void
JackLayer::read(AudioBuffer &buffer)
{
for (unsigned i = 0; i < in_ringbuffers_.size(); ++i) {
const size_t incomingSamples = jack_ringbuffer_read_space(in_ringbuffers_[i]) / sizeof(captureFloatBuffer_[0]);
if (!incomingSamples)
continue;
captureFloatBuffer_.resize(incomingSamples);
buffer.resize(incomingSamples);
// write to output
const size_t from_ringbuffer = jack_ringbuffer_read_space(in_ringbuffers_[i]);
const size_t expected_bytes = std::min(incomingSamples * sizeof(captureFloatBuffer_[0]), from_ringbuffer);
// FIXME: while we have samples to write AND while we have space to write them
const size_t read_bytes = jack_ringbuffer_read(in_ringbuffers_[i],
(char *) captureFloatBuffer_.data(), expected_bytes);
if (read_bytes < expected_bytes) {
RING_WARN("Dropped %zu bytes", expected_bytes - read_bytes);
break;
}
/* Write the data one frame at a time. This is
* inefficient, but makes things simpler. */
// FIXME: this is braindead, we should write blocks of samples at a time
// convert a vector of samples from 1 channel to a float vector
convertFromFloat(captureFloatBuffer_, *buffer.getChannel(i));
}
}
void
AudioSender::process()
{
auto& mainBuffer = Manager::instance().getRingBufferPool();
auto mainBuffFormat = mainBuffer.getInternalAudioFormat();
// compute nb of byte to get corresponding to 1 audio frame
const std::size_t samplesToGet = std::chrono::duration_cast<std::chrono::seconds>(mainBuffFormat.sample_rate * secondsPerPacket_).count();
if (mainBuffer.availableForGet(id_) < samplesToGet) {
const auto wait_time = std::chrono::duration_cast<std::chrono::milliseconds>(secondsPerPacket_);
if (not mainBuffer.waitForDataAvailable(id_, samplesToGet, wait_time))
return;
}
// get data
micData_.setFormat(mainBuffFormat);
micData_.resize(samplesToGet);
const auto samples = mainBuffer.getData(micData_, id_);
if (samples != samplesToGet)
return;
// down/upmix as needed
auto accountAudioCodec = std::static_pointer_cast<AccountAudioCodecInfo>(args_.codec);
micData_.setChannelNum(accountAudioCodec->audioformat.nb_channels, true);
if (mainBuffFormat.sample_rate != accountAudioCodec->audioformat.sample_rate) {
if (not resampler_) {
RING_DBG("Creating audio resampler");
resampler_.reset(new Resampler(accountAudioCodec->audioformat));
}
resampledData_.setFormat(accountAudioCodec->audioformat);
resampledData_.resize(samplesToGet);
resampler_->resample(micData_, resampledData_);
Smartools::getInstance().setLocalAudioCodec(audioEncoder_->getEncoderName());
if (audioEncoder_->encode_audio(resampledData_) < 0)
RING_ERR("encoding failed");
} else {
if (audioEncoder_->encode_audio(micData_) < 0)
RING_ERR("encoding failed");
}
}
void JackLayer::fillWithUrgent(AudioBuffer &buffer, size_t samplesToGet)
{
// Urgent data (dtmf, incoming call signal) come first.
samplesToGet = std::min(samplesToGet, hardwareBufferSize_);
buffer.resize(samplesToGet);
urgentRingBuffer_.get(buffer, RingBufferPool::DEFAULT_ID);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
// Consume the regular one as well (same amount of samples)
Manager::instance().getRingBufferPool().discard(samplesToGet, RingBufferPool::DEFAULT_ID);
}
size_t OpenSLLayer::audioPlaybackFillWithVoice(AudioBuffer &buffer)
{
RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();
size_t got = mainBuffer.getAvailableData(buffer, RingBufferPool::DEFAULT_ID);
buffer.resize(got);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (audioFormat_.sample_rate != mainBuffer.getInternalSamplingRate()) {
DEBUG("OpenSLLayer::audioPlaybackFillWithVoice sample_rate != mainBuffer.getInternalSamplingRate() \n");
AudioBuffer out(buffer, false);
out.setSampleRate(audioFormat_.sample_rate);
resampler_->resample(buffer, out);
buffer = out;
}
return buffer.size();
}
void JackLayer::fillWithVoice(AudioBuffer &buffer, size_t samplesAvail)
{
RingBufferPool &mainBuffer = Manager::instance().getRingBufferPool();
buffer.resize(samplesAvail);
mainBuffer.getData(buffer, RingBufferPool::DEFAULT_ID);
buffer.applyGain(isPlaybackMuted_ ? 0.0 : playbackGain_);
if (audioFormat_.sample_rate != (unsigned) mainBuffer.getInternalSamplingRate()) {
RING_DBG("fillWithVoice sample_rate != mainBuffer.getInternalSamplingRate() \n");
AudioBuffer out(buffer, false);
out.setSampleRate(audioFormat_.sample_rate);
resampler_->resample(buffer, out);
buffer = out;
}
}
void JackLayer::fillWithToneOrRingtone(AudioBuffer &buffer)
{
buffer.resize(hardwareBufferSize_);
AudioLoop *tone = Manager::instance().getTelephoneTone();
AudioLoop *file_tone = Manager::instance().getTelephoneFile();
// In case of a dtmf, the pointers will be set to nullptr once the dtmf length is
// reached. For this reason we need to fill audio buffer with zeros if pointer is nullptr
if (tone) {
tone->getNext(buffer, playbackGain_);
} else if (file_tone) {
file_tone->getNext(buffer, playbackGain_);
} else {
buffer.reset();
}
}
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;
}