DynamicAudioNormalizer | Dynamic Audio Normalizer | Speech library

by lordmulder C++ Version: 2.10 License: Non-SPDX

X-Ray Key Features Code Snippets Community Discussions(10)Vulnerabilities Install Support

kandi X-RAY | DynamicAudioNormalizer Summary

DynamicAudioNormalizer is a C++ library typically used in Artificial Intelligence, Speech applications. DynamicAudioNormalizer has no bugs, it has no vulnerabilities and it has low support. However DynamicAudioNormalizer has a Non-SPDX License. You can download it from GitHub.

Dynamic Audio Normalizer is a library for advanced audio normalization purposes. It applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in contrast to more "simple" normalization algorithms, the Dynamic Audio Normalizer dynamically re-adjusts the gain factor to the input audio. This allows for applying extra gain to the "quiet" sections of the audio while avoiding distortions or clipping the "loud" sections. In other words: The Dynamic Audio Normalizer will "even out" the volume of quiet and loud sections, in the sense that the volume of each section is brought to the same target level. Note, however, that the Dynamic Audio Normalizer achieves this goal without applying "dynamic range compressing". It will retain 100% of the dynamic range within each "local" region of the audio file. The Dynamic Audio Normalizer is available as a small standalone command-line utility and also as an effect in the SoX audio processor as well as in the FFmpeg audio/video converter. Furthermore, it can be integrated into your favourite DAW (digital audio workstation), as a VST plug-in, or into your favourite media player, as a Winamp plug-in. Last but not least, the "core" library can be integrated into custom applications easily, thanks to a straightforward API (application programming interface). The "native" API is written in C++, but language bindings for C99, Microsoft.NET, Java, Python and Pascal are provided.

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Quality

Security

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Support

DynamicAudioNormalizer has a low active ecosystem.

It has 209 star(s) with 34 fork(s). There are 19 watchers for this library.

It had no major release in the last 12 months.

There are 3 open issues and 21 have been closed. On average issues are closed in 130 days. There are 2 open pull requests and 0 closed requests.

It has a neutral sentiment in the developer community.

The latest version of DynamicAudioNormalizer is 2.10

Quality

DynamicAudioNormalizer has 0 bugs and 0 code smells.

Security

DynamicAudioNormalizer has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

DynamicAudioNormalizer code analysis shows 0 unresolved vulnerabilities.

There are 0 security hotspots that need review.

License

DynamicAudioNormalizer has a Non-SPDX License.

Non-SPDX licenses can be open source with a non SPDX compliant license, or non open source licenses, and you need to review them closely before use.

Reuse

DynamicAudioNormalizer releases are available to install and integrate.

Installation instructions, examples and code snippets are available.

It has 2833 lines of code, 98 functions and 15 files.

It has high code complexity. Code complexity directly impacts maintainability of the code.

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DynamicAudioNormalizer Key Features

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DynamicAudioNormalizer Examples and Code Snippets

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Community Discussions

Trending Discussions on Speech

Enable use of images from the local library on Kubernetes

IndexError: tuple index out of range when I try to create an executable from a python script using auto-py-to-exe

Google Actions Builder stops execution when selecting a visual item from a List

How to use muti-language in 'gTTS' for single input line?

Assigning True/False if a token is present in a data-frame

speechSynthesis.getVoices (Web Speech API) doesn't show some of the locally installed voices

Combining Object Detection with Text to Speech Code

Yielding values from consecutive parallel parse functions via meta in Scrapy

Rails. Puma stops working when instantiating a client of Google Cloud Text-to-Speech (Windows)

R - Regular Expression to Extract Text Between Parentheses That Contain Keyword

QUESTION

Enable use of images from the local library on Kubernetes

Asked 2022-Mar-20 at 13:23

I'm following a tutorial https://docs.openfaas.com/tutorials/first-python-function/,

currently, I have the right image

...

ANSWER

Answered 2022-Mar-16 at 08:10

If your image has a latest tag, the Pod's ImagePullPolicy will be automatically set to Always. Each time the pod is created, Kubernetes tries to pull the newest image.

Try not tagging the image as latest or manually setting the Pod's ImagePullPolicy to Never. If you're using static manifest to create a Pod, the setting will be like the following:

Source https://stackoverflow.com/questions/71493306

QUESTION

IndexError: tuple index out of range when I try to create an executable from a python script using auto-py-to-exe

Asked 2022-Feb-24 at 15:03

I have been trying out an open-sourced personal AI assistant script. The script works fine but I want to create an executable so that I can gift the executable to one of my friends. However, when I try to create the executable using the auto-py-to-exe, it states the below error:

...

