audio: Cleaned out most remaining `/* */` comments for `//` style.

Fully committing to it...!

This left SDL_wave.* alone for now, since there's a ton of comments in there
and this code hasn't changed much from SDL2 so far. But as SDL2 ages out a
little more, I'll likely switch this over, too.

build-scripts/gen_audio_channel_conversion.c

@@ -269,7 +269,7 @@ static void write_converter(const int fromchans, const int tochans)
            "\n", lowercase(fromstr), lowercase(tostr));
 
     if (convert_backwards) {  /* must convert backwards when growing the output in-place. */
-        printf("    /* convert backwards, since output is growing in-place. */\n");
+        printf("    // convert backwards, since output is growing in-place.\n");
         printf("    src += (num_frames-1)");
         if (fromchans != 1) {
             printf(" * %d", fromchans);
@@ -425,7 +425,7 @@ int main(void)
         "  3. This notice may not be removed or altered from any source distribution.\n"
         "*/\n"
         "\n"
-        "/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_channel_conversion.c */\n"
+        "// DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_channel_conversion.c\n"
         "\n"
         "\n"
         "typedef void (*SDL_AudioChannelConverter)(float *dst, const float *src, int num_frames);\n"

build-scripts/gen_audio_resampler_filter.c

@@ -25,7 +25,7 @@ Built with:
 
 gcc -o genfilter build-scripts/gen_audio_resampler_filter.c -lm && ./genfilter > src/audio/SDL_audio_resampler_filter.h
 
- */
+*/
 
 /*
    SDL's resampler uses a "bandlimited interpolation" algorithm:
@@ -128,7 +128,7 @@ int main(void)
         "  3. This notice may not be removed or altered from any source distribution.\n"
         "*/\n"
         "\n"
-        "/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c */\n"
+        "// DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c\n"
         "\n"
         "#define RESAMPLER_ZERO_CROSSINGS %d\n"
         "#define RESAMPLER_BITS_PER_SAMPLE %d\n"

src/audio/SDL_audio_channel_converters.h

@@ -19,7 +19,7 @@
   3. This notice may not be removed or altered from any source distribution.
 */
 
-/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_channel_conversion.c */
+// DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_channel_conversion.c
 
 
 typedef void (*SDL_AudioChannelConverter)(float *dst, const float *src, int num_frames);
@@ -30,7 +30,7 @@ static void SDL_ConvertMonoToStereo(float *dst, const float *src, int num_frames
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "stereo");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 2;
     for (i = num_frames; i; i--, src--, dst -= 2) {
@@ -47,7 +47,7 @@ static void SDL_ConvertMonoTo21(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "2.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 3;
     for (i = num_frames; i; i--, src--, dst -= 3) {
@@ -65,7 +65,7 @@ static void SDL_ConvertMonoToQuad(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "quad");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 4;
     for (i = num_frames; i; i--, src--, dst -= 4) {
@@ -84,7 +84,7 @@ static void SDL_ConvertMonoTo41(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "4.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 5;
     for (i = num_frames; i; i--, src--, dst -= 5) {
@@ -104,7 +104,7 @@ static void SDL_ConvertMonoTo51(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "5.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 6;
     for (i = num_frames; i; i--, src--, dst -= 6) {
@@ -125,7 +125,7 @@ static void SDL_ConvertMonoTo61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src--, dst -= 7) {
@@ -147,7 +147,7 @@ static void SDL_ConvertMonoTo71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("mono", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1);
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src--, dst -= 8) {
@@ -182,7 +182,7 @@ static void SDL_ConvertStereoTo21(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "2.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 3;
     for (i = num_frames; i; i--, src -= 2, dst -= 3) {
@@ -199,7 +199,7 @@ static void SDL_ConvertStereoToQuad(float *dst, const float *src, int num_frames
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "quad");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 4;
     for (i = num_frames; i; i--, src -= 2, dst -= 4) {
@@ -217,7 +217,7 @@ static void SDL_ConvertStereoTo41(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "4.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 5;
     for (i = num_frames; i; i--, src -= 2, dst -= 5) {
@@ -236,7 +236,7 @@ static void SDL_ConvertStereoTo51(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "5.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 6;
     for (i = num_frames; i; i--, src -= 2, dst -= 6) {
@@ -256,7 +256,7 @@ static void SDL_ConvertStereoTo61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src -= 2, dst -= 7) {
@@ -277,7 +277,7 @@ static void SDL_ConvertStereoTo71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("stereo", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 2;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 2, dst -= 8) {
@@ -325,7 +325,7 @@ static void SDL_Convert21ToQuad(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("2.1", "quad");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 3;
     dst += (num_frames-1) * 4;
     for (i = num_frames; i; i--, src -= 3, dst -= 4) {
@@ -344,7 +344,7 @@ static void SDL_Convert21To41(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("2.1", "4.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 3;
     dst += (num_frames-1) * 5;
     for (i = num_frames; i; i--, src -= 3, dst -= 5) {
@@ -363,7 +363,7 @@ static void SDL_Convert21To51(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("2.1", "5.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 3;
     dst += (num_frames-1) * 6;
     for (i = num_frames; i; i--, src -= 3, dst -= 6) {
@@ -383,7 +383,7 @@ static void SDL_Convert21To61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("2.1", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 3;
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src -= 3, dst -= 7) {
@@ -404,7 +404,7 @@ static void SDL_Convert21To71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("2.1", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 3;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 3, dst -= 8) {
@@ -469,7 +469,7 @@ static void SDL_ConvertQuadTo41(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("quad", "4.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 4;
     dst += (num_frames-1) * 5;
     for (i = num_frames; i; i--, src -= 4, dst -= 5) {
@@ -488,7 +488,7 @@ static void SDL_ConvertQuadTo51(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("quad", "5.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 4;
     dst += (num_frames-1) * 6;
     for (i = num_frames; i; i--, src -= 4, dst -= 6) {
@@ -508,7 +508,7 @@ static void SDL_ConvertQuadTo61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("quad", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 4;
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src -= 4, dst -= 7) {
@@ -531,7 +531,7 @@ static void SDL_ConvertQuadTo71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("quad", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 4;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 4, dst -= 8) {
@@ -613,7 +613,7 @@ static void SDL_Convert41To51(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("4.1", "5.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 5;
     dst += (num_frames-1) * 6;
     for (i = num_frames; i; i--, src -= 5, dst -= 6) {
@@ -633,7 +633,7 @@ static void SDL_Convert41To61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("4.1", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 5;
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src -= 5, dst -= 7) {
@@ -656,7 +656,7 @@ static void SDL_Convert41To71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("4.1", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 5;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 5, dst -= 8) {
@@ -758,7 +758,7 @@ static void SDL_Convert51To61(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("5.1", "6.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 6;
     dst += (num_frames-1) * 7;
     for (i = num_frames; i; i--, src -= 6, dst -= 7) {
@@ -781,7 +781,7 @@ static void SDL_Convert51To71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("5.1", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 6;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 6, dst -= 8) {
@@ -911,7 +911,7 @@ static void SDL_Convert61To71(float *dst, const float *src, int num_frames)
 
     LOG_DEBUG_AUDIO_CONVERT("6.1", "7.1");
 
-    /* convert backwards, since output is growing in-place. */
+    // convert backwards, since output is growing in-place.
     src += (num_frames-1) * 7;
     dst += (num_frames-1) * 8;
     for (i = num_frames; i; i--, src -= 7, dst -= 8) {

src/audio/SDL_audio_resampler_filter.h

@@ -19,7 +19,7 @@
   3. This notice may not be removed or altered from any source distribution.
 */
 
-/* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c */
+// DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_resampler_filter.c
 
 #define RESAMPLER_ZERO_CROSSINGS 5
 #define RESAMPLER_BITS_PER_SAMPLE 16

src/audio/SDL_audiodev.c

@@ -20,14 +20,14 @@
 */
 #include "SDL_internal.h"
 
-/* Get the name of the audio device we use for output */
+// Get the name of the audio device we use for output
 
 #if defined(SDL_AUDIO_DRIVER_NETBSD) || defined(SDL_AUDIO_DRIVER_OSS)
 
 #include <fcntl.h>
 #include <sys/types.h>
 #include <sys/stat.h>
-#include <unistd.h> /* For close() */
+#include <unistd.h> // For close()
 
 #include "SDL_audiodev_c.h"
 
@@ -84,7 +84,7 @@ static void SDL_EnumUnixAudioDevices_Internal(const SDL_bool iscapture, const SD
         test = test_stub;
     }
 
-    /* Figure out what our audio device is */
+    // Figure out what our audio device is
     audiodev = SDL_getenv("SDL_PATH_DSP");
     if (audiodev == NULL) {
         audiodev = SDL_getenv("AUDIODEV");
@@ -95,7 +95,7 @@ static void SDL_EnumUnixAudioDevices_Internal(const SDL_bool iscapture, const SD
         } else {
             struct stat sb;
 
-            /* Added support for /dev/sound/\* in Linux 2.4 */
+            // Added support for /dev/sound/\* in Linux 2.4
             if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) && ((stat(SDL_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode))) {
                 audiodev = SDL_PATH_DEV_DSP24;
             } else {
@@ -122,4 +122,4 @@ void SDL_EnumUnixAudioDevices(const SDL_bool classic, SDL_bool (*test)(int))
     SDL_EnumUnixAudioDevices_Internal(SDL_FALSE, classic, test);
 }
 
-#endif /* Audio driver selection */
+#endif // Audio device selection

src/audio/SDL_audiodev_c.h

@@ -25,8 +25,8 @@
 #include "SDL_internal.h"
 #include "SDL_sysaudio.h"
 
-/* Open the audio device for playback, and don't block if busy */
-/* #define USE_BLOCKING_WRITES */
+// Open the audio device for playback, and don't block if busy
+//#define USE_BLOCKING_WRITES
 
 #ifdef USE_BLOCKING_WRITES
 #define OPEN_FLAGS_OUTPUT O_WRONLY
@@ -38,4 +38,4 @@
 
 extern void SDL_EnumUnixAudioDevices(const SDL_bool classic, SDL_bool (*test)(int));
 
-#endif /* SDL_audiodev_c_h_ */
+#endif // SDL_audiodev_c_h_

src/audio/SDL_audioresample.c

@@ -23,13 +23,13 @@
 #include "SDL_sysaudio.h"
 #include "SDL_audioresample.h"
 
-/* SDL's resampler uses a "bandlimited interpolation" algorithm:
-     https://ccrma.stanford.edu/~jos/resample/ */
+// SDL's resampler uses a "bandlimited interpolation" algorithm:
+//     https://ccrma.stanford.edu/~jos/resample/
 
 #include "SDL_audio_resampler_filter.h"
 
-/* For a given srcpos, `srcpos + frame` are sampled, where `-RESAMPLER_ZERO_CROSSINGS < frame <= RESAMPLER_ZERO_CROSSINGS`.
- * Note, when upsampling, it is also possible to start sampling from `srcpos = -1`. */
+// For a given srcpos, `srcpos + frame` are sampled, where `-RESAMPLER_ZERO_CROSSINGS < frame <= RESAMPLER_ZERO_CROSSINGS`.
+// Note, when upsampling, it is also possible to start sampling from `srcpos = -1`.
 #define RESAMPLER_MAX_PADDING_FRAMES (RESAMPLER_ZERO_CROSSINGS + 1)
 
 #define RESAMPLER_FILTER_INTERP_BITS  (32 - RESAMPLER_BITS_PER_ZERO_CROSSING)

src/audio/SDL_audiotypecvt.c

@@ -22,32 +22,31 @@
 
 #include "SDL_audio_c.h"
 
-/* TODO: NEON is disabled until https://github.com/libsdl-org/SDL/issues/8352
- * can be fixed */
+// TODO: NEON is disabled until https://github.com/libsdl-org/SDL/issues/8352 can be fixed
 #undef SDL_NEON_INTRINSICS
 
 #ifndef SDL_CPUINFO_DISABLED
 #if defined(__x86_64__) && defined(SDL_SSE2_INTRINSICS)
-#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* x86_64 guarantees SSE2. */
+#define NEED_SCALAR_CONVERTER_FALLBACKS 0 // x86_64 guarantees SSE2.
 #elif defined(__MACOS__) && defined(SDL_SSE2_INTRINSICS)
-#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* macOS/Intel guarantees SSE2. */
+#define NEED_SCALAR_CONVERTER_FALLBACKS 0 // macOS/Intel guarantees SSE2.
 #elif defined(__ARM_ARCH) && (__ARM_ARCH >= 8) && defined(SDL_NEON_INTRINSICS)
-#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* ARMv8+ promise NEON. */
+#define NEED_SCALAR_CONVERTER_FALLBACKS 0 // ARMv8+ promise NEON.
 #elif defined(__APPLE__) && defined(__ARM_ARCH) && (__ARM_ARCH >= 7) && defined(SDL_NEON_INTRINSICS)
-#define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* All Apple ARMv7 chips promise NEON support. */
+#define NEED_SCALAR_CONVERTER_FALLBACKS 0 // All Apple ARMv7 chips promise NEON support.
 #endif
 #endif
 
-/* Set to zero if platform is guaranteed to use a SIMD codepath here. */
+// Set to zero if platform is guaranteed to use a SIMD codepath here.
 #if !defined(NEED_SCALAR_CONVERTER_FALLBACKS) || defined(SDL_CPUINFO_DISABLED)
 #define NEED_SCALAR_CONVERTER_FALLBACKS 1
 #endif
 
-#define DIVBY2147483648 0.0000000004656612873077392578125f /* 0x1p-31f */
+#define DIVBY2147483648 0.0000000004656612873077392578125f // 0x1p-31f
 
 #if NEED_SCALAR_CONVERTER_FALLBACKS
 
-/* This code requires that floats are in the IEEE-754 binary32 format */
+// This code requires that floats are in the IEEE-754 binary32 format
 SDL_COMPILE_TIME_ASSERT(float_bits, sizeof(float) == sizeof(Uint32));
 
 union float_bits {
@@ -111,7 +110,7 @@ static void SDL_Convert_S32_to_F32_Scalar(float *dst, const Sint32 *src, int num
     }
 }
 
-/* Create a bit-mask based on the sign-bit. Should optimize to a single arithmetic-shift-right */
+// Create a bit-mask based on the sign-bit. Should optimize to a single arithmetic-shift-right
 #define SIGNMASK(x) (Uint32)(0u - ((Uint32)(x) >> 31))
 
 static void SDL_Convert_F32_to_S8_Scalar(Sint8 *dst, const float *src, int num_samples)
@@ -202,7 +201,7 @@ static void SDL_Convert_F32_to_S32_Scalar(Sint32 *dst, const float *src, int num
 
 #undef SIGNMASK
 
-#endif /* NEED_SCALAR_CONVERTER_FALLBACKS */
+#endif // NEED_SCALAR_CONVERTER_FALLBACKS
 
 #ifdef SDL_SSE2_INTRINSICS
 static void SDL_TARGETING("sse2") SDL_Convert_S8_to_F32_SSE2(float *dst, const Sint8 *src, int num_samples)
@@ -324,7 +323,7 @@ static void SDL_TARGETING("sse2") SDL_Convert_S32_to_F32_SSE2(float *dst, const
 {
     int i = num_samples;
 
-    /* dst[i] = f32(src[i]) / f32(0x80000000) */
+    // dst[i] = f32(src[i]) / f32(0x80000000)
     const __m128 scaler = _mm_set1_ps(DIVBY2147483648);
 
     LOG_DEBUG_AUDIO_CONVERT("S32", "F32 (using SSE2)");
@@ -543,9 +542,9 @@ static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S32_SSE2(Sint32 *dst, const
 #endif
 
 #ifdef SDL_NEON_INTRINSICS
-#define DIVBY128     0.0078125f /* 0x1p-7f */
-#define DIVBY32768   0.000030517578125f /* 0x1p-15f */
-#define DIVBY8388607 0.00000011920930376163766f /* 0x1.000002p-23f */
+#define DIVBY128     0.0078125f // 0x1p-7f
+#define DIVBY32768   0.000030517578125f // 0x1p-15f
+#define DIVBY8388607 0.00000011920930376163766f // 0x1.000002p-23f
 
 static void SDL_Convert_S8_to_F32_NEON(float *dst, const Sint8 *src, int num_samples)
 {
@@ -556,25 +555,25 @@ static void SDL_Convert_S8_to_F32_NEON(float *dst, const Sint8 *src, int num_sam
     src += num_samples - 1;
     dst += num_samples - 1;
 
-    /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
+    // Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src)
     for (i = num_samples; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) {
         *dst = ((float)*src) * DIVBY128;
     }
 
     src -= 15;
-    dst -= 15; /* adjust to read NEON blocks from the start. */
+    dst -= 15; // adjust to read NEON blocks from the start.
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const int8_t *mmsrc = (const int8_t *)src;
         const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
-        while (i >= 16) {                                            /* 16 * 8-bit */
-            const int8x16_t bytes = vld1q_s8(mmsrc);                 /* get 16 sint8 into a NEON register. */
-            const int16x8_t int16hi = vmovl_s8(vget_high_s8(bytes)); /* convert top 8 bytes to 8 int16 */
-            const int16x8_t int16lo = vmovl_s8(vget_low_s8(bytes));  /* convert bottom 8 bytes to 8 int16 */
-            /* split int16 to two int32, then convert to float, then multiply to normalize, store. */
+        while (i >= 16) {                                            // 16 * 8-bit
+            const int8x16_t bytes = vld1q_s8(mmsrc);                 // get 16 sint8 into a NEON register.
+            const int16x8_t int16hi = vmovl_s8(vget_high_s8(bytes)); // convert top 8 bytes to 8 int16
+            const int16x8_t int16lo = vmovl_s8(vget_low_s8(bytes));  // convert bottom 8 bytes to 8 int16
+            // split int16 to two int32, then convert to float, then multiply to normalize, store.
             vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16lo))), divby128));
             vst1q_f32(dst + 4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16lo))), divby128));
             vst1q_f32(dst + 8, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16hi))), divby128));
@@ -588,9 +587,9 @@ static void SDL_Convert_S8_to_F32_NEON(float *dst, const Sint8 *src, int num_sam
     }
 
     src += 15;
-    dst += 15; /* adjust for any scalar finishing. */
+    dst += 15; // adjust for any scalar finishing.
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         *dst = ((float)*src) * DIVBY128;
         i--;
@@ -608,26 +607,26 @@ static void SDL_Convert_U8_to_F32_NEON(float *dst, const Uint8 *src, int num_sam
     src += num_samples - 1;
     dst += num_samples - 1;
 
-    /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
+    // Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src)
     for (i = num_samples; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) {
         *dst = (((float)*src) * DIVBY128) - 1.0f;
     }
 
     src -= 15;
-    dst -= 15; /* adjust to read NEON blocks from the start. */
+    dst -= 15; // adjust to read NEON blocks from the start.
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const uint8_t *mmsrc = (const uint8_t *)src;
         const float32x4_t divby128 = vdupq_n_f32(DIVBY128);
         const float32x4_t negone = vdupq_n_f32(-1.0f);
-        while (i >= 16) {                                              /* 16 * 8-bit */
-            const uint8x16_t bytes = vld1q_u8(mmsrc);                  /* get 16 uint8 into a NEON register. */
-            const uint16x8_t uint16hi = vmovl_u8(vget_high_u8(bytes)); /* convert top 8 bytes to 8 uint16 */
-            const uint16x8_t uint16lo = vmovl_u8(vget_low_u8(bytes));  /* convert bottom 8 bytes to 8 uint16 */
-            /* split uint16 to two uint32, then convert to float, then multiply to normalize, subtract to adjust for sign, store. */
+        while (i >= 16) {                                              // 16 * 8-bit
+            const uint8x16_t bytes = vld1q_u8(mmsrc);                  // get 16 uint8 into a NEON register.
+            const uint16x8_t uint16hi = vmovl_u8(vget_high_u8(bytes)); // convert top 8 bytes to 8 uint16
+            const uint16x8_t uint16lo = vmovl_u8(vget_low_u8(bytes));  // convert bottom 8 bytes to 8 uint16
+            // split uint16 to two uint32, then convert to float, then multiply to normalize, subtract to adjust for sign, store.
             vst1q_f32(dst, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16lo))), divby128));
             vst1q_f32(dst + 4, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16lo))), divby128));
             vst1q_f32(dst + 8, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16hi))), divby128));
@@ -641,9 +640,9 @@ static void SDL_Convert_U8_to_F32_NEON(float *dst, const Uint8 *src, int num_sam
     }
 
     src += 15;
-    dst += 15; /* adjust for any scalar finishing. */
+    dst += 15; // adjust for any scalar finishing.
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         *dst = (((float)*src) * DIVBY128) - 1.0f;
         i--;
@@ -661,22 +660,22 @@ static void SDL_Convert_S16_to_F32_NEON(float *dst, const Sint16 *src, int num_s
     src += num_samples - 1;
     dst += num_samples - 1;
 
-    /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */
+    // Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src)
     for (i = num_samples; i && (((size_t)(dst - 7)) & 15); --i, --src, --dst) {
         *dst = ((float)*src) * DIVBY32768;
     }
 
     src -= 7;
-    dst -= 7; /* adjust to read NEON blocks from the start. */
+    dst -= 7; // adjust to read NEON blocks from the start.
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768);
-        while (i >= 8) {                                            /* 8 * 16-bit */
-            const int16x8_t ints = vld1q_s16((int16_t const *)src); /* get 8 sint16 into a NEON register. */
-            /* split int16 to two int32, then convert to float, then multiply to normalize, store. */
+        while (i >= 8) {                                            // 8 * 16-bit
+            const int16x8_t ints = vld1q_s16((int16_t const *)src); // get 8 sint16 into a NEON register.
+            // split int16 to two int32, then convert to float, then multiply to normalize, store.
             vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(ints))), divby32768));
             vst1q_f32(dst + 4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(ints))), divby32768));
             i -= 8;
@@ -686,9 +685,9 @@ static void SDL_Convert_S16_to_F32_NEON(float *dst, const Sint16 *src, int num_s
     }
 
     src += 7;
-    dst += 7; /* adjust for any scalar finishing. */
+    dst += 7; // adjust for any scalar finishing.
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         *dst = ((float)*src) * DIVBY32768;
         i--;
@@ -703,20 +702,20 @@ static void SDL_Convert_S32_to_F32_NEON(float *dst, const Sint32 *src, int num_s
 
     LOG_DEBUG_AUDIO_CONVERT("S32", "F32 (using NEON)");
 
-    /* Get dst aligned to 16 bytes */
+    // Get dst aligned to 16 bytes
     for (i = num_samples; i && (((size_t)dst) & 15); --i, ++src, ++dst) {
         *dst = ((float)(*src >> 8)) * DIVBY8388607;
     }
 
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t divby8388607 = vdupq_n_f32(DIVBY8388607);
         const int32_t *mmsrc = (const int32_t *)src;
-        while (i >= 4) { /* 4 * sint32 */
-            /* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */
+        while (i >= 4) { // 4 * sint32
+            // shift out lowest bits so int fits in a float32. Small precision loss, but much faster.
             vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vshrq_n_s32(vld1q_s32(mmsrc), 8)), divby8388607));
             i -= 4;
             mmsrc += 4;
@@ -725,7 +724,7 @@ static void SDL_Convert_S32_to_F32_NEON(float *dst, const Sint32 *src, int num_s
         src = (const Sint32 *)mmsrc;
     }
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         *dst = ((float)(*src >> 8)) * DIVBY8388607;
         i--;
@@ -740,7 +739,7 @@ static void SDL_Convert_F32_to_S8_NEON(Sint8 *dst, const float *src, int num_sam
 
     LOG_DEBUG_AUDIO_CONVERT("F32", "S8 (using NEON)");
 
-    /* Get dst aligned to 16 bytes */
+    // Get dst aligned to 16 bytes
     for (i = num_samples; i && (((size_t)dst) & 15); --i, ++src, ++dst) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -754,21 +753,21 @@ static void SDL_Convert_F32_to_S8_NEON(Sint8 *dst, const float *src, int num_sam
 
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t one = vdupq_n_f32(1.0f);
         const float32x4_t negone = vdupq_n_f32(-1.0f);
         const float32x4_t mulby127 = vdupq_n_f32(127.0f);
         int8_t *mmdst = (int8_t *)dst;
-        while (i >= 16) {                                                                                                       /* 16 * float32 */
-            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby127));      /* load 4 floats, clamp, convert to sint32 */
-            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
-            const int32x4_t ints3 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), mulby127));  /* load 4 floats, clamp, convert to sint32 */
-            const int32x4_t ints4 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */
-            const int8x8_t i8lo = vmovn_s16(vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2)));                                  /* narrow to sint16, combine, narrow to sint8 */
-            const int8x8_t i8hi = vmovn_s16(vcombine_s16(vmovn_s32(ints3), vmovn_s32(ints4)));                                  /* narrow to sint16, combine, narrow to sint8 */
-            vst1q_s8(mmdst, vcombine_s8(i8lo, i8hi));                                                                           /* combine to int8x16_t, store out */
+        while (i >= 16) {                                                                                                       // 16 * float32
+            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby127));      // load 4 floats, clamp, convert to sint32
+            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby127));  // load 4 floats, clamp, convert to sint32
+            const int32x4_t ints3 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), mulby127));  // load 4 floats, clamp, convert to sint32
+            const int32x4_t ints4 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), mulby127)); // load 4 floats, clamp, convert to sint32
+            const int8x8_t i8lo = vmovn_s16(vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2)));                                  // narrow to sint16, combine, narrow to sint8
+            const int8x8_t i8hi = vmovn_s16(vcombine_s16(vmovn_s32(ints3), vmovn_s32(ints4)));                                  // narrow to sint16, combine, narrow to sint8
+            vst1q_s8(mmdst, vcombine_s8(i8lo, i8hi));                                                                           // combine to int8x16_t, store out
             i -= 16;
             src += 16;
             mmdst += 16;
@@ -776,7 +775,7 @@ static void SDL_Convert_F32_to_S8_NEON(Sint8 *dst, const float *src, int num_sam
         dst = (Sint8 *)mmdst;
     }
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -798,7 +797,7 @@ static void SDL_Convert_F32_to_U8_NEON(Uint8 *dst, const float *src, int num_sam
 
     LOG_DEBUG_AUDIO_CONVERT("F32", "U8 (using NEON)");
 
-    /* Get dst aligned to 16 bytes */
+    // Get dst aligned to 16 bytes
     for (i = num_samples; i && (((size_t)dst) & 15); --i, ++src, ++dst) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -812,21 +811,21 @@ static void SDL_Convert_F32_to_U8_NEON(Uint8 *dst, const float *src, int num_sam
 
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t one = vdupq_n_f32(1.0f);
         const float32x4_t negone = vdupq_n_f32(-1.0f);
         const float32x4_t mulby127 = vdupq_n_f32(127.0f);
         uint8_t *mmdst = (uint8_t *)dst;
-        while (i >= 16) {                                                                                                                         /* 16 * float32 */
-            const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby127));      /* load 4 floats, clamp, convert to uint32 */
-            const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
-            const uint32x4_t uints3 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), one), mulby127));  /* load 4 floats, clamp, convert to uint32 */
-            const uint32x4_t uints4 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */
-            const uint8x8_t ui8lo = vmovn_u16(vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2)));                                                /* narrow to uint16, combine, narrow to uint8 */
-            const uint8x8_t ui8hi = vmovn_u16(vcombine_u16(vmovn_u32(uints3), vmovn_u32(uints4)));                                                /* narrow to uint16, combine, narrow to uint8 */
-            vst1q_u8(mmdst, vcombine_u8(ui8lo, ui8hi));                                                                                           /* combine to uint8x16_t, store out */
+        while (i >= 16) {                                                                                                                         // 16 * float32
+            const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby127));      // load 4 floats, clamp, convert to uint32
+            const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), one), mulby127));  // load 4 floats, clamp, convert to uint32
+            const uint32x4_t uints3 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), one), mulby127));  // load 4 floats, clamp, convert to uint32
+            const uint32x4_t uints4 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), one), mulby127)); // load 4 floats, clamp, convert to uint32
+            const uint8x8_t ui8lo = vmovn_u16(vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2)));                                                // narrow to uint16, combine, narrow to uint8
+            const uint8x8_t ui8hi = vmovn_u16(vcombine_u16(vmovn_u32(uints3), vmovn_u32(uints4)));                                                // narrow to uint16, combine, narrow to uint8
+            vst1q_u8(mmdst, vcombine_u8(ui8lo, ui8hi));                                                                                           // combine to uint8x16_t, store out
             i -= 16;
             src += 16;
             mmdst += 16;
@@ -835,7 +834,7 @@ static void SDL_Convert_F32_to_U8_NEON(Uint8 *dst, const float *src, int num_sam
         dst = (Uint8 *)mmdst;
     }
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -857,7 +856,7 @@ static void SDL_Convert_F32_to_S16_NEON(Sint16 *dst, const float *src, int num_s
 
     LOG_DEBUG_AUDIO_CONVERT("F32", "S16 (using NEON)");
 
-    /* Get dst aligned to 16 bytes */
+    // Get dst aligned to 16 bytes
     for (i = num_samples; i && (((size_t)dst) & 15); --i, ++src, ++dst) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -871,17 +870,17 @@ static void SDL_Convert_F32_to_S16_NEON(Sint16 *dst, const float *src, int num_s
 
     SDL_assert(!i || !(((size_t)dst) & 15));
 
-    /* Make sure src is aligned too. */
+    // Make sure src is aligned too.
     if (!(((size_t)src) & 15)) {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t one = vdupq_n_f32(1.0f);
         const float32x4_t negone = vdupq_n_f32(-1.0f);
         const float32x4_t mulby32767 = vdupq_n_f32(32767.0f);
         int16_t *mmdst = (int16_t *)dst;
-        while (i >= 8) {                                                                                                         /* 8 * float32 */
-            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby32767));     /* load 4 floats, clamp, convert to sint32 */
-            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */
-            vst1q_s16(mmdst, vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2)));                                                  /* narrow to sint16, combine, store out. */
+        while (i >= 8) {                                                                                                         // 8 * float32
+            const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby32767));     // load 4 floats, clamp, convert to sint32
+            const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby32767)); // load 4 floats, clamp, convert to sint32
+            vst1q_s16(mmdst, vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2)));                                                  // narrow to sint16, combine, store out.
             i -= 8;
             src += 8;
             mmdst += 8;
@@ -889,7 +888,7 @@ static void SDL_Convert_F32_to_S16_NEON(Sint16 *dst, const float *src, int num_s
         dst = (Sint16 *)mmdst;
     }
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -911,7 +910,7 @@ static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_s
 
     LOG_DEBUG_AUDIO_CONVERT("F32", "S32 (using NEON)");
 
-    /* Get dst aligned to 16 bytes */
+    // Get dst aligned to 16 bytes
     for (i = num_samples; i && (((size_t)dst) & 15); --i, ++src, ++dst) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -927,12 +926,12 @@ static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_s
     SDL_assert(!i || !(((size_t)src) & 15));
 
     {
-        /* Aligned! Do NEON blocks as long as we have 16 bytes available. */
+        // Aligned! Do NEON blocks as long as we have 16 bytes available.
         const float32x4_t one = vdupq_n_f32(1.0f);
         const float32x4_t negone = vdupq_n_f32(-1.0f);
         const float32x4_t mulby8388607 = vdupq_n_f32(8388607.0f);
         int32_t *mmdst = (int32_t *)dst;
-        while (i >= 4) { /* 4 * float32 */
+        while (i >= 4) { // 4 * float32
             vst1q_s32(mmdst, vshlq_n_s32(vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby8388607)), 8));
             i -= 4;
             src += 4;
@@ -941,7 +940,7 @@ static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_s
         dst = (Sint32 *)mmdst;
     }
 
-    /* Finish off any leftovers with scalar operations. */
+    // Finish off any leftovers with scalar operations.
     while (i) {
         const float sample = *src;
         if (sample >= 1.0f) {
@@ -958,7 +957,7 @@ static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_s
 }
 #endif
 
-/* Function pointers set to a CPU-specific implementation. */
+// Function pointers set to a CPU-specific implementation.
 void (*SDL_Convert_S8_to_F32)(float *dst, const Sint8 *src, int num_samples) = NULL;
 void (*SDL_Convert_U8_to_F32)(float *dst, const Uint8 *src, int num_samples) = NULL;
 void (*SDL_Convert_S16_to_F32)(float *dst, const Sint16 *src, int num_samples) = NULL;

src/audio/SDL_mixer.c

@@ -20,7 +20,7 @@
 */
 #include "SDL_internal.h"
 
-/* This provides the default mixing callback for the SDL audio routines */
+// This provides the default mixing callback for the SDL audio routines
 
 #include "SDL_sysaudio.h"
 
@@ -77,12 +77,12 @@ static const Uint8 mix8[] = {
     0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF
 };
 
-/* The volume ranges from 0 - 128 */
+// The volume ranges from 0 - 128
 #define ADJUST_VOLUME(type, s, v) ((s) = (type)(((s) * (v)) / SDL_MIX_MAXVOLUME))
 #define ADJUST_VOLUME_U8(s, v)    ((s) = (Uint8)(((((s) - 128) * (v)) / SDL_MIX_MAXVOLUME) + 128))
 
 
-/* !!! FIXME: this needs some SIMD magic. */
+// !!! FIXME: this needs some SIMD magic.
 
 int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
                         Uint32 len, int volume)
@@ -237,7 +237,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
         float *dst32 = (float *)dst;
         float src1, src2;
         double dst_sample;
-        /* !!! FIXME: are these right? */
+        // !!! FIXME: are these right?
         const double max_audioval = 3.402823466e+38F;
         const double min_audioval = -3.402823466e+38F;
 
@@ -265,7 +265,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
         float *dst32 = (float *)dst;
         float src1, src2;
         double dst_sample;
-        /* !!! FIXME: are these right? */
+        // !!! FIXME: are these right?
         const double max_audioval = 3.402823466e+38F;
         const double min_audioval = -3.402823466e+38F;
 
@@ -285,7 +285,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format,
         }
     } break;
 
-    default: /* If this happens... FIXME! */
+    default: // If this happens... FIXME!
         return SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
     }
 

src/audio/aaudio/SDL_aaudio.h

@@ -35,4 +35,4 @@ void AAUDIO_PauseDevices(void);
 
 #endif
 
-#endif /* SDL_aaudio_h_ */
+#endif // SDL_aaudio_h_

src/audio/aaudio/SDL_aaudiofuncs.h

@@ -33,18 +33,18 @@ SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSharingMode, (AAudioStreamBuilder *
 SDL_PROC(void, AAudioStreamBuilder_setDirection, (AAudioStreamBuilder * builder, aaudio_direction_t direction))
 SDL_PROC_UNUSED(void, AAudioStreamBuilder_setBufferCapacityInFrames, (AAudioStreamBuilder * builder, int32_t numFrames))
 SDL_PROC(void, AAudioStreamBuilder_setPerformanceMode, (AAudioStreamBuilder * builder, aaudio_performance_mode_t mode))
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder * builder, aaudio_usage_t usage))                                         /* API 28 */
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder * builder, aaudio_content_type_t contentType))                      /* API 28 */
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder * builder, aaudio_input_preset_t inputPreset))                      /* API 28 */
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder * builder, aaudio_allowed_capture_policy_t capturePolicy)) /* API 29 */
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder * builder, aaudio_session_id_t sessionId))                            /* API 28 */
-SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder * builder, bool privacySensitive))                             /* API 30 */
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setUsage, (AAudioStreamBuilder * builder, aaudio_usage_t usage))                                         // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setContentType, (AAudioStreamBuilder * builder, aaudio_content_type_t contentType))                      // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setInputPreset, (AAudioStreamBuilder * builder, aaudio_input_preset_t inputPreset))                      // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setAllowedCapturePolicy, (AAudioStreamBuilder * builder, aaudio_allowed_capture_policy_t capturePolicy)) // API 29
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setSessionId, (AAudioStreamBuilder * builder, aaudio_session_id_t sessionId))                            // API 28
+SDL_PROC_UNUSED(void, AAudioStreamBuilder_setPrivacySensitive, (AAudioStreamBuilder * builder, bool privacySensitive))                             // API 30
 SDL_PROC(void, AAudioStreamBuilder_setDataCallback, (AAudioStreamBuilder * builder, AAudioStream_dataCallback callback, void *userData))
 SDL_PROC(void, AAudioStreamBuilder_setFramesPerDataCallback, (AAudioStreamBuilder * builder, int32_t numFrames))
 SDL_PROC(void, AAudioStreamBuilder_setErrorCallback, (AAudioStreamBuilder * builder, AAudioStream_errorCallback callback, void *userData))
 SDL_PROC(aaudio_result_t, AAudioStreamBuilder_openStream, (AAudioStreamBuilder * builder, AAudioStream **stream))
 SDL_PROC(aaudio_result_t, AAudioStreamBuilder_delete, (AAudioStreamBuilder * builder))
-SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_release, (AAudioStream * stream)) /* API 30 */
+SDL_PROC_UNUSED(aaudio_result_t, AAudioStream_release, (AAudioStream * stream)) // API 30
 SDL_PROC(aaudio_result_t, AAudioStream_close, (AAudioStream * stream))
 SDL_PROC(aaudio_result_t, AAudioStream_requestStart, (AAudioStream * stream))
 SDL_PROC(aaudio_result_t, AAudioStream_requestPause, (AAudioStream * stream))
@@ -70,13 +70,13 @@ SDL_PROC_UNUSED(aaudio_performance_mode_t, AAudioStream_getPerformanceMode, (AAu
 SDL_PROC_UNUSED(aaudio_direction_t, AAudioStream_getDirection, (AAudioStream * stream))
 SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesWritten, (AAudioStream * stream))
 SDL_PROC_UNUSED(int64_t, AAudioStream_getFramesRead, (AAudioStream * stream))
-SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream * stream)) /* API 28 */
+SDL_PROC_UNUSED(aaudio_session_id_t, AAudioStream_getSessionId, (AAudioStream * stream)) // API 28
 SDL_PROC(aaudio_result_t, AAudioStream_getTimestamp, (AAudioStream * stream, clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds))
-SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream * stream))                                 /* API 28 */
-SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream * stream))                    /* API 28 */
-SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream * stream))                    /* API 28 */
-SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, (AAudioStream * stream)) /* API 29 */
-SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream * stream))                                 /* API 30 */
+SDL_PROC_UNUSED(aaudio_usage_t, AAudioStream_getUsage, (AAudioStream * stream))                                 // API 28
+SDL_PROC_UNUSED(aaudio_content_type_t, AAudioStream_getContentType, (AAudioStream * stream))                    // API 28
+SDL_PROC_UNUSED(aaudio_input_preset_t, AAudioStream_getInputPreset, (AAudioStream * stream))                    // API 28
+SDL_PROC_UNUSED(aaudio_allowed_capture_policy_t, AAudioStream_getAllowedCapturePolicy, (AAudioStream * stream)) // API 29
+SDL_PROC_UNUSED(bool, AAudioStream_isPrivacySensitive, (AAudioStream * stream))                                 // API 30
 
 #undef SDL_PROC
 #undef SDL_PROC_UNUSED

src/audio/alsa/SDL_alsa_audio.h

@@ -29,14 +29,14 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The audio device handle */
+    // The audio device handle
     snd_pcm_t *pcm_handle;
 
-    /* Raw mixing buffer */
+    // Raw mixing buffer
     Uint8 *mixbuf;
 
-    /* swizzle function */
+    // swizzle function
     void (*swizzle_func)(SDL_AudioDevice *_this, void *buffer, Uint32 bufferlen);
 };
 
-#endif /* SDL_ALSA_audio_h_ */
+#endif // SDL_ALSA_audio_h_

src/audio/android/SDL_androidaudio.h

@@ -35,4 +35,4 @@ static void ANDROIDAUDIO_PauseDevices(void) {}
 
 #endif
 
-#endif /* SDL_androidaudio_h_ */
+#endif // SDL_androidaudio_h_

src/audio/coreaudio/SDL_coreaudio.h

@@ -39,7 +39,7 @@
 #include <AudioToolbox/AudioToolbox.h>
 #include <AudioUnit/AudioUnit.h>
 
-/* Things named "Master" were renamed to "Main" in macOS 12.0's SDK. */
+// Things named "Master" were renamed to "Main" in macOS 12.0's SDK.
 #ifdef MACOSX_COREAUDIO
 #include <AvailabilityMacros.h>
 #ifndef MAC_OS_VERSION_12_0
@@ -65,4 +65,4 @@ struct SDL_PrivateAudioData
 #endif
 };
 
-#endif /* SDL_coreaudio_h_ */
+#endif // SDL_coreaudio_h_

src/audio/coreaudio/SDL_coreaudio.m

@@ -79,9 +79,9 @@ static OSStatus DeviceAliveNotification(AudioObjectID devid, UInt32 num_addr, co
 
     SDL_bool dead = SDL_FALSE;
     if (error == kAudioHardwareBadDeviceError) {
-        dead = SDL_TRUE; /* device was unplugged. */
+        dead = SDL_TRUE; // device was unplugged.
     } else if ((error == kAudioHardwareNoError) && (!alive)) {
-        dead = SDL_TRUE; /* device died in some other way. */
+        dead = SDL_TRUE; // device died in some other way.
     }
 
     if (dead) {
@@ -263,7 +263,7 @@ static void COREAUDIO_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioD
 
     AudioObjectAddPropertyListener(kAudioObjectSystemObject, &devlist_address, DeviceListChangedNotification, NULL);
 
