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2. How can I extract audio from video with ffmpeg? - Stack Overflow

   stackoverflow.com/questions/9913032

   For people looking for the simpler way to extract audio from a video file while retaining the original video file's parameters, you can use: ffmpeg -i <video_file_name.extension> <audio_file_name.extension>. For example, running: ffmpeg -i screencap.mov screencap.mp3. extracts an mp3 audio file from a mov video file.

3. ffmpeg - Audacity FFMPG exporting partial file - Stack Overflow

   stackoverflow.com/questions/41397034

   I exported each track from audacity as a seperate file, and then used ffmpg to join them using the following options. F:\unsorted\Jurassic World 2015 (1080p Bluray x265 HEVC 10bit AAC 7.1 Tigole)\Mult>"C:\Program Files (x86)\FFmpeg for Audacity\ffmpeg".

4. Extracting Opus from WebM losslessly with FFmpeg or MKVToolNix

   superuser.com/questions/1474050

   Most lossy formats are going to have more or less rounding errors when decoding. Is there any difference between ffmpeg -i source.webm -c:a copy -vn output.opus && ffmpeg -i output.opus decoded_extracted.wav and ffmpeg -i source.webm decoded.wav?

5. How to specify the noise floor in FFmpeg's acompressor filter

   stackoverflow.com/questions/57190018/how-to-specify-the-noise-floor-in-ffmpegs...

   I've always used Audacity's Dynamic Range Compressor if I wanted to crank up the volume of an audio file. Today I was looking at whether I could do the same with FFmpeg's acompressor filter. I've found "how to user "compressor" with ffmpeg" and Gyan's answer, which was really helpful, but not satisfying enough.

6. Splitting an audio file into chunks of a specified length

   superuser.com/questions/525210

   To use this program, save it as audio_cutter.py and run it using Python. A window with a file input and a cut interval input will appear. Click "Browse" to upload an audio file, input the desired cut interval in seconds, and click "Cut Audio." The program will create a folder called "outputs" and save the audio segments in it.

7. ffmpeg how to reduce bass and increase treble like audacity

   stackoverflow.com/questions/39607741

   13. You could try the firequalizer filter. Each entry takes two parameters frequency and gain (dB). The gain for 0 Hz should be the bass value you used. The gain for 16000 Hz should be the treble value. 1000 Hz remains as is - no gain. 250 Hz gain should be half the bass value. 4000 Hz value should be half the treble value.

8. ffmpeg / Audacity channel splitting differences - Stack Overflow

   stackoverflow.com/questions/49196155

   I did this splitting through Audacity. The accuracy was about 80-90%, which was perfect for my purposes. However, once I tried to automate the splitting using ffmpeg (more specifically: ffmpeg -i input_filename.mp3 -map_channel 0.0.0 left.wav -map_channel 0.0.1 right.wav), the accuracy dropped drastically. I've been experimenting for about a ...

9. Audacity FFmpeg 7.1 AAC export Unsupported number of channels

   stackoverflow.com/questions/41390162

   Thanks I went to the windows build page, with shared libraries, and copy and pasted the internals of the bin folder to the FFMPEG for audicity in C:/Program files, and pointed Audacity at the 57.dll library instead of 55.dll and its now processing. I had assumed that Audacity would not work with the new version. –

10. ffmpeg cannot open a simple microsoft wav file exported with...

    stackoverflow.com/questions/17809407

    You can get ffmpeg to recognize it with some additional input options: ffmpeg -acodec pcm_s16le -i steps-stereo-16b-44khz.wav output.ogg. This was a bug in ffmpeg, ticket #2810: unsupported wav, that has been fixed a while ago. You can compile ffmpeg or get a recent build to take advantage of the fix. Note that for ogg output the default ...

11. Reduce background noise and optimize the speech from an audio...

    superuser.com/questions/733061

    59. FFmpeg now has 3 native filters to deal with noise background: afftdn: Denoises audio samples with FFT. anlmdn: Reduces broadband noise in audio samples using a Non-Local Means algorithm. arnndn: Reduces noise from speech using Recurrent Neural Networks. Examples for model files to load can be found here.