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Python Package Index (PyPI) : https://pypi.org/

 

PyPI · The Python Package Index

The Python Package Index (PyPI) is a repository of software for the Python programming language.

pypi.org

 

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Node.js 패키지 생성 및 실행 - Node.js package, npm init, npm run 

 

 

Node.js의 패키지 만들기
 - 폴더 생성
 - 콘솔에서 생성된 폴더로 이동
 - 패키지를 생성하기 위해 npm init 명령어를 실행
 - 폴더에 package.json 파일이 생성됩니다.
D:\_nodejs\nodePjt>
D:\_nodejs\nodePjt>npm init

{
  "name": "test1",
  "version": "1.0.0",
  "description": "test",
  "main": "index.js",
  "scripts": {
    "test": "echo \"Error: no test specified\" && exit 1"
  },
  "author": "carrotweb",
  "license": "ISC"
}



* 패키지를 설치하는 명령어입니다.
  Express.js 설치하기
  Express.js는 Node.js에서 HTTP와 관련된 컴포넌트를 기반으로 하는 웹 애플리케이션 프레임워크입니다.
  현재 패키지(애플리케이션)에 Express.js를 설치하기 위해 콘솔에서 npm install 명령어를 실행합니다.
  npm install에 옵션으로 --save를 추가하면 자동으로 package.json 파일의 "dependencies"에 "express" 항목이 추가됩니다.

D:\_nodejs\nodePjt>
D:\_nodejs\nodePjt>npm install express --save

{
  "name": "test1",
  "version": "1.0.0",
  "description": "test",
  "main": "index.js",
  "scripts": {
    "test": "echo \"Error: no test specified\" && exit 1"
  },
  "author": "carrotweb",
  "license": "ISC",
  "dependencies": {
    "express": "^4.17.1" --> 추가
  }
}





 * Node.js의 패키지(애플리케이션) 실행하기
 * Node.js를 종료는 콘솔에서 Ctrl + C를 누르면 됩니다.

D:\_nodejs\nodePjt>
D:\_nodejs\nodePjt>node index.js
Listening...
^C
D:\_nodejs\nodePjt>^C
D:\_nodejs\nodePjt>



=========================================================


접속 : http://localhost:8080/index.html


* npm으로 실행하기 위해 Script 추가하기
* 콘솔에서 npm start를 실행합니다.
  종료하려면 콘솔에서 Ctrl + C를 누르고 "Y"를 입력하고 엔터키를 누르면 됩니다.

D:\_nodejs\nodePjt>npm run start
npm WARN config global `--global`, `--local` are deprecated. Use `--location=global` instead.

> nodepjt@1.0.0 start
> node index.js

Listening...



* Express 정적 파일 적용하기
 - index.js를 오픈하여 이미지 파일이나 CSS 파일, JavaScript 파일 등과 같은 정적 파일을 제공하기 위해 
   Express.js의 express.static() 메서드를 추가합니다. 정적 파일들이 들어있는 폴더로 public 폴더를 설정하였습니다.
 - 폴더에 public 폴더를 생성합니다.
 - public 폴더에 index.html 파일을 생성합니다.
 - npm start를 실행합니다.
  

   D:\_nodejs\nodePjt>npm run start



 - 브라우저에서 "http://localhost:8080/index.html"를 입력


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CentOS yum 패키지 삭제

 

Yum을 삭제하는 것이 아니라, yum으로 다른 패키지를 삭제하는 것

 

 

## 설치 확인 

[root@ngio ~]# yum list installed java*jdk
... (생략)
Installed Packages
java-1.7.0-openjdk.x86_64            1:1.7.0.9-2.3.3.el5.1             installed


## 패키지 삭제

[root@ngio ~]# yum remove java-1.7.0-openjdk.x86_64
... (생략)
================================================================================
 Package                    Arch     Version                  Repository   Size
================================================================================
Removing:
 java-1.7.0-openjdk         x86_64   1:1.7.0.9-2.3.3.el5.1    installed    52 M
Removing for dependencies:
 java-1.7.0-openjdk-devel   x86_64   1:1.7.0.9-2.3.3.el5.1    installed    26 M

Transaction Summary
================================================================================
Remove        2 Package(s)
Reinstall     0 Package(s)
Downgrade     0 Package(s)

Is this ok [y/N]: y


## 재확인

[root@ngio ~]# yum list installed java*jdk
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
Error: No matching Packages to list
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StreamingRecognitionResult

A streaming speech recognition result corresponding to a portion of the audio that is currently being processed.

Fields

 

#alternatives

#channel_tag

 

alternatives[]

SpeechRecognitionAlternative

May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.

is_final

bool

If false, this StreamingRecognitionResult represents an interim result that may change. If true, this is the final time the speech service will return this particular StreamingRecognitionResult, the recognizer will not return any further hypotheses for this portion of the transcript and corresponding audio.

stability

float

An estimate of the likelihood that the recognizer will not change its guess about this interim result. Values range from 0.0 (completely unstable) to 1.0 (completely stable). This field is only provided for interim results (is_final=false). The default of 0.0 is a sentinel value indicating stability was not set.

result_end_time

Duration

Time offset of the end of this result relative to the beginning of the audio.

channel_tag

int32

For multi-channel audio, this is the channel number corresponding to the recognized result for the audio from that channel. For audio_channel_count = N, its output values can range from '1' to 'N'.

https://cloud.google.com/speech-to-text/docs/reference/rpc/google.cloud.speech.v1#streamingrecognitionresult

 

Package google.cloud.speech.v1  |  Cloud Speech-to-Text 문서

phrases[] string A list of strings containing words and phrases "hints" so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typicall

cloud.google.com

 

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Package google.cloud.speech.v1

RecognitionConfig

Provides information to the recognizer that specifies how to process the request.

Fields

encoding

AudioEncoding

Encoding of audio data sent in all RecognitionAudio messages. This field is optional for FLAC and WAV audio files and required for all other audio formats. For details, see AudioEncoding.

sample_rate_hertz

int32

Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that's not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, see AudioEncoding.

audio_channel_count

int32

The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are 1-8. Valid values for OGG_OPUS are '1'-'254'. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to 'true'.

enable_separate_recognition_per_channel

bool

This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately. The recognition result will contain a channel_tag field to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized: audio_channel_count multiplied by the length of the audio.

language_code

string

Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.

max_alternatives

int32

Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than max_alternatives. Valid values are 0-30. A value of 0 or 1 will return a maximum of one. If omitted, will return a maximum of one.

profanity_filter

bool

If set to true, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. "f***". If set to false or omitted, profanities won't be filtered out.

speech_contexts[]

SpeechContext

Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.

enable_word_time_offsets

bool

If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If false, no word-level time offset information is returned. The default is false.

enable_automatic_punctuation

bool

If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.

diarization_config

SpeakerDiarizationConfig

Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.

metadata

RecognitionMetadata

Metadata regarding this request.

model

string

Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig.

 

use_enhanced

bool

Set to true to use an enhanced model for speech recognition. If use_enhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio.

If use_enhanced is true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.

 

 

 

 

https://cloud.google.com/speech-to-text/docs/reference/rpc/google.cloud.speech.v1#recognitionconfig

 

Package google.cloud.speech.v1  |  Cloud Speech-to-Text 문서

phrases[] string A list of strings containing words and phrases "hints" so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typicall

cloud.google.com

 

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