void AudioBuffer::convert_rate(AudioBuffer &_dest, unsigned _frames_count, SRC_STATE *_SRC)
{
AudioSpec destspec{AUDIO_FORMAT_F32, m_spec.channels, _dest.rate()};
if(m_spec.format != AUDIO_FORMAT_F32 || _dest.spec() != destspec) {
throw std::logic_error("unsupported format");
}
_frames_count = std::min(frames(),_frames_count);
double rate_ratio = double(destspec.rate)/double(m_spec.rate);
unsigned out_frames = unsigned(ceil(double(_frames_count) * rate_ratio));
if(out_frames==0) {
return;
}
unsigned destpos = _dest.samples();
unsigned destframes = _dest.frames();
_dest.resize_frames(_dest.frames()+out_frames);
#if HAVE_LIBSAMPLERATE
SRC_DATA srcdata;
srcdata.data_in = &at<float>(0);
srcdata.data_out = &_dest.at<float>(destpos);
srcdata.input_frames = _frames_count;
srcdata.output_frames = out_frames;
srcdata.src_ratio = rate_ratio;
int srcresult;
if(_SRC != nullptr) {
srcdata.end_of_input = 0;
srcresult = src_process(_SRC, &srcdata);
} else {
srcdata.end_of_input = 1;
srcresult = src_simple(&srcdata, SRC_SINC_BEST_QUALITY, destspec.channels) ;
}
if(srcresult != 0) {
throw std::runtime_error(std::string("error resampling: ") + src_strerror(srcresult));
}
assert(srcdata.output_frames_gen>=0 && srcdata.output_frames_gen<=out_frames);
if(srcdata.output_frames_gen != out_frames) {
_dest.resize_frames(destframes + srcdata.output_frames_gen);
}
PDEBUGF(LOG_V2, LOG_MIXER, "convert rate: f-in: %d, f-out: %d, gen: %d\n",
_frames_count, out_frames, srcdata.output_frames_gen);
#else
for(unsigned i=destpos; i<_dest.samples(); ++i) {
_dest.operator[]<float>(i) = 0.f;
}
#endif
}
// This one puts some data inside the ring buffer.
void RingBuffer::put(AudioBuffer& buf)
{
std::lock_guard<std::mutex> l(lock_);
const size_t sample_num = buf.frames();
const size_t buffer_size = buffer_.frames();
if (buffer_size == 0)
return;
size_t len = putLength();
if (buffer_size - len < sample_num)
discard(sample_num);
size_t toCopy = sample_num;
// Add more channels if the input buffer holds more channels than the ring.
if (buffer_.channels() < buf.channels())
buffer_.setChannelNum(buf.channels());
size_t in_pos = 0;
size_t pos = endPos_;
while (toCopy) {
size_t block = toCopy;
if (block > buffer_size - pos) // Wrap block around ring ?
block = buffer_size - pos; // Fill in to the end of the buffer
buffer_.copy(buf, block, in_pos, pos);
in_pos += block;
pos = (pos + block) % buffer_size;
toCopy -= block;
}
endPos_ = pos;
not_empty_.notify_all();
}
void AudioBuffer::convert(const AudioSpec &_new_spec)
{
if(_new_spec == m_spec) {
return;
}
AudioSpec new_spec = _new_spec;
AudioBuffer dest[2];
unsigned bufidx = 0;
AudioBuffer *source = this;
if(source->rate() != new_spec.rate) {
#if HAVE_LIBSAMPLERATE
if(source->format() != AUDIO_FORMAT_F32) {
dest[1].set_spec({AUDIO_FORMAT_F32, source->channels(), source->rate()});
source->convert_format(dest[1], source->frames());
source = &dest[1];
}
dest[0].set_spec({source->format(), source->channels(), new_spec.rate});
source->convert_rate(dest[0], source->frames(), nullptr);
source = &dest[0];
bufidx = 1;
#else
new_spec.rate = source->rate();
#endif
}
if(source->channels() != new_spec.channels) {
dest[bufidx].set_spec({source->format(),new_spec.channels,source->rate()});
source->convert_channels(dest[bufidx], source->frames());
source = &dest[bufidx];
bufidx = (bufidx + 1) % 2;
}
if(source->format() != new_spec.format) {
dest[bufidx].set_spec({new_spec.format,source->channels(),source->rate()});
source->convert_format(dest[bufidx], source->frames());
source = &dest[bufidx];
}
if(new_spec != m_spec) {
m_data = source->m_data;
m_spec = new_spec;
}
}
void AudioBuffer::add_frames(const AudioBuffer &_source, unsigned _frames_count)
{
if(_source.spec() != m_spec) {
throw std::logic_error("sound buffers must have the same spec");
}
_frames_count = std::min(_frames_count, _source.frames());
if(_frames_count == 0) {
return;
}
unsigned datalen = _frames_count * frame_size();
auto srcstart = _source.m_data.begin();
m_data.insert(m_data.end(), srcstart, srcstart+datalen);
}
void Resampler::resample(const AudioBuffer &dataIn, AudioBuffer &dataOut)
{
const double inputFreq = dataIn.getSampleRate();
const double outputFreq = dataOut.getSampleRate();
const double sampleFactor = outputFreq / inputFreq;
if (sampleFactor == 1.0)
return;
const size_t nbFrames = dataIn.frames();
const size_t nbChans = dataIn.channels();
if (nbChans != format_.nb_channels) {
// change channel num if needed
int err;
src_delete(src_state_);
src_state_ = src_new(SRC_LINEAR, nbChans, &err);
format_.nb_channels = nbChans;
DEBUG("SRC channel number changed.");
}
if (nbChans != dataOut.channels()) {
DEBUG("Output buffer had the wrong number of channels (in: %d, out: %d).", nbChans, dataOut.channels());
dataOut.setChannelNum(nbChans);
}
size_t inSamples = nbChans * nbFrames;
size_t outSamples = inSamples * sampleFactor;
// grow buffer if needed
floatBufferIn_.resize(inSamples);
floatBufferOut_.resize(outSamples);
scratchBuffer_.resize(outSamples);
SRC_DATA src_data;
src_data.data_in = floatBufferIn_.data();
src_data.data_out = floatBufferOut_.data();
src_data.input_frames = nbFrames;
src_data.output_frames = nbFrames * sampleFactor;
src_data.src_ratio = sampleFactor;
src_data.end_of_input = 0; // More data will come
dataIn.interleaveFloat(floatBufferIn_.data());
src_process(src_state_, &src_data);
/*
TODO: one-shot deinterleave and float-to-short conversion
*/
src_float_to_short_array(floatBufferOut_.data(), scratchBuffer_.data(), outSamples);
dataOut.deinterleave(scratchBuffer_.data(), src_data.output_frames, nbChans);
}
void
JackLayer::write(AudioBuffer &buffer, std::vector<float> &floatBuffer)
{
for (unsigned i = 0; i < out_ringbuffers_.size(); ++i) {
const unsigned inChannel = std::min(i, buffer.channels() - 1);
convertToFloat(*buffer.getChannel(inChannel), floatBuffer);
// write to output
const size_t to_ringbuffer = jack_ringbuffer_write_space(out_ringbuffers_[i]);
const size_t write_bytes = std::min(buffer.frames() * sizeof(floatBuffer[0]), to_ringbuffer);
// FIXME: while we have samples to write AND while we have space to write them
const size_t written_bytes = jack_ringbuffer_write(out_ringbuffers_[i],
(const char *) floatBuffer.data(), write_bytes);
if (written_bytes < write_bytes)
RING_WARN("Dropped %zu bytes for channel %u", write_bytes - written_bytes, i);
}
}
void OpenSLLayer::audioCaptureFillBuffer(AudioBuffer &buffer)
{
RingBufferPool &mbuffer = Manager::instance().getRingBufferPool();
const AudioFormat mainBufferFormat = mbuffer.getInternalAudioFormat();
const bool resample = mainBufferFormat.sample_rate != audioFormat_.sample_rate;
buffer.applyGain(isCaptureMuted_ ? 0.0 : captureGain_);
if (resample) {
int outSamples = buffer.frames() * (static_cast<double>(audioFormat_.sample_rate) / mainBufferFormat.sample_rate);
AudioBuffer out(outSamples, mainBufferFormat);
resampler_->resample(buffer, out);
dcblocker_.process(out);
mainRingBuffer_->put(out);
} else {
dcblocker_.process(buffer);
mainRingBuffer_->put(buffer);
}
}
size_t RingBuffer::get(AudioBuffer& buf, const std::string &call_id)
{
std::lock_guard<std::mutex> l(lock_);
if (hasNoReadOffsets())
return 0;
if (not hasThisReadOffset(call_id))
return 0;
const size_t buffer_size = buffer_.frames();
if (buffer_size == 0)
return 0;
size_t len = getLength(call_id);
const size_t sample_num = buf.frames();
size_t toCopy = std::min(sample_num, len);
if (toCopy and toCopy != sample_num) {
RING_DBG("Partial get: %zu/%zu", toCopy, sample_num);
}
const size_t copied = toCopy;
size_t dest = 0;
size_t startPos = getReadOffset(call_id);
while (toCopy > 0) {
size_t block = toCopy;
if (block > buffer_size - startPos)
block = buffer_size - startPos;
buf.copy(buffer_, block, startPos, dest);
dest += block;
startPos = (startPos + block) % buffer_size;
toCopy -= block;
}
storeReadOffset(startPos, call_id);
return copied;
}
void AudioBuffer::convert_channels(const AudioBuffer &_source, AudioBuffer &_dest,
unsigned _frames_count)
{
unsigned d = _dest.samples();
_dest.resize_frames(_dest.frames()+_frames_count);
if(_source.channels()==1 && _dest.channels()==2) {
//from mono to stereo
for(unsigned i=0; i<_frames_count; ++i) {
T ch1 = _source.at<T>(i);
_dest.operator[]<T>(d+i*2) = ch1;
_dest.operator[]<T>(d+i*2+1) = ch1;
}
} else if(_source.channels()==2 && _dest.channels()==1) {
//from stereo to mono
for(unsigned i=0; i<_frames_count; ++i) {
double ch1 = double(_source.at<T>(i*2));
double ch2 = double(_source.at<T>(i*2+1));
_dest.operator[]<T>(d+i) = T((ch1 + ch2) / 2.0);
}
}
}
size_t
RingBufferPool::getAvailableData(AudioBuffer& buffer, const std::string& call_id)
{
std::lock_guard<std::recursive_mutex> lk(stateLock_);
auto bindings = getReadBindings(call_id);
if (not bindings)
return 0;
// No mixing
if (bindings->size() == 1) {
return (*bindings->cbegin())->get(buffer, call_id);
}
size_t availableSamples = std::numeric_limits<size_t>::max();
for (const auto& rbuf : *bindings)
availableSamples = std::min(availableSamples,
rbuf->availableForGet(call_id));
if (availableSamples == std::numeric_limits<size_t>::max())
return 0;
availableSamples = std::min(availableSamples, buffer.frames());
buffer.resize(availableSamples);
buffer.reset();
buffer.setFormat(internalAudioFormat_);
AudioBuffer mixBuffer(buffer);
for (const auto &rbuf : *bindings) {
if (rbuf->get(mixBuffer, call_id) > 0)
buffer.mix(mixBuffer);
}
return availableSamples;
}
void AudioBuffer::add_frames(const AudioBuffer &_source)
{
add_frames(_source, _source.frames());
}
