And start with uncompressed formats. You aren't anywhere near ready to deal with figuring out how much processing power is needed for dealing with compressed formats.

It is very important to know two parameters for audio file, sampling rate and word size.

An audio file format is a file format for storing digital audio data on a computer system.

The most common uncompressed audio formats are WAV, AIFF, AU, PCM or BWF.

In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence.

Sample rates are usually measured per second, The number of samples the processer takes per second is known as the sample rate. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth.

Digital audio resolution is measured in sample depth. Most digital audio formats use a sample depth of either 16-bit, 24-bit, and 32-bit. Common sample rates are 44.1 kHz, 48 kHz, and 96 kHz. kHz stands for kilohertz, or 1000 samples per second, so 44.1 kHz represents 44100 samples per second. This many samples are needed to capture the full complexity of sound waves.

AIFF files = sample rate of 44.1 kHz and a bit depth of 16 bits.

 

When we first started with computers to do audio we used "DSP" computers that are designed to process audio & analog signals. DSP is not a good general purpose micro but they process audio very well.

 

It is very important to know two parameters for audio file, sampling rate and word size.

An audio file format is a file format for storing digital audio data on a computer system.

The most common uncompressed audio formats are WAV, AIFF, AU, PCM or BWF.

In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence.

Sample rates are usually measured per second, The number of samples the processer takes per second is known as the sample rate. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth.

Digital audio resolution is measured in sample depth. Most digital audio formats use a sample depth of either 16-bit, 24-bit, and 32-bit. Common sample rates are 44.1 kHz, 48 kHz, and 96 kHz. kHz stands for kilohertz, or 1000 samples per second, so 44.1 kHz represents 44100 samples per second. This many samples are needed to capture the full complexity of sound waves.

AIFF files = sample rate of 44.1 kHz and a bit depth of 16 bits.

Is this YOUR understanding of things? This reads like something either lifted wholesale from some website or something generated by ChatGPT or its ilk. We can't have a discussion to help you understand a topic better unless what you post accurately reflects your level of understanding of the topic.

 

Is this YOUR understanding of things? This reads like something either lifted wholesale from some website or something generated by ChatGPT or its ilk. We can't have a discussion to help you understand a topic better unless what you post accurately reflects your level of understanding of the topic.

English is not my first language. so you may find my words strange. I was reading all posts yesterday but I did not reply quickly, I spent my time in reading and understanding. Obviously I have visited many websites, I have read the information given there.

I found that for audio file it is important to know two things sampling rate and word size.

I don't want to give up because I have spent a lot of time researching and am still researching. But if it's too far from my current knowledge so maybe I'll leave it

 

You are asking two separate questions.

If the sampling rate is 44.1kHz then the MCU has to output one data point every 22.6μs.
Many MCUs are capable of outputting one data point every 20μs but that might not leave any time for the MCU to do anything else such as acquiring the data in the first place, for example, reading it from a file or from a web server.

That is not how it is done. Very often the data is acquired and put into a buffer using DMA. Then the DAC accesses the data from the buffer using DMA. No MCU cycles are needed to do this, i.e. it takes 0% of the MCU time.

In terms of capability of the MCU, it has to have DMA capability.

 

English is not my first language. so you may find my words strange. I was reading all posts yesterday but I did not reply quickly, I spent my time in reading and understanding. Obviously I have visited many websites, I have read the information given there.

I found that for audio file it is important to know two things sampling rate and word size.

I don't want to give up because I have spent a lot of time researching and am still researching. But if it's too far from my current knowledge so maybe I'll leave it

Hi K,
I'm English, so understand the language ok, but have similar difficulties to you, for different reasons.
I've been working on a project for years, and had lots of help, from forums like this.
There will be some frustrations, but if you don't want to give up, then don't.

Apart from the my headaches also headaches I give to forum members, I have enjoyed moving slowly forward, and I hope you will too.
Good luck C.

 

I've been working on a project for years, and had lots of help, from forums like this.
There will be some frustrations, but if you don't want to give up, then don't.

There are many members on this forum who helped me. I thank all the members who give their valuable time to help others. @WBahn is a member who helped me a lot in understanding c programming. I salute his patience.

Sometimes it is very frustrating when even after trying hard, we don't understand things.

i like to be on this forum