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From j...@apache.org
Subject [10/83] [abbrv] [partial] incubator-corinthia git commit: removed SDL2
Date Mon, 23 Mar 2015 16:18:58 GMT
http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_mixer.c
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_mixer.c b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_mixer.c
deleted file mode 100644
index 42a1c68..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_mixer.c
+++ /dev/null
@@ -1,321 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../SDL_internal.h"
-
-/* This provides the default mixing callback for the SDL audio routines */
-
-#include "SDL_cpuinfo.h"
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "SDL_sysaudio.h"
-
-/* This table is used to add two sound values together and pin
- * the value to avoid overflow.  (used with permission from ARDI)
- * Changed to use 0xFE instead of 0xFF for better sound quality.
- */
-static const Uint8 mix8[] = {
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
-    0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
-    0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
-    0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
-    0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
-    0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
-    0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
-    0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
-    0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
-    0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
-    0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
-    0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
-    0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
-    0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
-    0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
-    0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
-    0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
-    0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
-    0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
-    0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
-    0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
-    0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
-    0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
-    0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
-    0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE
-};
-
-/* The volume ranges from 0 - 128 */
-#define ADJUST_VOLUME(s, v) (s = (s*v)/SDL_MIX_MAXVOLUME)
-#define ADJUST_VOLUME_U8(s, v)  (s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
-
-
-void
-SDL_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
-                   Uint32 len, int volume)
-{
-    if (volume == 0) {
-        return;
-    }
-
-    switch (format) {
-
-    case AUDIO_U8:
-        {
-#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
-            SDL_MixAudio_m68k_U8((char *) dst, (char *) src,
-                                 (unsigned long) len, (long) volume,
-                                 (char *) mix8);
-#else
-            Uint8 src_sample;
-
-            while (len--) {
-                src_sample = *src;
-                ADJUST_VOLUME_U8(src_sample, volume);
-                *dst = mix8[*dst + src_sample];
-                ++dst;
-                ++src;
-            }
-#endif
-        }
-        break;
-
-    case AUDIO_S8:
-        {
-            Sint8 *dst8, *src8;
-            Sint8 src_sample;
-            int dst_sample;
-            const int max_audioval = ((1 << (8 - 1)) - 1);
-            const int min_audioval = -(1 << (8 - 1));
-
-            src8 = (Sint8 *) src;
-            dst8 = (Sint8 *) dst;
-            while (len--) {
-                src_sample = *src8;
-                ADJUST_VOLUME(src_sample, volume);
-                dst_sample = *dst8 + src_sample;
-                if (dst_sample > max_audioval) {
-                    *dst8 = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    *dst8 = min_audioval;
-                } else {
-                    *dst8 = dst_sample;
-                }
-                ++dst8;
-                ++src8;
-            }
-        }
-        break;
-
-    case AUDIO_S16LSB:
-        {
-            Sint16 src1, src2;
-            int dst_sample;
-            const int max_audioval = ((1 << (16 - 1)) - 1);
-            const int min_audioval = -(1 << (16 - 1));
-
-            len /= 2;
-            while (len--) {
-                src1 = ((src[1]) << 8 | src[0]);
-                ADJUST_VOLUME(src1, volume);
-                src2 = ((dst[1]) << 8 | dst[0]);
-                src += 2;
-                dst_sample = src1 + src2;
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                dst[0] = dst_sample & 0xFF;
-                dst_sample >>= 8;
-                dst[1] = dst_sample & 0xFF;
-                dst += 2;
-            }
-        }
-        break;
-
-    case AUDIO_S16MSB:
-        {
-#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
-            SDL_MixAudio_m68k_S16MSB((short *) dst, (short *) src,
-                                     (unsigned long) len, (long) volume);
-#else
-            Sint16 src1, src2;
-            int dst_sample;
-            const int max_audioval = ((1 << (16 - 1)) - 1);
-            const int min_audioval = -(1 << (16 - 1));
-
-            len /= 2;
-            while (len--) {
-                src1 = ((src[0]) << 8 | src[1]);
-                ADJUST_VOLUME(src1, volume);
-                src2 = ((dst[0]) << 8 | dst[1]);
-                src += 2;
-                dst_sample = src1 + src2;
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                dst[1] = dst_sample & 0xFF;
-                dst_sample >>= 8;
-                dst[0] = dst_sample & 0xFF;
-                dst += 2;
-            }
-#endif
-        }
-        break;
-
-    case AUDIO_S32LSB:
-        {
-            const Uint32 *src32 = (Uint32 *) src;
-            Uint32 *dst32 = (Uint32 *) dst;
-            Sint64 src1, src2;
-            Sint64 dst_sample;
-            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
-            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
-
-            len /= 4;
-            while (len--) {
-                src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
-                src32++;
-                ADJUST_VOLUME(src1, volume);
-                src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
-                dst_sample = src1 + src2;
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                *(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
-            }
-        }
-        break;
-
-    case AUDIO_S32MSB:
-        {
-            const Uint32 *src32 = (Uint32 *) src;
-            Uint32 *dst32 = (Uint32 *) dst;
-            Sint64 src1, src2;
-            Sint64 dst_sample;
-            const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
-            const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
-
-            len /= 4;
-            while (len--) {
-                src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
-                src32++;
-                ADJUST_VOLUME(src1, volume);
-                src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
-                dst_sample = src1 + src2;
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                *(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
-            }
-        }
-        break;
-
-    case AUDIO_F32LSB:
-        {
-            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
-            const float fvolume = (float) volume;
-            const float *src32 = (float *) src;
-            float *dst32 = (float *) dst;
-            float src1, src2;
-            double dst_sample;
-            /* !!! FIXME: are these right? */
-            const double max_audioval = 3.402823466e+38F;
-            const double min_audioval = -3.402823466e+38F;
-
-            len /= 4;
-            while (len--) {
-                src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
-                src2 = SDL_SwapFloatLE(*dst32);
-                src32++;
-
-                dst_sample = ((double) src1) + ((double) src2);
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                *(dst32++) = SDL_SwapFloatLE((float) dst_sample);
-            }
-        }
-        break;
-
-    case AUDIO_F32MSB:
-        {
-            const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
-            const float fvolume = (float) volume;
-            const float *src32 = (float *) src;
-            float *dst32 = (float *) dst;
-            float src1, src2;
-            double dst_sample;
-            /* !!! FIXME: are these right? */
-            const double max_audioval = 3.402823466e+38F;
-            const double min_audioval = -3.402823466e+38F;
-
-            len /= 4;
