From 716e3fe5122bc96bdc40fde71a50e4dc7f49b759 Mon Sep 17 00:00:00 2001
From: lyblsgo <lyblsgo@163.com>
Date: 星期五, 21 四月 2023 17:12:10 +0800
Subject: [PATCH] rename executable file;rm some unnecessary deps
---
funasr/runtime/onnxruntime/src/Audio.cpp | 20 ++++++++++----------
1 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/funasr/runtime/onnxruntime/src/Audio.cpp b/funasr/runtime/onnxruntime/src/Audio.cpp
index 72e90a2..5c10cf1 100644
--- a/funasr/runtime/onnxruntime/src/Audio.cpp
+++ b/funasr/runtime/onnxruntime/src/Audio.cpp
@@ -187,13 +187,13 @@
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()
{
- return (float)speech_len / model_sample_rate;
+ return (float)speech_len / MODEL_SAMPLE_RATE;
}
void Audio::wavResample(int32_t sampling_rate, const float *waveform,
@@ -203,9 +203,9 @@
"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;
@@ -213,7 +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
@@ -270,7 +270,7 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
wavResample(*sampling_rate, speech_data, speech_len);
}
@@ -317,7 +317,7 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
wavResample(*sampling_rate, speech_data, speech_len);
}
@@ -360,7 +360,7 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
wavResample(*sampling_rate, speech_data, speech_len);
}
@@ -411,7 +411,7 @@
}
//resample
- if(*sampling_rate != model_sample_rate){
+ if(*sampling_rate != MODEL_SAMPLE_RATE){
wavResample(*sampling_rate, speech_data, speech_len);
}
@@ -511,7 +511,7 @@
std::vector<float> pcm_data(speech_data, speech_data+sp_len);
vector<std::vector<int>> vad_segments = pRecogObj->vad_seg(pcm_data);
- int seg_sample = model_sample_rate/1000;
+ int seg_sample = MODEL_SAMPLE_RATE/1000;
for(vector<int> segment:vad_segments)
{
frame = new AudioFrame();
--
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