From 233789b0d1789adef4f9bd2d9fff72c11f710a28 Mon Sep 17 00:00:00 2001 From: Sam Lantinga Date: Mon, 4 Sep 2023 09:17:29 -0700 Subject: [PATCH] Audio types have the same naming convention as other SDL endian types, e.g. [S|U][BITS][LE|BE] Native endian types have no LE/BE suffix --- build-scripts/SDL_migration.cocci | 24 +++++------ docs/README-migration.md | 24 +++++------ include/SDL3/SDL_audio.h | 43 +++++++++---------- include/SDL3/SDL_oldnames.h | 48 ++++++++++----------- src/audio/SDL_audio.c | 46 ++++++++++---------- src/audio/SDL_audiocvt.c | 26 +++++------ src/audio/SDL_mixer.c | 12 +++--- src/audio/SDL_wave.c | 8 ++-- src/audio/aaudio/SDL_aaudio.c | 10 ++--- src/audio/alsa/SDL_alsa_audio.c | 12 +++--- src/audio/coreaudio/SDL_coreaudio.m | 12 +++--- src/audio/dsp/SDL_dspaudio.c | 4 +- src/audio/haiku/SDL_haikuaudio.cc | 12 +++--- src/audio/jack/SDL_jackaudio.c | 2 +- src/audio/netbsd/SDL_netbsdaudio.c | 8 ++-- src/audio/openslES/SDL_openslES.c | 6 +-- src/audio/pipewire/SDL_pipewire.c | 12 +++--- src/audio/psp/SDL_pspaudio.c | 2 +- src/audio/pulseaudio/SDL_pulseaudio.c | 24 +++++------ src/audio/qnx/SDL_qsa_audio.c | 2 +- src/audio/sndio/SDL_sndioaudio.c | 8 ++-- src/audio/vita/SDL_vitaaudio.c | 2 +- src/core/windows/SDL_windows.c | 12 +++--- src/test/SDL_test_common.c | 33 +++++++++++--- test/testaudio.c | 14 +++--- test/testaudiostreamdynamicresample.c | 12 +++--- test/testautomation_audio.c | 62 +++++++++++++++++---------- test/testsurround.c | 2 +- 28 files changed, 258 insertions(+), 224 deletions(-) diff --git a/build-scripts/SDL_migration.cocci b/build-scripts/SDL_migration.cocci index a2243cff3..1ade7e891 100644 --- a/build-scripts/SDL_migration.cocci +++ b/build-scripts/SDL_migration.cocci @@ -2591,51 +2591,51 @@ typedef SDL_cond, SDL_Condition; @@ @@ - AUDIO_F32 -+ SDL_AUDIO_F32 ++ SDL_AUDIO_F32LE @@ @@ - AUDIO_F32LSB -+ SDL_AUDIO_F32LSB ++ SDL_AUDIO_F32LE @@ @@ - AUDIO_F32MSB -+ SDL_AUDIO_F32MSB ++ SDL_AUDIO_F32BE @@ @@ - AUDIO_F32SYS -+ SDL_AUDIO_F32SYS ++ SDL_AUDIO_F32 @@ @@ - AUDIO_S16 -+ SDL_AUDIO_S16 ++ SDL_AUDIO_S16LE @@ @@ - AUDIO_S16LSB -+ SDL_AUDIO_S16LSB ++ SDL_AUDIO_S16LE @@ @@ - AUDIO_S16MSB -+ SDL_AUDIO_S16MSB ++ SDL_AUDIO_S16BE @@ @@ - AUDIO_S16SYS -+ SDL_AUDIO_S16SYS ++ SDL_AUDIO_S16 @@ @@ - AUDIO_S32 -+ SDL_AUDIO_S32 ++ SDL_AUDIO_S32LE @@ @@ - AUDIO_S32LSB -+ SDL_AUDIO_S32LSB ++ SDL_AUDIO_S32LE @@ @@ - AUDIO_S32MSB -+ SDL_AUDIO_S32MSB ++ SDL_AUDIO_S32BE @@ @@ - AUDIO_S32SYS -+ SDL_AUDIO_S32SYS ++ SDL_AUDIO_S32 @@ @@ - AUDIO_S8 diff --git a/docs/README-migration.md b/docs/README-migration.md index 2f00a5d70..5347eca8e 100644 --- a/docs/README-migration.md +++ b/docs/README-migration.md @@ -255,18 +255,18 @@ The following functions have been removed: * SDL_GetQueuedAudioSize() The following symbols have been renamed: -* AUDIO_F32 => SDL_AUDIO_F32 -* AUDIO_F32LSB => SDL_AUDIO_F32LSB -* AUDIO_F32MSB => SDL_AUDIO_F32MSB -* AUDIO_F32SYS => SDL_AUDIO_F32SYS -* AUDIO_S16 => SDL_AUDIO_S16 -* AUDIO_S16LSB => SDL_AUDIO_S16LSB -* AUDIO_S16MSB => SDL_AUDIO_S16MSB -* AUDIO_S16SYS => SDL_AUDIO_S16SYS -* AUDIO_S32 => SDL_AUDIO_S32 -* AUDIO_S32LSB => SDL_AUDIO_S32LSB -* AUDIO_S32MSB => SDL_AUDIO_S32MSB -* AUDIO_S32SYS => SDL_AUDIO_S32SYS +* AUDIO_F32 => SDL_AUDIO_F32LE +* AUDIO_F32LSB => SDL_AUDIO_F32LE +* AUDIO_F32MSB => SDL_AUDIO_F32BE +* AUDIO_F32SYS => SDL_AUDIO_F32 +* AUDIO_S16 => SDL_AUDIO_S16LE +* AUDIO_S16LSB => SDL_AUDIO_S16LE +* AUDIO_S16MSB => SDL_AUDIO_S16BE +* AUDIO_S16SYS => SDL_AUDIO_S16 +* AUDIO_S32 => SDL_AUDIO_S32LE +* AUDIO_S32LSB => SDL_AUDIO_S32LE +* AUDIO_S32MSB => SDL_AUDIO_S32BE +* AUDIO_S32SYS => SDL_AUDIO_S32 * AUDIO_S8 => SDL_AUDIO_S8 * AUDIO_U8 => SDL_AUDIO_U8 diff --git a/include/SDL3/SDL_audio.h b/include/SDL3/SDL_audio.h index a8270a0af..e88934f0b 100644 --- a/include/SDL3/SDL_audio.h +++ b/include/SDL3/SDL_audio.h @@ -80,15 +80,17 @@ typedef Uint16 SDL_AudioFormat; /* @{ */ #define SDL_AUDIO_MASK_BITSIZE (0xFF) -#define SDL_AUDIO_MASK_DATATYPE (1<<8) -#define SDL_AUDIO_MASK_ENDIAN (1<<12) +#define SDL_AUDIO_MASK_FLOAT (1<<8) +#define SDL_AUDIO_MASK_LIL_ENDIAN (1<<12) +#define SDL_AUDIO_MASK_BIG_ENDIAN (1<<13) +#define SDL_AUDIO_MASK_ENDIAN (SDL_AUDIO_MASK_BIG_ENDIAN|SDL_AUDIO_MASK_LIL_ENDIAN) #define SDL_AUDIO_MASK_SIGNED (1<<15) -#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) -#define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) -#define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) -#define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) +#define SDL_AUDIO_BITSIZE(x) ((x) & SDL_AUDIO_MASK_BITSIZE) +#define SDL_AUDIO_ISFLOAT(x) ((x) & SDL_AUDIO_MASK_FLOAT) +#define SDL_AUDIO_ISBIGENDIAN(x) (((x) & SDL_AUDIO_MASK_ENDIAN) == SDL_AUDIO_MASK_BIG_ENDIAN) +#define