#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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#ifdef _MSC_VER
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#pragma warning(disable:4996)
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#endif
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#if defined(__APPLE__)
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#include <string.h>
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#else
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extern "C" {
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#include <libavutil/opt.h>
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/samplefmt.h>
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#include <libswresample/swresample.h>
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}
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#endif
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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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window = nullptr;
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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(const AudioFrame &other)
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{
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start = other.start;
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end = other.end;
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len = other.len;
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is_final = other.is_final;
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}
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AudioFrame::AudioFrame(int start, int end, bool is_final):start(start),end(end),is_final(is_final){
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len = end - start;
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}
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AudioFrame::~AudioFrame(){
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if(data != nullptr){
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free(data);
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data = nullptr;
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}
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}
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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) : dest_sample_rate(MODEL_SAMPLE_RATE), data_type(data_type)
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{
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speech_buff = nullptr;
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speech_data = nullptr;
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align_size = 1360;
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seg_sample = dest_sample_rate / 1000;
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}
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Audio::Audio(int model_sample_rate, int data_type) : dest_sample_rate(model_sample_rate), data_type(data_type)
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{
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speech_buff = nullptr;
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speech_data = nullptr;
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align_size = 1360;
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seg_sample = dest_sample_rate / 1000;
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}
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Audio::Audio(int model_sample_rate, int data_type, int size) : dest_sample_rate(model_sample_rate), data_type(data_type)
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{
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speech_buff = nullptr;
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speech_data = nullptr;
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align_size = (float)size;
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seg_sample = dest_sample_rate / 1000;
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}
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Audio::~Audio()
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{
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if (speech_buff != nullptr) {
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free(speech_buff);
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speech_buff = nullptr;
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}
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if (speech_data != nullptr) {
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free(speech_data);
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speech_data = nullptr;
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}
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if (speech_char != nullptr) {
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free(speech_char);
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speech_char = nullptr;
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}
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ClearQueue(frame_queue);
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ClearQueue(asr_online_queue);
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ClearQueue(asr_offline_queue);
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}
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void Audio::ClearQueue(std::queue<AudioFrame*>& q) {
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while (!q.empty()) {
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AudioFrame* frame = q.front();
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delete frame;
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q.pop();
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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 / dest_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 / dest_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: "
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<< " in_sample_rate: "<< sampling_rate
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<< " output_sample_rate: " << static_cast<int32_t>(dest_sample_rate);
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float min_freq =
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std::min<int32_t>(sampling_rate, dest_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, dest_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 != nullptr) {
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free(speech_data);
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speech_data = nullptr;
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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::FfmpegLoad(const char *filename, bool copy2char){
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#if defined(__APPLE__)
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return false;
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#else
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// from file
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AVFormatContext* formatContext = avformat_alloc_context();
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if (avformat_open_input(&formatContext, filename, nullptr, nullptr) != 0) {
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LOG(ERROR) << "Error: Could not open input file.";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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if (avformat_find_stream_info(formatContext, nullptr) < 0) {
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LOG(ERROR) << "Error: Could not open input file.";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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const AVCodec* codec = nullptr;
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AVCodecParameters* codecParameters = nullptr;
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int audioStreamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
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if (audioStreamIndex >= 0) {
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codecParameters = formatContext->streams[audioStreamIndex]->codecpar;
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}else {
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LOG(ERROR) << "Error: Could not open input file.";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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AVCodecContext* codecContext = avcodec_alloc_context3(codec);
