VulcanoLE/src/VulcanoLE/Audio/AudioGrabber.cpp

199 lines
6.7 KiB
C++

#include <cstring>
#include <cmath>
#include <VulcanoLE/Audio/AudioGrabber.h>
#include <VUtils/Logging.h>
AudioGrabber::AudioGrabber() = default;
AudioGrabber::~AudioGrabber() = default;
bool AudioGrabber::read(stereoSample *buffer, uint32_t bufferSize) {
auto buffer_size_bytes = static_cast<size_t>(sizeof(stereoSample) * bufferSize);
if (m_pulseaudioSimple == nullptr) {
openPulseaudioSource(static_cast<uint32_t>(buffer_size_bytes));
}
if (m_pulseaudioSimple != nullptr) {
memset(buffer, 0, buffer_size_bytes);
int32_t error_code;
auto return_code = pa_simple_read(m_pulseaudioSimple, buffer,
buffer_size_bytes, &error_code);
if (return_code < 0) {
WARN("Could not finish reading pulse Audio stream buffer\n bytes read: %d buffer\n size: %d", return_code,
buffer_size_bytes)
// zero out buffer
memset(buffer, 0, buffer_size_bytes);
pa_simple_free(m_pulseaudioSimple);
m_pulseaudioSimple = nullptr;
return false;
}
// Success fully read entire buffer
return true;
}
return false;
}
void AudioGrabber::populateDefaultSourceName() {
pa_mainloop_api *mainloop_api;
pa_context *pulseaudio_context;
m_pulseaudioMainloop = pa_mainloop_new();
mainloop_api = pa_mainloop_get_api(m_pulseaudioMainloop);
pulseaudio_context = pa_context_new(mainloop_api, "VulcanoLE device list");
pa_context_connect(pulseaudio_context, nullptr, PA_CONTEXT_NOFLAGS,
nullptr);
pa_context_set_state_callback(pulseaudio_context,
pulseaudioContextStateCallback,
reinterpret_cast<void *>(this));
int ret;
if (pa_mainloop_run(m_pulseaudioMainloop, &ret) < 0) {
ERR("Could not open pulseaudio mainloop to find default device name: %d", ret)
}
}
bool AudioGrabber::openPulseaudioSource(uint32_t maxBufferSize) {
int32_t error_code = 0;
static const pa_sample_spec sample_spec = { PA_SAMPLE_FLOAT32NE, sampleRate,
channels };
static const pa_buffer_attr buffer_attr = { maxBufferSize, 0, 0, 0,
(maxBufferSize / 2) };
populateDefaultSourceName();
if (!m_PulseaudioDefaultSourceName.empty()) {
m_pulseaudioSimple =
pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
m_PulseaudioDefaultSourceName.c_str(),
recStreamDescription, &sample_spec,
nullptr, &buffer_attr, &error_code);
}
if (m_pulseaudioSimple == nullptr) {
m_pulseaudioSimple =
pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
nullptr, recStreamDescription,
&sample_spec, nullptr, &buffer_attr, &error_code);
}
if (m_pulseaudioSimple == nullptr) {
m_pulseaudioSimple =
pa_simple_new(nullptr, recStreamName, PA_STREAM_RECORD,
"0", recStreamDescription, &sample_spec,
nullptr, &buffer_attr, &error_code);
}
if (m_pulseaudioSimple != nullptr) {
return true;
}
ERR("Could not open pulseaudio source: %s", pa_strerror(error_code))
return false;
}
void AudioGrabber::pulseaudioContextStateCallback(pa_context *c, void *userdata) {
switch (pa_context_get_state(c)) {
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
case PA_CONTEXT_READY: {
pa_operation_unref(pa_context_get_server_info(
c, pulseaudioServerInfoCallback, userdata));
break;
}
case PA_CONTEXT_FAILED:
case PA_CONTEXT_TERMINATED:
auto *src =
reinterpret_cast<AudioGrabber *>(userdata);
pa_mainloop_quit(src->m_pulseaudioMainloop, 0);
break;
}
}
void AudioGrabber::pulseaudioServerInfoCallback(pa_context *context, const pa_server_info *i, void *userdata) {
if (i != nullptr) {
auto *src = reinterpret_cast<AudioGrabber *>(userdata);
std::string name = i->default_sink_name;
name.append(defaultMonitorPostfix);
src->m_PulseaudioDefaultSourceName = name;
// stop mainloop after finding default name
pa_mainloop_quit(src->m_pulseaudioMainloop, 0);
}
}
AudioGrabber *AudioGrabber::createAudioGrabber() {
auto *grabber = new AudioGrabber();
return grabber;
}
void AudioGrabber::init() {
m_buffer = static_cast<stereoSample *>(calloc(BUFFER_SIZE, sizeof(stereoSample)));
if (env != nullptr)
m_scale = env->getAsDouble("audio_scale", 1.0);
DBG("SET Audio Scale: %.3f", m_scale)
loudness = { 0.0, 0.0 };
}
void AudioGrabber::calculateRMS(stereoSample *pFrame) {
float squareL = 0, meanL;
float squareR = 0, meanR;
for (int i = 0; i < BUFFER_SIZE; i++) {
squareL += std::pow(pFrame[0].l, 2);
squareR += std::pow(pFrame[0].r, 2);
}
meanL = (squareL / (float) (BUFFER_SIZE));
meanR = (squareR / (float) (BUFFER_SIZE));
loudness = { std::sqrt(meanL), std::sqrt(meanR) };
}
void AudioGrabber::calculatePEAK(stereoSample *pFrame) {
stereoSampleFrame max = { 0, 0 };
for (int i = 0; i < BUFFER_SIZE; i++) {
float left = std::abs(pFrame[0].l);
float right = std::abs(pFrame[0].r);
if (left > max.l)
max.l = left;
if (right > max.r)
max.r = right;
}
loudness = max;
}
stereoSampleFrame AudioGrabber::getLoudness() {
std::unique_lock<std::mutex> lck(m_mtx);
return loudness;
}
bool AudioGrabber::work() {
std::unique_lock<std::mutex> lck(m_mtx);
if (this->read(m_buffer, BUFFER_SIZE)) {
switch (requestMode) {
case ReqMode::FFT:
// FFT get's better results with maybe a bit scaling.. for RMS and PEAK this gets "worse"
for (int i = 0; i < BUFFER_SIZE; ++i) {
m_buffer[i].l *= m_scale;
m_buffer[i].r *= m_scale;
}
fft.process(m_buffer);
break;
case ReqMode::RMS:
calculateRMS(m_buffer);
break;
case ReqMode::PEAK:
calculatePEAK(m_buffer);
break;
default:
fft.process(m_buffer);
calculateRMS(m_buffer);
calculatePEAK(m_buffer);
}
return true;
} else {
DBG("Wait for Data")
return false;
}
}