SDL_mixer: add SDL_ prefix to AUDIO_* constants

From 82fc2ac7f82c1dc896703a9052c9d523bb899f33 Mon Sep 17 00:00:00 2001
From: Nhalrath <[EMAIL REDACTED]>
Date: Wed, 3 May 2023 14:15:54 +0800
Subject: [PATCH] add SDL_ prefix to AUDIO_* constants

---
 include/SDL3/SDL_mixer.h                   | 16 +++----
 playmus.c                                  |  4 +-
 playwave.c                                 |  4 +-
 src/codecs/load_aiff.c                     |  4 +-
 src/codecs/load_sndfile.c                  |  2 +-
 src/codecs/load_voc.c                      |  2 +-
 src/codecs/music_drflac.c                  | 12 ++---
 src/codecs/music_drmp3.c                   | 12 ++---
 src/codecs/music_flac.c                    |  8 +++-
 src/codecs/music_fluidsynth.c              |  4 +-
 src/codecs/music_gme.c                     |  7 ++-
 src/codecs/music_modplug.c                 |  8 +++-
 src/codecs/music_mpg123.c                  | 10 ++--
 src/codecs/music_ogg.c                     |  7 ++-
 src/codecs/music_ogg_stb.c                 |  8 +++-
 src/codecs/music_opus.c                    |  8 +++-
 src/codecs/music_wav.c                     | 54 +++++++++++-----------
 src/codecs/music_wavpack.c                 |  6 +--
 src/codecs/music_xmp.c                     |  7 ++-
 src/codecs/native_midi/native_midi_win32.c |  2 +-
 src/codecs/timidity/timidity.c             | 14 +++---
 src/effect_position.c                      | 14 +++---
 22 files changed, 119 insertions(+), 94 deletions(-)

diff --git a/include/SDL3/SDL_mixer.h b/include/SDL3/SDL_mixer.h
index 6e3f0bf5..846a3e52 100644
--- a/include/SDL3/SDL_mixer.h
+++ b/include/SDL3/SDL_mixer.h
@@ -221,9 +221,9 @@ extern DECLSPEC void SDLCALL Mix_Quit(void);
 /* Good default values for a PC soundcard */
 #define MIX_DEFAULT_FREQUENCY   44100
 #if SDL_BYTEORDER == SDL_LIL_ENDIAN
-#define MIX_DEFAULT_FORMAT  AUDIO_S16LSB
+#define MIX_DEFAULT_FORMAT  SDL_AUDIO_S16LSB
 #else
-#define MIX_DEFAULT_FORMAT  AUDIO_S16MSB
+#define MIX_DEFAULT_FORMAT  SDL_AUDIO_S16MSB
 #endif
 #define MIX_DEFAULT_CHANNELS    2
 #define MIX_MAX_VOLUME          SDL_MIX_MAXVOLUME /* Volume of a chunk */
@@ -314,9 +314,9 @@ typedef struct _Mix_Music Mix_Music;
  * The audio device frequency is specified in Hz; in modern times, 48000 is
  * often a reasonable default.
  *
- * The audio device format is one of SDL's AUDIO_* constants. AUDIO_S16SYS
+ * The audio device format is one of SDL's SDL_AUDIO_* constants. SDL_AUDIO_S16SYS
  * (16-bit audio) is probably a safe default. More modern systems may prefer
- * AUDIO_F32SYS (32-bit floating point audio).
+ * SDL_AUDIO_F32SYS (32-bit floating point audio).
  *
  * The audio device channels are generally 1 for mono output, or 2 for stereo,
  * but the brave can try surround sound configs with 4 (quad), 6 (5.1), 7
@@ -347,7 +347,7 @@ typedef struct _Mix_Music Mix_Music;
  * should dispose of the device with Mix_CloseAudio().
  *
  * \param frequency the frequency to playback audio at (in Hz).
- * \param format audio format, one of SDL's AUDIO_* values.
+ * \param format audio format, one of SDL's SDL_AUDIO_* values.
  * \param channels number of channels (1 is mono, 2 is stereo, etc).
  * \param chunksize audio buffer size in sample FRAMES (total samples divided
  *                  by channel count).
@@ -397,9 +397,9 @@ extern DECLSPEC int SDLCALL Mix_OpenAudio(int frequency, Uint16 format, int chan
  * The audio device frequency is specified in Hz; in modern times, 48000 is
  * often a reasonable default.
  *
- * The audio device format is one of SDL's AUDIO_* constants. AUDIO_S16SYS
+ * The audio device format is one of SDL's SDL_AUDIO_* constants. SDL_AUDIO_S16SYS
  * (16-bit audio) is probably a safe default. More modern systems may prefer
- * AUDIO_F32SYS (32-bit floating point audio).
+ * SDL_AUDIO_F32SYS (32-bit floating point audio).
  *
  * The audio device channels are generally 1 for mono output, or 2 for stereo,
  * but the brave can try surround sound configs with 4 (quad), 6 (5.1), 7
@@ -448,7 +448,7 @@ extern DECLSPEC int SDLCALL Mix_OpenAudio(int frequency, Uint16 format, int chan
  * should dispose of the device with Mix_CloseDevice().
  *
  * \param frequency the frequency to playback audio at (in Hz).
- * \param format audio format, one of SDL's AUDIO_* values.
+ * \param format audio format, one of SDL's SDL_AUDIO_* values.
  * \param channels number of channels (1 is mono, 2 is stereo, etc).
  * \param chunksize audio buffer size in sample FRAMES (total samples divided
  *                  by channel count).
diff --git a/playmus.c b/playmus.c
index 832be82d..a68ee930 100644
--- a/playmus.c
+++ b/playmus.c
@@ -165,10 +165,10 @@ int main(int argc, char *argv[])
             interactive = 1;
         } else
         if (strcmp(argv[i], "-8") == 0) {
-            audio_format = AUDIO_U8;
+            audio_format = SDL_AUDIO_U8;
         } else
         if (strcmp(argv[i], "-f32") == 0) {
-            audio_format = AUDIO_F32;
+            audio_format = SDL_AUDIO_F32;
         } else
         if (strcmp(argv[i], "-rwops") == 0) {
             rwops = 1;
diff --git a/playwave.c b/playwave.c
index d75cbcbd..117a5509 100644
--- a/playwave.c
+++ b/playwave.c
@@ -395,10 +395,10 @@ int main(int argc, char *argv[])
             loops = -1;
         } else
         if (strcmp(argv[i], "-8") == 0) {
-            audio_format = AUDIO_U8;
+            audio_format = SDL_AUDIO_U8;
         } else
         if (strcmp(argv[i], "-f32") == 0) {
-            audio_format = AUDIO_F32;
+            audio_format = SDL_AUDIO_F32;
         } else
         if (strcmp(argv[i], "-f") == 0) { /* rcg06122001 flip stereo */
             reverse_stereo = 1;
diff --git a/src/codecs/load_aiff.c b/src/codecs/load_aiff.c
index 5984048c..415e890c 100644
--- a/src/codecs/load_aiff.c
+++ b/src/codecs/load_aiff.c
@@ -210,10 +210,10 @@ SDL_AudioSpec *Mix_LoadAIFF_RW (SDL_RWops *src, int freesrc,
     spec->freq = frequency;
     switch (samplesize) {
         case 8:
-            spec->format = AUDIO_S8;
+            spec->format = SDL_AUDIO_S8;
             break;
         case 16:
-            spec->format = AUDIO_S16MSB;
+            spec->format = SDL_AUDIO_S16MSB;
             break;
         default:
             Mix_SetError("Unsupported AIFF samplesize");
diff --git a/src/codecs/load_sndfile.c b/src/codecs/load_sndfile.c
index b05692c4..d0a4beb2 100644
--- a/src/codecs/load_sndfile.c
+++ b/src/codecs/load_sndfile.c
@@ -186,7 +186,7 @@ SDL_AudioSpec *Mix_LoadSndFile_RW (SDL_RWops *src, int freesrc,
 
     spec->channels = sfinfo.channels;
     spec->freq = sfinfo.samplerate;
-    spec->format = AUDIO_S16;
+    spec->format = SDL_AUDIO_S16;
 
     *audio_buf = (Uint8 *)buf;
     *audio_len = len;
diff --git a/src/codecs/load_voc.c b/src/codecs/load_voc.c
index 506d2055..956b2cae 100644
--- a/src/codecs/load_voc.c
+++ b/src/codecs/load_voc.c
@@ -407,7 +407,7 @@ SDL_AudioSpec *Mix_LoadVOC_RW (SDL_RWops *src, int freesrc,
         goto done;
     }
 
-    spec->format = ((v.size == ST_SIZE_WORD) ? AUDIO_S16 : AUDIO_U8);
+    spec->format = ((v.size == ST_SIZE_WORD) ? SDL_AUDIO_S16 : SDL_AUDIO_U8);
     if (spec->channels == 0)
         spec->channels = v.channels;
 
diff --git a/src/codecs/music_drflac.c b/src/codecs/music_drflac.c
index 7ce2a89b..4a5b54dc 100644
--- a/src/codecs/music_drflac.c
+++ b/src/codecs/music_drflac.c
@@ -184,12 +184,12 @@ static void *DRFLAC_CreateFromRW(SDL_RWops *src, int freesrc)
     }
 
     /* We should have channels and sample rate set up here */
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS,
-                                       (Uint8)music->channels,
-                                       music->sample_rate,
-                                       music_spec.format,
-                                       music_spec.channels,
-                                       music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS,
+                                          (Uint8)music->channels,
+                                          music->sample_rate,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         SDL_OutOfMemory();
         drflac_close(music->dec);
diff --git a/src/codecs/music_drmp3.c b/src/codecs/music_drmp3.c
index d1483ac8..6cdc0499 100644
--- a/src/codecs/music_drmp3.c
+++ b/src/codecs/music_drmp3.c
@@ -111,12 +111,12 @@ static void *DRMP3_CreateFromRW(SDL_RWops *src, int freesrc)
     }
 
     music->channels = music->dec.channels;
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS,
-                                       (Uint8)music->channels,
-                                       (int)music->dec.sampleRate,
-                                       music_spec.format,
-                                       music_spec.channels,
-                                       music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS,
+                                          (Uint8)music->channels,
+                                          (int)music->dec.sampleRate,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         SDL_OutOfMemory();
         drmp3_uninit(&music->dec);
diff --git a/src/codecs/music_flac.c b/src/codecs/music_flac.c
index a36078a3..a5427a85 100644
--- a/src/codecs/music_flac.c
+++ b/src/codecs/music_flac.c
@@ -388,8 +388,12 @@ static void flac_metadata_music_cb(
 
         /* We check for NULL stream later when we get data */
         SDL_assert(!music->stream);
-        music->stream = SDL_CreateAudioStream(AUDIO_S16SYS, (Uint8)channels, (int)music->sample_rate,
-                                          music_spec.format, music_spec.channels, music_spec.freq);
+        music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS,
+                                              (Uint8)channels,
+                                              (int)music->sample_rate,
+                                              music_spec.format,
+                                              music_spec.channels,
+                                              music_spec.freq);
     } else if (metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
         FLAC__uint32 i;
 
diff --git a/src/codecs/music_fluidsynth.c b/src/codecs/music_fluidsynth.c
index 1bbb308a..856ccc1f 100644
--- a/src/codecs/music_fluidsynth.c
+++ b/src/codecs/music_fluidsynth.c
@@ -175,7 +175,7 @@ static FLUIDSYNTH_Music *FLUIDSYNTH_LoadMusic(void *data)
     FLUIDSYNTH_Music *music;
     double samplerate; /* as set by the lib. */
     const Uint8 channels = 2;
-    int src_format = AUDIO_S16SYS;
+    int src_format = SDL_AUDIO_S16SYS;
     void *rw_mem;
     size_t rw_size;
     int ret;
@@ -189,7 +189,7 @@ static FLUIDSYNTH_Music *FLUIDSYNTH_LoadMusic(void *data)
     music->buffer_size = music_spec.samples * sizeof(Sint16) * channels;
     music->synth_write = fluidsynth.fluid_synth_write_s16;
     if (music_spec.format & 0x0020) { /* 32 bit. */
-        src_format = AUDIO_F32SYS;
+        src_format = SDL_AUDIO_F32SYS;
         music->buffer_size <<= 1;
         music->synth_write = fluidsynth.fluid_synth_write_float;
     }
diff --git a/src/codecs/music_gme.c b/src/codecs/music_gme.c
index ff2fe314..fa3db80d 100644
--- a/src/codecs/music_gme.c
+++ b/src/codecs/music_gme.c
@@ -222,8 +222,11 @@ static void *GME_CreateFromRW(struct SDL_RWops *src, int freesrc)
     music->tempo = 1.0;
     music->gain = 1.0;
 