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if (!codecContext) {
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LOG(ERROR) << "Failed to allocate codec context";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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if (avcodec_parameters_to_context(codecContext, codecParameters) != 0) {
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LOG(ERROR) << "Error: Could not copy codec parameters to codec context.";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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return false;
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}
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if (avcodec_open2(codecContext, codec, nullptr) < 0) {
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LOG(ERROR) << "Error: Could not open audio decoder.";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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return false;
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}
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SwrContext *swr_ctx = swr_alloc_set_opts(
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nullptr, // allocate a new context
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AV_CH_LAYOUT_MONO, // output channel layout (stereo)
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AV_SAMPLE_FMT_S16, // output sample format (signed 16-bit)
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dest_sample_rate, // output sample rate (same as input)
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av_get_default_channel_layout(codecContext->channels), // input channel layout
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codecContext->sample_fmt, // input sample format
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codecContext->sample_rate, // input sample rate
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0, // logging level
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nullptr // parent context
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);
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if (swr_ctx == nullptr) {
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LOG(ERROR) << "Could not initialize resampler";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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return false;
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}
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if (swr_init(swr_ctx) != 0) {
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LOG(ERROR) << "Could not initialize resampler";
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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swr_free(&swr_ctx);
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return false;
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}
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// to pcm
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AVPacket* packet = av_packet_alloc();
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AVFrame* frame = av_frame_alloc();
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std::vector<uint8_t> resampled_buffers;
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while (av_read_frame(formatContext, packet) >= 0) {
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if (packet->stream_index == audioStreamIndex) {
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if (avcodec_send_packet(codecContext, packet) >= 0) {
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while (avcodec_receive_frame(codecContext, frame) >= 0) {
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// Resample audio if necessary
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std::vector<uint8_t> resampled_buffer;
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int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, codecContext->sample_rate) + frame->nb_samples,
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dest_sample_rate,
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codecContext->sample_rate,
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AV_ROUND_DOWN);
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int resampled_size = out_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
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if (resampled_buffer.size() < resampled_size) {
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resampled_buffer.resize(resampled_size);
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}
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uint8_t *resampled_data = resampled_buffer.data();
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int ret = swr_convert(
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swr_ctx,
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&resampled_data, // output buffer
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out_samples, // output buffer size
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(const uint8_t **)(frame->data), // choose channel
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frame->nb_samples // input buffer size
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);
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if (ret < 0) {
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LOG(ERROR) << "Error resampling audio";
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break;
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}
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resampled_buffers.insert(resampled_buffers.end(), resampled_buffer.begin(), resampled_buffer.begin() + resampled_size);
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}
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}
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}
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av_packet_unref(packet);
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}
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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swr_free(&swr_ctx);
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av_packet_free(&packet);
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av_frame_free(&frame);
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if (speech_data != nullptr) {
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free(speech_data);
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speech_data = nullptr;
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}
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if (speech_char != nullptr) {
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free(speech_char);
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speech_char = nullptr;
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}
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offset = 0;
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if(copy2char){
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speech_char = (char *)malloc(resampled_buffers.size());
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memset(speech_char, 0, resampled_buffers.size());
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memcpy((void*)speech_char, (const void*)resampled_buffers.data(), resampled_buffers.size());
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}
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speech_len = (resampled_buffers.size()) / 2;
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speech_data = (float*)malloc(sizeof(float) * speech_len);
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if(speech_data){
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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 = 32768.0f;
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}
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for (int32_t i = 0; i < speech_len; ++i) {
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int16_t val = (int16_t)((resampled_buffers[2 * i + 1] << 8) | resampled_buffers[2 * i]);
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speech_data[i] = (float)val / scale;
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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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}else{
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return false;
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}
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#endif
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}
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bool Audio::FfmpegLoad(const char* buf, int n_file_len){
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#if defined(__APPLE__)
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return false;
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#else
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// from buf
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void* buf_copy = av_malloc(n_file_len);
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memcpy(buf_copy, buf, n_file_len);
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AVIOContext* avio_ctx = avio_alloc_context(
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(unsigned char*)buf_copy, // buffer
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n_file_len, // buffer size
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0, // write flag (0 for read-only)
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nullptr, // opaque pointer (not used here)
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nullptr, // read callback (not used here)
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nullptr, // write callback (not used here)
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nullptr // seek callback (not used here)
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);
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if (!avio_ctx) {
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av_free(buf_copy);
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return false;
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}
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AVFormatContext* formatContext = avformat_alloc_context();
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formatContext->pb = avio_ctx;
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if (avformat_open_input(&formatContext, "", nullptr, nullptr) != 0) {
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LOG(ERROR) << "Error: Could not open input file.";
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avio_context_free(&avio_ctx);
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
|
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if (avformat_find_stream_info(formatContext, nullptr) < 0) {
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LOG(ERROR) << "Error: Could not find stream information.";
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avio_context_free(&avio_ctx);
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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const AVCodec* codec = nullptr;
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AVCodecParameters* codecParameters = nullptr;
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int audioStreamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
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if (audioStreamIndex >= 0) {
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codecParameters = formatContext->streams[audioStreamIndex]->codecpar;
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}
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AVCodecContext* codecContext = avcodec_alloc_context3(codec);
|
if (!codecContext) {
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LOG(ERROR) << "Failed to allocate codec context";
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avio_context_free(&avio_ctx);
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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return false;
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}
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if (avcodec_parameters_to_context(codecContext, codecParameters) != 0) {
|
LOG(ERROR) << "Error: Could not copy codec parameters to codec context.";
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avio_context_free(&avio_ctx);
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
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return false;
|
}
|
if (avcodec_open2(codecContext, codec, nullptr) < 0) {
|
LOG(ERROR) << "Error: Could not open audio decoder.";
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avio_context_free(&avio_ctx);
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avformat_close_input(&formatContext);
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avformat_free_context(formatContext);
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avcodec_free_context(&codecContext);
|
return false;
|
}
|
SwrContext *swr_ctx = swr_alloc_set_opts(
|
nullptr, // allocate a new context
|
AV_CH_LAYOUT_MONO, // output channel layout (stereo)
|
AV_SAMPLE_FMT_S16, // output sample format (signed 16-bit)
|
dest_sample_rate, // output sample rate (same as input)
|
av_get_default_channel_layout(codecContext->channels), // input channel layout
|
codecContext->sample_fmt, // input sample format
|
codecContext->sample_rate, // input sample rate
|
0, // logging level
|
nullptr // parent context
|
);
|
if (swr_ctx == nullptr) {
|
LOG(ERROR) << "Could not initialize resampler";
|
avio_context_free(&avio_ctx);
|
avformat_close_input(&formatContext);
|
avformat_free_context(formatContext);
|
avcodec_free_context(&codecContext);
|
return false;
|
}
|
if (swr_init(swr_ctx) != 0) {
|
LOG(ERROR) << "Could not initialize resampler";
|
avio_context_free(&avio_ctx);
|
avformat_close_input(&formatContext);
|
avformat_free_context(formatContext);
|
avcodec_free_context(&codecContext);
|
swr_free(&swr_ctx);
|
return false;
|
}
|
|
// to pcm
|
AVPacket* packet = av_packet_alloc();
|
AVFrame* frame = av_frame_alloc();
|
std::vector<uint8_t> resampled_buffers;
|
while (av_read_frame(formatContext, packet) >= 0) {
|
if (packet->stream_index == audioStreamIndex) {
|
if (avcodec_send_packet(codecContext, packet) >= 0) {
|
while (avcodec_receive_frame(codecContext, frame) >= 0) {
|
// Resample audio if necessary
|
std::vector<uint8_t> resampled_buffer;
|
int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, codecContext->sample_rate) + frame->nb_samples,
|
dest_sample_rate,
|
codecContext->sample_rate,
|
AV_ROUND_DOWN);
|
|
int resampled_size = out_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
|
if (resampled_buffer.size() < resampled_size) {
|
resampled_buffer.resize(resampled_size);
|
}
|
uint8_t *resampled_data = resampled_buffer.data();
|
int ret = swr_convert(
|
swr_ctx,
|
&resampled_data, // output buffer
|
out_samples, // output buffer size
|
(const uint8_t **)(frame->data), // choose channel: channel_data
|
frame->nb_samples // input buffer size
|
);
|
if (ret < 0) {
|
LOG(ERROR) << "Error resampling audio";
|
break;
|
}
|
resampled_buffers.insert(resampled_buffers.end(), resampled_buffer.begin(), resampled_buffer.begin() + resampled_size);
|
}
|
}
|
}
|
av_packet_unref(packet);
|
}
|
|
//avio_context_free(&avio_ctx);
|
av_freep(&avio_ctx ->buffer);
|
av_freep(&avio_ctx);
|
avformat_close_input(&formatContext);
|
avformat_free_context(formatContext);
|
avcodec_free_context(&codecContext);
|
swr_free(&swr_ctx);
|
av_packet_free(&packet);
|
av_frame_free(&frame);
|
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
|
speech_len = (resampled_buffers.size()) / 2;
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
if(speech_data){
|
memset(speech_data, 0, sizeof(float) * speech_len);
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768.0f;
|
}
|
for (int32_t i = 0; i < speech_len; ++i) {
|
int16_t val = (int16_t)((resampled_buffers[2 * i + 1] << 8) | resampled_buffers[2 * i]);
|
speech_data[i] = (float)val / scale;
|
}
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}else{
|
return false;
|
}
|
|
#endif
|
}
|
|
|
bool Audio::LoadWav(const char *filename, int32_t* sampling_rate, bool resample)
|
{
|
WaveHeader header;
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
if (speech_buff != nullptr) {
|
free(speech_buff);
|
speech_buff = nullptr;
|
}
|
|
offset = 0;
|
std::ifstream is(filename, std::ifstream::binary);
|
is.read(reinterpret_cast<char *>(&header), sizeof(header));
|
if(!is){
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
|
if (!header.Validate()) {
|
return false;
|
}
|
|
header.SeekToDataChunk(is);
|
if (!is) {
|
return false;
|
}
|
|
if (!header.Validate()) {
|
return false;
|
}
|
|
header.SeekToDataChunk(is);
|
if (!is) {
|
return false;
|
}
|
|
*sampling_rate = header.sample_rate;
|
// header.subchunk2_size contains the number of bytes in the data.
