Text-to-Speech | A sweet and simple Text to Speech android app | Android library

I have checked that Android offers one option to "Amplify Speech Volume" under Text-to-Speech settings and I was wondering how to offer the same capability using their AudioManager. As far as I am concerned, AudioManager.setStreamVolume() could be a good candidate for this purpose [LINK], but I did not find any StreamType for Text-To-Speech.

I am assuming that there is a way to access that option programmatically in Java, but I might be wrong.

I am using the Google TextToSpeech API in Node.js to generate speech from text. I was able to get an output file with the same name as the text that is generated for the speech. However, I need to tweak this a bit. I wish I could generate multiple files at the same time. The point is that I have, for example, 5 words (or sentences) to generate, e.g. cat, dog, house, sky, sun. I would like to generate them each to a separate file: cat.wav, dog.wav, etc.

I also want the application to be able to read these words from the * .txt file (each word/sentence on a separate line of the * .txt file).

Is there such a possibility? Below I am pasting the * .js file code and the * .json file code that I am using.

*.js

I'm writing a script in PHP to automate the process of grabbing data from a Database and inserting it into a google sheet. Since it's going to be running at 15 minute intervals and Oauth consent screen isn't really the best option. Is there another option to get the necessary authorization to write to the sheet without the need of signing in via browser?

Most of the examples I've seen in the documentation and online use a consent screen. If I try this:

I want to create a web application(Flask- A Flashcard AI), a part of which is a bot which needs to directly interact with the human through speech recognition and text-to-speech. I have pyttsx3 and speech_recognition installed for that, where I am confused is how am I supposed to get the user's audio as input and then send it to the backend. I have tried to look up YouTube tutorials and asked other people about the same, the only success I've had is learning about Navigator.MediaDevices.getUserMedia. I want to make the communication fluent, and I will have to send the data to the back-end as well. I am not sure how to send it to the back-end and get the user media fluently, I could use Navigator.MediaDevices.getUserMedia and convert it into an audio file(not sure how to do that yet but I think I'll figure it out eventually, and having the user upload a audio recording won't be nice at all), but then that'll take up a lot of space on the database.

I'm using Google Cloud Text-to-speech to synthesize speech from text. How can I specify the region for the API calls? This is similar to this question Specify Region for Google Speech API? but my question is for text-to-speech, not speech-to-text.

For speech-to-text there wasn't an available endpoint in Europe but there is one now: https://cloud.google.com/speech-to-text/docs/endpoints

I can't find the same type of endpoint documentation for text-to-speech, the closest I find is this page: https://cloud.google.com/text-to-speech/docs/reference/rest that specifies a single endpoint: https://texttospeech.googleapis.com

Does this mean that I cannot keep the text-to-speech requests within Europe? It could also be that the region is fetched from the Google Cloud project region or something like that but I cannot find such an option.

I am making use of the Google TTS API and would like to use timepoints in order to show words of a sentence at the right time. (like subtitles). Unfortunately, I can not get this to work.

HTTP request

I've made a simple app for Android that provides Accessibility service.

When I install the app on Android 10 (Samsung) and activate the Accessibility service, OS says:

APP does not support the current system language. The text-to-speech language will be set to English (United States), but this might cause unexpected behavior.

It happens when system language is en_US/es_US and text-to-speech language is Spanish.

I've added values-es/strings.xml with translations but it does not help.

What can I do to provide support for different text-to-speech languages?

I am trying to build a websocket server using Autobahn python which acts as a man-in-the-middle or relay for IBM Watson's text-to-speech service. I have already managed to receive and forward the streaming audio from the client to Watson by use of a queue, and I am receiving back transcription hypotheses as JSON data from Watson to my server, but I am not sure how to then forward that JSON data on to the client. It seems that the Watson transcription-side callback and the Autobahn client-side callback exist independently and I can't call a routine from one callback within the other or access data from one callback within the other.

Do I need to set up some kind of shared text message queue? I am sure it should be something simple but I think the problem may be my lack of understanding of the "self" keyword which seems to be isolating the two routines. Would also appreciate any resources on understanding "self".

My server receives udp packets of audio buffers. I would like to stream these incoming chunks to Google's text-to-speech API.

Using fs streams, is it possible to achieve this in a manner that goes something like this?: