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Use consistent style for pointer declarations and casts

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This commit is contained in:
Sam Lantinga
2025-06-18 09:26:51 -07:00
parent 390fe65323
commit d7939abf42
105 changed files with 339 additions and 346 deletions
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44
src/audio/SDL_audiocvt.c
View File

@@ -280,7 +280,7 @@ void ConvertAudio(int num_frames,
// swizzle input to "standard" format if necessary.
if (src_map) {
void* buf = scratch ? scratch : dst; // use scratch if available, since it has to be big enough to hold src, unless it's NULL, then dst has to be.
void *buf = scratch ? scratch : dst; // use scratch if available, since it has to be big enough to hold src, unless it's NULL, then dst has to be.
SwizzleAudio(num_frames, buf, src, src_channels, src_map, src_format);
src = buf;
}
@@ -318,7 +318,7 @@ void ConvertAudio(int num_frames,
// get us to float format.
if (srcconvert) {
void* buf = (channelconvert || dstconvert) ? scratch : dst;
void *buf = (channelconvert || dstconvert) ? scratch : dst;
ConvertAudioToFloat((float *) buf, src, num_frames * src_channels, src_format);
src = buf;
}
@@ -368,7 +368,7 @@ void ConvertAudio(int num_frames,
channel_converter = override;
}
void* buf = dstconvert ? scratch : dst;
void *buf = dstconvert ? scratch : dst;
channel_converter((float *) buf, (const float *) src, num_frames);
src = buf;
}
@@ -399,7 +399,7 @@ static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, S
return max_format_size * max_channels;
}
static Sint64 GetAudioStreamResampleRate(SDL_AudioStream* stream, int src_freq, Sint64 resample_offset)
static Sint64 GetAudioStreamResampleRate(SDL_AudioStream *stream, int src_freq, Sint64 resample_offset)
{
src_freq = (int)((float)src_freq * stream->freq_ratio);
@@ -778,9 +778,9 @@ static bool CheckAudioStreamIsFullySetup(SDL_AudioStream *stream)
}
// you MUST hold `stream->lock` when calling this, and validate your parameters!
static bool PutAudioStreamBufferInternal(SDL_AudioStream *stream, const SDL_AudioSpec *spec, const int *chmap, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void* userdata)
static bool PutAudioStreamBufferInternal(SDL_AudioStream *stream, const SDL_AudioSpec *spec, const int *chmap, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void *userdata)
{
SDL_AudioTrack* track = NULL;
SDL_AudioTrack *track = NULL;
if (callback) {
track = SDL_CreateAudioTrack(stream->queue, spec, chmap, (Uint8 *)buf, len, len, callback, userdata);
@@ -809,7 +809,7 @@ static bool PutAudioStreamBufferInternal(SDL_AudioStream *stream, const SDL_Audi
return retval;
}
static bool PutAudioStreamBuffer(SDL_AudioStream *stream, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void* userdata)
static bool PutAudioStreamBuffer(SDL_AudioStream *stream, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void *userdata)
{
#if DEBUG_AUDIOSTREAM
SDL_Log("AUDIOSTREAM: wants to put %d bytes", len);
@@ -836,7 +836,7 @@ static bool PutAudioStreamBuffer(SDL_AudioStream *stream, const void *buf, int l
static void SDLCALL FreeAllocatedAudioBuffer(void *userdata, const void *buf, int len)
{
SDL_free((void*) buf);
SDL_free((void *)buf);
}
bool SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len)
@@ -1081,8 +1081,8 @@ static Uint8 *EnsureAudioStreamWorkBufferSize(SDL_AudioStream *stream, size_t ne
return ptr;
}
static Sint64 NextAudioStreamIter(SDL_AudioStream* stream, void** inout_iter,
Sint64* inout_resample_offset, SDL_AudioSpec* out_spec, int **out_chmap, bool* out_flushed)
static Sint64 NextAudioStreamIter(SDL_AudioStream *stream, void **inout_iter,
Sint64 *inout_resample_offset, SDL_AudioSpec *out_spec, int **out_chmap, bool *out_flushed)
{
SDL_AudioSpec spec;
bool flushed;
@@ -1136,9 +1136,9 @@ static Sint64 NextAudioStreamIter(SDL_AudioStream* stream, void** inout_iter,
return output_frames;
}
static Sint64 GetAudioStreamAvailableFrames(SDL_AudioStream* stream, Sint64* out_resample_offset)
static Sint64 GetAudioStreamAvailableFrames(SDL_AudioStream *stream, Sint64 *out_resample_offset)
{
void* iter = SDL_BeginAudioQueueIter(stream->queue);
void *iter = SDL_BeginAudioQueueIter(stream->queue);
Sint64 resample_offset = stream->resample_offset;
Sint64 output_frames = 0;
@@ -1160,9 +1160,9 @@ static Sint64 GetAudioStreamAvailableFrames(SDL_AudioStream* stream, Sint64* out
return output_frames;
}
static Sint64 GetAudioStreamHead(SDL_AudioStream* stream, SDL_AudioSpec* out_spec, int **out_chmap, bool* out_flushed)
static Sint64 GetAudioStreamHead(SDL_AudioStream *stream, SDL_AudioSpec *out_spec, int **out_chmap, bool *out_flushed)
{
void* iter = SDL_BeginAudioQueueIter(stream->queue);
void *iter = SDL_BeginAudioQueueIter(stream->queue);
if (!iter) {
SDL_zerop(out_spec);
@@ -1178,8 +1178,8 @@ static Sint64 GetAudioStreamHead(SDL_AudioStream* stream, SDL_AudioSpec* out_spe
// Enough input data MUST be available!
static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int output_frames, float gain)
{
const SDL_AudioSpec* src_spec = &stream->input_spec;
const SDL_AudioSpec* dst_spec = &stream->dst_spec;
const SDL_AudioSpec *src_spec = &stream->input_spec;
const SDL_AudioSpec *dst_spec = &stream->dst_spec;
const SDL_AudioFormat src_format = src_spec->format;
const int src_channels = src_spec->channels;
@@ -1199,7 +1199,7 @@ static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int o
// Not resampling? It's an easy conversion (and maybe not even that!)
if (resample_rate == 0) {
Uint8* work_buffer = NULL;
Uint8 *work_buffer = NULL;
// Ensure we have enough scratch space for any conversions
if ((src_format != dst_format) || (src_channels != dst_channels) || (gain != 1.0f)) {
@@ -1269,7 +1269,7 @@ static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int o
work_buffer_capacity += resample_bytes;
}
Uint8* work_buffer = EnsureAudioStreamWorkBufferSize(stream, work_buffer_capacity);
Uint8 *work_buffer = EnsureAudioStreamWorkBufferSize(stream, work_buffer_capacity);
if (!work_buffer) {
return false;
@@ -1281,7 +1281,7 @@ static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int o
const float postresample_gain = (input_frames > output_frames) ? gain : 1.0f;
// (dst channel map is NULL because we'll do the final swizzle on ConvertAudio after resample.)
const Uint8* input_buffer = SDL_ReadFromAudioQueue(stream->queue,
const Uint8 *input_buffer = SDL_ReadFromAudioQueue(stream->queue,
NULL, resample_format, resample_channels, NULL,
padding_frames, input_frames, padding_frames, work_buffer, preresample_gain);
@@ -1292,11 +1292,11 @@ static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int o
input_buffer += padding_frames * resample_frame_size;
// Decide where the resampled output goes
void* resample_buffer = (resample_buffer_offset != -1) ? (work_buffer + resample_buffer_offset) : buf;
void *resample_buffer = (resample_buffer_offset != -1) ? (work_buffer + resample_buffer_offset) : buf;
SDL_ResampleAudio(resample_channels,
(const float *) input_buffer, input_frames,
(float*) resample_buffer, output_frames,
(const float *)input_buffer, input_frames,
(float *)resample_buffer, output_frames,
resample_rate, &stream->resample_offset);
// Convert to the final format, if necessary (src channel map is NULL because SDL_ReadFromAudioQueue already handled this).

98
src/audio/SDL_audiotypecvt.c
View File

@@ -185,7 +185,7 @@ static void SDL_Convert_F32_to_S32_Scalar(Sint32 *dst, const float *src, int num
#undef SIGNMASK
static void SDL_Convert_Swap16_Scalar(Uint16* dst, const Uint16* src, int num_samples)
static void SDL_Convert_Swap16_Scalar(Uint16 *dst, const Uint16 *src, int num_samples)
{
int i;
@@ -194,7 +194,7 @@ static void SDL_Convert_Swap16_Scalar(Uint16* dst, const Uint16* src, int num_sa
}
}
static void SDL_Convert_Swap32_Scalar(Uint32* dst, const Uint32* src, int num_samples)
static void SDL_Convert_Swap32_Scalar(Uint32 *dst, const Uint32 *src, int num_samples)
{
int i;
@@ -375,7 +375,7 @@ static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S8_SSE2(Sint8 *dst, const f
const __m128i bytes = _mm_packus_epi16(shorts0, shorts1);
_mm_store_si128((__m128i*)&dst[i], bytes);
_mm_store_si128((__m128i *)&dst[i], bytes);
})
}
@@ -409,7 +409,7 @@ static void SDL_TARGETING("sse2") SDL_Convert_F32_to_U8_SSE2(Uint8 *dst, const f
const __m128i bytes = _mm_packus_epi16(shorts0, shorts1);
_mm_store_si128((__m128i*)&dst[i], bytes);
_mm_store_si128((__m128i *)&dst[i], bytes);
})
}
@@ -441,8 +441,8 @@ static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S16_SSE2(Sint16 *dst, const
const __m128i shorts0 = _mm_packs_epi32(ints0, ints1);
const __m128i shorts1 = _mm_packs_epi32(ints2, ints3);
_mm_store_si128((__m128i*)&dst[i], shorts0);
_mm_store_si128((__m128i*)&dst[i + 8], shorts1);
_mm_store_si128((__m128i *)&dst[i], shorts0);
_mm_store_si128((__m128i *)&dst[i + 8], shorts1);
})
}
@@ -477,55 +477,55 @@ static void SDL_TARGETING("sse2") SDL_Convert_F32_to_S32_SSE2(Sint32 *dst, const
const __m128i ints2 = _mm_xor_si128(_mm_cvttps_epi32(values3), _mm_castps_si128(_mm_cmpge_ps(values3, limit)));
const __m128i ints3 = _mm_xor_si128(_mm_cvttps_epi32(values4), _mm_castps_si128(_mm_cmpge_ps(values4, limit)));
_mm_store_si128((__m128i*)&dst[i], ints0);
_mm_store_si128((__m128i*)&dst[i + 4], ints1);
_mm_store_si128((__m128i*)&dst[i + 8], ints2);
_mm_store_si128((__m128i*)&dst[i + 12], ints3);
_mm_store_si128((__m128i *)&dst[i], ints0);
_mm_store_si128((__m128i *)&dst[i + 4], ints1);
_mm_store_si128((__m128i *)&dst[i + 8], ints2);
_mm_store_si128((__m128i *)&dst[i + 12], ints3);
})
}
#endif
// FIXME: SDL doesn't have SSSE3 detection, so use the next one up
#ifdef SDL_SSE4_1_INTRINSICS
static void SDL_TARGETING("ssse3") SDL_Convert_Swap16_SSSE3(Uint16* dst, const Uint16* src, int num_samples)
static void SDL_TARGETING("ssse3") SDL_Convert_Swap16_SSSE3(Uint16 *dst, const Uint16 *src, int num_samples)
{
const __m128i shuffle = _mm_set_epi8(14, 15, 12, 13, 10, 11, 8, 9, 6, 7, 4, 5, 2, 3, 0, 1);
CONVERT_16_FWD({
dst[i] = SDL_Swap16(src[i]);
}, {
__m128i ints0 = _mm_loadu_si128((const __m128i*)&src[i]);
__m128i ints1 = _mm_loadu_si128((const __m128i*)&src[i + 8]);
__m128i ints0 = _mm_loadu_si128((const __m128i *)&src[i]);
__m128i ints1 = _mm_loadu_si128((const __m128i *)&src[i + 8]);
ints0 = _mm_shuffle_epi8(ints0, shuffle);
ints1 = _mm_shuffle_epi8(ints1, shuffle);
_mm_store_si128((__m128i*)&dst[i], ints0);
_mm_store_si128((__m128i*)&dst[i + 8], ints1);
_mm_store_si128((__m128i *)&dst[i], ints0);
_mm_store_si128((__m128i *)&dst[i + 8], ints1);
})
}
static void SDL_TARGETING("ssse3") SDL_Convert_Swap32_SSSE3(Uint32* dst, const Uint32* src, int num_samples)
static void SDL_TARGETING("ssse3") SDL_Convert_Swap32_SSSE3(Uint32 *dst, const Uint32 *src, int num_samples)
{
const __m128i shuffle = _mm_set_epi8(12, 13, 14, 15, 8, 9, 10, 11, 4, 5, 6, 7, 0, 1, 2, 3);
CONVERT_16_FWD({
dst[i] = SDL_Swap32(src[i]);
}, {
__m128i ints0 = _mm_loadu_si128((const __m128i*)&src[i]);
__m128i ints1 = _mm_loadu_si128((const __m128i*)&src[i + 4]);
__m128i ints2 = _mm_loadu_si128((const __m128i*)&src[i + 8]);
__m128i ints3 = _mm_loadu_si128((const __m128i*)&src[i + 12]);
__m128i ints0 = _mm_loadu_si128((const __m128i *)&src[i]);
__m128i ints1 = _mm_loadu_si128((const __m128i *)&src[i + 4]);
__m128i ints2 = _mm_loadu_si128((const __m128i *)&src[i + 8]);
__m128i ints3 = _mm_loadu_si128((const __m128i *)&src[i + 12]);
ints0 = _mm_shuffle_epi8(ints0, shuffle);
ints1 = _mm_shuffle_epi8(ints1, shuffle);
ints2 = _mm_shuffle_epi8(ints2, shuffle);
ints3 = _mm_shuffle_epi8(ints3, shuffle);
_mm_store_si128((__m128i*)&dst[i], ints0);
_mm_store_si128((__m128i*)&dst[i + 4], ints1);
_mm_store_si128((__m128i*)&dst[i + 8], ints2);
_mm_store_si128((__m128i*)&dst[i + 12], ints3);
_mm_store_si128((__m128i *)&dst[i], ints0);
_mm_store_si128((__m128i *)&dst[i + 4], ints1);
_mm_store_si128((__m128i *)&dst[i + 8], ints2);
_mm_store_si128((__m128i *)&dst[i + 12], ints3);
})
}
#endif
@@ -774,41 +774,41 @@ static void SDL_Convert_F32_to_S32_NEON(Sint32 *dst, const float *src, int num_s
fesetenv(&fenv);
}
static void SDL_Convert_Swap16_NEON(Uint16* dst, const Uint16* src, int num_samples)
static void SDL_Convert_Swap16_NEON(Uint16 *dst, const Uint16 *src, int num_samples)
{
CONVERT_16_FWD({
dst[i] = SDL_Swap16(src[i]);
}, {
uint8x16_t ints0 = vld1q_u8((const Uint8*)&src[i]);
uint8x16_t ints1 = vld1q_u8((const Uint8*)&src[i + 8]);
uint8x16_t ints0 = vld1q_u8((const Uint8 *)&src[i]);
uint8x16_t ints1 = vld1q_u8((const Uint8 *)&src[i + 8]);
ints0 = vrev16q_u8(ints0);
ints1 = vrev16q_u8(ints1);
vst1q_u8((Uint8*)&dst[i], ints0);
vst1q_u8((Uint8*)&dst[i + 8], ints1);
vst1q_u8((Uint8 *)&dst[i], ints0);
vst1q_u8((Uint8 *)&dst[i + 8], ints1);
})
}
static void SDL_Convert_Swap32_NEON(Uint32* dst, const Uint32* src, int num_samples)
static void SDL_Convert_Swap32_NEON(Uint32 *dst, const Uint32 *src, int num_samples)
{
CONVERT_16_FWD({
dst[i] = SDL_Swap32(src[i]);
}, {
uint8x16_t ints0 = vld1q_u8((const Uint8*)&src[i]);
uint8x16_t ints1 = vld1q_u8((const Uint8*)&src[i + 4]);
uint8x16_t ints2 = vld1q_u8((const Uint8*)&src[i + 8]);
uint8x16_t ints3 = vld1q_u8((const Uint8*)&src[i + 12]);
uint8x16_t ints0 = vld1q_u8((const Uint8 *)&src[i]);
uint8x16_t ints1 = vld1q_u8((const Uint8 *)&src[i + 4]);
uint8x16_t ints2 = vld1q_u8((const Uint8 *)&src[i + 8]);
uint8x16_t ints3 = vld1q_u8((const Uint8 *)&src[i + 12]);
ints0 = vrev32q_u8(ints0);
ints1 = vrev32q_u8(ints1);
ints2 = vrev32q_u8(ints2);
ints3 = vrev32q_u8(ints3);
vst1q_u8((Uint8*)&dst[i], ints0);
vst1q_u8((Uint8*)&dst[i + 4], ints1);
vst1q_u8((Uint8*)&dst[i + 8], ints2);
vst1q_u8((Uint8*)&dst[i + 12], ints3);
vst1q_u8((Uint8 *)&dst[i], ints0);
vst1q_u8((Uint8 *)&dst[i + 4], ints1);
vst1q_u8((Uint8 *)&dst[i + 8], ints2);
vst1q_u8((Uint8 *)&dst[i + 12], ints3);
})
}
@@ -839,8 +839,8 @@ static void (*SDL_Convert_F32_to_U8)(Uint8 *dst, const float *src, int num_sampl
static void (*SDL_Convert_F32_to_S16)(Sint16 *dst, const float *src, int num_samples) = NULL;
static void (*SDL_Convert_F32_to_S32)(Sint32 *dst, const float *src, int num_samples) = NULL;
static void (*SDL_Convert_Swap16)(Uint16* dst, const Uint16* src, int num_samples) = NULL;
static void (*SDL_Convert_Swap32)(Uint32* dst, const Uint32* src, int num_samples) = NULL;
static void (*SDL_Convert_Swap16)(Uint16 *dst, const Uint16 *src, int num_samples) = NULL;
static void (*SDL_Convert_Swap32)(Uint32 *dst, const Uint32 *src, int num_samples) = NULL;
void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_AudioFormat src_fmt)
{
@@ -858,7 +858,7 @@ void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_Audio
break;
case SDL_AUDIO_S16 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) src, num_samples);
SDL_Convert_Swap16((Uint16 *)dst, (const Uint16 *)src, num_samples);
SDL_Convert_S16_to_F32(dst, (const Sint16 *) dst, num_samples);
break;
@@ -867,12 +867,12 @@ void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_Audio
break;
case SDL_AUDIO_S32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
SDL_Convert_Swap32((Uint32 *)dst, (const Uint32 *)src, num_samples);
SDL_Convert_S32_to_F32(dst, (const Sint32 *) dst, num_samples);
break;
case SDL_AUDIO_F32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
SDL_Convert_Swap32((Uint32 *)dst, (const Uint32 *)src, num_samples);
break;
default: SDL_assert(!"Unexpected audio format!"); break;
@@ -896,7 +896,7 @@ void ConvertAudioFromFloat(void *dst, const float *src, int num_samples, SDL_Aud
case SDL_AUDIO_S16 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_F32_to_S16((Sint16 *) dst, src, num_samples);
SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) dst, num_samples);
SDL_Convert_Swap16((Uint16 *)dst, (const Uint16 *)dst, num_samples);
break;
case SDL_AUDIO_S32:
@@ -905,22 +905,22 @@ void ConvertAudioFromFloat(void *dst, const float *src, int num_samples, SDL_Aud
case SDL_AUDIO_S32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_F32_to_S32((Sint32 *) dst, src, num_samples);
SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) dst, num_samples);
SDL_Convert_Swap32((Uint32 *)dst, (const Uint32 *)dst, num_samples);
break;
case SDL_AUDIO_F32 ^ SDL_AUDIO_MASK_BIG_ENDIAN:
SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples);
SDL_Convert_Swap32((Uint32 *)dst, (const Uint32 *)src, num_samples);
break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
}
void ConvertAudioSwapEndian(void* dst, const void* src, int num_samples, int bitsize)
void ConvertAudioSwapEndian(void *dst, const void *src, int num_samples, int bitsize)
{
switch (bitsize) {
case 16: SDL_Convert_Swap16((Uint16*) dst, (const Uint16*) src, num_samples); break;
case 32: SDL_Convert_Swap32((Uint32*) dst, (const Uint32*) src, num_samples); break;
case 16: SDL_Convert_Swap16((Uint16 *)dst, (const Uint16 *)src, num_samples); break;
case 32: SDL_Convert_Swap32((Uint32 *)dst, (const Uint32 *)src, num_samples); break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
}

8
src/audio/SDL_sysaudio.h
View File

@@ -112,7 +112,7 @@ extern void SDL_AudioThreadFinalize(SDL_AudioDevice *device);
extern void ConvertAudioToFloat(float *dst, const void *src, int num_samples, SDL_AudioFormat src_fmt);
extern void ConvertAudioFromFloat(void *dst, const float *src, int num_samples, SDL_AudioFormat dst_fmt);
extern void ConvertAudioSwapEndian(void* dst, const void* src, int num_samples, int bitsize);
extern void ConvertAudioSwapEndian(void *dst, const void *src, int num_samples, int bitsize);
extern bool SDL_ChannelMapIsDefault(const int *map, int channels);
extern bool SDL_ChannelMapIsBogus(const int *map, int channels);
@@ -121,7 +121,7 @@ extern bool SDL_ChannelMapIsBogus(const int *map, int channels);
extern void ConvertAudio(int num_frames,
const void *src, SDL_AudioFormat src_format, int src_channels, const int *src_map,
void *dst, SDL_AudioFormat dst_format, int dst_channels, const int *dst_map,
void* scratch, float gain);
void *scratch, float gain);
// Compare two SDL_AudioSpecs, return true if they match exactly.
// Using SDL_memcmp directly isn't safe, since potential padding might not be initialized.
@@ -201,7 +201,7 @@ struct SDL_AudioQueue; // forward decl.
struct SDL_AudioStream
{
SDL_Mutex* lock;
SDL_Mutex *lock;
SDL_PropertiesID props;
@@ -217,7 +217,7 @@ struct SDL_AudioStream
float freq_ratio;
float gain;
struct SDL_AudioQueue* queue;
struct SDL_AudioQueue *queue;
SDL_AudioSpec input_spec; // The spec of input data currently being processed
int *input_chmap;

10
src/audio/alsa/SDL_alsa_audio.c
View File

@@ -87,7 +87,7 @@ static int (*ALSA_snd_device_name_free_hint)(void **);
static snd_pcm_sframes_t (*ALSA_snd_pcm_avail)(snd_pcm_t *);
static size_t (*ALSA_snd_ctl_card_info_sizeof)(void);
static size_t (*ALSA_snd_pcm_info_sizeof)(void);
static int (*ALSA_snd_card_next)(int*);
static int (*ALSA_snd_card_next)(int *);
static int (*ALSA_snd_ctl_open)(snd_ctl_t **,const char *,int);
static int (*ALSA_snd_ctl_close)(snd_ctl_t *);
static int (*ALSA_snd_ctl_card_info)(snd_ctl_t *, snd_ctl_card_info_t *);
@@ -670,7 +670,7 @@ static void swizzle_map_compute(const struct ALSA_pcm_cfg_ctx *ctx, int *swizzle
static int alsa_chmap_install(struct ALSA_pcm_cfg_ctx *ctx, const unsigned int *chmap)
{
bool isstack;
snd_pcm_chmap_t *chmap_to_install = (snd_pcm_chmap_t*)SDL_small_alloc(unsigned int, 1 + ctx->chans_n, &isstack);
snd_pcm_chmap_t *chmap_to_install = (snd_pcm_chmap_t *)SDL_small_alloc(unsigned int, 1 + ctx->chans_n, &isstack);
if (!chmap_to_install) {
return -1;
}
@@ -1214,7 +1214,7 @@ static int hotplug_device_process(snd_ctl_t *ctl, snd_ctl_card_info_t *ctl_card_
unsigned int subdev_idx = 0;
const bool recording = direction == SND_PCM_STREAM_CAPTURE ? true : false; // used for the unicity of the device
bool isstack;
snd_pcm_info_t *pcm_info = (snd_pcm_info_t*)SDL_small_alloc(Uint8, ALSA_snd_pcm_info_sizeof(), &isstack);
snd_pcm_info_t *pcm_info = (snd_pcm_info_t *)SDL_small_alloc(Uint8, ALSA_snd_pcm_info_sizeof(), &isstack);
SDL_memset(pcm_info, 0, ALSA_snd_pcm_info_sizeof());
while (true) {
@@ -1453,10 +1453,10 @@ static void ALSA_DetectDevices(SDL_AudioDevice **default_playback, SDL_AudioDevi
bool has_default_playback = false, has_default_recording = false;
ALSA_HotplugIteration(&has_default_playback, &has_default_recording); // run once now before a thread continues to check.
if (has_default_playback) {
*default_playback = SDL_AddAudioDevice(/*recording=*/false, "ALSA default playback device", NULL, (void*)&default_playback_handle);
*default_playback = SDL_AddAudioDevice(/*recording=*/false, "ALSA default playback device", NULL, (void *)&default_playback_handle);
}
if (has_default_recording) {
*default_recording = SDL_AddAudioDevice(/*recording=*/true, "ALSA default recording device", NULL, (void*)&default_recording_handle);
*default_recording = SDL_AddAudioDevice(/*recording=*/true, "ALSA default recording device", NULL, (void *)&default_recording_handle);
}
#if SDL_ALSA_HOTPLUG_THREAD

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

@@ -56,7 +56,7 @@ static bool EMSCRIPTENAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buf
}
for (var j = 0; j < $1; ++j) {
channelData[j] = HEAPF32[buf + (j*numChannels + c)];
channelData[j] = HEAPF32[buf + (j * numChannels + c)];
}
}
}, buffer, buffer_size / framelen);

2
src/audio/pipewire/SDL_pipewire.c
View File

@@ -546,7 +546,7 @@ static void node_event_info(void *object, const struct pw_node_info *info)
// Need to parse the parameters to get the sample rate
for (i = 0; i < info->n_params; ++i) {
pw_node_enum_params((struct pw_node*)node->proxy, 0, info->params[i].id, 0, 0, NULL);
pw_node_enum_params((struct pw_node *)node->proxy, 0, info->params[i].id, 0, 0, NULL);
}
hotplug_core_sync(node);

2
src/audio/wasapi/SDL_wasapi.c
View File

@@ -750,7 +750,7 @@ static bool mgmtthrtask_PrepDevice(void *userdata)
// Try querying IAudioClient3 if sharemode is AUDCLNT_SHAREMODE_SHARED
if (sharemode == AUDCLNT_SHAREMODE_SHARED) {
IAudioClient3 *client3 = NULL;
ret = IAudioClient_QueryInterface(client, &SDL_IID_IAudioClient3, (void**)&client3);
ret = IAudioClient_QueryInterface(client, &SDL_IID_IAudioClient3, (void **)&client3);
if (SUCCEEDED(ret)) {
UINT32 default_period_in_frames = 0;
UINT32 fundamental_period_in_frames = 0;