|
// As we assume each sample contains two bytes, so it is divided by 2 here
|
speech_len = header.subchunk2_size / 2;
|
speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len);
|
|
if (speech_buff)
|
{
|
memset(speech_buff, 0, sizeof(int16_t) * speech_len);
|
is.read(reinterpret_cast<char *>(speech_buff), header.subchunk2_size);
|
if (!is) {
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
memset(speech_data, 0, sizeof(float) * speech_len);
|
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768;
|
}
|
for (int32_t i = 0; i != speech_len; ++i) {
|
speech_data[i] = (float)speech_buff[i] / scale;
|
}
|
|
//resample
|
if(resample && *sampling_rate != dest_sample_rate){
|
WavResample(*sampling_rate, speech_data, speech_len);
|
}
|
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}
|
else
|
return false;
|
}
|
|
bool Audio::LoadWav2Char(const char *filename, int32_t* sampling_rate)
|
{
|
WaveHeader header;
|
if (speech_char != nullptr) {
|
free(speech_char);
|
speech_char = nullptr;
|
}
|
offset = 0;
|
std::ifstream is(filename, std::ifstream::binary);
|
is.read(reinterpret_cast<char *>(&header), sizeof(header));
|
if(!is){
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
if (!header.Validate()) {
|
return false;
|
}
|
header.SeekToDataChunk(is);
|
if (!is) {
|
return false;
|
}
|
if (!header.Validate()) {
|
return false;
|
}
|
header.SeekToDataChunk(is);
|
if (!is) {
|
return false;
|
}
|
|
*sampling_rate = header.sample_rate;
|
// header.subchunk2_size contains the number of bytes in the data.
|
// As we assume each sample contains two bytes, so it is divided by 2 here
|
speech_len = header.subchunk2_size / 2;
|
speech_char = (char *)malloc(header.subchunk2_size);
|
memset(speech_char, 0, header.subchunk2_size);
|
is.read(speech_char, header.subchunk2_size);
|
|
return true;
|
}
|
|
bool Audio::LoadWav(const char* buf, int n_file_len, int32_t* sampling_rate)
|
{
|
WaveHeader header;
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
if (speech_buff != nullptr) {
|
free(speech_buff);
|
speech_buff = nullptr;
|
}
|
|
std::memcpy(&header, buf, sizeof(header));
|
|
*sampling_rate = header.sample_rate;
|
speech_len = header.subchunk2_size / 2;
|
speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len);
|
if (speech_buff)
|
{
|
memset(speech_buff, 0, sizeof(int16_t) * speech_len);
|
memcpy((void*)speech_buff, (const void*)(buf + WAV_HEADER_SIZE), speech_len * sizeof(int16_t));
|
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
memset(speech_data, 0, sizeof(float) * speech_len);
|
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768;
|
}
|
|
for (int32_t i = 0; i != speech_len; ++i) {
|
speech_data[i] = (float)speech_buff[i] / scale;
|
}
|
|
//resample
|
if(*sampling_rate != dest_sample_rate){
|
WavResample(*sampling_rate, speech_data, speech_len);
|
}
|
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}
|
else
|
return false;
|
}
|
|
bool Audio::LoadPcmwav(const char* buf, int n_buf_len, int32_t* sampling_rate)
|
{
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
|
speech_len = n_buf_len / 2;
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
if(speech_data){
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768.0f;
|
}
|
const uint8_t* byte_buf = reinterpret_cast<const uint8_t*>(buf);
|
for (int32_t i = 0; i < speech_len; ++i) {
|
int16_t val = (int16_t)((byte_buf[2 * i + 1] << 8) | byte_buf[2 * i]);
|
speech_data[i] = (float)val / scale;
|
}
|
|
//resample
|
if(*sampling_rate != dest_sample_rate){
|
WavResample(*sampling_rate, speech_data, speech_len);
|
}
|
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}else{
|
return false;
|
}
|
}
|
|
bool Audio::LoadPcmwavOnline(const char* buf, int n_buf_len, int32_t* sampling_rate)
|
{
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
|
speech_len = n_buf_len / 2;
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
if(speech_data){
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768.0f;
|
}
|
const uint8_t* byte_buf = reinterpret_cast<const uint8_t*>(buf);
|
for (int32_t i = 0; i < speech_len; ++i) {
|
int16_t val = (int16_t)((byte_buf[2 * i + 1] << 8) | byte_buf[2 * i]);
|
speech_data[i] = (float)val / scale;
|
}
|
|
//resample
|
if(*sampling_rate != dest_sample_rate){
|
WavResample(*sampling_rate, speech_data, speech_len);
|
}
|
|
for (int32_t i = 0; i != speech_len; ++i) {
|
all_samples.emplace_back(speech_data[i]);
|
}
|
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}else{
|
return false;
|
}
|
}
|
|
bool Audio::LoadPcmwav(const char* filename, int32_t* sampling_rate, bool resample)
|
{
|
if (speech_data != nullptr) {
|
free(speech_data);
|
speech_data = nullptr;
|
}
|
if (speech_buff != nullptr) {
|
free(speech_buff);
|
speech_buff = nullptr;
|
}
|
offset = 0;
|
|
FILE* fp;
|
fp = fopen(filename, "rb");
|
if (fp == nullptr)
|
{
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
fseek(fp, 0, SEEK_END);
|
uint32_t n_file_len = ftell(fp);
|
fseek(fp, 0, SEEK_SET);
|
|
speech_len = (n_file_len) / 2;
|
speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
|
if (speech_buff)
|
{
|
memset(speech_buff, 0, sizeof(int16_t) * speech_len);
|
int ret = fread(speech_buff, sizeof(int16_t), speech_len, fp);
|
fclose(fp);
|
|
speech_data = (float*)malloc(sizeof(float) * speech_len);
|
memset(speech_data, 0, sizeof(float) * speech_len);
|
|
float scale = 1;
|
if (data_type == 1) {
|
scale = 32768;
|
}
|
for (int32_t i = 0; i != speech_len; ++i) {
|
speech_data[i] = (float)speech_buff[i] / scale;
|
}
|
|
//resample
|
if(resample && *sampling_rate != dest_sample_rate){
|
WavResample(*sampling_rate, speech_data, speech_len);
|
}
|
|
AudioFrame* frame = new AudioFrame(speech_len);
|
frame_queue.push(frame);
|
|
return true;
|
}
|
else
|
return false;
|
|
}
|
|
bool Audio::LoadPcmwav2Char(const char* filename, int32_t* sampling_rate)
|
{
|
if (speech_char != nullptr) {
|
free(speech_char);
|
speech_char = nullptr;
|
}
|
offset = 0;
|
|
FILE* fp;
|
fp = fopen(filename, "rb");
|
if (fp == nullptr)
|
{
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
fseek(fp, 0, SEEK_END);
|
uint32_t n_file_len = ftell(fp);
|
fseek(fp, 0, SEEK_SET);
|
|
speech_len = (n_file_len) / 2;
|
speech_char = (char *)malloc(n_file_len);
|
memset(speech_char, 0, n_file_len);
|
fread(speech_char, sizeof(int16_t), n_file_len/2, fp);
|
fclose(fp);
|
|
return true;
|
}
|
|
bool Audio::LoadOthers2Char(const char* filename)
|
{
|
if (speech_char != nullptr) {
|
free(speech_char);
|
speech_char = nullptr;
|
}
|
|
FILE* fp;
|
fp = fopen(filename, "rb");
|
if (fp == nullptr)
|
{
|
LOG(ERROR) << "Failed to read " << filename;
|
return false;
|
}
|
fseek(fp, 0, SEEK_END);
|
uint32_t n_file_len = ftell(fp);
|
fseek(fp, 0, SEEK_SET);
|
|
speech_len = n_file_len;
|
speech_char = (char *)malloc(n_file_len);
|
memset(speech_char, 0, n_file_len);
|
fread(speech_char, 1, n_file_len, fp);
|
fclose(fp);
|
|
return true;
|
}
|
|
int Audio::FetchTpass(AudioFrame *&frame)
|
{
|
if (asr_offline_queue.size() > 0) {
|
frame = asr_offline_queue.front();
|
asr_offline_queue.pop();
|
return 1;
|
} else {
|
return 0;
|
}
|
}
|
|
int Audio::FetchChunck(AudioFrame *&frame)
|
{
|
if (asr_online_queue.size() > 0) {
|
frame = asr_online_queue.front();
|
asr_online_queue.pop();
|
return 1;
|
} else {
|
return 0;
|
}
|
}
|
|
int Audio::Fetch(float *&dout, int &len, int &flag)
|
{
|
if (frame_queue.size() > 0) {
|
AudioFrame *frame = frame_queue.front();
|
frame_queue.pop();
|
|
dout = speech_data + frame->GetStart();
|
len = frame->GetLen();
|
delete frame;
|
flag = S_END;
|
return 1;
|
} else {
|
return 0;
|
}
|
}
|
|
int Audio::Fetch(float *&dout, int &len, int &flag, float &start_time)
|
{
|
if (frame_queue.size() > 0) {
|
AudioFrame *frame = frame_queue.front();
|
frame_queue.pop();
|
|
start_time = (float)(frame->GetStart())/ dest_sample_rate;
|
dout = speech_data + frame->GetStart();
|
len = frame->GetLen();
|
delete frame;
|
flag = S_END;
|
return 1;
|
} else {
|
return 0;
|
}
|
}
|
|
void Audio::Padding()
|
{
|
float num_samples = speech_len;
|
float frame_length = 400;
|
float frame_shift = 160;
|
float num_frames = floor((num_samples + (frame_shift / 2)) / frame_shift);
|
float num_new_samples = (num_frames - 1) * frame_shift + frame_length;
|
float num_padding = num_new_samples - num_samples;
|
float num_left_padding = (frame_length - frame_shift) / 2;
|
float num_right_padding = num_padding - num_left_padding;
|
|
float *new_data = (float *)malloc(num_new_samples * sizeof(float));
|
int i;
|
int tmp_off = 0;
|
for (i = 0; i < num_left_padding; i++) {
|
int ii = num_left_padding - i - 1;
|
new_data[i] = speech_data[ii];
|
}
|
tmp_off = num_left_padding;
|
memcpy(new_data + tmp_off, speech_data, speech_len * sizeof(float));
|
tmp_off += speech_len;
|
|
for (i = 0; i < num_right_padding; i++) {
|
int ii = speech_len - i - 1;
|
new_data[tmp_off + i] = speech_data[ii];
|
}
|
free(speech_data);
|
speech_data = nullptr;
|
speech_data = new_data;
|
speech_len = num_new_samples;
|
|
AudioFrame *frame = new AudioFrame(num_new_samples);
|
frame_queue.push(frame);
|
frame = frame_queue.front();
|
frame_queue.pop();
|
delete frame;
|
}
|
|
void Audio::Split(OfflineStream* offline_stream)
|
{
|
AudioFrame *frame;
|
|
frame = frame_queue.front();
|
frame_queue.pop();
|
int sp_len = frame->GetLen();
|
delete frame;
|
frame = nullptr;
|
|
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
|
vector<std::vector<int>> vad_segments = (offline_stream->vad_handle)->Infer(pcm_data);
|
for(vector<int> segment:vad_segments)
|
{
|
frame = new AudioFrame();
|
int start = segment[0]*seg_sample;
|
int end = segment[1]*seg_sample;
|
frame->SetStart(start);
|
frame->SetEnd(end);
|
frame_queue.push(frame);
|
frame = nullptr;
|
}
|
}
|
|
void Audio::CutSplit(OfflineStream* offline_stream)
|
{
|
std::unique_ptr<VadModel> vad_online_handle = make_unique<FsmnVadOnline>((FsmnVad*)(offline_stream->vad_handle).get());
|
AudioFrame *frame;
|
|
frame = frame_queue.front();
|
frame_queue.pop();
|
int sp_len = frame->GetLen();
|
delete frame;
|
frame = nullptr;
|
|
int step = dest_sample_rate*1;
|
bool is_final=false;
|
vector<std::vector<int>> vad_segments;
|
for (int sample_offset = 0; sample_offset < speech_len; sample_offset += std::min(step, speech_len - sample_offset)) {
|
if (sample_offset + step >= speech_len - 1) {
|
step = speech_len - sample_offset;
|
is_final = true;
|
} else {
|
is_final = false;
|
}
|
std::vector<float> pcm_data(speech_data+sample_offset, speech_data+sample_offset+step);
|
vector<std::vector<int>> cut_segments = vad_online_handle->Infer(pcm_data, is_final);
|
vad_segments.insert(vad_segments.end(), cut_segments.begin(), cut_segments.end());
|
}
|
|
int speech_start_i = -1, speech_end_i =-1;
|
for(vector<int> vad_segment:vad_segments)
|
{
|
if(vad_segment.size() != 2){
|
LOG(ERROR) << "Size of vad_segment is not 2.";
|
break;
|
}
|
if(vad_segment[0] != -1){
|
speech_start_i = vad_segment[0];
|
}
|
if(vad_segment[1] != -1){
|
speech_end_i = vad_segment[1];
|
}
|
|
if(speech_start_i!=-1 && speech_end_i!=-1){
|
frame = new AudioFrame();
|
int start = speech_start_i*seg_sample;
|
int end = speech_end_i*seg_sample;
|
frame->SetStart(start);
|
frame->SetEnd(end);
|
frame_queue.push(frame);
|
frame = nullptr;
|
speech_start_i=-1;
|
speech_end_i=-1;
|
}
|
}
|
}
|
|
void Audio::Split(VadModel* vad_obj, vector<std::vector<int>>& vad_segments, bool input_finished)
|
{
|
AudioFrame *frame;
|
|
frame = frame_queue.front();
|
frame_queue.pop();
|
int sp_len = frame->GetLen();
|
delete frame;
|
frame = nullptr;
|
|
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
|
vad_segments = vad_obj->Infer(pcm_data, input_finished);
|
}
|
|
// 2pass
|
void Audio::Split(VadModel* vad_obj, int chunk_len, bool input_finished, ASR_TYPE asr_mode)
|
{
|
AudioFrame *frame;
|
|
frame = frame_queue.front();
|
frame_queue.pop();
|
int sp_len = frame->GetLen();
|
delete frame;
|
frame = nullptr;
|
|
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
|
vector<std::vector<int>> vad_segments = vad_obj->Infer(pcm_data, input_finished);
|
|
speech_end += sp_len/seg_sample;
|
if(vad_segments.size() == 0){
|
if(speech_start != -1){
|
int start = speech_start*seg_sample;
|
int end = speech_end*seg_sample;
|
int buff_len = end-start;
|
int step = chunk_len;
|
|
if(asr_mode != ASR_OFFLINE){
|
if(buff_len >= step){
|
frame = new AudioFrame(step);
|
frame->global_start = speech_start;
|
frame->global_end = speech_start + step/seg_sample;
|
frame->data = (float*)malloc(sizeof(float) * step);
|
memcpy(frame->data, all_samples.data()+start-offset, step*sizeof(float));
|
asr_online_queue.push(frame);
|
frame = nullptr;
|
speech_start += step/seg_sample;
|
}
|
}
|
}
|
}else{
|
for(auto vad_segment: vad_segments){
|
int speech_start_i=-1, speech_end_i=-1;
|
if(vad_segment[0] != -1){
|
speech_start_i = vad_segment[0];
|
}
|
if(vad_segment[1] != -1){
|
speech_end_i = vad_segment[1];
|
}
|
|
// [1, 100]
|
if(speech_start_i != -1 && speech_end_i != -1){
|
int start = speech_start_i*seg_sample;
|
int end = speech_end_i*seg_sample;
|
|
if(asr_mode != ASR_OFFLINE){
|
frame = new AudioFrame(end-start);
|
frame->is_final = true;
|
frame->global_start = speech_start_i;
|
frame->global_end = speech_end_i;
|
frame->data = (float*)malloc(sizeof(float) * (end-start));
|
memcpy(frame->data, all_samples.data()+start-offset, (end-start)*sizeof(float));
|
asr_online_queue.push(frame);
|
frame = nullptr;
|
}
|
|
if(asr_mode != ASR_ONLINE){
|
frame = new AudioFrame(end-start);
|
frame->is_final = true;
|
frame->global_start = speech_start_i;
|
frame->global_end = speech_end_i;
|
frame->data = (float*)malloc(sizeof(float) * (end-start));
|
memcpy(frame->data, all_samples.data()+start-offset, (end-start)*sizeof(float));
|
asr_offline_queue.push(frame);
|
frame = nullptr;
|
}
|
|
speech_start = -1;
|
speech_offline_start = -1;
|
// [70, -1]
|
}else if(speech_start_i != -1){
|
speech_start = speech_start_i;
|
speech_offline_start = speech_start_i;
|
|
int start = speech_start*seg_sample;
|
int end = speech_end*seg_sample;
|
int buff_len = end-start;
|
int step = chunk_len;
|
|
if(asr_mode != ASR_OFFLINE){
|
if(buff_len >= step){
|
frame = new AudioFrame(step);
|
frame->global_start = speech_start;
|
frame->global_end = speech_start + step/seg_sample;
|
frame->data = (float*)malloc(sizeof(float) * step);
|
memcpy(frame->data, all_samples.data()+start-offset, step*sizeof(float));
|
asr_online_queue.push(frame);
|
frame = nullptr;
|
speech_start += step/seg_sample;
|
}
|
}
|
|
}else if(speech_end_i != -1){ // [-1,100]
|
if(speech_start == -1 || speech_offline_start == -1){
|
LOG(ERROR) <<"Vad start is null while vad end is available. Set vad start 0" ;
|
speech_start = 0;
|
}
|
|
int start = speech_start*seg_sample;
|
int offline_start = speech_offline_start*seg_sample;
|
int end = speech_end_i*seg_sample;
|
int buff_len = end-start;
|
int step = chunk_len;
|
|
if(asr_mode != ASR_ONLINE){
|
frame = new AudioFrame(end-offline_start);
|
frame->is_final = true;
|
frame->global_start = speech_offline_start;
|
frame->global_end = speech_end_i;
|
frame->data = (float*)malloc(sizeof(float) * (end-offline_start));
|
memcpy(frame->data, all_samples.data()+offline_start-offset, (end-offline_start)*sizeof(float));
|
asr_offline_queue.push(frame);
|
frame = nullptr;
|
}
|
|
if(asr_mode != ASR_OFFLINE){
|
if(buff_len > 0){
|
for (int sample_offset = 0; sample_offset < buff_len; sample_offset += std::min(step, buff_len - sample_offset)) {
|
bool is_final = false;
|
if (sample_offset + step >= buff_len - 1) {
|
step = buff_len - sample_offset;
|
is_final = true;
|
}
|
frame = new AudioFrame(step);
|
frame->is_final = is_final;
|
frame->global_start = (int)((start+sample_offset)/seg_sample);
|
frame->global_end = frame->global_start + step/seg_sample;
|
frame->data = (float*)malloc(sizeof(float) * step);
|
memcpy(frame->data, all_samples.data()+start-offset+sample_offset, step*sizeof(float));
|
asr_online_queue.push(frame);
|
frame = nullptr;
|
}
|
}else{
|
frame = new AudioFrame(0);
|
frame->is_final = true;
|
frame->global_start = speech_start; // in this case start >= end
|
frame->global_end = speech_end_i;
|
asr_online_queue.push(frame);
|
frame = nullptr;
|
}
|
}
|
speech_start = -1;
|
speech_offline_start = -1;
|
}
|
}
|
}
|
|
// erase all_samples
|
int vector_cache = dest_sample_rate*2;
|
if(speech_offline_start == -1){
|
if(all_samples.size() > vector_cache){
|
int erase_num = all_samples.size() - vector_cache;
|
all_samples.erase(all_samples.begin(), all_samples.begin()+erase_num);
|
offset += erase_num;
|
}
|
}else{
|
int offline_start = speech_offline_start*seg_sample;
|
if(offline_start-offset > vector_cache){
|
int erase_num = offline_start-offset - vector_cache;
|
all_samples.erase(all_samples.begin(), all_samples.begin()+erase_num);
|
offset += erase_num;
|
}
|
}
|
|
}
|
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} // namespace funasr
|
