gitea/SDL3_fork
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Cleanup add brace (#6545)

Browse Source 
* Add braces after if conditions

* More add braces after if conditions

* Add braces after while() conditions

* Fix compilation because of macro being modified

* Add braces to for loop

* Add braces after if/goto

* Move comments up

* Remove extra () in the 'return ...;' statements

* More remove extra () in the 'return ...;' statements

* More remove extra () in the 'return ...;' statements after merge

* Fix inconsistent patterns are xxx == NULL vs !xxx

* More "{}" for "if() break;"  and "if() continue;"

* More "{}" after if() short statement

* More "{}" after "if () return;" statement

* More fix inconsistent patterns are xxx == NULL vs !xxx

* Revert some modificaion on SDL_RLEaccel.c

* SDL_RLEaccel: no short statement

* Cleanup 'if' where the bracket is in a new line

* Cleanup 'while' where the bracket is in a new line

* Cleanup 'for' where the bracket is in a new line

* Cleanup 'else' where the bracket is in a new line
This commit is contained in:
Sylvain Becker
2022-11-27 17:38:43 +01:00
committed by GitHub
parent 4958dafdc3
commit 6a2200823c
387 changed files with 6094 additions and 4633 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
src/audio/SDL_audio.c
View File

@@ -509,11 +509,9 @@ SDL_RemoveAudioDevice(const SDL_bool iscapture, void *handle)
} else {
mark_device_removed(handle, current_audio.outputDevices, &current_audio.outputDevicesRemoved);
}
for (device_index = 0; device_index < SDL_arraysize(open_devices); device_index++)
{
for (device_index = 0; device_index < SDL_arraysize(open_devices); device_index++) {
device = open_devices[device_index];
if (device != NULL && device->handle == handle)
{
if (device != NULL && device->handle == handle) {
device_was_opened = SDL_TRUE;
SDL_OpenedAudioDeviceDisconnected(device);
break;
@@ -977,7 +975,7 @@ SDL_AudioInit(const char *driver_name)
}
} else {
for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
if(bootstrap[i]->demand_only) {
if (bootstrap[i]->demand_only) {
continue;
}

42
src/audio/SDL_audiocvt.c
View File

@@ -180,7 +180,7 @@ ResamplerPadding(const int inrate, const int outrate)
return 0;
}
if (inrate > outrate) {
return (int) SDL_ceilf(((float) (RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float) outrate)));
return (int)SDL_ceilf(((float)(RESAMPLER_SAMPLES_PER_ZERO_CROSSING * inrate) / ((float)outrate)));
}
return RESAMPLER_SAMPLES_PER_ZERO_CROSSING;
}
@@ -246,7 +246,7 @@ SDL_ResampleAudio(const int chans, const int inrate, const int outrate,
outtime = outtimeincr * i;
}
return outframes * chans * sizeof (float);
return outframes * chans * sizeof(float);
}
int
@@ -485,7 +485,7 @@ SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format
/* we keep no streaming state here, so pad with silence on both ends. */
padding = (float *) SDL_calloc(paddingsamples ? paddingsamples : 1, sizeof (float));
if (!padding) {
if (padding == NULL) {
SDL_OutOfMemory();
return;
}
@@ -583,7 +583,7 @@ SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
!!! FIXME in 2.1: We need to store data for this resampler, because the cvt structure doesn't store the original sample rates,
!!! FIXME in 2.1: so we steal the ninth and tenth slot. :( */
if (cvt->filter_index >= (SDL_AUDIOCVT_MAX_FILTERS-2)) {
return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS-2);
return SDL_SetError("Too many filters needed for conversion, exceeded maximum of %d", SDL_AUDIOCVT_MAX_FILTERS - 2);
}
cvt->filters[SDL_AUDIOCVT_MAX_FILTERS-1] = (SDL_AudioFilter) (uintptr_t) src_rate;
cvt->filters[SDL_AUDIOCVT_MAX_FILTERS] = (SDL_AudioFilter) (uintptr_t) dst_rate;
@@ -785,7 +785,7 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
}
cvt->needed = (cvt->filter_index != 0);
return (cvt->needed);
return cvt->needed;
}
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const void *inbuf, const int inbuflen, void *outbuf, const int outbuflen);
@@ -832,7 +832,7 @@ EnsureStreamBufferSize(SDL_AudioStream *stream, const int newlen)
ptr = stream->work_buffer_base;
} else {
ptr = (Uint8 *) SDL_realloc(stream->work_buffer_base, newlen + 32);
if (!ptr) {
if (ptr == NULL) {
SDL_OutOfMemory();
return NULL;
}
@@ -908,12 +908,12 @@ SetupLibSampleRateResampling(SDL_AudioStream *stream)
if (SRC_available) {
state = SRC_src_new(SRC_converter, stream->pre_resample_channels, &result);
if (!state) {
if (state == NULL) {
SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
}
}
if (!state) {
if (state == NULL) {
SDL_CleanupAudioStreamResampler_SRC(stream);
return SDL_FALSE;
}
@@ -980,7 +980,7 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
SDL_AudioStream *retval;
retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
if (!retval) {
if (retval == NULL) {
SDL_OutOfMemory();
return NULL;
}
@@ -1123,7 +1123,7 @@ SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, in
#endif
workbuf = EnsureStreamBufferSize(stream, workbuflen);
if (!workbuf) {
if (workbuf == NULL) {
return -1; /* probably out of memory. */
}
@@ -1190,8 +1190,9 @@ SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, in
if (maxputbytes) {
const int maxbytes = *maxputbytes;
if (buflen > maxbytes)
if (buflen > maxbytes) {
buflen = maxbytes;
}
*maxputbytes -= buflen;
}
@@ -1214,10 +1215,10 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
SDL_Log("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
#endif
if (!stream) {
if (stream == NULL) {
return SDL_InvalidParamError("stream");
}
if (!buf) {
if (buf == NULL) {
return SDL_InvalidParamError("buf");
}
if (len == 0) {
@@ -1269,7 +1270,7 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
int SDL_AudioStreamFlush(SDL_AudioStream *stream)
{
if (!stream) {
if (stream == NULL) {
return SDL_InvalidParamError("stream");
}
@@ -1287,8 +1288,9 @@ int SDL_AudioStreamFlush(SDL_AudioStream *stream)
const SDL_bool first_run = stream->first_run;
const int filled = stream->staging_buffer_filled;
int actual_input_frames = filled / stream->src_sample_frame_size;
if (!first_run)
if (!first_run) {
actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels;
}
if (actual_input_frames > 0) { /* don't bother if nothing to flush. */
/* This is how many bytes we're expecting without silence appended. */
@@ -1327,10 +1329,10 @@ SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len)
SDL_Log("AUDIOSTREAM: want to get %d converted bytes\n", len);
#endif
if (!stream) {
if (stream == NULL) {
return SDL_InvalidParamError("stream");
}
if (!buf) {
if (buf == NULL) {
return SDL_InvalidParamError("buf");
}
if (len <= 0) {
@@ -1340,20 +1342,20 @@ SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len)
return SDL_SetError("Can't request partial sample frames");
}
return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
return (int)SDL_ReadFromDataQueue(stream->queue, buf, len);
}
/* number of converted/resampled bytes available */
int
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
{
return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
return stream ? (int)SDL_CountDataQueue(stream->queue) : 0;
}
void
SDL_AudioStreamClear(SDL_AudioStream *stream)
{
if (!stream) {
if (stream == NULL) {
SDL_InvalidParamError("stream");
} else {
SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);

3
src/audio/SDL_audiodev.c
View File

@@ -83,8 +83,9 @@ SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*
const char *audiodev;
char audiopath[1024];
if (test == NULL)
if (test == NULL) {
test = test_stub;
}
/* Figure out what our audio device is */
if (((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) &&

14
src/audio/SDL_wave.c
View File

@@ -818,7 +818,7 @@ IMA_ADPCM_Init(WaveFile *file, size_t datalength)
/* There's no specification for this, but it's basically the same
* format because the extensible header has wSampePerBlocks too.
*/
} else {
} else {
/* The Standards Update says there 'should' be 2 bytes for wSamplesPerBlock. */
if (chunk->size >= 20 && format->extsize >= 2) {
format->samplesperblock = chunk->data[18] | ((Uint16)chunk->data[19] << 8);
@@ -899,14 +899,18 @@ IMA_ADPCM_ProcessNibble(Sint8 *cindex, Sint16 lastsample, Uint8 nybble)
* (nybble & 0x8 ? -1 : 1) * ((nybble & 0x7) * step / 4 + step / 8)
*/
delta = step >> 3;
if (nybble & 0x04)
if (nybble & 0x04) {
delta += step;
if (nybble & 0x02)
}
if (nybble & 0x02) {
delta += step >> 1;
if (nybble & 0x01)
}
if (nybble & 0x01) {
delta += step >> 2;
if (nybble & 0x08)
}
if (nybble & 0x08) {
delta = -delta;
}
sample = lastsample + delta;

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

@@ -439,7 +439,7 @@ SDL_bool aaudio_DetectBrokenPlayState( void )
int64_t framePosition, timeNanoseconds;
aaudio_result_t res;
if ( !audioDevice || !audioDevice->hidden ) {
if (audioDevice == NULL || !audioDevice->hidden ) {
return SDL_FALSE;
}

69
src/audio/alsa/SDL_alsa_audio.c
View File

@@ -230,7 +230,7 @@ get_audio_device(void *handle, const int channels)
const char *device;
if (handle != NULL) {
return (const char *) handle;
return (const char *)handle;
}
/* !!! FIXME: we also check "SDL_AUDIO_DEVICE_NAME" at the higher level. */
@@ -398,8 +398,7 @@ ALSA_PlayDevice(_THIS)
return;
}
continue;
}
else if (status == 0) {
} else if (status == 0) {
/* No frames were written (no available space in pcm device).
Allow other threads to catch up. */
Uint32 delay = (frames_left / 2 * 1000) / this->spec.freq;
@@ -414,7 +413,7 @@ ALSA_PlayDevice(_THIS)
static Uint8 *
ALSA_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
return this->hidden->mixbuf;
}
static int
@@ -438,8 +437,7 @@ ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
if (status == -EAGAIN) {
ALSA_snd_pcm_wait(this->hidden->pcm_handle, wait_time);
status = 0;
}
else if (status < 0) {
} else if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
@@ -500,7 +498,7 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
status = ALSA_snd_pcm_hw_params_set_period_size_near(
this->hidden->pcm_handle, hwparams, &persize, NULL);
if ( status < 0 ) {
return(-1);
return -1;
}
/* Need to at least double buffer */
@@ -508,19 +506,19 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
status = ALSA_snd_pcm_hw_params_set_periods_min(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
return -1;
}
status = ALSA_snd_pcm_hw_params_set_periods_first(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
return -1;
}
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
if ( status < 0 ) {
return(-1);
return -1;
}
this->spec.samples = persize;
@@ -536,7 +534,7 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params)
persize, periods, bufsize);
}
return(0);
return 0;
}
static int
@@ -572,8 +570,7 @@ ALSA_OpenDevice(_THIS, const char *devname)
SND_PCM_NONBLOCK);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't open audio device: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't open audio device: %s", ALSA_snd_strerror(status));
}
this->hidden->pcm_handle = pcm_handle;
@@ -582,16 +579,14 @@ ALSA_OpenDevice(_THIS, const char *devname)
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't get hardware config: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't get hardware config: %s", ALSA_snd_strerror(status));
}
/* SDL only uses interleaved sample output */
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't set interleaved access: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't set interleaved access: %s", ALSA_snd_strerror(status));
}
/* Try for a closest match on audio format */
@@ -673,8 +668,7 @@ ALSA_OpenDevice(_THIS, const char *devname)
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't set audio frequency: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't set audio frequency: %s", ALSA_snd_strerror(status));
}
this->spec.freq = rate;
@@ -688,24 +682,20 @@ ALSA_OpenDevice(_THIS, const char *devname)
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't get software config: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't get software config: %s", ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
if (status < 0) {
return SDL_SetError("Couldn't set minimum available samples: %s",
ALSA_snd_strerror(status));
return SDL_SetError("Couldn't set minimum available samples: %s", ALSA_snd_strerror(status));
}
status =
ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
if (status < 0) {
return SDL_SetError("ALSA: Couldn't set start threshold: %s",
ALSA_snd_strerror(status));
return SDL_SetError("ALSA: Couldn't set start threshold: %s", ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
if (status < 0) {
return SDL_SetError("Couldn't set software audio parameters: %s",
ALSA_snd_strerror(status));
return SDL_SetError("Couldn't set software audio parameters: %s", ALSA_snd_strerror(status));
}
/* Calculate the final parameters for this audio specification */
@@ -746,7 +736,7 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
char *handle = NULL;
char *ptr;
if (!dev) {
if (dev == NULL) {
return;
}
@@ -756,7 +746,7 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
Make sure not to free the storage associated with desc in this case */
if (hint) {
desc = ALSA_snd_device_name_get_hint(hint, "DESC");
if (!desc) {
if (desc == NULL) {
SDL_free(dev);
return;
}
@@ -776,7 +766,7 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
/*printf("ALSA: adding %s device '%s' (%s)\n", iscapture ? "capture" : "output", name, desc);*/
handle = SDL_strdup(name);
if (!handle) {
if (handle == NULL) {
if (hint) {
free(desc);
}
@@ -789,8 +779,9 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
* enumeration time
*/
SDL_AddAudioDevice(iscapture, desc, NULL, handle);
if (hint)
if (hint) {
free(desc);
}
dev->name = handle;
dev->iscapture = iscapture;
dev->next = *pSeen;
@@ -829,7 +820,7 @@ ALSA_HotplugIteration(void)
if we can find a preferred prefix for the system. */
for (i = 0; hints[i]; i++) {
char *name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
if (!name) {
if (name == NULL) {
continue;
}
@@ -858,17 +849,17 @@ ALSA_HotplugIteration(void)
char *name;
/* if we didn't find a device name prefix we like at all... */
if ((!match) && (defaultdev != i)) {
if ((match == NULL) && (defaultdev != i)) {
continue; /* ...skip anything that isn't the default device. */
}
name = ALSA_snd_device_name_get_hint(hints[i], "NAME");
if (!name) {
if (name == NULL) {
continue;
}
/* only want physical hardware interfaces */
if (!match || (SDL_strncmp(name, match, match_len) == 0)) {
if (match == NULL || (SDL_strncmp(name, match, match_len) == 0)) {
char *ioid = ALSA_snd_device_name_get_hint(hints[i], "IOID");
const SDL_bool isoutput = (ioid == NULL) || (SDL_strcmp(ioid, "Output") == 0);
const SDL_bool isinput = (ioid == NULL) || (SDL_strcmp(ioid, "Input") == 0);
@@ -893,8 +884,12 @@ ALSA_HotplugIteration(void)
}
dev->next = seen;
seen = dev;
if (isinput) have_input = SDL_TRUE;
if (isoutput) have_output = SDL_TRUE;
if (isinput) {
have_input = SDL_TRUE;
}
if (isoutput) {
have_output = SDL_TRUE;
}
} else {
prev = dev;
}

14
src/audio/android/SDL_androidaudio.c
View File

@@ -146,26 +146,24 @@ void ANDROIDAUDIO_PauseDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (SDL_AtomicGet(&audioDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
} else {
SDL_LockMutex(audioDevice->mixer_lock);
SDL_AtomicSet(&audioDevice->paused, 1);
private->resume = SDL_TRUE;
}
}
if(captureDevice != NULL && captureDevice->hidden != NULL) {
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
if (SDL_AtomicGet(&captureDevice->paused)) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
} else {
SDL_LockMutex(captureDevice->mixer_lock);
SDL_AtomicSet(&captureDevice->paused, 1);
private->resume = SDL_TRUE;
@@ -178,7 +176,7 @@ void ANDROIDAUDIO_ResumeDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
if (audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&audioDevice->paused, 0);
@@ -187,7 +185,7 @@ void ANDROIDAUDIO_ResumeDevices(void)
}
}
if(captureDevice != NULL && captureDevice->hidden != NULL) {
if (captureDevice != NULL && captureDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) captureDevice->hidden;
if (private->resume) {
SDL_AtomicSet(&captureDevice->paused, 0);

6
src/audio/directsound/SDL_directsound.c
View File

@@ -293,7 +293,7 @@ DSOUND_GetDeviceBuf(_THIS)
}
if (result != DS_OK) {
SetDSerror("DirectSound GetCurrentPosition", result);
return (NULL);
return NULL;
}
cursor /= this->spec.size;
#ifdef DEBUG_SOUND
@@ -328,9 +328,9 @@ DSOUND_GetDeviceBuf(_THIS)
}
if (result != DS_OK) {
SetDSerror("DirectSound Lock", result);
return (NULL);
return NULL;
}
return (this->hidden->locked_buf);
return this->hidden->locked_buf;
}
static int

2
src/audio/disk/SDL_diskaudio.c
View File

@@ -65,7 +65,7 @@ DISKAUDIO_PlayDevice(_THIS)
static Uint8 *
DISKAUDIO_GetDeviceBuf(_THIS)
{
return (_this->hidden->mixbuf);
return _this->hidden->mixbuf;
}
static int

11
src/audio/dsp/SDL_dspaudio.c
View File

@@ -80,12 +80,13 @@ DSP_OpenDevice(_THIS, const char *devname)
/* Make sure fragment size stays a power of 2, or OSS fails. */
/* I don't know which of these are actually legal values, though... */
if (this->spec.channels > 8)
if (this->spec.channels > 8) {
this->spec.channels = 8;
else if (this->spec.channels > 4)
} else if (this->spec.channels > 4) {
this->spec.channels = 4;
else if (this->spec.channels > 2)
} else if (this->spec.channels > 2) {
this->spec.channels = 2;
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
@@ -258,13 +259,13 @@ DSP_PlayDevice(_THIS)
static Uint8 *
DSP_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
return this->hidden->mixbuf;
}
static int
DSP_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
return (int) read(this->hidden->audio_fd, buffer, buflen);
return (int)read(this->hidden->audio_fd, buffer, buflen);
}
static void

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

@@ -206,7 +206,7 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, const char *devname)
/* create context */
result = MAIN_THREAD_EM_ASM_INT({
if(typeof(Module['SDL3']) === 'undefined') {
if (typeof(Module['SDL3']) === 'undefined') {
Module['SDL3'] = {};
}
var SDL3 = Module['SDL3'];

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

@@ -310,7 +310,7 @@ JACK_OpenDevice(_THIS, const char *devname)
}
devports = JACK_jack_get_ports(client, NULL, NULL, JackPortIsPhysical | sysportflags);
if (!devports || !devports[0]) {
if (devports == NULL || !devports[0]) {
return SDL_SetError("No physical JACK ports available");
}

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

@@ -144,7 +144,7 @@ NETBSDAUDIO_PlayDevice(_THIS)
static Uint8 *
NETBSDAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
return this->hidden->mixbuf;
}

4
src/audio/openslES/SDL_openslES.c
View File

@@ -423,7 +423,7 @@ openslES_CreatePCMPlayer(_THIS)
it can be done as described here:
https://developer.android.com/ndk/guides/audio/opensl/android-extensions.html#floating-point
*/
if(SDL_GetAndroidSDKVersion() >= 21) {
if (SDL_GetAndroidSDKVersion() >= 21) {
SDL_AudioFormat test_format;
for (test_format = SDL_FirstAudioFormat(this->spec.format); test_format; test_format = SDL_NextAudioFormat()) {
if (SDL_AUDIO_ISSIGNED(test_format)) {
@@ -498,7 +498,7 @@ openslES_CreatePCMPlayer(_THIS)
break;
}
if(SDL_AUDIO_ISFLOAT(this->spec.format)) {
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
/* Copy all setup into PCM EX structure */
format_pcm_ex.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
format_pcm_ex.endianness = format_pcm.endianness;

10
src/audio/pipewire/SDL_pipewire.c
View File

@@ -1062,7 +1062,7 @@ input_callback(void *data)
}
pw_buf = PIPEWIRE_pw_stream_dequeue_buffer(stream);
if (!pw_buf) {
if (pw_buf == NULL) {
return;
}
@@ -1186,15 +1186,15 @@ PIPEWIRE_OpenDevice(_THIS, const char *devname)
/* Get the hints for the application name, stream name and role */
app_name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_APP_NAME);
if (!app_name || *app_name == '\0') {
if (app_name == NULL || *app_name == '\0') {
app_name = SDL_GetHint(SDL_HINT_APP_NAME);
if (!app_name || *app_name == '\0') {
if (app_name == NULL || *app_name == '\0') {
app_name = "SDL Application";
}
}
stream_name = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_STREAM_NAME);
if (!stream_name || *stream_name == '\0') {
if (stream_name == NULL || *stream_name == '\0') {
stream_name = "Audio Stream";
}
@@ -1203,7 +1203,7 @@ PIPEWIRE_OpenDevice(_THIS, const char *devname)
* but 'Game' seems more appropriate for the majority of SDL applications.
*/
stream_role = SDL_GetHint(SDL_HINT_AUDIO_DEVICE_STREAM_ROLE);
if (!stream_role || *stream_role == '\0') {
if (stream_role == NULL || *stream_role == '\0') {
stream_role = "Game";
}

3
src/audio/ps2/SDL_ps2audio.c
View File

@@ -154,8 +154,9 @@ static void PS2AUDIO_Deinitialize(void)
static SDL_bool PS2AUDIO_Init(SDL_AudioDriverImpl * impl)
{
if(init_audio_driver() < 0)
if (init_audio_driver() < 0) {
return SDL_FALSE;
}
/* Set the function pointers */
impl->OpenDevice = PS2AUDIO_OpenDevice;

4
src/audio/psp/SDL_pspaudio.c
View File

@@ -102,7 +102,7 @@ PSPAUDIO_OpenDevice(_THIS, const char *devname)
static void PSPAUDIO_PlayDevice(_THIS)
{
if (this->spec.freq != 44100){
if (this->spec.freq != 44100) {
Uint8 *mixbuf = this->hidden->mixbufs[this->hidden->next_buffer];
SDL_assert(this->spec.channels == 2);
sceAudioSRCOutputBlocking(PSP_AUDIO_VOLUME_MAX, mixbuf);
@@ -128,7 +128,7 @@ static Uint8 *PSPAUDIO_GetDeviceBuf(_THIS)
static void PSPAUDIO_CloseDevice(_THIS)
{
if (this->hidden->channel >= 0) {
if (this->spec.freq != 44100){
if (this->spec.freq != 44100) {
sceAudioSRCChRelease();
} else {
sceAudioChRelease(this->hidden->channel);

14
src/audio/pulseaudio/SDL_pulseaudio.c
View File

@@ -49,17 +49,11 @@ static SDL_bool include_monitors = SDL_FALSE;
#if (PA_API_VERSION < 12)
/** Return non-zero if the passed state is one of the connected states */
static SDL_INLINE int PA_CONTEXT_IS_GOOD(pa_context_state_t x) {
return
x == PA_CONTEXT_CONNECTING ||
x == PA_CONTEXT_AUTHORIZING ||
x == PA_CONTEXT_SETTING_NAME ||
x == PA_CONTEXT_READY;
return x == PA_CONTEXT_CONNECTING || x == PA_CONTEXT_AUTHORIZING || x == PA_CONTEXT_SETTING_NAME || x == PA_CONTEXT_READY;
}
/** Return non-zero if the passed state is one of the connected states */
static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x) {
return
x == PA_STREAM_CREATING ||
x == PA_STREAM_READY;
return x == PA_STREAM_CREATING || x == PA_STREAM_READY;
}
#endif /* pulseaudio <= 0.9.10 */
@@ -308,7 +302,7 @@ ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
SDL_assert(mainloop_api); /* this never fails, right? */
context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());
if (!context) {
if (context == NULL) {
PULSEAUDIO_pa_mainloop_free(mainloop);
return SDL_SetError("pa_context_new() failed");
}
@@ -541,7 +535,7 @@ FindDeviceName(struct SDL_PrivateAudioData *h, const SDL_bool iscapture, void *h
SinkDeviceNameCallback, &h->device_name));
}
return (h->device_name != NULL);
return h->device_name != NULL;
}
static int

4
src/audio/vita/SDL_vitaaudio.c
View File

@@ -78,7 +78,7 @@ VITAAUD_OpenDevice(_THIS, const char *devname)
}
}
if(!test_format) {
if (!test_format) {
return SDL_SetError("Unsupported audio format");
}
@@ -108,7 +108,7 @@ VITAAUD_OpenDevice(_THIS, const char *devname)
format = SCE_AUDIO_OUT_MODE_STEREO;
}
if(this->spec.freq < 48000) {
if (this->spec.freq < 48000) {
port = SCE_AUDIO_OUT_PORT_TYPE_BGM;
}

5
src/audio/wasapi/SDL_wasapi_winrt.cpp
View File

@@ -387,8 +387,7 @@ WASAPI_RemoveDevice(const SDL_bool iscapture, LPCWSTR devid)
if (SDL_wcscmp(i->str, devid) == 0) {
if (prev) {
prev->next = next;
}
else {
} else {
deviceid_list = next;
}
SDL_RemoveAudioDevice(iscapture, i->str);
@@ -419,7 +418,7 @@ WASAPI_AddDevice(const SDL_bool iscapture, const char *devname, WAVEFORMATEXTENS
}
devidlist = (DevIdList *)SDL_malloc(sizeof(*devidlist));
if (!devidlist) {
if (devidlist == NULL) {
return; /* oh well. */
}