gzdoom-gles/src/sound/sndfile_decoder.cpp

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#include "sndfile_decoder.h"
#include "files.h"
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#ifdef HAVE_SNDFILE
sf_count_t SndFileDecoder::file_get_filelen(void *user_data)
{
FileReader *reader = reinterpret_cast<SndFileDecoder*>(user_data)->Reader;
return reader->GetLength();
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}
sf_count_t SndFileDecoder::file_seek(sf_count_t offset, int whence, void *user_data)
{
FileReader *reader = reinterpret_cast<SndFileDecoder*>(user_data)->Reader;
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if(reader->Seek(offset, whence) != 0)
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return -1;
return reader->Tell();
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}
sf_count_t SndFileDecoder::file_read(void *ptr, sf_count_t count, void *user_data)
{
FileReader *reader = reinterpret_cast<SndFileDecoder*>(user_data)->Reader;
return reader->Read(ptr, count);
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}
sf_count_t SndFileDecoder::file_write(const void *ptr, sf_count_t count, void *user_data)
{
return -1;
}
sf_count_t SndFileDecoder::file_tell(void *user_data)
{
FileReader *reader = reinterpret_cast<SndFileDecoder*>(user_data)->Reader;
return reader->Tell();
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}
SndFileDecoder::~SndFileDecoder()
{
if(SndFile)
sf_close(SndFile);
SndFile = 0;
}
bool SndFileDecoder::open(FileReader *reader)
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{
SF_VIRTUAL_IO sfio = { file_get_filelen, file_seek, file_read, file_write, file_tell };
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Reader = reader;
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SndFile = sf_open_virtual(&sfio, SFM_READ, &SndInfo, this);
if(SndFile)
{
if(SndInfo.channels == 1 || SndInfo.channels == 2)
return true;
sf_close(SndFile);
SndFile = 0;
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}
return false;
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}
void SndFileDecoder::getInfo(int *samplerate, ChannelConfig *chans, SampleType *type)
{
*samplerate = SndInfo.samplerate;
if(SndInfo.channels == 2)
*chans = ChannelConfig_Stereo;
else
*chans = ChannelConfig_Mono;
*type = SampleType_Int16;
}
size_t SndFileDecoder::read(char *buffer, size_t bytes)
{
short *out = (short*)buffer;
size_t frames = bytes / SndInfo.channels / 2;
size_t total = 0;
// It seems libsndfile has a bug with converting float samples from Vorbis
// to the 16-bit shorts we use, which causes some PCM samples to overflow
// and wrap, creating static. So instead, read the samples as floats and
// convert to short ourselves.
// Use a loop to convert a handful of samples at a time, avoiding a heap
// allocation for temporary storage. 64 at a time works, though maybe it
// could be more.
while(total < frames)
{
size_t todo = std::min<size_t>(frames-total, 64/SndInfo.channels);
float tmp[64];
size_t got = sf_readf_float(SndFile, tmp, todo);
if(got < todo) frames = total + got;
for(size_t i = 0;i < got*SndInfo.channels;i++)
*out++ = (short)((std::min)((std::max)(tmp[i] * 32767.f, -32768.f), 32767.f));
total += got;
}
return total * SndInfo.channels * 2;
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}
std::vector<char> SndFileDecoder::readAll()
{
if(SndInfo.frames <= 0)
return SoundDecoder::readAll();
int framesize = 2 * SndInfo.channels;
std::vector<char> output;
output.resize(SndInfo.frames * framesize);
size_t got = read(&output[0], output.size());
output.resize(got);
return output;
}
bool SndFileDecoder::seek(size_t ms_offset)
{
size_t smp_offset = (size_t)((double)ms_offset / 1000. * SndInfo.samplerate);
if(sf_seek(SndFile, smp_offset, SEEK_SET) < 0)
return false;
return true;
}
size_t SndFileDecoder::getSampleOffset()
{
return sf_seek(SndFile, 0, SEEK_CUR);
}
size_t SndFileDecoder::getSampleLength()
{
return (SndInfo.frames > 0) ? SndInfo.frames : 0;
}
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#endif