-            while (len--) {
-                src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
-                src2 = SDL_SwapFloatBE(*dst32);
-                src32++;
-
-                dst_sample = ((double) src1) + ((double) src2);
-                if (dst_sample > max_audioval) {
-                    dst_sample = max_audioval;
-                } else if (dst_sample < min_audioval) {
-                    dst_sample = min_audioval;
-                }
-                *(dst32++) = SDL_SwapFloatBE((float) dst_sample);
-            }
-        }
-        break;
-
-    default:                   /* If this happens... FIXME! */
-        SDL_SetError("SDL_MixAudio(): unknown audio format");
-        return;
-    }
-}
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_sysaudio.h
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_sysaudio.h b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_sysaudio.h
deleted file mode 100644
index 9fe31c8..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_sysaudio.h
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../SDL_internal.h"
-
-#ifndef _SDL_sysaudio_h
-#define _SDL_sysaudio_h
-
-#include "SDL_mutex.h"
-#include "SDL_thread.h"
-
-/* The SDL audio driver */
-typedef struct SDL_AudioDevice SDL_AudioDevice;
-#define _THIS   SDL_AudioDevice *_this
-
-/* Used by audio targets during DetectDevices() */
-typedef void (*SDL_AddAudioDevice)(const char *name);
-
-typedef struct SDL_AudioDriverImpl
-{
-    void (*DetectDevices) (int iscapture, SDL_AddAudioDevice addfn);
-    int (*OpenDevice) (_THIS, const char *devname, int iscapture);
-    void (*ThreadInit) (_THIS); /* Called by audio thread at start */
-    void (*WaitDevice) (_THIS);
-    void (*PlayDevice) (_THIS);
-    Uint8 *(*GetDeviceBuf) (_THIS);
-    void (*WaitDone) (_THIS);
-    void (*CloseDevice) (_THIS);
-    void (*LockDevice) (_THIS);
-    void (*UnlockDevice) (_THIS);
-    void (*Deinitialize) (void);
-
-    /* !!! FIXME: add pause(), so we can optimize instead of mixing silence. */
-
-    /* Some flags to push duplicate code into the core and reduce #ifdefs. */
-    int ProvidesOwnCallbackThread;
-    int SkipMixerLock;  /* !!! FIXME: do we need this anymore? */
-    int HasCaptureSupport;
-    int OnlyHasDefaultOutputDevice;
-    int OnlyHasDefaultInputDevice;
-} SDL_AudioDriverImpl;
-
-
-typedef struct SDL_AudioDriver
-{
-    /* * * */
-    /* The name of this audio driver */
-    const char *name;
-
-    /* * * */
-    /* The description of this audio driver */
-    const char *desc;
-
-    SDL_AudioDriverImpl impl;
-
-    char **outputDevices;
-    int outputDeviceCount;
-
-    char **inputDevices;
-    int inputDeviceCount;
-} SDL_AudioDriver;
-
-
-/* Streamer */
-typedef struct
-{
-    Uint8 *buffer;
-    int max_len;                /* the maximum length in bytes */
-    int read_pos, write_pos;    /* the position of the write and read heads in bytes */
-} SDL_AudioStreamer;
-
-
-/* Define the SDL audio driver structure */
-struct SDL_AudioDevice
-{
-    /* * * */
-    /* Data common to all devices */
-
-    /* The current audio specification (shared with audio thread) */
-    SDL_AudioSpec spec;
-
-    /* An audio conversion block for audio format emulation */
-    SDL_AudioCVT convert;
-
-    /* The streamer, if sample rate conversion necessitates it */
-    int use_streamer;
-    SDL_AudioStreamer streamer;
-
-    /* Current state flags */
-    int iscapture;
-    int enabled;
-    int paused;
-    int opened;
-
-    /* Fake audio buffer for when the audio hardware is busy */
-    Uint8 *fake_stream;
-
-    /* A semaphore for locking the mixing buffers */
-    SDL_mutex *mixer_lock;
-
-    /* A thread to feed the audio device */
-    SDL_Thread *thread;
-    SDL_threadID threadid;
-
-    /* * * */
-    /* Data private to this driver */
-    struct SDL_PrivateAudioData *hidden;
-};
-#undef _THIS
-
-typedef struct AudioBootStrap
-{
-    const char *name;
-    const char *desc;
-    int (*init) (SDL_AudioDriverImpl * impl);
-    int demand_only;  /* 1==request explicitly, or it won't be available. */
-} AudioBootStrap;
-
-#endif /* _SDL_sysaudio_h */
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.c
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.c b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.c
deleted file mode 100644
index 903264c..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.c
+++ /dev/null
@@ -1,623 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../SDL_internal.h"
-
-/* Microsoft WAVE file loading routines */
-
-#include "SDL_audio.h"
-#include "SDL_wave.h"
-
-
-static int ReadChunk(SDL_RWops * src, Chunk * chunk);
-
-struct MS_ADPCM_decodestate
-{
-    Uint8 hPredictor;
-    Uint16 iDelta;
-    Sint16 iSamp1;
-    Sint16 iSamp2;
-};
-static struct MS_ADPCM_decoder
-{
-    WaveFMT wavefmt;
-    Uint16 wSamplesPerBlock;
-    Uint16 wNumCoef;
-    Sint16 aCoeff[7][2];
-    /* * * */
-    struct MS_ADPCM_decodestate state[2];
-} MS_ADPCM_state;
-
-static int
-InitMS_ADPCM(WaveFMT * format)
-{
-    Uint8 *rogue_feel;
-    int i;
-
-    /* Set the rogue pointer to the MS_ADPCM specific data */
-    MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
-    MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
-    MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
-    MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
-    MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
-    MS_ADPCM_state.wavefmt.bitspersample =
-        SDL_SwapLE16(format->bitspersample);
-    rogue_feel = (Uint8 *) format + sizeof(*format);
-    if (sizeof(*format) == 16) {
-        /* const Uint16 extra_info = ((rogue_feel[1] << 8) | rogue_feel[0]); */
-        rogue_feel += sizeof(Uint16);
-    }
-    MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) | rogue_feel[0]);
-    rogue_feel += sizeof(Uint16);
-    MS_ADPCM_state.wNumCoef = ((rogue_feel[1] << 8) | rogue_feel[0]);
-    rogue_feel += sizeof(Uint16);
-    if (MS_ADPCM_state.wNumCoef != 7) {
-        SDL_SetError("Unknown set of MS_ADPCM coefficients");
-        return (-1);
-    }
-    for (i = 0; i < MS_ADPCM_state.wNumCoef; ++i) {
-        MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1] << 8) | rogue_feel[0]);
-        rogue_feel += sizeof(Uint16);
-        MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1] << 8) | rogue_feel[0]);
-        rogue_feel += sizeof(Uint16);
-    }
-    return (0);
-}
-
-static Sint32
-MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
-                Uint8 nybble, Sint16 * coeff)
-{
-    const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
-    const Sint32 min_audioval = -(1 << (16 - 1));
-    const Sint32 adaptive[] = {
-        230, 230, 230, 230, 307, 409, 512, 614,
-        768, 614, 512, 409, 307, 230, 230, 230
-    };
-    Sint32 new_sample, delta;
-
-    new_sample = ((state->iSamp1 * coeff[0]) +
-                  (state->iSamp2 * coeff[1])) / 256;
-    if (nybble & 0x08) {
-        new_sample += state->iDelta * (nybble - 0x10);
-    } else {
-        new_sample += state->iDelta * nybble;
-    }
-    if (new_sample < min_audioval) {
-        new_sample = min_audioval;
-    } else if (new_sample > max_audioval) {
-        new_sample = max_audioval;
-    }
-    delta = ((Sint32) state->iDelta * adaptive[nybble]) / 256;
-    if (delta < 16) {
-        delta = 16;
-    }
-    state->iDelta = (Uint16) delta;
-    state->iSamp2 = state->iSamp1;
-    state->iSamp1 = (Sint16) new_sample;
-    return (new_sample);
-}
-
-static int
-MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
-{
-    struct MS_ADPCM_decodestate *state[2];
-    Uint8 *freeable, *encoded, *decoded;
-    Sint32 encoded_len, samplesleft;
-    Sint8 nybble, stereo;
-    Sint16 *coeff[2];
-    Sint32 new_sample;
-
-    /* Allocate the proper sized output buffer */
-    encoded_len = *audio_len;
-    encoded = *audio_buf;
-    freeable = *audio_buf;
-    *audio_len = (encoded_len / MS_ADPCM_state.wavefmt.blockalign) *
-        MS_ADPCM_state.wSamplesPerBlock *
-        MS_ADPCM_state.wavefmt.channels * sizeof(Sint16);
-    *audio_buf = (Uint8 *) SDL_malloc(*audio_len);
-    if (*audio_buf == NULL) {
-        return SDL_OutOfMemory();
-    }
-    decoded = *audio_buf;
-
-    /* Get ready... Go! */
-    stereo = (MS_ADPCM_state.wavefmt.channels == 2);
-    state[0] = &MS_ADPCM_state.state[0];
-    state[1] = &MS_ADPCM_state.state[stereo];
-    while (encoded_len >= MS_ADPCM_state.wavefmt.blockalign) {
-        /* Grab the initial information for this block */
-        state[0]->hPredictor = *encoded++;
-        if (stereo) {
-            state[1]->hPredictor = *encoded++;
-        }
-        state[0]->iDelta = ((encoded[1] << 8) | encoded[0]);
-        encoded += sizeof(Sint16);
-        if (stereo) {
-            state[1]->iDelta = ((encoded[1] << 8) | encoded[0]);
-            encoded += sizeof(Sint16);
-        }
-        state[0]->iSamp1 = ((encoded[1] << 8) | encoded[0]);
-        encoded += sizeof(Sint16);
-        if (stereo) {
-            state[1]->iSamp1 = ((encoded[1] << 8) | encoded[0]);
-            encoded += sizeof(Sint16);
-        }
-        state[0]->iSamp2 = ((encoded[1] << 8) | encoded[0]);
-        encoded += sizeof(Sint16);
-        if (stereo) {
-            state[1]->iSamp2 = ((encoded[1] << 8) | encoded[0]);
-            encoded += sizeof(Sint16);
-        }
-        coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
-        coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
-
-        /* Store the two initial samples we start with */
-        decoded[0] = state[0]->iSamp2 & 0xFF;
-        decoded[1] = state[0]->iSamp2 >> 8;
-        decoded += 2;
-        if (stereo) {
-            decoded[0] = state[1]->iSamp2 & 0xFF;
-            decoded[1] = state[1]->iSamp2 >> 8;
-            decoded += 2;
-        }
-        decoded[0] = state[0]->iSamp1 & 0xFF;
-        decoded[1] = state[0]->iSamp1 >> 8;
-        decoded += 2;
-        if (stereo) {
-            decoded[0] = state[1]->iSamp1 & 0xFF;
-            decoded[1] = state[1]->iSamp1 >> 8;
-            decoded += 2;
-        }
-
-        /* Decode and store the other samples in this block */
-        samplesleft = (MS_ADPCM_state.wSamplesPerBlock - 2) *
-            MS_ADPCM_state.wavefmt.channels;
-        while (samplesleft > 0) {
-            nybble = (*encoded) >> 4;
-            new_sample = MS_ADPCM_nibble(state[0], nybble, coeff[0]);
-            decoded[0] = new_sample & 0xFF;
-            new_sample >>= 8;
-            decoded[1] = new_sample & 0xFF;
-            decoded += 2;
-
-            nybble = (*encoded) & 0x0F;
-            new_sample = MS_ADPCM_nibble(state[1], nybble, coeff[1]);
-            decoded[0] = new_sample & 0xFF;
-            new_sample >>= 8;
-            decoded[1] = new_sample & 0xFF;
-            decoded += 2;
-
-            ++encoded;
-            samplesleft -= 2;
-        }
-        encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
-    }
-    SDL_free(freeable);
-    return (0);
-}
-
-struct IMA_ADPCM_decodestate
-{
-    Sint32 sample;
-    Sint8 index;
-};
-static struct IMA_ADPCM_decoder
-{
-    WaveFMT wavefmt;
-    Uint16 wSamplesPerBlock;
-    /* * * */
-    struct IMA_ADPCM_decodestate state[2];
-} IMA_ADPCM_state;
-
-static int
-InitIMA_ADPCM(WaveFMT * format)
-{
-    Uint8 *rogue_feel;
-
-    /* Set the rogue pointer to the IMA_ADPCM specific data */
-    IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
-    IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
-    IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
-    IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
-    IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
-    IMA_ADPCM_state.wavefmt.bitspersample =
-        SDL_SwapLE16(format->bitspersample);
-    rogue_feel = (Uint8 *) format + sizeof(*format);
-    if (sizeof(*format) == 16) {
-        /* const Uint16 extra_info = ((rogue_feel[1] << 8) | rogue_feel[0]); */
-        rogue_feel += sizeof(Uint16);
-    }
-    IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) | rogue_feel[0]);
-    return (0);
-}
-
-static Sint32
-IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
-{
-    const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
-    const Sint32 min_audioval = -(1 << (16 - 1));
-    const int index_table[16] = {
-        -1, -1, -1, -1,
-        2, 4, 6, 8,
-        -1, -1, -1, -1,
-        2, 4, 6, 8
-    };
-    const Sint32 step_table[89] = {
-        7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,
-        34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,
-        143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,
-        449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
-        1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
-        3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
-        9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
-        22385, 24623, 27086, 29794, 32767
-    };
-    Sint32 delta, step;
-
-    /* Compute difference and new sample value */
-    if (state->index > 88) {
-        state->index = 88;
-    } else if (state->index < 0) {
-        state->index = 0;
-    }
-    step = step_table[state->index];
-    delta = step >> 3;
-    if (nybble & 0x04)
-        delta += step;
-    if (nybble & 0x02)
-        delta += (step >> 1);
-    if (nybble & 0x01)
-        delta += (step >> 2);
-    if (nybble & 0x08)
-        delta = -delta;
-    state->sample += delta;
-
-    /* Update index value */
-    state->index += index_table[nybble];
-
-    /* Clamp output sample */
-    if (state->sample > max_audioval) {
-        state->sample = max_audioval;
-    } else if (state->sample < min_audioval) {
-        state->sample = min_audioval;
-    }
-    return (state->sample);
-}
-
-/* Fill the decode buffer with a channel block of data (8 samples) */
-static void
-Fill_IMA_ADPCM_block(Uint8 * decoded, Uint8 * encoded,
-                     int channel, int numchannels,
-                     struct IMA_ADPCM_decodestate *state)
-{
-    int i;
-    Sint8 nybble;
-    Sint32 new_sample;
-
-    decoded += (channel * 2);
-    for (i = 0; i < 4; ++i) {
-        nybble = (*encoded) & 0x0F;
-        new_sample = IMA_ADPCM_nibble(state, nybble);
-        decoded[0] = new_sample & 0xFF;
-        new_sample >>= 8;
-        decoded[1] = new_sample & 0xFF;
-        decoded += 2 * numchannels;
-
-        nybble = (*encoded) >> 4;
-        new_sample = IMA_ADPCM_nibble(state, nybble);
-        decoded[0] = new_sample & 0xFF;
-        new_sample >>= 8;
-        decoded[1] = new_sample & 0xFF;
-        decoded += 2 * numchannels;
-
-        ++encoded;
-    }
-}
-
-static int
-IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
-{
-    struct IMA_ADPCM_decodestate *state;
-    Uint8 *freeable, *encoded, *decoded;
-    Sint32 encoded_len, samplesleft;
-    unsigned int c, channels;
-
-    /* Check to make sure we have enough variables in the state array */
-    channels = IMA_ADPCM_state.wavefmt.channels;
-    if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
-        SDL_SetError("IMA ADPCM decoder can only handle %d channels",
-                     SDL_arraysize(IMA_ADPCM_state.state));
-        return (-1);
-    }
-    state = IMA_ADPCM_state.state;
-
-    /* Allocate the proper sized output buffer */
-    encoded_len = *audio_len;
-    encoded = *audio_buf;
-    freeable = *audio_buf;
-    *audio_len = (encoded_len / IMA_ADPCM_state.wavefmt.blockalign) *
-        IMA_ADPCM_state.wSamplesPerBlock *
-        IMA_ADPCM_state.wavefmt.channels * sizeof(Sint16);
-    *audio_buf = (Uint8 *) SDL_malloc(*audio_len);
-    if (*audio_buf == NULL) {
-        return SDL_OutOfMemory();
-    }
-    decoded = *audio_buf;
-
-    /* Get ready... Go! */
-    while (encoded_len >= IMA_ADPCM_state.wavefmt.blockalign) {
-        /* Grab the initial information for this block */
-        for (c = 0; c < channels; ++c) {
-            /* Fill the state information for this block */
-            state[c].sample = ((encoded[1] << 8) | encoded[0]);
-            encoded += 2;
-            if (state[c].sample & 0x8000) {
-                state[c].sample -= 0x10000;
-            }
-            state[c].index = *encoded++;
-            /* Reserved byte in buffer header, should be 0 */
-            if (*encoded++ != 0) {
-                /* Uh oh, corrupt data?  Buggy code? */ ;
-            }
-
-            /* Store the initial sample we start with */
-            decoded[0] = (Uint8) (state[c].sample & 0xFF);
-            decoded[1] = (Uint8) (state[c].sample >> 8);
-            decoded += 2;
-        }
-
-        /* Decode and store the other samples in this block */
-        samplesleft = (IMA_ADPCM_state.wSamplesPerBlock - 1) * channels;
-        while (samplesleft > 0) {
-            for (c = 0; c < channels; ++c) {
-                Fill_IMA_ADPCM_block(decoded, encoded,
-                                     c, channels, &state[c]);
-                encoded += 4;
-                samplesleft -= 8;
-            }
-            decoded += (channels * 8 * 2);
-        }
-        encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
-    }
-    SDL_free(freeable);
-    return (0);
-}
-
-SDL_AudioSpec *
-SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
-               SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
-{
-    int was_error;
-    Chunk chunk;
-    int lenread;
-    int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
-    int samplesize;
-
-    /* WAV magic header */
-    Uint32 RIFFchunk;
-    Uint32 wavelen = 0;
-    Uint32 WAVEmagic;
-    Uint32 headerDiff = 0;
-
-    /* FMT chunk */
-    WaveFMT *format = NULL;
-
-    SDL_zero(chunk);
-
-    /* Make sure we are passed a valid data source */
-    was_error = 0;
-    if (src == NULL) {
-        was_error = 1;
-        goto done;
-    }
-
-    /* Check the magic header */
-    RIFFchunk = SDL_ReadLE32(src);
-    wavelen = SDL_ReadLE32(src);
-    if (wavelen == WAVE) {      /* The RIFFchunk has already been read */
-        WAVEmagic = wavelen;
-        wavelen = RIFFchunk;
-        RIFFchunk = RIFF;
-    } else {
-        WAVEmagic = SDL_ReadLE32(src);
-    }
-    if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
-        SDL_SetError("Unrecognized file type (not WAVE)");
-        was_error = 1;
-        goto done;
-    }
-    headerDiff += sizeof(Uint32);       /* for WAVE */
-
-    /* Read the audio data format chunk */
-    chunk.data = NULL;
-    do {
-        SDL_free(chunk.data);
-        chunk.data = NULL;
-        lenread = ReadChunk(src, &chunk);
-        if (lenread < 0) {
-            was_error = 1;
-            goto done;
-        }
-        /* 2 Uint32's for chunk header+len, plus the lenread */
-        headerDiff += lenread + 2 * sizeof(Uint32);
-    } while ((chunk.magic == FACT) || (chunk.magic == LIST));
-
-    /* Decode the audio data format */
-    format = (WaveFMT *) chunk.data;
-    if (chunk.magic != FMT) {
-        SDL_SetError("Complex WAVE files not supported");
-        was_error = 1;
-        goto done;
-    }
-    IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
-    switch (SDL_SwapLE16(format->encoding)) {
-    case PCM_CODE:
-        /* We can understand this */
-        break;
-    case IEEE_FLOAT_CODE:
-        IEEE_float_encoded = 1;
-        /* We can understand this */
-        break;
-    case MS_ADPCM_CODE:
-        /* Try to understand this */
-        if (InitMS_ADPCM(format) < 0) {
-            was_error = 1;
-            goto done;
-        }
-        MS_ADPCM_encoded = 1;
-        break;
-    case IMA_ADPCM_CODE:
-        /* Try to understand this */
-        if (InitIMA_ADPCM(format) < 0) {
-            was_error = 1;
-            goto done;
-        }
-        IMA_ADPCM_encoded = 1;
-        break;
-    case MP3_CODE:
-        SDL_SetError("MPEG Layer 3 data not supported",
-                     SDL_SwapLE16(format->encoding));
-        was_error = 1;
-        goto done;
-    default:
-        SDL_SetError("Unknown WAVE data format: 0x%.4x",
-                     SDL_SwapLE16(format->encoding));
-        was_error = 1;
-        goto done;
-    }
-    SDL_memset(spec, 0, (sizeof *spec));
-    spec->freq = SDL_SwapLE32(format->frequency);
-
-    if (IEEE_float_encoded) {
-        if ((SDL_SwapLE16(format->bitspersample)) != 32) {
-            was_error = 1;
-        } else {
-            spec->format = AUDIO_F32;
-        }
-    } else {
-        switch (SDL_SwapLE16(format->bitspersample)) {
-        case 4:
-            if (MS_ADPCM_encoded || IMA_ADPCM_encoded) {
-                spec->format = AUDIO_S16;
-            } else {
-                was_error = 1;
-            }
-            break;
-        case 8:
-            spec->format = AUDIO_U8;
-            break;
-        case 16:
-            spec->format = AUDIO_S16;
-            break;
-        case 32:
-            spec->format = AUDIO_S32;
-            break;
-        default:
-            was_error = 1;
-            break;
-        }
-    }
-
-    if (was_error) {
-        SDL_SetError("Unknown %d-bit PCM data format",
-                     SDL_SwapLE16(format->bitspersample));
-        goto done;
-    }
-    spec->channels = (Uint8) SDL_SwapLE16(format->channels);
-    spec->samples = 4096;       /* Good default buffer size */
-
-    /* Read the audio data chunk */
-    *audio_buf = NULL;
-    do {
-        SDL_free(*audio_buf);
-        *audio_buf = NULL;
-        lenread = ReadChunk(src, &chunk);
-        if (lenread < 0) {
-            was_error = 1;
-            goto done;
-        }
-        *audio_len = lenread;
-        *audio_buf = chunk.data;
-        if (chunk.magic != DATA)
-            headerDiff += lenread + 2 * sizeof(Uint32);
-    } while (chunk.magic != DATA);
-    headerDiff += 2 * sizeof(Uint32);   /* for the data chunk and len */
-
-    if (MS_ADPCM_encoded) {
-        if (MS_ADPCM_decode(audio_buf, audio_len) < 0) {
-            was_error = 1;
-            goto done;
-        }
-    }
-    if (IMA_ADPCM_encoded) {
-        if (IMA_ADPCM_decode(audio_buf, audio_len) < 0) {
-            was_error = 1;
-            goto done;
-        }
-    }
-
-    /* Don't return a buffer that isn't a multiple of samplesize */
-    samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
-    *audio_len &= ~(samplesize - 1);
-
-  done:
-    SDL_free(format);
-    if (src) {
-        if (freesrc) {
-            SDL_RWclose(src);
-        } else {
-            /* seek to the end of the file (given by the RIFF chunk) */
-            SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
-        }
-    }
-    if (was_error) {
-        spec = NULL;
-    }
-    return (spec);
-}
-
-/* Since the WAV memory is allocated in the shared library, it must also
-   be freed here.  (Necessary under Win32, VC++)
- */
-void
-SDL_FreeWAV(Uint8 * audio_buf)
-{
-    SDL_free(audio_buf);
-}
-
-static int
-ReadChunk(SDL_RWops * src, Chunk * chunk)
-{
-    chunk->magic = SDL_ReadLE32(src);
-    chunk->length = SDL_ReadLE32(src);
-    chunk->data = (Uint8 *) SDL_malloc(chunk->length);
-    if (chunk->data == NULL) {
-        return SDL_OutOfMemory();
-    }
-    if (SDL_RWread(src, chunk->data, chunk->length, 1) != 1) {
-        SDL_free(chunk->data);
-        chunk->data = NULL;
-        return SDL_Error(SDL_EFREAD);
-    }
-    return (chunk->length);
-}
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.h
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.h b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.h
deleted file mode 100644
index c53ad59..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/SDL_wave.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../SDL_internal.h"
-
-/* WAVE files are little-endian */
-
-/*******************************************/
-/* Define values for Microsoft WAVE format */
-/*******************************************/
-#define RIFF            0x46464952      /* "RIFF" */
-#define WAVE            0x45564157      /* "WAVE" */
-#define FACT            0x74636166      /* "fact" */
-#define LIST            0x5453494c      /* "LIST" */
-#define FMT             0x20746D66      /* "fmt " */
-#define DATA            0x61746164      /* "data" */
-#define PCM_CODE        0x0001
-#define MS_ADPCM_CODE   0x0002
-#define IEEE_FLOAT_CODE 0x0003
-#define IMA_ADPCM_CODE  0x0011
-#define MP3_CODE        0x0055
-#define WAVE_MONO       1
-#define WAVE_STEREO     2
-
-/* Normally, these three chunks come consecutively in a WAVE file */
-typedef struct WaveFMT
-{
-/* Not saved in the chunk we read:
-    Uint32  FMTchunk;
-    Uint32  fmtlen;
-*/
-    Uint16 encoding;
-    Uint16 channels;            /* 1 = mono, 2 = stereo */
-    Uint32 frequency;           /* One of 11025, 22050, or 44100 Hz */
-    Uint32 byterate;            /* Average bytes per second */
-    Uint16 blockalign;          /* Bytes per sample block */
-    Uint16 bitspersample;       /* One of 8, 12, 16, or 4 for ADPCM */
-} WaveFMT;
-
-/* The general chunk found in the WAVE file */
-typedef struct Chunk
-{
-    Uint32 magic;
-    Uint32 length;
-    Uint8 *data;
-} Chunk;
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.c
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.c b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.c
deleted file mode 100644
index 1f3def3..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.c
+++ /dev/null
@@ -1,685 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#if SDL_AUDIO_DRIVER_ALSA
-
-/* Allow access to a raw mixing buffer */
-
-#include <sys/types.h>
-#include <signal.h>             /* For kill() */
-#include <errno.h>
-#include <string.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_alsa_audio.h"
-
-#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
-#include "SDL_loadso.h"
-#endif
-
-static int (*ALSA_snd_pcm_open)
-  (snd_pcm_t **, const char *, snd_pcm_stream_t, int);
-static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
-static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
-  (snd_pcm_t *, const void *, snd_pcm_uframes_t);
-static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
-static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
-static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
-static const char *(*ALSA_snd_strerror) (int);
-static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void);
-static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void);
-static void (*ALSA_snd_pcm_hw_params_copy)
-  (snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
-static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *);
-static int (*ALSA_snd_pcm_hw_params_set_access)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
-static int (*ALSA_snd_pcm_hw_params_set_format)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
-static int (*ALSA_snd_pcm_hw_params_set_channels)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
-static int (*ALSA_snd_pcm_hw_params_get_channels)
-  (const snd_pcm_hw_params_t *, unsigned int *);
-static int (*ALSA_snd_pcm_hw_params_set_rate_near)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
-static int (*ALSA_snd_pcm_hw_params_set_period_size_near)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
-static int (*ALSA_snd_pcm_hw_params_get_period_size)
-  (const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
-static int (*ALSA_snd_pcm_hw_params_set_periods_near)
-  (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
-static int (*ALSA_snd_pcm_hw_params_get_periods)
-  (const snd_pcm_hw_params_t *, unsigned int *, int *);
-static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)
-  (snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
-static int (*ALSA_snd_pcm_hw_params_get_buffer_size)
-  (const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
-static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *);
-static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *,
-                                              snd_pcm_sw_params_t *);
-static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
-  (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
-static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *);
-static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
-static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
-static int (*ALSA_snd_pcm_sw_params_set_avail_min)
-  (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
-
-#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
-#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
-#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof
-
-static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
-static void *alsa_handle = NULL;
-
-static int
-load_alsa_sym(const char *fn, void **addr)
-{
-    *addr = SDL_LoadFunction(alsa_handle, fn);
-    if (*addr == NULL) {
-        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
-        return 0;
-    }
-
-    return 1;
-}
-
-/* cast funcs to char* first, to please GCC's strict aliasing rules. */
-#define SDL_ALSA_SYM(x) \
-    if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
-#else
-#define SDL_ALSA_SYM(x) ALSA_##x = x
-#endif
-
-static int
-load_alsa_syms(void)
-{
-    SDL_ALSA_SYM(snd_pcm_open);
-    SDL_ALSA_SYM(snd_pcm_close);
-    SDL_ALSA_SYM(snd_pcm_writei);
-    SDL_ALSA_SYM(snd_pcm_recover);
-    SDL_ALSA_SYM(snd_pcm_prepare);
-    SDL_ALSA_SYM(snd_pcm_drain);
-    SDL_ALSA_SYM(snd_strerror);
-    SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
-    SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
-    SDL_ALSA_SYM(snd_pcm_hw_params_copy);
-    SDL_ALSA_SYM(snd_pcm_hw_params_any);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
-    SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
-    SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);
-    SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
-    SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);
-    SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);
-    SDL_ALSA_SYM(snd_pcm_hw_params);
-    SDL_ALSA_SYM(snd_pcm_sw_params_current);
-    SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
-    SDL_ALSA_SYM(snd_pcm_sw_params);
-    SDL_ALSA_SYM(snd_pcm_nonblock);
-    SDL_ALSA_SYM(snd_pcm_wait);
-    SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
-    return 0;
-}
-
-#undef SDL_ALSA_SYM
-
-#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
-
-static void
-UnloadALSALibrary(void)
-{
-    if (alsa_handle != NULL) {
-        SDL_UnloadObject(alsa_handle);
-        alsa_handle = NULL;
-    }
-}
-
-static int
-LoadALSALibrary(void)
-{
-    int retval = 0;
-    if (alsa_handle == NULL) {
-        alsa_handle = SDL_LoadObject(alsa_library);
-        if (alsa_handle == NULL) {
-            retval = -1;
-            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
-        } else {
-            retval = load_alsa_syms();
-            if (retval < 0) {
-                UnloadALSALibrary();
-            }
-        }
-    }
-    return retval;
-}
-
-#else
-
-static void
-UnloadALSALibrary(void)
-{
-}
-
-static int
-LoadALSALibrary(void)
-{
-    load_alsa_syms();
-    return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
-
-static const char *
-get_audio_device(int channels)
-{
-    const char *device;
-
-    device = SDL_getenv("AUDIODEV");    /* Is there a standard variable name? */
-    if (device == NULL) {
-        switch (channels) {
-        case 6:
-            device = "plug:surround51";
-            break;
-        case 4:
-            device = "plug:surround40";
-            break;
-        default:
-            device = "default";
-            break;
-        }
-    }
-    return device;
-}
-
-
-/* This function waits until it is possible to write a full sound buffer */
-static void
-ALSA_WaitDevice(_THIS)
-{
-    /* We're in blocking mode, so there's nothing to do here */
-}
-
-
-/* !!! FIXME: is there a channel swizzler in alsalib instead? */
-/*
- * http://bugzilla.libsdl.org/show_bug.cgi?id=110
- * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
- *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
- */
-#define SWIZ6(T) \
-    T *ptr = (T *) this->hidden->mixbuf; \
-    Uint32 i; \
-    for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
-        T tmp; \
-        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
-        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
-    }
-
-static SDL_INLINE void
-swizzle_alsa_channels_6_64bit(_THIS)
-{
-    SWIZ6(Uint64);
-}
-
-static SDL_INLINE void
-swizzle_alsa_channels_6_32bit(_THIS)
-{
-    SWIZ6(Uint32);
-}
-
-static SDL_INLINE void
-swizzle_alsa_channels_6_16bit(_THIS)
-{
-    SWIZ6(Uint16);
-}
-
-static SDL_INLINE void
-swizzle_alsa_channels_6_8bit(_THIS)
-{
-    SWIZ6(Uint8);
-}
-
-#undef SWIZ6
-
-
-/*
- * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
- *  channels from Windows/Mac order to the format alsalib will want.
- */
-static SDL_INLINE void
-swizzle_alsa_channels(_THIS)
-{
-    if (this->spec.channels == 6) {
-        const Uint16 fmtsize = (this->spec.format & 0xFF);      /* bits/channel. */
-        if (fmtsize == 16)
-            swizzle_alsa_channels_6_16bit(this);
-        else if (fmtsize == 8)
-            swizzle_alsa_channels_6_8bit(this);
-        else if (fmtsize == 32)
-            swizzle_alsa_channels_6_32bit(this);
-        else if (fmtsize == 64)
-            swizzle_alsa_channels_6_64bit(this);
-    }
-
-    /* !!! FIXME: update this for 7.1 if needed, later. */
-}
-
-
-static void
-ALSA_PlayDevice(_THIS)
-{
-    int status;
-    const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
-    const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *
-                                this->spec.channels;
-    snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
-
-    swizzle_alsa_channels(this);
-
-    while ( frames_left > 0 && this->enabled ) {
-        /* !!! FIXME: This works, but needs more testing before going live */
-        /* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
-        status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
-                                     sample_buf, frames_left);
-
-        if (status < 0) {
-            if (status == -EAGAIN) {
-                /* Apparently snd_pcm_recover() doesn't handle this case -
-                   does it assume snd_pcm_wait() above? */
-                SDL_Delay(1);
-                continue;
-            }
-            status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
-            if (status < 0) {
-                /* Hmm, not much we can do - abort */
-                fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
-                        ALSA_snd_strerror(status));
-                this->enabled = 0;
-                return;
-            }
-            continue;
-        }
-        sample_buf += status * frame_size;
-        frames_left -= status;
-    }
-}
-
-static Uint8 *
-ALSA_GetDeviceBuf(_THIS)
-{
-    return (this->hidden->mixbuf);
-}
-
-static void
-ALSA_CloseDevice(_THIS)
-{
-    if (this->hidden != NULL) {
-        SDL_FreeAudioMem(this->hidden->mixbuf);
-        this->hidden->mixbuf = NULL;
-        if (this->hidden->pcm_handle) {
-            ALSA_snd_pcm_drain(this->hidden->pcm_handle);
-            ALSA_snd_pcm_close(this->hidden->pcm_handle);
-            this->hidden->pcm_handle = NULL;
-        }
-        SDL_free(this->hidden);
-        this->hidden = NULL;
-    }
-}
-
-static int
-ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
-{
-    int status;
-    snd_pcm_uframes_t bufsize;
-
-    /* "set" the hardware with the desired parameters */
-    status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
-    if ( status < 0 ) {
-        return(-1);
-    }
-
-    /* Get samples for the actual buffer size */
-    status = ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
