mirror of https://github.com/AxioDL/amuse.git
devkitA64 compilation fixes
This commit is contained in:
parent
6da262355a
commit
fe04c9a137
|
@ -514,19 +514,19 @@ void AudioGroupSampleDirectory::EntryData::patchMetadataWAV(SystemStringView wav
|
||||||
|
|
||||||
/* File timestamps reflect actual audio content, not loop/pitch data */
|
/* File timestamps reflect actual audio content, not loop/pitch data */
|
||||||
static void SetAudioFileTime(const SystemString& path, const Sstat& stat) {
|
static void SetAudioFileTime(const SystemString& path, const Sstat& stat) {
|
||||||
|
#ifndef __SWITCH__
|
||||||
#if _WIN32
|
#if _WIN32
|
||||||
__utimbuf64 times = {stat.st_atime, stat.st_mtime};
|
__utimbuf64 times = {stat.st_atime, stat.st_mtime};
|
||||||
_wutime64(path.c_str(), ×);
|
_wutime64(path.c_str(), ×);
|
||||||
#else
|
#else
|
||||||
#if __APPLE__
|
#if __APPLE__
|
||||||
struct timespec times[] = {stat.st_atimespec, stat.st_mtimespec};
|
struct timespec times[] = {stat.st_atimespec, stat.st_mtimespec};
|
||||||
#elif __SWITCH__
|
|
||||||
struct timespec times[] = {stat.st_atime, stat.st_mtime};
|
|
||||||
#else
|
#else
|
||||||
struct timespec times[] = {stat.st_atim, stat.st_mtim};
|
struct timespec times[] = {stat.st_atim, stat.st_mtim};
|
||||||
#endif
|
#endif
|
||||||
utimensat(AT_FDCWD, path.c_str(), times, 0);
|
utimensat(AT_FDCWD, path.c_str(), times, 0);
|
||||||
#endif
|
#endif
|
||||||
|
#endif
|
||||||
}
|
}
|
||||||
|
|
||||||
void AudioGroupSampleDirectory::Entry::patchSampleMetadata(SystemStringView basePath) const {
|
void AudioGroupSampleDirectory::Entry::patchSampleMetadata(SystemStringView basePath) const {
|
||||||
|
@ -718,7 +718,7 @@ void AudioGroupSampleDirectory::_extractCompressed(SampleId id, const EntryData&
|
||||||
#endif
|
#endif
|
||||||
|
|
||||||
uint32_t numSamples = ent.getNumSamples();
|
uint32_t numSamples = ent.getNumSamples();
|
||||||
atUint64 dataLen;
|
atUint64 dataLen = 0;
|
||||||
if (fmt == SampleFormat::DSP || fmt == SampleFormat::DSP_DRUM) {
|
if (fmt == SampleFormat::DSP || fmt == SampleFormat::DSP_DRUM) {
|
||||||
DSPADPCMHeader header;
|
DSPADPCMHeader header;
|
||||||
header.x0_num_samples = numSamples;
|
header.x0_num_samples = numSamples;
|
||||||
|
|
|
@ -186,7 +186,7 @@ static T ApplyVolume(float vol, T samp) {
|
||||||
}
|
}
|
||||||
|
|
||||||
void Voice::_procSamplePre(int16_t& samp) {
|
void Voice::_procSamplePre(int16_t& samp) {
|
||||||
double dt;
|
double dt = 0.0;
|
||||||
|
|
||||||
/* Block linearized will use a larger `dt` for amplitude sampling;
|
/* Block linearized will use a larger `dt` for amplitude sampling;
|
||||||
* significantly reducing the processing expense */
|
* significantly reducing the processing expense */
|
||||||
|
|
Loading…
Reference in New Issue