ANSWER

Answered 2021-Nov-05 at 02:20

42681 INFO: PyInstaller: 4.6
42690 INFO: Python: 3.10.0

Source https://stackoverflow.com/questions/69776492

QUESTION

Google Actions Builder stops execution when selecting a visual item from a List

Asked 2022-Feb-23 at 15:32

I'm pulling my hairs here. I have a Google Assistant application that I build with Jovo 4 and Google Actions Builder.

The goal is to create a HelpScene, which shows some options that explain the possibilities/features of the app on selection. This is the response I return from my Webhook. (This is Jovo code, but doesn't matter as this returns a JSON when the Assistant calls the webhook.)

...

ANSWER

Answered 2022-Feb-23 at 15:32

Okay, after days of searching, I finally figured it out. It did have something to do with the Jovo framework/setup and/or the scene parameter in the native response.

This is my component, in which I redirect new users to the HelpScene. This scene should show multiple cards in a list/collection/whatever on which the user can tap to receive more information about the application's features.

Source https://stackoverflow.com/questions/70968978

QUESTION

How to use muti-language in 'gTTS' for single input line?

Asked 2022-Jan-29 at 07:05

I want to convert text to speech from a document where multiple languages are included. When I am trying to do the following code, I fetch problems to record each language clearly. How can I save such type mixer text-audio clearly?

...

ANSWER

Answered 2022-Jan-29 at 07:05

It's not enough to use just text to speech, since it can work with one language only.
To solve this problem we need to detect language for each part of the sentence.
Then run it through text to speech and append it to our final spoken sentence.
It would be ideal to use some neural network (there are plenty) to do this categorization for You.
Just for a sake of proof of concept I used googletrans to detect language for each part of the sentences and gtts to make a mp3 file from it.

It's not bullet proof, especially with arabic text. googletrans somehow detect different language code, which is not recognized by gtts. For that reason we have to use code_table to pick proper language code that works with gtts.

Here is working example:

Source https://stackoverflow.com/questions/70852444

QUESTION

Assigning True/False if a token is present in a data-frame

Asked 2022-Jan-06 at 12:38

My current data-frame is:

...

ANSWER

Answered 2022-Jan-06 at 12:13

try

Source https://stackoverflow.com/questions/70606847

QUESTION

speechSynthesis.getVoices (Web Speech API) doesn't show some of the locally installed voices

Asked 2021-Dec-31 at 08:19

I'm trying to use Web Speech API to read text on my web page. But I found that some of the SAPI5 voices installed in my Windows 10 would not show up in the output of speechSynthesis.getVoices(), including the Microsoft Eva Mobile on Windows 10 "unlock"ed by importing a registry file. These voices could work fine in local TTS programs like Balabolka but they just don't show in the browser. Are there any specific rules by which the browser chooses whether to list the voices or not?

...

ANSWER

Answered 2021-Dec-31 at 08:19

OK, I found out what was wrong. I was using Microsoft Edge and it seems that Edge only shows some of Microsoft voices. If I use Firefox, the other installed voices will also show up. So it was Edge's fault.

Source https://stackoverflow.com/questions/70490870

QUESTION

Combining Object Detection with Text to Speech Code

Asked 2021-Dec-28 at 16:46

I am trying to write an object detection + text-to-speech code to detect objects and produce a voice output on the raspberry pi 4. However, as of right now, I am trying to write a simple python script that incorporates both elements into a single .py file and preferably as a function. I will then run this script on the raspberry pi. I want to give credit to Murtaza's Workshop "Object Detection OpenCV Python | Easy and Fast (2020)" and https://pypi.org/project/pyttsx3/ for the Text to speech documentation for pyttsx3. I have attached the code below. I have tried running the program and I always keep getting errors with the Text to speech code (commented lines 33-36 for reference). I believe it is some looping error but I just can't seem to get the program to run continuously. For instance, if I run the code without the TTS part, it works fine. Otherwise, it runs for perhaps 3-5 seconds and suddenly stops. I am a beginner but highly passionate in computer vision, and any help is appreciated!

...

ANSWER

Answered 2021-Dec-28 at 16:46

I installed pyttsx3 using the two commands in the terminal on the Raspberry Pi:

sudo apt update && sudo apt install espeak ffmpeg libespeak1
pip install pyttsx3

I followed the video youtube.com/watch?v=AWhDDl-7Iis&ab_channel=AiPhile to install pyttsx3. My functional code should also be listed above. My question should be resolved but hopefully useful to anyone looking to write a similar program. I have made minor tweaks to my code.

