------------------------------------------------- static Music musicStreams[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time -static MixChannel *mixChannels[MAX_MIX_CHANNELS]; // What mix channels are currently active +static MixChannel *mixChannels[MAX_MIX_CHANNELS]; // Mix channels currently active (from music streams) static int lastAudioError = 0; // Registers last audio error @@ -157,13 +170,11 @@ static void UnloadWave(Wave wave); // Unload wave data static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data static void EmptyMusicStream(int index); // Empty music buffers -static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels. +static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); static void CloseMixChannel(MixChannel *mixc); // Frees mix channel -static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses -static int FillAlBufferWithSilence(MixChannel *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed -static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in -static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in -static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled +static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements); // Pushes more audio data into mix channel +//static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in +//static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in #if defined(AUDIO_STANDALONE) const char *GetExtension(const char *fileName); // Get the extension for a filename @@ -204,7 +215,7 @@ void InitAudioDevice(void) // Close the audio device for all contexts void CloseAudioDevice(void) { - for (int index=0; index= MAX_MIX_CHANNELS) return NULL; @@ -280,7 +291,20 @@ static MixChannel *InitMixChannel(unsigned short sampleRate, unsigned char mixCh alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); // Fill buffers - for (int i = 0; i < MAX_STREAM_BUFFERS; i++) FillAlBufferWithSilence(mixc, mixc->alBuffer[i]); + for (int i = 0; i < MAX_STREAM_BUFFERS; i++) + { + // Initialize buffer with zeros by default + if (mixc->floatingPoint) + { + float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f }; + alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate); + } + else + { + short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 }; + alBufferData(mixc->alBuffer[i], mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate); + } + } alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer); mixc->playing = true; @@ -320,9 +344,9 @@ static void CloseMixChannel(MixChannel *mixc) } } -// Pushes more audio data into mixc mix channel, only one buffer per call +// Pushes more audio data into mix channel, only one buffer per call // Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio. -// @Returns number of samples that where processed. +// Returns number of samples that where processed. static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements) { if (!mixc || (mixChannels[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples @@ -368,28 +392,11 @@ static int BufferMixChannel(MixChannel *mixc, void *data, int numberElements) return numberElements; } -// fill buffer with zeros, returns number processed -static int FillAlBufferWithSilence(MixChannel *mixc, ALuint buffer) -{ - if (mixc->floatingPoint) - { - float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f }; - alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate); - - return MUSIC_BUFFER_SIZE_FLOAT; - } - else - { - short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 }; - alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate); - - return MUSIC_BUFFER_SIZE_SHORT; - } -} - +/* +// Convert data from short to float // example usage: -// short sh[3] = {1,2,3};float fl[3]; -// ResampleShortToFloat(sh,fl,3); +// short sh[3] = {1,2,3};float fl[3]; +// ResampleShortToFloat(sh,fl,3); static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len) { for (int i = 0; i < len; i++) @@ -399,9 +406,10 @@ static void ResampleShortToFloat(short *shorts, float *floats, unsigned short le } } +// Convert data from float to short // example usage: -// char ch[3] = {1,2,3};float fl[3]; -// ResampleByteToFloat(ch,fl,3); +// char ch[3] = {1,2,3};float fl[3]; +// ResampleByteToFloat(ch,fl,3); static void ResampleByteToFloat(char *chars, float *floats, unsigned short len) { for (int i = 0; i < len; i++) @@ -410,43 +418,55 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len) else floats[i] = (float)chars[i]/128.0f; } } +*/ -// used to output raw audio streams, returns negative numbers on error, + number represents the mix channel index -// if floating point is false the data size is 16bit short, otherwise it is float 32bit -RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint) +// Initialize raw audio mix channel for audio buffering +// NOTE: Returns mix channel index or -1 if it fails (errors are registered on lastAudioError) +int InitRawMixChannel(int sampleRate, int channels, bool floatingPoint) { int mixIndex; + for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot { if (mixChannels[mixIndex] == NULL) break; - else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error + else if (mixIndex == (MAX_MIX_CHANNELS - 1)) + { + lastAudioError = ERROR_OUT_OF_MIX_CHANNELS; + return -1; + } } if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex; - else return ERROR_RAW_CONTEXT_CREATION; // error -} - -void CloseRawAudioContext(RawAudioContext ctx) -{ - if (mixChannels[ctx]) CloseMixChannel(mixChannels[ctx]); + else + { + lastAudioError = ERROR_RAW_CONTEXT_CREATION; + return -1; + } } -// if 0 is returned, the buffers are still full and you need to keep trying with the same data until a + number is returned. -// any + number returned is the number of samples that was processed and passed into buffer. -// data either needs to be array of floats or shorts. -int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements) +// Buffers data directly to raw mix channel +// if 0 is returned, buffers are still full and you need to keep trying with the same data +// otherwise it will return number of samples buffered. +// NOTE: Data could be either be an array of floats or shorts, depending on the created context +int BufferRawAudioContext(int ctx, void *data, unsigned short numberElements) { int numBuffered = 0; if (ctx >= 0) { - MixChannel* mixc = mixChannels[ctx]; + MixChannel *mixc = mixChannels[ctx]; numBuffered = BufferMixChannel(mixc, data, numberElements); } return numBuffered; } +// Closes and frees raw mix channel +void CloseRawAudioContext(int ctx) +{ + if (mixChannels[ctx]) CloseMixChannel(mixChannels[ctx]); +} + //---------------------------------------------------------------------------------- // Module Functions Definition - Sounds loading and playing (.WAV) //---------------------------------------------------------------------------------- @@ -804,7 +824,7 @@ void SetSoundPitch(Sound sound, float pitch) //---------------------------------------------------------------------------------- // Start music playing (open stream) -// returns 0 on success +// returns 0 on success or error code int PlayMusicStream(int index, char *fileName) { int mixIndex; @@ -866,7 +886,7 @@ int PlayMusicStream(int index, char *fileName) musicStreams[index].loop = true; jar_xm_set_max_loop_count(musicStreams[index].xmctx, 0); // infinite number of loops musicStreams[index].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicStreams[index].xmctx); - musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft) / 48000.f; + musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft)/48000.0f; musicStreams[index].enabled = true; TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicStreams[index].totalSamplesLeft); @@ -893,7 +913,7 @@ int PlayMusicStream(int index, char *fileName) musicStreams[index].chipTune = true; musicStreams[index].loop = true; musicStreams[index].totalSamplesLeft = (unsigned int)jar_mod_max_samples(&musicStreams[index].modctx); - musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft) / 48000.f; + musicStreams[index].totalLengthSeconds = ((float)musicStreams[index].totalSamplesLeft)/48000.0f; musicStreams[index].enabled = true; TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicStreams[index].totalSamplesLeft); @@ -944,6 +964,51 @@ void StopMusicStream(int index) } } +// Update (re-fill) music buffers if data already processed +void UpdateMusicStream(int index) +{ + ALenum state; + bool active = true; + ALint processed = 0; + + // Determine if music stream is ready to be written + alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); + + if (musicStreams[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicStreams[index].enabled && musicStreams[index].mixc && (processed > 0)) + { + active = BufferMusicStream(index, processed); + + if (!active && musicStreams[index].loop) + { + if (musicStreams[index].chipTune) + { + if(musicStreams[index].modctx.mod_loaded) jar_mod_seek_start(&musicStreams[index].modctx); + + musicStreams[index].totalSamplesLeft = musicStreams[index].totalLengthSeconds*48000.0f; + } + else + { + stb_vorbis_seek_start(musicStreams[index].stream); + musicStreams[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicStreams[index].stream)*musicStreams[index].mixc->channels; + } + + // Determine if music stream is ready to be written + alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); + + active = BufferMusicStream(index, processed); + } + + if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); + + alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state); + + if (state != AL_PLAYING && active) alSourcePlay(musicStreams[index].mixc->alSource); + + if (!active) StopMusicStream(index); + + } +} + //get number of music channels active at this time, this does not mean they are playing int GetMusicStreamCount(void) { @@ -1045,18 +1110,18 @@ float GetMusicTimePlayed(int index) { uint64_t samples; jar_xm_get_position(musicStreams[index].xmctx, NULL, NULL, NULL, &samples); - secondsPlayed = (float)samples / (48000.f * musicStreams[index].mixc->channels); // Not sure if this is the correct value + secondsPlayed = (float)samples/(48000.0f*musicStreams[index].mixc->channels); // Not sure if this is the correct value } else if(musicStreams[index].chipTune && musicStreams[index].modctx.mod_loaded) { long numsamp = jar_mod_current_samples(&musicStreams[index].modctx); - secondsPlayed = (float)numsamp / (48000.f); + secondsPlayed = (float)numsamp/(48000.0f); } else { - int totalSamples = stb_vorbis_stream_length_in_samples(musicStreams[index].stream) * musicStreams[index].mixc->channels; + int totalSamples = stb_vorbis_stream_length_in_samples(musicStreams[index].stream)*musicStreams[index].mixc->channels; int samplesPlayed = totalSamples - musicStreams[index].totalSamplesLeft; - secondsPlayed = (float)samplesPlayed / (musicStreams[index].mixc->sampleRate * musicStreams[index].mixc->channels); + secondsPlayed = (float)samplesPlayed/(musicStreams[index].mixc->sampleRate*musicStreams[index].mixc->channels); } } @@ -1144,53 +1209,6 @@ static void EmptyMusicStream(int index) } } -// Determine if a music stream is ready to be written -static int IsMusicStreamReadyForBuffering(int index) -{ - ALint processed = 0; - alGetSourcei(musicStreams[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); - return processed; -} - -// Update (re-fill) music buffers if data already processed -void UpdateMusicStream(int index) -{ - ALenum state; - bool active = true; - int numBuffers = IsMusicStreamReadyForBuffering(index); - - if (musicStreams[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicStreams[index].enabled && musicStreams[index].mixc && numBuffers) - { - active = BufferMusicStream(index, numBuffers); - - if (!active && musicStreams[index].loop) - { - if (musicStreams[index].chipTune) - { - if(musicStreams[index].modctx.mod_loaded) jar_mod_seek_start(&musicStreams[index].modctx); - - musicStreams[index].totalSamplesLeft = musicStreams[index].totalLengthSeconds * 48000.f; - } - else - { - stb_vorbis_seek_start(musicStreams[index].stream); - musicStreams[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicStreams[index].stream) * musicStreams[index].mixc->channels; - } - - active = BufferMusicStream(index, IsMusicStreamReadyForBuffering(index)); - } - - if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); - - alGetSourcei(musicStreams[index].mixc->alSource, AL_SOURCE_STATE, &state); - - if (state != AL_PLAYING && active) alSourcePlay(musicStreams[index].mixc->alSource); - - if (!active) StopMusicStream(index); - - } -} - // Load WAV file into Wave structure static Wave LoadWAV(const char *fileName) { diff --git a/src/audio.h b/src/audio.h index 3ffe575c4..f4a82a550 100644 --- a/src/audio.h +++ b/src/audio.h @@ -2,13 +2,26 @@ * * raylib.audio * -* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles +* Basic functions to manage Audio: +* Manage audio device (init/close) +* Load and Unload audio files +* Play/Stop/Pause/Resume loaded audio +* Manage mixing channels +* Manage raw audio context * * Uses external lib: * OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html) * stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/) +* jar_xm - XM module file loading +* jar_mod - MOD audio file loading * -* Copyright (c) 2015 Ramon Santamaria (@raysan5) +* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions: +* XM audio module support (jar_xm) +* MOD audio module support (jar_mod) +* Mixing channels support +* Raw audio context support +* +* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5) * * This software is provided "as-is", without any express or implied warranty. In no event * will the authors be held liable for any damages arising from the use of this software. @@ -63,9 +76,6 @@ typedef struct Wave { short channels; } Wave; -typedef int RawAudioContext; - - #ifdef __cplusplus extern "C" { // Prevents name mangling of functions #endif @@ -80,7 +90,7 @@ extern "C" { // Prevents name mangling of functions //---------------------------------------------------------------------------------- void InitAudioDevice(void); // Initialize audio device and context void CloseAudioDevice(void); // Close the audio device and context (and music stream) -bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet +bool IsAudioDeviceReady(void); // Check if device has been initialized successfully Sound LoadSound(char *fileName); // Load sound to memory Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data @@ -105,12 +115,9 @@ float GetMusicTimeLength(int index); // Get music tim float GetMusicTimePlayed(int index); // Get current music time played (in seconds) int GetMusicStreamCount(void); // Get number of streams loaded -// used to output raw audio streams, returns negative numbers on error -// if floating point is false the data size is 16bit short, otherwise it is float 32bit -RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); - -void CloseRawAudioContext(RawAudioContext ctx); -int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements); // returns number of elements buffered +int InitRawMixChannel(int sampleRate, int channels, bool floatingPoint); // Initialize raw audio mix channel for audio buffering +int BufferRawMixChannel(int mixc, void *data, unsigned short numberElements); // Buffers data directly to raw mix channel +void CloseRawMixChannel(int mixc); // Closes and frees raw mix channel #ifdef __cplusplus } diff --git a/src/raylib.h b/src/raylib.h index 9225c5ee6..e3a17ebbc 100644 --- a/src/raylib.h +++ b/src/raylib.h @@ -468,8 +468,6 @@ typedef struct Wave { short channels; } Wave; -typedef int RawAudioContext; - // Texture formats // NOTE: Support depends on OpenGL version and platform typedef enum {