From 7ab2e5cf22bbb31808bcacf84c054c710e4e6a93 Mon Sep 17 00:00:00 2001
From: Yabin Li <wucong.lyb@alibaba-inc.com>
Date: 星期一, 24 四月 2023 16:19:17 +0800
Subject: [PATCH] Merge pull request #400 from alibaba-damo-academy/dev_knf
---
funasr/runtime/onnxruntime/src/audio.cpp | 150 ++++++++++++++++----------------------------------
1 files changed, 48 insertions(+), 102 deletions(-)
diff --git a/funasr/runtime/onnxruntime/src/Audio.cpp b/funasr/runtime/onnxruntime/src/audio.cpp
similarity index 76%
rename from funasr/runtime/onnxruntime/src/Audio.cpp
rename to funasr/runtime/onnxruntime/src/audio.cpp
index 38b6de8..ef48fa1 100644
--- a/funasr/runtime/onnxruntime/src/Audio.cpp
+++ b/funasr/runtime/onnxruntime/src/audio.cpp
@@ -6,7 +6,7 @@
#include <fstream>
#include <assert.h>
-#include "Audio.h"
+#include "audio.h"
#include "precomp.h"
using namespace std;
@@ -128,39 +128,30 @@
start = 0;
};
AudioFrame::~AudioFrame(){};
-int AudioFrame::set_start(int val)
+int AudioFrame::SetStart(int val)
{
start = val < 0 ? 0 : val;
return start;
};
-int AudioFrame::set_end(int val, int max_len)
+int AudioFrame::SetEnd(int val)
{
-
- float num_samples = val - start;
- float frame_length = 400;
- float frame_shift = 160;
- float num_new_samples =
- ceil((num_samples - frame_length) / frame_shift) * frame_shift + frame_length;
-
- end = start + num_new_samples;
- len = (int)num_new_samples;
- if (end > max_len)
- printf("frame end > max_len!!!!!!!\n");
+ end = val;
+ len = end - start;
return end;
};
-int AudioFrame::get_start()
+int AudioFrame::GetStart()
{
return start;
};
-int AudioFrame::get_len()
+int AudioFrame::GetLen()
{
return len;
};
-int AudioFrame::disp()
+int AudioFrame::Disp()
{
printf("not imp!!!!\n");
@@ -194,27 +185,27 @@
}
}
-void Audio::disp()
+void Audio::Disp()
{
- printf("Audio time is %f s. len is %d\n", (float)speech_len / model_sample_rate,
+ printf("Audio time is %f s. len is %d\n", (float)speech_len / MODEL_SAMPLE_RATE,
speech_len);
}
-float Audio::get_time_len()
+float Audio::GetTimeLen()
{
- return (float)speech_len / model_sample_rate;
+ return (float)speech_len / MODEL_SAMPLE_RATE;
}
-void Audio::wavResample(int32_t sampling_rate, const float *waveform,
+void Audio::WavResample(int32_t sampling_rate, const float *waveform,
int32_t n)
{
printf(
"Creating a resampler:\n"
" in_sample_rate: %d\n"
" output_sample_rate: %d\n",
- sampling_rate, static_cast<int32_t>(model_sample_rate));
+ sampling_rate, static_cast<int32_t>(MODEL_SAMPLE_RATE));
float min_freq =
- std::min<int32_t>(sampling_rate, model_sample_rate);
+ std::min<int32_t>(sampling_rate, MODEL_SAMPLE_RATE);
float lowpass_cutoff = 0.99 * 0.5 * min_freq;
int32_t lowpass_filter_width = 6;
@@ -222,7 +213,7 @@
//auto resampler = new LinearResample(
// sampling_rate, model_sample_rate, lowpass_cutoff, lowpass_filter_width);
auto resampler = std::make_unique<LinearResample>(
- sampling_rate, model_sample_rate, lowpass_cutoff, lowpass_filter_width);
+ sampling_rate, MODEL_SAMPLE_RATE, lowpass_cutoff, lowpass_filter_width);
std::vector<float> samples;
resampler->Resample(waveform, n, true, &samples);
//reset speech_data
@@ -235,7 +226,7 @@
copy(samples.begin(), samples.end(), speech_data);
}
-bool Audio::loadwav(const char *filename, int32_t* sampling_rate)
+bool Audio::LoadWav(const char *filename, int32_t* sampling_rate)
{
WaveHeader header;
if (speech_data != NULL) {
@@ -279,8 +270,8 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
- wavResample(*sampling_rate, speech_data, speech_len);
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
+ WavResample(*sampling_rate, speech_data, speech_len);
}
AudioFrame* frame = new AudioFrame(speech_len);
@@ -292,7 +283,7 @@
return false;
}
-bool Audio::loadwav(const char* buf, int nFileLen, int32_t* sampling_rate)
+bool Audio::LoadWav(const char* buf, int n_file_len, int32_t* sampling_rate)
{
WaveHeader header;
if (speech_data != NULL) {
@@ -326,8 +317,8 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
- wavResample(*sampling_rate, speech_data, speech_len);
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
+ WavResample(*sampling_rate, speech_data, speech_len);
}
AudioFrame* frame = new AudioFrame(speech_len);
@@ -339,7 +330,7 @@
return false;
}
-bool Audio::loadpcmwav(const char* buf, int nBufLen, int32_t* sampling_rate)
+bool Audio::LoadPcmwav(const char* buf, int n_buf_len, int32_t* sampling_rate)
{
if (speech_data != NULL) {
free(speech_data);
@@ -349,7 +340,7 @@
}
offset = 0;
- speech_len = nBufLen / 2;
+ speech_len = n_buf_len / 2;
speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
if (speech_buff)
{
@@ -369,8 +360,8 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
- wavResample(*sampling_rate, speech_data, speech_len);
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
+ WavResample(*sampling_rate, speech_data, speech_len);
}
AudioFrame* frame = new AudioFrame(speech_len);
@@ -382,7 +373,7 @@
return false;
}
-bool Audio::loadpcmwav(const char* filename, int32_t* sampling_rate)
+bool Audio::LoadPcmwav(const char* filename, int32_t* sampling_rate)
{
if (speech_data != NULL) {
free(speech_data);
@@ -397,10 +388,10 @@
if (fp == nullptr)
return false;
fseek(fp, 0, SEEK_END);
- uint32_t nFileLen = ftell(fp);
+ uint32_t n_file_len = ftell(fp);
fseek(fp, 0, SEEK_SET);
- speech_len = (nFileLen) / 2;
+ speech_len = (n_file_len) / 2;
speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len);
if (speech_buff)
{
@@ -420,8 +411,8 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
- wavResample(*sampling_rate, speech_data, speech_len);
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
+ WavResample(*sampling_rate, speech_data, speech_len);
}
AudioFrame* frame = new AudioFrame(speech_len);
@@ -434,7 +425,7 @@
}
-int Audio::fetch_chunck(float *&dout, int len)
+int Audio::FetchChunck(float *&dout, int len)
{
if (offset >= speech_align_len) {
dout = NULL;
@@ -455,14 +446,14 @@
}
}
-int Audio::fetch(float *&dout, int &len, int &flag)
+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->get_start();
- len = frame->get_len();
+ dout = speech_data + frame->GetStart();
+ len = frame->GetLen();
delete frame;
flag = S_END;
return 1;
@@ -471,9 +462,8 @@
}
}
-void Audio::padding()
+void Audio::Padding()
{
-
float num_samples = speech_len;
float frame_length = 400;
float frame_shift = 160;
@@ -509,71 +499,27 @@
delete frame;
}
-#define UNTRIGGERED 0
-#define TRIGGERED 1
-
-#define SPEECH_LEN_5S (16000 * 5)
-#define SPEECH_LEN_10S (16000 * 10)
-#define SPEECH_LEN_20S (16000 * 20)
-#define SPEECH_LEN_30S (16000 * 30)
-
-/*
-void Audio::split()
+void Audio::Split(Model* recog_obj)
{
- VadInst *handle = WebRtcVad_Create();
- WebRtcVad_Init(handle);
- WebRtcVad_set_mode(handle, 2);
- int window_size = 10;
- AudioWindow audiowindow(window_size);
- int status = UNTRIGGERED;
- int offset = 0;
- int fs = 16000;
- int step = 480;
-
AudioFrame *frame;
frame = frame_queue.front();
frame_queue.pop();
+ int sp_len = frame->GetLen();
delete frame;
frame = NULL;
- while (offset < speech_len - step) {
- int n = WebRtcVad_Process(handle, fs, speech_buff + offset, step);
- if (status == UNTRIGGERED && audiowindow.put(n) >= window_size - 1) {
- frame = new AudioFrame();
- int start = offset - step * (window_size - 1);
- frame->set_start(start);
- status = TRIGGERED;
- } else if (status == TRIGGERED) {
- int win_weight = audiowindow.put(n);
- int voice_len = (offset - frame->get_start());
- int gap = 0;
- if (voice_len < SPEECH_LEN_5S) {
- offset += step;
- continue;
- } else if (voice_len < SPEECH_LEN_10S) {
- gap = 1;
- } else if (voice_len < SPEECH_LEN_20S) {
- gap = window_size / 5;
- } else {
- gap = window_size / 2;
- }
-
- if (win_weight < gap) {
- status = UNTRIGGERED;
- offset = frame->set_end(offset, speech_align_len);
- frame_queue.push(frame);
- frame = NULL;
- }
- }
- offset += step;
- }
-
- if (frame != NULL) {
- frame->set_end(speech_len, speech_align_len);
+ std::vector<float> pcm_data(speech_data, speech_data+sp_len);
+ vector<std::vector<int>> vad_segments = recog_obj->VadSeg(pcm_data);
+ int seg_sample = MODEL_SAMPLE_RATE/1000;
+ 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 = NULL;
}
- WebRtcVad_Free(handle);
}
-*/
\ No newline at end of file
--
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