-    /* Get the Device ID */
+    // Get the Device ID
     UInt32 size;
     AudioDeviceID devid;
 
@@ -428,7 +428,7 @@ static SDL_bool UpdateAudioSession(SDL_AudioDevice *device, SDL_bool open, SDL_b
             /* AVAudioSessionCategoryOptionAllowBluetooth isn't available in the SDK for
                Apple TV but is still needed in order to output to Bluetooth devices.
              */
-            options |= 0x4; /* AVAudioSessionCategoryOptionAllowBluetooth; */
+            options |= 0x4; // AVAudioSessionCategoryOptionAllowBluetooth;
         }
         if (category == AVAudioSessionCategoryPlayAndRecord) {
             options |= AVAudioSessionCategoryOptionAllowBluetoothA2DP |
@@ -441,7 +441,7 @@ static SDL_bool UpdateAudioSession(SDL_AudioDevice *device, SDL_bool open, SDL_b
 
         if ([session respondsToSelector:@selector(setCategory:mode:options:error:)]) {
             if (![session.category isEqualToString:category] || session.categoryOptions != options) {
-                /* Stop the current session so we don't interrupt other application audio */
+                // Stop the current session so we don't interrupt other application audio
                 PauseAudioDevices();
                 [session setActive:NO error:nil];
                 session_active = SDL_FALSE;
@@ -454,7 +454,7 @@ static SDL_bool UpdateAudioSession(SDL_AudioDevice *device, SDL_bool open, SDL_b
             }
         } else {
             if (![session.category isEqualToString:category]) {
-                /* Stop the current session so we don't interrupt other application audio */
+                // Stop the current session so we don't interrupt other application audio
                 PauseAudioDevices();
                 [session setActive:NO error:nil];
                 session_active = SDL_FALSE;
@@ -498,7 +498,7 @@ static SDL_bool UpdateAudioSession(SDL_AudioDevice *device, SDL_bool open, SDL_b
             /* An interruption end notification is not guaranteed to be sent if
              we were previously interrupted... resuming if needed when the app
              becomes active seems to be the way to go. */
-            // Note: object: below needs to be nil, as otherwise it filters by the object, and session doesn't send foreground / active notifications.  johna
+            // Note: object: below needs to be nil, as otherwise it filters by the object, and session doesn't send foreground / active notifications.
             [center addObserver:listener
                        selector:@selector(applicationBecameActive:)
                            name:UIApplicationDidBecomeActiveNotification
@@ -717,7 +717,7 @@ static int PrepareAudioQueue(SDL_AudioDevice *device)
 
     SDL_UpdatedAudioDeviceFormat(device);  // make sure this is correct.
 
-    /* Set the channel layout for the audio queue */
+    // Set the channel layout for the audio queue
     AudioChannelLayout layout;
     SDL_zero(layout);
     switch (device->spec.channels) {
@@ -740,7 +740,7 @@ static int PrepareAudioQueue(SDL_AudioDevice *device)
         layout.mChannelLayoutTag = kAudioChannelLayoutTag_MPEG_5_1_A;
         break;
     case 7:
-        /* FIXME: Need to move channel[4] (BC) to channel[6] */
+        // FIXME: Need to move channel[4] (BC) to channel[6]
         layout.mChannelLayoutTag = kAudioChannelLayoutTag_MPEG_6_1_A;
         break;
     case 8:
@@ -763,7 +763,7 @@ static int PrepareAudioQueue(SDL_AudioDevice *device)
 
     int numAudioBuffers = 2;
     const double msecs = (device->sample_frames / ((double)device->spec.freq)) * 1000.0;
-    if (msecs < MINIMUM_AUDIO_BUFFER_TIME_MS) { /* use more buffers if we have a VERY small sample set. */
+    if (msecs < MINIMUM_AUDIO_BUFFER_TIME_MS) { // use more buffers if we have a VERY small sample set.
         numAudioBuffers = ((int)SDL_ceil(MINIMUM_AUDIO_BUFFER_TIME_MS / msecs) * 2);
     }
 
@@ -782,7 +782,7 @@ static int PrepareAudioQueue(SDL_AudioDevice *device)
         CHECK_RESULT("AudioQueueAllocateBuffer");
         SDL_memset(device->hidden->audioBuffer[i]->mAudioData, device->silence_value, device->hidden->audioBuffer[i]->mAudioDataBytesCapacity);
         device->hidden->audioBuffer[i]->mAudioDataByteSize = device->hidden->audioBuffer[i]->mAudioDataBytesCapacity;
-        /* !!! FIXME: should we use AudioQueueEnqueueBufferWithParameters and specify all frames be "trimmed" so these are immediately ready to refill with SDL callback data? */
+        // !!! FIXME: should we use AudioQueueEnqueueBufferWithParameters and specify all frames be "trimmed" so these are immediately ready to refill with SDL callback data?
         result = AudioQueueEnqueueBuffer(device->hidden->audioQueue, device->hidden->audioBuffer[i], 0, NULL);
         CHECK_RESULT("AudioQueueEnqueueBuffer");
     }
@@ -831,7 +831,7 @@ static int AudioQueueThreadEntry(void *arg)
 
 static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
 {
-    /* Initialize all variables that we clean on shutdown */
+    // Initialize all variables that we clean on shutdown
     device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
     if (device->hidden == NULL) {
         return SDL_OutOfMemory();
@@ -842,7 +842,7 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
         return -1;
     }
 
-    /* Stop CoreAudio from doing expensive audio rate conversion */
+    // Stop CoreAudio from doing expensive audio rate conversion
     @autoreleasepool {
         AVAudioSession *session = [AVAudioSession sharedInstance];
         [session setPreferredSampleRate:device->spec.freq error:nil];
@@ -856,12 +856,12 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
             device->spec.channels = session.preferredOutputNumberOfChannels;
         }
         #else
-        /* Calling setPreferredOutputNumberOfChannels seems to break audio output on iOS */
-        #endif /* TARGET_OS_TV */
+        // Calling setPreferredOutputNumberOfChannels seems to break audio output on iOS
+        #endif // TARGET_OS_TV
     }
     #endif
 
-    /* Setup a AudioStreamBasicDescription with the requested format */
+    // Setup a AudioStreamBasicDescription with the requested format
     AudioStreamBasicDescription *strdesc = &device->hidden->strdesc;
     strdesc->mFormatID = kAudioFormatLinearPCM;
     strdesc->mFormatFlags = kLinearPCMFormatFlagIsPacked;
@@ -872,7 +872,7 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
     const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
     SDL_AudioFormat test_format;
     while ((test_format = *(closefmts++)) != 0) {
-        /* CoreAudio handles most of SDL's formats natively. */
+        // CoreAudio handles most of SDL's formats natively.
         switch (test_format) {
         case SDL_AUDIO_U8:
         case SDL_AUDIO_S8:
@@ -890,7 +890,7 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
         break;
     }
 
-    if (!test_format) { /* shouldn't happen, but just in case... */
+    if (!test_format) { // shouldn't happen, but just in case...
         return SDL_SetError("%s: Unsupported audio format", "coreaudio");
     }
     device->spec.format = test_format;
@@ -914,10 +914,10 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device)
     }
 #endif
 
-    /* This has to init in a new thread so it can get its own CFRunLoop. :/ */
+    // This has to init in a new thread so it can get its own CFRunLoop. :/
     device->hidden->ready_semaphore = SDL_CreateSemaphore(0);
     if (!device->hidden->ready_semaphore) {
-        return -1; /* oh well. */
+        return -1; // oh well.
     }
 
     char threadname[64];
@@ -951,7 +951,6 @@ static void COREAUDIO_DeinitializeStart(void)
 
 static SDL_bool COREAUDIO_Init(SDL_AudioDriverImpl *impl)
 {
-    /* Set the function pointers */
     impl->OpenDevice = COREAUDIO_OpenDevice;
     impl->PlayDevice = COREAUDIO_PlayDevice;
     impl->GetDeviceBuf = COREAUDIO_GetDeviceBuf;
@@ -971,11 +970,11 @@ static SDL_bool COREAUDIO_Init(SDL_AudioDriverImpl *impl)
     impl->ProvidesOwnCallbackThread = SDL_TRUE;
     impl->HasCaptureSupport = SDL_TRUE;
 
-    return SDL_TRUE; /* this audio target is available. */
+    return SDL_TRUE;
 }
 
 AudioBootStrap COREAUDIO_bootstrap = {
     "coreaudio", "CoreAudio", COREAUDIO_Init, SDL_FALSE
 };
 
-#endif /* SDL_AUDIO_DRIVER_COREAUDIO */
+#endif // SDL_AUDIO_DRIVER_COREAUDIO

src/audio/disk/SDL_diskaudio.h

@@ -27,10 +27,10 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The file descriptor for the audio device */
+    // The file descriptor for the audio device
     SDL_RWops *io;
     Uint32 io_delay;
     Uint8 *mixbuf;
 };
 
-#endif /* SDL_diskaudio_h_ */
+#endif // SDL_diskaudio_h_

src/audio/dsp/SDL_dspaudio.c

@@ -24,8 +24,8 @@
 
 #ifdef SDL_AUDIO_DRIVER_OSS
 
-#include <stdio.h>  /* For perror() */
-#include <string.h> /* For strerror() */
+#include <stdio.h>  // For perror()
+#include <string.h> // For strerror()
 #include <errno.h>
 #include <unistd.h>
 #include <fcntl.h>
@@ -97,7 +97,7 @@ static int DSP_OpenDevice(SDL_AudioDevice *device)
         return SDL_SetError("Couldn't get audio format list");
     }
 
-    /* Try for a closest match on audio format */
+    // Try for a closest match on audio format
     int format = 0;
     SDL_AudioFormat test_format;
     const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
@@ -156,7 +156,7 @@ static int DSP_OpenDevice(SDL_AudioDevice *device)
     }
     device->spec.freq = value;
 
-    /* Calculate the final parameters for this audio specification */
+    // Calculate the final parameters for this audio specification
     SDL_UpdatedAudioDeviceFormat(device);
 
     /* Determine the power of two of the fragment size
@@ -168,9 +168,9 @@ static int DSP_OpenDevice(SDL_AudioDevice *device)
     while ((0x01U << frag_spec) < device->buffer_size) {
         frag_spec++;
     }
-    frag_spec |= 0x00020000; /* two fragments, for low latency */
+    frag_spec |= 0x00020000; // two fragments, for low latency
 
-    /* Set the audio buffering parameters */
+    // Set the audio buffering parameters
 #ifdef DEBUG_AUDIO
     fprintf(stderr, "Requesting %d fragments of size %d\n",
             (frag_spec >> 16), 1 << (frag_spec & 0xFFFF));
@@ -189,7 +189,7 @@ static int DSP_OpenDevice(SDL_AudioDevice *device)
     }
 #endif
 
-    /* Allocate mixing buffer */
+    // Allocate mixing buffer
     if (!device->iscapture) {
         device->hidden->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
         if (device->hidden->mixbuf == NULL) {
@@ -269,8 +269,8 @@ static void DSP_FlushCapture(SDL_AudioDevice *device)
 static SDL_bool InitTimeDevicesExist = SDL_FALSE;
 static SDL_bool look_for_devices_test(int fd)
 {
-    InitTimeDevicesExist = SDL_TRUE; /* note that _something_ exists. */
-    /* Don't add to the device list, we're just seeing if any devices exist. */
+    InitTimeDevicesExist = SDL_TRUE; // note that _something_ exists.
+    // Don't add to the device list, we're just seeing if any devices exist.
     return SDL_FALSE;
 }
 
@@ -280,10 +280,9 @@ static SDL_bool DSP_Init(SDL_AudioDriverImpl *impl)
     SDL_EnumUnixAudioDevices(SDL_FALSE, look_for_devices_test);
     if (!InitTimeDevicesExist) {
         SDL_SetError("dsp: No such audio device");
-        return SDL_FALSE; /* maybe try a different backend. */
+        return SDL_FALSE; // maybe try a different backend.
     }
 
-    /* Set the function pointers */
     impl->DetectDevices = DSP_DetectDevices;
     impl->OpenDevice = DSP_OpenDevice;
     impl->WaitDevice = DSP_WaitDevice;
@@ -296,11 +295,11 @@ static SDL_bool DSP_Init(SDL_AudioDriverImpl *impl)
 
     impl->HasCaptureSupport = SDL_TRUE;
 
-    return SDL_TRUE; /* this audio target is available. */
+    return SDL_TRUE;
 }
 
 AudioBootStrap DSP_bootstrap = {
     "dsp", "Open Sound System (/dev/dsp)", DSP_Init, SDL_FALSE
 };
 
-#endif /* SDL_AUDIO_DRIVER_OSS */
+#endif // SDL_AUDIO_DRIVER_OSS

src/audio/dsp/SDL_dspaudio.h

@@ -27,11 +27,11 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The file descriptor for the audio device */
+    // The file descriptor for the audio device
     int audio_fd;
 
-    /* Raw mixing buffer */
+    // Raw mixing buffer
     Uint8 *mixbuf;
 };
 
-#endif /* SDL_dspaudio_h_ */
+#endif // SDL_dspaudio_h_

src/audio/emscripten/SDL_emscriptenaudio.h

@@ -30,4 +30,4 @@ struct SDL_PrivateAudioData
     Uint8 *mixbuf;
 };
 
-#endif /* SDL_emscriptenaudio_h_ */
+#endif // SDL_emscriptenaudio_h_

src/audio/haiku/SDL_haikuaudio.h

@@ -32,4 +32,4 @@ struct SDL_PrivateAudioData
     int current_buffer_len;
 };
 
-#endif /* SDL_haikuaudio_h_ */
+#endif // SDL_haikuaudio_h_

src/audio/jack/SDL_jackaudio.c

@@ -53,19 +53,19 @@ static int load_jack_syms(void);
 static const char *jack_library = SDL_AUDIO_DRIVER_JACK_DYNAMIC;
 static void *jack_handle = NULL;
 