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS, 2, music_spec.freq,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS, 2,
+                                          music_spec.freq,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         GME_Delete(music);
         return NULL;
diff --git a/src/codecs/music_modplug.c b/src/codecs/music_modplug.c
index 51e81b90..486b60eb 100644
--- a/src/codecs/music_modplug.c
+++ b/src/codecs/music_modplug.c
@@ -175,8 +175,12 @@ void *MODPLUG_CreateFromRW(SDL_RWops *src, int freesrc)
 
     music->volume = MIX_MAX_VOLUME;
 
-    music->stream = SDL_CreateAudioStream((settings.mBits == 8) ? AUDIO_U8 : AUDIO_S16SYS, (Uint8)settings.mChannels, settings.mFrequency,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream((settings.mBits == 8) ? SDL_AUDIO_U8 : SDL_AUDIO_S16SYS,
+                                          (Uint8)settings.mChannels,
+                                          settings.mFrequency,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         MODPLUG_Delete(music);
         return NULL;
diff --git a/src/codecs/music_mpg123.c b/src/codecs/music_mpg123.c
index 65143530..a95c91ab 100644
--- a/src/codecs/music_mpg123.c
+++ b/src/codecs/music_mpg123.c
@@ -156,11 +156,11 @@ static void MPG123_Delete(void *context);
 static int mpg123_format_to_sdl(int fmt)
 {
     switch (fmt) {
-        case MPG123_ENC_SIGNED_8:       return AUDIO_S8;
-        case MPG123_ENC_UNSIGNED_8:     return AUDIO_U8;
-        case MPG123_ENC_SIGNED_16:      return AUDIO_S16SYS;
-        case MPG123_ENC_SIGNED_32:      return AUDIO_S32SYS;
-        case MPG123_ENC_FLOAT_32:       return AUDIO_F32SYS;
+        case MPG123_ENC_SIGNED_8:       return SDL_AUDIO_S8;
+        case MPG123_ENC_UNSIGNED_8:     return SDL_AUDIO_U8;
+        case MPG123_ENC_SIGNED_16:      return SDL_AUDIO_S16SYS;
+        case MPG123_ENC_SIGNED_32:      return SDL_AUDIO_S32SYS;
+        case MPG123_ENC_FLOAT_32:       return SDL_AUDIO_F32SYS;
         default:                        return -1;
     }
 }
diff --git a/src/codecs/music_ogg.c b/src/codecs/music_ogg.c
index 0e95ab2f..952955c4 100644
--- a/src/codecs/music_ogg.c
+++ b/src/codecs/music_ogg.c
@@ -211,8 +211,11 @@ static int OGG_UpdateSection(OGG_music *music)
         music->stream = NULL;
     }
 
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS, (Uint8)vi->channels, (int)vi->rate,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS,
+                                          (Uint8)vi->channels, (int)vi->rate,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         return -1;
     }
diff --git a/src/codecs/music_ogg_stb.c b/src/codecs/music_ogg_stb.c
index 7bd3b30c..715c4fd5 100644
--- a/src/codecs/music_ogg_stb.c
+++ b/src/codecs/music_ogg_stb.c
@@ -141,8 +141,12 @@ static int OGG_UpdateSection(OGG_music *music)
         music->stream = NULL;
     }
 
