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Add new functions, update C sources

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This commit is contained in:
Milan Nikolic 2017-03-10 15:01:42 +01:00
parent 1aabddd935
commit 732563d5c2
19 changed files with 541 additions and 647 deletions
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69
raylib/audio.c
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@ -3,33 +3,51 @@
* raylib.audio
*
* This module provides basic functionality to work with audio:
* Manage audio device (init/close)
* Load and Unload audio files (WAV, OGG, FLAC, XM, MOD)
* Play/Stop/Pause/Resume loaded audio
* Manage mixing channels
* Manage raw audio context
* Manage audio device (init/close)
* Load and Unload audio files (WAV, OGG, FLAC, XM, MOD)
* Play/Stop/Pause/Resume loaded audio
* Manage mixing channels
* Manage raw audio context
*
* External libs:
* NOTES:
*
* Only up to two channels supported: MONO and STEREO (for additional channels, use AL_EXT_MCFORMATS)
* Only the following sample sizes supported: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32)
*
* CONFIGURATION:
*
* #define AUDIO_STANDALONE
* If defined, the module can be used as standalone library (independently of raylib).
* Required types and functions are defined in the same module.
*
* #define SUPPORT_FILEFORMAT_WAV / SUPPORT_LOAD_WAV / ENABLE_LOAD_WAV
* #define SUPPORT_FILEFORMAT_OGG
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* #define SUPPORT_FILEFORMAT_FLAC
* Selected desired fileformats to be supported for loading. Some of those formats are
* supported by default, to remove support, just comment unrequired #define in this module
*
* #define SUPPORT_RAW_AUDIO_BUFFERS
*
* DEPENDENCIES:
* OpenAL Soft - Audio device management (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
* dr_flac - FLAC audio file loading
*
* Module Configuration Flags:
* AUDIO_STANDALONE - Use this module as standalone library (independently of raylib)
*
* Some design decisions:
* Support only up to two channels: MONO and STEREO (for additional channels, AL_EXT_MCFORMATS)
* Support only the following sample sizes: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32)
* CONTRIBUTORS:
*
* 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
* XM audio module support (jar_xm)
* MOD audio module support (jar_mod)
* Mixing channels support
* Raw audio context support
*
*
* LICENSE: zlib/libpng
*
* Copyright (c) 2014-2016 Ramon Santamaria (@raysan5)
*
* This software is provided "as-is", without any express or implied warranty. In no event
@ -246,11 +264,11 @@ Wave LoadWave(const char *fileName)
else if (strcmp(GetExtension(fileName), "flac") == 0) wave = LoadFLAC(fileName);
else if (strcmp(GetExtension(fileName),"rres") == 0)
{
RRESData rres = LoadResource(fileName);
RRES rres = LoadResource(fileName, 0);
// NOTE: Parameters for RRES_WAVE type are: sampleCount, sampleRate, sampleSize, channels
// NOTE: Parameters for RRES_TYPE_WAVE are: sampleCount, sampleRate, sampleSize, channels
if (rres.type == RRES_WAVE) wave = LoadWaveEx(rres.data, rres.param1, rres.param2, rres.param3, rres.param4);
if (rres[0].type == RRES_TYPE_WAVE) wave = LoadWaveEx(rres[0].data, rres[0].param1, rres[0].param2, rres[0].param3, rres[0].param4);
else TraceLog(WARNING, "[%s] Resource file does not contain wave data", fileName);
UnloadResource(rres);
@ -586,7 +604,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
void *data = malloc(sampleCount*wave->channels*wave->sampleSize/8);
memcpy(data, wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
memcpy(data, (unsigned char*)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
free(wave->data);
wave->data = data;
@ -744,7 +762,18 @@ void ResumeMusicStream(Music music)
void StopMusicStream(Music music)
{
alSourceStop(music->stream.source);
// Clear stream buffers
void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1);
for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
{
alBufferData(music->stream.buffers[i], music->stream.format, pcm, AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, music->stream.sampleRate);
}
free(pcm);
// Restart music context
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;