-/* !!! FIXME: this is copy/pasted in several places now */
+// !!! FIXME: this is copy/pasted in several places now
 static int load_jack_sym(const char *fn, void **addr)
 {
     *addr = SDL_LoadFunction(jack_handle, fn);
     if (*addr == NULL) {
-        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        // 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. */
+// cast funcs to char* first, to please GCC's strict aliasing rules.
 #define SDL_JACK_SYM(x)                                 \
     if (!load_jack_sym(#x, (void **)(char *)&JACK_##x)) \
     return -1
@@ -85,7 +85,7 @@ static int LoadJackLibrary(void)
         jack_handle = SDL_LoadObject(jack_library);
         if (jack_handle == NULL) {
             retval = -1;
-            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+            // Don't call SDL_SetError(): SDL_LoadObject already did.
         } else {
             retval = load_jack_syms();
             if (retval < 0) {
@@ -110,7 +110,7 @@ static int LoadJackLibrary(void)
     return 0;
 }
 
-#endif /* SDL_AUDIO_DRIVER_JACK_DYNAMIC */
+#endif // SDL_AUDIO_DRIVER_JACK_DYNAMIC
 
 static int load_jack_syms(void)
 {
@@ -137,7 +137,7 @@ static int load_jack_syms(void)
     return 0;
 }
 
-static void jackShutdownCallback(void *arg) /* JACK went away; device is lost. */
+static void jackShutdownCallback(void *arg) // JACK went away; device is lost.
 {
     SDL_AudioDeviceDisconnected((SDL_AudioDevice *)arg);
 }
@@ -294,7 +294,7 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
     int ports = 0;
     int i;
 
-    /* Initialize all variables that we clean on shutdown */
+    // Initialize all variables that we clean on shutdown
     device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
     if (device->hidden == NULL) {
         return SDL_OutOfMemory();
@@ -314,16 +314,16 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
     }
 
     while (devports[++ports]) {
-        /* spin to count devports */
+        // spin to count devports
     }
 
-    /* Filter out non-audio ports */
+    // Filter out non-audio ports
     audio_ports = SDL_calloc(ports, sizeof(*audio_ports));
     for (i = 0; i < ports; i++) {
         const jack_port_t *dport = JACK_jack_port_by_name(client, devports[i]);
         const char *type = JACK_jack_port_type(dport);
         const int len = SDL_strlen(type);
-        /* See if type ends with "audio" */
+        // See if type ends with "audio"
         if (len >= 5 && !SDL_memcmp(type + len - 5, "audio", 5)) {
             audio_ports[channels++] = i;
         }
@@ -335,7 +335,7 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
 
     /* !!! FIXME: docs say about buffer size: "This size may change, clients that depend on it must register a bufsize_callback so they will be notified if it does." */
 
-    /* Jack pretty much demands what it wants. */
+    // Jack pretty much demands what it wants.
     device->spec.format = SDL_AUDIO_F32;
     device->spec.freq = JACK_jack_get_sample_rate(client);
     device->spec.channels = channels;
@@ -351,7 +351,7 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
         }
     }
 
-    /* Build SDL's ports, which we will connect to the device ports. */
+    // Build SDL's ports, which we will connect to the device ports.
     device->hidden->sdlports = (jack_port_t **)SDL_calloc(channels, sizeof(jack_port_t *));
     if (device->hidden->sdlports == NULL) {
         SDL_free(audio_ports);
@@ -386,7 +386,7 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
         return SDL_SetError("Failed to activate JACK client");
     }
 
-    /* once activated, we can connect all the ports. */
+    // once activated, we can connect all the ports.
     for (i = 0; i < channels; i++) {
         const char *sdlport = JACK_jack_port_name(device->hidden->sdlports[i]);
         const char *srcport = iscapture ? devports[audio_ports[i]] : sdlport;
@@ -397,11 +397,11 @@ static int JACK_OpenDevice(SDL_AudioDevice *device)
         }
     }
 
-    /* don't need these anymore. */
+    // don't need these anymore.
     JACK_jack_free(devports);
     SDL_free(audio_ports);
 
-    /* We're ready to rock and roll. :-) */
+    // We're ready to rock and roll. :-)
     return 0;
 }
 
@@ -415,7 +415,7 @@ static SDL_bool JACK_Init(SDL_AudioDriverImpl *impl)
     if (LoadJackLibrary() < 0) {
         return SDL_FALSE;
     } else {
-        /* Make sure a JACK server is running and available. */
+        // Make sure a JACK server is running and available.
         jack_status_t status;
         jack_client_t *client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
         if (client == NULL) {
@@ -425,7 +425,6 @@ static SDL_bool JACK_Init(SDL_AudioDriverImpl *impl)
         JACK_jack_client_close(client);
     }
 
-    /* Set the function pointers */
     impl->OpenDevice = JACK_OpenDevice;
     impl->GetDeviceBuf = JACK_GetDeviceBuf;
     impl->PlayDevice = JACK_PlayDevice;
@@ -438,11 +437,11 @@ static SDL_bool JACK_Init(SDL_AudioDriverImpl *impl)
     impl->HasCaptureSupport = SDL_TRUE;
     impl->ProvidesOwnCallbackThread = SDL_TRUE;
 
-    return SDL_TRUE; /* this audio target is available. */
+    return SDL_TRUE;
 }
 
 AudioBootStrap JACK_bootstrap = {
     "jack", "JACK Audio Connection Kit", JACK_Init, SDL_FALSE
 };
 
-#endif /* SDL_AUDIO_DRIVER_JACK */
+#endif // SDL_AUDIO_DRIVER_JACK

src/audio/jack/SDL_jackaudio.h

@@ -32,4 +32,4 @@ struct SDL_PrivateAudioData
     float *iobuffer;
 };
 
-#endif /* SDL_jackaudio_h_ */
+#endif // SDL_jackaudio_h_

src/audio/n3ds/SDL_n3dsaudio.h

@@ -24,11 +24,11 @@
 
 #include <3ds.h>
 
-#define NUM_BUFFERS 2 /* -- Don't lower this! */
+#define NUM_BUFFERS 2 // -- Don't lower this!
 
 struct SDL_PrivateAudioData
 {
-    /* Speaker data */
+    // Speaker data
     Uint8 *mixbuf;
     Uint32 nextbuf;
     ndspWaveBuf waveBuf[NUM_BUFFERS];
@@ -37,4 +37,4 @@ struct SDL_PrivateAudioData
     SDL_bool isCancelled;
 };
 
-#endif /* SDL_n3dsaudio_h */
+#endif // SDL_n3dsaudio_h

src/audio/netbsd/SDL_netbsdaudio.c

@@ -135,7 +135,7 @@ static int NETBSDAUDIO_WaitDevice(SDL_AudioDevice *device)
         } else if (iscapture && (remain < device->buffer_size)) {
             SDL_Delay(10);
         } else {
-            break; /* ready to go! */
+            break; // ready to go!
         }
     }
 

src/audio/netbsd/SDL_netbsdaudio.h

@@ -27,18 +27,18 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The file descriptor for the audio device */
+    // The file descriptor for the audio device
     int audio_fd;
 
-    /* Raw mixing buffer */
+    // Raw mixing buffer
     Uint8 *mixbuf;
     int mixlen;
 
-    /* Support for audio timing using a timer, in addition to SDL_IOReady() */
+    // Support for audio timing using a timer, in addition to SDL_IOReady()
     float frame_ticks;
     float next_frame;
 };
 
-#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
+#define FUDGE_TICKS 10 // The scheduler overhead ticks per frame
 
-#endif /* SDL_netbsdaudio_h_ */
+#endif // SDL_netbsdaudio_h_

src/audio/openslES/SDL_openslES.h

@@ -35,4 +35,4 @@ static void OPENSLES_PauseDevices(void) {}
 
 #endif
 
-#endif /* SDL_openslesaudio_h_ */
+#endif // SDL_openslesaudio_h_

src/audio/pipewire/SDL_pipewire.c

@@ -63,7 +63,7 @@
  * This seems to be a sane lower limit as Pipewire
  * uses it in several of it's own modules.
  */
-#define PW_MIN_SAMPLES     32 /* About 0.67ms at 48kHz */
+#define PW_MIN_SAMPLES     32 // About 0.67ms at 48kHz
 #define PW_BASE_CLOCK_RATE 48000
 
 #define PW_POD_BUFFER_LENGTH         1024
@@ -82,7 +82,7 @@ enum PW_READY_FLAGS
 
 static SDL_bool pipewire_initialized = SDL_FALSE;
 
-/* Pipewire entry points */
+// Pipewire entry points
 static const char *(*PIPEWIRE_pw_get_library_version)(void);
 static void (*PIPEWIRE_pw_init)(int *, char ***);
 static void (*PIPEWIRE_pw_deinit)(void);
@@ -127,7 +127,7 @@ static int pipewire_dlsym(const char *fn, void **addr)
 {
     *addr = SDL_LoadFunction(pipewire_handle, fn);
     if (*addr == NULL) {
-        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        // Don't call SDL_SetError(): SDL_LoadFunction already did.
         return 0;
     }
 
@@ -163,10 +163,11 @@ static int load_pipewire_library(void)
 }
 
 static void unload_pipewire_library(void)
-{ /* Nothing to do */
+{
+    // Nothing to do
 }
 
-#endif /* SDL_AUDIO_DRIVER_PIPEWIRE_DYNAMIC */
+#endif // SDL_AUDIO_DRIVER_PIPEWIRE_DYNAMIC
 
 static int load_pipewire_syms(void)
 {
@@ -220,7 +221,7 @@ static int init_pipewire_library(void)
                 return -1;
             }
 
-            /* SDL can build against 0.3.20, but requires 0.3.24 */
+            // SDL can build against 0.3.20, but requires 0.3.24
             if (pipewire_version_at_least(0, 3, 24)) {
                 PIPEWIRE_pw_init(NULL, NULL);
                 return 0;
@@ -237,7 +238,7 @@ static void deinit_pipewire_library(void)
     unload_pipewire_library();
 }
 
-/* A generic Pipewire node object used for enumeration. */
+// A generic Pipewire node object used for enumeration.
 struct node_object
 {
     struct spa_list link;
@@ -260,7 +261,7 @@ struct node_object
     struct spa_hook core_listener;
 };
 
-/* A sink/source node used for stream I/O. */
+// A sink/source node used for stream I/O.
 struct io_node
 {
     struct spa_list link;
@@ -269,13 +270,13 @@ struct io_node
     SDL_bool is_capture;
     SDL_AudioSpec spec;
 
-    const char *name; /* Friendly name */
-    const char *path; /* OS identifier (i.e. ALSA endpoint) */
+    const char *name; // Friendly name
+    const char *path; // OS identifier (i.e. ALSA endpoint)
 
-    char buf[]; /* Buffer to hold the name and path strings. */
+    char buf[]; // Buffer to hold the name and path strings.
 };
 
-/* The global hotplug thread and associated objects. */
+// The global hotplug thread and associated objects.
 static struct pw_thread_loop *hotplug_loop;
 static struct pw_core *hotplug_core;
 static struct pw_context *hotplug_context;
@@ -291,13 +292,13 @@ static SDL_bool hotplug_events_enabled;
 static char *pipewire_default_sink_id = NULL;
 static char *pipewire_default_source_id = NULL;
 
-/* The active node list */
+// The active node list
 static SDL_bool io_list_check_add(struct io_node *node)
 {
     struct io_node *n;
     SDL_bool ret = SDL_TRUE;
 
-    /* See if the node is already in the list */
+    // See if the node is already in the list
     spa_list_for_each (n, &hotplug_io_list, link) {
         if (n->id == node->id) {
             ret = SDL_FALSE;
@@ -305,7 +306,7 @@ static SDL_bool io_list_check_add(struct io_node *node)
         }
     }
 
-    /* Add to the list if the node doesn't already exist */
+    // Add to the list if the node doesn't already exist
     spa_list_append(&hotplug_io_list, &node->link);
 
     if (hotplug_events_enabled) {
@@ -321,7 +322,7 @@ static void io_list_remove(Uint32 id)
 {
     struct io_node *n, *temp;
 
-    /* Find and remove the node from the list */
+    // Find and remove the node from the list
     spa_list_for_each_safe (n, temp, &hotplug_io_list, link) {
         if (n->id == id) {
             spa_list_remove(&n->link);
@@ -369,7 +370,7 @@ static void node_object_destroy(struct node_object *node)
     PIPEWIRE_pw_proxy_destroy(node->proxy);
 }
 
-/* The pending node list */
+// The pending node list
 static void pending_list_add(struct node_object *node)
 {
     SDL_assert(node);
@@ -401,7 +402,7 @@ static void *node_object_new(Uint32 id, const char *type, Uint32 version, const
     struct pw_proxy *proxy;
     struct node_object *node;
 
-    /* Create the proxy object */
+    // Create the proxy object
     proxy = pw_registry_bind(hotplug_registry, id, type, version, sizeof(struct node_object));
     if (proxy == NULL) {
         SDL_SetError("Pipewire: Failed to create proxy object (%i)", errno);
@@ -414,24 +415,24 @@ static void *node_object_new(Uint32 id, const char *type, Uint32 version, const
     node->id = id;
     node->proxy = proxy;
 
-    /* Add the callbacks */
+    // Add the callbacks
     pw_core_add_listener(hotplug_core, &node->core_listener, core_events, node);
     PIPEWIRE_pw_proxy_add_object_listener(node->proxy, &node->node_listener, funcs, node);
 
-    /* Add the node to the active list */
+    // Add the node to the active list
     pending_list_add(node);
 
     return node;
 }
 
-/* Core sync points */
+// Core sync points
 static void core_events_hotplug_init_callback(void *object, uint32_t id, int seq)
 {
     if (id == PW_ID_CORE && seq == hotplug_init_seq_val) {
-        /* This core listener is no longer needed. */
+        // This core listener is no longer needed.
         spa_hook_remove(&hotplug_core_listener);
 
-        /* Signal that the initial I/O list is populated */
+        // Signal that the initial I/O list is populated
         hotplug_init_complete = SDL_TRUE;
         PIPEWIRE_pw_thread_loop_signal(hotplug_loop, false);
     }
@@ -483,7 +484,7 @@ static void hotplug_core_sync(struct node_object *node)
     }
 }
 
-/* Helpers for retrieving values from params */
+// Helpers for retrieving values from params
 static SDL_bool get_range_param(const struct spa_pod *param, Uint32 key, int *def, int *min, int *max)
 {
     const struct spa_pod_prop *prop;
@@ -535,7 +536,7 @@ static SDL_bool get_int_param(const struct spa_pod *param, Uint32 key, int *val)
     return SDL_FALSE;
 }
 
-/* Interface node callbacks */
+// Interface node callbacks
 static void node_event_info(void *object, const struct pw_node_info *info)
 {
     struct node_object *node = object;
@@ -549,7 +550,7 @@ static void node_event_info(void *object, const struct pw_node_info *info)
             io->spec.channels = (Uint8)SDL_atoi(prop_val);
         }
 
-        /* Need to parse the parameters to get the sample rate */
+        // Need to parse the parameters to get the sample rate
         for (i = 0; i < info->n_params; ++i) {
             pw_node_enum_params(node->proxy, 0, info->params[i].id, 0, 0, NULL);
         }
@@ -563,7 +564,7 @@ static void node_event_param(void *object, int seq, uint32_t id, uint32_t index,
     struct node_object *node = object;
     struct io_node *io = node->userdata;
 
-    /* Get the default frequency */
+    // Get the default frequency
     if (io->spec.freq == 0) {
         get_range_param(param, SPA_FORMAT_AUDIO_rate, &io->spec.freq, NULL, NULL);
     }
@@ -586,19 +587,19 @@ static const struct pw_node_events interface_node_events = { PW_VERSION_NODE_EVE
 static char *get_name_from_json(const char *json)
 {
     struct spa_json parser[2];
-    char key[7]; /* "name" */
+    char key[7]; // "name"
     char value[PW_MAX_IDENTIFIER_LENGTH];
     spa_json_init(&parser[0], json, SDL_strlen(json));
     if (spa_json_enter_object(&parser[0], &parser[1]) <= 0) {
-        /* Not actually JSON */
+        // Not actually JSON
         return NULL;
     }
     if (spa_json_get_string(&parser[1], key, sizeof(key)) <= 0) {
-        /* Not actually a key/value pair */
+        // Not actually a key/value pair
         return NULL;
     }
     if (spa_json_get_string(&parser[1], value, sizeof(value)) <= 0) {
-        /* Somehow had a key with no value? */
+        // Somehow had a key with no value?
         return NULL;
     }
     return SDL_strdup(value);
@@ -617,7 +618,7 @@ static void change_default_device(const char *path)
     }
 }
 
-/* Metadata node callback */
+// Metadata node callback
 static int metadata_property(void *object, Uint32 subject, const char *key, const char *type, const char *value)
 {
     struct node_object *node = object;
@@ -645,13 +646,13 @@ static int metadata_property(void *object, Uint32 subject, const char *key, cons
 
 static const struct pw_metadata_events metadata_node_events = { PW_VERSION_METADATA_EVENTS, .property = metadata_property };
 
-/* Global registry callbacks */
+// Global registry callbacks
 static void registry_event_global_callback(void *object, uint32_t id, uint32_t permissions, const char *type, uint32_t version,
                                            const struct spa_dict *props)
 {
     struct node_object *node;
 
-    /* We're only interested in interface and metadata nodes. */
+    // We're only interested in interface and metadata nodes.
     if (!SDL_strcmp(type, PW_TYPE_INTERFACE_Node)) {
         const char *media_class = spa_dict_lookup(props, PW_KEY_MEDIA_CLASS);
 
@@ -663,7 +664,7 @@ static void registry_event_global_callback(void *object, uint32_t id, uint32_t p
             int desc_buffer_len;
             int path_buffer_len;
 
-            /* Just want sink and capture */
+            // Just want sink and capture
             if (!SDL_strcasecmp(media_class, "Audio/Sink")) {
                 is_capture = SDL_FALSE;
             } else if (!SDL_strcasecmp(media_class, "Audio/Source")) {
@@ -682,7 +683,7 @@ static void registry_event_global_callback(void *object, uint32_t id, uint32_t p
                     return;
                 }
 
-                /* Allocate and initialize the I/O node information struct */
+                // Allocate and initialize the I/O node information struct
                 desc_buffer_len = SDL_strlen(node_desc) + 1;
                 path_buffer_len = SDL_strlen(node_path) + 1;
                 node->userdata = io = SDL_calloc(1, sizeof(struct io_node) + desc_buffer_len + path_buffer_len);
@@ -692,16 +693,16 @@ static void registry_event_global_callback(void *object, uint32_t id, uint32_t p
                     return;
                 }
 
-                /* Begin setting the node properties */
+                // Begin setting the node properties
                 io->id = id;
                 io->is_capture = is_capture;
-                io->spec.format = SDL_AUDIO_F32; /* Pipewire uses floats internally, other formats require conversion. */
+                io->spec.format = SDL_AUDIO_F32; // Pipewire uses floats internally, other formats require conversion.
                 io->name = io->buf;
                 io->path = io->buf + desc_buffer_len;
                 SDL_strlcpy(io->buf, node_desc, desc_buffer_len);
                 SDL_strlcpy(io->buf + desc_buffer_len, node_path, path_buffer_len);
 
-                /* Update sync points */
+                // Update sync points
                 hotplug_core_sync(node);
             }
         }
@@ -712,7 +713,7 @@ static void registry_event_global_callback(void *object, uint32_t id, uint32_t p
             return;
         }
 
-        /* Update sync points */
+        // Update sync points
         hotplug_core_sync(node);
     }
 }
@@ -726,7 +727,7 @@ static void registry_event_remove_callback(void *object, uint32_t id)
 static const struct pw_registry_events registry_events = { PW_VERSION_REGISTRY_EVENTS, .global = registry_event_global_callback,
                                                            .global_remove = registry_event_remove_callback };
 
-/* The hotplug thread */
+// The hotplug thread
 static int hotplug_loop_init(void)
 {
     int res;
@@ -818,7 +819,7 @@ static void PIPEWIRE_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDe
 
     PIPEWIRE_pw_thread_loop_lock(hotplug_loop);
 
-    /* Wait until the initial registry enumeration is complete */
+    // Wait until the initial registry enumeration is complete
     if (!hotplug_init_complete) {
         PIPEWIRE_pw_thread_loop_wait(hotplug_loop);
     }
@@ -839,7 +840,7 @@ static void PIPEWIRE_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDe
     PIPEWIRE_pw_thread_loop_unlock(hotplug_loop);
 }
 
-/* Channel maps that match the order in SDL_Audio.h */
+// Channel maps that match the order in SDL_Audio.h
 static const enum spa_audio_channel PIPEWIRE_channel_map_1[] = { SPA_AUDIO_CHANNEL_MONO };
 static const enum spa_audio_channel PIPEWIRE_channel_map_2[] = { SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR };
 static const enum spa_audio_channel PIPEWIRE_channel_map_3[] = { SPA_AUDIO_CHANNEL_FL, SPA_AUDIO_CHANNEL_FR, SPA_AUDIO_CHANNEL_LFE };
@@ -890,7 +891,7 @@ static void initialize_spa_info(const SDL_AudioSpec *spec, struct spa_audio_info
         break;
     }
 
-    /* Pipewire natively supports all of SDL's sample formats */
+    // Pipewire natively supports all of SDL's sample formats
     switch (spec->format) {
     case SDL_AUDIO_U8:
         info->format = SPA_AUDIO_FORMAT_U8;
@@ -1065,10 +1066,10 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
     const SDL_bool iscapture = device->iscapture;
     int res;
 
-    /* Clamp the period size to sane values */
+    // Clamp the period size to sane values
     const int min_period = PW_MIN_SAMPLES * SPA_MAX(device->spec.freq / PW_BASE_CLOCK_RATE, 1);
 
-    /* Get the hints for the application name, stream name and role */
+    // Get the hints for the application name, stream name and role
     app_name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_NAME);
     if (app_name == NULL || *app_name == '\0') {
         app_name = SDL_GetHint(SDL_HINT_APP_NAME);
@@ -1077,7 +1078,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         }
     }
 
-    /* App ID. Default to NULL if not available. */
+    // App ID. Default to NULL if not available.
     app_id = SDL_GetHint(SDL_HINT_APP_ID);
 
     stream_name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_STREAM_NAME);
@@ -1094,7 +1095,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         stream_role = "Game";
     }
 
-    /* Initialize the Pipewire stream info from the SDL audio spec */
+    // Initialize the Pipewire stream info from the SDL audio spec
     initialize_spa_info(&device->spec, &spa_info);
     params = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &spa_info);
     if (params == NULL) {
@@ -1107,7 +1108,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         return SDL_OutOfMemory();
     }
 
-    /* Size of a single audio frame in bytes */
+    // Size of a single audio frame in bytes
     priv->stride = SDL_AUDIO_FRAMESIZE(device->spec);
 
     if (device->sample_frames < min_period) {
@@ -1122,7 +1123,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         return SDL_SetError("Pipewire: Failed to create stream loop (%i)", errno);
     }
 
-    /* Load the realtime module so Pipewire can set the loop thread to the appropriate priority. */
+    // Load the realtime module so Pipewire can set the loop thread to the appropriate priority.
     props = PIPEWIRE_pw_properties_new(PW_KEY_CONFIG_NAME, "client-rt.conf", NULL);
     if (props == NULL) {
         return SDL_SetError("Pipewire: Failed to create stream context properties (%i)", errno);
@@ -1173,7 +1174,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         }
     }
 
-    /* Create the new stream */
+    // Create the new stream
     priv->stream = PIPEWIRE_pw_stream_new_simple(PIPEWIRE_pw_thread_loop_get_loop(priv->loop), stream_name, props,
                                                  iscapture ? &stream_input_events : &stream_output_events, device);
     if (priv->stream == NULL) {
@@ -1191,7 +1192,7 @@ static int PIPEWIRE_OpenDevice(SDL_AudioDevice *device)
         return SDL_SetError("Pipewire: Failed to start stream loop");
     }
 
-    /* Wait until all init flags are set or the stream has failed. */
+    // Wait until all init flags are set or the stream has failed.
     PIPEWIRE_pw_thread_loop_lock(priv->loop);
     while (priv->stream_init_status != PW_READY_FLAG_ALL_BITS &&
            PIPEWIRE_pw_stream_get_state(priv->stream, NULL) != PW_STREAM_STATE_ERROR) {
@@ -1264,7 +1265,6 @@ static SDL_bool PIPEWIRE_Init(SDL_AudioDriverImpl *impl)
         }
     }
 
-    /* Set the function pointers */
     impl->DetectDevices = PIPEWIRE_DetectDevices;
     impl->OpenDevice = PIPEWIRE_OpenDevice;
     impl->DeinitializeStart = PIPEWIRE_DeinitializeStart;
@@ -1283,4 +1283,4 @@ static SDL_bool PIPEWIRE_Init(SDL_AudioDriverImpl *impl)
 
 AudioBootStrap PIPEWIRE_bootstrap = { "pipewire", "Pipewire", PIPEWIRE_Init, SDL_FALSE };
 
-#endif /* SDL_AUDIO_DRIVER_PIPEWIRE */
+#endif // SDL_AUDIO_DRIVER_PIPEWIRE

src/audio/pipewire/SDL_pipewire.h

@@ -33,11 +33,11 @@ struct SDL_PrivateAudioData
     struct pw_stream *stream;
     struct pw_context *context;
 
-    Sint32 stride; /* Bytes-per-frame */
+    Sint32 stride; // Bytes-per-frame
     int stream_init_status;
 
     // Set in GetDeviceBuf, filled in AudioThreadIterate, queued in PlayDevice
     struct pw_buffer *pw_buf;
 };
 
-#endif /* SDL_pipewire_h_ */
+#endif // SDL_pipewire_h_

src/audio/ps2/SDL_ps2audio.h

@@ -29,14 +29,14 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The hardware output channel. */
+    // The hardware output channel.
     int channel;
-    /* The raw allocated mixing buffer. */
+    // The raw allocated mixing buffer.
     Uint8 *rawbuf;
-    /* Individual mixing buffers. */
+    // Individual mixing buffers.
     Uint8 *mixbufs[NUM_BUFFERS];
-    /* Index of the next available mixing buffer. */
+    // Index of the next available mixing buffer.
     int next_buffer;
 };
 
-#endif /* SDL_ps2audio_h_ */
+#endif // SDL_ps2audio_h_

src/audio/psp/SDL_pspaudio.h

@@ -28,14 +28,14 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The hardware output channel. */
+    // The hardware output channel.
     int channel;
-    /* The raw allocated mixing buffer. */
+    // The raw allocated mixing buffer.
     Uint8 *rawbuf;
-    /* Individual mixing buffers. */
+    // Individual mixing buffers.
     Uint8 *mixbufs[NUM_BUFFERS];
-    /* Index of the next available mixing buffer. */
+    // Index of the next available mixing buffer.
     int next_buffer;
 };
 
-#endif /* SDL_pspaudio_h_ */
+#endif // SDL_pspaudio_h_

src/audio/pulseaudio/SDL_pulseaudio.c

@@ -23,7 +23,7 @@
 
 #ifdef SDL_AUDIO_DRIVER_PULSEAUDIO
 
-/* Allow access to a raw mixing buffer */
+// Allow access to a raw mixing buffer
 
 #ifdef HAVE_SIGNAL_H
 #include <signal.h>
@@ -39,7 +39,7 @@
 typedef void (*pa_operation_notify_cb_t) (pa_operation *o, void *userdata);
 #endif
 
