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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.
This commit is contained in:
Ryan C. Gordon 2023-10-18 15:35:09 -04:00
parent 0ff67dc21b
commit 9d7c57234a
No known key found for this signature in database
GPG Key ID: FA148B892AB48044
37 changed files with 378 additions and 384 deletions
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4
build-scripts/gen_audio_channel_conversion.c
View File

@ -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"

4
build-scripts/gen_audio_resampler_filter.c
View File

@ -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"

58
src/audio/SDL_audio_channel_converters.h
View File

@ -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) {

2
src/audio/SDL_audio_resampler_filter.h
View File

@ -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

10
src/audio/SDL_audiodev.c
View File

@ -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

6
src/audio/SDL_audiodev_c.h
View File

@ -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_

8
src/audio/SDL_audioresample.c
View File

@ -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)

177
src/audio/SDL_audiotypecvt.c
View File

@ -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;

12
src/audio/SDL_mixer.c
View File

@ -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");
}

2
src/audio/aaudio/SDL_aaudio.h
View File

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

26
src/audio/aaudio/SDL_aaudiofuncs.h
View File

@ -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

8
src/audio/alsa/SDL_alsa_audio.h
View File

@ -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_

2
src/audio/android/SDL_androidaudio.h
View File

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

4
src/audio/coreaudio/SDL_coreaudio.h
View File

@ -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_

45
src/audio/coreaudio/SDL_coreaudio.m
View File

@ -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

4
src/audio/disk/SDL_diskaudio.h
View File

@ -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_

25
src/audio/dsp/SDL_dspaudio.c
View File

@ -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

6
src/audio/dsp/SDL_dspaudio.h
View File

@ -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_

2
src/audio/emscripten/SDL_emscriptenaudio.h
View File

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

2
src/audio/haiku/SDL_haikuaudio.h
View File

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

37
src/audio/jack/SDL_jackaudio.c
View File

@ -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

2
src/audio/jack/SDL_jackaudio.h
View File

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

6
src/audio/n3ds/SDL_n3dsaudio.h
View File

@ -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

2
src/audio/netbsd/SDL_netbsdaudio.c
View File

@ -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!
}
}

10
src/audio/netbsd/SDL_netbsdaudio.h
View File

@ -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_

2
src/audio/openslES/SDL_openslES.h
View File

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

108
src/audio/pipewire/SDL_pipewire.c
View File

@ -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

4
src/audio/pipewire/SDL_pipewire.h
View File

@ -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_

10
src/audio/ps2/SDL_ps2audio.h
View File

@ -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_

10
src/audio/psp/SDL_pspaudio.h
View File

@ -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_

109
src/audio/pulseaudio/SDL_pulseaudio.c
View File

@ -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

8
src/audio/pulseaudio/SDL_pulseaudio.h
View File

@ -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_

3
src/audio/qnx/SDL_qsa_audio.h
View File

@ -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: */

2
src/audio/sndio/SDL_sndioaudio.h
View File

@ -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_

10
src/audio/vita/SDL_vitaaudio.h
View File

@ -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

6
src/audio/wasapi/SDL_wasapi.h
View File

@ -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_

26
src/audio/wasapi/SDL_wasapi_win32.c
View File

@ -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__)