#include <math.h>
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#include <stdint.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <fstream>
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#include <assert.h>
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#include "audio.h"
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#include "precomp.h"
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using namespace std;
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namespace funasr {
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// see http://soundfile.sapp.org/doc/WaveFormat/
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// Note: We assume little endian here
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struct WaveHeader {
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bool Validate() const {
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// F F I R
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if (chunk_id != 0x46464952) {
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printf("Expected chunk_id RIFF. Given: 0x%08x\n", chunk_id);
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return false;
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}
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// E V A W
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if (format != 0x45564157) {
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printf("Expected format WAVE. Given: 0x%08x\n", format);
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return false;
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}
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if (subchunk1_id != 0x20746d66) {
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printf("Expected subchunk1_id 0x20746d66. Given: 0x%08x\n",
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subchunk1_id);
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return false;
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}
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if (subchunk1_size != 16) { // 16 for PCM
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printf("Expected subchunk1_size 16. Given: %d\n",
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subchunk1_size);
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return false;
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}
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if (audio_format != 1) { // 1 for PCM
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printf("Expected audio_format 1. Given: %d\n", audio_format);
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return false;
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}
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if (num_channels != 1) { // we support only single channel for now
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printf("Expected single channel. Given: %d\n", num_channels);
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return false;
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}
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if (byte_rate != (sample_rate * num_channels * bits_per_sample / 8)) {
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return false;
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}
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if (block_align != (num_channels * bits_per_sample / 8)) {
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return false;
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}
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if (bits_per_sample != 16) { // we support only 16 bits per sample
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printf("Expected bits_per_sample 16. Given: %d\n",
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bits_per_sample);
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return false;
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}
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return true;
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}
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// See https://en.wikipedia.org/wiki/WAV#Metadata and
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// https://www.robotplanet.dk/audio/wav_meta_data/riff_mci.pdf
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void SeekToDataChunk(std::istream &is) {
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// a t a d
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while (is && subchunk2_id != 0x61746164) {
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// const char *p = reinterpret_cast<const char *>(&subchunk2_id);
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// printf("Skip chunk (%x): %c%c%c%c of size: %d\n", subchunk2_id, p[0],
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// p[1], p[2], p[3], subchunk2_size);
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is.seekg(subchunk2_size, std::istream::cur);
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is.read(reinterpret_cast<char *>(&subchunk2_id), sizeof(int32_t));
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is.read(reinterpret_cast<char *>(&subchunk2_size), sizeof(int32_t));
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}
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}
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int32_t chunk_id;
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int32_t chunk_size;
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int32_t format;
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int32_t subchunk1_id;
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int32_t subchunk1_size;
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int16_t audio_format;
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int16_t num_channels;
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int32_t sample_rate;
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int32_t byte_rate;
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int16_t block_align;
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int16_t bits_per_sample;
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int32_t subchunk2_id; // a tag of this chunk
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int32_t subchunk2_size; // size of subchunk2
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};
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static_assert(sizeof(WaveHeader) == WAV_HEADER_SIZE, "");
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class AudioWindow {
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private:
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int *window;
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int in_idx;
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int out_idx;
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int sum;
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int window_size = 0;
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public:
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AudioWindow(int window_size) : window_size(window_size)
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{
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window = (int *)calloc(sizeof(int), window_size + 1);
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in_idx = 0;
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out_idx = 1;
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sum = 0;
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};
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~AudioWindow(){
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free(window);
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};
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int put(int val)
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{
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sum = sum + val - window[out_idx];
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window[in_idx] = val;
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in_idx = in_idx == window_size ? 0 : in_idx + 1;
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out_idx = out_idx == window_size ? 0 : out_idx + 1;
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return sum;
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};
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};
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AudioFrame::AudioFrame(){};
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AudioFrame::AudioFrame(int len) : len(len)
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{
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start = 0;
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};
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AudioFrame::~AudioFrame(){};
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int AudioFrame::SetStart(int val)
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{
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start = val < 0 ? 0 : val;
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return start;
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};
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int AudioFrame::SetEnd(int val)
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{
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end = val;
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len = end - start;
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return end;
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};
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int AudioFrame::GetStart()
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{
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return start;
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};
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int AudioFrame::GetLen()
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{
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return len;
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};
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int AudioFrame::Disp()
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{
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LOG(ERROR) << "Not imp!!!!";
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return 0;
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};
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Audio::Audio(int data_type) : data_type(data_type)
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{
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speech_buff = NULL;
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speech_data = NULL;
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align_size = 1360;
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}
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Audio::Audio(int data_type, int size) : data_type(data_type)
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{
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speech_buff = NULL;
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speech_data = NULL;
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align_size = (float)size;
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}
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Audio::~Audio()
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{
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if (speech_buff != NULL) {
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free(speech_buff);
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}
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if (speech_data != NULL) {
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free(speech_data);
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}
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}
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void Audio::Disp()
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{
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LOG(INFO) << "Audio time is " << (float)speech_len / MODEL_SAMPLE_RATE << " s. len is " << speech_len;
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}
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float Audio::GetTimeLen()
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{
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return (float)speech_len / MODEL_SAMPLE_RATE;
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}
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void Audio::WavResample(int32_t sampling_rate, const float *waveform,
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int32_t n)
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{
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LOG(INFO) << "Creating a resampler:\n"
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<< " in_sample_rate: "<< sampling_rate << "\n"
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<< " output_sample_rate: " << static_cast<int32_t>(MODEL_SAMPLE_RATE);
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float min_freq =
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std::min<int32_t>(sampling_rate, MODEL_SAMPLE_RATE);
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float lowpass_cutoff = 0.99 * 0.5 * min_freq;
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int32_t lowpass_filter_width = 6;
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auto resampler = std::make_unique<LinearResample>(
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sampling_rate, MODEL_SAMPLE_RATE, lowpass_cutoff, lowpass_filter_width);
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std::vector<float> samples;
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resampler->Resample(waveform, n, true, &samples);
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//reset speech_data
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speech_len = samples.size();
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if (speech_data != NULL) {
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free(speech_data);
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}
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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memset(speech_data, 0, sizeof(float) * speech_len);
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copy(samples.begin(), samples.end(), speech_data);
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}
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bool Audio::LoadWav(const char *filename, int32_t* sampling_rate)
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{
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WaveHeader header;
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if (speech_data != NULL) {
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free(speech_data);
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}
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if (speech_buff != NULL) {
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free(speech_buff);
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}
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offset = 0;
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std::ifstream is(filename, std::ifstream::binary);
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is.read(reinterpret_cast<char *>(&header), sizeof(header));
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if(!is){
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LOG(ERROR) << "Failed to read " << filename;
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return false;
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}
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if (!header.Validate()) {
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return false;
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}
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header.SeekToDataChunk(is);
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if (!is) {
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return false;
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}
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if (!header.Validate()) {
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return false;
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}
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header.SeekToDataChunk(is);
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if (!is) {
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return false;
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}
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*sampling_rate = header.sample_rate;
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// header.subchunk2_size contains the number of bytes in the data.
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// As we assume each sample contains two bytes, so it is divided by 2 here
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speech_len = header.subchunk2_size / 2;
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speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len);
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if (speech_buff)
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{
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memset(speech_buff, 0, sizeof(int16_t) * speech_len);
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is.read(reinterpret_cast<char *>(speech_buff), header.subchunk2_size);
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if (!is) {
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LOG(ERROR) << "Failed to read " << filename;
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return false;
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}
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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memset(speech_data, 0, sizeof(float) * speech_len);
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float scale = 1;
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if (data_type == 1) {
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scale = 32768;
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}
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for (int32_t i = 0; i != speech_len; ++i) {
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speech_data[i] = (float)speech_buff[i] / scale;
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}
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//resample
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if(*sampling_rate != MODEL_SAMPLE_RATE){
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WavResample(*sampling_rate, speech_data, speech_len);
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}
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AudioFrame* frame = new AudioFrame(speech_len);
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frame_queue.push(frame);
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return true;
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}
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else
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return false;
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}
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bool Audio::LoadWav(const char* buf, int n_file_len, int32_t* sampling_rate)
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{
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WaveHeader header;
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if (speech_data != NULL) {
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free(speech_data);
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}
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if (speech_buff != NULL) {
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free(speech_buff);
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}
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offset = 0;
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std::memcpy(&header, buf, sizeof(header));
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*sampling_rate = header.sample_rate;
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speech_len = header.subchunk2_size / 2;
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speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len);
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if (speech_buff)
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{
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memset(speech_buff, 0, sizeof(int16_t) * speech_len);
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memcpy((void*)speech_buff, (const void*)(buf + WAV_HEADER_SIZE), speech_len * sizeof(int16_t));
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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memset(speech_data, 0, sizeof(float) * speech_len);
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float scale = 1;
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if (data_type == 1) {
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scale = 32768;
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}
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for (int32_t i = 0; i != speech_len; ++i) {
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speech_data[i] = (float)speech_buff[i] / scale;
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}
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//resample
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if(*sampling_rate != MODEL_SAMPLE_RATE){
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WavResample(*sampling_rate, speech_data, speech_len);
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}
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AudioFrame* frame = new AudioFrame(speech_len);
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frame_queue.push(frame);
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return true;
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}
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else
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return false;
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}
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bool Audio::LoadPcmwav(const char* buf, int n_buf_len, int32_t* sampling_rate)
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{
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if (speech_data != NULL) {
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free(speech_data);
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}
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if (speech_buff != NULL) {
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free(speech_buff);
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}
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offset = 0;
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speech_len = n_buf_len / 2;
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speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
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if (speech_buff)
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{
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memset(speech_buff, 0, sizeof(int16_t) * speech_len);
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memcpy((void*)speech_buff, (const void*)buf, speech_len * sizeof(int16_t));
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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memset(speech_data, 0, sizeof(float) * speech_len);
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float scale = 1;
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if (data_type == 1) {
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scale = 32768;
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}
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for (int32_t i = 0; i != speech_len; ++i) {
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speech_data[i] = (float)speech_buff[i] / scale;
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}
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//resample
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if(*sampling_rate != MODEL_SAMPLE_RATE){
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WavResample(*sampling_rate, speech_data, speech_len);
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}
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AudioFrame* frame = new AudioFrame(speech_len);
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frame_queue.push(frame);
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return true;
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}
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else
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return false;
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}
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bool Audio::LoadPcmwav(const char* filename, int32_t* sampling_rate)
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{
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if (speech_data != NULL) {
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free(speech_data);
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}
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if (speech_buff != NULL) {
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free(speech_buff);
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}
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offset = 0;
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FILE* fp;
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fp = fopen(filename, "rb");
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if (fp == nullptr)
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{
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LOG(ERROR) << "Failed to read " << filename;
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return false;
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}
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fseek(fp, 0, SEEK_END);
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uint32_t n_file_len = ftell(fp);
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fseek(fp, 0, SEEK_SET);
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speech_len = (n_file_len) / 2;
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speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
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if (speech_buff)
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{
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memset(speech_buff, 0, sizeof(int16_t) * speech_len);
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int ret = fread(speech_buff, sizeof(int16_t), speech_len, fp);
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fclose(fp);
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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memset(speech_data, 0, sizeof(float) * speech_len);
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float scale = 1;
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if (data_type == 1) {
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scale = 32768;
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}
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for (int32_t i = 0; i != speech_len; ++i) {
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speech_data[i] = (float)speech_buff[i] / scale;
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}
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//resample
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if(*sampling_rate != MODEL_SAMPLE_RATE){
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WavResample(*sampling_rate, speech_data, speech_len);
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}
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AudioFrame* frame = new AudioFrame(speech_len);
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frame_queue.push(frame);
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return true;
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}
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else
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return false;
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}
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int Audio::FetchChunck(float *&dout, int len)
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{
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if (offset >= speech_align_len) {
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dout = NULL;
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return S_ERR;
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} else if (offset == speech_align_len - len) {
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dout = speech_data + offset;
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offset = speech_align_len;
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// 临时解决
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AudioFrame *frame = frame_queue.front();
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frame_queue.pop();
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delete frame;
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return S_END;
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} else {
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dout = speech_data + offset;
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offset += len;
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return S_MIDDLE;
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}
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}
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int Audio::Fetch(float *&dout, int &len, int &flag)
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{
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if (frame_queue.size() > 0) {
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AudioFrame *frame = frame_queue.front();
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frame_queue.pop();
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dout = speech_data + frame->GetStart();
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len = frame->GetLen();
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delete frame;
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flag = S_END;
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return 1;
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} else {
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return 0;
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}
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}
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void Audio::Padding()
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{
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float num_samples = speech_len;
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float frame_length = 400;
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float frame_shift = 160;
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float num_frames = floor((num_samples + (frame_shift / 2)) / frame_shift);
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float num_new_samples = (num_frames - 1) * frame_shift + frame_length;
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float num_padding = num_new_samples - num_samples;
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float num_left_padding = (frame_length - frame_shift) / 2;
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float num_right_padding = num_padding - num_left_padding;
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float *new_data = (float *)malloc(num_new_samples * sizeof(float));
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int i;
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int tmp_off = 0;
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for (i = 0; i < num_left_padding; i++) {
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int ii = num_left_padding - i - 1;
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new_data[i] = speech_data[ii];
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}
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tmp_off = num_left_padding;
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memcpy(new_data + tmp_off, speech_data, speech_len * sizeof(float));
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tmp_off += speech_len;
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for (i = 0; i < num_right_padding; i++) {
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int ii = speech_len - i - 1;
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new_data[tmp_off + i] = speech_data[ii];
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}
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free(speech_data);
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speech_data = new_data;
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speech_len = num_new_samples;
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AudioFrame *frame = new AudioFrame(num_new_samples);
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frame_queue.push(frame);
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frame = frame_queue.front();
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frame_queue.pop();
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delete frame;
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}
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void Audio::Split(OfflineStream* offline_stream)
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{
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AudioFrame *frame;
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frame = frame_queue.front();
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frame_queue.pop();
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int sp_len = frame->GetLen();
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delete frame;
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frame = NULL;
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std::vector<float> pcm_data(speech_data, speech_data+sp_len);
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vector<std::vector<int>> vad_segments = (offline_stream->vad_handle)->Infer(pcm_data);
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int seg_sample = MODEL_SAMPLE_RATE/1000;
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for(vector<int> segment:vad_segments)
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{
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frame = new AudioFrame();
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int start = segment[0]*seg_sample;
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int end = segment[1]*seg_sample;
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frame->SetStart(start);
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frame->SetEnd(end);
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frame_queue.push(frame);
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frame = NULL;
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}
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}
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void Audio::Split(VadModel* vad_obj, vector<std::vector<int>>& vad_segments)
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{
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AudioFrame *frame;
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frame = frame_queue.front();
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frame_queue.pop();
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int sp_len = frame->GetLen();
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delete frame;
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frame = NULL;
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std::vector<float> pcm_data(speech_data, speech_data+sp_len);
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vad_segments = vad_obj->Infer(pcm_data);
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}
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} // namespace funasr
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