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AudioOutput.cpp
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AudioOutput.cpp
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#include "AudioOutput.h"
/*
* Audioout fills the audio output buffer.
* If there are no samples available (underrun) mutted sound is send
* Sound data is pulled from databuffer and copied to rtaudio buffer
* A underrun counter is increased for adjusting samplerate of the radio
**/
AudioOutput *audio_output;
void AudioOutput::CopyUnderrunSamples(bool copyUnderrun_)
{
copyUnderrun = copyUnderrun_;
}
bool AudioOutput::createAudioDevice(int SampleRate, unsigned int bufferFrames)
{
auto RtApi = RtAudio::LINUX_ALSA;
string s = Settings_file.find_audio("device");
audio_output = new AudioOutput(SampleRate,bufferFrames, RtApi);
if (audio_output)
{
audio_output->set_volume(50);
audio_output->open(s);
return true;
}
fprintf(stderr, "ERROR: AudioOutput\n");
return false;
}
int AudioOutput::Audioout_class(void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status)
{
double *buffer = (double *) outputBuffer;
if (status)
std::cout << "Stream underflow detected!\n" << std::endl;
// Write interleaved audio data.
if (databuffer.queued_samples() == 0)
{
//Use previous samples incase of buffer underrun
int bytes = nBufferFrames * min(audio_output->get_channels(), 2);
if (underrunSamples.size() && copyUnderrun)
{
int i = 0;
for (auto &col : underrunSamples)
{
Sample v = col;
((double *)buffer)[i++] = v;
}
}
else
{
for (int i = 0; i < bytes; i++)
{
((double *)buffer)[i] = 0.0;
}
}
if (audio_output != nullptr)
audio_output->inc_underrun();
return 0;
}
SampleVector samples = databuffer.pull();
underrunSamples = samples;
//cout << "nBufferFrames " << nBufferFrames << " nSamples " << samples.size() << endl;
int i = 0;
for (auto& col : samples)
{
Sample v = col;
((double *)buffer)[i++] = v;
}
return 0;
}
int AudioOutput::getDevices(std::string device)
{
std::vector<unsigned int> ids = getDeviceIds();
if (ids.size() == 0)
{
std::cout << "No devices found." << std::endl;
return 0;
}
RtAudio::DeviceInfo info;
for (auto col : ids)
{
info = getDeviceInfo(col);
// Print, for example, the name and maximum number of output channels for each device
std::cout << "device name = " << info.name << std::endl;
std::cout << ": maximum output channels = " << info.outputChannels << std::endl;
if (std::string(info.name).find(device) != string::npos && info.outputChannels > 1)
return col;
}
std::cout << "No matching device found." << std::endl;
return 0;
}
AudioOutput::AudioOutput(int pcmrate, unsigned int bufferFrames_, RtAudio::Api api)
: RtAudio(api),
parameters{}, bufferFrames{bufferFrames_}, volume{}, underrun{0}, info{0}
{
sampleRate = pcmrate;
parameters.nChannels = 2;
parameters.firstChannel = 0;
parameters.deviceId = 0;
}
/*
* Open sound device based on name
* if name is default open default device
* GetDevics() fills the map with device names and ID's
* Use samplerate which is optimized for device
**/
bool AudioOutput::open(std::string device)
{
int retry{0};
RtAudioErrorType err;
StreamOptions option{{0}, {0}, {0}, {0}};
option.flags = RTAUDIO_MINIMIZE_LATENCY;
parameters.deviceId = 0;
parameters.firstChannel = 0;
parameters.nChannels = 2;
if (device == "default")
parameters.deviceId = getDefaultOutputDevice(); //getDefaultOutputDevice();
else
parameters.deviceId = getDevices(device);
info = getDeviceInfo(parameters.deviceId);
if (info.preferredSampleRate)
sampleRate = info.preferredSampleRate;
parameters.nChannels = info.outputChannels;
printf("audio device = %d %s samplerate %d channels %d\n", parameters.deviceId, device.c_str(), sampleRate, parameters.nChannels);
err = openStream(¶meters, NULL, RTAUDIO_FLOAT64, sampleRate, &bufferFrames, (RtAudioCallback)Audioout_, (void *)this, NULL);
if (err != RTAUDIO_NO_ERROR)
{
printf("Cannot open audio output stream\n");
return false;
}
startStream();
return true;
}
/*
* Set volume of output use log scale
**/
void AudioOutput::set_volume(int vol)
{
// log volume
volume.store(exp(((double)vol * 6.908) / 100.0) / 1000);
//printf("vol %f\n", (float)m_volume.load());
}
void AudioOutput::adjust_gain(SampleVector& samples)
{
double gain = volume.load();
for (unsigned int i = 0, n = samples.size(); i < n; i++) {
samples[i] *= gain;
}
}
void AudioOutput::adjust_gain(SampleVector &samples_in, SampleVector &samples_out)
{
double gain = volume.load();
for (auto sample : samples_in)
{
samples_out.push_back(gain * sample);
}
}
void AudioOutput::close()
{
if (isStreamOpen())
{
stopStream();
closeStream();
}
}
AudioOutput::~AudioOutput()
{
close();
}
/*
* Write data to audio buffer
**/
bool AudioOutput::write(SampleVector& audiosamples)
{
if (isStreamOpen())
databuffer.push(move(audiosamples));
else
audiosamples.clear();
return true;
}
int AudioOutput::queued_samples()
{
return databuffer.queued_samples();
}
void AudioOutput::writeSamples(const SampleVector &audioSamples)
{
for (auto &col : audioSamples)
{
// split the stream in blocks of samples of the size framesize
audioFrames.insert(audioFrames.end(), col);
if (audioFrames.size() == get_framesize())
{
if ((queued_samples() / 2) < 2048)
{
SampleVector audioStereoSamples;
mono_to_left_right(audioFrames, audioStereoSamples);
write(audioStereoSamples);
audioFrames.clear();
}
else
{
audioFrames.clear();
}
}
}
}
// copy mono signal to both sereo channels
void AudioOutput::mono_to_left_right(const SampleVector &samples_mono,
SampleVector &audio)
{
unsigned int n = samples_mono.size();
if (audio_output->get_channels() < 2)
{
audio = samples_mono;
return;
}
audio.resize(2 * n);
for (unsigned int i = 0; i < n; i++)
{
Sample m = samples_mono[i];
audio[2 * i] = m;
audio[2 * i + 1] = m;
}
}