-/* should we include monitors in the device list? Set at SDL_Init time */
+// should we include monitors in the device list? Set at SDL_Init time
 static SDL_bool include_monitors = SDL_FALSE;
 
 static pa_threaded_mainloop *pulseaudio_threaded_mainloop = NULL;
@@ -128,14 +128,14 @@ static int load_pulseaudio_sym(const char *fn, void **addr)
 {
     *addr = SDL_LoadFunction(pulseaudio_handle, fn);
     if (*addr == NULL) {
-        /* Don't call SDL_SetError(): SDL_LoadFunction already did. */
+        // 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. */
+// cast funcs to char* first, to please GCC's strict aliasing rules.
 #define SDL_PULSEAUDIO_SYM(x)                                       \
     if (!load_pulseaudio_sym(#x, (void **)(char *)&PULSEAUDIO_##x)) \
     return -1
@@ -155,7 +155,7 @@ static int LoadPulseAudioLibrary(void)
         pulseaudio_handle = SDL_LoadObject(pulseaudio_library);
         if (pulseaudio_handle == NULL) {
             retval = -1;
-            /* Don't call SDL_SetError(): SDL_LoadObject already did. */
+            // Don't call SDL_SetError(): SDL_LoadObject already did.
         } else {
             retval = load_pulseaudio_syms();
             if (retval < 0) {
@@ -180,7 +180,7 @@ static int LoadPulseAudioLibrary(void)
     return 0;
 }
 
-#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC */
+#endif // SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC
 
 static int load_pulseaudio_syms(void)
 {
@@ -231,7 +231,7 @@ static int load_pulseaudio_syms(void)
     SDL_PULSEAUDIO_SYM(pa_stream_set_read_callback);
     SDL_PULSEAUDIO_SYM(pa_context_get_server_info);
 
-    /* optional */
+    // optional
 #ifdef SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC
     load_pulseaudio_sym("pa_operation_set_state_callback", (void **)(char *)&PULSEAUDIO_pa_operation_set_state_callback);  // needs pulseaudio 4.0
     load_pulseaudio_sym("pa_threaded_mainloop_set_name", (void **)(char *)&PULSEAUDIO_pa_threaded_mainloop_set_name);  // needs pulseaudio 5.0
@@ -254,7 +254,7 @@ static SDL_INLINE int squashVersion(const int major, const int minor, const int
     return ((major & 0xFF) << 16) | ((minor & 0xFF) << 8) | (patch & 0xFF);
 }
 
-/* Workaround for older pulse: pa_context_new() must have non-NULL appname */
+// Workaround for older pulse: pa_context_new() must have non-NULL appname
 static const char *getAppName(void)
 {
     const char *retval = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_NAME);
@@ -266,12 +266,12 @@ static const char *getAppName(void)
         return retval;
     } else {
         const char *verstr = PULSEAUDIO_pa_get_library_version();
-        retval = "SDL Application"; /* the "oh well" default. */
+        retval = "SDL Application"; // the "oh well" default.
         if (verstr != NULL) {
             int maj, min, patch;
             if (SDL_sscanf(verstr, "%d.%d.%d", &maj, &min, &patch) == 3) {
                 if (squashVersion(maj, min, patch) >= squashVersion(0, 9, 15)) {
-                    retval = NULL; /* 0.9.15+ handles NULL correctly. */
+                    retval = NULL; // 0.9.15+ handles NULL correctly.
                 }
             }
         }
@@ -288,7 +288,7 @@ static void OperationStateChangeCallback(pa_operation *o, void *userdata)
    you did the work in the callback and just want to know it's done, though. */
 static void WaitForPulseOperation(pa_operation *o)
 {
-    /* This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about. */
+    // This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about.
     SDL_assert(pulseaudio_threaded_mainloop != NULL);
     if (o) {
         // note that if PULSEAUDIO_pa_operation_set_state_callback == NULL, then `o` must have a callback that will signal pulseaudio_threaded_mainloop.
@@ -299,7 +299,7 @@ static void WaitForPulseOperation(pa_operation *o)
             PULSEAUDIO_pa_operation_set_state_callback(o, OperationStateChangeCallback, NULL);
         }
         while (PULSEAUDIO_pa_operation_get_state(o) == PA_OPERATION_RUNNING) {
-            PULSEAUDIO_pa_threaded_mainloop_wait(pulseaudio_threaded_mainloop);  /* this releases the lock and blocks on an internal condition variable. */
+            PULSEAUDIO_pa_threaded_mainloop_wait(pulseaudio_threaded_mainloop);  // this releases the lock and blocks on an internal condition variable.
         }
         PULSEAUDIO_pa_operation_unref(o);
     }
@@ -323,7 +323,7 @@ static void DisconnectFromPulseServer(void)
 
 static void PulseContextStateChangeCallback(pa_context *context, void *userdata)
 {
-    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  /* just signal any waiting code, it can look up the details. */
+    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  // just signal any waiting code, it can look up the details.
 }
 
 static int ConnectToPulseServer(void)
@@ -334,7 +334,7 @@ static int ConnectToPulseServer(void)
     SDL_assert(pulseaudio_threaded_mainloop == NULL);
     SDL_assert(pulseaudio_context == NULL);
 
-    /* Set up a new main loop */
+    // Set up a new main loop
     if (!(pulseaudio_threaded_mainloop = PULSEAUDIO_pa_threaded_mainloop_new())) {
         return SDL_SetError("pa_threaded_mainloop_new() failed");
     }
@@ -352,7 +352,7 @@ static int ConnectToPulseServer(void)
     PULSEAUDIO_pa_threaded_mainloop_lock(pulseaudio_threaded_mainloop);
 
     mainloop_api = PULSEAUDIO_pa_threaded_mainloop_get_api(pulseaudio_threaded_mainloop);
-    SDL_assert(mainloop_api); /* this never fails, right? */
+    SDL_assert(mainloop_api); // this never fails, right?
 
     pulseaudio_context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());
     if (pulseaudio_context == NULL) {
@@ -362,7 +362,7 @@ static int ConnectToPulseServer(void)
 
     PULSEAUDIO_pa_context_set_state_callback(pulseaudio_context, PulseContextStateChangeCallback, NULL);
 
-    /* Connect to the PulseAudio server */
+    // Connect to the PulseAudio server
     if (PULSEAUDIO_pa_context_connect(pulseaudio_context, NULL, 0, NULL) < 0) {
         SDL_SetError("Could not setup connection to PulseAudio");
         goto failed;
@@ -381,7 +381,7 @@ static int ConnectToPulseServer(void)
 
     PULSEAUDIO_pa_threaded_mainloop_unlock(pulseaudio_threaded_mainloop);
 
-    return 0; /* connected and ready! */
+    return 0; // connected and ready!
 
 failed:
     PULSEAUDIO_pa_threaded_mainloop_unlock(pulseaudio_threaded_mainloop);
@@ -392,27 +392,27 @@ failed:
 static void WriteCallback(pa_stream *p, size_t nbytes, void *userdata)
 {
     struct SDL_PrivateAudioData *h = (struct SDL_PrivateAudioData *)userdata;
-    /*printf("PULSEAUDIO WRITE CALLBACK! nbytes=%u\n", (unsigned int) nbytes);*/
+    //printf("PULSEAUDIO WRITE CALLBACK! nbytes=%u\n", (unsigned int) nbytes);
     h->bytes_requested += nbytes;
     PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);
 }
 
-/* This function waits until it is possible to write a full sound buffer */
+// This function waits until it is possible to write a full sound buffer
 static int PULSEAUDIO_WaitDevice(SDL_AudioDevice *device)
 {
     struct SDL_PrivateAudioData *h = device->hidden;
     int retval = 0;
 
-    /*printf("PULSEAUDIO PLAYDEVICE START! mixlen=%d\n", available);*/
+    //printf("PULSEAUDIO PLAYDEVICE START! mixlen=%d\n", available);
 
     PULSEAUDIO_pa_threaded_mainloop_lock(pulseaudio_threaded_mainloop);
 
     while (!SDL_AtomicGet(&device->shutdown) && (h->bytes_requested == 0)) {
-        /*printf("PULSEAUDIO WAIT IN WAITDEVICE!\n");*/
+        //printf("PULSEAUDIO WAIT IN WAITDEVICE!\n");
         PULSEAUDIO_pa_threaded_mainloop_wait(pulseaudio_threaded_mainloop);
 
         if ((PULSEAUDIO_pa_context_get_state(pulseaudio_context) != PA_CONTEXT_READY) || (PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY)) {
-            /*printf("PULSEAUDIO DEVICE FAILURE IN WAITDEVICE!\n");*/
+            //printf("PULSEAUDIO DEVICE FAILURE IN WAITDEVICE!\n");
             retval = -1;
             break;
         }
@@ -427,7 +427,7 @@ static int PULSEAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, i
 {
     struct SDL_PrivateAudioData *h = device->hidden;
 
-    /*printf("PULSEAUDIO PLAYDEVICE START! mixlen=%d\n", available);*/
+    //printf("PULSEAUDIO PLAYDEVICE START! mixlen=%d\n", available);
 
     SDL_assert(h->bytes_requested >= buffer_size);
 
@@ -439,10 +439,10 @@ static int PULSEAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, i
         return -1;
     }
 
-    /*printf("PULSEAUDIO FEED! nbytes=%d\n", buffer_size);*/
+    //printf("PULSEAUDIO FEED! nbytes=%d\n", buffer_size);
     h->bytes_requested -= buffer_size;
 
-    /*printf("PULSEAUDIO PLAYDEVICE END! written=%d\n", written);*/
+    //printf("PULSEAUDIO PLAYDEVICE END! written=%d\n", written);
     return 0;
 }
 
@@ -464,8 +464,8 @@ static Uint8 *PULSEAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
 
 static void ReadCallback(pa_stream *p, size_t nbytes, void *userdata)
 {
-    /*printf("PULSEAUDIO READ CALLBACK! nbytes=%u\n", (unsigned int) nbytes);*/
-    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  /* the capture code queries what it needs, we just need to signal to end any wait */
+    //printf("PULSEAUDIO READ CALLBACK! nbytes=%u\n", (unsigned int) nbytes);
+    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  // the capture code queries what it needs, we just need to signal to end any wait
 }
 
 static int PULSEAUDIO_WaitCaptureDevice(SDL_AudioDevice *device)
@@ -527,7 +527,7 @@ static int PULSEAUDIO_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, i
             PULSEAUDIO_pa_stream_drop(h->stream); // done with this fragment.
             PULSEAUDIO_pa_threaded_mainloop_unlock(pulseaudio_threaded_mainloop);
         }
-        return cpy; /* new data, return it. */
+        return cpy; // new data, return it.
     }
 
     return 0;
@@ -550,15 +550,15 @@ static void PULSEAUDIO_FlushCapture(SDL_AudioDevice *device)
     while (!SDL_AtomicGet(&device->shutdown) && (PULSEAUDIO_pa_stream_readable_size(h->stream) > 0)) {
         PULSEAUDIO_pa_threaded_mainloop_wait(pulseaudio_threaded_mainloop);
         if ((PULSEAUDIO_pa_context_get_state(pulseaudio_context) != PA_CONTEXT_READY) || (PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY)) {
-            /*printf("PULSEAUDIO DEVICE FAILURE IN FLUSHCAPTURE!\n");*/
+            //printf("PULSEAUDIO DEVICE FAILURE IN FLUSHCAPTURE!\n");
             SDL_AudioDeviceDisconnected(device);
             break;
         }
 
         if (PULSEAUDIO_pa_stream_readable_size(h->stream) > 0) {
-            /* a new fragment is available! Just dump it. */
+            // a new fragment is available! Just dump it.
             PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
-            PULSEAUDIO_pa_stream_drop(h->stream); /* drop this fragment. */
+            PULSEAUDIO_pa_stream_drop(h->stream); // drop this fragment.
         }
     }
 
@@ -619,7 +619,7 @@ static SDL_bool FindDeviceName(SDL_AudioDevice *device)
 
 static void PulseStreamStateChangeCallback(pa_stream *stream, void *userdata)
 {
-    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  /* just signal any waiting code, it can look up the details. */
+    PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);  // just signal any waiting code, it can look up the details.
 }
 
 static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
@@ -638,13 +638,13 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
     SDL_assert(pulseaudio_threaded_mainloop != NULL);
     SDL_assert(pulseaudio_context != NULL);
 
-    /* Initialize all variables that we clean on shutdown */
+    // Initialize all variables that we clean on shutdown
     h = device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
     if (device->hidden == NULL) {
         return SDL_OutOfMemory();
     }
 
-    /* Try for a closest match on audio format */
+    // Try for a closest match on audio format
     closefmts = SDL_ClosestAudioFormats(device->spec.format);
     while ((test_format = *(closefmts++)) != 0) {
 #ifdef DEBUG_AUDIO
@@ -683,10 +683,10 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
     device->spec.format = test_format;
     paspec.format = format;
 
-    /* Calculate the final parameters for this audio specification */
+    // Calculate the final parameters for this audio specification
     SDL_UpdatedAudioDeviceFormat(device);
 
-    /* Allocate mixing buffer */
+    // Allocate mixing buffer
     if (!iscapture) {
         h->mixbuf = (Uint8 *)SDL_malloc(device->buffer_size);
         if (h->mixbuf == NULL) {
@@ -698,7 +698,7 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
     paspec.channels = device->spec.channels;
     paspec.rate = device->spec.freq;
 
-    /* Reduced prebuffering compared to the defaults. */
+    // Reduced prebuffering compared to the defaults.
     paattr.fragsize = device->buffer_size;   // despite the name, this is only used for capture devices, according to PulseAudio docs!
     paattr.tlength = device->buffer_size;
     paattr.prebuf = -1;
@@ -712,15 +712,15 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
         retval = SDL_SetError("Requested PulseAudio sink/source missing?");
     } else {
         const char *name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_STREAM_NAME);
-        /* The SDL ALSA output hints us that we use Windows' channel mapping */
-        /* https://bugzilla.libsdl.org/show_bug.cgi?id=110 */
+        // The SDL ALSA output hints us that we use Windows' channel mapping
+        // https://bugzilla.libsdl.org/show_bug.cgi?id=110
         PULSEAUDIO_pa_channel_map_init_auto(&pacmap, device->spec.channels, PA_CHANNEL_MAP_WAVEEX);
 
         h->stream = PULSEAUDIO_pa_stream_new(
             pulseaudio_context,
-            (name && *name) ? name : "Audio Stream", /* stream description */
-            &paspec,                                 /* sample format spec */
-            &pacmap                                  /* channel map */
+            (name && *name) ? name : "Audio Stream", // stream description
+            &paspec,                                 // sample format spec
+            &pacmap                                  // channel map
         );
 
         if (h->stream == NULL) {
@@ -767,11 +767,11 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device)
 
     PULSEAUDIO_pa_threaded_mainloop_unlock(pulseaudio_threaded_mainloop);
 
-    /* We're (hopefully) ready to rock and roll. :-) */
+    // We're (hopefully) ready to rock and roll. :-)
     return retval;
 }
 
-/* device handles are device index + 1, cast to void*, so we never pass a NULL. */
+// device handles are device index + 1, cast to void*, so we never pass a NULL.
 
 static SDL_AudioFormat PulseFormatToSDLFormat(pa_sample_format_t format)
 {
@@ -817,11 +817,11 @@ static void SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last,
     PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);
 }
 
-/* This is called when PulseAudio adds a capture ("source") device. */
+// This is called when PulseAudio adds a capture ("source") device.
 // !!! FIXME: this is almost identical to SinkInfoCallback, merge the two.
 static void SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
 {
-    /* Maybe skip "monitor" sources. These are just output from other sinks. */
+    // Maybe skip "monitor" sources. These are just output from other sinks.
     if (i && (include_monitors || (i->monitor_of_sink == PA_INVALID_INDEX))) {
         const SDL_bool add = (SDL_bool) ((intptr_t)data);
 
@@ -843,13 +843,13 @@ static void SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_la
 static void ServerInfoCallback(pa_context *c, const pa_server_info *i, void *data)
 {
     if (!default_sink_path || (SDL_strcmp(i->default_sink_name, default_sink_path) != 0)) {
-        /*printf("DEFAULT SINK PATH CHANGED TO '%s'\n", i->default_sink_name);*/
+        //printf("DEFAULT SINK PATH CHANGED TO '%s'\n", i->default_sink_name);
         SDL_free(default_sink_path);
         default_sink_path = SDL_strdup(i->default_sink_name);
     }
 
     if (!default_source_path || (SDL_strcmp(i->default_source_name, default_source_path) != 0)) {
-        /*printf("DEFAULT SOURCE PATH CHANGED TO '%s'\n", i->default_source_name);*/
+        //printf("DEFAULT SOURCE PATH CHANGED TO '%s'\n", i->default_source_name);
         SDL_free(default_source_path);
         default_source_path = SDL_strdup(i->default_source_name);
     }
@@ -857,14 +857,14 @@ static void ServerInfoCallback(pa_context *c, const pa_server_info *i, void *dat
     PULSEAUDIO_pa_threaded_mainloop_signal(pulseaudio_threaded_mainloop, 0);
 }
 
-// This is called when PulseAudio has a device connected/removed/changed. */
+// This is called when PulseAudio has a device connected/removed/changed.
 static void HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)
 {
     const SDL_bool added = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW);
     const SDL_bool removed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE);
     const SDL_bool changed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_CHANGE);
 
-    if (added || removed || changed) { /* we only care about add/remove events. */
+    if (added || removed || changed) { // we only care about add/remove events.
         const SDL_bool sink = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK);
         const SDL_bool source = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE);
 
@@ -909,7 +909,7 @@ static int SDLCALL HotplugThread(void *data)
     PULSEAUDIO_pa_threaded_mainloop_lock(pulseaudio_threaded_mainloop);
     PULSEAUDIO_pa_context_set_subscribe_callback(pulseaudio_context, HotplugCallback, NULL);
 
-    /* don't WaitForPulseOperation on the subscription; when it's done we'll be able to get hotplug events, but waiting doesn't changing anything. */
+    // don't WaitForPulseOperation on the subscription; when it's done we'll be able to get hotplug events, but waiting doesn't changing anything.
     op = PULSEAUDIO_pa_context_subscribe(pulseaudio_context, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE | PA_SUBSCRIPTION_MASK_SERVER, NULL, NULL);
 
     SDL_PostSemaphore((SDL_Semaphore *) data);
@@ -961,7 +961,7 @@ static void PULSEAUDIO_DetectDevices(SDL_AudioDevice **default_output, SDL_Audio
         *default_capture = device;
     }
 
-    /* ok, we have a sane list, let's set up hotplug notifications now... */
+    // ok, we have a sane list, let's set up hotplug notifications now...
     SDL_AtomicSet(&pulseaudio_hotplug_thread_active, 1);
     pulseaudio_hotplug_thread = SDL_CreateThreadInternal(HotplugThread, "PulseHotplug", 256 * 1024, ready_sem);  // !!! FIXME: this can probably survive in significantly less stack space.
     SDL_WaitSemaphore(ready_sem);
@@ -1006,7 +1006,6 @@ static SDL_bool PULSEAUDIO_Init(SDL_AudioDriverImpl *impl)
 
     include_monitors = SDL_GetHintBoolean(SDL_HINT_AUDIO_INCLUDE_MONITORS, SDL_FALSE);
 
-    /* Set the function pointers */
     impl->DetectDevices = PULSEAUDIO_DetectDevices;
     impl->OpenDevice = PULSEAUDIO_OpenDevice;
     impl->PlayDevice = PULSEAUDIO_PlayDevice;
@@ -1021,11 +1020,11 @@ static SDL_bool PULSEAUDIO_Init(SDL_AudioDriverImpl *impl)
 
     impl->HasCaptureSupport = SDL_TRUE;
 
-    return SDL_TRUE; /* this audio target is available. */
+    return SDL_TRUE;
 }
 
 AudioBootStrap PULSEAUDIO_bootstrap = {
     "pulseaudio", "PulseAudio", PULSEAUDIO_Init, SDL_FALSE
 };
 
-#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO */
+#endif // SDL_AUDIO_DRIVER_PULSEAUDIO

src/audio/pulseaudio/SDL_pulseaudio.h

@@ -31,16 +31,16 @@ struct SDL_PrivateAudioData
 {
     char *device_name;
 
-    /* pulseaudio structures */
+    // pulseaudio structures
     pa_stream *stream;
 
-    /* Raw mixing buffer */
+    // Raw mixing buffer
     Uint8 *mixbuf;
 
-    int bytes_requested; /* bytes of data the hardware wants _now_. */
+    int bytes_requested; // bytes of data the hardware wants _now_.
 
     const Uint8 *capturebuf;
     int capturelen;
 };
 
-#endif /* SDL_pulseaudio_h_ */
+#endif // SDL_pulseaudio_h_

src/audio/qnx/SDL_qsa_audio.h

@@ -36,6 +36,5 @@ struct SDL_PrivateAudioData
     Uint8 *pcm_buf;  // Raw mixing buffer
 };
 
-#endif /* __SDL_QSA_AUDIO_H__ */
+#endif // __SDL_QSA_AUDIO_H__
 
-/* vi: set ts=4 sw=4 expandtab: */

src/audio/sndio/SDL_sndioaudio.h

@@ -35,4 +35,4 @@ struct SDL_PrivateAudioData
     struct pollfd *pfd;  // Polling structures for non-blocking sndio devices
 };
 
-#endif /* SDL_sndioaudio_h_ */
+#endif // SDL_sndioaudio_h_

src/audio/vita/SDL_vitaaudio.h

@@ -28,14 +28,14 @@
 
 struct SDL_PrivateAudioData
 {
-    /* The hardware input/output port. */
+    // The hardware input/output port.
     int port;
-    /* The raw allocated mixing buffer. */
+    // The raw allocated mixing buffer.
     Uint8 *rawbuf;
-    /* Individual mixing buffers. */
+    // Individual mixing buffers.
     Uint8 *mixbufs[NUM_BUFFERS];
-    /* Index of the next available mixing buffer. */
+    // Index of the next available mixing buffer.
     int next_buffer;
 };
 
-#endif /* SDL_vitaaudio_h */
+#endif // SDL_vitaaudio_h

src/audio/wasapi/SDL_wasapi.h

@@ -45,7 +45,7 @@ struct SDL_PrivateAudioData
     void *activation_handler;
 };
 
-/* win32 and winrt implementations call into these. */
+// win32 and winrt implementations call into these.
 int WASAPI_PrepDevice(SDL_AudioDevice *device);
 void WASAPI_DisconnectDevice(SDL_AudioDevice *device);  // don't hold the device lock when calling this!
 
@@ -54,7 +54,7 @@ void WASAPI_DisconnectDevice(SDL_AudioDevice *device);  // don't hold the device
 typedef int (*ManagementThreadTask)(void *userdata);
 int WASAPI_ProxyToManagementThread(ManagementThreadTask task, void *userdata, int *wait_until_complete);
 
-/* These are functions that are implemented differently for Windows vs WinRT. */
+// These are functions that are implemented differently for Windows vs WinRT.
 // UNLESS OTHERWISE NOTED THESE ALL HAPPEN ON THE MANAGEMENT THREAD.
 int WASAPI_PlatformInit(void);
 void WASAPI_PlatformDeinit(void);
@@ -70,4 +70,4 @@ void WASAPI_PlatformFreeDeviceHandle(SDL_AudioDevice *device);
 }
 #endif
 
-#endif /* SDL_wasapi_h_ */
+#endif // SDL_wasapi_h_

src/audio/wasapi/SDL_wasapi_win32.c

@@ -37,7 +37,7 @@
 
 #include "SDL_wasapi.h"
 
-/* handle to Avrt.dll--Vista and later!--for flagging the callback thread as "Pro Audio" (low latency). */
+// handle to Avrt.dll--Vista and later!--for flagging the callback thread as "Pro Audio" (low latency).
 static HMODULE libavrt = NULL;
 typedef HANDLE(WINAPI *pfnAvSetMmThreadCharacteristicsW)(LPCWSTR, LPDWORD);
 typedef BOOL(WINAPI *pfnAvRevertMmThreadCharacteristics)(HANDLE);
@@ -46,7 +46,7 @@ static pfnAvRevertMmThreadCharacteristics pAvRevertMmThreadCharacteristics = NUL
 
 static SDL_bool immdevice_initialized = SDL_FALSE;
 
-/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
+// Some GUIDs we need to know without linking to libraries that aren't available before Vista.
 static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, { 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
 
 int WASAPI_PlatformInit(void)
@@ -59,7 +59,7 @@ int WASAPI_PlatformInit(void)
 
     immdevice_initialized = SDL_TRUE;
 
-    libavrt = LoadLibrary(TEXT("avrt.dll")); /* this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now! */
+    libavrt = LoadLibrary(TEXT("avrt.dll")); // this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now!
     if (libavrt) {
         pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW)GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
         pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics)GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
@@ -98,12 +98,12 @@ void WASAPI_PlatformDeinitializeStart(void)
 
 void WASAPI_PlatformThreadInit(SDL_AudioDevice *device)
 {
-    /* this thread uses COM. */
-    if (SUCCEEDED(WIN_CoInitialize())) { /* can't report errors, hope it worked! */
+    // this thread uses COM.
+    if (SUCCEEDED(WIN_CoInitialize())) { // can't report errors, hope it worked!
         device->hidden->coinitialized = SDL_TRUE;
     }
 
-    /* Set this thread to very high "Pro Audio" priority. */
+    // Set this thread to very high "Pro Audio" priority.
     if (pAvSetMmThreadCharacteristicsW) {
         DWORD idx = 0;
         device->hidden->task = pAvSetMmThreadCharacteristicsW(L"Pro Audio", &idx);
@@ -115,7 +115,7 @@ void WASAPI_PlatformThreadInit(SDL_AudioDevice *device)
 
 void WASAPI_PlatformThreadDeinit(SDL_AudioDevice *device)
 {
-    /* Set this thread back to normal priority. */
+    // Set this thread back to normal priority.
     if (device->hidden->task && pAvRevertMmThreadCharacteristics) {
         pAvRevertMmThreadCharacteristics(device->hidden->task);
         device->hidden->task = NULL;
@@ -132,10 +132,10 @@ int WASAPI_ActivateDevice(SDL_AudioDevice *device)
     IMMDevice *immdevice = NULL;
     if (SDL_IMMDevice_Get(device, &immdevice, device->iscapture) < 0) {
         device->hidden->client = NULL;
-        return -1; /* This is already set by SDL_IMMDevice_Get */
+        return -1; // This is already set by SDL_IMMDevice_Get
     }
 
-    /* this is _not_ async in standard win32, yay! */
+    // this is _not_ async in standard win32, yay!
     HRESULT ret = IMMDevice_Activate(immdevice, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&device->hidden->client);
     IMMDevice_Release(immdevice);
 
@@ -145,11 +145,11 @@ int WASAPI_ActivateDevice(SDL_AudioDevice *device)
     }
 
     SDL_assert(device->hidden->client != NULL);
-    if (WASAPI_PrepDevice(device) == -1) { /* not async, fire it right away. */
+    if (WASAPI_PrepDevice(device) == -1) { // not async, fire it right away.
         return -1;
     }
 
-    return 0; /* good to go. */
+    return 0; // good to go.
 }
 
 void WASAPI_EnumerateEndpoints(SDL_AudioDevice **default_output, SDL_AudioDevice **default_capture)
@@ -159,7 +159,7 @@ void WASAPI_EnumerateEndpoints(SDL_AudioDevice **default_output, SDL_AudioDevice
 
 void WASAPI_PlatformDeleteActivationHandler(void *handler)
 {
-    /* not asynchronous. */
+    // not asynchronous.
     SDL_assert(!"This function should have only been called on WinRT.");
 }
 
@@ -168,4 +168,4 @@ void WASAPI_PlatformFreeDeviceHandle(SDL_AudioDevice *device)
     SDL_IMMDevice_FreeDeviceHandle(device);
 }
 
-#endif /* SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__) */
+#endif // SDL_AUDIO_DRIVER_WASAPI && !defined(__WINRT__)