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(FFmpeg) How to Change the Pitch / Sample Rate of an Audio ...

    http://johnriselvato.com/ffmpeg-how-to-change-the-pitch-sample-rate-of-an-audio-track-mp3/#:~:text=%24%20ffmpeg%20-i%20input.mp3%20-af%20%22asetrate%3D44100%2A0.5%22%20output.mp3%20Tip%3A,the%20filter%20aresample%20is%20needed%2C%20as%20seen%20below%3A
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(FFmpeg) How to Change the Pitch / Sample Rate of an Audio ...

    http://johnriselvato.com/ffmpeg-how-to-change-the-pitch-sample-rate-of-an-audio-track-mp3/
    $ ffmpeg -i input.mp3 -af "asetrate=44100*0.5" output.mp3 . Tip: Changing the sample rate to change the pitch might create a conflict because some players or websites (like Bandcamp) require audio with a sample rate to be 44.1kHz. To keep the pitch change while setting the preferred sample rate the filter aresample is needed, as seen below:

FFmpeg change audio sample rate - Stack Overflow

    https://stackoverflow.com/questions/44569142/ffmpeg-change-audio-sample-rate

    ffmpeg Documentation

      https://ffmpeg.org/ffmpeg.html

      How to change sample rate in FFMPEG - VideoHelp Forum

        https://forum.videohelp.com/threads/246255-How-to-change-sample-rate-in-FFMPEG
        The output-file name should be the last thing in the command line. Perhaps you could try the following to verify: ffmpeg -i movie.avi -ar 22050 movie.flv Since the original sample rate, in this case, was 16000 Hertz, the -ar 44100 wouldn't give you any advantage over -ar 22050, and 22050 will generate a smaller output file (less bandwidth).

      How to use FFMpeg to do Simple Audio Conversion

        https://www.howtoforge.com/tutorial/ffmpeg-audio-conversion/

        ffmpeg guide · GitHub

          https://gist.github.com/protrolium/e0dbd4bb0f1a396fcb55

          FFMpeg Audio Decoding to Raw Samples · GitHub

            https://gist.github.com/jimjibone/6569303
            FFMpeg Audio Decoding to Raw Samples. GitHub Gist: instantly share code, notes, and snippets.

          FFmpeg Resampler Documentation

            https://ffmpeg.org/ffmpeg-resampler.html
            Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr (which, with a sample-rate of 44100, preserves the entire audio band to 20kHz). precision. For soxr only, the precision in bits to which the resampled signal will be calculated.

          using ffmpeg to extract audio from video files · GitHub

            https://gist.github.com/whizkydee/804d7e290f46c73f55a84db8a8936d74
            ffmpeg -formats. Convert WAV to MP3, mix down to mono (use 1 audio channel), set bit rate to 64 kbps and sample rate to 22050 Hz: ffmpeg -i input.wav -ac 1 -ab 64000 -ar 22050 output.mp3. Convert any MP3 file to WAV 16khz mono 16bit: ffmpeg -i 111.mp3 -acodec pcm_s16le -ac 1 …

          FFmpeg: Audio resampling

            https://ffmpeg.org/doxygen/0.11/group__lavc__resample.html
            Typedefs: typedef struct ReSampleContext : ReSampleContext: Functions: ReSampleContext * : av_audio_resample_init (int output_channels, int input_channels, int output_rate, int input_rate, enum AVSampleFormat sample_fmt_out, enum AVSampleFormat sample_fmt_in, int filter_length, int log2_phase_count, int linear, double cutoff): Initialize audio resampling context

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