SDL_AUDIO_ISLITTLEENDIAN(x) (((x) & SDL_AUDIO_MASK_ENDIAN) == SDL_AUDIO_MASK_LIL_ENDIAN) +#define SDL_AUDIO_ISSIGNED(x) ((x) & SDL_AUDIO_MASK_SIGNED) #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) -#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) /** @@ -99,27 +101,24 @@ typedef Uint16 SDL_AudioFormat; /* @{ */ #define SDL_AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ #define SDL_AUDIO_S8 0x8008 /**< Signed 8-bit samples */ -#define SDL_AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ -#define SDL_AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ -#define SDL_AUDIO_S16 SDL_AUDIO_S16LSB +#define SDL_AUDIO_S16LE 0x9010 /**< Signed 16-bit samples */ +#define SDL_AUDIO_S16BE 0xA010 /**< As above, but big-endian byte order */ /* @} */ /** * \name int32 support */ /* @{ */ -#define SDL_AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ -#define SDL_AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ -#define SDL_AUDIO_S32 SDL_AUDIO_S32LSB +#define SDL_AUDIO_S32LE 0x9020 /**< 32-bit integer samples */ +#define SDL_AUDIO_S32BE 0xA020 /**< As above, but big-endian byte order */ /* @} */ /** * \name float32 support */ /* @{ */ -#define SDL_AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ -#define SDL_AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ -#define SDL_AUDIO_F32 SDL_AUDIO_F32LSB +#define SDL_AUDIO_F32LE 0x9120 /**< 32-bit floating point samples */ +#define SDL_AUDIO_F32BE 0xA120 /**< As above, but big-endian byte order */ /* @} */ /** @@ -127,13 +126,13 @@ typedef Uint16 SDL_AudioFormat; */ /* @{ */ #if SDL_BYTEORDER == SDL_LIL_ENDIAN -#define SDL_AUDIO_S16SYS SDL_AUDIO_S16LSB -#define SDL_AUDIO_S32SYS SDL_AUDIO_S32LSB -#define SDL_AUDIO_F32SYS SDL_AUDIO_F32LSB +#define SDL_AUDIO_S16 SDL_AUDIO_S16LE +#define SDL_AUDIO_S32 SDL_AUDIO_S32LE +#define SDL_AUDIO_F32 SDL_AUDIO_F32LE #else -#define SDL_AUDIO_S16SYS SDL_AUDIO_S16MSB -#define SDL_AUDIO_S32SYS SDL_AUDIO_S32MSB -#define SDL_AUDIO_F32SYS SDL_AUDIO_F32MSB +#define SDL_AUDIO_S16 SDL_AUDIO_S16BE +#define SDL_AUDIO_S32 SDL_AUDIO_S32BE +#define SDL_AUDIO_F32 SDL_AUDIO_F32BE #endif /* @} */ diff --git a/include/SDL3/SDL_oldnames.h b/include/SDL3/SDL_oldnames.h index e58ec3a44..0ecea4073 100644 --- a/include/SDL3/SDL_oldnames.h +++ b/include/SDL3/SDL_oldnames.h @@ -43,18 +43,18 @@ #define SDL_atomic_t SDL_AtomicInt /* ##SDL_audio.h */ -#define AUDIO_F32 SDL_AUDIO_F32 -#define AUDIO_F32LSB SDL_AUDIO_F32LSB -#define AUDIO_F32MSB SDL_AUDIO_F32MSB -#define AUDIO_F32SYS SDL_AUDIO_F32SYS -#define AUDIO_S16 SDL_AUDIO_S16 -#define AUDIO_S16LSB SDL_AUDIO_S16LSB -#define AUDIO_S16MSB SDL_AUDIO_S16MSB -#define AUDIO_S16SYS SDL_AUDIO_S16SYS -#define AUDIO_S32 SDL_AUDIO_S32 -#define AUDIO_S32LSB SDL_AUDIO_S32LSB -#define AUDIO_S32MSB SDL_AUDIO_S32MSB -#define AUDIO_S32SYS SDL_AUDIO_S32SYS +#define AUDIO_F32 SDL_AUDIO_F32LE +#define AUDIO_F32LSB SDL_AUDIO_F32LE +#define AUDIO_F32MSB SDL_AUDIO_F32BE +#define AUDIO_F32SYS SDL_AUDIO_F32 +#define AUDIO_S16 SDL_AUDIO_S16LE +#define AUDIO_S16LSB SDL_AUDIO_S16LE +#define AUDIO_S16MSB SDL_AUDIO_S16BE +#define AUDIO_S16SYS SDL_AUDIO_S16 +#define AUDIO_S32 SDL_AUDIO_S32LE +#define AUDIO_S32LSB SDL_AUDIO_S32LE +#define AUDIO_S32MSB SDL_AUDIO_S32BE +#define AUDIO_S32SYS SDL_AUDIO_S32 #define AUDIO_S8 SDL_AUDIO_S8 #define AUDIO_U8 SDL_AUDIO_U8 #define SDL_AudioStreamAvailable SDL_GetAudioStreamAvailable @@ -494,18 +494,18 @@ #elif !defined(SDL_DISABLE_OLD_NAMES) /* ##SDL_audio.h */ -#define AUDIO_F32 AUDIO_F32_renamed_SDL_AUDIO_F32 -#define AUDIO_F32LSB AUDIO_F32LSB_renamed_SDL_AUDIO_F32LSB -#define AUDIO_F32MSB AUDIO_F32MSB_renamed_SDL_AUDIO_F32MSB -#define AUDIO_F32SYS AUDIO_F32SYS_renamed_SDL_AUDIO_F32SYS -#define AUDIO_S16 AUDIO_S16_renamed_SDL_AUDIO_S16 -#define AUDIO_S16LSB AUDIO_S16LSB_renamed_SDL_AUDIO_S16LSB -#define AUDIO_S16MSB AUDIO_S16MSB_renamed_SDL_AUDIO_S16MSB -#define AUDIO_S16SYS AUDIO_S16SYS_renamed_SDL_AUDIO_S16SYS -#define AUDIO_S32 AUDIO_S32_renamed_SDL_AUDIO_S32 -#define AUDIO_S32LSB AUDIO_S32LSB_renamed_SDL_AUDIO_S32LSB -#define AUDIO_S32MSB AUDIO_S32MSB_renamed_SDL_AUDIO_S32MSB -#define AUDIO_S32SYS AUDIO_S32SYS_renamed_SDL_AUDIO_S32SYS +#define AUDIO_F32 AUDIO_F32_renamed_SDL_AUDIO_F32LE +#define AUDIO_F32LSB AUDIO_F32LSB_renamed_SDL_AUDIO_F32LE +#define AUDIO_F32MSB AUDIO_F32MSB_renamed_SDL_AUDIO_F32BE +#define AUDIO_F32SYS AUDIO_F32SYS_renamed_SDL_AUDIO_F32 +#define AUDIO_S16 AUDIO_S16_renamed_SDL_AUDIO_S16LE +#define AUDIO_S16LSB AUDIO_S16LSB_renamed_SDL_AUDIO_S16LE +#define AUDIO_S16MSB AUDIO_S16MSB_renamed_SDL_AUDIO_S16BE +#define AUDIO_S16SYS AUDIO_S16SYS_renamed_SDL_AUDIO_S16 +#define AUDIO_S32 