Ffmpegを使用してバッチ変換ファイルを作成する方法for sample rate conversion
、 といった:
$ ffmpeg -i *.wav -ar 22050 *.wav
私はサンプルレート変換しか行っていないことに注意してください。最終的には、すべての* .wavを22050 * .wavファイルに一括変換し、変換されたすべてのファイルに同じファイル名を保持します。
FFMpegは、エラーが発生する可能性があるため、読み取り中に同じファイルに書き込むことはできません。
それを行う唯一の方法は、別のファイルに変換して元のファイルを置き換えることです。
または、別のフォルダーに変換して元のフォルダーを置き換えることもできます。
Windowsの場合:
mkdir outdir
for %i in (*.bmp) do (
ffmpeg -i %i -ar 22050 outdir\%i
)
注:Windowsバッチファイルに配置する場合は、%iを%% iに置き換えます。
Linuxの場合:
mkdir outdir
for i in *.wav; do
ffmpeg -i $i -ar 22050 outdir/$i;
done
次に、ディレクトリをoutdirに置き換えます。
あなたの場合:
Counter Strike:GOまたはHLDJに必要な場合は、 ここ を参照してください。 HLにはさらに多くの設定が必要です。
オーディオはモノ(FFMpegの-ac 1
フラグを追加)および16ビット(-acodec pcm_s16le
を追加)である必要があります。
FFMpegコマンド(Linuxの場合)は次のようになります。
mkdir outdir
for i in *.wav; do
ffmpeg -i $i -acodec pcm_s16le -ac 1 -ar 22050 outdir/$i;
done
私はまさにそれを行うためのスクリプトを書きました。また、ワンライナーまたは短いbashスニペットで実行できる機能に加えて、いくつかの機能があります。
https://github.com/clone206/ffmpeg-batch-audio-resampler
投稿時のコード:
#!/bin/bash
########################
# CONSTANTS
########################
# The requested output sample rate.
MAX_SR=$1
# The specified output file extension
OUTFILE_EXT=$2
# The specified volume boost for output file
VOL_BOOST=$3
# Don't go below this sample rate:
MIN_SR=44100
# The directory to output converted files to. Use trailing slash. Needs manual escaping (not quotes)
# if this is customized and there are special chars (spaces, etc) in the pathname supplied, eg:
# OUT_DIR=~/Music/iTunes/iTunes\ Media/Automatically\ Add\ to\ iTunes.localized/
OUT_DIR=~/batch_resampled/
########################
# UTILITY FUNCTIONS
########################
# Send string to stderr
err() {
>&2 echo "$1"
}
# Display usage info
print_usage() {
err ""
err "USAGE: "$(basename $0)" <sample_rate> <outfile_extension> [vol_adjust_db]"
err ""
err "Converts all supported audio files in the current directory to the format corresponding "
err "to the given file extension (don't include the dot), at the speciied sample rate (in Hz). "
err "To specify a maximum output sample rate, where any input file of a greater rate gets downsampled "
err "to the nearest even multiple of either 44100 or 48000, add an 'm' to the end of the number, "
err "eg. '96000m'. If an input file has a sample rate that is already below this, it will not be upsampled. "
err ""
err "An optional volume adjust (in dB) can be given (positive number for boost, "
err "negative for cut). "
err ""
err "Renames file basenames on conversion and doesn't re-convert already "
err "converted files on subsequent runs."
err ""
err "Supported infile types: flac,dsf,dff,wav,aiff,m4a,mp3"
err "Supported outfile types: flac,wav,aiff,m4a(alac),mp3"
err ""
}
########################
# ARGS CHECKING
########################
if [[ -z "$OUTFILE_EXT" || $OUTFILE_EXT == "dsf" || $OUTFILE_EXT == "dff" ]]
then
print_usage
exit 1
fi
########################
# GLOBALS
########################
# Added to the end of the file basename
suffix=""
# Args to pass to the resampler
filter_args=""
pre_filter_args=""
# Args for the output file, like codec
out_args=""
# Volume filter level. Setting to 0dB explicitly seems to prevent replaygain-related clipping.
# Also comes in handy for boosting DSD file levels on conversion to PCM.
vol_level=""
# The sample rate of each input file
infile_sr=""
# The sample format of each input file
infile_sfmt=""
# Output file name
output_file=""
# Level of any errors encountered.
error_level=0
# Lowest factor of input file
lowest_factor=0
# Destination sample rate of a given output file
dest_sr=$MAX_SR
########################
# FUNCTIONS
########################
# Recalculate sample rate to nearest even multiple
# of 44.1/48K, depending on the input file's sample rate and the max sample rate given
recalc_sr() {
infile_sr=$1
lowest_factor=$2
max_sr=$(echo $MAX_SR | sed 's/m$//')
dest_sr=0
# Make sure our output sample rate is a multiple of either 44.1K or 48K
if [[ ! $(($max_sr % 44100)) -eq 0 && ! $(($max_sr % 48000)) -eq 0 ]]
then
err "ERROR: Only even multiples of 44100 and 48000 are allowed to be specified for maximum output sample rates!"
return 2
fi
# If the input is already at or below the specified max...
if [[ $infile_sr -le $max_sr ]]
then
# Nothing to calculate. Keep sr the same
dest_sr=$infile_sr
err "INFO: Input sample rate <= output rate. Will not be upsampled."
echo "$dest_sr"
return 0
# If user specified the min sr as the max
Elif [[ $max_sr -eq $MIN_SR ]]
then
# Downsample even if not even multiple
dest_sr=$max_sr
err "INFO: No even multiple available below minimum rate of ${MIN_SR}, so downsampling to ${MIN_SR}."
echo "$dest_sr"
return 0
else
# Find closest even multiple below the max. Takes advantage of the dropping of decimals by bash for easy math.
dest_sr=$(( ($max_sr/$lowest_factor)*$lowest_factor ))
err "INFO: Downsampling to even multiple of ${lowest_factor}"
echo "$dest_sr"
return 0
fi
}
# Takes input file sample rate as the only param.
# Determines whether this is an even multiple or 44.1K or 48K
# Delegates to the recalc_sr function and passes through the results via echo's
find_lowest_factor() {
infile_sr=$1
if [[ $infile_sr && $(($infile_sr % 44100)) -eq 0 ]]
then
echo "44100"
return 0
Elif [[ $infile_sr && $(($infile_sr % 48000)) -eq 0 ]]
then
echo "48000"
return 0
else
err "ERROR: Only multiples of 44100 and 48000 are allowed for input sample rates when in maximum mode!"
return 1
fi
}
# Takes an input file as the only argument and converts it
# based on certain conditions, or returns an error if necessary
conv() {
# The input file with "./" stripped from the beginning
item=${1#./}
err "item: $item"
# Get some specifics about this input file
# To see all info about a particular input file on the command line, type ffprobe -v error -show_streams <file>
infile_sfmt=$(ffprobe -v error -select_streams a:0 -show_entries stream=sample_fmt -of default=noprint_wrappers=1:nokey=1 "$item" \
| xargs echo -n)
infile_sr=$(ffprobe -v error -select_streams a:0 -show_entries stream=sample_rate -of default=noprint_wrappers=1:nokey=1 "$item" \
| xargs echo -n)
# If user appended an "m" for "maximum" to the end of
# the sample rate param, find nearest even multiple of
# this infile's sample rate
if [[ $MAX_SR =~ m$ ]]
then
err "INFO: Maximum sample rate specified for output. Recalculating destination sample rate"
# Get lowest factor from infile sr and skip this file on error
if ! lowest_factor=$(find_lowest_factor $infile_sr)
then
error_level=1
err "SKIPPING ${item}..."
return 0
fi
# Get destination sample rate for this file. Stop all conversion on error
if ! dest_sr=$(recalc_sr $infile_sr $lowest_factor)
then
error_level=2
err "FATAL"
return 1
fi
fi
# Append a marker to the end of the output filename indicating new sample rate
suffix="_ff"$(($dest_sr / 1000))"k"
filter_args="aresample=resampler=soxr:precision=32:dither_method=triangular:osr=${dest_sr}"
# Double pass for signed to signed resampling, as ffmpeg has some kind of problem
# with this while setting the output sample rate of a signed audio codec type w sox.
# This way it gets double dithered with sox, which seems to be the best way of
# avoiding the bug, which causes pops in the converted audio.
if [[ $infile_sfmt =~ ^s[0-9]+ && ($item =~ .flac$ || $item =~ .m[^\.]+$) ]]
then
pre_filter_args="aresample=resampler=soxr:precision=32:dither_method=triangular,"
fi
# Set the right output codec/sfmt for m4a, wav, aif, flac
if [[ $OUTFILE_EXT == "aiff" ]]
then
out_args="-acodec pcm_${infile_sfmt}be"
Elif [[ $OUTFILE_EXT == "wav" ]]
then
out_args="-acodec pcm_${infile_sfmt}le"
Elif [[ $OUTFILE_EXT == "m4a" ]]
then
out_args="-acodec alac"
# For some reason (perhaps bc of the precision set in sox resampler),
# signed pcm files are having their bit depth increased from 16 to 24 if we
# don't specify to keep the sample format the same.
Elif [[ $OUTFILE_EXT == "flac" && $infile_sfmt =~ ^s[0-9]+ ]]
then
if [[ $infile_sfmt =~ p$ ]]
then
# Flac doesn't support the "p" variants of signed sample fmts
infile_sfmt=${infile_sfmt%p}
fi
out_args="-sample_fmt ${infile_sfmt}"
fi
# Prepend output directory, strip infile extension, add suffix, add outfile extension
output_file=${OUT_DIR}${item%.*}${suffix}.${OUTFILE_EXT}
# Skip if we've already created this output file and it's not zero-size
if [[ -e $output_file && -s $output_file ]]
then
err "INFO: Output file already exists!"
err "SKIPPING ${output_file}"
return 0
fi
# Print some info about the output file to be created
err "filter_args: $pre_filter_args$filter_args$vol_level"
err "out_args: $out_args"
err "output_file: $output_file"
# Where the magic happens
ffmpeg -y -i "$item" -af "$pre_filter_args$filter_args$vol_level" $out_args "$output_file"
return 0
}
########################
# DRIVER CODE
########################
# Set volume boost/cut if given as argument
if [[ $VOL_BOOST ]]
then
vol_level=",volume=${3}dB"
fi
# Recreate the directory structure from the input dir in the output dir
# First create the output directory in case there are no subdirectories in CWD
mkdir "$OUT_DIR"
find . -mindepth 1 -type d -exec mkdir -p -- "${OUT_DIR}{}" \;
# Recursively loop through all supported audio files, call the conv function on each
# Using brace expansion to search both the current dir, and subdirs
for item in ./{*,**/*}.{flac,dsf,dff,wav,aiff,m4a,mp3}
do
# Skip unexpanded globs
if [[ -e $item ]]
then
# Reset some file-specific resampler args
pre_filter_args=""
out_args=""
# If we got a fatal error on trying to convert this file...
if ! conv "$item"
then
# Exit loop
break
fi
fi
done
# Not doing anything specific with the various error levels for now. Just checking for existence.
if [[ $error_level -ne 0 ]]
then
err ""
err "WARNING: Errors were encountered. Some or all files may not have been converted"
print_usage
exit 1
else
exit 0
fi