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: