mirrors/SDL
1
0
Fork 0
mirror of https://github.com/libsdl-org/SDL.git synced 2026-05-03 12:34:41 +00:00
Code Issues Packages Projects Releases Wiki Activity

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

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

46
src/audio/SDL_audio.c
View File

@@ -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;

26
src/audio/SDL_audiocvt.c
View File

@@ -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;

12
src/audio/SDL_mixer.c
View File

@@ -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;

8
src/audio/SDL_wave.c
View File

@@ -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. */

10
src/audio/aaudio/SDL_aaudio.c
View File

@@ -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);
}

12
src/audio/alsa/SDL_alsa_audio.c
View File

@@ -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:

12
src/audio/coreaudio/SDL_coreaudio.m
View File

@@ -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:

4
src/audio/dsp/SDL_dspaudio.c
View File

@@ -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;
}

12
src/audio/haiku/SDL_haikuaudio.cc
View File

@@ -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;

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

@@ -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);

8
src/audio/netbsd/SDL_netbsdaudio.c
View File

@@ -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:

6
src/audio/openslES/SDL_openslES.c
View File

@@ -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

12
src/audio/pipewire/SDL_pipewire.c
View File

@@ -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;
}

2
src/audio/psp/SDL_pspaudio.c
View File

@@ -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,

24
src/audio/pulseaudio/SDL_pulseaudio.c
View File

@@ -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;
}

2
src/audio/qnx/SDL_qsa_audio.c
View File

@@ -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)

8
src/audio/sndio/SDL_sndioaudio.c
View File

@@ -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)) {

2
src/audio/vita/SDL_vitaaudio.c
View File

@@ -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;
}

12
src/core/windows/SDL_windows.c
View File

@@ -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;

33
src/test/SDL_test_common.c
View File

@@ -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) {