-    music->stream = SDL_CreateAudioStream(AUDIO_F32SYS, (Uint8)vi.channels, (int)vi.sample_rate,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_F32SYS,
+                                          (Uint8)vi.channels,
+                                          (int)vi.sample_rate,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         return -1;
     }
diff --git a/src/codecs/music_opus.c b/src/codecs/music_opus.c
index 09619938..ee936ae1 100644
--- a/src/codecs/music_opus.c
+++ b/src/codecs/music_opus.c
@@ -191,8 +191,12 @@ static int OPUS_UpdateSection(OPUS_music *music)
         music->stream = NULL;
     }
 
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS, (Uint8)op_info->channel_count, 48000,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS,
+                                          (Uint8)op_info->channel_count,
+                                          48000,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         return -1;
     }
diff --git a/src/codecs/music_wav.c b/src/codecs/music_wav.c
index 50e42738..450ace61 100644
--- a/src/codecs/music_wav.c
+++ b/src/codecs/music_wav.c
@@ -730,15 +730,15 @@ static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
     switch (bits) {
         case 8:
             switch(wave->encoding) {
-            case PCM_CODE:  spec->format = AUDIO_U8; break;
-            case ALAW_CODE: spec->format = AUDIO_S16; break;
-            case uLAW_CODE: spec->format = AUDIO_S16; break;
+            case PCM_CODE:  spec->format = SDL_AUDIO_U8; break;
+            case ALAW_CODE: spec->format = SDL_AUDIO_S16; break;
+            case uLAW_CODE: spec->format = SDL_AUDIO_S16; break;
             default: goto unknown_bits;
             }
             break;
         case 16:
             switch(wave->encoding) {
-            case PCM_CODE: spec->format = AUDIO_S16; break;
+            case PCM_CODE: spec->format = SDL_AUDIO_S16; break;
             default: goto unknown_bits;
             }
             break;
@@ -746,15 +746,15 @@ static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
             switch(wave->encoding) {
             case PCM_CODE:
                 wave->decode = fetch_pcm24le;
-                spec->format = AUDIO_S32;
+                spec->format = SDL_AUDIO_S32;
                 break;
             default: goto unknown_bits;
             }
             break;
         case 32:
             switch(wave->encoding) {
-            case PCM_CODE:   spec->format = AUDIO_S32; break;
-            case FLOAT_CODE: spec->format = AUDIO_F32; break;
+            case PCM_CODE:   spec->format = SDL_AUDIO_S32; break;
+            case FLOAT_CODE: spec->format = SDL_AUDIO_F32; break;
             default: goto unknown_bits;
             }
             break;
@@ -762,7 +762,7 @@ static SDL_bool ParseFMT(WAV_Music *wave, Uint32 chunk_length)
             switch(wave->encoding) {
             case FLOAT_CODE:
                 wave->decode = fetch_float64le;
-                spec->format = AUDIO_F32;
+                spec->format = SDL_AUDIO_F32;
                 break;
             default: goto unknown_bits;
             }
@@ -1176,17 +1176,17 @@ static SDL_bool LoadAIFFMusic(WAV_Music *wave)
     switch (samplesize) {
     case 8:
         if (!is_AIFC)
-            spec->format = AUDIO_S8;
+            spec->format = SDL_AUDIO_S8;
         else switch (compressionType) {
-        case raw_: spec->format = AUDIO_U8; break;
-        case sowt: spec->format = AUDIO_S8; break;
+        case raw_: spec->format = SDL_AUDIO_U8; break;
+        case sowt: spec->format = SDL_AUDIO_S8; break;
         case ulaw:
-            spec->format = AUDIO_S16LSB;
+            spec->format = SDL_AUDIO_S16LSB;
             wave->encoding = uLAW_CODE;
             wave->decode = fetch_ulaw;
             break;
         case alaw:
-            spec->format = AUDIO_S16LSB;
+            spec->format = SDL_AUDIO_S16LSB;
             wave->encoding = ALAW_CODE;
             wave->decode = fetch_alaw;
             break;
@@ -1195,17 +1195,17 @@ static SDL_bool LoadAIFFMusic(WAV_Music *wave)
         break;
     case 16:
         if (!is_AIFC)
-            spec->format = AUDIO_S16MSB;
+            spec->format = SDL_AUDIO_S16MSB;
         else switch (compressionType) {
-        case sowt: spec->format = AUDIO_S16LSB; break;
-        case NONE: spec->format = AUDIO_S16MSB; break;
+        case sowt: spec->format = SDL_AUDIO_S16LSB; break;
+        case NONE: spec->format = SDL_AUDIO_S16MSB; break;
         case ULAW:
-            spec->format = AUDIO_S16LSB;
+            spec->format = SDL_AUDIO_S16LSB;
             wave->encoding = uLAW_CODE;
             wave->decode = fetch_ulaw;
             break;
         case ALAW:
-            spec->format = AUDIO_S16LSB;
+            spec->format = SDL_AUDIO_S16LSB;
             wave->encoding = ALAW_CODE;
             wave->decode = fetch_alaw;
             break;
@@ -1216,21 +1216,21 @@ static SDL_bool LoadAIFFMusic(WAV_Music *wave)
         wave->encoding = PCM_CODE;
         wave->decode = fetch_pcm24be;
         if (!is_AIFC)
-            spec->format = AUDIO_S32MSB;
+            spec->format = SDL_AUDIO_S32MSB;
         else switch (compressionType) {
-        case sowt: spec->format = AUDIO_S32LSB; break;
-        case NONE: spec->format = AUDIO_S32MSB; break;
+        case sowt: spec->format = SDL_AUDIO_S32LSB; break;
+        case NONE: spec->format = SDL_AUDIO_S32MSB; break;
         default: goto unsupported_format;
         }
         break;
     case 32:
         if (!is_AIFC)
-            spec->format = AUDIO_S32MSB;
+            spec->format = SDL_AUDIO_S32MSB;
         else switch (compressionType) {
-        case sowt: spec->format = AUDIO_S32LSB; break;
-        case NONE: spec->format = AUDIO_S32MSB; break;
+        case sowt: spec->format = SDL_AUDIO_S32LSB; break;
+        case NONE: spec->format = SDL_AUDIO_S32MSB; break;
         case fl32:
-        case FL32: spec->format = AUDIO_F32MSB; break;
+        case FL32: spec->format = SDL_AUDIO_F32MSB; break;
         default: goto unsupported_format;
         }
         break;
@@ -1238,10 +1238,10 @@ static SDL_bool LoadAIFFMusic(WAV_Music *wave)
         wave->encoding = FLOAT_CODE;
         wave->decode = fetch_float64be;
         if (!is_AIFC)
-            spec->format = AUDIO_F32;
+            spec->format = SDL_AUDIO_F32;
         else switch (compressionType) {
         case fl64:
-            spec->format = AUDIO_F32;
+            spec->format = SDL_AUDIO_F32;
             break;
         default: goto unsupported_format;
         }
diff --git a/src/codecs/music_wavpack.c b/src/codecs/music_wavpack.c
index bb723aa6..205d376e 100644
--- a/src/codecs/music_wavpack.c
+++ b/src/codecs/music_wavpack.c
@@ -404,13 +404,13 @@ static void *WAVPACK_CreateFromRW_internal(SDL_RWops *src1, SDL_RWops *src2, int
      * always in an int32_t[] buffer, in signed host-endian format. */
     switch (music->bps) {
     case 8:
-        format = AUDIO_U8;
+        format = SDL_AUDIO_U8;
         break;
     case 16:
-        format = AUDIO_S16SYS;
+        format = SDL_AUDIO_S16SYS;
         break;
     default:
-        format = (music->mode & MODE_FLOAT) ? AUDIO_F32SYS : AUDIO_S32SYS;
+        format = (music->mode & MODE_FLOAT) ? SDL_AUDIO_F32SYS : SDL_AUDIO_S32SYS;
         break;
     }
     music->stream = SDL_CreateAudioStream(format, (Uint8)music->channels, (int)music->samplerate / music->decimation,
diff --git a/src/codecs/music_xmp.c b/src/codecs/music_xmp.c
index bd44764c..cc3d48e3 100644
--- a/src/codecs/music_xmp.c
+++ b/src/codecs/music_xmp.c
@@ -243,8 +243,11 @@ void *XMP_CreateFromRW(SDL_RWops *src, int freesrc)
     }
 
     music->volume = MIX_MAX_VOLUME;
-    music->stream = SDL_CreateAudioStream(AUDIO_S16SYS, 2, music_spec.freq,
-                                       music_spec.format, music_spec.channels, music_spec.freq);
+    music->stream = SDL_CreateAudioStream(SDL_AUDIO_S16SYS, 2,
+                                          music_spec.freq,
+                                          music_spec.format,
+                                          music_spec.channels,
+                                          music_spec.freq);
     if (!music->stream) {
         goto e3;
     }
diff --git a/src/codecs/native_midi/native_midi_win32.c b/src/codecs/native_midi/native_midi_win32.c
index 82817169..241226f6 100644
--- a/src/codecs/native_midi/native_midi_win32.c
+++ b/src/codecs/native_midi/native_midi_win32.c
@@ -41,7 +41,7 @@ struct _NativeMidiSong {
   Uint16 ppqn;
   int Size;
   int NewPos;
-  SDL_mutex *mutex;
+  SDL_Mutex *mutex;
 };
 
 static UINT MidiDevice=MIDI_MAPPER;
diff --git a/src/codecs/timidity/timidity.c b/src/codecs/timidity/timidity.c
index e0325fd0..f5d81c5c 100644
--- a/src/codecs/timidity/timidity.c
+++ b/src/codecs/timidity/timidity.c
@@ -552,25 +552,25 @@ static void do_song_load(SDL_RWops *rw, SDL_AudioSpec *audio, MidiSong **out)
       goto fail;
   }
   switch (audio->format) {
-  case AUDIO_S8:
+  case SDL_AUDIO_S8 :
     song->write = timi_s32tos8;
     break;
-  case AUDIO_U8:
+  case SDL_AUDIO_U8 :
     song->write = timi_s32tou8;
     break;
-  case AUDIO_S16LSB:
+  case SDL_AUDIO_S16LSB :
     song->write = timi_s32tos16l;
     break;
-  case AUDIO_S16MSB:
+  case SDL_AUDIO_S16MSB :
     song->write = timi_s32tos16b;
     break;
-  case AUDIO_S32LSB:
+  case SDL_AUDIO_S32LSB :
     song->write = timi_s32tos32l;
     break;
-  case AUDIO_S32MSB:
+  case SDL_AUDIO_S32MSB :
     song->write = timi_s32tos32b;
     break;
-  case AUDIO_F32SYS:
+  case SDL_AUDIO_F32SYS :
     song->write = timi_s32tof32;
     break;
   default:
diff --git a/src/effect_position.c b/src/effect_position.c
index c8ce9391..f99cae6b 100644
--- a/src/effect_position.c
+++ b/src/effect_position.c
@@ -1340,7 +1340,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
     Mix_EffectFunc_t f = NULL;
 
     switch (format) {
-        case AUDIO_U8:
+        case SDL_AUDIO_U8 :
             switch (channels) {
             case 1:
             case 2:
@@ -1359,7 +1359,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_S8:
+        case SDL_AUDIO_S8 :
             switch (channels) {
             case 1:
             case 2:
@@ -1378,7 +1378,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_S16LSB:
+        case SDL_AUDIO_S16LSB :
             switch (channels) {
             case 1:
             case 2:
@@ -1396,7 +1396,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_S16MSB:
+        case SDL_AUDIO_S16MSB :
             switch (channels) {
             case 1:
             case 2:
@@ -1414,7 +1414,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_S32MSB:
+        case SDL_AUDIO_S32MSB :
             switch (channels) {
             case 1:
             case 2:
@@ -1432,7 +1432,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_S32LSB:
+        case SDL_AUDIO_S32LSB :
             switch (channels) {
             case 1:
             case 2:
@@ -1450,7 +1450,7 @@ static Mix_EffectFunc_t get_position_effect_func(Uint16 format, int channels)
             }
             break;
 
-        case AUDIO_F32SYS:
+        case SDL_AUDIO_F32SYS :
             switch (channels) {
             case 1:
             case 2: