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Audio types have the same naming convention as other SDL endian types, e.g. [S|U][BITS][LE|BE]

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Native endian types have no LE/BE suffix
This commit is contained in:
Sam Lantinga
2023-09-04 09:17:29 -07:00
parent 36b5f3e35c
commit 233789b0d1
28 changed files with 258 additions and 224 deletions
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14
test/testaudio.c
View File

@@ -416,12 +416,12 @@ static const char *AudioFmtToString(const SDL_AudioFormat fmt)
#define FMTCASE(x) case SDL_AUDIO_##x: return #x
FMTCASE(U8);
FMTCASE(S8);
FMTCASE(S16LSB);
FMTCASE(S16MSB);
FMTCASE(S32LSB);
FMTCASE(S32MSB);
FMTCASE(F32LSB);
FMTCASE(F32MSB);
FMTCASE(S16LE);
FMTCASE(S16BE);
FMTCASE(S32LE);
FMTCASE(S32BE);
FMTCASE(F32LE);
FMTCASE(F32BE);
#undef FMTCASE
}
return "?";
@@ -965,7 +965,7 @@ static void Loop(void)
case SDL_EVENT_AUDIO_DEVICE_ADDED:
CreatePhysicalDeviceThing(event.adevice.which, event.adevice.iscapture);
break;
case SDL_EVENT_AUDIO_DEVICE_REMOVED: {
const SDL_AudioDeviceID which = event.adevice.which;
Thing *i, *next;

12
test/testaudiostreamdynamicresample.c
View File

@@ -180,12 +180,12 @@ static const char *AudioFmtToString(const SDL_AudioFormat fmt)
#define FMTCASE(x) case SDL_AUDIO_##x: return #x
FMTCASE(U8);
FMTCASE(S8);
FMTCASE(S16LSB);
FMTCASE(S16MSB);
FMTCASE(S32LSB);
FMTCASE(S32MSB);
FMTCASE(F32LSB);
FMTCASE(F32MSB);
FMTCASE(S16LE);
FMTCASE(S16BE);
FMTCASE(S32LE);
FMTCASE(S32BE);
FMTCASE(F32LE);
FMTCASE(F32BE);
#undef FMTCASE
}
return "?";

62
test/testautomation_audio.c
View File

@@ -176,13 +176,13 @@ static int audio_initOpenCloseQuitAudio(void *arg)
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16SYS;
desired.format = SDL_AUDIO_S16;
desired.channels = 2;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32SYS;
desired.format = SDL_AUDIO_F32;
desired.channels = 2;
break;
}
@@ -261,14 +261,14 @@ static int audio_pauseUnpauseAudio(void *arg)
case 0:
/* Set standard desired spec */
desired.freq = 22050;
desired.format = SDL_AUDIO_S16SYS;
desired.format = SDL_AUDIO_S16;
desired.channels = 2;
break;
case 1:
/* Set custom desired spec */
desired.freq = 48000;
desired.format = SDL_AUDIO_F32SYS;
desired.format = SDL_AUDIO_F32;
desired.channels = 2;
break;
}
@@ -441,18 +441,34 @@ static int audio_printCurrentAudioDriver(void *arg)
}
/* Definition of all formats, channels, and frequencies used to test audio conversions */
static SDL_AudioFormat g_audioFormats[] = { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S16SYS, SDL_AUDIO_S16,
SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_S32SYS, SDL_AUDIO_S32,
SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_F32SYS, SDL_AUDIO_F32 };
static const char *g_audioFormatsVerbose[] = { "SDL_AUDIO_S8", "SDL_AUDIO_U8", "SDL_AUDIO_S16LSB", "SDL_AUDIO_S16MSB", "SDL_AUDIO_S16SYS", "SDL_AUDIO_S16",
"SDL_AUDIO_S32LSB", "SDL_AUDIO_S32MSB", "SDL_AUDIO_S32SYS", "SDL_AUDIO_S32",
"SDL_AUDIO_F32LSB", "SDL_AUDIO_F32MSB", "SDL_AUDIO_F32SYS", "SDL_AUDIO_F32" };
static SDL_AudioFormat g_audioFormats[] = {
SDL_AUDIO_S8, SDL_AUDIO_U8,
SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_S16,
SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_S32,
SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, SDL_AUDIO_F32
};
static const char *g_audioFormatsVerbose[] = {
"SDL_AUDIO_S8", "SDL_AUDIO_U8",
"SDL_AUDIO_S16LE", "SDL_AUDIO_S16BE", "SDL_AUDIO_S16",
"SDL_AUDIO_S32LE", "SDL_AUDIO_S32BE", "SDL_AUDIO_S32",
"SDL_AUDIO_F32LE", "SDL_AUDIO_F32BE", "SDL_AUDIO_F32"
};
static const int g_numAudioFormats = SDL_arraysize(g_audioFormats);
static Uint8 g_audioChannels[] = { 1, 2, 4, 6 };
static const int g_numAudioChannels = SDL_arraysize(g_audioChannels);
static int g_audioFrequencies[] = { 11025, 22050, 44100, 48000 };
static const int g_numAudioFrequencies = SDL_arraysize(g_audioFrequencies);
/* Verify the audio formats are laid out as expected */
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_U8_FORMAT, SDL_AUDIO_U8 == SDL_AUDIO_BITSIZE(8));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S8_FORMAT, SDL_AUDIO_S8 == (SDL_AUDIO_BITSIZE(8) | SDL_AUDIO_MASK_SIGNED));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S16LE_FORMAT, SDL_AUDIO_S16LE == (SDL_AUDIO_BITSIZE(16) | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_LIL_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S16BE_FORMAT, SDL_AUDIO_S16BE == (SDL_AUDIO_BITSIZE(16) | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BIG_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S32LE_FORMAT, SDL_AUDIO_S32LE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_LIL_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_S32BE_FORMAT, SDL_AUDIO_S32BE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BIG_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_F32LE_FORMAT, SDL_AUDIO_F32LE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_FLOAT | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_LIL_ENDIAN));
SDL_COMPILE_TIME_ASSERT(SDL_AUDIO_F32BE_FORMAT, SDL_AUDIO_F32BE == (SDL_AUDIO_BITSIZE(32) | SDL_AUDIO_MASK_FLOAT | SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BIG_ENDIAN));
/**
* \brief Builds various audio conversion structures
*
@@ -466,7 +482,7 @@ static int audio_buildAudioStream(void *arg)
int i, ii, j, jj, k, kk;
/* No conversion needed */
spec1.format = SDL_AUDIO_S16LSB;
spec1.format = SDL_AUDIO_S16LE;
spec1.channels = 2;
spec1.freq = 22050;
stream = SDL_CreateAudioStream(&spec1, &spec1);
@@ -478,7 +494,7 @@ static int audio_buildAudioStream(void *arg)
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LSB;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
stream = SDL_CreateAudioStream(&spec1, &spec2);
@@ -533,7 +549,7 @@ static int audio_buildAudioStreamNegative(void *arg)
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LSB;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
@@ -546,7 +562,7 @@ static int audio_buildAudioStreamNegative(void *arg)
spec1.format = SDL_AUDIO_S8;
spec1.channels = 1;
spec1.freq = 22050;
spec2.format = SDL_AUDIO_S16LSB;
spec2.format = SDL_AUDIO_S16LE;
spec2.channels = 2;
spec2.freq = 44100;
@@ -842,14 +858,14 @@ static int audio_resampleLoss(void *arg)
SDLTest_AssertPass("Test resampling of %i s %i Hz %f phase sine wave from sampling rate of %i Hz to %i Hz",
spec->time, spec->freq, spec->phase, spec->rate_in, spec->rate_out);
tmpspec1.format = SDL_AUDIO_F32SYS;
tmpspec1.format = SDL_AUDIO_F32;
tmpspec1.channels = num_channels;
tmpspec1.freq = spec->rate_in;
tmpspec2.format = SDL_AUDIO_F32SYS;
tmpspec2.format = SDL_AUDIO_F32;
tmpspec2.channels = num_channels;
tmpspec2.freq = spec->rate_out;
stream = SDL_CreateAudioStream(&tmpspec1, &tmpspec2);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32SYS, 1, %i, SDL_AUDIO_F32SYS, 1, %i)", spec->rate_in, spec->rate_out);
SDLTest_AssertPass("Call to SDL_CreateAudioStream(SDL_AUDIO_F32, 1, %i, SDL_AUDIO_F32, 1, %i)", spec->rate_in, spec->rate_out);
SDLTest_AssertCheck(stream != NULL, "Expected SDL_CreateAudioStream to succeed.");
if (stream == NULL) {
return TEST_ABORTED;
@@ -929,7 +945,7 @@ static int audio_resampleLoss(void *arg)
signal_to_noise, spec->signal_to_noise);
SDLTest_AssertCheck(max_error <= spec->max_error, "Maximum conversion error %f should be no more than %f.",
max_error, spec->max_error);
if (++num_channels > max_channels) {
num_channels = min_channels;
++spec_idx;
@@ -946,7 +962,7 @@ static int audio_resampleLoss(void *arg)
*/
static int audio_convertAccuracy(void *arg)
{
static SDL_AudioFormat formats[] = { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16SYS, SDL_AUDIO_S32SYS };
static SDL_AudioFormat formats[] = { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16, SDL_AUDIO_S32 };
static const char* format_names[] = { "S8", "U8", "S16", "S32" };
int src_num = 65537 + 2048 + 48 + 256 + 100000;
@@ -1020,7 +1036,7 @@ static int audio_convertAccuracy(void *arg)
float target_max_delta = (bits > 23) ? 0.0f : (1.0f / (float)(1 << bits));
float target_min_delta = -target_max_delta;
src_spec.format = SDL_AUDIO_F32SYS;
src_spec.format = SDL_AUDIO_F32;
src_spec.channels = 1;
src_spec.freq = 44100;
@@ -1099,15 +1115,15 @@ static int audio_formatChange(void *arg)
double target_signal_to_noise = 75.0;
int sine_freq = 500;
spec1.format = SDL_AUDIO_F32SYS;
spec1.format = SDL_AUDIO_F32;
spec1.channels = 1;
spec1.freq = 20000;
spec2.format = SDL_AUDIO_F32SYS;
spec2.format = SDL_AUDIO_F32;
spec2.channels = 1;
spec2.freq = 40000;
spec3.format = SDL_AUDIO_F32SYS;
spec3.format = SDL_AUDIO_F32;
spec3.channels = 1;
spec3.freq = 80000;

2
test/testsurround.c
View File

@@ -205,7 +205,7 @@ int main(int argc, char *argv[])
SDL_Log(" (%d channels)\n", spec.channels);
spec.freq = SAMPLE_RATE_HZ;
spec.format = SDL_AUDIO_S16SYS;
spec.format = SDL_AUDIO_S16;
/* These are used by the fill_buffer callback */
total_channels = spec.channels;