-    if ( status < 0 ) {
-        return(-1);
-    }
-    if ( !override && bufsize != this->spec.samples * 2 ) {
-        return(-1);
-    }
-
-    /* !!! FIXME: Is this safe to do? */
-    this->spec.samples = bufsize / 2;
-
-    /* This is useful for debugging */
-    if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
-        snd_pcm_uframes_t persize = 0;
-        unsigned int periods = 0;
-
-        ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);
-        ALSA_snd_pcm_hw_params_get_periods(hwparams, &periods, NULL);
-
-        fprintf(stderr,
-            "ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
-            persize, periods, bufsize);
-    }
-
-    return(0);
-}
-
-static int
-ALSA_set_period_size(_THIS, snd_pcm_hw_params_t *params, int override)
-{
-    const char *env;
-    int status;
-    snd_pcm_hw_params_t *hwparams;
-    snd_pcm_uframes_t frames;
-    unsigned int periods;
-
-    /* Copy the hardware parameters for this setup */
-    snd_pcm_hw_params_alloca(&hwparams);
-    ALSA_snd_pcm_hw_params_copy(hwparams, params);
-
-    if ( !override ) {
-        env = SDL_getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");
-        if ( env ) {
-            override = SDL_atoi(env);
-            if ( override == 0 ) {
-                return(-1);
-            }
-        }
-    }
-
-    frames = this->spec.samples;
-    status = ALSA_snd_pcm_hw_params_set_period_size_near(
-                this->hidden->pcm_handle, hwparams, &frames, NULL);
-    if ( status < 0 ) {
-        return(-1);
-    }
-
-    periods = 2;
-    status = ALSA_snd_pcm_hw_params_set_periods_near(
-                this->hidden->pcm_handle, hwparams, &periods, NULL);
-    if ( status < 0 ) {
-        return(-1);
-    }
-
-    return ALSA_finalize_hardware(this, hwparams, override);
-}
-
-static int
-ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params, int override)
-{
-    const char *env;
-    int status;
-    snd_pcm_hw_params_t *hwparams;
-    snd_pcm_uframes_t frames;
-
-    /* Copy the hardware parameters for this setup */
-    snd_pcm_hw_params_alloca(&hwparams);
-    ALSA_snd_pcm_hw_params_copy(hwparams, params);
-
-    if ( !override ) {
-        env = SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");
-        if ( env ) {
-            override = SDL_atoi(env);
-            if ( override == 0 ) {
-                return(-1);
-            }
-        }
-    }
-
-    frames = this->spec.samples * 2;
-    status = ALSA_snd_pcm_hw_params_set_buffer_size_near(
-                    this->hidden->pcm_handle, hwparams, &frames);
-    if ( status < 0 ) {
-        return(-1);
-    }
-
-    return ALSA_finalize_hardware(this, hwparams, override);
-}
-
-static int
-ALSA_OpenDevice(_THIS, const char *devname, int iscapture)
-{
-    int status = 0;
-    snd_pcm_t *pcm_handle = NULL;
-    snd_pcm_hw_params_t *hwparams = NULL;
-    snd_pcm_sw_params_t *swparams = NULL;
-    snd_pcm_format_t format = 0;
-    SDL_AudioFormat test_format = 0;
-    unsigned int rate = 0;
-    unsigned int channels = 0;
-
-    /* Initialize all variables that we clean on shutdown */
-    this->hidden = (struct SDL_PrivateAudioData *)
-        SDL_malloc((sizeof *this->hidden));
-    if (this->hidden == NULL) {
-        return SDL_OutOfMemory();
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Open the audio device */
-    /* Name of device should depend on # channels in spec */
-    status = ALSA_snd_pcm_open(&pcm_handle,
-                               get_audio_device(this->spec.channels),
-                               SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't open audio device: %s",
-                            ALSA_snd_strerror(status));
-    }
-
-    this->hidden->pcm_handle = pcm_handle;
-
-    /* Figure out what the hardware is capable of */
-    snd_pcm_hw_params_alloca(&hwparams);
-    status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't get hardware config: %s",
-                            ALSA_snd_strerror(status));
-    }
-
-    /* SDL only uses interleaved sample output */
-    status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
-                                               SND_PCM_ACCESS_RW_INTERLEAVED);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't set interleaved access: %s",
-                     ALSA_snd_strerror(status));
-    }
-
-    /* Try for a closest match on audio format */
-    status = -1;
-    for (test_format = SDL_FirstAudioFormat(this->spec.format);
-         test_format && (status < 0);) {
-        status = 0;             /* if we can't support a format, it'll become -1. */
-        switch (test_format) {
-        case AUDIO_U8:
-            format = SND_PCM_FORMAT_U8;
-            break;
-        case AUDIO_S8:
-            format = SND_PCM_FORMAT_S8;
-            break;
-        case AUDIO_S16LSB:
-            format = SND_PCM_FORMAT_S16_LE;
-            break;
-        case AUDIO_S16MSB:
-            format = SND_PCM_FORMAT_S16_BE;
-            break;
-        case AUDIO_U16LSB:
-            format = SND_PCM_FORMAT_U16_LE;
-            break;
-        case AUDIO_U16MSB:
-            format = SND_PCM_FORMAT_U16_BE;
-            break;
-        case AUDIO_S32LSB:
-            format = SND_PCM_FORMAT_S32_LE;
-            break;
-        case AUDIO_S32MSB:
-            format = SND_PCM_FORMAT_S32_BE;
-            break;
-        case AUDIO_F32LSB:
-            format = SND_PCM_FORMAT_FLOAT_LE;
-            break;
-        case AUDIO_F32MSB:
-            format = SND_PCM_FORMAT_FLOAT_BE;
-            break;
-        default:
-            status = -1;
-            break;
-        }
-        if (status >= 0) {
-            status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,
-                                                       hwparams, format);
-        }
-        if (status < 0) {
-            test_format = SDL_NextAudioFormat();
-        }
-    }
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't find any hardware audio formats");
-    }
-    this->spec.format = test_format;
-
-    /* Set the number of channels */
-    status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
-                                                 this->spec.channels);
-    channels = this->spec.channels;
-    if (status < 0) {
-        status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
-        if (status < 0) {
-            ALSA_CloseDevice(this);
-            return SDL_SetError("ALSA: Couldn't set audio channels");
-        }
-        this->spec.channels = channels;
-    }
-
-    /* Set the audio rate */
-    rate = this->spec.freq;
-    status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
-                                                  &rate, NULL);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't set audio frequency: %s",
-                            ALSA_snd_strerror(status));
-    }
-    this->spec.freq = rate;
-
-    /* Set the buffer size, in samples */
-    if ( ALSA_set_period_size(this, hwparams, 0) < 0 &&
-         ALSA_set_buffer_size(this, hwparams, 0) < 0 ) {
-        /* Failed to set desired buffer size, do the best you can... */
-        if ( ALSA_set_period_size(this, hwparams, 1) < 0 ) {
-            ALSA_CloseDevice(this);
-            return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
-        }
-    }
-    /* Set the software parameters */
-    snd_pcm_sw_params_alloca(&swparams);
-    status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't get software config: %s",
-                            ALSA_snd_strerror(status));
-    }
-    status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("Couldn't set minimum available samples: %s",
-                            ALSA_snd_strerror(status));
-    }
-    status =
-        ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("ALSA: Couldn't set start threshold: %s",
-                            ALSA_snd_strerror(status));
-    }
-    status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
-    if (status < 0) {
-        ALSA_CloseDevice(this);
-        return SDL_SetError("Couldn't set software audio parameters: %s",
-                            ALSA_snd_strerror(status));
-    }
-
-    /* Calculate the final parameters for this audio specification */
-    SDL_CalculateAudioSpec(&this->spec);
-
-    /* Allocate mixing buffer */
-    this->hidden->mixlen = this->spec.size;
-    this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
-    if (this->hidden->mixbuf == NULL) {
-        ALSA_CloseDevice(this);
-        return SDL_OutOfMemory();
-    }
-    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
-
-    /* Switch to blocking mode for playback */
-    ALSA_snd_pcm_nonblock(pcm_handle, 0);
-
-    /* We're ready to rock and roll. :-) */
-    return 0;
-}
-
-static void
-ALSA_Deinitialize(void)
-{
-    UnloadALSALibrary();
-}
-
-static int
-ALSA_Init(SDL_AudioDriverImpl * impl)
-{
-    if (LoadALSALibrary() < 0) {
-        return 0;
-    }
-
-    /* Set the function pointers */
-    impl->OpenDevice = ALSA_OpenDevice;
-    impl->WaitDevice = ALSA_WaitDevice;
-    impl->GetDeviceBuf = ALSA_GetDeviceBuf;
-    impl->PlayDevice = ALSA_PlayDevice;
-    impl->CloseDevice = ALSA_CloseDevice;
-    impl->Deinitialize = ALSA_Deinitialize;
-    impl->OnlyHasDefaultOutputDevice = 1;       /* !!! FIXME: Add device enum! */
-
-    return 1;   /* this audio target is available. */
-}
-
-
-AudioBootStrap ALSA_bootstrap = {
-    "alsa", "ALSA PCM audio", ALSA_Init, 0
-};
-
-#endif /* SDL_AUDIO_DRIVER_ALSA */
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.h
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.h b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.h
deleted file mode 100644
index 4535388..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/alsa/SDL_alsa_audio.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#ifndef _SDL_ALSA_audio_h
-#define _SDL_ALSA_audio_h
-
-#include <alsa/asoundlib.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS   SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-    /* The audio device handle */
-    snd_pcm_t *pcm_handle;
-
-    /* Raw mixing buffer */
-    Uint8 *mixbuf;
-    int mixlen;
-};
-
-#endif /* _SDL_ALSA_audio_h */
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.c
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.c b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.c
deleted file mode 100644
index 0c85b83..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.c
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#if SDL_AUDIO_DRIVER_ANDROID
-
-/* Output audio to Android */
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "SDL_androidaudio.h"
-
-#include "../../core/android/SDL_android.h"
-
-#include <android/log.h>
-
-static void * audioDevice;
-
-static int
-AndroidAUD_OpenDevice(_THIS, const char *devname, int iscapture)
-{
-    SDL_AudioFormat test_format;
-
-    if (iscapture) {
-        /* TODO: implement capture */
-        return SDL_SetError("Capture not supported on Android");
-    }
-
-    if (audioDevice != NULL) {
-        return SDL_SetError("Only one audio device at a time please!");
-    }
-
-    audioDevice = this;
-
-    test_format = SDL_FirstAudioFormat(this->spec.format);
-    while (test_format != 0) { /* no "UNKNOWN" constant */
-        if ((test_format == AUDIO_U8) || (test_format == AUDIO_S16LSB)) {
-            this->spec.format = test_format;
-            break;
-        }
-        test_format = SDL_NextAudioFormat();
-    }
-
-    if (test_format == 0) {
-        /* Didn't find a compatible format :( */
-        return SDL_SetError("No compatible audio format!");
-    }
-
-    if (this->spec.channels > 1) {
-        this->spec.channels = 2;
-    } else {
-        this->spec.channels = 1;
-    }
-
-    if (this->spec.freq < 8000) {
-        this->spec.freq = 8000;
-    }
-    if (this->spec.freq > 48000) {
-        this->spec.freq = 48000;
-    }
-
-    /* TODO: pass in/return a (Java) device ID, also whether we're opening for input or output */
-    this->spec.samples = Android_JNI_OpenAudioDevice(this->spec.freq, this->spec.format == AUDIO_U8 ? 0 : 1, this->spec.channels, this->spec.samples);
-    SDL_CalculateAudioSpec(&this->spec);
-
-    if (this->spec.samples == 0) {
-        /* Init failed? */
-        return SDL_SetError("Java-side initialization failed!");
-    }
-
-    return 0;
-}
-
-static void
-AndroidAUD_PlayDevice(_THIS)
-{
-    Android_JNI_WriteAudioBuffer();
-}
-
-static Uint8 *
-AndroidAUD_GetDeviceBuf(_THIS)
-{
-    return Android_JNI_GetAudioBuffer();
-}
-
-static void
-AndroidAUD_CloseDevice(_THIS)
-{
-    /* At this point SDL_CloseAudioDevice via close_audio_device took care of terminating the audio thread
-       so it's safe to terminate the Java side buffer and AudioTrack
-     */
-    Android_JNI_CloseAudioDevice();
-
-    if (audioDevice == this) {
-        audioDevice = NULL;
-    }
-}
-
-static int
-AndroidAUD_Init(SDL_AudioDriverImpl * impl)
-{
-    /* Set the function pointers */
-    impl->OpenDevice = AndroidAUD_OpenDevice;
-    impl->PlayDevice = AndroidAUD_PlayDevice;
-    impl->GetDeviceBuf = AndroidAUD_GetDeviceBuf;
-    impl->CloseDevice = AndroidAUD_CloseDevice;
-
-    /* and the capabilities */
-    impl->HasCaptureSupport = 0; /* TODO */
-    impl->OnlyHasDefaultOutputDevice = 1;
-    impl->OnlyHasDefaultInputDevice = 1;
-
-    return 1;   /* this audio target is available. */
-}
-
-AudioBootStrap ANDROIDAUD_bootstrap = {
-    "android", "SDL Android audio driver", AndroidAUD_Init, 0
-};
-
-#endif /* SDL_AUDIO_DRIVER_ANDROID */
-
-/* vi: set ts=4 sw=4 expandtab: */
-

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.h
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.h b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.h
deleted file mode 100644
index ab49f00..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/android/SDL_androidaudio.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#ifndef _SDL_androidaudio_h
-#define _SDL_androidaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS   SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-};
-
-static void AndroidAUD_CloseDevice(_THIS);
-
-#endif /* _SDL_androidaudio_h */
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.c
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.c b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.c
deleted file mode 100644
index 72fba70..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.c
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#if SDL_AUDIO_DRIVER_ARTS
-
-/* Allow access to a raw mixing buffer */
-
-#ifdef HAVE_SIGNAL_H
-#include <signal.h>
-#endif
-#include <unistd.h>
-#include <errno.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_artsaudio.h"
-
-#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
-#include "SDL_name.h"
-#include "SDL_loadso.h"
-#else
-#define SDL_NAME(X)	X
-#endif
-
-#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
-
-static const char *arts_library = SDL_AUDIO_DRIVER_ARTS_DYNAMIC;
-static void *arts_handle = NULL;
-
-/* !!! FIXME: I hate this SDL_NAME clutter...it makes everything so messy! */
-static int (*SDL_NAME(arts_init)) (void);
-static void (*SDL_NAME(arts_free)) (void);
-static arts_stream_t(*SDL_NAME(arts_play_stream)) (int rate, int bits,
-                                                   int channels,
-                                                   const char *name);
-static int (*SDL_NAME(arts_stream_set)) (arts_stream_t s,
-                                         arts_parameter_t param, int value);
-static int (*SDL_NAME(arts_stream_get)) (arts_stream_t s,
-                                         arts_parameter_t param);
-static int (*SDL_NAME(arts_write)) (arts_stream_t s, const void *buffer,
-                                    int count);
-static void (*SDL_NAME(arts_close_stream)) (arts_stream_t s);
-static int (*SDL_NAME(arts_suspend))(void);
-static int (*SDL_NAME(arts_suspended)) (void);
-static const char *(*SDL_NAME(arts_error_text)) (int errorcode);
-
-#define SDL_ARTS_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
-static struct
-{
-    const char *name;
-    void **func;
-} arts_functions[] = {
-/* *INDENT-OFF* */
-    SDL_ARTS_SYM(arts_init),
-    SDL_ARTS_SYM(arts_free),
-    SDL_ARTS_SYM(arts_play_stream),
-    SDL_ARTS_SYM(arts_stream_set),
-    SDL_ARTS_SYM(arts_stream_get),
-    SDL_ARTS_SYM(arts_write),
-    SDL_ARTS_SYM(arts_close_stream),
-    SDL_ARTS_SYM(arts_suspend),
-    SDL_ARTS_SYM(arts_suspended),
-    SDL_ARTS_SYM(arts_error_text),
-/* *INDENT-ON* */
-};
-
-#undef SDL_ARTS_SYM
-
-static void
-UnloadARTSLibrary()
-{
-    if (arts_handle != NULL) {
-        SDL_UnloadObject(arts_handle);
-        arts_handle = NULL;
-    }
-}
-
-static int
-LoadARTSLibrary(void)
-{
-    int i, retval = -1;
-
-    if (arts_handle == NULL) {
-        arts_handle = SDL_LoadObject(arts_library);
-        if (arts_handle != NULL) {
-            retval = 0;
-            for (i = 0; i < SDL_arraysize(arts_functions); ++i) {
-                *arts_functions[i].func =
-                    SDL_LoadFunction(arts_handle, arts_functions[i].name);
-                if (!*arts_functions[i].func) {
-                    retval = -1;
-                    UnloadARTSLibrary();
-                    break;
-                }
-            }
-        }
-    }
-
-    return retval;
-}
-
-#else
-
-static void
-UnloadARTSLibrary()
-{
-    return;
-}
-
-static int
-LoadARTSLibrary(void)
-{
-    return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_ARTS_DYNAMIC */
-
-/* This function waits until it is possible to write a full sound buffer */
-static void
-ARTS_WaitDevice(_THIS)
-{
-    Sint32 ticks;
-
-    /* Check to see if the thread-parent process is still alive */
-    {
-        static int cnt = 0;
-        /* Note that this only works with thread implementations
-           that use a different process id for each thread.