Source https://stackoverflow.com/questions/70129247

QUESTION

Yielding values from consecutive parallel parse functions via meta in Scrapy

Asked 2021-Dec-20 at 07:53

In my scrapy code I'm trying to yield the following figures from parliament's website where all the members of parliament (MPs) are listed. Opening the links for each MP, I'm making parallel requests to get the figures I'm trying to count. I'm intending to yield each three figures below in the company of the name and the party of the MP

Here are the figures I'm trying to scrape

How many bill proposals that each MP has their signature on
How many question proposals that each MP has their signature on
How many times that each MP spoke on the parliament

In order to count and yield out how many bills has each member of parliament has their signature on, I'm trying to write a scraper on the members of parliament which works with 3 layers:

Starting with the link where all MPs are listed
From (1) accessing the individual page of each MP where the three information defined above is displayed
3a) Requesting the page with bill proposals and counting the number of them by len function 3b) Requesting the page with question proposals and counting the number of them by len function 3c) Requesting the page with speeches and counting the number of them by len function

What I want: I want to yield the inquiries of 3a,3b,3c with the name and the party of the MP in the same raw

Problem 1) When I get an output to csv it only creates fields of speech count, name, part. It doesn't show me the fields of bill proposals and question proposals
Problem 2) There are two empty values for each MP, which I guess corresponds to the values I described above at Problem1
Problem 3) What is the better way of restructuring my code to output the three values in the same line, rather than printing each MP three times for each value that I'm scraping

...

ANSWER

Answered 2021-Dec-18 at 06:26

This is happening because you are yielding dicts instead of item objects, so spider engine will not have a guide of fields you want to have as default.

In order to make the csv output fields bill_prop_count and res_prop_count, you should make the following changes in your code:

1 - Create a base item object with all desirable fields - you can create this in the items.py file of your scrapy project:

Source https://stackoverflow.com/questions/70399191

QUESTION

Rails. Puma stops working when instantiating a client of Google Cloud Text-to-Speech (Windows)

Asked 2021-Dec-15 at 22:07

I've upgraded my Ruby version from 2.5.x to 2.6.x (and uninstalled the 2.5.x version). And now Puma server stops working when instantiating a client of Google Cloud Text-to-Speech:

...

ANSWER

Answered 2021-Dec-07 at 08:52

Try reinstalling ruby-debug

Source https://stackoverflow.com/questions/70173479

QUESTION

R - Regular Expression to Extract Text Between Parentheses That Contain Keyword

Asked 2021-Nov-13 at 22:41

I need to extract the text from between parentheses if a keyword is inside the parentheses.

So if I have a string that looks like this:

('one', 'CARDINAL'), ('Castro', 'PERSON'), ('Latin America', 'LOC'), ('Somoza', 'PERSON')

And my keyword is "LOC", I just want to extract ('Latin America', 'LOC'), not the others.

Help is appreciated!!

This is a sample of my data set, a csv file:

...

ANSWER

Answered 2021-Nov-13 at 22:41

You can use

Source https://stackoverflow.com/questions/69958102

Community Discussions, Code Snippets contain sources that include Stack Exchange Network

Vulnerabilities

No vulnerabilities reported

Install DynamicAudioNormalizer

Dynamic Audio Normalizer can be downloaded from one of the following official mirror sites:.
https://github.com/lordmulder/DynamicAudioNormalizer/releases/latest
https://bitbucket.org/muldersoft/dynamic-audio-normalizer/downloads
https://sourceforge.net/projects/muldersoft/files/Dynamic%20Audio%20Normalizer/
https://www.mediafire.com/folder/flrb14nitnh8i/Dynamic_Audio_Normalizer
https://www.assembla.com/spaces/dynamicaudionormalizer/documents
The exact steps that are required to load, activate and configure a VST plug-in differ from application to application. However, it will generally be required to make the application "recognize" the new VST plug-in, i.e. DynamicAudioNormalizerVST.dll first. Most applications will either pick up the VST plug-in from the global VST directory – usually located at C:\Program Files (x86)\Steinberg\Vstplugins on Windows – or provide an option to choose the directory from where to load the VST plug-in. This means that, depending on the individual application, you will either need to copy the Dynamic Audio Normalizer VST plug-in into the global VST directory or tell the application where the Dynamic Audio Normalizer VST plug-in is located. Note that, with some applications, it may also be required to explicitly scan for new VST pluig-ins. See the manual for details! The screen capture bellow shows the situation in the Acoustica 6.0 software by Acon AS. Here we simply open the "VST Directories" dialogue from the "Plug-Ins" menu, then add the Dynamic Audio Normalizer directory and finally click "OK". Furthermore, note that – unless you are using the static build of the Dynamic Audio Normalizer – the VST plug-in DLL, i.e. DynamicAudioNormalizerVST.dll, also requires the Dynamic Audio Normalizer core library, i.e. DynamicAudioNormalizerAPI.dll. This means that the core library must be made available to the VST host in addition to the VST plug-in itself. Otherwise, loading the VST plug-in DLL is going to fail! Copying the core library to the same directory, where the VST plug-in DLL, is located generally is not sufficient. Instead, the core library must be located in one of those directories that are checked for additional DLL dependencies (see here for details). Therefore, it is recommended to copy the DynamicAudioNormalizerAPI.dll file into the same directory where the VST host's "main" executable (EXE file) is located.
If you have not installed the latest version of Winamp yet, please update your Winamp installation first. Although the development of Winamp seems to be discontinued and its future remains uncertain at this point, you can still download the last Winamp release (as of January 2017), which is Winamp 5.666 "Redux", from one of the following download mirrors:.
http://meggamusic.co.uk/winamp/Winamp_Download.htm
http://www.filehorse.com/download-winamp/
http://codecpack.co/download/Nullsoft_Winamp.html
http://www.free-codecs.com/winamp_download.htm
The following listing summarizes the steps that an application needs to follow when using the API:.
Create a new MDynamicAudioNormalizer instance. This allocates required resources.
Call the initialize() method, once, in order to initialize the MDynamicAudioNormalizer instance.
Call the processInplace() method, in a loop, until all input samples have been processed. Note: At the beginning, this function returns less output samples than input samples have been passed. Samples are guaranteed to be returned in FIFO order, but there is a certain "delay"; call getInternalDelay() for details.
Call the flushBuffer() method, in a loop, until all the pending "delayed" output samples have been flushed.
Destroy the MDynamicAudioNormalizer instance. This will free up all allocated resources.
The following build environments are currently supported:.
Microsoft Windows with Visual C++, tested with Visual Studio 2013 and Visual Studio 2015: You can build Dynamic Audio Normalizer by using the provided solution file: DynamicAudioNormalizer.sln Optionally, you may run the deployment script z_build.bat, which will build the application in various configurations also create deployment packages. Note that you may need to edit the paths in the build script first! Be sure that your environment variables JAVA_HOME (JDK path) and QTDIR (Qt4 path) are set correctly!
Linux with GCC/G++ and GNU Make, tested under Ubuntu 16.04 LTS and openSUSE Leap 42.2: You can build Dynamic Audio Normalizer by invoking the provided Makefile via make command. We assume that the essential build tools (make, g++, libc, etc), as contained in Debian's build-essential package, are installed. Optionally, you may pass the MODE=no-gui or MODE=mininmal options to Make in order to build the software without the GUI program or to create a minimal build (core library + CLI front-end only), respectively Be sure that your environment variables JAVA_HOME (JDK path) and QTDIR (Qt4 path) are set correctly!
Building the Dynamic Audio Normalizer requires some third-party tools and libraries:.
POSIX Threads (PThreads) is always required (on Windows use pthreads-w32, by Ross P. Johnson)
libsndfile, by Erik de Castro Lopo, is required for building the command-line program
libmpg123, by Michael Hipp et al., is required for building the command-line program
Qt Framework, by Qt Project, is required for building the log viewer GUI program (recommended version: Qt 4.x)
Java Develpment Kit (JDK), by Oracle, is required for building the JNI bindings (recommended version: JDK 8)
Ant, by Apache Software Foundation, is required for building the JNI bindings (recommended version: 1.9.x)
VST SDK v2.4, by Steinberg GmbH, is required for building the VST plug-in (it's still included in the VST 3.x SDK!)
Winamp SDK, by Nullsoft Inc, is required for building the Winamp plug-in (recommended version: 5.55)
Pandoc, by John MacFarlane, is required for generating the HTML documentation
UPX, by Markus Franz Xaver Johannes Oberhumer et al., is required for "packing" the libraries/executables

Support

This chapter describes the MDynamicAudioNormalizer application programming interface (API). The API specified in this document allows software developer to call the Dynamic Audio Normalizer library from their own programs.

Find more information at: