void MusicPlayer::play(const char *filename)
{
#ifdef USE_OGG
stb_vorbis_close(v);
stb_vorbis_alloc t;
t.alloc_buffer=this->buf;
t.alloc_buffer_length_in_bytes=STB_BUFSIZE;
// stb_vorbis has char* pointer even though it is not really writing to buffer
v = stb_vorbis_open_filename((char *)filename, &error, &t);
#endif
#ifdef USE_MP3
mpg123_close(mh);
error = mpg123_open(mh, filename);
if (!error)
{
error = mpg123_getformat(mh, &rate, &channels, &encoding);
rate = AUDIO_FREQUENCY;
channels = 2;
mpg123_format_none(mh);
mpg123_format(mh, rate, channels, encoding);
}
#endif
}
Sound::Sound(const std::string& fileName) :
_channel_count(0),
_sample_count(0),
_sample_rate(0)
{
stb_vorbis* file = stb_vorbis_open_filename(const_cast<char*>(fileName.c_str()), 0, 0);
if (file)
{
stb_vorbis_info info = stb_vorbis_get_info(file);
_channel_count = info.channels;
_sample_rate = info.sample_rate;
int samples = stb_vorbis_stream_length_in_samples(file) * _channel_count;
_sample_count = samples;
_samples.resize(samples);
stb_vorbis_get_samples_short_interleaved(file, _channel_count, &_samples.front(), samples);
stb_vorbis_close(file);
}
else
{
throw std::runtime_error("Unable to open audio file");
}
}
void play_file_from_filename(const std::string name) {
currently_playing = std::string(name);
if (audiobuf) linearFree(audiobuf);
if (name.rfind(".flac") != std::string::npos) {
if (!FLAC_decoder) {
FLAC_decoder = FLAC__stream_decoder_new();
FLAC__stream_decoder_set_md5_checking(FLAC_decoder, true);
}
audiobuf_index = 0;
decode_mode = AUDIO_MODE_FLAC;
FLAC__StreamDecoderInitStatus init_status = FLAC__stream_decoder_init_file(FLAC_decoder, name.c_str(), FLAC_write_callback, FLAC_metadata_callback, FLAC_error_callback, NULL);
if(init_status != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
printf("ERROR: initializing decoder: %s\n", FLAC__StreamDecoderInitStatusString[init_status]);
}
FLAC__stream_decoder_process_until_end_of_metadata(FLAC_decoder);
} else {
decode_mode = AUDIO_MODE_VORBIS;
v = stb_vorbis_open_filename(name.c_str(), &error, NULL);
info = stb_vorbis_get_info(v);
Samples = info.sample_rate;
audiobuf_size = Samples * sizeof(s16) * 2;
audiobuf = (s16*)linearAlloc(audiobuf_size);
}
paused = false;
}
// Start music playing (open stream)
void PlayMusicStream(char *fileName)
{
if (strcmp(GetExtension(fileName),"ogg") == 0)
{
// Stop current music, clean buffers, unload current stream
StopMusicStream();
// Open audio stream
currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
if (currentMusic.stream == NULL)
{
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
}
else
{
// Get file info
stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
currentMusic.channels = info.channels;
currentMusic.sampleRate = info.sample_rate;
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(INFO, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
else currentMusic.format = AL_FORMAT_MONO16;
currentMusic.loop = true; // We loop by default
musicEnabled = true;
// Create an audio source
alGenSources(1, ¤tMusic.source); // Generate pointer to audio source
alSourcef(currentMusic.source, AL_PITCH, 1);
alSourcef(currentMusic.source, AL_GAIN, 1);
alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
//alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue!
// Generate two OpenAL buffers
alGenBuffers(2, currentMusic.buffers);
// Fill buffers with music...
