Merge branch 'dev_infer' of https://github.com/alibaba-damo-academy/FunASR into dev_infer
| | |
| | | ./runtime/onnxruntime_python.md |
| | | ./runtime/onnxruntime_cpp.md |
| | | ./runtime/libtorch_python.md |
| | | ./runtime/grpc_python.md |
| | | ./runtime/grpc_cpp.md |
| | | ./runtime/html5.md |
| | | ./runtime/websocket_python.md |
| | | ./runtime/websocket_cpp.md |
| | | ./runtime/grpc_python.md |
| | | ./runtime/grpc_cpp.md |
| | | |
| | | |
| | | .. toctree:: |
| | | :maxdepth: 1 |
| New file |
| | |
| | | ../../funasr/runtime/html5/readme.md |
| | |
| | | --data_dir ${feats_dir}/data \ |
| | | --train_set ${train_set} \ |
| | | --valid_set ${valid_set} \ |
| | | --data_file_names "wav.scp,text,embed.scp" \ |
| | | --data_file_names "wav.scp,text,embeds.scp" \ |
| | | --cmvn_file ${feats_dir}/data/${train_set}/cmvn/cmvn.mvn \ |
| | | --speed_perturb ${speed_perturb} \ |
| | | --resume true \ |
| | |
| | | --data_dir ${feats_dir}/data \ |
| | | --train_set ${train_set} \ |
| | | --valid_set ${valid_set} \ |
| | | --data_file_names "wav.scp,text,embed.scp" \ |
| | | --data_file_names "wav.scp,text,embeds.scp" \ |
| | | --cmvn_file ${feats_dir}/data/${train_set}/cmvn/cmvn.mvn \ |
| | | --speed_perturb ${speed_perturb} \ |
| | | --dataset_type $dataset_type \ |
| New file |
| | |
| | | beam_size: 5 |
| | | penalty: 0.0 |
| | | maxlenratio: 0.0 |
| | | minlenratio: 0.0 |
| | | ctc_weight: 0.3 |
| | | lm_weight: 0.0 |
| New file |
| | |
| | | beam_size: 60 |
| | | penalty: 0.0 |
| | | maxlenratio: 0.0 |
| | | minlenratio: 0.0 |
| | | ctc_weight: 0.3 |
| | | lm_weight: 0.0 |
| | |
| | | asr_config=conf/train_asr_conformer.yaml |
| | | model_dir="baseline_$(basename "${asr_config}" .yaml)_${lang}_${token_type}_${tag}" |
| | | |
| | | inference_config=conf/decode_asr_transformer.yaml |
| | | #inference_config=conf/decode_asr_transformer_beam60_ctc0.3.yaml |
| | | inference_config=conf/decode_asr_transformer_ctc0.3_beam5yaml |
| | | #inference_config=conf/decode_asr_transformer_ctc0.3_beam60.yaml |
| | | inference_asr_model=valid.acc.ave_10best.pb |
| | | |
| | | # you can set gpu num for decoding here |
| New file |
| | |
| | | beam_size: 1 |
| | | penalty: 0.0 |
| | | maxlenratio: 0.0 |
| | | minlenratio: 0.0 |
| | | ctc_weight: 0.3 |
| | | lm_weight: 0.0 |
| New file |
| | |
| | | beam_size: 20 |
| | | penalty: 0.0 |
| | | maxlenratio: 0.0 |
| | | minlenratio: 0.0 |
| | | ctc_weight: 0.3 |
| | | lm_weight: 0.0 |
| New file |
| | |
| | | beam_size: 5 |
| | | penalty: 0.0 |
| | | maxlenratio: 0.0 |
| | | minlenratio: 0.0 |
| | | ctc_weight: 0.3 |
| | | lm_weight: 0.0 |
| | |
| | | asr_config=conf/train_asr_conformer.yaml |
| | | model_dir="baseline_$(basename "${asr_config}" .yaml)_${lang}_${token_type}_${tag}" |
| | | |
| | | inference_config=conf/decode_asr_transformer.yaml |
| | | #inference_config=conf/decode_asr_transformer_beam60_ctc0.3.yaml |
| | | #inference_config=conf/decode_asr_transformer_ctc0.3_beam1.yaml |
| | | inference_config=conf/decode_asr_transformer_ctc0.3_beam5.yaml |
| | | #inference_config=conf/decode_asr_transformer_ctc0.3_beam20.yaml |
| | | inference_asr_model=valid.acc.ave_10best.pb |
| | | |
| | | # you can set gpu num for decoding here |
| | |
| | | inference_pipeline = pipeline( |
| | | task=Tasks.punctuation, |
