Convert an Audio File
Description
Re-encode an audio (or A/V) file’s audio to a new container, sample rate, channel count, or codec via ffmpeg. The output format is inferred from the output extension. With no optional arguments it is a straight transcode.
Usage
audio_convert(input, output, sample_rate = NULL, channels = NULL,
bitrate = NULL, codec = NULL, overwrite = TRUE, dry_run = FALSE)
Arguments
input: Path to input audio (or A/V) file.output: Path for output file; extension selects the format.sample_rate: Output sample rate in Hz, or NULL to keep the source rate.channels: Output channel count (1 = mono, 2 = stereo), or NULL to keep.bitrate: Output audio bitrate (e.g. “192k”), or NULL for the encoder default. Ignored by lossless formats such as WAV.codec: Explicit audio codec (e.g. “libmp3lame”, “pcm_s16le”), or NULL to let ffmpeg pick from the output extension.overwrite: If TRUE (default), overwrite the output file.dry_run: If TRUE, return the ffmpeg command without executing.
Details
Resampling to 16 kHz mono is the usual preparation for speech-to-text:
audio_convert("audio.mp3", "audio.wav", sample_rate = 16000).
Value
Invisibly returns the output path. If dry_run, returns the command string.
Examples
audio_convert("audio.mp3", "speech.wav", sample_rate = 16000)
audio_convert("take.wav", "take.mp3", bitrate = "192k")