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add brpc to use paraformer (#3149)

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* add brpc to use paraformer
speechx
吴伟鹏 2 years ago committed by GitHub
parent ce4af0e765
commit 8340ad5472
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22
runtime/engine/asr/server/brpc/BCLOUD
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#edit-mode: -*- python -*-
#coding:gbk
WORKROOT('../../../')
GCC('gcc82')
CPPFLAGS('-g -DNDEBUG -pipe -W -Wall -Werror -fPIC -Wno-deprecated -Wno-unused-parameter -fno-omit-frame-pointer -D__const__= -std=c++11 -D__STDC_FORMAT_MACROS -DBAIDU_RPC_ENABLE_CPU_PROFILER -DBAIDU_RPC_ENABLE_HEAP_PROFILER')
LDFLAGS('-lpthread -pthread -lrt -ldl -lz')
CONFIGS('baidu/base/baidu-rpc@stable')
CONFIGS('baidu/third-party/openssl@openssl_V1.0.2.10_GCC820_6U3_K2_GEN_PD_BL@git_tag')
CONFIGS('baidu/third-party/tcmalloc@tcmalloc_V2.7.0.7_GCC820_4U3_K3_GEN_PD_BL@git_tag')
CONFIGS("baidu/third-party/protobuf@protobuf_V2.6.1.1_GCC820_4U3_K3_GEN_PD_BL@git_tag")
CONFIGS('baidu/base/ullib@stable')
HEADERS(GLOB_GEN_SRCS('./proto/*.pb.h'), '$INC')
PROTOC(ENV.WorkRoot() + '/third-64/protobuf/bin/protoc')
PROTOFLAGS('-I../../../third-64/protobuf/include')
PROTOFLAGS('--proto_path=.', '--cpp_out=.')
Directory("paraformerCPP")

4
runtime/engine/asr/server/brpc/README.md
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## Extra Description
this codes are using BCLOUD to compile, and this method does not satisfy the rule of repos.

232
runtime/engine/asr/server/brpc/paraformerCPP/Audio.cpp
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#include <math.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <webrtc_vad.h>
#include "ComDefine.h"
#include "Audio.h"
using namespace std;
class AudioWindow {
private:
int *window;
int in_idx;
int out_idx;
int sum;
int window_size = 0;
public:
AudioWindow(int window_size) : window_size(window_size)
{
window = (int *)calloc(sizeof(int), window_size + 1);
in_idx = 0;
out_idx = 1;
sum = 0;
};
~AudioWindow(){
free(window);
};
int put(int val)
{
sum = sum + val - window[out_idx];
window[in_idx] = val;
in_idx = in_idx == window_size ? 0 : in_idx + 1;
out_idx = out_idx == window_size ? 0 : out_idx + 1;
return sum;
};
};
AudioFrame::AudioFrame(){};
AudioFrame::AudioFrame(int len) : len(len)
{
start = 0;
};
AudioFrame::~AudioFrame(){};
int AudioFrame::set_start(int val)
{
start = val < 0 ? 0 : val;
return start;
};
int AudioFrame::set_end(int val, int max_len)
{
float num_samples = val - start;
float frame_length = 400;
float frame_shift = 160;
float num_new_samples =
ceil((num_samples - frame_length) / frame_shift) * frame_shift + frame_length;
end = start + num_new_samples;
len = (int)num_new_samples;
if (end > max_len){
printf("frame end > max_len!!!!!!!\n");
}
return end;
};
int AudioFrame::get_start()
{
return start;
};
int AudioFrame::get_len()
{
return len;
};
int AudioFrame::disp()
{
printf("not imp!!!!\n");
return 0;
};
Audio::Audio(int data_type) : data_type(data_type)
{
speech_buff = NULL;
align_size = 1360;
}
Audio::Audio(int data_type, int size) : data_type(data_type)
{
speech_buff = NULL;
align_size = (float)size;
}
Audio::~Audio()
{
if (speech_buff != NULL) {
free(speech_buff);
speech_data.clear();
}
}
void Audio::disp()
{
printf("Audio time is %f s. len is %d\n", (float)speech_len / 16000,
speech_len);
}
void Audio::loadwavfrommem(AudioFile<float>audio)
{
if (speech_buff != NULL) {
free(speech_buff);
speech_data.clear();
}
int wav_length = audio.getNumSamplesPerChannel();
int channelNum = audio.getNumChannels();
speech_len = wav_length * channelNum;
printf("wav_length:%d, channelNum: %d", wav_length, channelNum);
speech_align_len = (int)(ceil((float)speech_len / align_size) * align_size);
speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_align_len);
memset(speech_buff, 0, sizeof(int16_t) * speech_align_len);
for (int i = 0; i < wav_length; i++)
{
for (int channel = 0; channel < channelNum; channel++)
{
speech_buff[i * channelNum + channel] = (int16_t)(audio.samples[channel][i] * 32768);
}
}
for (int i = 0; i < speech_len; i++) {
float temp = (float)speech_buff[i];
speech_data.emplace_back(temp);
}
AudioFrame *frame = new AudioFrame(speech_len);
frame_queue.push(frame);
}
int Audio::fetch(vector<float> &dout, int &len, int &flag)
{
if (frame_queue.size() > 0) {
AudioFrame *frame = frame_queue.front();
frame_queue.pop();
len = frame->get_len();
int speech_len = speech_data.size();
auto last = min(speech_len, frame->get_start() + len);
dout.insert(dout.begin(), speech_data.begin() + frame->get_start(), speech_data.begin() + last);
delete frame;
flag = S_END;
return 1;
} else {
return 0;
}
}
#define UNTRIGGERED 0
#define TRIGGERED 1
#define SPEECH_LEN_5S (16000 * 5)
#define SPEECH_LEN_10S (16000 * 10)
#define SPEECH_LEN_15S (16000 * 15)
#define SPEECH_LEN_20S (16000 * 20)
#define SPEECH_LEN_30S (16000 * 30)
#define SPEECH_LEN_60S (16000 * 60)
void Audio::split()
{
VadInst *handle = WebRtcVad_Create();
WebRtcVad_Init(handle);
WebRtcVad_set_mode(handle, 2);
int window_size = 10;
AudioWindow audiowindow(window_size);
int status = UNTRIGGERED;
int offset = 0;
int fs = 16000;
int step = 160;
AudioFrame *frame;
frame = frame_queue.front();
frame_queue.pop();
delete frame;
while (offset < speech_len - step) {
int n = WebRtcVad_Process(handle, fs, speech_buff + offset, step);
if (status == UNTRIGGERED && audiowindow.put(n) >= window_size - 1) {
frame = new AudioFrame();
int start = offset - step * (window_size - 1);
frame->set_start(start);
status = TRIGGERED;
} else if (status == TRIGGERED) {
int win_weight = audiowindow.put(n);
int voice_len = (offset - frame->get_start());
int gap = 0;
if (voice_len < SPEECH_LEN_5S) {
offset += step;
continue;
} else if (voice_len < SPEECH_LEN_10S) {
gap = 1;
} else if (voice_len < SPEECH_LEN_20S) {
gap = window_size / 5;
} else {
gap = window_size - 1;
}
if (win_weight < gap || voice_len >= SPEECH_LEN_15S) {
status = UNTRIGGERED;
offset = frame->set_end(offset, speech_align_len);
frame_queue.push(frame);
frame = NULL;
}
}
offset += step;
}
if (frame != NULL) {
frame->set_end(speech_len, speech_align_len);
frame_queue.push(frame);
frame = NULL;
}
WebRtcVad_Free(handle);
}

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runtime/engine/asr/server/brpc/paraformerCPP/Audio.h
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#ifndef AUDIO_H
#define AUDIO_H
#include <queue>
#include <stdint.h>
#include <vector>
#include <AudioFile.h>
using namespace std;
class AudioFrame {
private:
int start;
int end;
int len;
public:
AudioFrame();
AudioFrame(int len);
~AudioFrame();
int set_start(int val);
int set_end(int val, int max_len);
int get_start();
int get_len();
int disp();
};
class Audio {
private:
vector<float> speech_data;
int16_t *speech_buff;
int speech_len;
int speech_align_len;
int16_t sample_rate;
int offset;
float align_size;
int data_type;
queue<AudioFrame *> frame_queue;
public:
vector<float> speech_vec;
Audio(int data_type);
Audio(int data_type, int size);
~Audio();
void disp();
void loadwav(const char *filename);
void loadwavfrommem(AudioFile<float>audio);
int fetch_chunck(float *&dout, int len);
int fetch(vector<float> &dout, int &len, int &flag);
void padding();
void split();
};
#endif

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runtime/engine/asr/server/brpc/paraformerCPP/AudioFile.h
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runtime/engine/asr/server/brpc/paraformerCPP/ComDefine.h
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#ifndef COMDEFINE_H
#define COMDEFINE_H
#define S_BEGIN 0
#define S_MIDDLE 1
#define S_END 2
#define S_ALL 3
#define S_ERR 4
#endif

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runtime/engine/asr/server/brpc/paraformerCPP/Vocab.cpp
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#include "Vocab.h"
#include <fstream>
#include <iostream>
#include <list>
#include <sstream>
#include <string>
using namespace std;
Vocab::Vocab(const char *filename)
{
ifstream in(filename);
string line;
if (in) // 有该文件
{
while (getline(in, line)) // line中不包括每行的换行符
{
vocab.push_back(line);
}
// cout << vocab[1719] << endl;
}
// else // 没有该文件
//{
// cout << "no such file" << endl;
// }
}
Vocab::~Vocab()
{
}
string Vocab::vector2string(vector<int> in)
{
stringstream ss;
for (auto it = in.begin(); it != in.end(); it++) {
ss << vocab[*it];
}
return ss.str();
}
int str2int(string str)
{
const char *ch_array = str.c_str();
if (((ch_array[0] & 0xf0) != 0xe0) || ((ch_array[1] & 0xc0) != 0x80) ||
((ch_array[2] & 0xc0) != 0x80))
return 0;
int val = ((ch_array[0] & 0x0f) << 12) | ((ch_array[1] & 0x3f) << 6) |
(ch_array[2] & 0x3f);
return val;
}
bool Vocab::isChinese(string ch)
{
if (ch.size() != 3) {
return false;
}
int unicode = str2int(ch);
if (unicode >= 19968 && unicode <= 40959) {
return true;
}
return false;
}
string Vocab::vector2stringV2(vector<int> in)
{
int i;
list<string> words;
int is_pre_english = false;
int pre_english_len = 0;
int is_combining = false;
string combine = "";
for (auto it = in.begin(); it != in.end(); it++) {
string word = vocab[*it];
// step1 space character skips
if (word == "<s>" || word == "</s>" || word == "<unk>")
continue;
// step2 combie phoneme to full word
{
int sub_word = !(word.find("@@") == string::npos);
// process word start and middle part
if (sub_word) {
combine += word.erase(word.length() - 2);
is_combining = true;
continue;
}
// process word end part
else if (is_combining) {
combine += word;
is_combining = false;
word = combine;
combine = "";
}
}
// step3 process english word deal with space , turn abbreviation to upper case
{
// input word is chinese, not need process
if (isChinese(word)) {
words.push_back(word);
is_pre_english = false;
}
// input word is english word
else {
// pre word is chinese
if (!is_pre_english) {
word[0] = word[0] - 32;
words.push_back(word);
pre_english_len = word.size();
}
// pre word is english word
else {
// single letter turn to upper case
if (word.size() == 1) {
word[0] = word[0] - 32;
}
if (pre_english_len > 1) {
words.push_back(" ");
words.push_back(word);
pre_english_len = word.size();
}
else {
if (word.size() > 1) {
words.push_back(" ");
}
words.push_back(word);
pre_english_len = word.size();
}
}
is_pre_english = true;
}
}
}
// for (auto it = words.begin(); it != words.end(); it++) {
// cout << *it << endl;
// }
stringstream ss;
for (auto it = words.begin(); it != words.end(); it++) {
ss << *it;
}
return ss.str();
}
string Vocab::vector2stringV3(string in)
{
int i;
list<string> words;
words.push_back(in.c_str());
int is_pre_english = false;
int pre_english_len = 0;
int is_combining = false;
string combine = "";
for (auto it = in.begin(); it != in.end(); it++) {
string word = vocab[*it];
// step1 space character skips
if (word == "<s>" || word == "</s>" || word == "<unk>")
continue;
// step2 combie phoneme to full word
{
int sub_word = !(word.find("@@") == string::npos);
// process word start and middle part
if (sub_word) {
combine += word.erase(word.length() - 2);
is_combining = true;
continue;
}
// process word end part
else if (is_combining) {
combine += word;
is_combining = false;
word = combine;
combine = "";
}
}
// step3 process english word deal with space , turn abbreviation to upper case
{
// input word is chinese, not need process
if (isChinese(word)) {
words.push_back(word);
is_pre_english = false;
}
// input word is english word
else {
// pre word is chinese
if (!is_pre_english) {
word[0] = word[0] - 32;
words.push_back(word);
pre_english_len = word.size();
}
// pre word is english word
else {
// single letter turn to upper case
if (word.size() == 1) {
word[0] = word[0] - 32;
}
if (pre_english_len > 1) {
words.push_back(" ");
words.push_back(word);
pre_english_len = word.size();
}
else {
if (word.size() > 1) {
words.push_back(" ");
}
words.push_back(word);
pre_english_len = word.size();
}
}
is_pre_english = true;
}
}
}
// for (auto it = words.begin(); it != words.end(); it++) {
// cout << *it << endl;
// }
stringstream ss;
for (auto it = words.begin(); it != words.end(); it++) {
ss << *it;
}
return ss.str();
}
int Vocab::size()
{
return vocab.size();
}

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runtime/engine/asr/server/brpc/paraformerCPP/Vocab.h
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#ifndef VOCAB_H
#define VOCAB_H
#include <stdint.h>
#include <string>
#include <vector>
using namespace std;
class Vocab {
private:
vector<string> vocab;
bool isChinese(string ch);
bool isEnglish(string ch);
public:
Vocab(const char *filename);
~Vocab();
int size();
string vector2string(vector<int> in);
string vector2stringV2(vector<int> in);
string vector2stringV2(string in);
};
#endif

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runtime/engine/asr/server/brpc/paraformerCPP/asr.conf
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# comlog related params
##########################################
# 进程名
COMLOG_PROCNAME : asr-service
#日志等级
#[默认配置(uint)]
COMLOG_LEVEL : 4
#设备数目
#[默认配置(uint), COMLOG_DEVICE_NUM : 2]
COMLOG_DEVICE_NUM : 2
#设备0 名
#[默认配置(字符串), COMLOG_DEVICE0 : FILE]
COMLOG_DEVICE0 : FILE
#设备0 名
#[默认配置(字符串), COMLOG_DEVICE1 : TTY]
COMLOG_DEVICE1 : TTY
#设备类型, ULLOG
#[默认配置(字符串)]
FILE_TYPE : ULLOG
FILE_SIZE : 2048
FILE_SPLITE_TYPE : DATECUT
FILE_DATA_CRONOCUT : 1
FILE_RESERVED1 : %Y%m%d%H
FILE_QUOTA_DAY : 30
#日志名
#[默认配置(字符串)]
FILE_NAME : asr-service.log
#日志路径
#[默认配置(字符串), FILE_PATH : ./log]
FILE_PATH : ./log
#是否打开这个设备
#[默认配置(uint), FILE_OPEN : 1]
FILE_OPEN : 1
#设备类型, TTY
#[默认配置(字符串), TTY_TYPE : TTY]
TTY_TYPE : TTY
#日志名
#[默认配置(字符串), TTY_NAME : receiver]
TTY_NAME : asr-service
#日志路径
#[默认配置(字符串), TTY_PATH : ./log]
TTY_PATH : ./log
#是否打开这个设备
#[默认配置(uint), TTY_OPEN : 1]
TTY_OPEN : 1

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runtime/engine/asr/server/brpc/paraformerCPP/client.cc
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// Copyright (c) 2023 PaddlePaddle Authors. All Rights Reserved.
//
// 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.
#include "recognizer.h"
#include <iostream>
#include <string>
using std::vector;
int main(int argc, char* argv[]) {
std::string model_file = argv[1];
std::string word_symbol_file = argv[2];
std::string wav_audio = argv[3];
InitRecognizer(model_file, word_symbol_file);
int idx = AddRecognizerInstance(); // idx == 0
for (int i = 0; i < 50; i++){
AcceptWav(wav_audio, idx);
std::string result = GetResult(idx);
std::cout << "idx:" << idx << "result :" << result << std::endl;
Reset(idx);
}
}

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runtime/engine/asr/server/brpc/paraformerCPP/main.cc
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// Copyright (c) 2023 PaddlePaddle Authors. All Rights Reserved.
//
// 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.
#include "recognizer.h"
#include <iostream>
#include <string>
using std::vector;
int main(int argc, char* argv[]) {
std::string model_file = "model.onnx";
std::string word_symbol_file = "words.txt";
std::string wav_audio = argv[1];
InitRecognizer(model_file, word_symbol_file);
AcceptWav(wav_audio);
std::string result = GetResult();
std::cout << "result :" << result << std::endl;
Reset();
}

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runtime/engine/asr/server/brpc/paraformerCPP/recognizer.h
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// Copyright (c) 2023 PaddlePaddle Authors. All Rights Reserved.
//
// 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.
#pragma once
#include <string>
#include <vector>
bool InitRecognizer(const std::string& model_file,
const std::string& word_symbol_table_file);
int AddRecognizerInstance(); // instance id is from 0 to size - 1
int GetRecognizerInstanceSize(); // return size
void Accept(const std::vector<float>& waves, int instance_id);
void AcceptWav(const std::string wav_file, int instance_id);
std::string GetResult(int instance_id);
void Reset(int instance_id);

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runtime/engine/asr/server/brpc/paraformerCPP/run.sh
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export LD_LIBRARY_PATH=./fdlib/lib:$LD_LIBRARY_PATH
../bin/asrcpuserver --port 8765

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runtime/engine/asr/server/brpc/paraformerCPP/server.cpp
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// Copyright (c) 2014 baidu-rpc authors.
//
// 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.
// A server to receive EchoRequest and send back EchoResponse.
#include <iostream>
#ifndef _WIN32
#include <sys/time.h>
#else
#include <win_func.h>
#endif
#include <gflags/gflags.h>
#include <baas-lib-c/baas.h>
#include <baas-lib-c/giano_mock_helper.h>
#include <base/logging.h>
#include "base/base64.h"
#include <baidu/rpc/server.h>
#include <baidu/rpc/policy/giano_authenticator.h>
#include "echo.pb.h"
#include <typeinfo>
#include <stdexcept>
#include <fstream>
#include <string>
#include <com_log.h>
#include <vector>
#include "rapidjson/document.h"
#include "rapidjson/stringbuffer.h"
#include "rapidjson/writer.h"
#include "recognizer.h"
#include <AudioFile.h>
#include <Audio.h>
#include <cstdlib>
using namespace std;
DEFINE_int32(port, 8857, "TCP Port of this server");
DEFINE_string(modelpath, "/home/bae/sourcecode/paddlespeech_cli", "the path of model");
DEFINE_int32(max_concurrency, 2, "Limit of request processing in parallel");
namespace example {
struct subRes{
int s; // start time
int e; // end time
string t; // text
};
class AudioServiceImpl : public EchoService {
public:
AudioServiceImpl() {};
virtual ~AudioServiceImpl() {};
string to_string(rapidjson::Document& out_doc) {
rapidjson::StringBuffer out_buffer;
rapidjson::Writer<rapidjson::StringBuffer> out_writer(out_buffer);
out_doc.Accept(out_writer);
std::string out_params = out_buffer.GetString();
return out_params;
}
string convertvector(vector<subRes> result){
rapidjson::Document document;
document.SetArray();
rapidjson::Document::AllocatorType& allocator = document.GetAllocator();
for (const auto sub : result) {
rapidjson::Value obj(rapidjson::kObjectType);
obj.AddMember("s", sub.s, allocator);
obj.AddMember("e", sub.e, allocator);
obj.AddMember("t", sub.t.c_str(), allocator);
document.PushBack(obj, allocator);
}
rapidjson::StringBuffer strbuf;
rapidjson::Writer<rapidjson::StringBuffer> writer(strbuf);
document.Accept(writer);
return strbuf.GetString();
}
string convertvectorV2(vector<subRes> result){
rapidjson::Document document;
document.SetObject();
rapidjson::Document::AllocatorType& allocator = document.GetAllocator();
rapidjson::Value ObjectArray(rapidjson::kArrayType);
for (const auto sub : result) {
rapidjson::Value obj(rapidjson::kObjectType);
obj.AddMember("s", sub.s, allocator);
obj.AddMember("e", sub.e, allocator);
obj.AddMember("t", sub.t.c_str(), allocator);
ObjectArray.PushBack(obj, allocator);
}
document.AddMember("AllTrans", ObjectArray, allocator);
rapidjson::StringBuffer strbuf;
rapidjson::Writer<rapidjson::StringBuffer> writer(strbuf);
document.Accept(writer);
return strbuf.GetString();
}
virtual void audiorecognition(google::protobuf::RpcController* cntl_base,
const AudioRequest* request,
AudioResponse* response,
google::protobuf::Closure* done) {
// This object helps you to call done->Run() in RAII style. If you need
// to process the request asynchronously, pass done_guard.release().
baidu::rpc::ClosureGuard done_guard(done);
string decode_audio_buffer;
base::Base64Decode(request->audio(), &decode_audio_buffer);
vector<uint8_t> vec;
vec.assign(decode_audio_buffer.begin(), decode_audio_buffer.end());
AudioFile<float> a;
bool res = a.loadFromMemory(vec);
Audio audi(0);
audi.loadwavfrommem(a);
audi.split();
vector<float> buff;
int len = 0;
int flag = 1;
vector<subRes> results;
int tmp_len = 0;
while (audi.fetch(buff, len, flag) > 0) {
int do_idx = rand() % 2; //random number [0,1)
Accept(buff, do_idx);
std::string subtxt = GetResult(do_idx);
Reset(do_idx);
buff.clear();
int start_time = (int)(tmp_len/16000.0 * 1000);
int end_time = (int)((tmp_len + len)/16000.0 * 1000);
struct subRes subres = {
start_time,
end_time,
subtxt.c_str(),
};
tmp_len += len;
results.push_back(subres);
com_writelog(COMLOG_NOTICE, "using process: %d, start: %d, end: %d, result: %s", do_idx, start_time, end_time, subtxt.c_str());
}
// vector<float> inputAudio;
// for (int i = 0; i < a.getNumSamplesPerChannel(); i++)
// {
// float tempval = 0.0;
// for (int channel = 0; channel < a.getNumChannels(); channel++)
// {
// tempval += a.samples[channel][i] * 32768;
// }
// inputAudio.emplace_back(tempval);
// }
// int do_idx = rand() % 2; //random number [0,1)
// Accept(inputAudio, do_idx);
// std::string result = GetResult(do_idx);
// Reset(do_idx);
// com_writelog(COMLOG_NOTICE, "using process: %d, result: %s", do_idx, result.c_str());
// std::cout << "Result: " << result << std::endl;
response->set_err_no(0);
response->set_err_msg("");
string jsonresult = convertvector(results);
response->set_result(jsonresult);
response->set_cost_time(0);
results.clear();
}
};
} // namespace example
int main(int argc, char* argv[]) {
// Parse gflags. We recommend you to use gflags as well.
google::ParseCommandLineFlags(&argc, &argv, true);
bool flag_auth = false;
// Setup for `GianoAuthenticator'.
std::unique_ptr<baidu::rpc::policy::GianoAuthenticator> auth;
if (flag_auth) {
if (baas::BAAS_Init() != 0) {
LOG(ERROR) << "Fail to init BAAS";
return -1;
}
baas::CredentialVerifier
ver = baas::ServerUtility::CreateCredentialVerifier();
auth.reset(new baidu::rpc::policy::GianoAuthenticator(NULL, &ver));
}
int ret = com_loadlog("./", "asr.conf");
if (ret != 0)
{
fprintf(stderr, "load err\n");
return -1;
}
// 打印日志，线程安全
com_writelog(COMLOG_NOTICE, "server start1");
// Generally you only need one Server.
baidu::rpc::Server server;
// Instance of your service.
example::AudioServiceImpl audio_service_impl;
// Add the service into server. Notice the second parameter, because the
// service is put on stack, we don't want server to delete it, otherwise
// use baidu::rpc::SERVER_OWNS_SERVICE.
if (server.AddService(&audio_service_impl,
baidu::rpc::SERVER_DOESNT_OWN_SERVICE,
"/v1/audiorecognition => audiorecognition") != 0) {
LOG(ERROR) << "Fail to add service";
return -1;
}
InitRecognizer("model.onnx", "words.txt");
for (int i =0 ;i<= 1; i++){
int idx = AddRecognizerInstance();
}
srand((unsigned)time(NULL));
// Start the server.
baidu::rpc::ServerOptions options;
options.idle_timeout_sec = -1;
options.auth = auth.get();
options.max_concurrency = FLAGS_max_concurrency;
if (server.Start(FLAGS_port, &options) != 0) {
LOG(ERROR) << "Fail to start EchoServer";
return -1;
}
// Wait until Ctrl-C is pressed, then Stop() and Join() the server.
server.RunUntilAskedToQuit();
return 0;
}

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc/CMakeLists.txt
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if(WIN32)
add_definitions(-DWEBRTC_WIN)
else()
add_definitions(-DWEBRTC_POSIX)
endif()
include_directories("..")
file(GLOB_RECURSE files "*.c" "rtc_base/checks.cc")
message("${files}")
add_library(webrtcvad ${files})

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
/* Tables for data buffer indexes that are bit reversed and thus need to be
* swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
* operations, while index_7[{1, 3, 5, ...}] are for the right side of the
* operation. Same for index_8.
*/
/* Indexes for the case of stages == 7. */
static const int16_t index_7[112] = {
1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
103, 115, 111, 123
};
/* Indexes for the case of stages == 8. */
static const int16_t index_8[240] = {
1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
203, 211, 207, 243, 215, 235, 223, 251, 239, 247
};
void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
/* For any specific value of stages, we know exactly the indexes that are
* bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
* stages are 7 and 8, so we use tables to save unnecessary iterations and
* calculations for these two cases.
*/
if (stages == 7 || stages == 8) {
int m = 0;
int length = 112;
const int16_t* index = index_7;
if (stages == 8) {
length = 240;
index = index_8;
}
/* Decimation in time. Swap the elements with bit-reversed indexes. */
for (m = 0; m < length; m += 2) {
/* We declare a int32_t* type pointer, to load both the 16-bit real
* and imaginary elements from complex_data in one instruction, reducing
* complexity.
*/
int32_t* complex_data_ptr = (int32_t*)complex_data;
int32_t temp = 0;
temp = complex_data_ptr[index[m]]; /* Real and imaginary */
complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
complex_data_ptr[index[m + 1]] = temp;
}
}
else {
int m = 0, mr = 0, l = 0;
int n = 1 << stages;
int nn = n - 1;
/* Decimation in time - re-order data */
for (m = 1; m <= nn; ++m) {
int32_t* complex_data_ptr = (int32_t*)complex_data;
int32_t temp = 0;
/* Find out indexes that are bit-reversed. */
l = n;
do {
l >>= 1;
} while (l > nn - mr);
mr = (mr & (l - 1)) + l;
if (mr <= m) {
continue;
}
/* Swap the elements with bit-reversed indexes.
* This is similar to the loop in the stages == 7 or 8 cases.
*/
temp = complex_data_ptr[m]; /* Real and imaginary */
complex_data_ptr[m] = complex_data_ptr[mr];
complex_data_ptr[mr] = temp;
}
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the function WebRtcSpl_ComplexFFT().
* The description header can be found in signal_processing_library.h
*
*/
#include "webrtc/common_audio/signal_processing/complex_fft_tables.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/rtc_base/system/arch.h"
#define CFFTSFT 14
#define CFFTRND 1
#define CFFTRND2 16384
#define CIFFTSFT 14
#define CIFFTRND 1
int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
{
int i, j, l, k, istep, n, m;
int16_t wr, wi;
int32_t tr32, ti32, qr32, qi32;
/* The 1024-value is a constant given from the size of kSinTable1024[],
* and should not be changed depending on the input parameter 'stages'
*/
n = 1 << stages;
if (n > 1024)
return -1;
l = 1;
k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
depending on the input parameter 'stages' */
if (mode == 0)
{
// mode==0: Low-complexity and Low-accuracy mode
while (l < n)
{
istep = l << 1;
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = -kSinTable1024[j];
for (i = m; i < n; i += istep)
{
j = i + l;
tr32 = (wr * frfi[2 * j] - wi * frfi[2 * j + 1]) >> 15;
ti32 = (wr * frfi[2 * j + 1] + wi * frfi[2 * j]) >> 15;
qr32 = (int32_t)frfi[2 * i];
qi32 = (int32_t)frfi[2 * i + 1];
frfi[2 * j] = (int16_t)((qr32 - tr32) >> 1);
frfi[2 * j + 1] = (int16_t)((qi32 - ti32) >> 1);
frfi[2 * i] = (int16_t)((qr32 + tr32) >> 1);
frfi[2 * i + 1] = (int16_t)((qi32 + ti32) >> 1);
}
}
--k;
l = istep;
}
} else
{
// mode==1: High-complexity and High-accuracy mode
while (l < n)
{
istep = l << 1;
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = -kSinTable1024[j];
#ifdef WEBRTC_ARCH_ARM_V7
int32_t wri = 0;
__asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
"r"((int32_t)wr), "r"((int32_t)wi));
#endif
for (i = m; i < n; i += istep)
{
j = i + l;
#ifdef WEBRTC_ARCH_ARM_V7
register int32_t frfi_r;
__asm __volatile(
"pkhbt %[frfi_r], %[frfi_even], %[frfi_odd],"
" lsl #16\n\t"
"smlsd %[tr32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
"smladx %[ti32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
:[frfi_r]"=&r"(frfi_r),
[tr32]"=&r"(tr32),
[ti32]"=r"(ti32)
:[frfi_even]"r"((int32_t)frfi[2*j]),
[frfi_odd]"r"((int32_t)frfi[2*j +1]),
[wri]"r"(wri),
[cfftrnd]"r"(CFFTRND));
#else
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CFFTRND;
#endif
tr32 >>= 15 - CFFTSFT;
ti32 >>= 15 - CFFTSFT;
qr32 = ((int32_t)frfi[2 * i]) * (1 << CFFTSFT);
qi32 = ((int32_t)frfi[2 * i + 1]) * (1 << CFFTSFT);
frfi[2 * j] = (int16_t)(
(qr32 - tr32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * j + 1] = (int16_t)(
(qi32 - ti32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * i] = (int16_t)(
(qr32 + tr32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * i + 1] = (int16_t)(
(qi32 + ti32 + CFFTRND2) >> (1 + CFFTSFT));
}
}
--k;
l = istep;
}
}
return 0;
}
int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
{
size_t i, j, l, istep, n, m;
int k, scale, shift;
int16_t wr, wi;
int32_t tr32, ti32, qr32, qi32;
int32_t tmp32, round2;
/* The 1024-value is a constant given from the size of kSinTable1024[],
* and should not be changed depending on the input parameter 'stages'
*/
n = ((size_t)1) << stages;
if (n > 1024)
return -1;
scale = 0;
l = 1;
k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
depending on the input parameter 'stages' */
while (l < n)
{
// variable scaling, depending upon data
shift = 0;
round2 = 8192;
tmp32 = WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
if (tmp32 > 13573)
{
shift++;
scale++;
round2 <<= 1;
}
if (tmp32 > 27146)
{
shift++;
scale++;
round2 <<= 1;
}
istep = l << 1;
if (mode == 0)
{
// mode==0: Low-complexity and Low-accuracy mode
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = kSinTable1024[j];
for (i = m; i < n; i += istep)
{
j = i + l;
tr32 = (wr * frfi[2 * j] - wi * frfi[2 * j + 1]) >> 15;
ti32 = (wr * frfi[2 * j + 1] + wi * frfi[2 * j]) >> 15;
qr32 = (int32_t)frfi[2 * i];
qi32 = (int32_t)frfi[2 * i + 1];
frfi[2 * j] = (int16_t)((qr32 - tr32) >> shift);
frfi[2 * j + 1] = (int16_t)((qi32 - ti32) >> shift);
frfi[2 * i] = (int16_t)((qr32 + tr32) >> shift);
frfi[2 * i + 1] = (int16_t)((qi32 + ti32) >> shift);
}
}
} else
{
// mode==1: High-complexity and High-accuracy mode
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = kSinTable1024[j];
#ifdef WEBRTC_ARCH_ARM_V7
int32_t wri = 0;
__asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
"r"((int32_t)wr), "r"((int32_t)wi));
#endif
for (i = m; i < n; i += istep)
{
j = i + l;
#ifdef WEBRTC_ARCH_ARM_V7
register int32_t frfi_r;
__asm __volatile(
"pkhbt %[frfi_r], %[frfi_even], %[frfi_odd], lsl #16\n\t"
"smlsd %[tr32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
"smladx %[ti32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
:[frfi_r]"=&r"(frfi_r),
[tr32]"=&r"(tr32),
[ti32]"=r"(ti32)
:[frfi_even]"r"((int32_t)frfi[2*j]),
[frfi_odd]"r"((int32_t)frfi[2*j +1]),
[wri]"r"(wri),
[cifftrnd]"r"(CIFFTRND)
);
#else
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CIFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CIFFTRND;
#endif
tr32 >>= 15 - CIFFTSFT;
ti32 >>= 15 - CIFFTSFT;
qr32 = ((int32_t)frfi[2 * i]) * (1 << CIFFTSFT);
qi32 = ((int32_t)frfi[2 * i + 1]) * (1 << CIFFTSFT);
frfi[2 * j] = (int16_t)(
(qr32 - tr32 + round2) >> (shift + CIFFTSFT));
frfi[2 * j + 1] = (int16_t)(
(qi32 - ti32 + round2) >> (shift + CIFFTSFT));
frfi[2 * i] = (int16_t)(
(qr32 + tr32 + round2) >> (shift + CIFFTSFT));
frfi[2 * i + 1] = (int16_t)(
(qi32 + ti32 + round2) >> (shift + CIFFTSFT));
}
}
}
--k;
l = istep;
}
return scale;
}

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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
#include <stdint.h>
static const int16_t kSinTable1024[] = {
0, 201, 402, 603, 804, 1005, 1206, 1406, 1607,
1808, 2009, 2209, 2410, 2610, 2811, 3011, 3211, 3411,
3611, 3811, 4011, 4210, 4409, 4608, 4807, 5006, 5205,
5403, 5601, 5799, 5997, 6195, 6392, 6589, 6786, 6982,
7179, 7375, 7571, 7766, 7961, 8156, 8351, 8545, 8739,
8932, 9126, 9319, 9511, 9703, 9895, 10087, 10278, 10469,
10659, 10849, 11038, 11227, 11416, 11604, 11792, 11980, 12166,
12353, 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268, 15446,
15623, 15799, 15975, 16150, 16325, 16499, 16672, 16845, 17017,
17189, 17360, 17530, 17699, 17868, 18036, 18204, 18371, 18537,
18702, 18867, 19031, 19194, 19357, 19519, 19680, 19840, 20000,
20159, 20317, 20474, 20631, 20787, 20942, 21096, 21249, 21402,
21554, 21705, 21855, 22004, 22153, 22301, 22448, 22594, 22739,
22883, 23027, 23169, 23311, 23452, 23592, 23731, 23869, 24006,
24143, 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198, 26318,
26437, 26556, 26673, 26789, 26905, 27019, 27132, 27244, 27355,
27466, 27575, 27683, 27790, 27896, 28001, 28105, 28208, 28309,
28410, 28510, 28608, 28706, 28802, 28897, 28992, 29085, 29177,
29268, 29358, 29446, 29534, 29621, 29706, 29790, 29873, 29955,
30036, 30116, 30195, 30272, 30349, 30424, 30498, 30571, 30643,
30713, 30783, 30851, 30918, 30984, 31049, 31113, 31175, 31236,
31297, 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097, 32137,
32176, 32213, 32249, 32284, 32318, 32350, 32382, 32412, 32441,
32468, 32495, 32520, 32544, 32567, 32588, 32609, 32628, 32646,
32662, 32678, 32692, 32705, 32717, 32727, 32736, 32744, 32751,
32757, 32761, 32764, 32766, 32767, 32766, 32764, 32761, 32757,
32751, 32744, 32736, 32727, 32717, 32705, 32692, 32678, 32662,
32646, 32628, 32609, 32588, 32567, 32544, 32520, 32495, 32468,
32441, 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833, 31785,
31735, 31684, 31633, 31580, 31525, 31470, 31413, 31356, 31297,
31236, 31175, 31113, 31049, 30984, 30918, 30851, 30783, 30713,
30643, 30571, 30498, 30424, 30349, 30272, 30195, 30116, 30036,
29955, 29873, 29790, 29706, 29621, 29534, 29446, 29358, 29268,
29177, 29085, 28992, 28897, 28802, 28706, 28608, 28510, 28410,
28309, 28208, 28105, 28001, 27896, 27790, 27683, 27575, 27466,
27355, 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456, 25329,
25201, 25072, 24942, 24811, 24679, 24546, 24413, 24278, 24143,
24006, 23869, 23731, 23592, 23452, 23311, 23169, 23027, 22883,
22739, 22594, 22448, 22301, 22153, 22004, 21855, 21705, 21554,
21402, 21249, 21096, 20942, 20787, 20631, 20474, 20317, 20159,
20000, 19840, 19680, 19519, 19357, 19194, 19031, 18867, 18702,
18537, 18371, 18204, 18036, 17868, 17699, 17530, 17360, 17189,
17017, 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191, 14009,
13827, 13645, 13462, 13278, 13094, 12909, 12724, 12539, 12353,
12166, 11980, 11792, 11604, 11416, 11227, 11038, 10849, 10659,
10469, 10278, 10087, 9895, 9703, 9511, 9319, 9126, 8932,
8739, 8545, 8351, 8156, 7961, 7766, 7571, 7375, 7179,
6982, 6786, 6589, 6392, 6195, 5997, 5799, 5601, 5403,
5205, 5006, 4807, 4608, 4409, 4210, 4011, 3811, 3611,
3411, 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
1607, 1406, 1206, 1005, 804, 603, 402, 201, 0,
-201, -402, -603, -804, -1005, -1206, -1406, -1607, -1808,
-2009, -2209, -2410, -2610, -2811, -3011, -3211, -3411, -3611,
-3811, -4011, -4210, -4409, -4608, -4807, -5006, -5205, -5403,
-5601, -5799, -5997, -6195, -6392, -6589, -6786, -6982, -7179,
-7375, -7571, -7766, -7961, -8156, -8351, -8545, -8739, -8932,
-9126, -9319, -9511, -9703, -9895, -10087, -10278, -10469, -10659,
-10849, -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
-12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827, -14009,
-14191, -14372, -14552, -14732, -14911, -15090, -15268, -15446, -15623,
-15799, -15975, -16150, -16325, -16499, -16672, -16845, -17017, -17189,
-17360, -17530, -17699, -17868, -18036, -18204, -18371, -18537, -18702,
-18867, -19031, -19194, -19357, -19519, -19680, -19840, -20000, -20159,
-20317, -20474, -20631, -20787, -20942, -21096, -21249, -21402, -21554,
-21705, -21855, -22004, -22153, -22301, -22448, -22594, -22739, -22883,
-23027, -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
-24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201, -25329,
-25456, -25582, -25707, -25831, -25954, -26077, -26198, -26318, -26437,
-26556, -26673, -26789, -26905, -27019, -27132, -27244, -27355, -27466,
-27575, -27683, -27790, -27896, -28001, -28105, -28208, -28309, -28410,
-28510, -28608, -28706, -28802, -28897, -28992, -29085, -29177, -29268,
-29358, -29446, -29534, -29621, -29706, -29790, -29873, -29955, -30036,
-30116, -30195, -30272, -30349, -30424, -30498, -30571, -30643, -30713,
-30783, -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
-31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735, -31785,
-31833, -31880, -31926, -31970, -32014, -32056, -32097, -32137, -32176,
-32213, -32249, -32284, -32318, -32350, -32382, -32412, -32441, -32468,
-32495, -32520, -32544, -32567, -32588, -32609, -32628, -32646, -32662,
-32678, -32692, -32705, -32717, -32727, -32736, -32744, -32751, -32757,
-32761, -32764, -32766, -32767, -32766, -32764, -32761, -32757, -32751,
-32744, -32736, -32727, -32717, -32705, -32692, -32678, -32662, -32646,
-32628, -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
-32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176, -32137,
-32097, -32056, -32014, -31970, -31926, -31880, -31833, -31785, -31735,
-31684, -31633, -31580, -31525, -31470, -31413, -31356, -31297, -31236,
-31175, -31113, -31049, -30984, -30918, -30851, -30783, -30713, -30643,
-30571, -30498, -30424, -30349, -30272, -30195, -30116, -30036, -29955,
-29873, -29790, -29706, -29621, -29534, -29446, -29358, -29268, -29177,
-29085, -28992, -28897, -28802, -28706, -28608, -28510, -28410, -28309,
-28208, -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
-27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437, -26318,
-26198, -26077, -25954, -25831, -25707, -25582, -25456, -25329, -25201,
-25072, -24942, -24811, -24679, -24546, -24413, -24278, -24143, -24006,
-23869, -23731, -23592, -23452, -23311, -23169, -23027, -22883, -22739,
-22594, -22448, -22301, -22153, -22004, -21855, -21705, -21554, -21402,
-21249, -21096, -20942, -20787, -20631, -20474, -20317, -20159, -20000,
-19840, -19680, -19519, -19357, -19194, -19031, -18867, -18702, -18537,
-18371, -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
-16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623, -15446,
-15268, -15090, -14911, -14732, -14552, -14372, -14191, -14009, -13827,
-13645, -13462, -13278, -13094, -12909, -12724, -12539, -12353, -12166,
-11980, -11792, -11604, -11416, -11227, -11038, -10849, -10659, -10469,
-10278, -10087, -9895, -9703, -9511, -9319, -9126, -8932, -8739,
-8545, -8351, -8156, -7961, -7766, -7571, -7375, -7179, -6982,
-6786, -6589, -6392, -6195, -5997, -5799, -5601, -5403, -5205,
-5006, -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
-3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808, -1607,
-1406, -1206, -1005, -804, -603, -402, -201};
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_

30
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/cross_correlation.c
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
/* C version of WebRtcSpl_CrossCorrelation() for generic platforms. */
void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
const int16_t* seq1,
const int16_t* seq2,
size_t dim_seq,
size_t dim_cross_correlation,
int right_shifts,
int step_seq2) {
size_t i = 0, j = 0;
for (i = 0; i < dim_cross_correlation; i++) {
int32_t corr = 0;
for (j = 0; j < dim_seq; j++)
corr += (seq1[j] * seq2[j]) >> right_shifts;
seq2 += step_seq2;
*cross_correlation++ = corr;
}
}

141
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/division_operations.c
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains implementations of the divisions
* WebRtcSpl_DivU32U16()
* WebRtcSpl_DivW32W16()
* WebRtcSpl_DivW32W16ResW16()
* WebRtcSpl_DivResultInQ31()
* WebRtcSpl_DivW32HiLow()
*
* The description header can be found in signal_processing_library.h
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/rtc_base/sanitizer.h"
uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (uint32_t)(num / den);
} else
{
return (uint32_t)0xFFFFFFFF;
}
}
int32_t WebRtcSpl_DivW32W16(int32_t num, int16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (int32_t)(num / den);
} else
{
return (int32_t)0x7FFFFFFF;
}
}
int16_t WebRtcSpl_DivW32W16ResW16(int32_t num, int16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (int16_t)(num / den);
} else
{
return (int16_t)0x7FFF;
}
}
int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den)
{
int32_t L_num = num;
int32_t L_den = den;
int32_t div = 0;
int k = 31;
int change_sign = 0;
if (num == 0)
return 0;
if (num < 0)
{
change_sign++;
L_num = -num;
}
if (den < 0)
{
change_sign++;
L_den = -den;
}
while (k--)
{
div <<= 1;
L_num <<= 1;
if (L_num >= L_den)
{
L_num -= L_den;
div++;
}
}
if (change_sign == 1)
{
div = -div;
}
return div;
}
int32_t RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
{
int16_t approx, tmp_hi, tmp_low, num_hi, num_low;
int32_t tmpW32;
approx = (int16_t)WebRtcSpl_DivW32W16((int32_t)0x1FFFFFFF, den_hi);
// result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
// tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
tmpW32 = (den_hi * approx << 1) + ((den_low * approx >> 15) << 1);
// tmpW32 = den * approx
tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
// UBSan: 2147483647 - -2 cannot be represented in type 'int'
// Store tmpW32 in hi and low format
tmp_hi = (int16_t)(tmpW32 >> 16);
tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
// tmpW32 = 1/den in Q29
tmpW32 = (tmp_hi * approx + (tmp_low * approx >> 15)) << 1;
// 1/den in hi and low format
tmp_hi = (int16_t)(tmpW32 >> 16);
tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
// Store num in hi and low format
num_hi = (int16_t)(num >> 16);
num_low = (int16_t)((num - ((int32_t)num_hi << 16)) >> 1);
// num * (1/den) by 32 bit multiplication (result in Q28)
tmpW32 = num_hi * tmp_hi + (num_hi * tmp_low >> 15) +
(num_low * tmp_hi >> 15);
// Put result in Q31 (convert from Q28)
tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
return tmpW32;
}

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/dot_product_with_scale.cc
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/signal_processing/dot_product_with_scale.h"
#include "webrtc/rtc_base/numerics/safe_conversions.h"
int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
const int16_t* vector2,
size_t length,
int scaling) {
int64_t sum = 0;
size_t i = 0;
/* Unroll the loop to improve performance. */
for (i = 0; i + 3 < length; i += 4) {
sum += (vector1[i + 0] * vector2[i + 0]) >> scaling;
sum += (vector1[i + 1] * vector2[i + 1]) >> scaling;
sum += (vector1[i + 2] * vector2[i + 2]) >> scaling;
sum += (vector1[i + 3] * vector2[i + 3]) >> scaling;
}
for (; i < length; i++) {
sum += (vector1[i] * vector2[i]) >> scaling;
}
return rtc::saturated_cast<int32_t>(sum);
}

40
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/dot_product_with_scale.h
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/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
#include <stdint.h>
#include <string.h>
#ifdef __cplusplus
extern "C" {
#endif
// Calculates the dot product between two (int16_t) vectors.
//
// Input:
// - vector1 : Vector 1
// - vector2 : Vector 2
// - vector_length : Number of samples used in the dot product
// - scaling : The number of right bit shifts to apply on each term
// during calculation to avoid overflow, i.e., the
// output will be in Q(-|scaling|)
//
// Return value : The dot product in Q(-scaling)
int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
const int16_t* vector2,
size_t length,
int scaling);
#ifdef __cplusplus
}
#endif // __cplusplus
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_

65
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/downsample_fast.c
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/sanitizer.h"
// TODO(Bjornv): Change the function parameter order to WebRTC code style.
// C version of WebRtcSpl_DownsampleFast() for generic platforms.
int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
size_t data_in_length,
int16_t* data_out,
size_t data_out_length,
const int16_t* __restrict coefficients,
size_t coefficients_length,
int factor,
size_t delay) {
int16_t* const original_data_out = data_out;
size_t i = 0;
size_t j = 0;
int32_t out_s32 = 0;
size_t endpos = delay + factor * (data_out_length - 1) + 1;
// Return error if any of the running conditions doesn't meet.
if (data_out_length == 0 || coefficients_length == 0
|| data_in_length < endpos) {
return -1;
}
rtc_MsanCheckInitialized(coefficients, sizeof(coefficients[0]),
coefficients_length);
for (i = delay; i < endpos; i += factor) {
out_s32 = 2048; // Round value, 0.5 in Q12.
for (j = 0; j < coefficients_length; j++) {
// Negative overflow is permitted here, because this is
// auto-regressive filters, and the state for each batch run is
// stored in the "negative" positions of the output vector.
rtc_MsanCheckInitialized(&data_in[(ptrdiff_t) i - (ptrdiff_t) j],
sizeof(data_in[0]), 1);
// out_s32 is in Q12 domain.
out_s32 += coefficients[j] * data_in[(ptrdiff_t) i - (ptrdiff_t) j];
}
out_s32 >>= 12; // Q0.
// Saturate and store the output.
*data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
}
RTC_DCHECK_EQ(original_data_out + data_out_length, data_out);
rtc_MsanCheckInitialized(original_data_out, sizeof(original_data_out[0]),
data_out_length);
return 0;
}

39
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/energy.c
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the function WebRtcSpl_Energy().
* The description header can be found in signal_processing_library.h
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
int32_t WebRtcSpl_Energy(int16_t* vector,
size_t vector_length,
int* scale_factor)
{
int32_t en = 0;
size_t i;
int scaling =
WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
size_t looptimes = vector_length;
int16_t *vectorptr = vector;
for (i = 0; i < looptimes; i++)
{
en += (*vectorptr * *vectorptr) >> scaling;
vectorptr++;
}
*scale_factor = scaling;
return en;
}

46
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/get_scaling_square.c
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the function WebRtcSpl_GetScalingSquare().
* The description header can be found in signal_processing_library.h
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
size_t in_vector_length,
size_t times)
{
int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
size_t i;
int16_t smax = -1;
int16_t sabs;
int16_t *sptr = in_vector;
int16_t t;
size_t looptimes = in_vector_length;
for (i = looptimes; i > 0; i--)
{
sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
smax = (sabs > smax ? sabs : smax);
}
t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
if (smax == 0)
{
return 0; // Since norm(0) returns 0
} else
{
return (t > nbits) ? 0 : nbits - t;
}
}

96
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/include/real_fft.h
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
#include <stdint.h>
// For ComplexFFT(), the maximum fft order is 10;
// WebRTC APM uses orders of only 7 and 8.
enum { kMaxFFTOrder = 10 };
struct RealFFT;
#ifdef __cplusplus
extern "C" {
#endif
struct RealFFT* WebRtcSpl_CreateRealFFT(int order);
void WebRtcSpl_FreeRealFFT(struct RealFFT* self);
// Compute an FFT for a real-valued signal of length of 2^order,
// where 1 < order <= MAX_FFT_ORDER. Transform length is determined by the
// specification structure, which must be initialized prior to calling the FFT
// function with WebRtcSpl_CreateRealFFT().
// The relationship between the input and output sequences can
// be expressed in terms of the DFT, i.e.:
// x[n] = (2^(-scalefactor)/N) . SUM[k=0,...,N-1] X[k].e^(jnk.2.pi/N)
// n=0,1,2,...N-1
// N=2^order.
// The conjugate-symmetric output sequence is represented using a CCS vector,
// which is of length N+2, and is organized as follows:
// Index: 0 1 2 3 4 5 . . . N-2 N-1 N N+1
// Component: R0 0 R1 I1 R2 I2 . . . R[N/2-1] I[N/2-1] R[N/2] 0
// where R[n] and I[n], respectively, denote the real and imaginary components
// for FFT bin 'n'. Bins are numbered from 0 to N/2, where N is the FFT length.
// Bin index 0 corresponds to the DC component, and bin index N/2 corresponds to
// the foldover frequency.
//
// Input Arguments:
// self - pointer to preallocated and initialized FFT specification structure.
// real_data_in - the input signal. For an ARM Neon platform, it must be
// aligned on a 32-byte boundary.
//
// Output Arguments:
// complex_data_out - the output complex signal with (2^order + 2) 16-bit
// elements. For an ARM Neon platform, it must be different
// from real_data_in, and aligned on a 32-byte boundary.
//
// Return Value:
// 0 - FFT calculation is successful.
// -1 - Error with bad arguments (null pointers).
int WebRtcSpl_RealForwardFFT(struct RealFFT* self,
const int16_t* real_data_in,
int16_t* complex_data_out);
// Compute the inverse FFT for a conjugate-symmetric input sequence of length of
// 2^order, where 1 < order <= MAX_FFT_ORDER. Transform length is determined by
// the specification structure, which must be initialized prior to calling the
// FFT function with WebRtcSpl_CreateRealFFT().
// For a transform of length M, the input sequence is represented using a packed
// CCS vector of length M+2, which is explained in the comments for
// WebRtcSpl_RealForwardFFTC above.
//
// Input Arguments:
// self - pointer to preallocated and initialized FFT specification structure.
// complex_data_in - the input complex signal with (2^order + 2) 16-bit
// elements. For an ARM Neon platform, it must be aligned on
// a 32-byte boundary.
//
// Output Arguments:
// real_data_out - the output real signal. For an ARM Neon platform, it must
// be different to complex_data_in, and aligned on a 32-byte
// boundary.
//
// Return Value:
// 0 or a positive number - a value that the elements in the |real_data_out|
// should be shifted left with in order to get
// correct physical values.
// -1 - Error with bad arguments (null pointers).
int WebRtcSpl_RealInverseFFT(struct RealFFT* self,
const int16_t* complex_data_in,
int16_t* real_data_out);
#ifdef __cplusplus
}
#endif
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This header file includes the inline functions in
// the fix point signal processing library.
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
#include "webrtc/rtc_base/compile_assert_c.h"
extern const int8_t kWebRtcSpl_CountLeadingZeros32_Table[64];
// Don't call this directly except in tests!
static __inline int WebRtcSpl_CountLeadingZeros32_NotBuiltin(uint32_t n) {
// Normalize n by rounding up to the nearest number that is a sequence of 0
// bits followed by a sequence of 1 bits. This number has the same number of
// leading zeros as the original n. There are exactly 33 such values.
n |= n >> 1;
n |= n >> 2;
n |= n >> 4;
n |= n >> 8;
n |= n >> 16;
// Multiply the modified n with a constant selected (by exhaustive search)
// such that each of the 33 possible values of n give a product whose 6 most
// significant bits are unique. Then look up the answer in the table.
return kWebRtcSpl_CountLeadingZeros32_Table[(n * 0x8c0b2891) >> 26];
}
// Don't call this directly except in tests!
static __inline int WebRtcSpl_CountLeadingZeros64_NotBuiltin(uint64_t n) {
const int leading_zeros = n >> 32 == 0 ? 32 : 0;
return leading_zeros + WebRtcSpl_CountLeadingZeros32_NotBuiltin(
(uint32_t)(n >> (32 - leading_zeros)));
}
// Returns the number of leading zero bits in the argument.
static __inline int WebRtcSpl_CountLeadingZeros32(uint32_t n) {
#ifdef __GNUC__
RTC_COMPILE_ASSERT(sizeof(unsigned int) == sizeof(uint32_t));
return n == 0 ? 32 : __builtin_clz(n);
#else
return WebRtcSpl_CountLeadingZeros32_NotBuiltin(n);
#endif
}
// Returns the number of leading zero bits in the argument.
static __inline int WebRtcSpl_CountLeadingZeros64(uint64_t n) {
#ifdef __GNUC__
RTC_COMPILE_ASSERT(sizeof(unsigned long long) == sizeof(uint64_t)); // NOLINT
return n == 0 ? 64 : __builtin_clzll(n);
#else
return WebRtcSpl_CountLeadingZeros64_NotBuiltin(n);
#endif
}
#ifdef WEBRTC_ARCH_ARM_V7
#include "webrtc/common_audio/signal_processing/include/spl_inl_armv7.h"
#else
#if defined(MIPS32_LE)
#include "webrtc/common_audio/signal_processing/include/spl_inl_mips.h"
#endif
#if !defined(MIPS_DSP_R1_LE)
static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
int16_t out16 = (int16_t)value32;
if (value32 > 32767)
out16 = 32767;
else if (value32 < -32768)
out16 = -32768;
return out16;
}
static __inline int32_t WebRtcSpl_AddSatW32(int32_t a, int32_t b) {
// Do the addition in unsigned numbers, since signed overflow is undefined
// behavior.
const int32_t sum = (int32_t)((uint32_t)a + (uint32_t)b);
// a + b can't overflow if a and b have different signs. If they have the
// same sign, a + b also has the same sign iff it didn't overflow.
if ((a < 0) == (b < 0) && (a < 0) != (sum < 0)) {
// The direction of the overflow is obvious from the sign of a + b.
return sum < 0 ? INT32_MAX : INT32_MIN;
}
return sum;
}
static __inline int32_t WebRtcSpl_SubSatW32(int32_t a, int32_t b) {
// Do the subtraction in unsigned numbers, since signed overflow is undefined
// behavior.
const int32_t diff = (int32_t)((uint32_t)a - (uint32_t)b);
// a - b can't overflow if a and b have the same sign. If they have different
// signs, a - b has the same sign as a iff it didn't overflow.
if ((a < 0) != (b < 0) && (a < 0) != (diff < 0)) {
// The direction of the overflow is obvious from the sign of a - b.
return diff < 0 ? INT32_MAX : INT32_MIN;
}
return diff;
}
static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
return WebRtcSpl_SatW32ToW16((int32_t)a + (int32_t)b);
}
static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
return WebRtcSpl_SatW32ToW16((int32_t)var1 - (int32_t)var2);
}
#endif // #if !defined(MIPS_DSP_R1_LE)
#if !defined(MIPS32_LE)
static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
return 32 - WebRtcSpl_CountLeadingZeros32(n);
}
// Return the number of steps a can be left-shifted without overflow,
// or 0 if a == 0.
static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a < 0 ? ~a : a) - 1;
}
// Return the number of steps a can be left-shifted without overflow,
// or 0 if a == 0.
static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a);
}
// Return the number of steps a can be left-shifted without overflow,
// or 0 if a == 0.
static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
const int32_t a32 = a;
return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a < 0 ? ~a32 : a32) - 17;
}
static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
return (a * b + c);
}
#endif // #if !defined(MIPS32_LE)
#endif // WEBRTC_ARCH_ARM_V7
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the implementation of functions
* WebRtcSpl_MaxAbsValueW16C()
* WebRtcSpl_MaxAbsValueW32C()
* WebRtcSpl_MaxValueW16C()
* WebRtcSpl_MaxValueW32C()
* WebRtcSpl_MinValueW16C()
* WebRtcSpl_MinValueW32C()
* WebRtcSpl_MaxAbsIndexW16()
* WebRtcSpl_MaxIndexW16()
* WebRtcSpl_MaxIndexW32()
* WebRtcSpl_MinIndexW16()
* WebRtcSpl_MinIndexW32()
*
*/
#include <stdlib.h>
#include "webrtc/rtc_base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
// TODO(bjorn/kma): Consolidate function pairs (e.g. combine
// WebRtcSpl_MaxAbsValueW16C and WebRtcSpl_MaxAbsIndexW16 into a single one.)
// TODO(kma): Move the next six functions into min_max_operations_c.c.
// Maximum absolute value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length) {
size_t i = 0;
int absolute = 0, maximum = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute = abs((int)vector[i]);
if (absolute > maximum) {
maximum = absolute;
}
}
// Guard the case for abs(-32768).
if (maximum > WEBRTC_SPL_WORD16_MAX) {
maximum = WEBRTC_SPL_WORD16_MAX;
}
return (int16_t)maximum;
}
// Maximum absolute value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length) {
// Use uint32_t for the local variables, to accommodate the return value
// of abs(0x80000000), which is 0x80000000.
uint32_t absolute = 0, maximum = 0;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute = abs((int)vector[i]);
if (absolute > maximum) {
maximum = absolute;
}
}
maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX);
return (int32_t)maximum;
}
// Maximum value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length) {
int16_t maximum = WEBRTC_SPL_WORD16_MIN;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum)
maximum = vector[i];
}
return maximum;
}
// Maximum value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length) {
int32_t maximum = WEBRTC_SPL_WORD32_MIN;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum)
maximum = vector[i];
}
return maximum;
}
// Minimum value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length) {
int16_t minimum = WEBRTC_SPL_WORD16_MAX;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum)
minimum = vector[i];
}
return minimum;
}
// Minimum value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length) {
int32_t minimum = WEBRTC_SPL_WORD32_MAX;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum)
minimum = vector[i];
}
return minimum;
}
// Index of maximum absolute value in a word16 vector.
size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length) {
// Use type int for local variables, to accomodate the value of abs(-32768).
size_t i = 0, index = 0;
int absolute = 0, maximum = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute = abs((int)vector[i]);
if (absolute > maximum) {
maximum = absolute;
index = i;
}
}
return index;
}
// Index of maximum value in a word16 vector.
size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length) {
size_t i = 0, index = 0;
int16_t maximum = WEBRTC_SPL_WORD16_MIN;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum) {
maximum = vector[i];
index = i;
}
}
return index;
}
// Index of maximum value in a word32 vector.
size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length) {
size_t i = 0, index = 0;
int32_t maximum = WEBRTC_SPL_WORD32_MIN;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum) {
maximum = vector[i];
index = i;
}
}
return index;
}
// Index of minimum value in a word16 vector.
size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length) {
size_t i = 0, index = 0;
int16_t minimum = WEBRTC_SPL_WORD16_MAX;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum) {
minimum = vector[i];
index = i;
}
}
return index;
}
// Index of minimum value in a word32 vector.
size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length) {
size_t i = 0, index = 0;
int32_t minimum = WEBRTC_SPL_WORD32_MAX;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum) {
minimum = vector[i];
index = i;
}
}
return index;
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains resampling functions between 48 kHz and nb/wb.
* The description header can be found in signal_processing_library.h
*
*/
#include <string.h>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
////////////////////////////
///// 48 kHz -> 16 kHz /////
////////////////////////////
// 48 -> 16 resampler
void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
WebRtcSpl_State48khzTo16khz* state, int32_t* tmpmem)
{
///// 48 --> 48(LP) /////
// int16_t in[480]
// int32_t out[480]
/////
WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
///// 48 --> 32 /////
// int32_t in[480]
// int32_t out[320]
/////
// copy state to and from input array
memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(int32_t));
memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(int32_t));
WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
///// 32 --> 16 /////
// int32_t in[320]
// int16_t out[160]
/////
WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
}
// initialize state of 48 -> 16 resampler
void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
{
memset(state->S_48_48, 0, 16 * sizeof(int32_t));
memset(state->S_48_32, 0, 8 * sizeof(int32_t));
memset(state->S_32_16, 0, 8 * sizeof(int32_t));
}
////////////////////////////
///// 16 kHz -> 48 kHz /////
////////////////////////////
// 16 -> 48 resampler
void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
WebRtcSpl_State16khzTo48khz* state, int32_t* tmpmem)
{
///// 16 --> 32 /////
// int16_t in[160]
// int32_t out[320]
/////
WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
///// 32 --> 24 /////
// int32_t in[320]
// int32_t out[240]
// copy state to and from input array
/////
memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(int32_t));
memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(int32_t));
WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
///// 24 --> 48 /////
// int32_t in[240]
// int16_t out[480]
/////
WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
}
// initialize state of 16 -> 48 resampler
void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
{
memset(state->S_16_32, 0, 8 * sizeof(int32_t));
memset(state->S_32_24, 0, 8 * sizeof(int32_t));
memset(state->S_24_48, 0, 8 * sizeof(int32_t));
}
////////////////////////////
///// 48 kHz -> 8 kHz /////
////////////////////////////
// 48 -> 8 resampler
void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* out,
WebRtcSpl_State48khzTo8khz* state, int32_t* tmpmem)
{
///// 48 --> 24 /////
// int16_t in[480]
// int32_t out[240]
/////
WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
///// 24 --> 24(LP) /////
// int32_t in[240]
// int32_t out[240]
/////
WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
///// 24 --> 16 /////
// int32_t in[240]
// int32_t out[160]
/////
// copy state to and from input array
memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(int32_t));
memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(int32_t));
WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
///// 16 --> 8 /////
// int32_t in[160]
// int16_t out[80]
/////
WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
}
// initialize state of 48 -> 8 resampler
void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
{
memset(state->S_48_24, 0, 8 * sizeof(int32_t));
memset(state->S_24_24, 0, 16 * sizeof(int32_t));
memset(state->S_24_16, 0, 8 * sizeof(int32_t));
memset(state->S_16_8, 0, 8 * sizeof(int32_t));
}
////////////////////////////
///// 8 kHz -> 48 kHz /////
////////////////////////////
// 8 -> 48 resampler
void WebRtcSpl_Resample8khzTo48khz(const int16_t* in, int16_t* out,
WebRtcSpl_State8khzTo48khz* state, int32_t* tmpmem)
{
///// 8 --> 16 /////
// int16_t in[80]
// int32_t out[160]
/////
WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
///// 16 --> 12 /////
// int32_t in[160]
// int32_t out[120]
/////
// copy state to and from input array
memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(int32_t));
memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(int32_t));
WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
///// 12 --> 24 /////
// int32_t in[120]
// int16_t out[240]
/////
WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
///// 24 --> 48 /////
// int32_t in[240]
// int16_t out[480]
/////
WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
}
// initialize state of 8 -> 48 resampler
void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
{
memset(state->S_8_16, 0, 8 * sizeof(int32_t));
memset(state->S_16_12, 0, 8 * sizeof(int32_t));
memset(state->S_12_24, 0, 8 * sizeof(int32_t));
memset(state->S_24_48, 0, 8 * sizeof(int32_t));
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This header file contains some internal resampling functions.
*
*/
#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
#include "webrtc/rtc_base/sanitizer.h"
// allpass filter coefficients.
static const int16_t kResampleAllpass[2][3] = {
{821, 6110, 12382},
{3050, 9368, 15063}
};
//
// decimator
// input: int32_t (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
// output: int16_t (saturated) (of length len/2)
// state: filter state array; length = 8
void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
int32_t *state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
len >>= 1;
// lower allpass filter (operates on even input samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i << 1];
diff = tmp0 - state[1];
// UBSan: -1771017321 - 999586185 cannot be represented in type 'int'
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// divide by two and store temporarily
in[i << 1] = (state[3] >> 1);
}
in++;
// upper allpass filter (operates on odd input samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i << 1];
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// divide by two and store temporarily
in[i << 1] = (state[7] >> 1);
}
in--;
// combine allpass outputs
for (i = 0; i < len; i += 2)
{
// divide by two, add both allpass outputs and round
tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
if (tmp0 > (int32_t)0x00007FFF)
tmp0 = 0x00007FFF;
if (tmp0 < (int32_t)0xFFFF8000)
tmp0 = 0xFFFF8000;
out[i] = (int16_t)tmp0;
if (tmp1 > (int32_t)0x00007FFF)
tmp1 = 0x00007FFF;
if (tmp1 < (int32_t)0xFFFF8000)
tmp1 = 0xFFFF8000;
out[i + 1] = (int16_t)tmp1;
}
}
//
// decimator
// input: int16_t
// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len/2)
// state: filter state array; length = 8
void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
WebRtcSpl_DownBy2ShortToInt(const int16_t *in,
int32_t len,
int32_t *out,
int32_t *state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
len >>= 1;
// lower allpass filter (operates on even input samples)
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// UBSan: -1379909682 - 834099714 cannot be represented in type 'int'
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// divide by two and store temporarily
out[i] = (state[3] >> 1);
}
in++;
// upper allpass filter (operates on odd input samples)
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// divide by two and store temporarily
out[i] += (state[7] >> 1);
}
in--;
}
//
// interpolator
// input: int16_t
// output: int32_t (normalized, not saturated) (of length len*2)
// state: filter state array; length = 8
void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len, int32_t *out,
int32_t *state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
// upper allpass filter (generates odd output samples)
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// scale down, round and store
out[i << 1] = state[7] >> 15;
}
out++;
// lower allpass filter (generates even output samples)
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// scale down, round and store
out[i << 1] = state[3] >> 15;
}
}
//
// interpolator
// input: int32_t (shifted 15 positions to the left, + offset 16384)
// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len*2)
// state: filter state array; length = 8
void WebRtcSpl_UpBy2IntToInt(const int32_t *in, int32_t len, int32_t *out,
int32_t *state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
// upper allpass filter (generates odd output samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i];
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// scale down, round and store
out[i << 1] = state[7];
}
out++;
// lower allpass filter (generates even output samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i];
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// scale down, round and store
out[i << 1] = state[3];
}
}
//
// interpolator
// input: int32_t (shifted 15 positions to the left, + offset 16384)
// output: int16_t (saturated) (of length len*2)
// state: filter state array; length = 8
void WebRtcSpl_UpBy2IntToShort(const int32_t *in, int32_t len, int16_t *out,
int32_t *state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
// upper allpass filter (generates odd output samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i];
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// scale down, saturate and store
tmp1 = state[7] >> 15;
if (tmp1 > (int32_t)0x00007FFF)
tmp1 = 0x00007FFF;
if (tmp1 < (int32_t)0xFFFF8000)
tmp1 = 0xFFFF8000;
out[i << 1] = (int16_t)tmp1;
}
out++;
// lower allpass filter (generates even output samples)
for (i = 0; i < len; i++)
{
tmp0 = in[i];
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// scale down, saturate and store
tmp1 = state[3] >> 15;
if (tmp1 > (int32_t)0x00007FFF)
tmp1 = 0x00007FFF;
if (tmp1 < (int32_t)0xFFFF8000)
tmp1 = 0xFFFF8000;
out[i << 1] = (int16_t)tmp1;
}
}
// lowpass filter
// input: int16_t
// output: int32_t (normalized, not saturated)
// state: filter state array; length = 8
void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len, int32_t* out,
int32_t* state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
len >>= 1;
// lower allpass filter: odd input -> even output samples
in++;
// initial state of polyphase delay element
tmp0 = state[12];
for (i = 0; i < len; i++)
{
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// scale down, round and store
out[i << 1] = state[3] >> 1;
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
}
in--;
// upper allpass filter: even input -> even output samples
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
diff = tmp0 - state[5];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// average the two allpass outputs, scale down and store
out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
}
// switch to odd output samples
out++;
// lower allpass filter: even input -> odd output samples
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
diff = tmp0 - state[9];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[8] + diff * kResampleAllpass[1][0];
state[8] = tmp0;
diff = tmp1 - state[10];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[9] + diff * kResampleAllpass[1][1];
state[9] = tmp1;
diff = tmp0 - state[11];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[11] = state[10] + diff * kResampleAllpass[1][2];
state[10] = tmp0;
// scale down, round and store
out[i << 1] = state[11] >> 1;
}
// upper allpass filter: odd input -> odd output samples
in++;
for (i = 0; i < len; i++)
{
tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
diff = tmp0 - state[13];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[12] + diff * kResampleAllpass[0][0];
state[12] = tmp0;
diff = tmp1 - state[14];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[13] + diff * kResampleAllpass[0][1];
state[13] = tmp1;
diff = tmp0 - state[15];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[15] = state[14] + diff * kResampleAllpass[0][2];
state[14] = tmp0;
// average the two allpass outputs, scale down and store
out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
}
}
// lowpass filter
// input: int32_t (shifted 15 positions to the left, + offset 16384)
// output: int32_t (normalized, not saturated)
// state: filter state array; length = 8
void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
int32_t* state)
{
int32_t tmp0, tmp1, diff;
int32_t i;
len >>= 1;
// lower allpass filter: odd input -> even output samples
in++;
// initial state of polyphase delay element
tmp0 = state[12];
for (i = 0; i < len; i++)
{
diff = tmp0 - state[1];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[0] + diff * kResampleAllpass[1][0];
state[0] = tmp0;
diff = tmp1 - state[2];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[1] + diff * kResampleAllpass[1][1];
state[1] = tmp1;
diff = tmp0 - state[3];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[3] = state[2] + diff * kResampleAllpass[1][2];
state[2] = tmp0;
// scale down, round and store
out[i << 1] = state[3] >> 1;
tmp0 = in[i << 1];
}
in--;
// upper allpass filter: even input -> even output samples
for (i = 0; i < len; i++)
{
tmp0 = in[i << 1];
diff = tmp0 - state[5];
// UBSan: -794814117 - 1566149201 cannot be represented in type 'int'
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[4] + diff * kResampleAllpass[0][0];
state[4] = tmp0;
diff = tmp1 - state[6];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[5] + diff * kResampleAllpass[0][1];
state[5] = tmp1;
diff = tmp0 - state[7];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[7] = state[6] + diff * kResampleAllpass[0][2];
state[6] = tmp0;
// average the two allpass outputs, scale down and store
out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
}
// switch to odd output samples
out++;
// lower allpass filter: even input -> odd output samples
for (i = 0; i < len; i++)
{
tmp0 = in[i << 1];
diff = tmp0 - state[9];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[8] + diff * kResampleAllpass[1][0];
state[8] = tmp0;
diff = tmp1 - state[10];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[9] + diff * kResampleAllpass[1][1];
state[9] = tmp1;
diff = tmp0 - state[11];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[11] = state[10] + diff * kResampleAllpass[1][2];
state[10] = tmp0;
// scale down, round and store
out[i << 1] = state[11] >> 1;
}
// upper allpass filter: odd input -> odd output samples
in++;
for (i = 0; i < len; i++)
{
tmp0 = in[i << 1];
diff = tmp0 - state[13];
// scale down and round
diff = (diff + (1 << 13)) >> 14;
tmp1 = state[12] + diff * kResampleAllpass[0][0];
state[12] = tmp0;
diff = tmp1 - state[14];
// scale down and round
diff = diff >> 14;
if (diff < 0)
diff += 1;
tmp0 = state[13] + diff * kResampleAllpass[0][1];
state[13] = tmp1;
diff = tmp0 - state[15];
// scale down and truncate
diff = diff >> 14;
if (diff < 0)
diff += 1;
state[15] = state[14] + diff * kResampleAllpass[0][2];
state[14] = tmp0;
// average the two allpass outputs, scale down and store
out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
}
}

60
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/resample_by_2_internal.h
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This header file contains some internal resampling functions.
*
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
#include <stdint.h>
/*******************************************************************
* resample_by_2_fast.c
* Functions for internal use in the other resample functions
******************************************************************/
void WebRtcSpl_DownBy2IntToShort(int32_t* in,
int32_t len,
int16_t* out,
int32_t* state);
void WebRtcSpl_DownBy2ShortToInt(const int16_t* in,
int32_t len,
int32_t* out,
int32_t* state);
void WebRtcSpl_UpBy2ShortToInt(const int16_t* in,
int32_t len,
int32_t* out,
int32_t* state);
void WebRtcSpl_UpBy2IntToInt(const int32_t* in,
int32_t len,
int32_t* out,
int32_t* state);
void WebRtcSpl_UpBy2IntToShort(const int32_t* in,
int32_t len,
int16_t* out,
int32_t* state);
void WebRtcSpl_LPBy2ShortToInt(const int16_t* in,
int32_t len,
int32_t* out,
int32_t* state);
void WebRtcSpl_LPBy2IntToInt(const int32_t* in,
int32_t len,
int32_t* out,
int32_t* state);
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc/common_audio/signal_processing/resample_fractional.c
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the resampling functions between 48, 44, 32 and 24 kHz.
* The description headers can be found in signal_processing_library.h
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
// interpolation coefficients
static const int16_t kCoefficients48To32[2][8] = {
{778, -2050, 1087, 23285, 12903, -3783, 441, 222},
{222, 441, -3783, 12903, 23285, 1087, -2050, 778}
};
static const int16_t kCoefficients32To24[3][8] = {
{767, -2362, 2434, 24406, 10620, -3838, 721, 90},
{386, -381, -2646, 19062, 19062, -2646, -381, 386},
{90, 721, -3838, 10620, 24406, 2434, -2362, 767}
};
static const int16_t kCoefficients44To32[4][9] = {
{117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
{-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
{50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
{-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
};
// Resampling ratio: 2/3
// input: int32_t (normalized, not saturated) :: size 3 * K
// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 2 * K
// K: number of blocks
void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
{
/////////////////////////////////////////////////////////////
// Filter operation:
//
// Perform resampling (3 input samples -> 2 output samples);
// process in sub blocks of size 3 samples.
int32_t tmp;
size_t m;
for (m = 0; m < K; m++)
{
tmp = 1 << 14;
tmp += kCoefficients48To32[0][0] * In[0];
tmp += kCoefficients48To32[0][1] * In[1];
tmp += kCoefficients48To32[0][2] * In[2];
tmp += kCoefficients48To32[0][3] * In[3];
tmp += kCoefficients48To32[0][4] * In[4];
tmp += kCoefficients48To32[0][5] * In[5];
tmp += kCoefficients48To32[0][6] * In[6];
tmp += kCoefficients48To32[0][7] * In[7];
Out[0] = tmp;
tmp = 1 << 14;
tmp += kCoefficients48To32[1][0] * In[1];
tmp += kCoefficients48To32[1][1] * In[2];
tmp += kCoefficients48To32[1][2] * In[3];
tmp += kCoefficients48To32[1][3] * In[4];
tmp += kCoefficients48To32[1][4] * In[5];
tmp += kCoefficients48To32[1][5] * In[6];
tmp += kCoefficients48To32[1][6] * In[7];
tmp += kCoefficients48To32[1][7] * In[8];
Out[1] = tmp;
// update pointers
In += 3;
Out += 2;
}
}
// Resampling ratio: 3/4
// input: int32_t (normalized, not saturated) :: size 4 * K
// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 3 * K
// K: number of blocks
void WebRtcSpl_Resample32khzTo24khz(const int32_t *In, int32_t *Out, size_t K)
{
/////////////////////////////////////////////////////////////
// Filter operation:
//
// Perform resampling (4 input samples -> 3 output samples);
// process in sub blocks of size 4 samples.
size_t m;
int32_t tmp;
for (m = 0; m < K; m++)
{
tmp = 1 << 14;
tmp += kCoefficients32To24[0][0] * In[0];
tmp += kCoefficients32To24[0][1] * In[1];
tmp += kCoefficients32To24[0][2] * In[2];
tmp += kCoefficients32To24[0][3] * In[3];
tmp += kCoefficients32To24[0][4] * In[4];
tmp += kCoefficients32To24[0][5] * In[5];
tmp += kCoefficients32To24[0][6] * In[6];
tmp += kCoefficients32To24[0][7] * In[7];
Out[0] = tmp;
tmp = 1 << 14;
tmp += kCoefficients32To24[1][0] * In[1];
tmp += kCoefficients32To24[1][1] * In[2];
tmp += kCoefficients32To24[1][2] * In[3];
tmp += kCoefficients32To24[1][3] * In[4];
tmp += kCoefficients32To24[1][4] * In[5];
tmp += kCoefficients32To24[1][5] * In[6];
tmp += kCoefficients32To24[1][6] * In[7];
tmp += kCoefficients32To24[1][7] * In[8];
Out[1] = tmp;
tmp = 1 << 14;
tmp += kCoefficients32To24[2][0] * In[2];
tmp += kCoefficients32To24[2][1] * In[3];
tmp += kCoefficients32To24[2][2] * In[4];
tmp += kCoefficients32To24[2][3] * In[5];
tmp += kCoefficients32To24[2][4] * In[6];
tmp += kCoefficients32To24[2][5] * In[7];
tmp += kCoefficients32To24[2][6] * In[8];
tmp += kCoefficients32To24[2][7] * In[9];
Out[2] = tmp;
// update pointers
In += 4;
Out += 3;
}
}
//
// fractional resampling filters
// Fout = 11/16 * Fin
// Fout = 8/11 * Fin
//
// compute two inner-products and store them to output array
static void WebRtcSpl_ResampDotProduct(const int32_t *in1, const int32_t *in2,
const int16_t *coef_ptr, int32_t *out1,
int32_t *out2)
{
int32_t tmp1 = 16384;
int32_t tmp2 = 16384;
int16_t coef;
coef = coef_ptr[0];
tmp1 += coef * in1[0];
tmp2 += coef * in2[-0];
coef = coef_ptr[1];
tmp1 += coef * in1[1];
tmp2 += coef * in2[-1];
coef = coef_ptr[2];
tmp1 += coef * in1[2];
tmp2 += coef * in2[-2];
coef = coef_ptr[3];
tmp1 += coef * in1[3];
tmp2 += coef * in2[-3];
coef = coef_ptr[4];
tmp1 += coef * in1[4];
tmp2 += coef * in2[-4];
coef = coef_ptr[5];
tmp1 += coef * in1[5];
tmp2 += coef * in2[-5];
coef = coef_ptr[6];
tmp1 += coef * in1[6];
tmp2 += coef * in2[-6];
coef = coef_ptr[7];
tmp1 += coef * in1[7];
tmp2 += coef * in2[-7];
coef = coef_ptr[8];
*out1 = tmp1 + coef * in1[8];
*out2 = tmp2 + coef * in2[-8];
}
// Resampling ratio: 8/11
// input: int32_t (normalized, not saturated) :: size 11 * K
// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 8 * K
// K: number of blocks
void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
{
/////////////////////////////////////////////////////////////
// Filter operation:
//
// Perform resampling (11 input samples -> 8 output samples);
// process in sub blocks of size 11 samples.
int32_t tmp;
size_t m;
for (m = 0; m < K; m++)
{
tmp = 1 << 14;
// first output sample
Out[0] = ((int32_t)In[3] << 15) + tmp;
// sum and accumulate filter coefficients and input samples
tmp += kCoefficients44To32[3][0] * In[5];
tmp += kCoefficients44To32[3][1] * In[6];
tmp += kCoefficients44To32[3][2] * In[7];
tmp += kCoefficients44To32[3][3] * In[8];
tmp += kCoefficients44To32[3][4] * In[9];
tmp += kCoefficients44To32[3][5] * In[10];
tmp += kCoefficients44To32[3][6] * In[11];
tmp += kCoefficients44To32[3][7] * In[12];
tmp += kCoefficients44To32[3][8] * In[13];
Out[4] = tmp;
// sum and accumulate filter coefficients and input samples
WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
// sum and accumulate filter coefficients and input samples
WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
// sum and accumulate filter coefficients and input samples
WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
// update pointers
In += 11;
Out += 8;
}
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/* The global function contained in this file initializes SPL function
* pointers, currently only for ARM platforms.
*
* Some code came from common/rtcd.c in the WebM project.
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
/* Declare function pointers. */
MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
MaxValueW16 WebRtcSpl_MaxValueW16;
MaxValueW32 WebRtcSpl_MaxValueW32;
MinValueW16 WebRtcSpl_MinValueW16;
MinValueW32 WebRtcSpl_MinValueW32;
CrossCorrelation WebRtcSpl_CrossCorrelation;
DownsampleFast WebRtcSpl_DownsampleFast;
ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
#if (!defined(WEBRTC_HAS_NEON)) && !defined(MIPS32_LE)
/* Initialize function pointers to the generic C version. */
static void InitPointersToC(void) {
WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16C;
WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32C;
WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16C;
WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32C;
WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationC;
WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastC;
WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
}
#endif
#if defined(WEBRTC_HAS_NEON)
/* Initialize function pointers to the Neon version. */
static void InitPointersToNeon(void) {
WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32Neon;
WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16Neon;
WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32Neon;
WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16Neon;
WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32Neon;
WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationNeon;
WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastNeon;
WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
}
#endif
#if defined(MIPS32_LE)
/* Initialize function pointers to the MIPS version. */
static void InitPointersToMIPS(void) {
WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16_mips;
WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16_mips;
WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32_mips;
WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16_mips;
WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32_mips;
WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelation_mips;
WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFast_mips;
#if defined(MIPS_DSP_R1_LE)
WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32_mips;
WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRound_mips;
#else
WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
#endif
}
#endif
static void InitFunctionPointers(void) {
#if defined(WEBRTC_HAS_NEON)
InitPointersToNeon();
#elif defined(MIPS32_LE)
InitPointersToMIPS();
#else
InitPointersToC();
#endif /* WEBRTC_HAS_NEON */
}
#include <pthread.h>
static void once(void (*func)(void)) {
static pthread_once_t lock = PTHREAD_ONCE_INIT;
pthread_once(&lock, func);
}
// #if defined(WEBRTC_POSIX)
// #include <pthread.h>
// static void once(void (*func)(void)) {
// static pthread_once_t lock = PTHREAD_ONCE_INIT;
// pthread_once(&lock, func);
// }
// #elif defined(_WIN32)
// #include <windows.h>
// static void once(void (*func)(void)) {
// /* Didn't use InitializeCriticalSection() since there's no race-free context
// * in which to execute it.
// *
// * TODO(kma): Change to different implementation (e.g.
// * InterlockedCompareExchangePointer) to avoid issues similar to
// * http://code.google.com/p/webm/issues/detail?id=467.
// */
// static CRITICAL_SECTION lock = {(void *)((size_t)-1), -1, 0, 0, 0, 0};
// static int done = 0;
// EnterCriticalSection(&lock);
// if (!done) {
// func();
// done = 1;
// }
// LeaveCriticalSection(&lock);
// }
// /* There's no fallback version as an #else block here to ensure thread safety.
// * In case of neither pthread for WEBRTC_POSIX nor _WIN32 is present, build
// * system should pick it up.
// */
// #endif /* WEBRTC_POSIX */
void WebRtcSpl_Init(void) {
once(InitFunctionPointers);
}

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/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdint.h>
#include "webrtc/common_audio/signal_processing/include/spl_inl.h"
// Table used by WebRtcSpl_CountLeadingZeros32_NotBuiltin. For each uint32_t n
// that's a sequence of 0 bits followed by a sequence of 1 bits, the entry at
// index (n * 0x8c0b2891) >> 26 in this table gives the number of zero bits in
// n.
const int8_t kWebRtcSpl_CountLeadingZeros32_Table[64] = {
32, 8, 17, -1, -1, 14, -1, -1, -1, 20, -1, -1, -1, 28, -1, 18,
-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 0, 26, 25, 24,
4, 11, 23, 31, 3, 7, 10, 16, 22, 30, -1, -1, 2, 6, 13, 9,
-1, 15, -1, 21, -1, 29, 19, -1, -1, -1, -1, -1, 1, 27, 5, 12,
};

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the function WebRtcSpl_Sqrt().
* The description header can be found in signal_processing_library.h
*
*/
#include "webrtc/rtc_base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
int32_t WebRtcSpl_SqrtLocal(int32_t in);
int32_t WebRtcSpl_SqrtLocal(int32_t in)
{
int16_t x_half, t16;
int32_t A, B, x2;
/* The following block performs:
y=in/2
x=y-2^30
x_half=x/2^31
t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ 0.875*((x_half)^5)
*/
B = in / 2;
B = B - ((int32_t)0x40000000); // B = in/2 - 1/2
x_half = (int16_t)(B >> 16); // x_half = x/2 = (in-1)/2
B = B + ((int32_t)0x40000000); // B = 1 + x/2
B = B + ((int32_t)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
x2 = ((int32_t)x_half) * ((int32_t)x_half) * 2; // A = (x/2)^2
A = -x2; // A = -(x/2)^2
B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
A >>= 16;
A = A * A * 2; // A = (x/2)^4
t16 = (int16_t)(A >> 16);
B += -20480 * t16 * 2; // B = B - 0.625*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
A = x_half * t16 * 2; // A = (x/2)^5
t16 = (int16_t)(A >> 16);
B += 28672 * t16 * 2; // B = B + 0.875*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
t16 = (int16_t)(x2 >> 16);
A = x_half * t16 * 2; // A = x/2^3
B = B + (A >> 1); // B = B + 0.5*A
// After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
B = B + ((int32_t)32768); // Round off bit
return B;
}
int32_t WebRtcSpl_Sqrt(int32_t value)
{
/*
Algorithm:
Six term Taylor Series is used here to compute the square root of a number
y^0.5 = (1+x)^0.5 where x = y-1
= 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
0.5 <= x < 1
Example of how the algorithm works, with ut=sqrt(in), and
with in=73632 and ut=271 (even shift value case):
in=73632
y= in/131072
x=y-1
t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
ut=t*(1/sqrt(2))*512
or:
in=73632
in2=73632*2^14
y= in2/2^31
x=y-1
t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
ut=t*(1/sqrt(2))
ut2=ut*2^9
which gives:
in = 73632
in2 = 1206386688
y = 0.56176757812500
x = -0.43823242187500
t = 0.74973506527313
ut = 0.53014274874797
ut2 = 2.714330873589594e+002
or:
in=73632
in2=73632*2^14
y=in2/2
x=y-2^30
x_half=x/2^31
t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ 0.875*((x_half)^5)
ut=t*(1/sqrt(2))
ut2=ut*2^9
which gives:
in = 73632
in2 = 1206386688
y = 603193344
x = -470548480
x_half = -0.21911621093750
t = 0.74973506527313
ut = 0.53014274874797
ut2 = 2.714330873589594e+002
*/
int16_t x_norm, nshift, t16, sh;
int32_t A;
int16_t k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
A = value;
// The convention in this function is to calculate sqrt(abs(A)). Negate the
// input if it is negative.
if (A < 0) {
if (A == WEBRTC_SPL_WORD32_MIN) {
// This number cannot be held in an int32_t after negating.
// Map it to the maximum positive value.
A = WEBRTC_SPL_WORD32_MAX;
} else {
A = -A;
}
} else if (A == 0) {
return 0; // sqrt(0) = 0
}
sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
{
A = A + ((int32_t)32768); // Round off bit
} else
{
A = WEBRTC_SPL_WORD32_MAX;
}
x_norm = (int16_t)(A >> 16); // x_norm = AH
nshift = (sh / 2);
RTC_DCHECK_GE(nshift, 0);
A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
if (2 * nshift == sh) {
// Even shift value case
t16 = (int16_t)(A >> 16); // t16 = AH
A = k_sqrt_2 * t16 * 2; // A = 1/sqrt(2)*t16
A = A + ((int32_t)32768); // Round off
A = A & ((int32_t)0x7fff0000); // Round off
A >>= 15; // A = A>>16
} else
{
A >>= 16; // A = A>>16
}
A = A & ((int32_t)0x0000ffff);
A >>= nshift; // De-normalize the result.
return A;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains implementations of the functions
* WebRtcSpl_VectorBitShiftW16()
* WebRtcSpl_VectorBitShiftW32()
* WebRtcSpl_VectorBitShiftW32ToW16()
* WebRtcSpl_ScaleVector()
* WebRtcSpl_ScaleVectorWithSat()
* WebRtcSpl_ScaleAndAddVectors()
* WebRtcSpl_ScaleAndAddVectorsWithRoundC()
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
void WebRtcSpl_VectorBitShiftW16(int16_t *res, size_t length,
const int16_t *in, int16_t right_shifts)
{
size_t i;
if (right_shifts > 0)
{
for (i = length; i > 0; i--)
{
(*res++) = ((*in++) >> right_shifts);
}
} else
{
for (i = length; i > 0; i--)
{
(*res++) = ((*in++) * (1 << (-right_shifts)));
}
}
}
void WebRtcSpl_VectorBitShiftW32(int32_t *out_vector,
size_t vector_length,
const int32_t *in_vector,
int16_t right_shifts)
{
size_t i;
if (right_shifts > 0)
{
for (i = vector_length; i > 0; i--)
{
(*out_vector++) = ((*in_vector++) >> right_shifts);
}
} else
{
for (i = vector_length; i > 0; i--)
{
(*out_vector++) = ((*in_vector++) << (-right_shifts));
}
}
}
void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, size_t length,
const int32_t* in, int right_shifts) {
size_t i;
int32_t tmp_w32;
if (right_shifts >= 0) {
for (i = length; i > 0; i--) {
tmp_w32 = (*in++) >> right_shifts;
(*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
}
} else {
int left_shifts = -right_shifts;
for (i = length; i > 0; i--) {
tmp_w32 = (*in++) << left_shifts;
(*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
}
}
}
void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector,
int16_t gain, size_t in_vector_length,
int16_t right_shifts)
{
// Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
size_t i;
const int16_t *inptr;
int16_t *outptr;
inptr = in_vector;
outptr = out_vector;
for (i = 0; i < in_vector_length; i++)
{
*outptr++ = (int16_t)((*inptr++ * gain) >> right_shifts);
}
}
void WebRtcSpl_ScaleVectorWithSat(const int16_t *in_vector, int16_t *out_vector,
int16_t gain, size_t in_vector_length,
int16_t right_shifts)
{
// Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
size_t i;
const int16_t *inptr;
int16_t *outptr;
inptr = in_vector;
outptr = out_vector;
for (i = 0; i < in_vector_length; i++) {
*outptr++ = WebRtcSpl_SatW32ToW16((*inptr++ * gain) >> right_shifts);
}
}
void WebRtcSpl_ScaleAndAddVectors(const int16_t *in1, int16_t gain1, int shift1,
const int16_t *in2, int16_t gain2, int shift2,
int16_t *out, size_t vector_length)
{
// Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
size_t i;
const int16_t *in1ptr;
const int16_t *in2ptr;
int16_t *outptr;
in1ptr = in1;
in2ptr = in2;
outptr = out;
for (i = 0; i < vector_length; i++)
{
*outptr++ = (int16_t)((gain1 * *in1ptr++) >> shift1) +
(int16_t)((gain2 * *in2ptr++) >> shift2);
}
}
// C version of WebRtcSpl_ScaleAndAddVectorsWithRound() for generic platforms.
int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
int16_t in_vector1_scale,
const int16_t* in_vector2,
int16_t in_vector2_scale,
int right_shifts,
int16_t* out_vector,
size_t length) {
size_t i = 0;
int round_value = (1 << right_shifts) >> 1;
if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
length == 0 || right_shifts < 0) {
return -1;
}
for (i = 0; i < length; i++) {
out_vector[i] = (int16_t)((
in_vector1[i] * in_vector1_scale + in_vector2[i] * in_vector2_scale +
round_value) >> right_shifts);
}
return 0;
}

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/*
* Written by Wilco Dijkstra, 1996. The following email exchange establishes the
* license.
*
* From: Wilco Dijkstra <Wilco.Dijkstra@ntlworld.com>
* Date: Fri, Jun 24, 2011 at 3:20 AM
* Subject: Re: sqrt routine
* To: Kevin Ma <kma@google.com>
* Hi Kevin,
* Thanks for asking. Those routines are public domain (originally posted to
* comp.sys.arm a long time ago), so you can use them freely for any purpose.
* Cheers,
* Wilco
*
* ----- Original Message -----
* From: "Kevin Ma" <kma@google.com>
* To: <Wilco.Dijkstra@ntlworld.com>
* Sent: Thursday, June 23, 2011 11:44 PM
* Subject: Fwd: sqrt routine
* Hi Wilco,
* I saw your sqrt routine from several web sites, including
* http://www.finesse.demon.co.uk/steven/sqrt.html.
* Just wonder if there's any copyright information with your Successive
* approximation routines, or if I can freely use it for any purpose.
* Thanks.
* Kevin
*/
// Minor modifications in code style for WebRTC, 2012.
#include "webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h"
/*
* Algorithm:
* Successive approximation of the equation (root + delta) ^ 2 = N
* until delta < 1. If delta < 1 we have the integer part of SQRT (N).
* Use delta = 2^i for i = 15 .. 0.
*
* Output precision is 16 bits. Note for large input values (close to
* 0x7FFFFFFF), bit 15 (the highest bit of the low 16-bit half word)
* contains the MSB information (a non-sign value). Do with caution
* if you need to cast the output to int16_t type.
*
* If the input value is negative, it returns 0.
*/
#define WEBRTC_SPL_SQRT_ITER(N) \
try1 = root + (1 << (N)); \
if (value >= try1 << (N)) \
{ \
value -= try1 << (N); \
root |= 2 << (N); \
}
int32_t WebRtcSpl_SqrtFloor(int32_t value)
{
int32_t root = 0, try1;
WEBRTC_SPL_SQRT_ITER (15);
WEBRTC_SPL_SQRT_ITER (14);
WEBRTC_SPL_SQRT_ITER (13);
WEBRTC_SPL_SQRT_ITER (12);
WEBRTC_SPL_SQRT_ITER (11);
WEBRTC_SPL_SQRT_ITER (10);
WEBRTC_SPL_SQRT_ITER ( 9);
WEBRTC_SPL_SQRT_ITER ( 8);
WEBRTC_SPL_SQRT_ITER ( 7);
WEBRTC_SPL_SQRT_ITER ( 6);
WEBRTC_SPL_SQRT_ITER ( 5);
WEBRTC_SPL_SQRT_ITER ( 4);
WEBRTC_SPL_SQRT_ITER ( 3);
WEBRTC_SPL_SQRT_ITER ( 2);
WEBRTC_SPL_SQRT_ITER ( 1);
WEBRTC_SPL_SQRT_ITER ( 0);
return root >> 1;
}

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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdint.h>
//
// WebRtcSpl_SqrtFloor(...)
//
// Returns the square root of the input value |value|. The precision of this
// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
// If |value| is a negative number then 0 is returned.
//
// Algorithm:
//
// An iterative 4 cylce/bit routine
//
// Input:
// - value : Value to calculate sqrt of
//
// Return value : Result of the sqrt calculation
//
int32_t WebRtcSpl_SqrtFloor(int32_t value);

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This header file includes the VAD API calls. Specific function calls are
* given below.
*/
#ifndef COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT
#define COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
#include <stddef.h>
#include <stdint.h>
typedef struct WebRtcVadInst VadInst;
#ifdef __cplusplus
extern "C" {
#endif
// Creates an instance to the VAD structure.
VadInst* WebRtcVad_Create(void);
// Frees the dynamic memory of a specified VAD instance.
//
// - handle [i] : Pointer to VAD instance that should be freed.
void WebRtcVad_Free(VadInst* handle);
// Initializes a VAD instance.
//
// - handle [i/o] : Instance that should be initialized.
//
// returns : 0 - (OK),
// -1 - (null pointer or Default mode could not be set).
int WebRtcVad_Init(VadInst* handle);
// Sets the VAD operating mode. A more aggressive (higher mode) VAD is more
// restrictive in reporting speech. Put in other words the probability of being
// speech when the VAD returns 1 is increased with increasing mode. As a
// consequence also the missed detection rate goes up.
//
// - handle [i/o] : VAD instance.
// - mode [i] : Aggressiveness mode (0, 1, 2, or 3).
//
// returns : 0 - (OK),
// -1 - (null pointer, mode could not be set or the VAD instance
// has not been initialized).
int WebRtcVad_set_mode(VadInst* handle, int mode);
// Calculates a VAD decision for the |audio_frame|. For valid sampling rates
// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
//
// - handle [i/o] : VAD Instance. Needs to be initialized by
// WebRtcVad_Init() before call.
// - fs [i] : Sampling frequency (Hz): 8000, 16000, or 32000
// - audio_frame [i] : Audio frame buffer.
// - frame_length [i] : Length of audio frame buffer in number of samples.
//
// returns : 1 - (Active Voice),
// 0 - (Non-active Voice),
// -1 - (Error)
int WebRtcVad_Process(VadInst* handle,
int fs,
const int16_t* audio_frame,
size_t frame_length);
// Checks for valid combinations of |rate| and |frame_length|. We support 10,
// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
//
// - rate [i] : Sampling frequency (Hz).
// - frame_length [i] : Speech frame buffer length in number of samples.
//
// returns : 0 - (valid combination), -1 - (invalid combination)
int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
#ifdef __cplusplus
}
#endif
#endif // COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/vad/vad_core.h"
#include "webrtc/rtc_base/sanitizer.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/vad/vad_filterbank.h"
#include "webrtc/common_audio/vad/vad_gmm.h"
#include "webrtc/common_audio/vad/vad_sp.h"
// Spectrum Weighting
static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
static const int16_t kNoiseUpdateConst = 655; // Q15
static const int16_t kSpeechUpdateConst = 6554; // Q15
static const int16_t kBackEta = 154; // Q8
// Minimum difference between the two models, Q5
static const int16_t kMinimumDifference[kNumChannels] = {
544, 544, 576, 576, 576, 576 };
// Upper limit of mean value for speech model, Q7
static const int16_t kMaximumSpeech[kNumChannels] = {
11392, 11392, 11520, 11520, 11520, 11520 };
// Minimum value for mean value
static const int16_t kMinimumMean[kNumGaussians] = { 640, 768 };
// Upper limit of mean value for noise model, Q7
static const int16_t kMaximumNoise[kNumChannels] = {
9216, 9088, 8960, 8832, 8704, 8576 };
// Start values for the Gaussian models, Q7
// Weights for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataWeights[kTableSize] = {
34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
// Weights for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataWeights[kTableSize] = {
48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
// Means for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataMeans[kTableSize] = {
6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
// Means for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataMeans[kTableSize] = {
8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
};
// Stds for the two Gaussians for the six channels (noise)
static const int16_t kNoiseDataStds[kTableSize] = {
378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
// Stds for the two Gaussians for the six channels (speech)
static const int16_t kSpeechDataStds[kTableSize] = {
555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
// Constants used in GmmProbability().
//
// Maximum number of counted speech (VAD = 1) frames in a row.
static const int16_t kMaxSpeechFrames = 6;
// Minimum standard deviation for both speech and noise.
static const int16_t kMinStd = 384;
// Constants in WebRtcVad_InitCore().
// Default aggressiveness mode.
static const short kDefaultMode = 0;
static const int kInitCheck = 42;
// Constants used in WebRtcVad_set_mode_core().
//
// Thresholds for different frame lengths (10 ms, 20 ms and 30 ms).
//
// Mode 0, Quality.
static const int16_t kOverHangMax1Q[3] = { 8, 4, 3 };
static const int16_t kOverHangMax2Q[3] = { 14, 7, 5 };
static const int16_t kLocalThresholdQ[3] = { 24, 21, 24 };
static const int16_t kGlobalThresholdQ[3] = { 57, 48, 57 };
// Mode 1, Low bitrate.
static const int16_t kOverHangMax1LBR[3] = { 8, 4, 3 };
static const int16_t kOverHangMax2LBR[3] = { 14, 7, 5 };
static const int16_t kLocalThresholdLBR[3] = { 37, 32, 37 };
static const int16_t kGlobalThresholdLBR[3] = { 100, 80, 100 };
// Mode 2, Aggressive.
static const int16_t kOverHangMax1AGG[3] = { 6, 3, 2 };
static const int16_t kOverHangMax2AGG[3] = { 9, 5, 3 };
static const int16_t kLocalThresholdAGG[3] = { 82, 78, 82 };
static const int16_t kGlobalThresholdAGG[3] = { 285, 260, 285 };
// Mode 3, Very aggressive.
static const int16_t kOverHangMax1VAG[3] = { 6, 3, 2 };
static const int16_t kOverHangMax2VAG[3] = { 9, 5, 3 };
static const int16_t kLocalThresholdVAG[3] = { 94, 94, 94 };
static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
// Calculates the weighted average w.r.t. number of Gaussians. The |data| are
// updated with an |offset| before averaging.
//
// - data [i/o] : Data to average.
// - offset [i] : An offset added to |data|.
// - weights [i] : Weights used for averaging.
//
// returns : The weighted average.
static int32_t WeightedAverage(int16_t* data, int16_t offset,
const int16_t* weights) {
int k;
int32_t weighted_average = 0;
for (k = 0; k < kNumGaussians; k++) {
data[k * kNumChannels] += offset;
weighted_average += data[k * kNumChannels] * weights[k * kNumChannels];
}
return weighted_average;
}
// An s16 x s32 -> s32 multiplication that's allowed to overflow. (It's still
// undefined behavior, so not a good idea; this just makes UBSan ignore the
// violation, so that our old code can continue to do what it's always been
// doing.)
static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
OverflowingMulS16ByS32ToS32(int16_t a, int32_t b) {
return a * b;
}
// Calculates the probabilities for both speech and background noise using
// Gaussian Mixture Models (GMM). A hypothesis-test is performed to decide which
// type of signal is most probable.
//
// - self [i/o] : Pointer to VAD instance
// - features [i] : Feature vector of length |kNumChannels|
// = log10(energy in frequency band)
// - total_power [i] : Total power in audio frame.
// - frame_length [i] : Number of input samples
//
// - returns : the VAD decision (0 - noise, 1 - speech).
static int16_t GmmProbability(VadInstT* self, int16_t* features,
int16_t total_power, size_t frame_length) {
int channel, k;
int16_t feature_minimum;
int16_t h0, h1;
int16_t log_likelihood_ratio;
int16_t vadflag = 0;
int16_t shifts_h0, shifts_h1;
int16_t tmp_s16, tmp1_s16, tmp2_s16;
int16_t diff;
int gaussian;
int16_t nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
int16_t delt, ndelt;
int16_t maxspe, maxmu;
int16_t deltaN[kTableSize], deltaS[kTableSize];
int16_t ngprvec[kTableSize] = { 0 }; // Conditional probability = 0.
int16_t sgprvec[kTableSize] = { 0 }; // Conditional probability = 0.
int32_t h0_test, h1_test;
int32_t tmp1_s32, tmp2_s32;
int32_t sum_log_likelihood_ratios = 0;
int32_t noise_global_mean, speech_global_mean;
int32_t noise_probability[kNumGaussians], speech_probability[kNumGaussians];
int16_t overhead1, overhead2, individualTest, totalTest;
// Set various thresholds based on frame lengths (80, 160 or 240 samples).
if (frame_length == 80) {
overhead1 = self->over_hang_max_1[0];
overhead2 = self->over_hang_max_2[0];
individualTest = self->individual[0];
totalTest = self->total[0];
} else if (frame_length == 160) {
overhead1 = self->over_hang_max_1[1];
overhead2 = self->over_hang_max_2[1];
individualTest = self->individual[1];
totalTest = self->total[1];
} else {
overhead1 = self->over_hang_max_1[2];
overhead2 = self->over_hang_max_2[2];
individualTest = self->individual[2];
totalTest = self->total[2];
}
if (total_power > kMinEnergy) {
// The signal power of current frame is large enough for processing. The
// processing consists of two parts:
// 1) Calculating the likelihood of speech and thereby a VAD decision.
// 2) Updating the underlying model, w.r.t., the decision made.
// The detection scheme is an LRT with hypothesis
// H0: Noise
// H1: Speech
//
// We combine a global LRT with local tests, for each frequency sub-band,
// here defined as |channel|.
for (channel = 0; channel < kNumChannels; channel++) {
// For each channel we model the probability with a GMM consisting of
// |kNumGaussians|, with different means and standard deviations depending
// on H0 or H1.
h0_test = 0;
h1_test = 0;
for (k = 0; k < kNumGaussians; k++) {
gaussian = channel + k * kNumChannels;
// Probability under H0, that is, probability of frame being noise.
// Value given in Q27 = Q7 * Q20.
tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
self->noise_means[gaussian],
self->noise_stds[gaussian],
&deltaN[gaussian]);
noise_probability[k] = kNoiseDataWeights[gaussian] * tmp1_s32;
h0_test += noise_probability[k]; // Q27
// Probability under H1, that is, probability of frame being speech.
// Value given in Q27 = Q7 * Q20.
tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
self->speech_means[gaussian],
self->speech_stds[gaussian],
&deltaS[gaussian]);
speech_probability[k] = kSpeechDataWeights[gaussian] * tmp1_s32;
h1_test += speech_probability[k]; // Q27
}
// Calculate the log likelihood ratio: log2(Pr{X|H1} / Pr{X|H1}).
// Approximation:
// log2(Pr{X|H1} / Pr{X|H1}) = log2(Pr{X|H1}*2^Q) - log2(Pr{X|H1}*2^Q)
// = log2(h1_test) - log2(h0_test)
// = log2(2^(31-shifts_h1)*(1+b1))
// - log2(2^(31-shifts_h0)*(1+b0))
// = shifts_h0 - shifts_h1
// + log2(1+b1) - log2(1+b0)
// ~= shifts_h0 - shifts_h1
//
// Note that b0 and b1 are values less than 1, hence, 0 <= log2(1+b0) < 1.
// Further, b0 and b1 are independent and on the average the two terms
// cancel.
shifts_h0 = WebRtcSpl_NormW32(h0_test);
shifts_h1 = WebRtcSpl_NormW32(h1_test);
if (h0_test == 0) {
shifts_h0 = 31;
}
if (h1_test == 0) {
shifts_h1 = 31;
}
log_likelihood_ratio = shifts_h0 - shifts_h1;
// Update |sum_log_likelihood_ratios| with spectrum weighting. This is
// used for the global VAD decision.
sum_log_likelihood_ratios +=
(int32_t) (log_likelihood_ratio * kSpectrumWeight[channel]);
// Local VAD decision.
if ((log_likelihood_ratio * 4) > individualTest) {
vadflag = 1;
}
// TODO(bjornv): The conditional probabilities below are applied on the
// hard coded number of Gaussians set to two. Find a way to generalize.
// Calculate local noise probabilities used later when updating the GMM.
h0 = (int16_t) (h0_test >> 12); // Q15
if (h0 > 0) {
// High probability of noise. Assign conditional probabilities for each
// Gaussian in the GMM.
tmp1_s32 = (noise_probability[0] & 0xFFFFF000) << 2; // Q29
ngprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h0); // Q14
ngprvec[channel + kNumChannels] = 16384 - ngprvec[channel];
} else {
// Low noise probability. Assign conditional probability 1 to the first
// Gaussian and 0 to the rest (which is already set at initialization).
ngprvec[channel] = 16384;
}
// Calculate local speech probabilities used later when updating the GMM.
h1 = (int16_t) (h1_test >> 12); // Q15
if (h1 > 0) {
// High probability of speech. Assign conditional probabilities for each
// Gaussian in the GMM. Otherwise use the initialized values, i.e., 0.
tmp1_s32 = (speech_probability[0] & 0xFFFFF000) << 2; // Q29
sgprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h1); // Q14
sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel];
}
}
// Make a global VAD decision.
vadflag |= (sum_log_likelihood_ratios >= totalTest);
// Update the model parameters.
maxspe = 12800;
for (channel = 0; channel < kNumChannels; channel++) {
// Get minimum value in past which is used for long term correction in Q4.
feature_minimum = WebRtcVad_FindMinimum(self, features[channel], channel);
// Compute the "global" mean, that is the sum of the two means weighted.
noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
&kNoiseDataWeights[channel]);
tmp1_s16 = (int16_t) (noise_global_mean >> 6); // Q8
for (k = 0; k < kNumGaussians; k++) {
gaussian = channel + k * kNumChannels;
nmk = self->noise_means[gaussian];
smk = self->speech_means[gaussian];
nsk = self->noise_stds[gaussian];
ssk = self->speech_stds[gaussian];
// Update noise mean vector if the frame consists of noise only.
nmk2 = nmk;
if (!vadflag) {
// deltaN = (x-mu)/sigma^2
// ngprvec[k] = |noise_probability[k]| /
// (|noise_probability[0]| + |noise_probability[1]|)
// (Q14 * Q11 >> 11) = Q14.
delt = (int16_t)((ngprvec[gaussian] * deltaN[gaussian]) >> 11);
// Q7 + (Q14 * Q15 >> 22) = Q7.
nmk2 = nmk + (int16_t)((delt * kNoiseUpdateConst) >> 22);
}
// Long term correction of the noise mean.
// Q8 - Q8 = Q8.
ndelt = (feature_minimum << 4) - tmp1_s16;
// Q7 + (Q8 * Q8) >> 9 = Q7.
nmk3 = nmk2 + (int16_t)((ndelt * kBackEta) >> 9);
// Control that the noise mean does not drift to much.
tmp_s16 = (int16_t) ((k + 5) << 7);
if (nmk3 < tmp_s16) {
nmk3 = tmp_s16;
}
tmp_s16 = (int16_t) ((72 + k - channel) << 7);
if (nmk3 > tmp_s16) {
nmk3 = tmp_s16;
}
self->noise_means[gaussian] = nmk3;
if (vadflag) {
// Update speech mean vector:
// |deltaS| = (x-mu)/sigma^2
// sgprvec[k] = |speech_probability[k]| /
// (|speech_probability[0]| + |speech_probability[1]|)
// (Q14 * Q11) >> 11 = Q14.
delt = (int16_t)((sgprvec[gaussian] * deltaS[gaussian]) >> 11);
// Q14 * Q15 >> 21 = Q8.
tmp_s16 = (int16_t)((delt * kSpeechUpdateConst) >> 21);
// Q7 + (Q8 >> 1) = Q7. With rounding.
smk2 = smk + ((tmp_s16 + 1) >> 1);
// Control that the speech mean does not drift to much.
maxmu = maxspe + 640;
if (smk2 < kMinimumMean[k]) {
smk2 = kMinimumMean[k];
}
if (smk2 > maxmu) {
smk2 = maxmu;
}
self->speech_means[gaussian] = smk2; // Q7.
// (Q7 >> 3) = Q4. With rounding.
tmp_s16 = ((smk + 4) >> 3);
tmp_s16 = features[channel] - tmp_s16; // Q4
// (Q11 * Q4 >> 3) = Q12.
tmp1_s32 = (deltaS[gaussian] * tmp_s16) >> 3;
tmp2_s32 = tmp1_s32 - 4096;
tmp_s16 = sgprvec[gaussian] >> 2;
// (Q14 >> 2) * Q12 = Q24.
tmp1_s32 = tmp_s16 * tmp2_s32;
tmp2_s32 = tmp1_s32 >> 4; // Q20
// 0.1 * Q20 / Q7 = Q13.
if (tmp2_s32 > 0) {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp2_s32, ssk * 10);
} else {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp2_s32, ssk * 10);
tmp_s16 = -tmp_s16;
}
// Divide by 4 giving an update factor of 0.025 (= 0.1 / 4).
// Note that division by 4 equals shift by 2, hence,
// (Q13 >> 8) = (Q13 >> 6) / 4 = Q7.
tmp_s16 += 128; // Rounding.
ssk += (tmp_s16 >> 8);
if (ssk < kMinStd) {
ssk = kMinStd;
}
self->speech_stds[gaussian] = ssk;
} else {
// Update GMM variance vectors.
// deltaN * (features[channel] - nmk) - 1
// Q4 - (Q7 >> 3) = Q4.
tmp_s16 = features[channel] - (nmk >> 3);
// (Q11 * Q4 >> 3) = Q12.
tmp1_s32 = (deltaN[gaussian] * tmp_s16) >> 3;
tmp1_s32 -= 4096;
// (Q14 >> 2) * Q12 = Q24.
tmp_s16 = (ngprvec[gaussian] + 2) >> 2;
tmp2_s32 = OverflowingMulS16ByS32ToS32(tmp_s16, tmp1_s32);
// Q20 * approx 0.001 (2^-10=0.0009766), hence,
// (Q24 >> 14) = (Q24 >> 4) / 2^10 = Q20.
tmp1_s32 = tmp2_s32 >> 14;
// Q20 / Q7 = Q13.
if (tmp1_s32 > 0) {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, nsk);
} else {
tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp1_s32, nsk);
tmp_s16 = -tmp_s16;
}
tmp_s16 += 32; // Rounding
nsk += tmp_s16 >> 6; // Q13 >> 6 = Q7.
if (nsk < kMinStd) {
nsk = kMinStd;
}
self->noise_stds[gaussian] = nsk;
}
}
// Separate models if they are too close.
// |noise_global_mean| in Q14 (= Q7 * Q7).
noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
&kNoiseDataWeights[channel]);
// |speech_global_mean| in Q14 (= Q7 * Q7).
speech_global_mean = WeightedAverage(&self->speech_means[channel], 0,
&kSpeechDataWeights[channel]);
// |diff| = "global" speech mean - "global" noise mean.
// (Q14 >> 9) - (Q14 >> 9) = Q5.
diff = (int16_t) (speech_global_mean >> 9) -
(int16_t) (noise_global_mean >> 9);
if (diff < kMinimumDifference[channel]) {
tmp_s16 = kMinimumDifference[channel] - diff;
// |tmp1_s16| = ~0.8 * (kMinimumDifference - diff) in Q7.
// |tmp2_s16| = ~0.2 * (kMinimumDifference - diff) in Q7.
tmp1_s16 = (int16_t)((13 * tmp_s16) >> 2);
tmp2_s16 = (int16_t)((3 * tmp_s16) >> 2);
// Move Gaussian means for speech model by |tmp1_s16| and update
// |speech_global_mean|. Note that |self->speech_means[channel]| is
// changed after the call.
speech_global_mean = WeightedAverage(&self->speech_means[channel],
tmp1_s16,
&kSpeechDataWeights[channel]);
// Move Gaussian means for noise model by -|tmp2_s16| and update
// |noise_global_mean|. Note that |self->noise_means[channel]| is
// changed after the call.
noise_global_mean = WeightedAverage(&self->noise_means[channel],
-tmp2_s16,
&kNoiseDataWeights[channel]);
}
// Control that the speech & noise means do not drift to much.
maxspe = kMaximumSpeech[channel];
tmp2_s16 = (int16_t) (speech_global_mean >> 7);
if (tmp2_s16 > maxspe) {
// Upper limit of speech model.
tmp2_s16 -= maxspe;
for (k = 0; k < kNumGaussians; k++) {
self->speech_means[channel + k * kNumChannels] -= tmp2_s16;
}
}
tmp2_s16 = (int16_t) (noise_global_mean >> 7);
if (tmp2_s16 > kMaximumNoise[channel]) {
tmp2_s16 -= kMaximumNoise[channel];
for (k = 0; k < kNumGaussians; k++) {
self->noise_means[channel + k * kNumChannels] -= tmp2_s16;
}
}
}
self->frame_counter++;
}
// Smooth with respect to transition hysteresis.
if (!vadflag) {
if (self->over_hang > 0) {
vadflag = 2 + self->over_hang;
self->over_hang--;
}
self->num_of_speech = 0;
} else {
self->num_of_speech++;
if (self->num_of_speech > kMaxSpeechFrames) {
self->num_of_speech = kMaxSpeechFrames;
self->over_hang = overhead2;
} else {
self->over_hang = overhead1;
}
}
return vadflag;
}
// Initialize the VAD. Set aggressiveness mode to default value.
int WebRtcVad_InitCore(VadInstT* self) {
int i;
if (self == NULL) {
return -1;
}
// Initialization of general struct variables.
self->vad = 1; // Speech active (=1).
self->frame_counter = 0;
self->over_hang = 0;
self->num_of_speech = 0;
// Initialization of downsampling filter state.
memset(self->downsampling_filter_states, 0,
sizeof(self->downsampling_filter_states));
// Initialization of 48 to 8 kHz downsampling.
WebRtcSpl_ResetResample48khzTo8khz(&self->state_48_to_8);
// Read initial PDF parameters.
for (i = 0; i < kTableSize; i++) {
self->noise_means[i] = kNoiseDataMeans[i];
self->speech_means[i] = kSpeechDataMeans[i];
self->noise_stds[i] = kNoiseDataStds[i];
self->speech_stds[i] = kSpeechDataStds[i];
}
// Initialize Index and Minimum value vectors.
for (i = 0; i < 16 * kNumChannels; i++) {
self->low_value_vector[i] = 10000;
self->index_vector[i] = 0;
}
// Initialize splitting filter states.
memset(self->upper_state, 0, sizeof(self->upper_state));
memset(self->lower_state, 0, sizeof(self->lower_state));
// Initialize high pass filter states.
memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
// Initialize mean value memory, for WebRtcVad_FindMinimum().
for (i = 0; i < kNumChannels; i++) {
self->mean_value[i] = 1600;
}
// Set aggressiveness mode to default (=|kDefaultMode|).
if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
return -1;
}
self->init_flag = kInitCheck;
return 0;
}
// Set aggressiveness mode
int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
int return_value = 0;
switch (mode) {
case 0:
// Quality mode.
memcpy(self->over_hang_max_1, kOverHangMax1Q,
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2Q,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdQ,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdQ,
sizeof(self->total));
break;
case 1:
// Low bitrate mode.
memcpy(self->over_hang_max_1, kOverHangMax1LBR,
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2LBR,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdLBR,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdLBR,
sizeof(self->total));
break;
case 2:
// Aggressive mode.
memcpy(self->over_hang_max_1, kOverHangMax1AGG,
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2AGG,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdAGG,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdAGG,
sizeof(self->total));
break;
case 3:
// Very aggressive mode.
memcpy(self->over_hang_max_1, kOverHangMax1VAG,
sizeof(self->over_hang_max_1));
memcpy(self->over_hang_max_2, kOverHangMax2VAG,
sizeof(self->over_hang_max_2));
memcpy(self->individual, kLocalThresholdVAG,
sizeof(self->individual));
memcpy(self->total, kGlobalThresholdVAG,
sizeof(self->total));
break;
default:
return_value = -1;
break;
}
return return_value;
}
// Calculate VAD decision by first extracting feature values and then calculate
// probability for both speech and background noise.
int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length) {
int vad;
size_t i;
int16_t speech_nb[240]; // 30 ms in 8 kHz.
// |tmp_mem| is a temporary memory used by resample function, length is
// frame length in 10 ms (480 samples) + 256 extra.
int32_t tmp_mem[480 + 256] = { 0 };
const size_t kFrameLen10ms48khz = 480;
const size_t kFrameLen10ms8khz = 80;
size_t num_10ms_frames = frame_length / kFrameLen10ms48khz;
for (i = 0; i < num_10ms_frames; i++) {
WebRtcSpl_Resample48khzTo8khz(speech_frame,
&speech_nb[i * kFrameLen10ms8khz],
&inst->state_48_to_8,
tmp_mem);
}
// Do VAD on an 8 kHz signal
vad = WebRtcVad_CalcVad8khz(inst, speech_nb, frame_length / 6);
return vad;
}
int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
size_t len;
int vad;
int16_t speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
// Downsample signal 32->16->8 before doing VAD
WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
frame_length);
len = frame_length / 2;
WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
len /= 2;
// Do VAD on an 8 kHz signal
vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
return vad;
}
int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
size_t len;
int vad;
int16_t speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
// Wideband: Downsample signal before doing VAD
WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
frame_length);
len = frame_length / 2;
vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
return vad;
}
int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
size_t frame_length)
{
int16_t feature_vector[kNumChannels], total_power;
// Get power in the bands
total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
feature_vector);
// Make a VAD
inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
return inst->vad;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This header file includes the descriptions of the core VAD calls.
*/
#ifndef COMMON_AUDIO_VAD_VAD_CORE_H_
#define COMMON_AUDIO_VAD_VAD_CORE_H_
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
enum { kNumChannels = 6 }; // Number of frequency bands (named channels).
enum { kNumGaussians = 2 }; // Number of Gaussians per channel in the GMM.
enum { kTableSize = kNumChannels * kNumGaussians };
enum { kMinEnergy = 10 }; // Minimum energy required to trigger audio signal.
typedef struct VadInstT_ {
int vad;
int32_t downsampling_filter_states[4];
WebRtcSpl_State48khzTo8khz state_48_to_8;
int16_t noise_means[kTableSize];
int16_t speech_means[kTableSize];
int16_t noise_stds[kTableSize];
int16_t speech_stds[kTableSize];
// TODO(bjornv): Change to |frame_count|.
int32_t frame_counter;
int16_t over_hang; // Over Hang
int16_t num_of_speech;
// TODO(bjornv): Change to |age_vector|.
int16_t index_vector[16 * kNumChannels];
int16_t low_value_vector[16 * kNumChannels];
// TODO(bjornv): Change to |median|.
int16_t mean_value[kNumChannels];
int16_t upper_state[5];
int16_t lower_state[5];
int16_t hp_filter_state[4];
int16_t over_hang_max_1[3];
int16_t over_hang_max_2[3];
int16_t individual[3];
int16_t total[3];
int init_flag;
} VadInstT;
// Initializes the core VAD component. The default aggressiveness mode is
// controlled by |kDefaultMode| in vad_core.c.
//
// - self [i/o] : Instance that should be initialized
//
// returns : 0 (OK), -1 (null pointer in or if the default mode can't be
// set)
int WebRtcVad_InitCore(VadInstT* self);
/****************************************************************************
* WebRtcVad_set_mode_core(...)
*
* This function changes the VAD settings
*
* Input:
* - inst : VAD instance
* - mode : Aggressiveness degree
* 0 (High quality) - 3 (Highly aggressive)
*
* Output:
* - inst : Changed instance
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcVad_set_mode_core(VadInstT* self, int mode);
/****************************************************************************
* WebRtcVad_CalcVad48khz(...)
* WebRtcVad_CalcVad32khz(...)
* WebRtcVad_CalcVad16khz(...)
* WebRtcVad_CalcVad8khz(...)
*
* Calculate probability for active speech and make VAD decision.
*
* Input:
* - inst : Instance that should be initialized
* - speech_frame : Input speech frame
* - frame_length : Number of input samples
*
* Output:
* - inst : Updated filter states etc.
*
* Return value : VAD decision
* 0 - No active speech
* 1-6 - Active speech
*/
int WebRtcVad_CalcVad48khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length);
int WebRtcVad_CalcVad32khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length);
int WebRtcVad_CalcVad16khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length);
int WebRtcVad_CalcVad8khz(VadInstT* inst,
const int16_t* speech_frame,
size_t frame_length);
#endif // COMMON_AUDIO_VAD_VAD_CORE_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/vad/vad_filterbank.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
// Constants used in LogOfEnergy().
static const int16_t kLogConst = 24660; // 160*log10(2) in Q9.
static const int16_t kLogEnergyIntPart = 14336; // 14 in Q10
// Coefficients used by HighPassFilter, Q14.
static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
// Allpass filter coefficients, upper and lower, in Q15.
// Upper: 0.64, Lower: 0.17
static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
// Adjustment for division with two in SplitFilter.
static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
// sampled at 500 Hz.
//
// - data_in [i] : Input audio data sampled at 500 Hz.
// - data_length [i] : Length of input and output data.
// - filter_state [i/o] : State of the filter.
// - data_out [o] : Output audio data in the frequency interval
// 80 - 250 Hz.
static void HighPassFilter(const int16_t* data_in, size_t data_length,
int16_t* filter_state, int16_t* data_out) {
size_t i;
const int16_t* in_ptr = data_in;
int16_t* out_ptr = data_out;
int32_t tmp32 = 0;
// The sum of the absolute values of the impulse response:
// The zero/pole-filter has a max amplification of a single sample of: 1.4546
// Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
// The all-zero section has a max amplification of a single sample of: 1.6189
// Impulse response: 0.4047 -0.8094 0.4047 0 0 0
// The all-pole section has a max amplification of a single sample of: 1.9931
// Impulse response: 1.0000 0.4734 -0.1189 -0.2187 -0.0627 0.04532
for (i = 0; i < data_length; i++) {
// All-zero section (filter coefficients in Q14).
tmp32 = kHpZeroCoefs[0] * *in_ptr;
tmp32 += kHpZeroCoefs[1] * filter_state[0];
tmp32 += kHpZeroCoefs[2] * filter_state[1];
filter_state[1] = filter_state[0];
filter_state[0] = *in_ptr++;
// All-pole section (filter coefficients in Q14).
tmp32 -= kHpPoleCoefs[1] * filter_state[2];
tmp32 -= kHpPoleCoefs[2] * filter_state[3];
filter_state[3] = filter_state[2];
filter_state[2] = (int16_t) (tmp32 >> 14);
*out_ptr++ = filter_state[2];
}
}
// All pass filtering of |data_in|, used before splitting the signal into two
// frequency bands (low pass vs high pass).
// Note that |data_in| and |data_out| can NOT correspond to the same address.
//
// - data_in [i] : Input audio signal given in Q0.
// - data_length [i] : Length of input and output data.
// - filter_coefficient [i] : Given in Q15.
// - filter_state [i/o] : State of the filter given in Q(-1).
// - data_out [o] : Output audio signal given in Q(-1).
static void AllPassFilter(const int16_t* data_in, size_t data_length,
int16_t filter_coefficient, int16_t* filter_state,
int16_t* data_out) {
// The filter can only cause overflow (in the w16 output variable)
// if more than 4 consecutive input numbers are of maximum value and
// has the the same sign as the impulse responses first taps.
// First 6 taps of the impulse response:
// 0.6399 0.5905 -0.3779 0.2418 -0.1547 0.0990
size_t i;
int16_t tmp16 = 0;
int32_t tmp32 = 0;
int32_t state32 = ((int32_t) (*filter_state) * (1 << 16)); // Q15
for (i = 0; i < data_length; i++) {
tmp32 = state32 + filter_coefficient * *data_in;
tmp16 = (int16_t) (tmp32 >> 16); // Q(-1)
*data_out++ = tmp16;
state32 = (*data_in * (1 << 14)) - filter_coefficient * tmp16; // Q14
state32 *= 2; // Q15.
data_in += 2;
}
*filter_state = (int16_t) (state32 >> 16); // Q(-1)
}
// Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to
// an upper (high pass) part and a lower (low pass) part respectively.
//
// - data_in [i] : Input audio data to be split into two frequency bands.
// - data_length [i] : Length of |data_in|.
// - upper_state [i/o] : State of the upper filter, given in Q(-1).
// - lower_state [i/o] : State of the lower filter, given in Q(-1).
// - hp_data_out [o] : Output audio data of the upper half of the spectrum.
// The length is |data_length| / 2.
// - lp_data_out [o] : Output audio data of the lower half of the spectrum.
// The length is |data_length| / 2.
static void SplitFilter(const int16_t* data_in, size_t data_length,
int16_t* upper_state, int16_t* lower_state,
int16_t* hp_data_out, int16_t* lp_data_out) {
size_t i;
size_t half_length = data_length >> 1; // Downsampling by 2.
int16_t tmp_out;
// All-pass filtering upper branch.
AllPassFilter(&data_in[0], half_length, kAllPassCoefsQ15[0], upper_state,
hp_data_out);
// All-pass filtering lower branch.
AllPassFilter(&data_in[1], half_length, kAllPassCoefsQ15[1], lower_state,
lp_data_out);
// Make LP and HP signals.
for (i = 0; i < half_length; i++) {
tmp_out = *hp_data_out;
*hp_data_out++ -= *lp_data_out;
*lp_data_out++ += tmp_out;
}
}
// Calculates the energy of |data_in| in dB, and also updates an overall
// |total_energy| if necessary.
//
// - data_in [i] : Input audio data for energy calculation.
// - data_length [i] : Length of input data.
// - offset [i] : Offset value added to |log_energy|.
// - total_energy [i/o] : An external energy updated with the energy of
// |data_in|.
// NOTE: |total_energy| is only updated if
// |total_energy| <= |kMinEnergy|.
// - log_energy [o] : 10 * log10("energy of |data_in|") given in Q4.
static void LogOfEnergy(const int16_t* data_in, size_t data_length,
int16_t offset, int16_t* total_energy,
int16_t* log_energy) {
// |tot_rshifts| accumulates the number of right shifts performed on |energy|.
int tot_rshifts = 0;
// The |energy| will be normalized to 15 bits. We use unsigned integer because
// we eventually will mask out the fractional part.
uint32_t energy = 0;
RTC_DCHECK(data_in);
RTC_DCHECK_GT(data_length, 0);
energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
&tot_rshifts);
if (energy != 0) {
// By construction, normalizing to 15 bits is equivalent with 17 leading
// zeros of an unsigned 32 bit value.
int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
// In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
// (14 << 10), which is what we initialize |log2_energy| with. For a more
// detailed derivations, see below.
int16_t log2_energy = kLogEnergyIntPart;
tot_rshifts += normalizing_rshifts;
// Normalize |energy| to 15 bits.
// |tot_rshifts| is now the total number of right shifts performed on
// |energy| after normalization. This means that |energy| is in
// Q(-tot_rshifts).
if (normalizing_rshifts < 0) {
energy <<= -normalizing_rshifts;
} else {
energy >>= normalizing_rshifts;
}
// Calculate the energy of |data_in| in dB, in Q4.
//
// 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
// 160 * log10(|energy| * 2^|tot_rshifts|) =
// 160 * log10(2) * log2(|energy| * 2^|tot_rshifts|) =
// 160 * log10(2) * (log2(|energy|) + log2(2^|tot_rshifts|)) =
// (160 * log10(2)) * (log2(|energy|) + |tot_rshifts|) =
// |kLogConst| * (|log2_energy| + |tot_rshifts|)
//
// We know by construction that |energy| is normalized to 15 bits. Hence,
// |energy| = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
// Further, we'd like |log2_energy| in Q10
// log2(|energy|) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
// 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
// 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
// (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
// (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
//
// Note that frac_Q15 = (|energy| & 0x00003FFF)
// Calculate and add the fractional part to |log2_energy|.
log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
// |kLogConst| is in Q9, |log2_energy| in Q10 and |tot_rshifts| in Q0.
// Note that we in our derivation above have accounted for an output in Q4.
*log_energy = (int16_t)(((kLogConst * log2_energy) >> 19) +
((tot_rshifts * kLogConst) >> 9));
if (*log_energy < 0) {
*log_energy = 0;
}
} else {
*log_energy = offset;
return;
}
*log_energy += offset;
// Update the approximate |total_energy| with the energy of |data_in|, if
// |total_energy| has not exceeded |kMinEnergy|. |total_energy| is used as an
// energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
if (*total_energy <= kMinEnergy) {
if (tot_rshifts >= 0) {
// We know by construction that the |energy| > |kMinEnergy| in Q0, so add
// an arbitrary value such that |total_energy| exceeds |kMinEnergy|.
*total_energy += kMinEnergy + 1;
} else {
// By construction |energy| is represented by 15 bits, hence any number of
// right shifted |energy| will fit in an int16_t. In addition, adding the
// value to |total_energy| is wrap around safe as long as
// |kMinEnergy| < 8192.
*total_energy += (int16_t) (energy >> -tot_rshifts); // Q0.
}
}
}
int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
size_t data_length, int16_t* features) {
int16_t total_energy = 0;
// We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
// 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
// have at most 120 samples after the first split and at most 60 samples after
// the second split.
int16_t hp_120[120], lp_120[120];
int16_t hp_60[60], lp_60[60];
const size_t half_data_length = data_length >> 1;
size_t length = half_data_length; // |data_length| / 2, corresponds to
// bandwidth = 2000 Hz after downsampling.
// Initialize variables for the first SplitFilter().
int frequency_band = 0;
const int16_t* in_ptr = data_in; // [0 - 4000] Hz.
int16_t* hp_out_ptr = hp_120; // [2000 - 4000] Hz.
int16_t* lp_out_ptr = lp_120; // [0 - 2000] Hz.
RTC_DCHECK_LE(data_length, 240);
RTC_DCHECK_LT(4, kNumChannels - 1); // Checking maximum |frequency_band|.
// Split at 2000 Hz and downsample.
SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
frequency_band = 1;
in_ptr = hp_120; // [2000 - 4000] Hz.
hp_out_ptr = hp_60; // [3000 - 4000] Hz.
lp_out_ptr = lp_60; // [2000 - 3000] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 3000 Hz - 4000 Hz.
length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
// Energy in 2000 Hz - 3000 Hz.
LogOfEnergy(lp_60, length, kOffsetVector[4], &total_energy, &features[4]);
// For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
frequency_band = 2;
in_ptr = lp_120; // [0 - 2000] Hz.
hp_out_ptr = hp_60; // [1000 - 2000] Hz.
lp_out_ptr = lp_60; // [0 - 1000] Hz.
length = half_data_length; // |data_length| / 2 <=> bandwidth = 2000 Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 1000 Hz - 2000 Hz.
length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
// For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
frequency_band = 3;
in_ptr = lp_60; // [0 - 1000] Hz.
hp_out_ptr = hp_120; // [500 - 1000] Hz.
lp_out_ptr = lp_120; // [0 - 500] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 500 Hz - 1000 Hz.
length >>= 1; // |data_length| / 8 <=> bandwidth = 500 Hz.
LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
// For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
frequency_band = 4;
in_ptr = lp_120; // [0 - 500] Hz.
hp_out_ptr = hp_60; // [250 - 500] Hz.
lp_out_ptr = lp_60; // [0 - 250] Hz.
SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
&self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
// Energy in 250 Hz - 500 Hz.
length >>= 1; // |data_length| / 16 <=> bandwidth = 250 Hz.
LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
// Remove 0 Hz - 80 Hz, by high pass filtering the lower band.
HighPassFilter(lp_60, length, self->hp_filter_state, hp_120);
// Energy in 80 Hz - 250 Hz.
LogOfEnergy(hp_120, length, kOffsetVector[0], &total_energy, &features[0]);
return total_energy;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file includes feature calculating functionality used in vad_core.c.
*/
#ifndef COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
#define COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
#include "webrtc/common_audio/vad/vad_core.h"
// Takes |data_length| samples of |data_in| and calculates the logarithm of the
// energy of each of the |kNumChannels| = 6 frequency bands used by the VAD:
// 80 Hz - 250 Hz
// 250 Hz - 500 Hz
// 500 Hz - 1000 Hz
// 1000 Hz - 2000 Hz
// 2000 Hz - 3000 Hz
// 3000 Hz - 4000 Hz
//
// The values are given in Q4 and written to |features|. Further, an approximate
// overall energy is returned. The return value is used in
// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
// the threshold |kMinEnergy|.
//
// - self [i/o] : State information of the VAD.
// - data_in [i] : Input audio data, for feature extraction.
// - data_length [i] : Audio data size, in number of samples.
// - features [o] : 10 * log10(energy in each frequency band), Q4.
// - returns : Total energy of the signal (NOTE! This value is not
// exact. It is only used in a comparison.)
int16_t WebRtcVad_CalculateFeatures(VadInstT* self,
const int16_t* data_in,
size_t data_length,
int16_t* features);
#endif // COMMON_AUDIO_VAD_VAD_FILTERBANK_H_

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/vad/vad_gmm.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
static const int32_t kCompVar = 22005;
static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.
// For a normal distribution, the probability of |input| is calculated and
// returned (in Q20). The formula for normal distributed probability is
//
// 1 / s * exp(-(x - m)^2 / (2 * s^2))
//
// where the parameters are given in the following Q domains:
// m = |mean| (Q7)
// s = |std| (Q7)
// x = |input| (Q4)
// in addition to the probability we output |delta| (in Q11) used when updating
// the noise/speech model.
int32_t WebRtcVad_GaussianProbability(int16_t input,
int16_t mean,
int16_t std,
int16_t* delta) {
int16_t tmp16, inv_std, inv_std2, exp_value = 0;
int32_t tmp32;
// Calculate |inv_std| = 1 / s, in Q10.
// 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
// Q-domain: Q17 / Q7 = Q10.
tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
// Calculate |inv_std2| = 1 / s^2, in Q14.
tmp16 = (inv_std >> 2); // Q10 -> Q8.
// Q-domain: (Q8 * Q8) >> 2 = Q14.
inv_std2 = (int16_t)((tmp16 * tmp16) >> 2);
// TODO(bjornv): Investigate if changing to
// inv_std2 = (int16_t)((inv_std * inv_std) >> 6);
// gives better accuracy.
tmp16 = (input << 3); // Q4 -> Q7
tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
// To be used later, when updating noise/speech model.
// |delta| = (x - m) / s^2, in Q11.
// Q-domain: (Q14 * Q7) >> 10 = Q11.
*delta = (int16_t)((inv_std2 * tmp16) >> 10);
// Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
// division by two with one shift.
// Q-domain: (Q11 * Q7) >> 8 = Q10.
tmp32 = (*delta * tmp16) >> 9;
// If the exponent is small enough to give a non-zero probability we calculate
// |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
// ~= exp2(-log2(exp(1)) * |tmp32|).
if (tmp32 < kCompVar) {
// Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
// Q-domain: (Q12 * Q10) >> 12 = Q10.
tmp16 = (int16_t)((kLog2Exp * tmp32) >> 12);
tmp16 = -tmp16;
exp_value = (0x0400 | (tmp16 & 0x03FF));
tmp16 ^= 0xFFFF;
tmp16 >>= 10;
tmp16 += 1;
// Get |exp_value| = exp(-|tmp32|) in Q10.
exp_value >>= tmp16;
}
// Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
// Q-domain: Q10 * Q10 = Q20.
return inv_std * exp_value;
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Gaussian probability calculations internally used in vad_core.c.
#ifndef COMMON_AUDIO_VAD_VAD_GMM_H_
#define COMMON_AUDIO_VAD_VAD_GMM_H_
#include <stdint.h>
// Calculates the probability for |input|, given that |input| comes from a
// normal distribution with mean and standard deviation (|mean|, |std|).
//
// Inputs:
// - input : input sample in Q4.
// - mean : mean input in the statistical model, Q7.
// - std : standard deviation, Q7.
//
// Output:
//
// - delta : input used when updating the model, Q11.
// |delta| = (|input| - |mean|) / |std|^2.
//
// Return:
// (probability for |input|) =
// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
int32_t WebRtcVad_GaussianProbability(int16_t input,
int16_t mean,
int16_t std,
int16_t* delta);
#endif // COMMON_AUDIO_VAD_VAD_GMM_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/vad/vad_sp.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/vad/vad_core.h"
// Allpass filter coefficients, upper and lower, in Q13.
// Upper: 0.64, Lower: 0.17.
static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13.
static const int16_t kSmoothingDown = 6553; // 0.2 in Q15.
static const int16_t kSmoothingUp = 32439; // 0.99 in Q15.
// TODO(bjornv): Move this function to vad_filterbank.c.
// Downsampling filter based on splitting filter and allpass functions.
void WebRtcVad_Downsampling(const int16_t* signal_in,
int16_t* signal_out,
int32_t* filter_state,
size_t in_length) {
int16_t tmp16_1 = 0, tmp16_2 = 0;
int32_t tmp32_1 = filter_state[0];
int32_t tmp32_2 = filter_state[1];
size_t n = 0;
// Downsampling by 2 gives half length.
size_t half_length = (in_length >> 1);
// Filter coefficients in Q13, filter state in Q0.
for (n = 0; n < half_length; n++) {
// All-pass filtering upper branch.
tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
((kAllPassCoefsQ13[0] * *signal_in) >> 14));
*signal_out = tmp16_1;
tmp32_1 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[0] * tmp16_1) >> 12);
// All-pass filtering lower branch.
tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
((kAllPassCoefsQ13[1] * *signal_in) >> 14));
*signal_out++ += tmp16_2;
tmp32_2 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[1] * tmp16_2) >> 12);
}
// Store the filter states.
filter_state[0] = tmp32_1;
filter_state[1] = tmp32_2;
}
// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
// smallest values the last 100 frames. Then calculates and returns the median
// of the five smallest values.
int16_t WebRtcVad_FindMinimum(VadInstT* self,
int16_t feature_value,
int channel) {
int i = 0, j = 0;
int position = -1;
// Offset to beginning of the 16 minimum values in memory.
const int offset = (channel << 4);
int16_t current_median = 1600;
int16_t alpha = 0;
int32_t tmp32 = 0;
// Pointer to memory for the 16 minimum values and the age of each value of
// the |channel|.
int16_t* age = &self->index_vector[offset];
int16_t* smallest_values = &self->low_value_vector[offset];
RTC_DCHECK_LT(channel, kNumChannels);
// Each value in |smallest_values| is getting 1 loop older. Update |age|, and
// remove old values.
for (i = 0; i < 16; i++) {
if (age[i] != 100) {
age[i]++;
} else {
// Too old value. Remove from memory and shift larger values downwards.
for (j = i; j < 15; j++) {
smallest_values[j] = smallest_values[j + 1];
age[j] = age[j + 1];
}
age[15] = 101;
smallest_values[15] = 10000;
}
}
// Check if |feature_value| is smaller than any of the values in
// |smallest_values|. If so, find the |position| where to insert the new value
// (|feature_value|).
if (feature_value < smallest_values[7]) {
if (feature_value < smallest_values[3]) {
if (feature_value < smallest_values[1]) {
if (feature_value < smallest_values[0]) {
position = 0;
} else {
position = 1;
}
} else if (feature_value < smallest_values[2]) {
position = 2;
} else {
position = 3;
}
} else if (feature_value < smallest_values[5]) {
if (feature_value < smallest_values[4]) {
position = 4;
} else {
position = 5;
}
} else if (feature_value < smallest_values[6]) {
position = 6;
} else {
position = 7;
}
} else if (feature_value < smallest_values[15]) {
if (feature_value < smallest_values[11]) {
if (feature_value < smallest_values[9]) {
if (feature_value < smallest_values[8]) {
position = 8;
} else {
position = 9;
}
} else if (feature_value < smallest_values[10]) {
position = 10;
} else {
position = 11;
}
} else if (feature_value < smallest_values[13]) {
if (feature_value < smallest_values[12]) {
position = 12;
} else {
position = 13;
}
} else if (feature_value < smallest_values[14]) {
position = 14;
} else {
position = 15;
}
}
// If we have detected a new small value, insert it at the correct position
// and shift larger values up.
if (position > -1) {
for (i = 15; i > position; i--) {
smallest_values[i] = smallest_values[i - 1];
age[i] = age[i - 1];
}
smallest_values[position] = feature_value;
age[position] = 1;
}
// Get |current_median|.
if (self->frame_counter > 2) {
current_median = smallest_values[2];
} else if (self->frame_counter > 0) {
current_median = smallest_values[0];
}
// Smooth the median value.
if (self->frame_counter > 0) {
if (current_median < self->mean_value[channel]) {
alpha = kSmoothingDown; // 0.2 in Q15.
} else {
alpha = kSmoothingUp; // 0.99 in Q15.
}
}
tmp32 = (alpha + 1) * self->mean_value[channel];
tmp32 += (WEBRTC_SPL_WORD16_MAX - alpha) * current_median;
tmp32 += 16384;
self->mean_value[channel] = (int16_t) (tmp32 >> 15);
return self->mean_value[channel];
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file includes specific signal processing tools used in vad_core.c.
#ifndef COMMON_AUDIO_VAD_VAD_SP_H_
#define COMMON_AUDIO_VAD_VAD_SP_H_
#include "webrtc/common_audio/vad/vad_core.h"
// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
//
// Inputs:
// - signal_in : Input signal.
// - in_length : Length of input signal in samples.
//
// Input & Output:
// - filter_state : Current filter states of the two all-pass filters. The
// |filter_state| is updated after all samples have been
// processed.
//
// Output:
// - signal_out : Downsampled signal (of length |in_length| / 2).
void WebRtcVad_Downsampling(const int16_t* signal_in,
int16_t* signal_out,
int32_t* filter_state,
size_t in_length);
// Updates and returns the smoothed feature minimum. As minimum we use the
// median of the five smallest feature values in a 100 frames long window.
// As long as |handle->frame_counter| is zero, that is, we haven't received any
// "valid" data, FindMinimum() outputs the default value of 1600.
//
// Inputs:
// - feature_value : New feature value to update with.
// - channel : Channel number.
//
// Input & Output:
// - handle : State information of the VAD.
//
// Returns:
// : Smoothed minimum value for a moving window.
int16_t WebRtcVad_FindMinimum(VadInstT* handle,
int16_t feature_value,
int channel);
#endif // COMMON_AUDIO_VAD_VAD_SP_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include <stdlib.h>
#include <string.h>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_audio/vad/vad_core.h"
static const int kInitCheck = 42;
static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
static const size_t kRatesSize = sizeof(kValidRates) / sizeof(*kValidRates);
static const int kMaxFrameLengthMs = 30;
VadInst* WebRtcVad_Create() {
VadInstT* self = (VadInstT*)malloc(sizeof(VadInstT));
WebRtcSpl_Init();
self->init_flag = 0;
return (VadInst*)self;
}
void WebRtcVad_Free(VadInst* handle) {
free(handle);
}
// TODO(bjornv): Move WebRtcVad_InitCore() code here.
int WebRtcVad_Init(VadInst* handle) {
// Initialize the core VAD component.
return WebRtcVad_InitCore((VadInstT*) handle);
}
// TODO(bjornv): Move WebRtcVad_set_mode_core() code here.
int WebRtcVad_set_mode(VadInst* handle, int mode) {
VadInstT* self = (VadInstT*) handle;
if (handle == NULL) {
return -1;
}
if (self->init_flag != kInitCheck) {
return -1;
}
return WebRtcVad_set_mode_core(self, mode);
}
int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
size_t frame_length) {
int vad = -1;
VadInstT* self = (VadInstT*) handle;
if (handle == NULL) {
return -1;
}
if (self->init_flag != kInitCheck) {
return -1;
}
if (audio_frame == NULL) {
return -1;
}
if (WebRtcVad_ValidRateAndFrameLength(fs, frame_length) != 0) {
return -1;
}
if (fs == 48000) {
vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
} else if (fs == 32000) {
vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length);
} else if (fs == 16000) {
vad = WebRtcVad_CalcVad16khz(self, audio_frame, frame_length);
} else if (fs == 8000) {
vad = WebRtcVad_CalcVad8khz(self, audio_frame, frame_length);
}
if (vad > 0) {
vad = 1;
}
return vad;
}
int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length) {
int return_value = -1;
size_t i;
int valid_length_ms;
size_t valid_length;
// We only allow 10, 20 or 30 ms frames. Loop through valid frame rates and
// see if we have a matching pair.
for (i = 0; i < kRatesSize; i++) {
if (kValidRates[i] == rate) {
for (valid_length_ms = 10; valid_length_ms <= kMaxFrameLengthMs;
valid_length_ms += 10) {
valid_length = (size_t)(kValidRates[i] / 1000 * valid_length_ms);
if (frame_length == valid_length) {
return_value = 0;
break;
}
}
break;
}
}
return return_value;
}

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/*
* Copyright 2006 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Most of this was borrowed (with minor modifications) from V8's and Chromium's
// src/base/logging.cc.
#include <cstdarg>
#include <cstdio>
#include <cstdlib>
#if defined(WEBRTC_ANDROID)
#define RTC_LOG_TAG_ANDROID "rtc"
#include <android/log.h> // NOLINT
#endif
#if defined(WEBRTC_WIN)
#include <windows.h>
#endif
#include <errno.h>
#define LAST_SYSTEM_ERROR (errno)
// #if defined(WEBRTC_WIN)
// #define LAST_SYSTEM_ERROR (::GetLastError())
// #elif defined(__native_client__) && __native_client__
// #define LAST_SYSTEM_ERROR (0)
// #elif defined(WEBRTC_POSIX)
// #include <errno.h>
// #define LAST_SYSTEM_ERROR (errno)
// #endif // WEBRTC_WIN
#include "webrtc/rtc_base/checks.h"
namespace {
#if defined(__GNUC__)
__attribute__((__format__(__printf__, 2, 3)))
#endif
void AppendFormat(std::string* s, const char* fmt, ...) {
va_list args, copy;
va_start(args, fmt);
va_copy(copy, args);
const int predicted_length = std::vsnprintf(nullptr, 0, fmt, copy);
va_end(copy);
if (predicted_length > 0) {
const size_t size = s->size();
s->resize(size + predicted_length);
// Pass "+ 1" to vsnprintf to include space for the '\0'.
std::vsnprintf(&((*s)[size]), predicted_length + 1, fmt, args);
}
va_end(args);
}
}
namespace rtc {
namespace webrtc_checks_impl {
// Reads one argument from args, appends it to s and advances fmt.
// Returns true iff an argument was sucessfully parsed.
bool ParseArg(va_list* args, const CheckArgType** fmt, std::string* s) {
if (**fmt == CheckArgType::kEnd)
return false;
switch (**fmt) {
case CheckArgType::kInt:
AppendFormat(s, "%d", va_arg(*args, int));
break;
case CheckArgType::kLong:
AppendFormat(s, "%ld", va_arg(*args, long));
break;
case CheckArgType::kLongLong:
AppendFormat(s, "%lld", va_arg(*args, long long));
break;
case CheckArgType::kUInt:
AppendFormat(s, "%u", va_arg(*args, unsigned));
break;
case CheckArgType::kULong:
AppendFormat(s, "%lu", va_arg(*args, unsigned long));
break;
case CheckArgType::kULongLong:
AppendFormat(s, "%llu", va_arg(*args, unsigned long long));
break;
case CheckArgType::kDouble:
AppendFormat(s, "%g", va_arg(*args, double));
break;
case CheckArgType::kLongDouble:
AppendFormat(s, "%Lg", va_arg(*args, long double));
break;
case CheckArgType::kCharP:
s->append(va_arg(*args, const char*));
break;
case CheckArgType::kStdString:
s->append(*va_arg(*args, const std::string*));
break;
case CheckArgType::kVoidP:
AppendFormat(s, "%p", va_arg(*args, const void*));
break;
default:
s->append("[Invalid CheckArgType]");
return false;
}
(*fmt)++;
return true;
}
RTC_NORETURN void FatalLog(const char* file,
int line,
const char* message,
const CheckArgType* fmt,
...) {
va_list args;
va_start(args, fmt);
std::string s;
AppendFormat(&s,
"\n\n"
"#\n"
"# Fatal error in: %s, line %d\n"
"# last system error: %u\n"
"# Check failed: %s",
file, line, LAST_SYSTEM_ERROR, message);
if (*fmt == CheckArgType::kCheckOp) {
// This log message was generated by RTC_CHECK_OP, so we have to complete
// the error message using the operands that have been passed as the first
// two arguments.
fmt++;
std::string s1, s2;
if (ParseArg(&args, &fmt, &s1) && ParseArg(&args, &fmt, &s2))
AppendFormat(&s, " (%s vs. %s)\n# ", s1.c_str(), s2.c_str());
} else {
s.append("\n# ");
}
// Append all the user-supplied arguments to the message.
while (ParseArg(&args, &fmt, &s))
;
va_end(args);
const char* output = s.c_str();
#if defined(WEBRTC_ANDROID)
__android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n", output);
#endif
fflush(stdout);
fprintf(stderr, "%s", output);
fflush(stderr);
abort();
}
} // namespace webrtc_checks_impl
} // namespace rtc
// Function to call from the C version of the RTC_CHECK and RTC_DCHECK macros.
RTC_NORETURN void rtc_FatalMessage(const char* file, int line,
const char* msg) {
static constexpr rtc::webrtc_checks_impl::CheckArgType t[] = {
rtc::webrtc_checks_impl::CheckArgType::kEnd};
FatalLog(file, line, msg, t);
}

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/*
* Copyright 2006 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_CHECKS_H_
#define RTC_BASE_CHECKS_H_
// If you for some reson need to know if DCHECKs are on, test the value of
// RTC_DCHECK_IS_ON. (Test its value, not if it's defined; it'll always be
// defined, to either a true or a false value.)
#if !defined(NDEBUG) || defined(DCHECK_ALWAYS_ON)
#define RTC_DCHECK_IS_ON 1
#else
#define RTC_DCHECK_IS_ON 0
#endif
// Annotate a function that will not return control flow to the caller.
#if defined(_MSC_VER)
#define RTC_NORETURN __declspec(noreturn)
#elif defined(__GNUC__)
#define RTC_NORETURN __attribute__ ((__noreturn__))
#else
#define RTC_NORETURN
#endif
#ifdef __cplusplus
extern "C" {
#endif
RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
#ifdef __cplusplus
} // extern "C"
#endif
#ifdef __cplusplus
// C++ version.
#include <string>
#include "webrtc/rtc_base/numerics/safe_compare.h"
#include "webrtc/rtc_base/system/inline.h"
// The macros here print a message to stderr and abort under various
// conditions. All will accept additional stream messages. For example:
// RTC_DCHECK_EQ(foo, bar) << "I'm printed when foo != bar.";
//
// - RTC_CHECK(x) is an assertion that x is always true, and that if it isn't,
// it's better to terminate the process than to continue. During development,
// the reason that it's better to terminate might simply be that the error
// handling code isn't in place yet; in production, the reason might be that
// the author of the code truly believes that x will always be true, but that
// she recognizes that if she is wrong, abrupt and unpleasant process
// termination is still better than carrying on with the assumption violated.
//
// RTC_CHECK always evaluates its argument, so it's OK for x to have side
// effects.
//
// - RTC_DCHECK(x) is the same as RTC_CHECK(x)---an assertion that x is always
// true---except that x will only be evaluated in debug builds; in production
// builds, x is simply assumed to be true. This is useful if evaluating x is
// expensive and the expected cost of failing to detect the violated
// assumption is acceptable. You should not handle cases where a production
// build fails to spot a violated condition, even those that would result in
// crashes. If the code needs to cope with the error, make it cope, but don't
// call RTC_DCHECK; if the condition really can't occur, but you'd sleep
// better at night knowing that the process will suicide instead of carrying
// on in case you were wrong, use RTC_CHECK instead of RTC_DCHECK.
//
// RTC_DCHECK only evaluates its argument in debug builds, so if x has visible
// side effects, you need to write e.g.
// bool w = x; RTC_DCHECK(w);
//
// - RTC_CHECK_EQ, _NE, _GT, ..., and RTC_DCHECK_EQ, _NE, _GT, ... are
// specialized variants of RTC_CHECK and RTC_DCHECK that print prettier
// messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
// RTC_DCHECK.
//
// - FATAL() aborts unconditionally.
//
// TODO(ajm): Ideally, checks.h would be combined with logging.h, but
// consolidation with system_wrappers/logging.h should happen first.
namespace rtc {
namespace webrtc_checks_impl {
enum class CheckArgType : int8_t {
kEnd = 0,
kInt,
kLong,
kLongLong,
kUInt,
kULong,
kULongLong,
kDouble,
kLongDouble,
kCharP,
kStdString,
kVoidP,
// kCheckOp doesn't represent an argument type. Instead, it is sent as the
// first argument from RTC_CHECK_OP to make FatalLog use the next two
// arguments to build the special CHECK_OP error message
// (the "a == b (1 vs. 2)" bit).
kCheckOp,
};
RTC_NORETURN void FatalLog(const char* file,
int line,
const char* message,
const CheckArgType* fmt,
...);
// Wrapper for log arguments. Only ever make values of this type with the
// MakeVal() functions.
template <CheckArgType N, typename T>
struct Val {
static constexpr CheckArgType Type() { return N; }
T GetVal() const { return val; }
T val;
};
inline Val<CheckArgType::kInt, int> MakeVal(int x) {
return {x};
}
inline Val<CheckArgType::kLong, long> MakeVal(long x) {
return {x};
}
inline Val<CheckArgType::kLongLong, long long> MakeVal(long long x) {
return {x};
}
inline Val<CheckArgType::kUInt, unsigned int> MakeVal(unsigned int x) {
return {x};
}
inline Val<CheckArgType::kULong, unsigned long> MakeVal(unsigned long x) {
return {x};
}
inline Val<CheckArgType::kULongLong, unsigned long long> MakeVal(
unsigned long long x) {
return {x};
}
inline Val<CheckArgType::kDouble, double> MakeVal(double x) {
return {x};
}
inline Val<CheckArgType::kLongDouble, long double> MakeVal(long double x) {
return {x};
}
inline Val<CheckArgType::kCharP, const char*> MakeVal(const char* x) {
return {x};
}
inline Val<CheckArgType::kStdString, const std::string*> MakeVal(
const std::string& x) {
return {&x};
}
inline Val<CheckArgType::kVoidP, const void*> MakeVal(const void* x) {
return {x};
}
// Ephemeral type that represents the result of the logging << operator.
template <typename... Ts>
class LogStreamer;
// Base case: Before the first << argument.
template <>
class LogStreamer<> final {
public:
template <
typename U,
typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
U arg) const {
return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
this);
}
template <
typename U,
typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
const U& arg) const {
return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
this);
}
template <typename... Us>
RTC_NORETURN RTC_FORCE_INLINE static void Call(const char* file,
const int line,
const char* message,
const Us&... args) {
static constexpr CheckArgType t[] = {Us::Type()..., CheckArgType::kEnd};
FatalLog(file, line, message, t, args.GetVal()...);
}
template <typename... Us>
RTC_NORETURN RTC_FORCE_INLINE static void CallCheckOp(const char* file,
const int line,
const char* message,
const Us&... args) {
static constexpr CheckArgType t[] = {CheckArgType::kCheckOp, Us::Type()...,
CheckArgType::kEnd};
FatalLog(file, line, message, t, args.GetVal()...);
}
};
// Inductive case: We've already seen at least one << argument. The most recent
// one had type `T`, and the earlier ones had types `Ts`.
template <typename T, typename... Ts>
class LogStreamer<T, Ts...> final {
public:
RTC_FORCE_INLINE LogStreamer(T arg, const LogStreamer<Ts...>* prior)
: arg_(arg), prior_(prior) {}
template <
typename U,
typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
operator<<(U arg) const {
return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
MakeVal(arg), this);
}
template <
typename U,
typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
operator<<(const U& arg) const {
return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
MakeVal(arg), this);
}
template <typename... Us>
RTC_NORETURN RTC_FORCE_INLINE void Call(const char* file,
const int line,
const char* message,
const Us&... args) const {
prior_->Call(file, line, message, arg_, args...);
}
template <typename... Us>
RTC_NORETURN RTC_FORCE_INLINE void CallCheckOp(const char* file,
const int line,
const char* message,
const Us&... args) const {
prior_->CallCheckOp(file, line, message, arg_, args...);
}
private:
// The most recent argument.
T arg_;
// Earlier arguments.
const LogStreamer<Ts...>* prior_;
};
template <bool isCheckOp>
class FatalLogCall final {
public:
FatalLogCall(const char* file, int line, const char* message)
: file_(file), line_(line), message_(message) {}
// This can be any binary operator with precedence lower than <<.
template <typename... Ts>
RTC_NORETURN RTC_FORCE_INLINE void operator&(
const LogStreamer<Ts...>& streamer) {
isCheckOp ? streamer.CallCheckOp(file_, line_, message_)
: streamer.Call(file_, line_, message_);
}
private:
const char* file_;
int line_;
const char* message_;
};
} // namespace webrtc_checks_impl
// The actual stream used isn't important. We reference |ignored| in the code
// but don't evaluate it; this is to avoid "unused variable" warnings (we do so
// in a particularly convoluted way with an extra ?: because that appears to be
// the simplest construct that keeps Visual Studio from complaining about
// condition being unused).
#define RTC_EAT_STREAM_PARAMETERS(ignored) \
(true ? true : ((void)(ignored), true)) \
? static_cast<void>(0) \
: rtc::webrtc_checks_impl::FatalLogCall<false>("", 0, "") & \
rtc::webrtc_checks_impl::LogStreamer<>()
// Call RTC_EAT_STREAM_PARAMETERS with an argument that fails to compile if
// values of the same types as |a| and |b| can't be compared with the given
// operation, and that would evaluate |a| and |b| if evaluated.
#define RTC_EAT_STREAM_PARAMETERS_OP(op, a, b) \
RTC_EAT_STREAM_PARAMETERS(((void)rtc::Safe##op(a, b)))
// RTC_CHECK dies with a fatal error if condition is not true. It is *not*
// controlled by NDEBUG or anything else, so the check will be executed
// regardless of compilation mode.
//
// We make sure RTC_CHECK et al. always evaluates |condition|, as
// doing RTC_CHECK(FunctionWithSideEffect()) is a common idiom.
#define RTC_CHECK(condition) \
while (!(condition)) \
rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
#condition) & \
rtc::webrtc_checks_impl::LogStreamer<>()
// Helper macro for binary operators.
// Don't use this macro directly in your code, use RTC_CHECK_EQ et al below.
#define RTC_CHECK_OP(name, op, val1, val2) \
while (!rtc::Safe##name((val1), (val2))) \
rtc::webrtc_checks_impl::FatalLogCall<true>(__FILE__, __LINE__, \
#val1 " " #op " " #val2) & \
rtc::webrtc_checks_impl::LogStreamer<>() << (val1) << (val2)
#define RTC_CHECK_EQ(val1, val2) RTC_CHECK_OP(Eq, ==, val1, val2)
#define RTC_CHECK_NE(val1, val2) RTC_CHECK_OP(Ne, !=, val1, val2)
#define RTC_CHECK_LE(val1, val2) RTC_CHECK_OP(Le, <=, val1, val2)
#define RTC_CHECK_LT(val1, val2) RTC_CHECK_OP(Lt, <, val1, val2)
#define RTC_CHECK_GE(val1, val2) RTC_CHECK_OP(Ge, >=, val1, val2)
#define RTC_CHECK_GT(val1, val2) RTC_CHECK_OP(Gt, >, val1, val2)
// The RTC_DCHECK macro is equivalent to RTC_CHECK except that it only generates
// code in debug builds. It does reference the condition parameter in all cases,
// though, so callers won't risk getting warnings about unused variables.
#if RTC_DCHECK_IS_ON
#define RTC_DCHECK(condition) RTC_CHECK(condition)
#define RTC_DCHECK_EQ(v1, v2) RTC_CHECK_EQ(v1, v2)
#define RTC_DCHECK_NE(v1, v2) RTC_CHECK_NE(v1, v2)
#define RTC_DCHECK_LE(v1, v2) RTC_CHECK_LE(v1, v2)
#define RTC_DCHECK_LT(v1, v2) RTC_CHECK_LT(v1, v2)
#define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2)
#define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2)
#else
#define RTC_DCHECK(condition) RTC_EAT_STREAM_PARAMETERS(condition)
#define RTC_DCHECK_EQ(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Eq, v1, v2)
#define RTC_DCHECK_NE(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Ne, v1, v2)
#define RTC_DCHECK_LE(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Le, v1, v2)
#define RTC_DCHECK_LT(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Lt, v1, v2)
#define RTC_DCHECK_GE(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Ge, v1, v2)
#define RTC_DCHECK_GT(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Gt, v1, v2)
#endif
#define RTC_UNREACHABLE_CODE_HIT false
#define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
// TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
#define FATAL() \
rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
"FATAL()") & \
rtc::webrtc_checks_impl::LogStreamer<>()
// Performs the integer division a/b and returns the result. CHECKs that the
// remainder is zero.
template <typename T>
inline T CheckedDivExact(T a, T b) {
RTC_CHECK_EQ(a % b, 0) << a << " is not evenly divisible by " << b;
return a / b;
}
} // namespace rtc
#else // __cplusplus not defined
// C version. Lacks many features compared to the C++ version, but usage
// guidelines are the same.
#define RTC_CHECK(condition) \
do { \
if (!(condition)) { \
rtc_FatalMessage(__FILE__, __LINE__, "CHECK failed: " #condition); \
} \
} while (0)
#define RTC_CHECK_EQ(a, b) RTC_CHECK((a) == (b))
#define RTC_CHECK_NE(a, b) RTC_CHECK((a) != (b))
#define RTC_CHECK_LE(a, b) RTC_CHECK((a) <= (b))
#define RTC_CHECK_LT(a, b) RTC_CHECK((a) < (b))
#define RTC_CHECK_GE(a, b) RTC_CHECK((a) >= (b))
#define RTC_CHECK_GT(a, b) RTC_CHECK((a) > (b))
#define RTC_DCHECK(condition) \
do { \
if (RTC_DCHECK_IS_ON && !(condition)) { \
rtc_FatalMessage(__FILE__, __LINE__, "DCHECK failed: " #condition); \
} \
} while (0)
#define RTC_DCHECK_EQ(a, b) RTC_DCHECK((a) == (b))
#define RTC_DCHECK_NE(a, b) RTC_DCHECK((a) != (b))
#define RTC_DCHECK_LE(a, b) RTC_DCHECK((a) <= (b))
#define RTC_DCHECK_LT(a, b) RTC_DCHECK((a) < (b))
#define RTC_DCHECK_GE(a, b) RTC_DCHECK((a) >= (b))
#define RTC_DCHECK_GT(a, b) RTC_DCHECK((a) > (b))
#endif // __cplusplus
#endif // RTC_BASE_CHECKS_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_COMPILE_ASSERT_C_H_
#define RTC_BASE_COMPILE_ASSERT_C_H_
// Use this macro to verify at compile time that certain restrictions are met.
// The argument is the boolean expression to evaluate.
// Example:
// RTC_COMPILE_ASSERT(sizeof(foo) < 128);
// Note: In C++, use static_assert instead!
#define RTC_COMPILE_ASSERT(expression) \
switch (0) { \
case 0: \
case expression:; \
}
#endif // RTC_BASE_COMPILE_ASSERT_C_H_

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file defines six constexpr functions:
//
// rtc::SafeEq // ==
// rtc::SafeNe // !=
// rtc::SafeLt // <
// rtc::SafeLe // <=
// rtc::SafeGt // >
// rtc::SafeGe // >=
//
// They each accept two arguments of arbitrary types, and in almost all cases,
// they simply call the appropriate comparison operator. However, if both
// arguments are integers, they don't compare them using C++'s quirky rules,
// but instead adhere to the true mathematical definitions. It is as if the
// arguments were first converted to infinite-range signed integers, and then
// compared, although of course nothing expensive like that actually takes
// place. In practice, for signed/signed and unsigned/unsigned comparisons and
// some mixed-signed comparisons with a compile-time constant, the overhead is
// zero; in the remaining cases, it is just a few machine instructions (no
// branches).
#ifndef RTC_BASE_NUMERICS_SAFE_COMPARE_H_
#define RTC_BASE_NUMERICS_SAFE_COMPARE_H_
#include <stddef.h>
#include <stdint.h>
#include <type_traits>
#include <utility>
#include "webrtc/rtc_base/type_traits.h"
namespace rtc {
namespace safe_cmp_impl {
template <size_t N>
struct LargerIntImpl : std::false_type {};
template <>
struct LargerIntImpl<sizeof(int8_t)> : std::true_type {
using type = int16_t;
};
template <>
struct LargerIntImpl<sizeof(int16_t)> : std::true_type {
using type = int32_t;
};
template <>
struct LargerIntImpl<sizeof(int32_t)> : std::true_type {
using type = int64_t;
};
// LargerInt<T1, T2>::value is true iff there's a signed type that's larger
// than T1 (and no larger than the larger of T2 and int*, for performance
// reasons); and if there is such a type, LargerInt<T1, T2>::type is an alias
// for it.
template <typename T1, typename T2>
struct LargerInt
: LargerIntImpl<sizeof(T1) < sizeof(T2) || sizeof(T1) < sizeof(int*)
? sizeof(T1)
: 0> {};
template <typename T>
constexpr typename std::make_unsigned<T>::type MakeUnsigned(T a) {
return static_cast<typename std::make_unsigned<T>::type>(a);
}
// Overload for when both T1 and T2 have the same signedness.
template <typename Op,
typename T1,
typename T2,
typename std::enable_if<std::is_signed<T1>::value ==
std::is_signed<T2>::value>::type* = nullptr>
constexpr bool Cmp(T1 a, T2 b) {
return Op::Op(a, b);
}
// Overload for signed - unsigned comparison that can be promoted to a bigger
// signed type.
template <typename Op,
typename T1,
typename T2,
typename std::enable_if<std::is_signed<T1>::value &&
std::is_unsigned<T2>::value &&
LargerInt<T2, T1>::value>::type* = nullptr>
constexpr bool Cmp(T1 a, T2 b) {
return Op::Op(a, static_cast<typename LargerInt<T2, T1>::type>(b));
}
// Overload for unsigned - signed comparison that can be promoted to a bigger
// signed type.
template <typename Op,
typename T1,
typename T2,
typename std::enable_if<std::is_unsigned<T1>::value &&
std::is_signed<T2>::value &&
LargerInt<T1, T2>::value>::type* = nullptr>
constexpr bool Cmp(T1 a, T2 b) {
return Op::Op(static_cast<typename LargerInt<T1, T2>::type>(a), b);
}
// Overload for signed - unsigned comparison that can't be promoted to a bigger
// signed type.
template <typename Op,
typename T1,
typename T2,
typename std::enable_if<std::is_signed<T1>::value &&
std::is_unsigned<T2>::value &&
!LargerInt<T2, T1>::value>::type* = nullptr>
constexpr bool Cmp(T1 a, T2 b) {
return a < 0 ? Op::Op(-1, 0) : Op::Op(safe_cmp_impl::MakeUnsigned(a), b);
}
// Overload for unsigned - signed comparison that can't be promoted to a bigger
// signed type.
template <typename Op,
typename T1,
typename T2,
typename std::enable_if<std::is_unsigned<T1>::value &&
std::is_signed<T2>::value &&
!LargerInt<T1, T2>::value>::type* = nullptr>
constexpr bool Cmp(T1 a, T2 b) {
return b < 0 ? Op::Op(0, -1) : Op::Op(a, safe_cmp_impl::MakeUnsigned(b));
}
#define RTC_SAFECMP_MAKE_OP(name, op) \
struct name { \
template <typename T1, typename T2> \
static constexpr bool Op(T1 a, T2 b) { \
return a op b; \
} \
};
RTC_SAFECMP_MAKE_OP(EqOp, ==)
RTC_SAFECMP_MAKE_OP(NeOp, !=)
RTC_SAFECMP_MAKE_OP(LtOp, <)
RTC_SAFECMP_MAKE_OP(LeOp, <=)
RTC_SAFECMP_MAKE_OP(GtOp, >)
RTC_SAFECMP_MAKE_OP(GeOp, >=)
#undef RTC_SAFECMP_MAKE_OP
} // namespace safe_cmp_impl
#define RTC_SAFECMP_MAKE_FUN(name) \
template <typename T1, typename T2> \
constexpr \
typename std::enable_if<IsIntlike<T1>::value && IsIntlike<T2>::value, \
bool>::type Safe##name(T1 a, T2 b) { \
/* Unary plus here turns enums into real integral types. */ \
return safe_cmp_impl::Cmp<safe_cmp_impl::name##Op>(+a, +b); \
} \
template <typename T1, typename T2> \
constexpr \
typename std::enable_if<!IsIntlike<T1>::value || !IsIntlike<T2>::value, \
bool>::type Safe##name(const T1& a, \
const T2& b) { \
return safe_cmp_impl::name##Op::Op(a, b); \
}
RTC_SAFECMP_MAKE_FUN(Eq)
RTC_SAFECMP_MAKE_FUN(Ne)
RTC_SAFECMP_MAKE_FUN(Lt)
RTC_SAFECMP_MAKE_FUN(Le)
RTC_SAFECMP_MAKE_FUN(Gt)
RTC_SAFECMP_MAKE_FUN(Ge)
#undef RTC_SAFECMP_MAKE_FUN
} // namespace rtc
#endif // RTC_BASE_NUMERICS_SAFE_COMPARE_H_

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_SANITIZER_H_
#define RTC_BASE_SANITIZER_H_
#include <stddef.h> // For size_t.
#ifdef __cplusplus
#include <type_traits>
#endif
#if defined(__has_feature)
#if __has_feature(address_sanitizer)
#define RTC_HAS_ASAN 1
#endif
#if __has_feature(memory_sanitizer)
#define RTC_HAS_MSAN 1
#endif
#endif
#ifndef RTC_HAS_ASAN
#define RTC_HAS_ASAN 0
#endif
#ifndef RTC_HAS_MSAN
#define RTC_HAS_MSAN 0
#endif
#if RTC_HAS_ASAN
#include <sanitizer/asan_interface.h>
#endif
#if RTC_HAS_MSAN
#include <sanitizer/msan_interface.h>
#endif
#ifdef __has_attribute
#if __has_attribute(no_sanitize)
#define RTC_NO_SANITIZE(what) __attribute__((no_sanitize(what)))
#endif
#endif
#ifndef RTC_NO_SANITIZE
#define RTC_NO_SANITIZE(what)
#endif
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being unaddressable, so that reads and writes are not allowed. ASan may
// narrow the range to the nearest alignment boundaries.
static inline void rtc_AsanPoison(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
}
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being addressable, so that reads and writes are allowed. ASan may widen
// the range to the nearest alignment boundaries.
static inline void rtc_AsanUnpoison(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
}
// Ask MSan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being uninitialized.
static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_poison(ptr, element_size * num_elements);
#endif
}
// Force an MSan check (if any bits in the memory range [ptr, ptr +
// element_size * num_elements) are uninitialized the call will crash with an
// MSan report).
static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_check_mem_is_initialized(ptr, element_size * num_elements);
#endif
}
#ifdef __cplusplus
namespace rtc {
namespace sanitizer_impl {
template <typename T>
constexpr bool IsTriviallyCopyable() {
return static_cast<bool>(std::is_trivially_copy_constructible<T>::value &&
(std::is_trivially_copy_assignable<T>::value ||
!std::is_copy_assignable<T>::value) &&
std::is_trivially_destructible<T>::value);
}
} // namespace sanitizer_impl
template <typename T>
inline void AsanPoison(const T& mem) {
rtc_AsanPoison(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline void AsanUnpoison(const T& mem) {
rtc_AsanUnpoison(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline void MsanMarkUninitialized(const T& mem) {
rtc_MsanMarkUninitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline T MsanUninitialized(T t) {
#if RTC_HAS_MSAN
// TODO(bugs.webrtc.org/8762): Switch to std::is_trivially_copyable when it
// becomes available in downstream projects.
static_assert(sanitizer_impl::IsTriviallyCopyable<T>(), "");
#endif
rtc_MsanMarkUninitialized(&t, sizeof(T), 1);
return t;
}
template <typename T>
inline void MsanCheckInitialized(const T& mem) {
rtc_MsanCheckInitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
}
} // namespace rtc
#endif // __cplusplus
#endif // RTC_BASE_SANITIZER_H_

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc/rtc_base/system/arch.h
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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file contains platform-specific typedefs and defines.
// Much of it is derived from Chromium's build/build_config.h.
#ifndef RTC_BASE_SYSTEM_ARCH_H_
#define RTC_BASE_SYSTEM_ARCH_H_
// Processor architecture detection. For more info on what's defined, see:
// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
// http://www.agner.org/optimize/calling_conventions.pdf
// or with gcc, run: "echo | gcc -E -dM -"
#if defined(_M_X64) || defined(__x86_64__)
#define WEBRTC_ARCH_X86_FAMILY
#define WEBRTC_ARCH_X86_64
#define WEBRTC_ARCH_64_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__aarch64__)
#define WEBRTC_ARCH_ARM_FAMILY
#define WEBRTC_ARCH_64_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(_M_IX86) || defined(__i386__)
#define WEBRTC_ARCH_X86_FAMILY
#define WEBRTC_ARCH_X86
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__ARMEL__)
#define WEBRTC_ARCH_ARM_FAMILY
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__MIPSEL__)
#define WEBRTC_ARCH_MIPS_FAMILY
#if defined(__LP64__)
#define WEBRTC_ARCH_64_BITS
#else
#define WEBRTC_ARCH_32_BITS
#endif
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__pnacl__)
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#else
#error Please add support for your architecture in typedefs.h
#endif
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
#endif
#endif // RTC_BASE_SYSTEM_ARCH_H_

31
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/rtc_base/system/inline.h
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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_SYSTEM_INLINE_H_
#define RTC_BASE_SYSTEM_INLINE_H_
#if defined(_MSC_VER)
#define RTC_FORCE_INLINE __forceinline
#define RTC_NO_INLINE __declspec(noinline)
#elif defined(__GNUC__)
#define RTC_FORCE_INLINE __attribute__((__always_inline__))
#define RTC_NO_INLINE __attribute__((__noinline__))
#else
#define RTC_FORCE_INLINE
#define RTC_NO_INLINE
#endif
#endif // RTC_BASE_SYSTEM_INLINE_H_

140
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/rtc_base/type_traits.h
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_TYPE_TRAITS_H_
#define RTC_BASE_TYPE_TRAITS_H_
#include <cstddef>
#include <type_traits>
namespace rtc {
// Determines if the given class has zero-argument .data() and .size() methods
// whose return values are convertible to T* and size_t, respectively.
template <typename DS, typename T>
class HasDataAndSize {
private:
template <
typename C,
typename std::enable_if<
std::is_convertible<decltype(std::declval<C>().data()), T*>::value &&
std::is_convertible<decltype(std::declval<C>().size()),
std::size_t>::value>::type* = nullptr>
static int Test(int);
template <typename>
static char Test(...);
public:
static constexpr bool value = std::is_same<decltype(Test<DS>(0)), int>::value;
};
namespace test_has_data_and_size {
template <typename DR, typename SR>
struct Test1 {
DR data();
SR size();
};
static_assert(HasDataAndSize<Test1<int*, int>, int>::value, "");
static_assert(HasDataAndSize<Test1<int*, int>, const int>::value, "");
static_assert(HasDataAndSize<Test1<const int*, int>, const int>::value, "");
static_assert(!HasDataAndSize<Test1<const int*, int>, int>::value,
"implicit cast of const int* to int*");
static_assert(!HasDataAndSize<Test1<char*, size_t>, int>::value,
"implicit cast of char* to int*");
struct Test2 {
int* data;
size_t size;
};
static_assert(!HasDataAndSize<Test2, int>::value,
".data and .size aren't functions");
struct Test3 {
int* data();
};
static_assert(!HasDataAndSize<Test3, int>::value, ".size() is missing");
class Test4 {
int* data();
size_t size();
};
static_assert(!HasDataAndSize<Test4, int>::value,
".data() and .size() are private");
} // namespace test_has_data_and_size
namespace type_traits_impl {
// Determines if the given type is an enum that converts implicitly to
// an integral type.
template <typename T>
struct IsIntEnum {
private:
// This overload is used if the type is an enum, and unary plus
// compiles and turns it into an integral type.
template <typename X,
typename std::enable_if<
std::is_enum<X>::value &&
std::is_integral<decltype(+std::declval<X>())>::value>::type* =
nullptr>
static int Test(int);
// Otherwise, this overload is used.
template <typename>
static char Test(...);
public:
static constexpr bool value =
std::is_same<decltype(Test<typename std::remove_reference<T>::type>(0)),
int>::value;
};
} // namespace type_traits_impl
// Determines if the given type is integral, or an enum that
// converts implicitly to an integral type.
template <typename T>
struct IsIntlike {
private:
using X = typename std::remove_reference<T>::type;
public:
static constexpr bool value =
std::is_integral<X>::value || type_traits_impl::IsIntEnum<X>::value;
};
namespace test_enum_intlike {
enum E1 { e1 };
enum { e2 };
enum class E3 { e3 };
struct S {};
static_assert(type_traits_impl::IsIntEnum<E1>::value, "");
static_assert(type_traits_impl::IsIntEnum<decltype(e2)>::value, "");
static_assert(!type_traits_impl::IsIntEnum<E3>::value, "");
static_assert(!type_traits_impl::IsIntEnum<int>::value, "");
static_assert(!type_traits_impl::IsIntEnum<float>::value, "");
static_assert(!type_traits_impl::IsIntEnum<S>::value, "");
static_assert(IsIntlike<E1>::value, "");
static_assert(IsIntlike<decltype(e2)>::value, "");
static_assert(!IsIntlike<E3>::value, "");
static_assert(IsIntlike<int>::value, "");
static_assert(!IsIntlike<float>::value, "");
static_assert(!IsIntlike<S>::value, "");
} // namespace test_enum_intlike
} // namespace rtc
#endif // RTC_BASE_TYPE_TRAITS_H_

48
runtime/engine/asr/server/brpc/paraformerCPP/webrtc/system_wrappers/include/cpu_features_wrapper.h
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
#define SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
#include <stdint.h>
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
#endif
// List of features in x86.
typedef enum { kSSE2, kSSE3 } CPUFeature;
// List of features in ARM.
enum {
kCPUFeatureARMv7 = (1 << 0),
kCPUFeatureVFPv3 = (1 << 1),
kCPUFeatureNEON = (1 << 2),
kCPUFeatureLDREXSTREX = (1 << 3)
};
typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
// Returns true if the CPU supports the feature.
extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
// No CPU feature is available => straight C path.
extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
// Return the features in an ARM device.
// It detects the features in the hardware platform, and returns supported
// values in the above enum definition as a bitmask.
extern uint64_t WebRtc_GetCPUFeaturesARM(void);
#if defined(__cplusplus) || defined(c_plusplus)
} // extern "C"
#endif
#endif // SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc/typedefs.h
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file contains platform-specific typedefs and defines.
// Much of it is derived from Chromium's build/build_config.h.
#ifndef WEBRTC_TYPEDEFS_H_
#define WEBRTC_TYPEDEFS_H_
// Processor architecture detection. For more info on what's defined, see:
// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
// http://www.agner.org/optimize/calling_conventions.pdf
// or with gcc, run: "echo | gcc -E -dM -"
#if defined(_M_X64) || defined(__x86_64__)
#define WEBRTC_ARCH_X86_FAMILY
#define WEBRTC_ARCH_X86_64
#define WEBRTC_ARCH_64_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__aarch64__)
#define WEBRTC_ARCH_64_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(_M_IX86) || defined(__i386__)
#define WEBRTC_ARCH_X86_FAMILY
#define WEBRTC_ARCH_X86
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__ARMEL__)
// TODO(ajm): We'd prefer to control platform defines here, but this is
// currently provided by the Android makefiles. Commented to avoid duplicate
// definition warnings.
//#define WEBRTC_ARCH_ARM
// TODO(ajm): Chromium uses the following two defines. Should we switch?
//#define WEBRTC_ARCH_ARM_FAMILY
//#define WEBRTC_ARCH_ARMEL
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__MIPSEL__)
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__pnacl__)
#define WEBRTC_ARCH_32_BITS
#define WEBRTC_ARCH_LITTLE_ENDIAN
#elif defined(__PPC__)
#if defined(__PPC64__)
#define WEBRTC_ARCH_64_BITS
#else
#define WEBRTC_ARCH_32_BITS
#endif
#define WEBRTC_ARCH_BIG_ENDIAN
#else
#error Please add support for your architecture in typedefs.h
#endif
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
#endif
#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)) || \
(defined(WEBRTC_ARCH_ARM_V7) && !defined(WEBRTC_ARCH_ARM_NEON))
#define WEBRTC_CPU_DETECTION
#endif
#if !defined(_MSC_VER)
#include <stdint.h>
#else
// Define C99 equivalent types, since pre-2010 MSVC doesn't provide stdint.h.
typedef signed char int8_t;
typedef signed short int16_t;
typedef signed int int32_t;
typedef __int64 int64_t;
typedef unsigned char uint8_t;
typedef unsigned short uint16_t;
typedef unsigned int uint32_t;
typedef unsigned __int64 uint64_t;
#endif
// Borrowed from Chromium's base/compiler_specific.h.
// Annotate a virtual method indicating it must be overriding a virtual
// method in the parent class.
// Use like:
// virtual void foo() OVERRIDE;
#if defined(_MSC_VER)
#define OVERRIDE override
#elif defined(__clang__)
// Clang defaults to C++03 and warns about using override. Squelch that.
// Intentionally no push/pop here so all users of OVERRIDE ignore the warning
// too. This is like passing -Wno-c++11-extensions, except that GCC won't die
// (because it won't see this pragma).
#pragma clang diagnostic ignored "-Wc++11-extensions"
#define OVERRIDE override
#elif defined(__GNUC__) && __cplusplus >= 201103 && \
(__GNUC__ * 10000 + __GNUC_MINOR__ * 100) >= 40700
// GCC 4.7 supports explicit virtual overrides when C++11 support is enabled.
#define OVERRIDE override
#else
#define OVERRIDE
#endif
// Annotate a function indicating the caller must examine the return value.
// Use like:
// int foo() WARN_UNUSED_RESULT;
// TODO(ajm): Hack to avoid multiple definitions until the base/ of webrtc and
// libjingle are merged.
#if !defined(WARN_UNUSED_RESULT)
#if defined(__GNUC__)
#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
#else
#define WARN_UNUSED_RESULT
#endif
#endif // WARN_UNUSED_RESULT
// Put after a variable that might not be used, to prevent compiler warnings:
// int result ATTRIBUTE_UNUSED = DoSomething();
// assert(result == 17);
#ifndef ATTRIBUTE_UNUSED
#if defined(__GNUC__) || defined(__clang__)
#define ATTRIBUTE_UNUSED __attribute__((unused))
#else
#define ATTRIBUTE_UNUSED
#endif
#endif
// Macro to be used for switch-case fallthrough (required for enabling
// -Wimplicit-fallthrough warning on Clang).
#ifndef FALLTHROUGH
#if defined(__clang__)
#define FALLTHROUGH() [[clang::fallthrough]]
#else
#define FALLTHROUGH() do { } while (0)
#endif
#endif
// Annotate a function that will not return control flow to the caller.
#if defined(_MSC_VER)
#define NO_RETURN __declspec(noreturn)
#elif defined(__GNUC__)
#define NO_RETURN __attribute__((noreturn))
#else
#define NO_RETURN
#endif
#endif // WEBRTC_TYPEDEFS_H_

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runtime/engine/asr/server/brpc/paraformerCPP/webrtc_vad.h
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This header file includes the VAD API calls. Specific function calls are
* given below.
*/
#ifndef COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT
#define COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
#include <stddef.h>
#include <stdint.h>
typedef struct WebRtcVadInst VadInst;
#ifdef __cplusplus
extern "C" {
#endif
// Creates an instance to the VAD structure.
VadInst* WebRtcVad_Create(void);
// Frees the dynamic memory of a specified VAD instance.
//
// - handle [i] : Pointer to VAD instance that should be freed.
void WebRtcVad_Free(VadInst* handle);
// Initializes a VAD instance.
//
// - handle [i/o] : Instance that should be initialized.
//
// returns : 0 - (OK),
// -1 - (null pointer or Default mode could not be set).
int WebRtcVad_Init(VadInst* handle);
// Sets the VAD operating mode. A more aggressive (higher mode) VAD is more
// restrictive in reporting speech. Put in other words the probability of being
// speech when the VAD returns 1 is increased with increasing mode. As a
// consequence also the missed detection rate goes up.
//
// - handle [i/o] : VAD instance.
// - mode [i] : Aggressiveness mode (0, 1, 2, or 3).
//
// returns : 0 - (OK),
// -1 - (null pointer, mode could not be set or the VAD instance
// has not been initialized).
int WebRtcVad_set_mode(VadInst* handle, int mode);
// Calculates a VAD decision for the |audio_frame|. For valid sampling rates
// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
//
// - handle [i/o] : VAD Instance. Needs to be initialized by
// WebRtcVad_Init() before call.
// - fs [i] : Sampling frequency (Hz): 8000, 16000, or 32000
// - audio_frame [i] : Audio frame buffer.
// - frame_length [i] : Length of audio frame buffer in number of samples.
//
// returns : 1 - (Active Voice),
// 0 - (Non-active Voice),
// -1 - (Error)
int WebRtcVad_Process(VadInst* handle,
int fs,
const int16_t* audio_frame,
size_t frame_length);
// Checks for valid combinations of |rate| and |frame_length|. We support 10,
// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
//
// - rate [i] : Sampling frequency (Hz).
// - frame_length [i] : Speech frame buffer length in number of samples.
//
// returns : 0 - (valid combination), -1 - (invalid combination)
int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
#ifdef __cplusplus
}
#endif
#endif // COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT

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runtime/engine/asr/server/brpc/paraformerCPP/win_func.h
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#include <time.h>
#ifdef WIN32
#include <windows.h>
#else
#include <sys/time.h>
#endif
#ifdef WIN32
int gettimeofday(struct timeval* tp, void* tzp)
{
time_t clock;
struct tm tm;
SYSTEMTIME wtm;
GetLocalTime(&wtm);
tm.tm_year = wtm.wYear - 1900;
tm.tm_mon = wtm.wMonth - 1;
tm.tm_mday = wtm.wDay;
tm.tm_hour = wtm.wHour;
tm.tm_min = wtm.wMinute;
tm.tm_sec = wtm.wSecond;
tm.tm_isdst = -1;
clock = mktime(&tm);
tp->tv_sec = clock;
tp->tv_usec = wtm.wMilliseconds * 1000;
return (0);
}
#endif

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runtime/engine/asr/server/brpc/proto/echo.proto
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syntax="proto2";
package example;
option cc_generic_services = true;
message AudioRequest {
required string audio = 1;
required string extra = 2;
};
message AudioResponse {
required int32 err_no = 1;
required string err_msg = 2;
required string result = 3;
required float cost_time = 4;
};
service EchoService {
rpc audiorecognition(AudioRequest) returns (AudioResponse);
};
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