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Why are audio files playing too fast?
Why do audio files recorded by call recording software play too fast? How can they be converted into normal, not accelerated speech?
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Most likely, the program is crooked and writes the wrong sample rate value to the file header. The solution, in this case, is to use ffmpeg to transfer to raw pcm, and then transfer from it with the correct (lower than now) sample rate value.
For example, so
ffmpeg -i input.wav -f s16le -acodec pcm_s16le output.pcm
ffmpeg -f s16le -ar 44.1k -ac 1 -i output.pcm file.wav
. s16le
you need to use s8
.
sox (from the command line) and audacity (GUI) can also help fix the sample rate.
If the menu is Asterisk, you can try to slip it mp3, or gsm with a sampling rate of 8 kHz
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