I used Powerpoint for the conversion, from an MP3 in to an MP4 file. Don’t use MP3 in production. Try exporting the cassette capture as a perfect quality WAV (Microsoft) 16-bit file and then use that in the video generation. Google and see if you can talk PowerPoint into a higher quality. MP3 is a popular lossy audio format. This allows you to maintain acceptable sound quality with small file size. For most users, this loss of sound quality is almost imperceptible, so the MP3 format quickly gained its popularity, allowing you to store more songs on compact digital devices such as smartphones and iPods. Assume you record at high quality 24 bit .wav. If you are sending your tracks to be mixed, you would want to send .wav so the quality is the best. If you are sending a rough mix or a quick mix for someone to add their voice you can send an mp3, shitty quality, but easier to email. In general, the compression rate used in MP3 is around 75% to 95%. Quality. WAV files provide the highest possible sound quality, making it the most suitable format for applications with critical audio quality, like music or sound production. MP3, on the other hand, loses some of the original audio data and provides moderate-quality sound. A 44.1kHz 256kb mp3 or m4a file should be converted to 16/44.1 WAV. You could convert to any higher bitrate/sample rate (e.g. 24/48), but there would be no benefit in terms of fidelity, and the file would be larger. You should not convert to any sample or bitrate lower than 16/44.1. All you will get is the same quality recording, just in a larger file. If I were you I'd stick to MP3. Don't convert from MP3 to another format or it will just sound the same as before maybe worse. If you want to change the format then you will need to re-rip or re-download from the original source using the new conversion settings you desire This is the basis of all the differences in a way among the three formats. The MP3 contains bits of digital noise and compressed sound files. The WMA file contains compressed sound files like MP3. While the WAV file contains data encoded with LPCM , ADPCM. This can even contain MP3 encoded data. Let’s modify the command to change the audio bitrate: $ ffmpeg -i sample.mp4 -c:v libx264 -c:a pcm_s16le -b:v 1200k -b:a 192k output.avi. The -b:a option specifies the audio bitrate. The command above outputs the output.avi file, which uses H.264 video codec with 1200 kbps and PCM audio with 192 kbps. 5. MP files compress audio data by eliminating sounds that are outside the hearing range of most people, thus reducing the file size without significantly affecting the perceived sound quality. Technical details: 🔵 The `.mp3` format, MPEG-1 Audio Layer III, is a popular digital audio format. To do so: Open Audacity. Click Edit (Windows) or Audacity (Mac) Click Preferences in the drop-down menu. Click the Quality tab. Click the "Default Sample Rate" drop-down box, then click 48000 Hz. Click the "Sample Rate Converter" drop-down box, then click Best Quality (Slowest) Click OK (Windows only). Part 2. nev1H.