AUDIO_S32_renamed_SDL_AUDIO_S32LE +#define AUDIO_S32LSB AUDIO_S32LSB_renamed_SDL_AUDIO_S32LE +#define AUDIO_S32MSB AUDIO_S32MSB_renamed_SDL_AUDIO_S32BE +#define AUDIO_S32SYS AUDIO_S32SYS_renamed_SDL_AUDIO_S32 #define AUDIO_S8 AUDIO_S8_renamed_SDL_AUDIO_S8 #define AUDIO_U8 AUDIO_U8_renamed_SDL_AUDIO_U8 #define SDL_AudioStreamAvailable SDL_AudioStreamAvailable_renamed_SDL_GetAudioStreamAvailable diff --git a/src/audio/SDL_audio.c b/src/audio/SDL_audio.c index de2e3a4f9..5171d326d 100644 --- a/src/audio/SDL_audio.c +++ b/src/audio/SDL_audio.c @@ -765,17 +765,17 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device) case MIXSTRATEGY_MIX: { //SDL_Log("MIX STRATEGY: MIX"); - float *mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32SYS) ? device_buffer : device->mix_buffer); + float *mix_buffer = (float *) ((device->spec.format == SDL_AUDIO_F32) ? device_buffer : device->mix_buffer); const int needed_samples = buffer_size / (SDL_AUDIO_BITSIZE(device->spec.format) / 8); const int work_buffer_size = needed_samples * sizeof (float); SDL_AudioSpec outspec; SDL_assert(work_buffer_size <= device->work_buffer_size); - outspec.format = SDL_AUDIO_F32SYS; + outspec.format = SDL_AUDIO_F32; outspec.channels = device->spec.channels; outspec.freq = device->spec.freq; - outspec.format = SDL_AUDIO_F32SYS; + outspec.format = SDL_AUDIO_F32; SDL_memset(mix_buffer, '\0', buffer_size); // start with silence. @@ -795,7 +795,7 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device) retval = SDL_FALSE; break; } else if (br > 0) { // it's okay if we get less than requested, we mix what we have. - if (SDL_MixAudioFormat((Uint8 *) mix_buffer, device->work_buffer, SDL_AUDIO_F32SYS, br, SDL_MIX_MAXVOLUME) < 0) { // !!! FIXME: allow streams to specify gain? + if (SDL_MixAudioFormat((Uint8 *) mix_buffer, device->work_buffer, SDL_AUDIO_F32, br, SDL_MIX_MAXVOLUME) < 0) { // !!! FIXME: allow streams to specify gain? SDL_assert(!"This shouldn't happen."); retval = SDL_FALSE; // uh...? break; @@ -806,8 +806,8 @@ SDL_bool SDL_OutputAudioThreadIterate(SDL_AudioDevice *device) if (((Uint8 *) mix_buffer) != device_buffer) { // !!! FIXME: we can't promise the device buf is aligned/padded for SIMD. - //ConvertAudio(needed_samples * device->spec.channels, mix_buffer, SDL_AUDIO_F32SYS, device->spec.channels, device_buffer, device->spec.format, device->spec.channels, device->work_buffer); - ConvertAudio(needed_samples / device->spec.channels, mix_buffer, SDL_AUDIO_F32SYS, device->spec.channels, device->work_buffer, device->spec.format, device->spec.channels, NULL); + //ConvertAudio(needed_samples * device->spec.channels, mix_buffer, SDL_AUDIO_F32, device->spec.channels, device_buffer, device->spec.format, device->spec.channels, device->work_buffer); + ConvertAudio(needed_samples / device->spec.channels, mix_buffer, SDL_AUDIO_F32, device->spec.channels, device->work_buffer, device->spec.format, device->spec.channels, NULL); SDL_memcpy(device_buffer, device->work_buffer, buffer_size); } break; @@ -1208,16 +1208,14 @@ static SDL_AudioFormat ParseAudioFormatString(const char *string) #define CHECK_FMT_STRING(x) if (SDL_strcmp(string, #x) == 0) { return SDL_AUDIO_##x; } CHECK_FMT_STRING(U8); CHECK_FMT_STRING(S8); - CHECK_FMT_STRING(S16LSB); - CHECK_FMT_STRING(S16MSB); + CHECK_FMT_STRING(S16LE); + CHECK_FMT_STRING(S16BE); CHECK_FMT_STRING(S16); - CHECK_FMT_STRING(S32LSB); - CHECK_FMT_STRING(S32MSB); - CHECK_FMT_STRING(S32SYS); + CHECK_FMT_STRING(S32LE); + CHECK_FMT_STRING(S32BE); CHECK_FMT_STRING(S32); - CHECK_FMT_STRING(F32LSB); - CHECK_FMT_STRING(F32MSB); - CHECK_FMT_STRING(F32SYS); + CHECK_FMT_STRING(F32LE); + CHECK_FMT_STRING(F32BE); CHECK_FMT_STRING(F32); #undef CHECK_FMT_STRING } @@ -1315,7 +1313,7 @@ static int OpenPhysicalAudioDevice(SDL_AudioDevice *device, const SDL_AudioSpec return SDL_OutOfMemory(); } - if (device->spec.format != SDL_AUDIO_F32SYS) { + if (device->spec.format != SDL_AUDIO_F32) { device->mix_buffer = (Uint8 *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size); if (device->mix_buffer == NULL) { ClosePhysicalAudioDevice(device); @@ -1660,14 +1658,14 @@ SDL_AudioStream *SDL_OpenAudioDeviceStream(SDL_AudioDeviceID devid, const SDL_Au #define NUM_FORMATS 8 static const SDL_AudioFormat format_list[NUM_FORMATS][NUM_FORMATS + 1] = { - { SDL_AUDIO_U8, SDL_AUDIO_S8, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, 0 }, - { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, 0 }, - { SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, - { SDL_AUDIO_S16MSB, SDL_AUDIO_S16LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, - { SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, - { SDL_AUDIO_S32MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S16LSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, - { SDL_AUDIO_F32LSB, SDL_AUDIO_F32MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_S16LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, - { SDL_AUDIO_F32MSB, SDL_AUDIO_F32LSB, SDL_AUDIO_S32MSB, SDL_AUDIO_S32LSB, SDL_AUDIO_S16MSB, SDL_AUDIO_S16LSB, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_U8, SDL_AUDIO_S8, SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, 0 }, + { SDL_AUDIO_S8, SDL_AUDIO_U8, SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, 0 }, + { SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_S16BE, SDL_AUDIO_S16LE, SDL_AUDIO_S32BE, SDL_AUDIO_S32LE, SDL_AUDIO_F32BE, SDL_AUDIO_F32LE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_S32BE, SDL_AUDIO_S32LE, SDL_AUDIO_F32BE, SDL_AUDIO_F32LE, SDL_AUDIO_S16BE, SDL_AUDIO_S16LE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_F32LE, SDL_AUDIO_F32BE, SDL_AUDIO_S32LE, SDL_AUDIO_S32BE, SDL_AUDIO_S16LE, SDL_AUDIO_S16BE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, + { SDL_AUDIO_F32BE, SDL_AUDIO_F32LE, SDL_AUDIO_S32BE, SDL_AUDIO_S32LE, SDL_AUDIO_S16BE, SDL_AUDIO_S16LE, SDL_AUDIO_U8, SDL_AUDIO_S8, 0 }, }; const SDL_AudioFormat *SDL_ClosestAudioFormats(SDL_AudioFormat format) @@ -1826,7 +1824,7 @@ int SDL_AudioDeviceFormatChangedAlreadyLocked(SDL_AudioDevice *device, const SDL SDL_aligned_free(device->mix_buffer); device->mix_buffer = NULL; - if (device->spec.format != SDL_AUDIO_F32SYS) { + if (device->spec.format != SDL_AUDIO_F32) { device->mix_buffer = (Uint8 *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), device->work_buffer_size); if (!device->mix_buffer) { kill_device = SDL_TRUE; diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c index 7d9120e94..5f9be7da8 100644 --- a/src/audio/SDL_audiocvt.c +++ b/src/audio/SDL_audiocvt.c @@ -860,8 +860,8 @@ static void AudioConvertToFloat(float *dst, const void *src, int num_samples, SD switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) { case SDL_AUDIO_S8: SDL_Convert_S8_to_F32(dst, (const Sint8 *) src, num_samples); break; case SDL_AUDIO_U8: SDL_Convert_U8_to_F32(dst, (const Uint8 *) src, num_samples); break; - case SDL_AUDIO_S16: SDL_Convert_S16_to_F32(dst, (const Sint16 *) src, num_samples); break; - case SDL_AUDIO_S32: SDL_Convert_S32_to_F32(dst, (const Sint32 *) src, num_samples); break; + case (SDL_AUDIO_S16 & ~SDL_AUDIO_MASK_ENDIAN): SDL_Convert_S16_to_F32(dst, (const Sint16 *) src, num_samples); break; + case (SDL_AUDIO_S32 & ~SDL_AUDIO_MASK_ENDIAN): SDL_Convert_S32_to_F32(dst, (const Sint32 *) src, num_samples); break; default: SDL_assert(!"Unexpected audio format!"); break; } } @@ -872,8 +872,8 @@ static void AudioConvertFromFloat(void *dst, const float *src, int num_samples, switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) { case SDL_AUDIO_S8: SDL_Convert_F32_to_S8((Sint8 *) dst, src, num_samples); break; case SDL_AUDIO_U8: SDL_Convert_F32_to_U8((Uint8 *) dst, src, num_samples); break; - case SDL_AUDIO_S16: SDL_Convert_F32_to_S16((Sint16 *) dst, src, num_samples); break; - case SDL_AUDIO_S32: SDL_Convert_F32_to_S32((Sint32 *) dst, src, num_samples); break; + case (SDL_AUDIO_S16 & ~SDL_AUDIO_MASK_ENDIAN): SDL_Convert_F32_to_S16((Sint16 *) dst, src, num_samples); break; + case (SDL_AUDIO_S32 & ~SDL_AUDIO_MASK_ENDIAN): SDL_Convert_F32_to_S32((Sint32 *) dst, src, num_samples); break; default: SDL_assert(!"Unexpected audio format!"); break; } } @@ -883,12 +883,12 @@ static SDL_bool SDL_IsSupportedAudioFormat(const SDL_AudioFormat fmt) switch (fmt) { case SDL_AUDIO_U8: case SDL_AUDIO_S8: - case SDL_AUDIO_S16LSB: - case SDL_AUDIO_S16MSB: - case SDL_AUDIO_S32LSB: - case SDL_AUDIO_S32MSB: - case SDL_AUDIO_F32LSB: - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_S16LE: + case SDL_AUDIO_S16BE: + case SDL_AUDIO_S32LE: + case SDL_AUDIO_S32BE: + case SDL_AUDIO_F32LE: + case SDL_AUDIO_F32BE: return SDL_TRUE; // supported. default: @@ -1540,7 +1540,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou // Check if we can resample directly into the output buffer. // Note, this is just to avoid extra copies. // Some other formats may fit directly into the output buffer, but i'd rather process data in a SIMD-aligned buffer. - if ((dst_format != SDL_AUDIO_F32SYS) || (dst_channels != resample_channels)) { + if ((dst_format != SDL_AUDIO_F32) || (dst_channels != resample_channels)) { // Allocate space for converting the resampled output to the destination format int resample_convert_bytes = output_frames * max_frame_size; work_buffer_capacity = SDL_max(work_buffer_capacity, resample_convert_bytes); @@ -1597,7 +1597,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou SDL_assert(work_buffer_frames == input_frames + (resampler_padding_frames * 2)); // Resampling! get the work buffer to float32 format, etc, in-place. - ConvertAudio(work_buffer_frames, work_buffer, src_format, src_channels, work_buffer, SDL_AUDIO_F32SYS, resample_channels, NULL); + ConvertAudio(work_buffer_frames, work_buffer, src_format, src_channels, work_buffer, SDL_AUDIO_F32, resample_channels, NULL); // Update the work_buffer pointers based on the new frame size input_buffer = work_buffer + ((input_buffer - work_buffer) / src_frame_size * resample_frame_size); @@ -1614,7 +1614,7 @@ static int GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int ou // Convert to the final format, if necessary if (buf != resample_buffer) { - ConvertAudio(output_frames, resample_buffer, SDL_AUDIO_F32SYS, resample_channels, buf, dst_format, dst_channels, work_buffer); + ConvertAudio(output_frames, resample_buffer, SDL_AUDIO_F32, resample_channels, buf, dst_format, dst_channels, work_buffer); } return 0; diff --git a/src/audio/SDL_mixer.c b/src/audio/SDL_mixer.c index c4630ef02..0a4992d82 100644 --- a/src/audio/SDL_mixer.c +++ b/src/audio/SDL_mixer.c @@ -131,7 +131,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: { Sint16 src1, src2; int dst_sample; @@ -155,7 +155,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: { Sint16 src1, src2; int dst_sample; @@ -179,7 +179,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: { const Uint32 *src32 = (Uint32 *)src; Uint32 *dst32 = (Uint32 *)dst; @@ -204,7 +204,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: { const Uint32 *src32 = (Uint32 *)src; Uint32 *dst32 = (Uint32 *)dst; @@ -229,7 +229,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_F32LSB: + case SDL_AUDIO_F32LE: { const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME); const float fvolume = (float)volume; @@ -257,7 +257,7 @@ int SDL_MixAudioFormat(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, } } break; - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_F32BE: { const float fmaxvolume = 1.0f / ((float)SDL_MIX_MAXVOLUME); const float fvolume = (float)volume; diff --git a/src/audio/SDL_wave.c b/src/audio/SDL_wave.c index 27d540c46..2afb70bd2 100644 --- a/src/audio/SDL_wave.c +++ b/src/audio/SDL_wave.c @@ -2039,10 +2039,10 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 * case ALAW_CODE: case MULAW_CODE: /* These can be easily stored in the byte order of the system. */ - spec->format = SDL_AUDIO_S16SYS; + spec->format = SDL_AUDIO_S16; break; case IEEE_FLOAT_CODE: - spec->format = SDL_AUDIO_F32LSB; + spec->format = SDL_AUDIO_F32LE; break; case PCM_CODE: switch (format->bitspersample) { @@ -2050,11 +2050,11 @@ static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 * spec->format = SDL_AUDIO_U8; break; case 16: - spec->format = SDL_AUDIO_S16LSB; + spec->format = SDL_AUDIO_S16LE; break; case 24: /* Has been shifted to 32 bits. */ case 32: - spec->format = SDL_AUDIO_S32LSB; + spec->format = SDL_AUDIO_S32LE; break; default: /* Just in case something unexpected happened in the checks. */ diff --git a/src/audio/aaudio/SDL_aaudio.c b/src/audio/aaudio/SDL_aaudio.c index 5b12d6f52..7692f4e2e 100644 --- a/src/audio/aaudio/SDL_aaudio.c +++ b/src/audio/aaudio/SDL_aaudio.c @@ -198,9 +198,9 @@ static int AAUDIO_OpenDevice(SDL_AudioDevice *device) const aaudio_direction_t direction = (iscapture ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT); ctx.AAudioStreamBuilder_setDirection(builder, direction); aaudio_format_t format; - if ((device->spec.format == SDL_AUDIO_S32SYS) && (SDL_GetAndroidSDKVersion() >= 31)) { + if ((device->spec.format == SDL_AUDIO_S32) && (SDL_GetAndroidSDKVersion() >= 31)) { format = AAUDIO_FORMAT_PCM_I32; - } else if (device->spec.format == SDL_AUDIO_F32SYS) { + } else if (device->spec.format == SDL_AUDIO_F32) { format = AAUDIO_FORMAT_PCM_FLOAT; } else { format = AAUDIO_FORMAT_PCM_I16; // sint16 is a safe bet for everything else. @@ -245,11 +245,11 @@ static int AAUDIO_OpenDevice(SDL_AudioDevice *device) format = ctx.AAudioStream_getFormat(hidden->stream); if (format == AAUDIO_FORMAT_PCM_I16) { - device->spec.format = SDL_AUDIO_S16SYS; + device->spec.format = SDL_AUDIO_S16; } else if (format == AAUDIO_FORMAT_PCM_I32) { - device->spec.format = SDL_AUDIO_S32SYS; + device->spec.format = SDL_AUDIO_S32; } else if (format == AAUDIO_FORMAT_PCM_FLOAT) { - device->spec.format = SDL_AUDIO_F32SYS; + device->spec.format = SDL_AUDIO_F32; } else { return SDL_SetError("Got unexpected audio format %d from AAudioStream_getFormat", (int) format); } diff --git a/src/audio/alsa/SDL_alsa_audio.c b/src/audio/alsa/SDL_alsa_audio.c index 816a0c217..73181761c 100644 --- a/src/audio/alsa/SDL_alsa_audio.c +++ b/src/audio/alsa/SDL_alsa_audio.c @@ -565,22 +565,22 @@ static int ALSA_OpenDevice(SDL_AudioDevice *device) case SDL_AUDIO_S8: format = SND_PCM_FORMAT_S8; break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: format = SND_PCM_FORMAT_S16_LE; break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: format = SND_PCM_FORMAT_S16_BE; break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: format = SND_PCM_FORMAT_S32_LE; break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: format = SND_PCM_FORMAT_S32_BE; break; - case SDL_AUDIO_F32LSB: + case SDL_AUDIO_F32LE: format = SND_PCM_FORMAT_FLOAT_LE; break; - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_F32BE: format = SND_PCM_FORMAT_FLOAT_BE; break; default: diff --git a/src/audio/coreaudio/SDL_coreaudio.m b/src/audio/coreaudio/SDL_coreaudio.m index 31be6db08..61e5d9ce9 100644 --- a/src/audio/coreaudio/SDL_coreaudio.m +++ b/src/audio/coreaudio/SDL_coreaudio.m @@ -876,12 +876,12 @@ static int COREAUDIO_OpenDevice(SDL_AudioDevice *device) switch (test_format) { case SDL_AUDIO_U8: case SDL_AUDIO_S8: - case SDL_AUDIO_S16LSB: - case SDL_AUDIO_S16MSB: - case SDL_AUDIO_S32LSB: - case SDL_AUDIO_S32MSB: - case SDL_AUDIO_F32LSB: - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_S16LE: + case SDL_AUDIO_S16BE: + case SDL_AUDIO_S32LE: + case SDL_AUDIO_S32BE: + case SDL_AUDIO_F32LE: + case SDL_AUDIO_F32BE: break; default: diff --git a/src/audio/dsp/SDL_dspaudio.c b/src/audio/dsp/SDL_dspaudio.c index a6550bf3b..05572d22f 100644 --- a/src/audio/dsp/SDL_dspaudio.c +++ b/src/audio/dsp/SDL_dspaudio.c @@ -111,12 +111,12 @@ static int DSP_OpenDevice(SDL_AudioDevice *device) format = AFMT_U8; } break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: if (value & AFMT_S16_LE) { format = AFMT_S16_LE; } break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: if (value & AFMT_S16_BE) { format = AFMT_S16_BE; } diff --git a/src/audio/haiku/SDL_haikuaudio.cc b/src/audio/haiku/SDL_haikuaudio.cc index 9ccaf3c43..15204362d 100644 --- a/src/audio/haiku/SDL_haikuaudio.cc +++ b/src/audio/haiku/SDL_haikuaudio.cc @@ -131,29 +131,29 @@ static int HAIKUAUDIO_OpenDevice(SDL_AudioDevice *device) format.format = media_raw_audio_format::B_AUDIO_UCHAR; break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: format.format = media_raw_audio_format::B_AUDIO_SHORT; break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: format.format = media_raw_audio_format::B_AUDIO_SHORT; format.byte_order = B_MEDIA_BIG_ENDIAN; break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: format.format = media_raw_audio_format::B_AUDIO_INT; break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: format.format = media_raw_audio_format::B_AUDIO_INT; format.byte_order = B_MEDIA_BIG_ENDIAN; break; - case SDL_AUDIO_F32LSB: + case SDL_AUDIO_F32LE: format.format = media_raw_audio_format::B_AUDIO_FLOAT; break; - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_F32BE: format.format = media_raw_audio_format::B_AUDIO_FLOAT; format.byte_order = B_MEDIA_BIG_ENDIAN; break; diff --git a/src/audio/jack/SDL_jackaudio.c b/src/audio/jack/SDL_jackaudio.c index e6567d4c7..ce2fde14c 100644 --- a/src/audio/jack/SDL_jackaudio.c +++ b/src/audio/jack/SDL_jackaudio.c @@ -309,7 +309,7 @@ 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. */ - device->spec.format = SDL_AUDIO_F32SYS; + device->spec.format = SDL_AUDIO_F32; device->spec.freq = JACK_jack_get_sample_rate(client); device->spec.channels = channels; device->sample_frames = JACK_jack_get_buffer_size(client); diff --git a/src/audio/netbsd/SDL_netbsdaudio.c b/src/audio/netbsd/SDL_netbsdaudio.c index 2d43a72b8..899fe3c34 100644 --- a/src/audio/netbsd/SDL_netbsdaudio.c +++ b/src/audio/netbsd/SDL_netbsdaudio.c @@ -248,16 +248,16 @@ static int NETBSDAUDIO_OpenDevice(SDL_AudioDevice *device) case SDL_AUDIO_S8: encoding = AUDIO_ENCODING_SLINEAR; break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: encoding = AUDIO_ENCODING_SLINEAR_LE; break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: encoding = AUDIO_ENCODING_SLINEAR_BE; break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: encoding = AUDIO_ENCODING_SLINEAR_LE; break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: encoding = AUDIO_ENCODING_SLINEAR_BE; break; default: diff --git a/src/audio/openslES/SDL_openslES.c b/src/audio/openslES/SDL_openslES.c index 3d29eb8f5..c4855eaef 100644 --- a/src/audio/openslES/SDL_openslES.c +++ b/src/audio/openslES/SDL_openslES.c @@ -248,7 +248,7 @@ static int openslES_CreatePCMRecorder(SDL_AudioDevice *device) } // Just go with signed 16-bit audio as it's the most compatible - device->spec.format = SDL_AUDIO_S16SYS; + device->spec.format = SDL_AUDIO_S16; device->spec.channels = 1; //device->spec.freq = SL_SAMPLINGRATE_16 / 1000;*/ @@ -424,12 +424,12 @@ static int openslES_CreatePCMPlayer(SDL_AudioDevice *device) if (!test_format) { // Didn't find a compatible format : LOGI("No compatible audio format, using signed 16-bit audio"); - test_format = SDL_AUDIO_S16SYS; + test_format = SDL_AUDIO_S16; } device->spec.format = test_format; } else { // Just go with signed 16-bit audio as it's the most compatible - device->spec.format = SDL_AUDIO_S16SYS; + device->spec.format = SDL_AUDIO_S16; } // Update the fragment size as size in bytes diff --git a/src/audio/pipewire/SDL_pipewire.c b/src/audio/pipewire/SDL_pipewire.c index edd1c4a5c..08083a01a 100644 --- a/src/audio/pipewire/SDL_pipewire.c +++ b/src/audio/pipewire/SDL_pipewire.c @@ -898,22 +898,22 @@ static void initialize_spa_info(const SDL_AudioSpec *spec, struct spa_audio_info case SDL_AUDIO_S8: info->format = SPA_AUDIO_FORMAT_S8; break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: info->format = SPA_AUDIO_FORMAT_S16_LE; break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: info->format = SPA_AUDIO_FORMAT_S16_BE; break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: info->format = SPA_AUDIO_FORMAT_S32_LE; break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: info->format = SPA_AUDIO_FORMAT_S32_BE; break; - case SDL_AUDIO_F32LSB: + case SDL_AUDIO_F32LE: info->format = SPA_AUDIO_FORMAT_F32_LE; break; - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_F32BE: info->format = SPA_AUDIO_FORMAT_F32_BE; break; } diff --git a/src/audio/psp/SDL_pspaudio.c b/src/audio/psp/SDL_pspaudio.c index d4900572a..d701b5f2c 100644 --- a/src/audio/psp/SDL_pspaudio.c +++ b/src/audio/psp/SDL_pspaudio.c @@ -47,7 +47,7 @@ static int PSPAUDIO_OpenDevice(SDL_AudioDevice *device) } // device only natively supports S16LSB - device->spec.format = SDL_AUDIO_S16LSB; + device->spec.format = SDL_AUDIO_S16LE; /* PSP has some limitations with the Audio. It fully supports 44.1KHz (Mono & Stereo), however with frequencies different than 44.1KHz, it just supports Stereo, diff --git a/src/audio/pulseaudio/SDL_pulseaudio.c b/src/audio/pulseaudio/SDL_pulseaudio.c index c3028c0a6..a0069cf2f 100644 --- a/src/audio/pulseaudio/SDL_pulseaudio.c +++ b/src/audio/pulseaudio/SDL_pulseaudio.c @@ -606,22 +606,22 @@ static int PULSEAUDIO_OpenDevice(SDL_AudioDevice *device) case SDL_AUDIO_U8: format = PA_SAMPLE_U8; break; - case SDL_AUDIO_S16LSB: + case SDL_AUDIO_S16LE: format = PA_SAMPLE_S16LE; break; - case SDL_AUDIO_S16MSB: + case SDL_AUDIO_S16BE: format = PA_SAMPLE_S16BE; break; - case SDL_AUDIO_S32LSB: + case SDL_AUDIO_S32LE: format = PA_SAMPLE_S32LE; break; - case SDL_AUDIO_S32MSB: + case SDL_AUDIO_S32BE: format = PA_SAMPLE_S32BE; break; - case SDL_AUDIO_F32LSB: + case SDL_AUDIO_F32LE: format = PA_SAMPLE_FLOAT32LE; break; - case SDL_AUDIO_F32MSB: + case SDL_AUDIO_F32BE: format = PA_SAMPLE_FLOAT32BE; break; default: @@ -723,17 +723,17 @@ static SDL_AudioFormat PulseFormatToSDLFormat(pa_sample_format_t format) case PA_SAMPLE_U8: return SDL_AUDIO_U8; case PA_SAMPLE_S16LE: - return SDL_AUDIO_S16LSB; + return SDL_AUDIO_S16LE; case PA_SAMPLE_S16BE: - return SDL_AUDIO_S16MSB; + return SDL_AUDIO_S16BE; case PA_SAMPLE_S32LE: - return SDL_AUDIO_S32LSB; + return SDL_AUDIO_S32LE; case PA_SAMPLE_S32BE: - return SDL_AUDIO_S32MSB; + return SDL_AUDIO_S32BE; case PA_SAMPLE_FLOAT32LE: - return SDL_AUDIO_F32LSB; + return SDL_AUDIO_F32LE; case PA_SAMPLE_FLOAT32BE: - return SDL_AUDIO_F32MSB; + return SDL_AUDIO_F32BE; default: return 0; } diff --git a/src/audio/qnx/SDL_qsa_audio.c b/src/audio/qnx/SDL_qsa_audio.c index d2552ce85..710371687 100644 --- a/src/audio/qnx/SDL_qsa_audio.c +++ b/src/audio/qnx/SDL_qsa_audio.c @@ -309,7 +309,7 @@ static SDL_AudioFormat QnxFormatToSDLFormat(const int32_t qnxfmt) #undef CHECKFMT default: break; } - return SDL_AUDIO_S16SYS; // oh well. + return SDL_AUDIO_S16; // oh well. } static void QSA_DetectDevices(SDL_AudioDevice **default_output, SDL_AudioDevice **default_capture) diff --git a/src/audio/sndio/SDL_sndioaudio.c b/src/audio/sndio/SDL_sndioaudio.c index d1685a747..0347c38ec 100644 --- a/src/audio/sndio/SDL_sndioaudio.c +++ b/src/audio/sndio/SDL_sndioaudio.c @@ -284,13 +284,13 @@ static int SNDIO_OpenDevice(SDL_AudioDevice *device) } if ((par.bps == 4) && (par.sig) && (par.le)) { - device->spec.format = SDL_AUDIO_S32LSB; + device->spec.format = SDL_AUDIO_S32LE; } else if ((par.bps == 4) && (par.sig) && (!par.le)) { - device->spec.format = SDL_AUDIO_S32MSB; + device->spec.format = SDL_AUDIO_S32BE; } else if ((par.bps == 2) && (par.sig) && (par.le)) { - device->spec.format = SDL_AUDIO_S16LSB; + device->spec.format = SDL_AUDIO_S16LE; } else if ((par.bps == 2) && (par.sig) && (!par.le)) { - device->spec.format = SDL_AUDIO_S16MSB; + device->spec.format = SDL_AUDIO_S16BE; } else if ((par.bps == 1) && (par.sig)) { device->spec.format = SDL_AUDIO_S8; } else if ((par.bps == 1) && (!par.sig)) { diff --git a/src/audio/vita/SDL_vitaaudio.c b/src/audio/vita/SDL_vitaaudio.c index 6e6190c95..53a54d104 100644 --- a/src/audio/vita/SDL_vitaaudio.c +++ b/src/audio/vita/SDL_vitaaudio.c @@ -71,7 +71,7 @@ static int VITAAUD_OpenDevice(SDL_AudioDevice *device) closefmts = SDL_ClosestAudioFormats(device->spec.format); while ((test_format = *(closefmts++)) != 0) { - if (test_format == SDL_AUDIO_S16LSB) { + if (test_format == SDL_AUDIO_S16LE) { device->spec.format = test_format; break; } diff --git a/src/core/windows/SDL_windows.c b/src/core/windows/SDL_windows.c index 467d4bb72..389862708 100644 --- a/src/core/windows/SDL_windows.c +++ b/src/core/windows/SDL_windows.c @@ -344,19 +344,19 @@ static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x SDL_AudioFormat SDL_WaveFormatExToSDLFormat(WAVEFORMATEX *waveformat) { if ((waveformat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) && (waveformat->wBitsPerSample == 32)) { - return SDL_AUDIO_F32SYS; + return SDL_AUDIO_F32; } else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 16)) { - return SDL_AUDIO_S16SYS; + return SDL_AUDIO_S16; } else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 32)) { - return SDL_AUDIO_S32SYS; + return SDL_AUDIO_S32; } else if (waveformat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { const WAVEFORMATEXTENSIBLE *ext = (const WAVEFORMATEXTENSIBLE *)waveformat; if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 32)) { - return SDL_AUDIO_F32SYS; + return SDL_AUDIO_F32; } else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 16)) { - return SDL_AUDIO_S16SYS; + return SDL_AUDIO_S16; } else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof(GUID)) == 0) && (waveformat->wBitsPerSample == 32)) { - return SDL_AUDIO_S32SYS; + return SDL_AUDIO_S32; } } return 0; diff --git a/src/test/SDL_test_common.c b/src/test/SDL_test_common.c index 928be46fc..9673add95 100644 --- a/src/test/SDL_test_common.c +++ b/src/test/SDL_test_common.c @@ -71,7 +71,7 @@ static const char *video_usage[] = { /* !!! FIXME: Float32? Sint32? */ static const char *audio_usage[] = { - "[--audio driver]", "[--rate N]", "[--format U8|S8|S16|S16LE|S16BE]", + "[--audio driver]", "[--rate N]", "[--format U8|S8|S16|S16LE|S16BE|S32|S32LE|S32BE|F32|F32LE|F32BE]", "[--channels N]" }; @@ -630,16 +630,37 @@ int SDLTest_CommonArg(SDLTest_CommonState *state, int index) return 2; } if (SDL_strcasecmp(argv[index], "S16LE") == 0) { - state->audio_format = SDL_AUDIO_S16LSB; + state->audio_format = SDL_AUDIO_S16LE; return 2; } if (SDL_strcasecmp(argv[index], "S16BE") == 0) { - state->audio_format = SDL_AUDIO_S16MSB; + state->audio_format = SDL_AUDIO_S16BE; + return 2; + } + if (SDL_strcasecmp(argv[index], "S32") == 0) { + state->audio_format = SDL_AUDIO_S32; + return 2; + } + if (SDL_strcasecmp(argv[index], "S32LE") == 0) { + state->audio_format = SDL_AUDIO_S32LE; + return 2; + } + if (SDL_strcasecmp(argv[index], "S32BE") == 0) { + state->audio_format = SDL_AUDIO_S32BE; + return 2; + } + if (SDL_strcasecmp(argv[index], "F32") == 0) { + state->audio_format = SDL_AUDIO_F32; + return 2; + } + if (SDL_strcasecmp(argv[index], "F32LE") == 0) { + state->audio_format = SDL_AUDIO_F32LE; + return 2; + } + if (SDL_strcasecmp(argv[index], "F32BE") == 0) { + state->audio_format = SDL_AUDIO_F32BE; return 2; } - - /* !!! FIXME: Float32? Sint32? */ - return -1; } if (SDL_strcasecmp(argv[index], "--channels") == 0) { diff --git a/test/testaudio.c b/test/testaudio.c index f74014865..d012c933a 100644 --- a/test/testaudio.c +++ b/test/testaudio.c @@ -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; diff --git a/test/testaudiostreamdynamicresample.c b/test/testaudiostreamdynamicresample.c index f95512a8e..a1f4c42c1 100644 --- a/test/testaudiostreamdynamicresample.c +++ b/test/testaudiostreamdynamicresample.c @@ -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 "?"; diff --git a/test/testautomation_audio.c b/test/testautomation_audio.c index 33cb68472..22296ba4d 100644 --- a/test/testautomation_audio.c +++ b/test/testautomation_audio.c @@ -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; diff --git a/test/testsurround.c b/test/testsurround.c index 53dc58dac..001952ea1 100644 --- a/test/testsurround.c +++ b/test/testsurround.c @@ -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;