Just use the command below:
ffmpeg -i INPUT_FILE \
-vf scale=3840x2160:flags=lanczos \
-c:v libx264 \
-crf 13 \
-c:a aac -b:a 512k \
-preset slow \
OUTPUT_FILE
Where:
INPUT_FILE
is the path to your input file-vf scale=3840x2160:flags=lanczos
is the new resolution using lanczos-c:v libx264
is the codec H264-crf 13
is the desired quality - From 0 to 51, the higher the number the worse. The smaller the number, the larger the file size-c:a aac -b:a 512k
is the audio codec and bitrate-preset slow
is the preset used. To go faster, useultrafast
OUTPUT_FILE
is the output file
Hope it helps.
This gist is great thanks for posting it and thanks to @jbrower95 for the awesome function. I made one minor tweak which was to add "" around the positional parameters to handle files with spaces in the names. Works flawless. Thanks again @luizomf!