| | | model='damo/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727', |
| | | output_dir="./tmp/" |
| | | model_revision = 'v1.0.2' |
| | | ) |
| | | |
| | | ##################text二进制数据##################### |
| | |
| | | return {'inputs': np.ones((1, text_length), dtype=np.int64), |
| | | 'text_lengths': np.array([text_length,], dtype=np.int32), |
| | | 'vad_masks': np.ones((1, 1, text_length, text_length), dtype=np.float32), |
| | | 'sub_masks': np.tril(np.ones((text_length, text_length), dtype=np.float32))[None, None, :, :].astype(np.float32) |
| | | 'sub_masks': np.ones((1, 1, text_length, text_length), dtype=np.float32), |
| | | } |
| | | |
| | | def _run(feed_dict): |
| | |
| | | ```
|
| | |
|
| | | ### javascript
|
| | | [html5录音](https://github.com/xiangyuecn/Recorder)
|
| | | [html5 recorder.js](https://github.com/xiangyuecn/Recorder)
|
| | | ```shell
|
| | | Recorder
|
| | | ```
|
| | |
|
| | | ### demo页面如下
|
| | | 
|
| | | ### demo
|
| | | 
|
| | |
|
| | | ## 两种ws_server_online连接模式
|
| | | ### 1)直接连接模式,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> wss asr online srv(证书生成请往后看)
|
| | | ## wss or ws protocol for ws_server_online
|
| | | 1) wss: browser microphone data --> html5 demo server --> js wss api --> wss asr online srv #for certificate generation just look back
|
| | |
|
| | | ### 2)nginx中转,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> nginx服务 --> ws asr online srv
|
| | | 2) ws: browser microphone data --> html5 demo server --> js wss api --> nginx wss server --> ws asr online srv
|
| | |
|
| | | ## 1.html5 demo服务启动
|
| | | ### 启动html5服务,需要ssl证书(自己生成请往后看)
|
| | | ## 1.html5 demo start
|
| | | ### ssl certificate is required
|
| | |
|
| | | ```shell
|
| | | usage: h5Server.py [-h] [--host HOST] [--port PORT] [--certfile CERTFILE]
|
| | | [--keyfile KEYFILE]
|
| | | python h5Server.py --port 1337
|
| | | ```
|
| | | ## 2.启动ws or wss asr online srv
|
| | | [具体请看online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
| | | online asr提供两种ws和wss模式,wss模式可以直接启动,无需nginx中转。否则需要通过nginx将wss转发到该online asr的ws端口上
|
| | | ### wss方式
|
| | | ## 2.asr online srv start
|
| | | [detail for online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
| | | Online asr provides wss or ws way. if started in ws way, nginx is required for relay.
|
| | | ### wss way, ssl certificate is required
|
| | | ```shell
|
| | | python ws_server_online.py --certfile server.crt --keyfile server.key --port 5921
|
| | | ```
|
| | | ### ws方式
|
| | | ### ws way
|
| | | ```shell
|
| | | python ws_server_online.py --port 5921
|
| | | ```
|
| | | ## 3.修改wsconnecter.js里asr接口地址
|
| | | wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
| | | ## 3.modify asr address in wsconnecter.js according to your environment
|
| | | asr address in wsconnecter.js must be wss, just like
|
| | | var Uri = "wss://xxx:xxx/"
|
| | |
|
| | | ## 4.浏览器打开地址测试
|
| | | https://127.0.0.1:1337/static/index.html
|
| | | ## 4.open browser to access html5 demo
|
| | | https://youraddress:port/static/index.html
|
| | |
|
| | |
|
| | |
|
| | |
|
| | | ## 自行生成证书
|
| | | 生成证书(注意这种证书并不能被所有浏览器认可,部分手动授权可以访问,最好使用其他认证的官方ssl证书)
|
| | | ## certificate generation by yourself
|
| | | generated certificate may not suitable for all browsers due to security concerns. you'd better buy or download an authenticated ssl certificate from authorized agency.
|
| | |
|
| | | ```shell
|
| | | ### 1)生成私钥,按照提示填写内容
|
| | | ### 1) Generate a private key
|
| | | openssl genrsa -des3 -out server.key 1024
|
| | |
|
| | | ### 2)生成csr文件 ,按照提示填写内容
|
| | | ### 2) Generate a csr file
|
| | | openssl req -new -key server.key -out server.csr
|
| | |
|
| | | ### 去掉pass
|
| | | ### 3) Remove pass
|
| | | cp server.key server.key.org
|
| | | openssl rsa -in server.key.org -out server.key
|
| | |
|
| | | ### 生成crt文件,有效期1年(365天)
|
| | | ### 4) Generated a crt file, valid for 1 year
|
| | | openssl x509 -req -days 365 -in server.csr -signkey server.key -out server.crt
|
| | | ```
|
| | |
|
| | | ## nginx配置说明(了解的可以跳过)
|
| | | h5打开麦克风需要https协议,同时后端的asr websocket也必须是wss协议,如果[online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)以ws方式运行,我们可以通过nginx配置实现wss协议到ws协议的转换。
|
| | |
|
| | | ### nginx转发配置示例
|
| | | ## nginx configuration (you can skip it if you known)
|
| | | https and wss protocol are required by browsers when want to open microphone and websocket. |
| | | if [online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket) run in ws way, you should use nginx to convert wss to ws.
|
| | | ### nginx wss->ws configuration example
|
| | | ```shell
|
| | | events { [0/1548]
|
| | | worker_connections 1024;
|
| New file |
| | |
| | | # online asr demo for html5
|
| | |
|
| | | ## requirement
|
| | | ### python
|
| | | ```shell
|
| | | flask
|
| | | gevent
|
| | | pyOpenSSL
|
| | | ```
|
| | |
|
| | | ### javascript
|
| | | [html5录音](https://github.com/xiangyuecn/Recorder)
|
| | | ```shell
|
| | | Recorder |
| | | ```
|
| | |
|
| | | ### demo页面如下
|
| | | 
|
| | |
|
| | | ## 两种ws_server_online连接模式
|
| | | ### 1)直接连接模式,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> wss asr online srv(证书生成请往后看)
|
| | |
|
| | | ### 2)nginx中转,浏览器https麦克风 --> html5 demo服务 --> js wss接口 --> nginx服务 --> ws asr online srv
|
| | |
|
| | | ## 1.html5 demo服务启动
|
| | | ### 启动html5服务,需要ssl证书(自己生成请往后看)
|
| | |
|
| | | ```shell
|
| | | usage: h5Server.py [-h] [--host HOST] [--port PORT] [--certfile CERTFILE]
|
| | | [--keyfile KEYFILE]
|
| | | python h5Server.py --port 1337
|
| | | ```
|
| | | ## 2.启动ws or wss asr online srv
|
| | | [具体请看online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)
|
| | | online asr提供两种ws和wss模式,wss模式可以直接启动,无需nginx中转。否则需要通过nginx将wss转发到该online asr的ws端口上
|
| | | ### wss方式
|
| | | ```shell
|
| | | python ws_server_online.py --certfile server.crt --keyfile server.key --port 5921
|
| | | ```
|
| | | ### ws方式
|
| | | ```shell
|
| | | python ws_server_online.py --port 5921
|
| | | ```
|
| | | ## 3.修改wsconnecter.js里asr接口地址
|
| | | wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
| | | var Uri = "wss://xxx:xxx/" |
| | |
|
| | | ## 4.浏览器打开地址测试
|
| | | https://127.0.0.1:1337/static/index.html
|
| | |
|
| | |
|
| | |
|
| | |
|
| | | ## 自行生成证书
|
| | | 生成证书(注意这种证书并不能被所有浏览器认可,部分手动授权可以访问,最好使用其他认证的官方ssl证书)
|
| | |
|
| | | ```shell
|
| | | ### 1)生成私钥,按照提示填写内容
|
| | | openssl genrsa -des3 -out server.key 1024
|
| | | |
| | | ### 2)生成csr文件 ,按照提示填写内容
|
| | | openssl req -new -key server.key -out server.csr
|
| | | |
| | | ### 去掉pass
|
| | | cp server.key server.key.org |
| | | openssl rsa -in server.key.org -out server.key
|
| | | |
| | | ### 生成crt文件,有效期1年(365天)
|
| | | openssl x509 -req -days 365 -in server.csr -signkey server.key -out server.crt
|
| | | ```
|
| | |
|
| | | ## nginx配置说明(了解的可以跳过)
|
| | | h5打开麦克风需要https协议,同时后端的asr websocket也必须是wss协议,如果[online asr](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/websocket)以ws方式运行,我们可以通过nginx配置实现wss协议到ws协议的转换。
|
| | |
|
| | | ### nginx转发配置示例
|
| | | ```shell
|
| | | events { [0/1548]
|
| | | worker_connections 1024;
|
| | | accept_mutex on;
|
| | | }
|
| | | http {
|
| | | error_log error.log;
|
| | | access_log access.log;
|
| | | server {
|
| | |
|
| | | listen 5921 ssl http2; # nginx listen port for wss
|
| | | server_name www.test.com;
|
| | |
|
| | | ssl_certificate /funasr/server.crt;
|
| | | ssl_certificate_key /funasr/server.key;
|
| | | ssl_protocols TLSv1 TLSv1.1 TLSv1.2;
|
| | | ssl_ciphers HIGH:!aNULL:!MD5;
|
| | |
|
| | | location /wss/ {
|
| | |
|
| | |
|
| | | proxy_pass http://127.0.0.1:1111/; # asr online model ws address and port
|
| | | proxy_http_version 1.1;
|
| | | proxy_set_header Upgrade $http_upgrade;
|
| | | proxy_set_header Connection "upgrade";
|
| | | proxy_read_timeout 600s;
|
| | |
|
| | | }
|
| | | }
|
| | | ```
|
| | | ### 修改wsconnecter.js里asr接口地址
|
| | | wsconnecter.js里配置online asr服务地址路径,这里配置的是wss端口
|
| | | var Uri = "wss://xxx:xxx/wss/" |
| | | ## Acknowledge
|
| | | 1. This project is maintained by [FunASR community](https://github.com/alibaba-damo-academy/FunASR).
|
| | | 2. We acknowledge [AiHealthx](http://www.aihealthx.com/) for contributing the html5 demo. |
| | |
| | | /* 2021-2023 by zhaoming,mali aihealthx.com */
|
| | |
|
| | | function WebSocketConnectMethod( config ) { //定义socket连接方法类
|
| | | var Uri = "wss://111.205.137.58:5821/wss/" //设置wss asr online接口地址 如 wss://X.X.X.X:port/wss/
|
| | | var Uri = "wss://30.220.136.139:5921/" // var Uri = "wss://30.221.177.46:5921/" //设置wss asr online接口地址 如 wss://X.X.X.X:port/wss/
|
| | | var speechSokt;
|
| | | var connKeeperID;
|
| | |
|
| | |
| | | public: |
| | | virtual ~VadModel(){}; |
| | | virtual void InitVad(const std::string &vad_model, const std::string &vad_cmvn, const std::string &vad_config, int thread_num)=0; |
| | | virtual std::vector<std::vector<int>> Infer(const std::vector<float> &waves)=0; |
| | | virtual std::vector<std::vector<int>> Infer(std::vector<float> &waves, bool input_finished=true)=0; |
| | | virtual void ReadModel(const char* vad_model)=0; |
| | | virtual void LoadConfigFromYaml(const char* filename)=0; |
| | | virtual void FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats, |
| | | const std::vector<float> &waves)=0; |
| | | virtual void LfrCmvn(std::vector<std::vector<float>> &vad_feats)=0; |
| | | virtual void Forward( |
| | | const std::vector<std::vector<float>> &chunk_feats, |
| | | std::vector<std::vector<float>> *out_prob)=0; |
| | | std::vector<float> &waves)=0; |
| | | virtual void LoadCmvn(const char *filename)=0; |
| | | virtual void InitCache()=0; |
| | | }; |
| | |
| | | ### funasr-onnx-offline-rtf |
| | | ```shell |
| | | ./funasr-onnx-offline-rtf --model-dir <string> [--quantize <string>] |
| | | [--vad-dir <string>] [--vad-quant <string>] |
| | | [--punc-dir <string>] [--punc-quant <string>] |
| | | --wav-path <string> --thread-num <int32_t> |
| | | [--] [--version] [-h] |
| | | Where: |
| | |
| | | (required) the model path, which contains model.onnx, config.yaml, am.mvn |
| | | --quantize <string> |
| | | false (Default), load the model of model.onnx in model_dir. If set true, load the model of model_quant.onnx in model_dir |
| | | |
| | | --vad-dir <string> |
| | | the vad model path, which contains model.onnx, vad.yaml, vad.mvn |
| | | --vad-quant <string> |
| | | false (Default), load the model of model.onnx in vad_dir. If set true, load the model of model_quant.onnx in vad_dir |
| | | |
| | | --punc-dir <string> |
| | | the punc model path, which contains model.onnx, punc.yaml |
| | | --punc-quant <string> |
| | | false (Default), load the model of model.onnx in punc_dir. If set true, load the model of model_quant.onnx in punc_dir |
| | | |
| | | --wav-path <string> |
| | | (required) the input could be: |
| | | wav_path, e.g.: asr_example.wav; |
| | |
| | | } |
| | | |
| | | // get 4 caches outputs,each size is 128*19 |
| | | for (int i = 1; i < 5; i++) { |
| | | float* data = vad_ort_outputs[i].GetTensorMutableData<float>(); |
| | | memcpy(in_cache_[i-1].data(), data, sizeof(float) * 128*19); |
| | | } |
| | | // for (int i = 1; i < 5; i++) { |
| | | // float* data = vad_ort_outputs[i].GetTensorMutableData<float>(); |
| | | // memcpy(in_cache_[i-1].data(), data, sizeof(float) * 128*19); |
| | | // } |
| | | } |
| | | |
| | | void FsmnVad::FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats, |
| | | const std::vector<float> &waves) { |
| | | std::vector<float> &waves) { |
| | | knf::OnlineFbank fbank(fbank_opts); |
| | | |
| | | fbank.AcceptWaveform(sample_rate, &waves[0], waves.size()); |
| | | std::vector<float> buf(waves.size()); |
| | | for (int32_t i = 0; i != waves.size(); ++i) { |
| | | buf[i] = waves[i] * 32768; |
| | | } |
| | | fbank.AcceptWaveform(sample_rate, buf.data(), buf.size()); |
| | | int32_t frames = fbank.NumFramesReady(); |
| | | for (int32_t i = 0; i != frames; ++i) { |
| | | const float *frame = fbank.GetFrame(i); |
| | |
| | | } |
| | | |
| | | std::vector<std::vector<int>> |
| | | FsmnVad::Infer(const std::vector<float> &waves) { |
| | | FsmnVad::Infer(std::vector<float> &waves, bool input_finished) { |
| | | std::vector<std::vector<float>> vad_feats; |
| | | std::vector<std::vector<float>> vad_probs; |
| | | FbankKaldi(vad_sample_rate_, vad_feats, waves); |
| | |
| | | ~FsmnVad(); |
| | | void Test(); |
| | | void InitVad(const std::string &vad_model, const std::string &vad_cmvn, const std::string &vad_config, int thread_num); |
| | | std::vector<std::vector<int>> Infer(const std::vector<float> &waves); |
| | | std::vector<std::vector<int>> Infer(std::vector<float> &waves, bool input_finished=true); |
| | | void Reset(); |
| | | |
| | | private: |
| | |
| | | std::vector<const char *> *in_names, std::vector<const char *> *out_names); |
| | | |
| | | void FbankKaldi(float sample_rate, std::vector<std::vector<float>> &vad_feats, |
| | | const std::vector<float> &waves); |
| | | std::vector<float> &waves); |
| | | |
| | | void LfrCmvn(std::vector<std::vector<float>> &vad_feats); |
| | | |
| | |
| | | // warm up |
| | | for (size_t i = 0; i < 1; i++) |
| | | { |
| | | FUNASR_RESULT result=FunASRInfer(asr_handle, wav_list[0].c_str(), RASR_NONE, NULL, 16000); |
| | | FUNASR_RESULT result=FunOfflineInfer(asr_handle, wav_list[0].c_str(), RASR_NONE, NULL, 16000); |
| | | } |
| | | |
| | | while (true) { |
| | |
| | | } |
| | | |
| | | gettimeofday(&start, NULL); |
| | | FUNASR_RESULT result=FunASRInfer(asr_handle, wav_list[i].c_str(), RASR_NONE, NULL, 16000); |
| | | FUNASR_RESULT result=FunOfflineInfer(asr_handle, wav_list[i].c_str(), RASR_NONE, NULL, 16000); |
| | | |
| | | gettimeofday(&end, NULL); |
| | | seconds = (end.tv_sec - start.tv_sec); |
| | |
| | | TCLAP::CmdLine cmd("funasr-onnx-offline-rtf", ' ', "1.0"); |
| | | TCLAP::ValueArg<std::string> model_dir("", MODEL_DIR, "the model path, which contains model.onnx, config.yaml, am.mvn", true, "", "string"); |
| | | TCLAP::ValueArg<std::string> quantize("", QUANTIZE, "false (Default), load the model of model.onnx in model_dir. If set true, load the model of model_quant.onnx in model_dir", false, "false", "string"); |
| | | TCLAP::ValueArg<std::string> vad_dir("", VAD_DIR, "the vad model path, which contains model.onnx, vad.yaml, vad.mvn", false, "", "string"); |
| | | TCLAP::ValueArg<std::string> vad_quant("", VAD_QUANT, "false (Default), load the model of model.onnx in vad_dir. If set true, load the model of model_quant.onnx in vad_dir", false, "false", "string"); |
| | | TCLAP::ValueArg<std::string> punc_dir("", PUNC_DIR, "the punc model path, which contains model.onnx, punc.yaml", false, "", "string"); |
| | | TCLAP::ValueArg<std::string> punc_quant("", PUNC_QUANT, "false (Default), load the model of model.onnx in punc_dir. If set true, load the model of model_quant.onnx in punc_dir", false, "false", "string"); |
| | | |
| | | TCLAP::ValueArg<std::string> wav_path("", WAV_PATH, "the input could be: wav_path, e.g.: asr_example.wav; pcm_path, e.g.: asr_example.pcm; wav.scp, kaldi style wav list (wav_id \t wav_path)", true, "", "string"); |
| | | TCLAP::ValueArg<std::int32_t> thread_num("", THREAD_NUM, "multi-thread num for rtf", true, 0, "int32_t"); |
| | | |
| | | cmd.add(model_dir); |
| | | cmd.add(quantize); |
| | | cmd.add(vad_dir); |
| | | cmd.add(vad_quant); |
| | | cmd.add(punc_dir); |
| | | cmd.add(punc_quant); |
| | | cmd.add(wav_path); |
| | | cmd.add(thread_num); |
| | | cmd.parse(argc, argv); |
| | |
| | | std::map<std::string, std::string> model_path; |
| | | GetValue(model_dir, MODEL_DIR, model_path); |
| | | GetValue(quantize, QUANTIZE, model_path); |
| | | GetValue(vad_dir, VAD_DIR, model_path); |
| | | GetValue(vad_quant, VAD_QUANT, model_path); |
| | | GetValue(punc_dir, PUNC_DIR, model_path); |
| | | GetValue(punc_quant, PUNC_QUANT, model_path); |
| | | GetValue(wav_path, WAV_PATH, model_path); |
| | | |
| | | struct timeval start, end; |
| | | gettimeofday(&start, NULL); |
| | | FUNASR_HANDLE asr_handle=FunASRInit(model_path, 1); |
| | | FUNASR_HANDLE asr_handle=FunOfflineInit(model_path, 1); |
| | | |
| | | if (!asr_handle) |
| | | { |
| | |
| | | long modle_init_micros = ((seconds * 1000000) + end.tv_usec) - (start.tv_usec); |
| | | LOG(INFO) << "Model initialization takes " << (double)modle_init_micros / 1000000 << " s"; |
| | | |
| | | // read wav_scp |
| | | // read wav_path |
| | | vector<string> wav_list; |
| | | string wav_path_ = model_path.at(WAV_PATH); |
| | | if(is_target_file(wav_path_, "wav") || is_target_file(wav_path_, "pcm")){ |
| | |
| | | LOG(INFO) << "total_rtf " << (double)total_time/ (total_length*1000000); |
| | | LOG(INFO) << "speedup " << 1.0/((double)total_time/ (total_length*1000000)); |
| | | |
| | | FunASRUninit(asr_handle); |
| | | FunOfflineUninit(asr_handle); |
| | | return 0; |
| | | } |
| | |
| | | |
| | | vector<float> Paraformer::FbankKaldi(float sample_rate, const float* waves, int len) { |
| | | knf::OnlineFbank fbank_(fbank_opts); |
| | | fbank_.AcceptWaveform(sample_rate, waves, len); |
| | | std::vector<float> buf(len); |
| | | for (int32_t i = 0; i != len; ++i) { |
| | | buf[i] = waves[i] * 32768; |
| | | } |
| | | fbank_.AcceptWaveform(sample_rate, buf.data(), buf.size()); |
| | | //fbank_->InputFinished(); |
| | | int32_t frames = fbank_.NumFramesReady(); |
| | | int32_t feature_dim = fbank_opts.mel_opts.num_bins; |
| | |
| | | mini_sentence = cache_sent + mini_sentence |
| | | mini_sentence_id = np.concatenate((cache_sent_id, mini_sentence_id), axis=0,dtype='int32') |
| | | text_length = len(mini_sentence_id) |
| | | vad_mask = self.vad_mask(text_length, len(cache))[None, None, :, :].astype(np.float32) |
| | | data = { |
| | | "input": mini_sentence_id[None,:], |
| | | "text_lengths": np.array([text_length], dtype='int32'), |
| | | "vad_mask": self.vad_mask(text_length, len(cache))[None, None, :, :].astype(np.float32), |
| | | "sub_masks": np.tril(np.ones((text_length, text_length), dtype=np.float32))[None, None, :, :].astype(np.float32) |
| | | "vad_mask": vad_mask, |
| | | "sub_masks": vad_mask |
| | | } |
| | | try: |
| | | outputs = self.infer(data['input'], data['text_lengths'], data['vad_mask'], data["sub_masks"]) |
| | |
| | | ncpu=args.ncpu, |
| | | model_revision='v1.0.4') |
| | | |
| | | # vad |
| | | inference_pipeline_vad = pipeline( |
| | | task=Tasks.voice_activity_detection, |
| | | model=args.vad_model, |
| | | model_revision=None, |
| | | output_dir=None, |
| | | batch_size=1, |
| | | mode='online', |
| | | ngpu=args.ngpu, |
| | | ncpu=1, |
| | | ) |
| | | |
| | | print("model loaded") |
| | | |
| | | |
| | | |
| | | async def ws_serve(websocket, path): |
| | | frames = [] |
| | | frames_asr_online = [] |
| | | global websocket_users |
| | | websocket_users.add(websocket) |
| | | websocket.param_dict_asr_online = {"cache": dict()} |
| | | websocket.param_dict_vad = {'in_cache': dict()} |
| | | websocket.wav_name = "microphone" |
| | | print("new user connected",flush=True) |
| | | try: |
| | |
| | | if "is_speaking" in messagejson: |
| | | websocket.is_speaking = messagejson["is_speaking"] |
| | | websocket.param_dict_asr_online["is_final"] = not websocket.is_speaking |
| | | websocket.param_dict_vad["is_final"] = not websocket.is_speaking |
| | | # need to fire engine manually if no data received any more |
| | | if not websocket.is_speaking: |
| | | await async_asr_online(websocket,b"") |
| | | await async_asr_online(websocket, b"") |
| | | if "chunk_interval" in messagejson: |
| | | websocket.chunk_interval=messagejson["chunk_interval"] |
| | | if "wav_name" in messagejson: |
| | |
| | | websocket.param_dict_asr_online["chunk_size"] = messagejson["chunk_size"] |
| | | # if has bytes in buffer or message is bytes |
| | | if len(frames_asr_online) > 0 or not isinstance(message, str): |
| | | if not isinstance(message,str): |
| | | if not isinstance(message, str): |
| | | frames_asr_online.append(message) |
| | | # frames.append(message) |
| | | # duration_ms = len(message) // 32 |
| | | # websocket.vad_pre_idx += duration_ms |
| | | speech_start_i, speech_end_i = await async_vad(websocket, message) |
| | | websocket.is_speaking = not speech_end_i |
| | | |
| | | if len(frames_asr_online) % websocket.chunk_interval == 0 or not websocket.is_speaking: |
| | | websocket.param_dict_asr_online["is_final"] = not websocket.is_speaking |
| | | audio_in = b"".join(frames_asr_online) |
| | | # if not websocket.is_speaking: |
| | | #padding 0.5s at end gurantee that asr engine can fire out last word |
| | | # audio_in=audio_in+b''.join(np.zeros(int(16000*0.5),dtype=np.int16)) |
| | | await async_asr_online(websocket,audio_in) |
| | | await async_asr_online(websocket, audio_in) |
| | | frames_asr_online = [] |
| | | |
| | | |
| | |
| | | |
| | | |
| | | async def async_asr_online(websocket,audio_in): |
| | | if len(audio_in) >=0: |
| | | if len(audio_in) >= 0: |
| | | audio_in = load_bytes(audio_in) |
| | | rec_result = inference_pipeline_asr_online(audio_in=audio_in, |
| | | param_dict=websocket.param_dict_asr_online) |
| | |
| | | await websocket.send(message) |
| | | |
| | | |
| | | async def async_vad(websocket, audio_in): |
| | | segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad) |
| | | |
| | | speech_start = False |
| | | speech_end = False |
| | | |
| | | if len(segments_result) == 0 or len(segments_result["text"]) > 1: |
| | | return speech_start, speech_end |
| | | if segments_result["text"][0][0] != -1: |
| | | speech_start = segments_result["text"][0][0] |
| | | if segments_result["text"][0][1] != -1: |
| | | speech_end = True |
| | | return speech_start, speech_end |
| | | |
| | | if len(args.certfile)>0: |
| | | ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER) |
| | | |
| | | # Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions |
| | | ssl_cert = args.certfile |
| | | ssl_key = args.keyfile |
| | | |
| | | ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key) |
| | | start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None,ssl=ssl_context) |
| | | ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER) |
| | | |
| | | # Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions |
| | | ssl_cert = args.certfile |
| | | ssl_key = args.keyfile |
| | | |
| | | ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key) |
| | | start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None,ssl=ssl_context) |
| | | else: |
| | | start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None) |
| | | start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None) |
| | | asyncio.get_event_loop().run_until_complete(start_server) |
| | | asyncio.get_event_loop().run_forever() |