Merge pull request #277 from yuGAN6/yugan6

Yugan6
This commit is contained in:
Alexander Veysov
2022-12-11 16:27:48 +03:00
committed by GitHub
4 changed files with 505 additions and 2 deletions

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@@ -20,9 +20,9 @@ This repository also includes Number Detector and Language classifier [models](h
<details>
<summary>Real Time Example</summary>
https://user-images.githubusercontent.com/36505480/144874384-95f80f6d-a4f1-42cc-9be7-004c891dd481.mp4
</details>
<br/>
@@ -110,4 +110,6 @@ Please see our [wiki](https://github.com/snakers4/silero-models/wiki) and [tiers
<h2 align="center">Examples and VAD-based Community Apps</h2>
<br/>
- Example of VAD ONNX Runtime model usage in [C++](https://github.com/snakers4/silero-vad/tree/master/examples/cpp)
- Voice activity detection for the [browser](https://github.com/ricky0123/vad) using ONNX Runtime Web

43
examples/cpp/README.md Normal file
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@@ -0,0 +1,43 @@
# Stream example in C++
Here's a simple example of the vad model in c++ onnxruntime.
## Requirements
Code are tested in the environments bellow, feel free to try others.
- WSL2 + Debian-bullseye (docker)
- gcc 12.2.0
- onnxruntime-linux-x64-1.12.1
## Usage
1. Install gcc 12.2.0, or just pull the docker image with `docker pull gcc:12.2.0-bullseye`
2. Install onnxruntime-linux-x64-1.12.1
- Download lib onnxruntime:
`wget https://github.com/microsoft/onnxruntime/releases/download/v1.12.1/onnxruntime-linux-x64-1.12.1.tgz`
- Unzip. Assume the path is `/root/onnxruntime-linux-x64-1.12.1`
3. Modify wav path & Test configs in main function
`wav::WavReader wav_reader("${path_to_your_wav_file}");`
test sample rate, frame per ms, threshold...
4. Build with gcc and run
```bash
# Build
g++ silero-vad-onnx.cpp -I /root/onnxruntime-linux-x64-1.12.1/include/ -L /root/onnxruntime-linux-x64-1.12.1/lib/ -lonnxruntime -Wl,-rpath,/root/onnxruntime-linux-x64-1.12.1/lib/ -o test
# Run
./test
```

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#include <iostream>
#include <vector>
#include <sstream>
#include <cstring>
#include <chrono>
#include "onnxruntime_cxx_api.h"
#include "wav.h"
class VadIterator
{
// OnnxRuntime 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);
public:
void init_engine_threads(int inter_threads, int intra_threads)
{
// The method should be called in each thread/proc in multi-thread/proc work
session_options.SetIntraOpNumThreads(intra_threads);
session_options.SetInterOpNumThreads(inter_threads);
session_options.SetGraphOptimizationLevel(GraphOptimizationLevel::ORT_ENABLE_ALL);
}
void init_onnx_model(const std::string &model_path)
{
// Init threads = 1 for
init_engine_threads(1, 1);
// Load model
session = std::make_shared<Ort::Session>(env, model_path.c_str(), session_options);
}
void reset_states()
{
// Call reset before each audio start
std::memset(_h.data(), 0.0f, _h.size() * sizeof(float));
std::memset(_c.data(), 0.0f, _c.size() * sizeof(float));
triggerd = false;
temp_end = 0;
current_sample = 0;
}
// Call it in predict func. if you prefer raw bytes input.
void bytes_to_float_tensor(const char *pcm_bytes)
{
std::memcpy(input.data(), pcm_bytes, window_size_samples * sizeof(int16_t));
for (int i = 0; i < window_size_samples; i++)
{
input[i] = static_cast<float>(input[i]) / 32768; // int16_t normalized to float
}
}
void predict(const std::vector<float> &data)
{
// bytes_to_float_tensor(data);
// Infer
// Create ort tensors
input.assign(data.begin(), data.end());
Ort::Value input_ort = Ort::Value::CreateTensor<float>(
memory_info, input.data(), input.size(), input_node_dims, 2);
Ort::Value sr_ort = Ort::Value::CreateTensor<int64_t>(
memory_info, sr.data(), sr.size(), sr_node_dims, 1);
Ort::Value h_ort = Ort::Value::CreateTensor<float>(
memory_info, _h.data(), _h.size(), hc_node_dims, 3);
Ort::Value c_ort = Ort::Value::CreateTensor<float>(
memory_info, _c.data(), _c.size(), hc_node_dims, 3);
// Clear and add inputs
ort_inputs.clear();
ort_inputs.emplace_back(std::move(input_ort));
ort_inputs.emplace_back(std::move(sr_ort));
ort_inputs.emplace_back(std::move(h_ort));
ort_inputs.emplace_back(std::move(c_ort));
// Infer
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());
// Output probability & update h,c recursively
float output = ort_outputs[0].GetTensorMutableData<float>()[0];
float *hn = ort_outputs[1].GetTensorMutableData<float>();
std::memcpy(_h.data(), hn, size_hc * sizeof(float));
float *cn = ort_outputs[2].GetTensorMutableData<float>();
std::memcpy(_c.data(), cn, size_hc * sizeof(float));
// Push forward sample index
current_sample += window_size_samples;
// Reset temp_end when > threshold
if ((output >= threshold) && (temp_end != 0))
{
temp_end = 0;
}
// 1) Silence
if ((output < threshold) && (triggerd == false))
{
// printf("{ silence: %.3f s }\n", 1.0 * current_sample / sample_rate);
}
// 2) Speaking
if ((output >= (threshold - 0.15)) && (triggerd == true))
{
// printf("{ speaking_2: %.3f s }\n", 1.0 * current_sample / sample_rate);
}
// 3) Start
if ((output >= threshold) && (triggerd == false))
{
triggerd = true;
speech_start = current_sample - window_size_samples - speech_pad_samples; // minus window_size_samples to get precise start time point.
printf("{ start: %.3f s }\n", 1.0 * speech_start / sample_rate);
}
// 4) End
if ((output < (threshold - 0.15)) && (triggerd == true))
{
if (temp_end != 0)
{
temp_end = current_sample;
}
// a. silence < min_slience_samples, continue speaking
if ((current_sample - temp_end) < min_silence_samples)
{
// printf("{ speaking_4: %.3f s }\n", 1.0 * current_sample / sample_rate);
// printf("");
}
// b. silence >= min_slience_samples, end speaking
else
{
speech_end = current_sample + speech_pad_samples;
temp_end = 0;
triggerd = false;
printf("{ end: %.3f s }\n", 1.0 * speech_end / sample_rate);
}
}
}
private:
// model config
int64_t window_size_samples; // Assign when init, support 256 512 768 for 8k; 512 1024 1536 for 16k.
int sample_rate;
int sr_per_ms; // Assign when init, support 8 or 16
float threshold;
int min_silence_samples; // sr_per_ms * #ms
int speech_pad_samples; // usually a
// model states
bool triggerd = false;
unsigned int speech_start = 0;
unsigned int speech_end = 0;
unsigned int temp_end = 0;
unsigned int current_sample = 0;
// MAX 4294967295 samples / 8sample per ms / 1000 / 60 = 8947 minutes
float output;
// Onnx model
// Inputs
std::vector<Ort::Value> ort_inputs;
std::vector<const char *> input_node_names = {"input", "sr", "h", "c"};
std::vector<float> input;
std::vector<int64_t> sr;
unsigned int size_hc = 2 * 1 * 64; // It's FIXED.
std::vector<float> _h;
std::vector<float> _c;
int64_t input_node_dims[2] = {};
const int64_t sr_node_dims[1] = {1};
const int64_t hc_node_dims[3] = {2, 1, 64};
// Outputs
std::vector<Ort::Value> ort_outputs;
std::vector<const char *> output_node_names = {"output", "hn", "cn"};
public:
// Construction
VadIterator(const std::string ModelPath,
int Sample_rate, int frame_size,
float Threshold, int min_silence_duration_ms, int speech_pad_ms)
{
init_onnx_model(ModelPath);
sample_rate = Sample_rate;
sr_per_ms = sample_rate / 1000;
threshold = Threshold;
min_silence_samples = sr_per_ms * min_silence_duration_ms;
speech_pad_samples = sr_per_ms * speech_pad_ms;
window_size_samples = frame_size * sr_per_ms;
input.resize(window_size_samples);
input_node_dims[0] = 1;
input_node_dims[1] = window_size_samples;
// std::cout << "== Input size" << input.size() << std::endl;
_h.resize(size_hc);
_c.resize(size_hc);
sr.resize(1);
}
};
int main()
{
// Read wav
wav::WavReader wav_reader("./test_for_vad.wav");
std::vector<int16_t> data(wav_reader.num_samples());
std::vector<float> input_wav(wav_reader.num_samples());
for (int i = 0; i < wav_reader.num_samples(); i++)
{
data[i] = static_cast<int16_t>(*(wav_reader.data() + i));
}
for (int i = 0; i < wav_reader.num_samples(); i++)
{
input_wav[i] = static_cast<float>(data[i]) / 32768;
}
// ===== Test configs =====
std::string path = "../files/silero_vad.onnx";
int test_sr = 8000;
int test_frame_ms = 64;
float test_threshold = 0.5f;
int test_min_silence_duration_ms = 0;
int test_speech_pad_ms = 0;
int test_window_samples = test_frame_ms * (test_sr/1000);
VadIterator vad(
path, test_sr, test_frame_ms, test_threshold,
test_min_silence_duration_ms, test_speech_pad_ms);
for (int j = 0; j < wav_reader.num_samples(); j += test_window_samples)
{
// std::cout << "== 4" << std::endl;
std::vector<float> r{&input_wav[0] + j, &input_wav[0] + j + test_window_samples};
auto start = std::chrono::high_resolution_clock::now();
// Predict and print throughout process time
vad.predict(r);
auto end = std::chrono::high_resolution_clock::now();
auto elapsed_time = std::chrono::duration_cast<std::chrono::nanoseconds>(end-start);
// std::cout << "== Elapsed time: " << 1.0*elapsed_time.count()/1000000 << "ms" << " ==" <<std::endl;
}
}

205
examples/cpp/wav.h Normal file
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@@ -0,0 +1,205 @@
// Copyright (c) 2016 Personal (Binbin Zhang)
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
// http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.
#ifndef FRONTEND_WAV_H_
#define FRONTEND_WAV_H_
#include <assert.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <string>
// #include "utils/log.h"
namespace wav {
struct WavHeader {
char riff[4]; // "riff"
unsigned int size;
char wav[4]; // "WAVE"
char fmt[4]; // "fmt "
unsigned int fmt_size;
uint16_t format;
uint16_t channels;
unsigned int sample_rate;
unsigned int bytes_per_second;
uint16_t block_size;
uint16_t bit;
char data[4]; // "data"
unsigned int data_size;
};
class WavReader {
public:
WavReader() : data_(nullptr) {}
explicit WavReader(const std::string& filename) { Open(filename); }
bool Open(const std::string& filename) {
FILE* fp = fopen(filename.c_str(), "rb"); //文件读取
if (NULL == fp) {
std::cout << "Error in read " << filename;
return false;
}
WavHeader header;
fread(&header, 1, sizeof(header), fp);
if (header.fmt_size < 16) {
fprintf(stderr,
"WaveData: expect PCM format data "
"to have fmt chunk of at least size 16.\n");
return false;
} else if (header.fmt_size > 16) {
int offset = 44 - 8 + header.fmt_size - 16;
fseek(fp, offset, SEEK_SET);
fread(header.data, 8, sizeof(char), fp);
}
// check "riff" "WAVE" "fmt " "data"
// Skip any sub-chunks between "fmt" and "data". Usually there will
// be a single "fact" sub chunk, but on Windows there can also be a
// "list" sub chunk.
while (0 != strncmp(header.data, "data", 4)) {
// We will just ignore the data in these chunks.
fseek(fp, header.data_size, SEEK_CUR);
// read next sub chunk
fread(header.data, 8, sizeof(char), fp);
}
num_channel_ = header.channels;
sample_rate_ = header.sample_rate;
bits_per_sample_ = header.bit;
int num_data = header.data_size / (bits_per_sample_ / 8);
data_ = new float[num_data]; // Create 1-dim array
num_samples_ = num_data / num_channel_;
for (int i = 0; i < num_data; ++i) {
switch (bits_per_sample_) {
case 8: {
char sample;
fread(&sample, 1, sizeof(char), fp);
data_[i] = static_cast<float>(sample);
break;
}
case 16: {
int16_t sample;
fread(&sample, 1, sizeof(int16_t), fp);
// std::cout << sample;
data_[i] = static_cast<float>(sample);
// std::cout << data_[i];
break;
}
case 32: {
int sample;
fread(&sample, 1, sizeof(int), fp);
data_[i] = static_cast<float>(sample);
break;
}
default:
fprintf(stderr, "unsupported quantization bits");
exit(1);
}
}
fclose(fp);
return true;
}
int num_channel() const { return num_channel_; }
int sample_rate() const { return sample_rate_; }
int bits_per_sample() const { return bits_per_sample_; }
int num_samples() const { return num_samples_; }
~WavReader() {
delete[] data_;
}
const float* data() const { return data_; }
private:
int num_channel_;
int sample_rate_;
int bits_per_sample_;
int num_samples_; // sample points per channel
float* data_;
};
class WavWriter {
public:
WavWriter(const float* data, int num_samples, int num_channel,
int sample_rate, int bits_per_sample)
: data_(data),
num_samples_(num_samples),
num_channel_(num_channel),
sample_rate_(sample_rate),
bits_per_sample_(bits_per_sample) {}
void Write(const std::string& filename) {
FILE* fp = fopen(filename.c_str(), "w");
// init char 'riff' 'WAVE' 'fmt ' 'data'
WavHeader header;
char wav_header[44] = {0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57,
0x41, 0x56, 0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00,
0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00};
memcpy(&header, wav_header, sizeof(header));
header.channels = num_channel_;
header.bit = bits_per_sample_;
header.sample_rate = sample_rate_;
header.data_size = num_samples_ * num_channel_ * (bits_per_sample_ / 8);
header.size = sizeof(header) - 8 + header.data_size;
header.bytes_per_second =
sample_rate_ * num_channel_ * (bits_per_sample_ / 8);
header.block_size = num_channel_ * (bits_per_sample_ / 8);
fwrite(&header, 1, sizeof(header), fp);
for (int i = 0; i < num_samples_; ++i) {
for (int j = 0; j < num_channel_; ++j) {
switch (bits_per_sample_) {
case 8: {
char sample = static_cast<char>(data_[i * num_channel_ + j]);
fwrite(&sample, 1, sizeof(sample), fp);
break;
}
case 16: {
int16_t sample = static_cast<int16_t>(data_[i * num_channel_ + j]);
fwrite(&sample, 1, sizeof(sample), fp);
break;
}
case 32: {
int sample = static_cast<int>(data_[i * num_channel_ + j]);
fwrite(&sample, 1, sizeof(sample), fp);
break;
}
}
}
}
fclose(fp);
}
private:
const float* data_;
int num_samples_; // total float points in data_
int num_channel_;
int sample_rate_;
int bits_per_sample_;
};
} // namespace wenet
#endif // FRONTEND_WAV_H_