Files
silero-vad/examples/cpp_libtorch/silero_torch.cc
2024-11-22 06:21:49 +00:00

286 lines
9.4 KiB
C++

//Author : Nathan Lee
//Created On : 2024-11-18
//Description : silero 5.1 system for torch-script(c++).
//Version : 1.0
#include "silero_torch.h"
namespace silero {
VadIterator::VadIterator(const std::string &model_path, float threshold, int sample_rate, int window_size_ms, int speech_pad_ms, int min_silence_duration_ms, int min_speech_duration_ms, int max_duration_merge_ms, bool print_as_samples)
:sample_rate(sample_rate), threshold(threshold), window_size_ms(window_size_ms), speech_pad_ms(speech_pad_ms), min_silence_duration_ms(min_silence_duration_ms), min_speech_duration_ms(min_speech_duration_ms), max_duration_merge_ms(max_duration_merge_ms), print_as_samples(print_as_samples)
{
init_torch_model(model_path);
//init_engine(window_size_ms);
}
VadIterator::~VadIterator(){
}
void VadIterator::SpeechProbs(std::vector<float>& input_wav){
// Set the sample rate (must match the model's expected sample rate)
// Process the waveform in chunks of 512 samples
int num_samples = input_wav.size();
int num_chunks = num_samples / window_size_samples;
int remainder_samples = num_samples % window_size_samples;
total_sample_size += num_samples;
torch::Tensor output;
std::vector<torch::Tensor> chunks;
for (int i = 0; i < num_chunks; i++) {
float* chunk_start = input_wav.data() + i *window_size_samples;
torch::Tensor chunk = torch::from_blob(chunk_start, {1,window_size_samples}, torch::kFloat32);
//std::cout<<"chunk size : "<<chunk.sizes()<<std::endl;
chunks.push_back(chunk);
if(i==num_chunks-1 && remainder_samples>0){//마지막 chunk && 나머지가 존재
int remaining_samples = num_samples - num_chunks * window_size_samples;
//std::cout<<"Remainder size : "<<remaining_samples;
float* chunk_start_remainder = input_wav.data() + num_chunks *window_size_samples;
torch::Tensor remainder_chunk = torch::from_blob(chunk_start_remainder, {1,remaining_samples},
torch::kFloat32);
// Pad the remainder chunk to match window_size_samples
torch::Tensor padded_chunk = torch::cat({remainder_chunk, torch::zeros({1, window_size_samples
- remaining_samples}, torch::kFloat32)}, 1);
//std::cout<<", padded_chunk size : "<<padded_chunk.size(1)<<std::endl;
chunks.push_back(padded_chunk);
}
}
if (!chunks.empty()) {
#ifdef USE_BATCH
torch::Tensor batched_chunks = torch::stack(chunks); // Stack all chunks into a single tensor
//batched_chunks = batched_chunks.squeeze(1);
batched_chunks = torch::cat({batched_chunks.squeeze(1)});
#ifdef USE_GPU
batched_chunks = batched_chunks.to(at::kCUDA); // Move the entire batch to GPU once
#endif
// Prepare input for model
std::vector<torch::jit::IValue> inputs;
inputs.push_back(batched_chunks); // Batch of chunks
inputs.push_back(sample_rate); // Assuming sample_rate is a valid input for the model
// Run inference on the batch
torch::NoGradGuard no_grad;
torch::Tensor output = model.forward(inputs).toTensor();
#ifdef USE_GPU
output = output.to(at::kCPU); // Move the output back to CPU once
#endif
// Collect output probabilities
for (int i = 0; i < chunks.size(); i++) {
float output_f = output[i].item<float>();
outputs_prob.push_back(output_f);
//std::cout << "Chunk " << i << " prob: " << output_f<< "\n";
}
#else
std::vector<torch::Tensor> outputs;
torch::Tensor batched_chunks = torch::stack(chunks);
#ifdef USE_GPU
batched_chunks = batched_chunks.to(at::kCUDA);
#endif
for (int i = 0; i < chunks.size(); i++) {
torch::NoGradGuard no_grad;
std::vector<torch::jit::IValue> inputs;
inputs.push_back(batched_chunks[i]);
inputs.push_back(sample_rate);
torch::Tensor output = model.forward(inputs).toTensor();
outputs.push_back(output);
}
torch::Tensor all_outputs = torch::stack(outputs);
#ifdef USE_GPU
all_outputs = all_outputs.to(at::kCPU);
#endif
for (int i = 0; i < chunks.size(); i++) {
float output_f = all_outputs[i].item<float>();
outputs_prob.push_back(output_f);
}
#endif
}
}
std::vector<SpeechSegment> VadIterator::GetSpeechTimestamps() {
std::vector<SpeechSegment> speeches = DoVad();
#ifdef USE_BATCH
//When you use BATCH inference. You would better use 'mergeSpeeches' function to arrage time stamp.
//It could be better get reasonable output because of distorted probs.
duration_merge_samples = sample_rate * max_duration_merge_ms / 1000;
std::vector<SpeechSegment> speeches_merge = mergeSpeeches(speeches, duration_merge_samples);
if(!print_as_samples){
for (auto& speech : speeches_merge) { //samples to second
speech.start /= sample_rate;
speech.end /= sample_rate;
}
}
return speeches_merge;
#else
if(!print_as_samples){
for (auto& speech : speeches) { //samples to second
speech.start /= sample_rate;
speech.end /= sample_rate;
}
}
return speeches;
#endif
}
void VadIterator::SetVariables(){
init_engine(window_size_ms);
}
void VadIterator::init_engine(int window_size_ms) {
min_silence_samples = sample_rate * min_silence_duration_ms / 1000;
speech_pad_samples = sample_rate * speech_pad_ms / 1000;
window_size_samples = sample_rate / 1000 * window_size_ms;
min_speech_samples = sample_rate * min_speech_duration_ms / 1000;
}
void VadIterator::init_torch_model(const std::string& model_path) {
at::set_num_threads(1);
model = torch::jit::load(model_path);
#ifdef USE_GPU
if (!torch::cuda::is_available()) {
std::cout<<"CUDA is not available! Please check your GPU settings"<<std::endl;
throw std::runtime_error("CUDA is not available!");
model.to(at::Device(at::kCPU));
} else {
std::cout<<"CUDA available! Running on '0'th GPU"<<std::endl;
model.to(at::Device(at::kCUDA, 0)); //select 0'th machine
}
#endif
model.eval();
torch::NoGradGuard no_grad;
std::cout << "Model loaded successfully"<<std::endl;
}
void VadIterator::reset_states() {
triggered = false;
current_sample = 0;
temp_end = 0;
outputs_prob.clear();
model.run_method("reset_states");
total_sample_size = 0;
}
std::vector<SpeechSegment> VadIterator::DoVad() {
std::vector<SpeechSegment> speeches;
for (size_t i = 0; i < outputs_prob.size(); ++i) {
float speech_prob = outputs_prob[i];
//std::cout << speech_prob << std::endl;
//std::cout << "Chunk " << i << " Prob: " << speech_prob << "\n";
//std::cout << speech_prob << " ";
current_sample += window_size_samples;
if (speech_prob >= threshold && temp_end != 0) {
temp_end = 0;
}
if (speech_prob >= threshold && !triggered) {
triggered = true;
SpeechSegment segment;
segment.start = std::max(static_cast<int>(0), current_sample - speech_pad_samples - window_size_samples);
speeches.push_back(segment);
continue;
}
if (speech_prob < threshold - 0.15f && triggered) {
if (temp_end == 0) {
temp_end = current_sample;
}
if (current_sample - temp_end < min_silence_samples) {
continue;
} else {
SpeechSegment& segment = speeches.back();
segment.end = temp_end + speech_pad_samples - window_size_samples;
temp_end = 0;
triggered = false;
}
}
}
if (triggered) { //만약 낮은 확률을 보이다가 마지막프레임 prbos만 딱 확률이 높게 나오면 위에서 triggerd = true 메핑과 동시에 segment start가 돼서 문제가 될것 같은데? start = end 같은값? 후처리가 있으니 문제가 없으려나?
std::cout<<"when last triggered is keep working until last Probs"<<std::endl;
SpeechSegment& segment = speeches.back();
segment.end = total_sample_size; // 현재 샘플을 마지막 구간의 종료 시간으로 설정
triggered = false; // VAD 상태 초기화
}
speeches.erase(
std::remove_if(
speeches.begin(),
speeches.end(),
[this](const SpeechSegment& speech) {
return ((speech.end - this->speech_pad_samples) - (speech.start + this->speech_pad_samples) < min_speech_samples);
//min_speech_samples is 4000samples(0.25sec)
//여기서 포인트!! 계산 할때는 start,end sample에'speech_pad_samples' 사이즈를 추가한후 길이를 측정함.
}
),
speeches.end()
);
//std::cout<<std::endl;
//std::cout<<"outputs_prob.size : "<<outputs_prob.size()<<std::endl;
reset_states();
return speeches;
}
std::vector<SpeechSegment> VadIterator::mergeSpeeches(const std::vector<SpeechSegment>& speeches, int duration_merge_samples) {
std::vector<SpeechSegment> mergedSpeeches;
if (speeches.empty()) {
return mergedSpeeches; // 빈 벡터 반환
}
// 첫 번째 구간으로 초기화
SpeechSegment currentSegment = speeches[0];
for (size_t i = 1; i < speeches.size(); ++i) { //첫번째 start,end 정보 건너뛰기. 그래서 i=1부터
// 두 구간의 차이가 threshold(duration_merge_samples)보다 작은 경우, 합침
if (speeches[i].start - currentSegment.end < duration_merge_samples) {
// 현재 구간의 끝점을 업데이트
currentSegment.end = speeches[i].end;
} else {
// 차이가 threshold(duration_merge_samples) 이상이면 현재 구간을 저장하고 새로운 구간 시작
mergedSpeeches.push_back(currentSegment);
currentSegment = speeches[i];
}
}
// 마지막 구간 추가
mergedSpeeches.push_back(currentSegment);
return mergedSpeeches;
}
}