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14.2 kB
| //#define __DEBUG_SPEECH_PROB___ | |
| // timestamp_t class: stores the start and end (in samples) of a speech segment. | |
| class timestamp_t { | |
| public: | |
| int start; | |
| int end; | |
| timestamp_t(int start = -1, int end = -1) | |
| : start(start), end(end) { } | |
| timestamp_t& operator=(const timestamp_t& a) { | |
| start = a.start; | |
| end = a.end; | |
| return *this; | |
| } | |
| bool operator==(const timestamp_t& a) const { | |
| return (start == a.start && end == a.end); | |
| } | |
| // Returns a formatted string of the timestamp. | |
| std::string c_str() const { | |
| return format("{start:%08d, end:%08d}", start, end); | |
| } | |
| private: | |
| // Helper function for formatting. | |
| std::string format(const char* fmt, ...) const { | |
| char buf[256]; | |
| va_list args; | |
| va_start(args, fmt); | |
| const auto r = std::vsnprintf(buf, sizeof(buf), fmt, args); | |
| va_end(args); | |
| if (r < 0) | |
| return {}; | |
| const size_t len = r; | |
| if (len < sizeof(buf)) | |
| return std::string(buf, len); | |
| std::string s(len, '\0'); | |
| va_start(args, fmt); | |
| std::vsnprintf(s.data(), len + 1, fmt, args); | |
| va_end(args); | |
| return s; | |
| auto vbuf = std::unique_ptr<char[]>(new char[len + 1]); | |
| va_start(args, fmt); | |
| std::vsnprintf(vbuf.get(), len + 1, fmt, args); | |
| va_end(args); | |
| return std::string(vbuf.get(), len); | |
| } | |
| }; | |
| // VadIterator class: uses ONNX Runtime to detect speech segments. | |
| class VadIterator { | |
| private: | |
| // ONNX Runtime resources | |
| Ort::Env env; | |
| Ort::SessionOptions session_options; | |
| std::shared_ptr<Ort::Session> session = nullptr; | |
| Ort::AllocatorWithDefaultOptions allocator; | |
| Ort::MemoryInfo memory_info = Ort::MemoryInfo::CreateCpu(OrtArenaAllocator, OrtMemTypeCPU); | |
| // ----- Context-related additions ----- | |
| const int context_samples = 64; // For 16kHz, 64 samples are added as context. | |
| std::vector<float> _context; // Holds the last 64 samples from the previous chunk (initialized to zero). | |
| // Original window size (e.g., 32ms corresponds to 512 samples) | |
| int window_size_samples; | |
| // Effective window size = window_size_samples + context_samples | |
| int effective_window_size; | |
| // Additional declaration: samples per millisecond | |
| int sr_per_ms; | |
| // ONNX Runtime input/output buffers | |
| std::vector<Ort::Value> ort_inputs; | |
| std::vector<const char*> input_node_names = { "input", "state", "sr" }; | |
| std::vector<float> input; | |
| unsigned int size_state = 2 * 1 * 128; | |
| std::vector<float> _state; | |
| std::vector<int64_t> sr; | |
| int64_t input_node_dims[2] = {}; | |
| const int64_t state_node_dims[3] = { 2, 1, 128 }; | |
| const int64_t sr_node_dims[1] = { 1 }; | |
| std::vector<Ort::Value> ort_outputs; | |
| std::vector<const char*> output_node_names = { "output", "stateN" }; | |
| // Model configuration parameters | |
| int sample_rate; | |
| float threshold; | |
| int min_silence_samples; | |
| int min_silence_samples_at_max_speech; | |
| int min_speech_samples; | |
| float max_speech_samples; | |
| int speech_pad_samples; | |
| int audio_length_samples; | |
| // State management | |
| bool triggered = false; | |
| unsigned int temp_end = 0; | |
| unsigned int current_sample = 0; | |
| int prev_end; | |
| int next_start = 0; | |
| std::vector<timestamp_t> speeches; | |
| timestamp_t current_speech; | |
| // Loads the ONNX model. | |
| void init_onnx_model(const std::wstring& model_path) { | |
| init_engine_threads(1, 1); | |
| session = std::make_shared<Ort::Session>(env, model_path.c_str(), session_options); | |
| } | |
| // Initializes threading settings. | |
| void init_engine_threads(int inter_threads, int intra_threads) { | |
| session_options.SetIntraOpNumThreads(intra_threads); | |
| session_options.SetInterOpNumThreads(inter_threads); | |
| session_options.SetGraphOptimizationLevel(GraphOptimizationLevel::ORT_ENABLE_ALL); | |
| } | |
| // Resets internal state (_state, _context, etc.) | |
| void reset_states() { | |
| std::memset(_state.data(), 0, _state.size() * sizeof(float)); | |
| triggered = false; | |
| temp_end = 0; | |
| current_sample = 0; | |
| prev_end = next_start = 0; | |
| speeches.clear(); | |
| current_speech = timestamp_t(); | |
| std::fill(_context.begin(), _context.end(), 0.0f); | |
| } | |
| // Inference: runs inference on one chunk of input data. | |
| // data_chunk is expected to have window_size_samples samples. | |
| void predict(const std::vector<float>& data_chunk) { | |
| // Build new input: first context_samples from _context, followed by the current chunk (window_size_samples). | |
| std::vector<float> new_data(effective_window_size, 0.0f); | |
| std::copy(_context.begin(), _context.end(), new_data.begin()); | |
| std::copy(data_chunk.begin(), data_chunk.end(), new_data.begin() + context_samples); | |
| input = new_data; | |
| // Create input tensor (input_node_dims[1] is already set to effective_window_size). | |
| Ort::Value input_ort = Ort::Value::CreateTensor<float>( | |
| memory_info, input.data(), input.size(), input_node_dims, 2); | |
| Ort::Value state_ort = Ort::Value::CreateTensor<float>( | |
| memory_info, _state.data(), _state.size(), state_node_dims, 3); | |
| Ort::Value sr_ort = Ort::Value::CreateTensor<int64_t>( | |
| memory_info, sr.data(), sr.size(), sr_node_dims, 1); | |
| ort_inputs.clear(); | |
| ort_inputs.emplace_back(std::move(input_ort)); | |
| ort_inputs.emplace_back(std::move(state_ort)); | |
| ort_inputs.emplace_back(std::move(sr_ort)); | |
| // Run inference. | |
| ort_outputs = session->Run( | |
| Ort::RunOptions{ nullptr }, | |
| input_node_names.data(), ort_inputs.data(), ort_inputs.size(), | |
| output_node_names.data(), output_node_names.size()); | |
| float speech_prob = ort_outputs[0].GetTensorMutableData<float>()[0]; | |
| float* stateN = ort_outputs[1].GetTensorMutableData<float>(); | |
| std::memcpy(_state.data(), stateN, size_state * sizeof(float)); | |
| current_sample += static_cast<unsigned int>(window_size_samples); // Advance by the original window size. | |
| // If speech is detected (probability >= threshold) | |
| if (speech_prob >= threshold) { | |
| float speech = current_sample - window_size_samples; | |
| printf("{ start: %.3f s (%.3f) %08d}\n", 1.0f * speech / sample_rate, speech_prob, current_sample - window_size_samples); | |
| if (temp_end != 0) { | |
| temp_end = 0; | |
| if (next_start < prev_end) | |
| next_start = current_sample - window_size_samples; | |
| } | |
| if (!triggered) { | |
| triggered = true; | |
| current_speech.start = current_sample - window_size_samples; | |
| } | |
| // Update context: copy the last context_samples from new_data. | |
| std::copy(new_data.end() - context_samples, new_data.end(), _context.begin()); | |
| return; | |
| } | |
| // If the speech segment becomes too long. | |
| if (triggered && ((current_sample - current_speech.start) > max_speech_samples)) { | |
| if (prev_end > 0) { | |
| current_speech.end = prev_end; | |
| speeches.push_back(current_speech); | |
| current_speech = timestamp_t(); | |
| if (next_start < prev_end) | |
| triggered = false; | |
| else | |
| current_speech.start = next_start; | |
| prev_end = 0; | |
| next_start = 0; | |
| temp_end = 0; | |
| } | |
| else { | |
| current_speech.end = current_sample; | |
| speeches.push_back(current_speech); | |
| current_speech = timestamp_t(); | |
| prev_end = 0; | |
| next_start = 0; | |
| temp_end = 0; | |
| triggered = false; | |
| } | |
| std::copy(new_data.end() - context_samples, new_data.end(), _context.begin()); | |
| return; | |
| } | |
| if ((speech_prob >= (threshold - 0.15)) && (speech_prob < threshold)) { | |
| // When the speech probability temporarily drops but is still in speech, update context without changing state. | |
| std::copy(new_data.end() - context_samples, new_data.end(), _context.begin()); | |
| return; | |
| } | |
| if (speech_prob < (threshold - 0.15)) { | |
| float speech = current_sample - window_size_samples - speech_pad_samples; | |
| printf("{ end: %.3f s (%.3f) %08d}\n", 1.0f * speech / sample_rate, speech_prob, current_sample - window_size_samples); | |
| if (triggered) { | |
| if (temp_end == 0) | |
| temp_end = current_sample; | |
| if (current_sample - temp_end > min_silence_samples_at_max_speech) | |
| prev_end = temp_end; | |
| if ((current_sample - temp_end) >= min_silence_samples) { | |
| current_speech.end = temp_end; | |
| if (current_speech.end - current_speech.start > min_speech_samples) { | |
| speeches.push_back(current_speech); | |
| current_speech = timestamp_t(); | |
| prev_end = 0; | |
| next_start = 0; | |
| temp_end = 0; | |
| triggered = false; | |
| } | |
| } | |
| } | |
| std::copy(new_data.end() - context_samples, new_data.end(), _context.begin()); | |
| return; | |
| } | |
| } | |
| public: | |
| // Process the entire audio input. | |
| void process(const std::vector<float>& input_wav) { | |
| reset_states(); | |
| audio_length_samples = static_cast<int>(input_wav.size()); | |
| // Process audio in chunks of window_size_samples (e.g., 512 samples) | |
| for (size_t j = 0; j < static_cast<size_t>(audio_length_samples); j += static_cast<size_t>(window_size_samples)) { | |
| if (j + static_cast<size_t>(window_size_samples) > static_cast<size_t>(audio_length_samples)) | |
| break; | |
| std::vector<float> chunk(&input_wav[j], &input_wav[j] + window_size_samples); | |
| predict(chunk); | |
| } | |
| if (current_speech.start >= 0) { | |
| current_speech.end = audio_length_samples; | |
| speeches.push_back(current_speech); | |
| current_speech = timestamp_t(); | |
| prev_end = 0; | |
| next_start = 0; | |
| temp_end = 0; | |
| triggered = false; | |
| } | |
| } | |
| // Returns the detected speech timestamps. | |
| const std::vector<timestamp_t> get_speech_timestamps() const { | |
| return speeches; | |
| } | |
| // Public method to reset the internal state. | |
| void reset() { | |
| reset_states(); | |
| } | |
| public: | |
| // Constructor: sets model path, sample rate, window size (ms), and other parameters. | |
| // The parameters are set to match the Python version. | |
| VadIterator(const std::wstring ModelPath, | |
| int Sample_rate = 16000, int windows_frame_size = 32, | |
| float Threshold = 0.5, int min_silence_duration_ms = 100, | |
| int speech_pad_ms = 30, int min_speech_duration_ms = 250, | |
| float max_speech_duration_s = std::numeric_limits<float>::infinity()) | |
| : sample_rate(Sample_rate), threshold(Threshold), speech_pad_samples(speech_pad_ms), prev_end(0) | |
| { | |
| sr_per_ms = sample_rate / 1000; // e.g., 16000 / 1000 = 16 | |
| window_size_samples = windows_frame_size * sr_per_ms; // e.g., 32ms * 16 = 512 samples | |
| effective_window_size = window_size_samples + context_samples; // e.g., 512 + 64 = 576 samples | |
| input_node_dims[0] = 1; | |
| input_node_dims[1] = effective_window_size; | |
| _state.resize(size_state); | |
| sr.resize(1); | |
| sr[0] = sample_rate; | |
| _context.assign(context_samples, 0.0f); | |
| min_speech_samples = sr_per_ms * min_speech_duration_ms; | |
| max_speech_samples = (sample_rate * max_speech_duration_s - window_size_samples - 2 * speech_pad_samples); | |
| min_silence_samples = sr_per_ms * min_silence_duration_ms; | |
| min_silence_samples_at_max_speech = sr_per_ms * 98; | |
| init_onnx_model(ModelPath); | |
| } | |
| }; | |
| int main() { | |
| // Read the WAV file (expects 16000 Hz, mono, PCM). | |
| wav::WavReader wav_reader("audio/recorder.wav"); // File located in the "audio" folder. | |
| int numSamples = wav_reader.num_samples(); | |
| std::vector<float> input_wav(static_cast<size_t>(numSamples)); | |
| for (size_t i = 0; i < static_cast<size_t>(numSamples); i++) { | |
| input_wav[i] = static_cast<float>(*(wav_reader.data() + i)); | |
| } | |
| // Set the ONNX model path (file located in the "model" folder). | |
| std::wstring model_path = L"model/silero_vad.onnx"; | |
| // Initialize the VadIterator. | |
| VadIterator vad(model_path); | |
| // Process the audio. | |
| vad.process(input_wav); | |
| // Retrieve the speech timestamps (in samples). | |
| std::vector<timestamp_t> stamps = vad.get_speech_timestamps(); | |
| // Convert timestamps to seconds and round to one decimal place (for 16000 Hz). | |
| const float sample_rate_float = 16000.0f; | |
| for (size_t i = 0; i < stamps.size(); i++) { | |
| float start_sec = std::rint((stamps[i].start / sample_rate_float) * 10.0f) / 10.0f; | |
| float end_sec = std::rint((stamps[i].end / sample_rate_float) * 10.0f) / 10.0f; | |
| std::cout << "Speech detected from " | |
| << std::fixed << std::setprecision(1) << start_sec | |
| << " s to " | |
| << std::fixed << std::setprecision(1) << end_sec | |
| << " s" << std::endl; | |
| } | |
| // Optionally, reset the internal state. | |
| vad.reset(); | |
| return 0; | |
| } | |