The best way to maintain quality whilst not having WAV-sized files is to use the FLAC (with -compression_level 12 for the lowest file size) or ALAC encoder. Unfortunately stream copying is not possible when using the -filter option (in this case -filter:a loudnorm). 24-bit WAV to 16-bit WAV) and maintaining the quality.Īnother way of keeping the audio codec and bitrate the same whilst maintaining quality is by using -c:a copy, which leaves the audio as-is (i.e. If you're using a lossless encoder, you are keeping the bitrate the same (unless you're reducing the bit depth, e.g. Just being pedantic and making things 100% clear for whoever reads this - keeping the bitrate the same when encoding will not retain the audio quality if you're using a lossy encoder. The user should choose an appropriate bitrate or VBR factor for their conversions. To use the FFmpeg examples / commands below, you'll need to have FFmpeg installed on your system. If the output looks good, then try a higher value. Choose the highest CRF value that still provides an acceptable quality. So I'm hesitant to put that in as an option. This article explains how to use FFmpeg to extract the audio stream from a video, either without re-encoding (keeping the original format), or converting the resulting audio file to CBR (constant bitrate) or VBR (variable bitrate) MP3 or Ogg Vorbis. The range is exponential, so increasing the CRF value +6 results in roughly half the bitrate / file size, while -6 leads to roughly twice the bitrate. Keeping the bitrate does not mean keeping the same quality, as files will have generation loss.
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