-         */
-        /* Check every 10 loops */
-        if (this->hidden->parent && (((++cnt) % 10) == 0)) {
-            if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
-                this->enabled = 0;
-            }
-        }
-    }
-
-    /* Use timer for general audio synchronization */
-    ticks =
-        ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
-    if (ticks > 0) {
-        SDL_Delay(ticks);
-    }
-}
-
-static void
-ARTS_PlayDevice(_THIS)
-{
-    /* Write the audio data */
-    int written = SDL_NAME(arts_write) (this->hidden->stream,
-                                        this->hidden->mixbuf,
-                                        this->hidden->mixlen);
-
-    /* If timer synchronization is enabled, set the next write frame */
-    if (this->hidden->frame_ticks) {
-        this->hidden->next_frame += this->hidden->frame_ticks;
-    }
-
-    /* If we couldn't write, assume fatal error for now */
-    if (written < 0) {
-        this->enabled = 0;
-    }
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
-#endif
-}
-
-static void
-ARTS_WaitDone(_THIS)
-{
-    /* !!! FIXME: camp here until buffer drains... SDL_Delay(???); */
-}
-
-
-static Uint8 *
-ARTS_GetDeviceBuf(_THIS)
-{
-    return (this->hidden->mixbuf);
-}
-
-
-static void
-ARTS_CloseDevice(_THIS)
-{
-    if (this->hidden != NULL) {
-        SDL_FreeAudioMem(this->hidden->mixbuf);
-        this->hidden->mixbuf = NULL;
-        if (this->hidden->stream) {
-            SDL_NAME(arts_close_stream) (this->hidden->stream);
-            this->hidden->stream = 0;
-        }
-        SDL_NAME(arts_free) ();
-        SDL_free(this->hidden);
-        this->hidden = NULL;
-    }
-}
-
-static int
-ARTS_Suspend(void)
-{
-    const Uint32 abortms = SDL_GetTicks() + 3000; /* give up after 3 secs */
-    while ( (!SDL_NAME(arts_suspended)()) && !SDL_TICKS_PASSED(SDL_GetTicks(), abortms) ) {
-        if ( SDL_NAME(arts_suspend)() ) {
-            break;
-        }
-    }
-    return SDL_NAME(arts_suspended)();
-}
-
-static int
-ARTS_OpenDevice(_THIS, const char *devname, int iscapture)
-{
-    int rc = 0;
-    int bits = 0, frag_spec = 0;
-    SDL_AudioFormat test_format = 0, format = 0;
-
-    /* Initialize all variables that we clean on shutdown */
-    this->hidden = (struct SDL_PrivateAudioData *)
-        SDL_malloc((sizeof *this->hidden));
-    if (this->hidden == NULL) {
-        return SDL_OutOfMemory();
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Try for a closest match on audio format */
-    for (test_format = SDL_FirstAudioFormat(this->spec.format);
-         !format && test_format;) {
-#ifdef DEBUG_AUDIO
-        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-        switch (test_format) {
-        case AUDIO_U8:
-            bits = 8;
-            format = 1;
-            break;
-        case AUDIO_S16LSB:
-            bits = 16;
-            format = 1;
-            break;
-        default:
-            format = 0;
-            break;
-        }
-        if (!format) {
-            test_format = SDL_NextAudioFormat();
-        }
-    }
-    if (format == 0) {
-        ARTS_CloseDevice(this);
-        return SDL_SetError("Couldn't find any hardware audio formats");
-    }
-    this->spec.format = test_format;
-
-    if ((rc = SDL_NAME(arts_init) ()) != 0) {
-        ARTS_CloseDevice(this);
-        return SDL_SetError("Unable to initialize ARTS: %s",
-                            SDL_NAME(arts_error_text) (rc));
-    }
-
-    if (!ARTS_Suspend()) {
-        ARTS_CloseDevice(this);
-        return SDL_SetError("ARTS can not open audio device");
-    }
-
-    this->hidden->stream = SDL_NAME(arts_play_stream) (this->spec.freq,
-                                                       bits,
-                                                       this->spec.channels,
-                                                       "SDL");
-
-    /* Play nothing so we have at least one write (server bug workaround). */
-    SDL_NAME(arts_write) (this->hidden->stream, "", 0);
-
-    /* Calculate the final parameters for this audio specification */
-    SDL_CalculateAudioSpec(&this->spec);
-
-    /* Determine the power of two of the fragment size */
-    for (frag_spec = 0; (0x01 << frag_spec) < this->spec.size; ++frag_spec);
-    if ((0x01 << frag_spec) != this->spec.size) {
-        ARTS_CloseDevice(this);
-        return SDL_SetError("Fragment size must be a power of two");
-    }
-    frag_spec |= 0x00020000;    /* two fragments, for low latency */
-
-#ifdef ARTS_P_PACKET_SETTINGS
-    SDL_NAME(arts_stream_set) (this->hidden->stream,
-                               ARTS_P_PACKET_SETTINGS, frag_spec);
-#else
-    SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_SIZE,
-                               frag_spec & 0xffff);
-    SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_COUNT,
-                               frag_spec >> 16);
-#endif
-    this->spec.size = SDL_NAME(arts_stream_get) (this->hidden->stream,
-                                                 ARTS_P_PACKET_SIZE);
-
-    /* Allocate mixing buffer */
-    this->hidden->mixlen = this->spec.size;
-    this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
-    if (this->hidden->mixbuf == NULL) {
-        ARTS_CloseDevice(this);
-        return SDL_OutOfMemory();
-    }
-    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
-
-    /* Get the parent process id (we're the parent of the audio thread) */
-    this->hidden->parent = getpid();
-
-    /* We're ready to rock and roll. :-) */
-    return 0;
-}
-
-
-static void
-ARTS_Deinitialize(void)
-{
-    UnloadARTSLibrary();
-}
-
-
-static int
-ARTS_Init(SDL_AudioDriverImpl * impl)
-{
-    if (LoadARTSLibrary() < 0) {
-        return 0;
-    } else {
-        if (SDL_NAME(arts_init) () != 0) {
-            UnloadARTSLibrary();
-            SDL_SetError("ARTS: arts_init failed (no audio server?)");
-            return 0;
-        }
-
-        /* Play a stream so aRts doesn't crash */
-        if (ARTS_Suspend()) {
-            arts_stream_t stream;
-            stream = SDL_NAME(arts_play_stream) (44100, 16, 2, "SDL");
-            SDL_NAME(arts_write) (stream, "", 0);
-            SDL_NAME(arts_close_stream) (stream);
-        }
-
-        SDL_NAME(arts_free) ();
-    }
-
-    /* Set the function pointers */
-    impl->OpenDevice = ARTS_OpenDevice;
-    impl->PlayDevice = ARTS_PlayDevice;
-    impl->WaitDevice = ARTS_WaitDevice;
-    impl->GetDeviceBuf = ARTS_GetDeviceBuf;
-    impl->CloseDevice = ARTS_CloseDevice;
-    impl->WaitDone = ARTS_WaitDone;
-    impl->Deinitialize = ARTS_Deinitialize;
-    impl->OnlyHasDefaultOutputDevice = 1;
-
-    return 1;   /* this audio target is available. */
-}
-
-
-AudioBootStrap ARTS_bootstrap = {
-    "arts", "Analog RealTime Synthesizer", ARTS_Init, 0
-};
-
-#endif /* SDL_AUDIO_DRIVER_ARTS */
-
-/* vi: set ts=4 sw=4 expandtab: */

http://git-wip-us.apache.org/repos/asf/incubator-corinthia/blob/d1484ae0/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.h
----------------------------------------------------------------------
diff --git a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.h b/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.h
deleted file mode 100644
index fb7706f..0000000
--- a/DocFormats/platform/3rdparty/SDL2-2.0.3/src/audio/arts/SDL_artsaudio.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
-  Simple DirectMedia Layer
-  Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
-
-  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.
-
-  Permission is granted to anyone to use this software for any purpose,
-  including commercial applications, and to alter it and redistribute it
-  freely, subject to the following restrictions:
-
-  1. The origin of this software must not be misrepresented; you must not
-     claim that you wrote the original software. If you use this software
-     in a product, an acknowledgment in the product documentation would be
-     appreciated but is not required.
-  2. Altered source versions must be plainly marked as such, and must not be
-     misrepresented as being the original software.
-  3. This notice may not be removed or altered from any source distribution.
-*/
-#include "../../SDL_internal.h"
-
-#ifndef _SDL_artscaudio_h
-#define _SDL_artscaudio_h
-
-#include <artsc.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS   SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-    /* The stream descriptor for the audio device */
-    arts_stream_t stream;
-
-    /* The parent process id, to detect when application quits */
-    pid_t parent;
-
-    /* Raw mixing buffer */
-    Uint8 *mixbuf;
-    int mixlen;
-
-    /* Support for audio timing using a timer, in addition to select() */
-    float frame_ticks;
-    float next_frame;
-};
-#define FUDGE_TICKS 10      /* The scheduler overhead ticks per frame */
-
-#endif /* _SDL_artscaudio_h */
-/* vi: set ts=4 sw=4 expandtab: */


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