BufferMusicStream(currentMusic.buffers[0]);
BufferMusicStream(currentMusic.buffers[1]);
// Queue buffers and start playing
alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
alSourcePlay(currentMusic.source);
// NOTE: Regularly, we must check if a buffer has been processed and refill it: MusicStreamUpdate()
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
}
}
else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
}
void LoadTestChannel(ShadertoyTestResource *channel_data, ShadertoyState *state, ShadertoyInputs *inputs, ShadertoyOutputs *outputs, ShadertoyPass pass, int channel_id) {
ImageFile *image = &image_files[channel_id][pass];
memset(image->filepath, 0, sizeof(image->filepath));
switch (channel_data->type) {
case SHADERTOY_RESOURCE_TEXTURE: {
GetFilePath(image->filepath, channel_data->filename[0]);
LoadTexture(state, image->filepath, pass, channel_id);
break;
}
case SHADERTOY_RESOURCE_CUBE_MAP: {
char *cubemap_paths[6];
for (int j = 0; j < 6; ++j) {
cubemap_paths[j] = new char[1024];
GetFilePath(cubemap_paths[j], channel_data->filename[j]);
if (j == 0) {
strcpy(image->filepath, cubemap_paths[j]);
}
}
LoadTexture(state, cubemap_paths, pass, channel_id);
for (int j = 0; j < 6; ++j) {
delete[] cubemap_paths[j];
}
break;
}
case SHADERTOY_RESOURCE_MUSIC: {
AudioFile *file = &audio_files[channel_id];
memset(file->filepath, 0, sizeof(file->filepath));
GetFilePath(file->filepath, channel_data->filename[0]);
int ogg_error;
stb_vorbis *ogg_data = stb_vorbis_open_filename(file->filepath, &ogg_error, NULL);
file->channels = ogg_data->channels;
file->samples_count = stb_vorbis_stream_length_in_samples(ogg_data) * ogg_data->channels;
file->samples = (float*)malloc(sizeof(float) * file->samples_count);
stb_vorbis_get_samples_float_interleaved(ogg_data, ogg_data->channels, file->samples, file->samples_count);
audio_data[channel_id] = { &inputs->audio_played_samples[channel_id], &file->samples, (int)ogg_data->sample_rate, ogg_data->channels, file->samples_count / ogg_data->channels };
ShadertoyLoadAudio(state, audio_data + channel_id, SHADERTOY_IMAGE_PASS, channel_id);
outputs->music_data_param[channel_id] = init_audio_output(ogg_data->channels, ogg_data->sample_rate, 32, file->samples_count * sizeof(float));
stb_vorbis_close(ogg_data);
break;
}
case SHADERTOY_RESOURCE_MICROPHONE: {
// 512 * sizeof(float) == FFT sampling data, so should be fine to use
inputs->micro_data_param = init_audio_input(2, 44100, 32, 2048 * sizeof(float));
break;
}
case SHADERTOY_RESOURCE_KEYBOARD:
case SHADERTOY_RESOURCE_NONE:
break;
}
}
void
sound_load(Sound *sound, const char *path)
{
int error = 0;
stb_vorbis *stream;
memset(sound, 0, sizeof(Sound));
stream = stb_vorbis_open_filename(path, &error, 0);
ASSERT(!error && stream);
punp_sound_load_stbv(sound, stream);
}
void test_get_frame_short_interleaved(FILE *g, char *filename)
{
short sbuffer[8000];
int n, error;
stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
if (!v) stb_fatal("Couldn't open {%s}", filename);
show_info(v);
while (0 != (n=stb_vorbis_get_frame_short_interleaved(v, 2, sbuffer, 8000))) {
fwrite(sbuffer, 2, n*2, g);
}
stb_vorbis_close(v);
}
bool cSoundFileOgg::IsFileSupported( const std::string& Filename, bool Read ) {
if ( Read ) {
// Open the vorbis stream
stb_vorbis* Stream = stb_vorbis_open_filename( const_cast<char*>( Filename.c_str() ), NULL, NULL );
if ( NULL != Stream ) {
stb_vorbis_close( Stream );
return true;
} else
return false;
} else // No support for writing ogg files yet...
return false;
}
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(char *fileName)
{
Wave wave;
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
if (oggFile == NULL)
{
TraceLog(WARNING, "[%s] OGG file could not be opened", fileName);
wave.data = NULL;
}
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
wave.sampleRate = info.sample_rate;
wave.bitsPerSample = 16;
wave.channels = info.channels;
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
int totalSamplesLength = (stb_vorbis_stream_length_in_samples(oggFile) * info.channels);
wave.dataSize = totalSamplesLength*sizeof(short); // Size must be in bytes
TraceLog(DEBUG, "[%s] Samples length: %i", fileName, totalSamplesLength);
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
TraceLog(DEBUG, "[%s] Total seconds: %f", fileName, totalSeconds);
if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
int totalSamples = totalSeconds*info.sample_rate*info.channels;
TraceLog(DEBUG, "[%s] Total samples calculated: %i", fileName, totalSamples);
wave.data = malloc(sizeof(short)*totalSamplesLength);
int samplesObtained = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, wave.data, totalSamplesLength);
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained);
TraceLog(INFO, "[%s] OGG file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
stb_vorbis_close(oggFile);
}
return wave;
}
AudioStream::AudioStream(const String & filename, AudioSource * source)
{
buffers[0] = buffers[1] = 0;
AudioStream::source = source;
stream = stb_vorbis_open_filename(filename.ToCString(), nullptr, nullptr);
if (!stream) return;
info = stb_vorbis_get_info(stream);
samplesLeft = stb_vorbis_stream_length_in_samples(stream) * info.channels;
alGenBuffers(2, buffers);
Stream(buffers[0]);
Stream(buffers[1]);
alSourceQueueBuffers(AudioStream::source->GetID(), 2, buffers);
streams.Add(this);
}
void test_get_samples_short_interleaved(FILE *g, char *filename)
{
int error;
stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
if (!v) stb_fatal("Couldn't open {%s}", filename);
show_info(v);
for(;;) {
int16 sbuffer[333];
int n;
n = stb_vorbis_get_samples_short_interleaved(v, 2, sbuffer, 333);
if (n == 0)
break;
fwrite(sbuffer, 2, n*2, g);
}
stb_vorbis_close(v);
}
bool AEAudioStreamOpen(AEAudioStream* self, const char* filename){
self->stream = stb_vorbis_open_filename((char*)filename, NULL, NULL);
if(not self->stream) return false;
// Get file info
self->info = stb_vorbis_get_info(self->stream);
if(self->info.channels == 2) self->format = AL_FORMAT_STEREO16;
else self->format = AL_FORMAT_MONO16;
if(not AEAudioStreamStream(self, self->buffers[0])) return false;
if(not AEAudioStreamStream(self, self->buffers[1])) return false;
alSourceQueueBuffers(self->source, 2, self->buffers);
alSourcePlay(self->source);
self->totalSamplesLeft=stb_vorbis_stream_length_in_samples(self->stream) * self->info.channels;
return true;
}
bool cSoundFileOgg::OpenRead( const std::string& Filename, std::size_t& SamplesCount, unsigned int& ChannelCount, unsigned int& SampleRate ) {
// Close the file if already opened
if ( NULL != mStream )
stb_vorbis_close( mStream );
// Open the vorbis stream
mStream = stb_vorbis_open_filename( const_cast<char*>( Filename.c_str() ), NULL, NULL );
if ( NULL == mStream ) {
eePRINTL( "Failed to read sound file %s (cannot open the file)", Filename.c_str() );
return false;
}
// Get the music parameters
stb_vorbis_info Infos = stb_vorbis_get_info( mStream );
ChannelCount = mChannelCount = Infos.channels;
SampleRate = Infos.sample_rate;
SamplesCount = static_cast<std::size_t>( stb_vorbis_stream_length_in_samples( mStream ) * ChannelCount );
