You cannot select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

399 lines
9.7 KiB
C++

3 years ago
/*
Copyright (C) 2019-2021 Doug McLain
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <https://www.gnu.org/licenses/>.
*/
#include "audioengine.h"
#include <QDebug>
#include <cmath>
#ifdef Q_OS_MACOS
#define MACHAK 1
#else
#define MACHAK 0
#endif
//AudioEngine::AudioEngine(QObject *parent) : QObject(parent)
AudioEngine::AudioEngine(QString in, QString out) :
m_outputdevice(out),
m_inputdevice(in),
m_out(nullptr),
m_in(nullptr),
m_srm(1)
{
m_audio_out_temp_buf_p = m_audio_out_temp_buf;
memset(m_aout_max_buf, 0, sizeof(float) * 200);
m_aout_max_buf_p = m_aout_max_buf;
m_aout_max_buf_idx = 0;
m_aout_gain = 100;
m_volume = 1.0f;
}
AudioEngine::~AudioEngine()
{
//m_indev->disconnect();
//m_in->stop();
//m_outdev->disconnect();
//m_out->stop();
//delete m_in;
//delete m_out;
}
QStringList AudioEngine::discover_audio_devices(uint8_t d)
{
QStringList list;
QAudio::Mode m = (d) ? QAudio::AudioOutput : QAudio::AudioInput;
QList<QAudioDeviceInfo> devices = QAudioDeviceInfo::availableDevices(m);
for (QList<QAudioDeviceInfo>::ConstIterator it = devices.constBegin(); it != devices.constEnd(); ++it ) {
//fprintf(stderr, "Playback device name = %s\n", (*it).deviceName().toStdString().c_str());fflush(stderr);
list.append((*it).deviceName());
}
return list;
}
void AudioEngine::init()
{
QAudioFormat format;
QAudioFormat tempformat;
format.setSampleRate(8000);
format.setChannelCount(1);
format.setSampleSize(16);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt);
m_agc = true;
QList<QAudioDeviceInfo> devices = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
if(devices.size() == 0){
fprintf(stderr, "No audio playback hardware found\n");fflush(stderr);
}
else{
QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
for (QList<QAudioDeviceInfo>::ConstIterator it = devices.constBegin(); it != devices.constEnd(); ++it ) {
if(MACHAK){
qDebug() << "Playback device name = " << (*it).deviceName();
qDebug() << (*it).supportedByteOrders();
qDebug() << (*it).supportedChannelCounts();
qDebug() << (*it).supportedCodecs();
qDebug() << (*it).supportedSampleRates();
qDebug() << (*it).supportedSampleSizes();
qDebug() << (*it).supportedSampleTypes();
qDebug() << (*it).preferredFormat();
}
if((*it).deviceName() == m_outputdevice){
info = *it;
}
}
if (!info.isFormatSupported(format)) {
qWarning() << "Raw audio format not supported by backend, trying nearest format.";
tempformat = info.nearestFormat(format);
qWarning() << "Format now set to " << format.sampleRate() << ":" << format.sampleSize();
}
else{
tempformat = format;
}
fprintf(stderr, "Using playback device %s\n", info.deviceName().toStdString().c_str());fflush(stderr);
m_out = new QAudioOutput(info, tempformat, this);
m_out->setBufferSize(19200);
connect(m_out, SIGNAL(stateChanged(QAudio::State)), this, SLOT(handleStateChanged(QAudio::State)));
//m_outdev = m_out->start();
}
devices = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
if(devices.size() == 0){
fprintf(stderr, "No audio recording hardware found\n");fflush(stderr);
}
else{
QAudioDeviceInfo info(QAudioDeviceInfo::defaultInputDevice());
for (QList<QAudioDeviceInfo>::ConstIterator it = devices.constBegin(); it != devices.constEnd(); ++it ) {
if(MACHAK){
qDebug() << "Capture device name = " << (*it).deviceName();
qDebug() << (*it).supportedByteOrders();
qDebug() << (*it).supportedChannelCounts();
qDebug() << (*it).supportedCodecs();
qDebug() << (*it).supportedSampleRates();
qDebug() << (*it).supportedSampleSizes();
qDebug() << (*it).supportedSampleTypes();
qDebug() << (*it).preferredFormat();
}
if((*it).deviceName() == m_inputdevice){
info = *it;
}
}
if (!info.isFormatSupported(format)) {
qWarning() << "Raw audio format not supported by backend, trying nearest format.";
tempformat = info.nearestFormat(format);
qWarning() << "Format now set to " << format.sampleRate() << ":" << format.sampleSize();
}
else{
tempformat = format;
}
int sr = 8000;
if(MACHAK){
sr = info.preferredFormat().sampleRate();
m_srm = (float)sr / 8000.0;
}
format.setSampleRate(sr);
m_in = new QAudioInput(info, format, this);
fprintf(stderr, "Capture device: %s SR: %d resample factor: %f\n", info.deviceName().toStdString().c_str(), sr, m_srm);fflush(stderr);
}
}
void AudioEngine::start_capture()
{
m_audioinq.clear();
if(m_in != nullptr){
m_indev = m_in->start();
connect(m_indev, SIGNAL(readyRead()), SLOT(input_data_received()));
}
}
void AudioEngine::stop_capture()
{
if(m_in != nullptr){
m_indev->disconnect();
m_in->stop();
}
}
void AudioEngine::start_playback()
{
//m_out->reset();
m_outdev = m_out->start();
}
void AudioEngine::stop_playback()
{
m_out->stop();
}
void AudioEngine::input_data_received()
{
QByteArray data;
qint64 len = m_in->bytesReady();
if (len > 0){
data.resize(len);
m_indev->read(data.data(), len);
/*
fprintf(stderr, "AUDIOIN: ");
for(int i = 0; i < len; ++i){
fprintf(stderr, "%02x ", (unsigned char)data.data()[i]);
}
fprintf(stderr, "\n");
fflush(stderr);
*/
if(MACHAK){
std::vector<int16_t> samples;
for(int i = 0; i < len; i += 2){
samples.push_back(((data.data()[i+1] << 8) & 0xff00) | (data.data()[i] & 0xff));
}
for(float i = 0; i < (float)len/2; i += m_srm){
m_audioinq.enqueue(samples[i]);
}
}
else{
for(int i = 0; i < len; i += (2 * m_srm)){
m_audioinq.enqueue(((data.data()[i+1] << 8) & 0xff00) | (data.data()[i] & 0xff));
}
}
}
}
void AudioEngine::write(int16_t *pcm, size_t s)
{
m_maxlevel = 0;
/*
fprintf(stderr, "AUDIOOUT: ");
for(int i = 0; i < s; ++i){
fprintf(stderr, "%04x ", (uint16_t)pcm[i]);
}
fprintf(stderr, "\n");
fflush(stderr);
*/
if(m_agc){
process_audio(pcm, s);
}
m_outdev->write((const char *) pcm, sizeof(int16_t) * s);
for(uint32_t i = 0; i < s; ++i){
if(pcm[i] > m_maxlevel){
m_maxlevel = pcm[i];
}
}
}
uint16_t AudioEngine::read(int16_t *pcm, int s)
{
m_maxlevel = 0;
if(m_audioinq.size() >= s){
for(int i = 0; i < s; ++i){
pcm[i] = m_audioinq.dequeue();
if(pcm[i] > m_maxlevel){
m_maxlevel = pcm[i];
}
}
return 1;
}
else if(m_in == nullptr){
memset(pcm, 0, sizeof(int16_t) * s);
return 1;
}
else{
//fprintf(stderr, "audio frame not avail size == %d\n", m_audioinq.size());
return 0;
}
}
uint16_t AudioEngine::read(int16_t *pcm)
{
int s;
m_maxlevel = 0;
if(m_audioinq.size() >= 160){
s = 160;
}
else{
s = m_audioinq.size();
}
for(int i = 0; i < s; ++i){
pcm[i] = m_audioinq.dequeue();
if(pcm[i] > m_maxlevel){
m_maxlevel = pcm[i];
}
}
return s;
}
// process_audio() based on code from DSD https://github.com/szechyjs/dsd
void AudioEngine::process_audio(int16_t *pcm, size_t s)
{
float aout_abs, max, gainfactor, gaindelta, maxbuf;
for(size_t i = 0; i < s; ++i){
m_audio_out_temp_buf[i] = static_cast<float>(pcm[i]);
}
// detect max level
max = 0;
m_audio_out_temp_buf_p = m_audio_out_temp_buf;
for (size_t i = 0; i < s; i++){
aout_abs = fabsf(*m_audio_out_temp_buf_p);
if (aout_abs > max){
max = aout_abs;
}
m_audio_out_temp_buf_p++;
}
*m_aout_max_buf_p = max;
m_aout_max_buf_p++;
m_aout_max_buf_idx++;
if (m_aout_max_buf_idx > 24){
m_aout_max_buf_idx = 0;
m_aout_max_buf_p = m_aout_max_buf;
}
// lookup max history
for (size_t i = 0; i < 25; i++){
maxbuf = m_aout_max_buf[i];
if (maxbuf > max){
max = maxbuf;
}
}
// determine optimal gain level
if (max > static_cast<float>(0)){
gainfactor = (static_cast<float>(30000) / max);
}
else{
gainfactor = static_cast<float>(50);
}
if (gainfactor < m_aout_gain){
m_aout_gain = gainfactor;
gaindelta = static_cast<float>(0);
}
else{
if (gainfactor > static_cast<float>(50)){
gainfactor = static_cast<float>(50);
}
gaindelta = gainfactor - m_aout_gain;
if (gaindelta > (static_cast<float>(0.05) * m_aout_gain)){
gaindelta = (static_cast<float>(0.05) * m_aout_gain);
}
}
gaindelta /= static_cast<float>(160);
// adjust output gain
m_audio_out_temp_buf_p = m_audio_out_temp_buf;
for (size_t i = 0; i < 160; i++){
*m_audio_out_temp_buf_p = (m_aout_gain + (static_cast<float>(i) * gaindelta)) * (*m_audio_out_temp_buf_p);
m_audio_out_temp_buf_p++;
}
m_aout_gain += (static_cast<float>(s) * gaindelta);
m_audio_out_temp_buf_p = m_audio_out_temp_buf;
for (size_t i = 0; i < s; i++){
*m_audio_out_temp_buf_p *= m_volume;
if (*m_audio_out_temp_buf_p > static_cast<float>(32760)){
*m_audio_out_temp_buf_p = static_cast<float>(32760);
}
else if (*m_audio_out_temp_buf_p < static_cast<float>(-32760)){
*m_audio_out_temp_buf_p = static_cast<float>(-32760);
}
pcm[i] = static_cast<int16_t>(*m_audio_out_temp_buf_p);
m_audio_out_temp_buf_p++;
}
}
void AudioEngine::handleStateChanged(QAudio::State newState)
{
switch (newState) {
case QAudio::ActiveState:
//qDebug() << "AudioOut state active";
break;
case QAudio::SuspendedState:
//qDebug() << "AudioOut state suspended";
break;
case QAudio::IdleState:
//qDebug() << "AudioOut state idle";
break;
case QAudio::StoppedState:
//qDebug() << "AudioOut state stopped";
break;
default:
break;
}
}