void PuReader::readNextPu()
{
clearSamples();
readHeader();
initSamples();
readSamples();
}
bool AudioFormatReader::read (int** destSamples,
int numDestChannels,
int64 startSampleInSource,
int numSamplesToRead,
const bool fillLeftoverChannelsWithCopies)
{
jassert (numDestChannels > 0); // you have to actually give this some channels to work with!
int startOffsetInDestBuffer = 0;
if (startSampleInSource < 0)
{
const int silence = (int) jmin (-startSampleInSource, (int64) numSamplesToRead);
for (int i = numDestChannels; --i >= 0;)
if (destSamples[i] != 0)
zeromem (destSamples[i], sizeof (int) * silence);
startOffsetInDestBuffer += silence;
numSamplesToRead -= silence;
startSampleInSource = 0;
}
if (numSamplesToRead <= 0)
return true;
if (! readSamples (destSamples, jmin ((int) numChannels, numDestChannels), startOffsetInDestBuffer,
startSampleInSource, numSamplesToRead))
return false;
if (numDestChannels > (int) numChannels)
{
if (fillLeftoverChannelsWithCopies)
{
int* lastFullChannel = destSamples[0];
for (int i = numDestChannels; --i > 0;)
{
if (destSamples[i] != 0)
{
lastFullChannel = destSamples[i];
break;
}
}
if (lastFullChannel != 0)
for (int i = numChannels; i < numDestChannels; ++i)
if (destSamples[i] != 0)
memcpy (destSamples[i], lastFullChannel, sizeof (int) * numSamplesToRead);
}
else
{
for (int i = numChannels; i < numDestChannels; ++i)
if (destSamples[i] != 0)
zeromem (destSamples[i], sizeof (int) * numSamplesToRead);
}
}
return true;
}
void CuReconReader::readNextCu()
{
PuReader puRead(this, sourceFile);
while (!cu->isCuCompleted())
{
puRead.readNextPu();
readSamples(puRead);
cu->addPu(*(puRead.getPu()));
}
}
void AudioMixer::_musicCallback(void *udata, uint8_t *stream, uint32_t len)
{
if (_paused) return;
auto pacm = (libfalltergeist::AcmFileType*)(udata);
if (pacm->samplesLeft() <= 0)
{
if (_loop)
{
pacm->rewind();
}
else
{
Mix_HookMusic(NULL,NULL);
return;
}
}
if (pacm->filename().find("music") != std::string::npos)
{
// music is stereo. just fetch
pacm->readSamples((short int*)stream, len/2);
}
else
{
//all other files are mono. double it
uint16_t* tmp = new uint16_t[len/2];
uint16_t* sstr = (uint16_t*)stream;
pacm->readSamples((short int*)tmp, len/4);
for (uint32_t i = 0; i < len/4; i++)
{
sstr[i*2] = tmp[i];
sstr[i*2+1] = tmp[i];
}
delete [] tmp;
}
}
long CoreAudioEncoder::inputDataProc(UInt32 *npackets, AudioBufferList *abl)
{
prepareInputBuffer(abl, *npackets);
AudioBuffer &ab = abl->mBuffers[0];
try {
*npackets = readSamples(ab.mData, *npackets);
} catch (...) {
/* treat as EOF */
*npackets = 0;
}
ab.mDataByteSize = *npackets * m_input_desc.mBytesPerFrame;
if (*npackets == 0)
ab.mData = 0;
return 0;
}
size_t CompositeSource::readSamples(void *buffer, size_t nsamples)
{
if (m_cur_file == m_sources.size())
return 0;
size_t rc = m_sources[m_cur_file]->readSamples(buffer, nsamples);
if (rc > 0) {
m_position += rc;
return rc;
} else {
++m_cur_file;
if (m_cur_file < m_sources.size())
m_sources[m_cur_file]->seekTo(0);
return readSamples(buffer, nsamples);
}
}
/*
Description: read IQ samples from a given WARP node and store them in a given array
Arguments:
samples (double my_complex*) - pointer to sample array
start_sample (int) - offset to the first sample to read
num_samples (int) - number of samples to read (between 1 and 2^15)
node_sock (int) - identifier of the node socket
node_id (int) - identifier of the node
buffer_id (int) - identifier of the buffer
host_id (int) - identifier of the host
*/
void readIQ(double my_complex* samples, int start_sample, int num_samples, int node_sock, int node_id, int buffer_id, int host_id){
assert(initialized==1);
int node_port = 9000 + node_id; // source port at host for node
int max_length = 8938;//1438, 8938 1422, 8928; // number of bytes available for IQ samples after all headers
int num_pkts = (int)(num_samples*4/max_length) + 1 ;
char readIQ_buffer[42] = {0, 0, 0, node_id, 0, host_id, 0, 1, 0, 28, 0, 10, 0, 0, 48, 0, 0, 8, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
char base_ip_addr[20];
char str[15];
sprintf(str, "%d", node_id+1);
strcpy(base_ip_addr, "10.0.0.");
strcat(base_ip_addr, str);
readSamples(samples, node_sock, readIQ_buffer , 42, base_ip_addr, node_port, num_samples, (uint32) buffer_id, start_sample, max_length, num_pkts);
}
void AudioMixer::playACMSound(std::string filename)
{
auto acm = ResourceManager::acmFileType(filename);
if (!acm) return;
Logger::debug("AudioMixer") << "playing: " << acm->filename() << std::endl;
Mix_Chunk *chunk = NULL;
auto it = _sfx.find(acm->filename());
if (it != _sfx.end())
{
chunk=it->second;
}
if (!chunk)
{
acm->init();
auto samples = acm->samples();
uint8_t *memory = new uint8_t [samples * 2];
auto cnt = acm->readSamples((short*)memory, samples)*2;
SDL_AudioCVT cvt;
SDL_BuildAudioCVT(&cvt, AUDIO_S16LSB, 1, 22050, AUDIO_S16LSB, 2, 22050); //convert from mono to stereo
cvt.buf = (Uint8*)malloc(cnt*cvt.len_mult);
memcpy(cvt.buf, (uint8_t*)memory, cnt);
cvt.len = cnt;
SDL_ConvertAudio(&cvt);
// free old buffer
delete [] memory;
// make SDL_mixer chunk
chunk = Mix_QuickLoad_RAW(cvt.buf, cvt.len*cvt.len_ratio);
if (_sfx.size() > 100) // TODO: make this configurable
{
Mix_FreeChunk(_sfx.begin()->second);
_sfx.erase(_sfx.begin());
}
_sfx.insert(std::pair<std::string, Mix_Chunk*>(acm->filename(), chunk));
}
Mix_PlayChannel(-1, chunk, 0);
}
void ExtAFSource::seekTo(int64_t count)
{
int npreroll = 0;
switch (m_iasbd.mFormatID) {
case kAudioFormatMPEGLayer1: npreroll = 1; break;
case kAudioFormatMPEGLayer2: npreroll = 1; break;
case kAudioFormatMPEGLayer3: npreroll = 10; break;
}
int64_t off
= std::max(0LL, count - m_iasbd.mFramesPerPacket * npreroll);
CHECKCA(ExtAudioFileSeek(m_eaf, off));
int32_t distance = count - off;
while (distance > 0) {
size_t nbytes = distance * m_asbd.mBytesPerFrame;
if (nbytes > m_buffer.size())
m_buffer.resize(nbytes);
size_t n = readSamples(&m_buffer[0], distance);
if (n <= 0) break;
distance -= n;
}
}