return true;
}
void test_get_frame_float(FILE *g, char *filename)
{
int error;
stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
if (!v) stb_fatal("Couldn't open {%s}", filename);
show_info(v);
for(;;) {
int n;
float *left, *right;
float **outputs;
int num_c;
n = stb_vorbis_get_frame_float(v, &num_c, &outputs);
if (n == 0)
break;
left = outputs[0];
right = outputs[num_c > 1];
write_floats(g, n, left, right);
}
stb_vorbis_close(v);
}
void MusicOGG::play(bool loop)
{
if (m_isPlaying) return;
m_loop = loop;
for (auto pBuffer : m_buffers) delete pBuffer;
m_buffers.clear();
m_bufferCount = 0;
m_pStream = stb_vorbis_open_filename((char*)m_filename.c_str(), NULL, NULL);
if (!m_pStream) return;
m_info = stb_vorbis_get_info(m_pStream);
m_sampleCount = stb_vorbis_stream_length_in_samples(m_pStream);
m_isPlaying = true;
m_paused = false;
m_samplingT = 0.0f;
// We will buffer 5 seconds
m_engineChannelCount = m_pStream->channels;
m_bufferMax = oAudioEngine->getSampleRate() * m_engineChannelCount;
for (int i = 0; i < MUSIC_BUFFER_COUNT; ++i)
{
auto pBuffer = new Buffer();
pBuffer->data.assign(m_bufferMax, 0.0f);
m_buffers.push_back(pBuffer);
}
m_bufferCount = 0;
if (m_thread.joinable()) m_thread.join();
m_thread = std::thread(std::bind(&MusicOGG::run, this));
m_musicSampleRate = (int)m_pStream->sample_rate;
m_musicChannelCount = (int)m_pStream->channels;
m_pSamplings = new float[4 * m_pStream->channels];
memset(m_pSamplings, 0, sizeof(float) * 4 * m_pStream->channels);
oAudioEngine->addInstance(OThis);
}
ALuint AEAudioContextBufferLoad(AEAudioContext* self, const char* filename){
ALuint buffer=0;
alGenBuffers(1, &buffer);
stb_vorbis *stream = stb_vorbis_open_filename((char*)filename, NULL, NULL);
if(not stream) return 0;
stb_vorbis_info info = stb_vorbis_get_info(stream);
ALenum format;
if(info.channels == 2) format = AL_FORMAT_STEREO16;
else format = AL_FORMAT_MONO16;
size_t sampleCount = stb_vorbis_stream_length_in_samples(stream) * info.channels;
void* data = malloc(sizeof(ALushort)*sampleCount);
stb_vorbis_get_samples_short_interleaved(stream, info.channels, data, (int)sampleCount);
stb_vorbis_close(stream);
alBufferData(buffer, format, data, sampleCount * sizeof(ALushort), info.sample_rate);
free(data);
return buffer;
}
//***** Streaming
int snd_LoadStreaming( const char* fileName, bool loops, unsigned int group )
{
assert( group >= 0 );
assert( group < sb_Count( sbSoundGroups ) );
int newIdx = -1;
for( int i = 0; i < MAX_STREAMING_SOUNDS; ++i ) {
if( streamingSounds[i].access == NULL ) {
newIdx = i;
break;
}
}
if( newIdx < 0 ) {
llog( LOG_ERROR, "Unable to find empty slot for streaming sound %s", fileName );
return -1;
}
int error;
streamingSounds[newIdx].access = stb_vorbis_open_filename( fileName, &error, NULL );
if( streamingSounds[newIdx].access == NULL ) {
llog( LOG_ERROR, "Unable to acquire a handle to streaming sound %s", fileName );
return -1;
}
streamingSounds[newIdx].playing = false;
streamingSounds[newIdx].loops = loops;
streamingSounds[newIdx].group = group;
streamingSounds[newIdx].channels = (Uint8)( streamingSounds[newIdx].access->channels );
if( streamingSounds[newIdx].channels > 2 ) {
streamingSounds[newIdx].channels = 2;
}
return newIdx;
}
OGGStream(String file, bool _loop)
{
loop = _loop;
finished = true;
valid = false;
std::string name = ToStdString(file);
//cout << "Load OGG " << name << endl;
//OGG file
int error = 0;
ogg = stb_vorbis_open_filename(&name[0], &error, NULL);
if (!ogg) {cout << " VORBIS FAIL" << endl; return;}
info = stb_vorbis_get_info(ogg);
//Info
output.channels = info.channels;
output.rate = info.sample_rate;
//Woot!
valid = true;
finished = false;
}
int main(int argc, char **argv)
{
int num_chan, samprate;
int i, j, test, phase;
short *output;
if (argc == 1) {
fprintf(stderr, "Usage: vorbseek {vorbisfile} [{vorbisfile]*]\n");
fprintf(stderr, "Tests various seek offsets to make sure they're sample exact.\n");
return 0;
}
#if 0
{
// check that outofmem occurs correctly
stb_vorbis_alloc va;
va.alloc_buffer = malloc(1024*1024);
for (i=0; i < 1024*1024; i += 10) {
int error=0;
stb_vorbis *v;
va.alloc_buffer_length_in_bytes = i;
v = stb_vorbis_open_filename(argv[1], &error, &va);
if (v != NULL)
break;
printf("Error %d at %d\n", error, i);
}
}
#endif
for (j=1; j < argc; ++j) {
unsigned int successes=0, attempts = 0;
unsigned int num_samples = stb_vorbis_decode_filename(argv[j], &num_chan, &samprate, &output);
break;
if (num_samples == 0xffffffff) {
fprintf(stderr, "Error: couldn't open file or not vorbis file: %s\n", argv[j]);
goto fail;
}
if (num_chan != 2) {
fprintf(stderr, "vorbseek testing only works with files with 2 channels, %s has %d\n", argv[j], num_chan);
goto fail;
}
for (test=0; test < 5; ++test) {
int error;
stb_vorbis *v = stb_vorbis_open_filename(argv[j], &error, NULL);
if (v == NULL) {
fprintf(stderr, "Couldn't re-open %s for test #%d\n", argv[j], test);
goto fail;
}
for (phase=0; phase < 3; ++phase) {
unsigned int base = phase == 0 ? 0 : phase == 1 ? num_samples - test_count[test]*test_spacing[test] : num_samples/3;
for (i=0; i < test_count[test]; ++i) {
unsigned int pos = base + i*test_spacing[test];
if (pos > num_samples) // this also catches underflows
continue;
successes += try_seeking(v, pos, output, num_samples);
attempts += 1;
}
}
stb_vorbis_close(v);
}
printf("%d of %d seeks failed in %s (%d samples)\n", attempts-successes, attempts, argv[j], num_samples);
free(output);
}
return 0;
fail:
return 1;
}
AudioBuffer::AudioBuffer(const String & filename):alBuffer(0)
{
if (filename.ExtractExt() == "wav") {
alGenBuffers(1, &alBuffer);
File file(filename, FILE_READ);
file.Seek(16);
uint32 fmtChunkSize;
file.ReadBytes(&fmtChunkSize, 4);
uint16 audioFormat;
file.ReadBytes(&audioFormat, 2);
uint16 channels;
file.ReadBytes(&channels, 2);
file.Seek(24);
uint32 sampleRate;
file.ReadBytes(&sampleRate, 4);
file.Seek(34);
uint16 bitsPerSample;
file.ReadBytes(&bitsPerSample, 2);
uint16 extraParamSize;
if (fmtChunkSize != 16) {
file.ReadBytes(&extraParamSize, 2);
file.Seek(38 + extraParamSize);
}
char texto[5];
texto[4] = '\0';
while (String(texto) != "data") {
file.ReadBytes(texto, 4);
}
uint32 dataSize;
uint32* data;
file.ReadBytes(&dataSize, 4);
data = (uint32*)malloc(dataSize);
file.ReadBytes(data, dataSize);
ALenum format;
if (bitsPerSample == 8) {
if (channels == 1) {
format = AL_FORMAT_MONO8;
}
else {
format = AL_FORMAT_STEREO8;
}
}
if (bitsPerSample == 16) {
if (channels == 1) {
format = AL_FORMAT_MONO16;
}
else {
format = AL_FORMAT_STEREO16;
}
}
alBufferData(alBuffer, format, data, dataSize, sampleRate);
free(data);
}
else if (filename.ExtractExt() == "ogg") {
stb_vorbis* stb_vorbis;
stb_vorbis = stb_vorbis_open_filename(filename.ToCString(), nullptr, nullptr);
if (stb_vorbis == nullptr) return;
stb_vorbis_info info = stb_vorbis_get_info(stb_vorbis);
uint32 dataSize = stb_vorbis_stream_length_in_samples(stb_vorbis) * info.channels;
int16* buffer = (int16*)malloc(dataSize * sizeof(uint16));
stb_vorbis_get_samples_short_interleaved(stb_vorbis, info.channels,(short *) buffer, dataSize);
alGenBuffers(1, &alBuffer);
ALenum format;
if (info.channels == 1) {
format = AL_FORMAT_MONO16;
}
else {
format = AL_FORMAT_STEREO16;
}
alBufferData(alBuffer, format, buffer, dataSize * sizeof(int16), info.sample_rate);
free(buffer);
stb_vorbis_close(stb_vorbis);
}
}
//----------------------------------------------------------------------------
int main(int argc, char **argv) {
//----------------------------------------------------------------------------
PrintConsole topScreen;
ndspWaveBuf waveBuf[2];
gfxInitDefault();
consoleInit(GFX_TOP, &topScreen);
consoleSelect(&topScreen);
printf("libctru streaming audio\n");
stb_vorbis_info info;
int error;
vorbisFile = stb_vorbis_open_filename("/mau5.ogg", &error, NULL);
info = stb_vorbis_get_info(vorbisFile);
Samples = info.sample_rate;
u32 *audioBuffer = (u32*)linearAlloc(Samples*sizeof(s16)*2);
bool fillBlock = false;
ndspInit();
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
ndspChnSetInterp(0, NDSP_INTERP_LINEAR);
ndspChnSetRate(0, Samples);
ndspChnSetFormat(0, NDSP_FORMAT_STEREO_PCM16);
float mix[12];
memset(mix, 0, sizeof(mix));
mix[0] = 1.0;
mix[1] = 1.0;
ndspChnSetMix(0, mix);
int note = 4;
memset(waveBuf,0,sizeof(waveBuf));
waveBuf[0].data_vaddr = &audioBuffer[0];
waveBuf[0].nsamples = Samples;
waveBuf[1].data_vaddr = &audioBuffer[Samples];
waveBuf[1].nsamples = Samples;
ndspChnWaveBufAdd(0, &waveBuf[0]);
ndspChnWaveBufAdd(0, &waveBuf[1]);
printf("Press up/down to change tone\n");
while(aptMainLoop()) {
gfxSwapBuffers();
gfxFlushBuffers();
gspWaitForVBlank();
hidScanInput();
u32 kDown = hidKeysDown();
if (kDown & KEY_START)
break; // break in order to return to hbmenu
if (waveBuf[fillBlock].status == NDSP_WBUF_DONE) {
fill_buffer(waveBuf[fillBlock].data_pcm16, waveBuf[fillBlock].nsamples);
ndspChnWaveBufAdd(0, &waveBuf[fillBlock]);
fillBlock = !fillBlock;
}
}
ndspExit();
linearFree(audioBuffer);
gfxExit();
return 0;
}
AudioBuffer::AudioBuffer(const String &filename) {
if (filename.ExtractExt() == "wav") {
m_alBuffer = 0;
File * file = new File(filename, FILE_READ);
int32 chunkId = file->ReadInt();
int32 riffChunkSize = file->ReadInt();
int32 format = file->ReadInt();
int32 subChunkId = file->ReadInt();
int32 fmtChunkSize = file->ReadInt();
int16 audioFormat = file->ReadInt16();
int16 channels = file->ReadInt16();
int32 sampleRate = file->ReadInt();
int32 byteRate = file->ReadInt();
int16 blockAlign = file->ReadInt16();
int16 bitsPerSample = file->ReadInt16();
if (fmtChunkSize > 16) {
int16 extraParamsSize = file->ReadInt16();
file->Seek(file->Pos() + extraParamsSize);
}
char text[5] = "";
file->ReadBytes(text, 4);
while (String(text) != "data" && !file->Eof()) {
file->ReadBytes(text, 4);
if (String(text) == "data")
break;
file->Seek(file->Pos() + 4);
}
int32 dataSize = file->ReadInt();
uint32 * data = (uint32 *)malloc(dataSize);
file->ReadBytes(data, dataSize);
ALenum bufferFormat;
if (bitsPerSample == 8) {
if (channels == 1)
bufferFormat = AL_FORMAT_MONO8;
else
bufferFormat = AL_FORMAT_STEREO8;
}
else {
if (channels == 1)
bufferFormat = AL_FORMAT_MONO16;
else
bufferFormat = AL_FORMAT_STEREO16;
}
alGenBuffers(1, &m_alBuffer);
alBufferData(m_alBuffer, bufferFormat, data, dataSize, sampleRate);
delete file;
free(data);
} else if (filename.ExtractExt() == "ogg") {
stb_vorbis * file = stb_vorbis_open_filename(filename.ToCString(), nullptr, nullptr);
if (file) {
stb_vorbis_info info = stb_vorbis_get_info(file);
uint32 size = stb_vorbis_stream_length_in_samples(file) * info.channels;
int16 * buffer = (int16 *)malloc(size * sizeof(int16));
stb_vorbis_get_samples_short_interleaved(file, info.channels, buffer, size);
alGenBuffers(1, &m_alBuffer);
ALenum bufferFormat;
if (info.channels == 1)
bufferFormat = AL_FORMAT_MONO16;
else
bufferFormat = AL_FORMAT_STEREO16;
alBufferData(m_alBuffer, bufferFormat, buffer, size, info.sample_rate);
stb_vorbis_close(file);
free(buffer);
}
}
}