ffmpeg: Removing parts with silence: Efficiently remove several parts of a Video

Question

I'm trying to remove parts of a video, where there is no (or little) audio. For this I use audacity (Analyze->Sound finder) and export the Label Track. This results in something like this:

0.000000    5.170000    1
12.520000   12.630000   2
14.400000   15.660000   3
17.430000   22.150000   4
...

With this regex: ([\d]+.[\d]{6})\t([\d]+.[\d]{6})\t([\d]+)\n > ffmpeg -i VL1.mp4 -strict -2 -ss \1 -to \2 buff.mp4\nffmpeg -f concat -i <(echo "file '/home/rene/videotest/output.mp4'\nfile '/home/rene/videotest/buff.mp4'") -c copy output2.mp4\nrm buff.mp4\nmv output2.mp4 output.mp4\n and some manual editing I get this:

#!/bin/bash

ffmpeg -i VL1.mp4 -strict -2 -ss 0.000000 -to 5.170000 output.mp4
ffmpeg -i VL1.mp4 -strict -2 -ss 12.520000 -to 12.630000 buff.mp4
ffmpeg -f concat -i <(echo "file '/home/rene/videotest/output.mp4'
file '/home/rene/videotest/buff.mp4'") -c copy output2.mp4
rm buff.mp4
mv output2.mp4 output.mp4
ffmpeg -i VL1.mp4 -strict -2 -ss 14.400000 -to 15.660000 buff.mp4
ffmpeg -f concat -i <(echo "file '/home/rene/videotest/output.mp4'
file '/home/rene/videotest/buff.mp4'") -c copy output2.mp4
rm buff.mp4
mv output2.mp4 output.mp4
ffmpeg -i VL1.mp4 -strict -2 -ss 17.430000 -to 22.150000 buff.mp4
ffmpeg -f concat -i <(echo "file '/home/rene/videotest/output.mp4'
file '/home/rene/videotest/buff.mp4'") -c copy output2.mp4
rm buff.mp4
mv output2.mp4 output.mp4
...

This works quite fine and I get a file output.mp4 that contains only the parts I want.

Unfortunately this is incredibly slow - and I fear it destroys my HDD (since for every part there is a file buff.mp4 created, a new file containing all existing parts plus this short clip is written to the HDD and the old file containing the existing parts is deleted). Since most of the clips are REALLY short (usually less than a few secs) my idea was to not write the file buff.mp4 to HDD but instead pass this to ffmpeg via < (so keep it in RAM?). The second thing was to append to output.mp4 and not rewriting the file each time.

Unfortunately for both things my knowledge of ffmpeg and bash in general is too slim. Could someone help me out?

I should be able to write a nice script without the manual regex stuff. And of course I'll publish this afterwards.

Answer 1

https://github.com/carykh/jumpcutter (MIT license) automatically removes parts of a video that have no or little audio. It relies on ffmpeg and the pipeline is coded in Python 3 (runs on Ubuntu, Microsoft Windows, and probably Mac OS X).

Explanation: https://www.youtube.com/watch?v=DQ8orIurGxw

Script (MIT license, author: carykh):

from contextlib import closing
from PIL import Image
import subprocess
from audiotsm import phasevocoder
from audiotsm.io.wav import WavReader, WavWriter
from scipy.io import wavfile
import numpy as np
import re
import math
from shutil import copyfile, rmtree
import os
import argparse
from pytube import YouTube

def downloadFile(url):
    name = YouTube(url).streams.first().download()
    newname = name.replace(' ','_')
    os.rename(name,newname)
    return newname

def getMaxVolume(s):
    maxv = float(np.max(s))
    minv = float(np.min(s))
    return max(maxv,-minv)

def copyFrame(inputFrame,outputFrame):
    src = TEMP_FOLDER+"/frame{:06d}".format(inputFrame+1)+".jpg"
    dst = TEMP_FOLDER+"/newFrame{:06d}".format(outputFrame+1)+".jpg"
    if not os.path.isfile(src):
        return False
    copyfile(src, dst)
    if outputFrame%20 == 19:
        print(str(outputFrame+1)+" time-altered frames saved.")
    return True

def inputToOutputFilename(filename):
    dotIndex = filename.rfind(".")
    return filename[:dotIndex]+"_ALTERED"+filename[dotIndex:]

def createPath(s):
    #assert (not os.path.exists(s)), "The filepath "+s+" already exists. Don't want to overwrite it. Aborting."

    try:  
        os.mkdir(s)
    except OSError:  
        assert False, "Creation of the directory %s failed. (The TEMP folder may already exist. Delete or rename it, and try again.)"

def deletePath(s): # Dangerous! Watch out!
    try:  
        rmtree(s,ignore_errors=False)
    except OSError:  
        print ("Deletion of the directory %s failed" % s)
        print(OSError)

parser = argparse.ArgumentParser(description='Modifies a video file to play at different speeds when there is sound vs. silence.')
parser.add_argument('--input_file', type=str,  help='the video file you want modified')
parser.add_argument('--url', type=str, help='A youtube url to download and process')
parser.add_argument('--output_file', type=str, default="", help="the output file. (optional. if not included, it'll just modify the input file name)")
parser.add_argument('--silent_threshold', type=float, default=0.03, help="the volume amount that frames' audio needs to surpass to be consider \"sounded\". It ranges from 0 (silence) to 1 (max volume)")
parser.add_argument('--sounded_speed', type=float, default=1.00, help="the speed that sounded (spoken) frames should be played at. Typically 1.")
parser.add_argument('--silent_speed', type=float, default=5.00, help="the speed that silent frames should be played at. 999999 for jumpcutting.")
parser.add_argument('--frame_margin', type=float, default=1, help="some silent frames adjacent to sounded frames are included to provide context. How many frames on either the side of speech should be included? That's this variable.")
parser.add_argument('--sample_rate', type=float, default=44100, help="sample rate of the input and output videos")
parser.add_argument('--frame_rate', type=float, default=30, help="frame rate of the input and output videos. optional... I try to find it out myself, but it doesn't always work.")
parser.add_argument('--frame_quality', type=int, default=3, help="quality of frames to be extracted from input video. 1 is highest, 31 is lowest, 3 is the default.")

args = parser.parse_args()



frameRate = args.frame_rate
SAMPLE_RATE = args.sample_rate
SILENT_THRESHOLD = args.silent_threshold
FRAME_SPREADAGE = args.frame_margin
NEW_SPEED = [args.silent_speed, args.sounded_speed]
if args.url != None:
    INPUT_FILE = downloadFile(args.url)
else:
    INPUT_FILE = args.input_file
URL = args.url
FRAME_QUALITY = args.frame_quality

assert INPUT_FILE != None , "why u put no input file, that dum"

if len(args.output_file) >= 1:
    OUTPUT_FILE = args.output_file
else:
    OUTPUT_FILE = inputToOutputFilename(INPUT_FILE)

TEMP_FOLDER = "TEMP"
AUDIO_FADE_ENVELOPE_SIZE = 400 # smooth out transitiion's audio by quickly fading in/out (arbitrary magic number whatever)

createPath(TEMP_FOLDER)

command = "ffmpeg -i "+INPUT_FILE+" -qscale:v "+str(FRAME_QUALITY)+" "+TEMP_FOLDER+"/frame%06d.jpg -hide_banner"
subprocess.call(command, shell=True)

command = "ffmpeg -i "+INPUT_FILE+" -ab 160k -ac 2 -ar "+str(SAMPLE_RATE)+" -vn "+TEMP_FOLDER+"/audio.wav"

subprocess.call(command, shell=True)

command = "ffmpeg -i "+TEMP_FOLDER+"/input.mp4 2>&1"
f = open(TEMP_FOLDER+"/params.txt", "w")
subprocess.call(command, shell=True, stdout=f)



sampleRate, audioData = wavfile.read(TEMP_FOLDER+"/audio.wav")
audioSampleCount = audioData.shape[0]
maxAudioVolume = getMaxVolume(audioData)

f = open(TEMP_FOLDER+"/params.txt", 'r+')
pre_params = f.read()
f.close()
params = pre_params.split('\n')
for line in params:
    m = re.search('Stream #.*Video.* ([0-9]*) fps',line)
    if m is not None:
        frameRate = float(m.group(1))

samplesPerFrame = sampleRate/frameRate

audioFrameCount = int(math.ceil(audioSampleCount/samplesPerFrame))

hasLoudAudio = np.zeros((audioFrameCount))



for i in range(audioFrameCount):
    start = int(i*samplesPerFrame)
    end = min(int((i+1)*samplesPerFrame),audioSampleCount)
    audiochunks = audioData[start:end]
    maxchunksVolume = float(getMaxVolume(audiochunks))/maxAudioVolume
    if maxchunksVolume >= SILENT_THRESHOLD:
        hasLoudAudio[i] = 1

chunks = [[0,0,0]]
shouldIncludeFrame = np.zeros((audioFrameCount))
for i in range(audioFrameCount):
    start = int(max(0,i-FRAME_SPREADAGE))
    end = int(min(audioFrameCount,i+1+FRAME_SPREADAGE))
    shouldIncludeFrame[i] = np.max(hasLoudAudio[start:end])
    if (i >= 1 and shouldIncludeFrame[i] != shouldIncludeFrame[i-1]): # Did we flip?
        chunks.append([chunks[-1][1],i,shouldIncludeFrame[i-1]])

chunks.append([chunks[-1][1],audioFrameCount,shouldIncludeFrame[i-1]])
chunks = chunks[1:]

outputAudioData = np.zeros((0,audioData.shape[1]))
outputPointer = 0

lastExistingFrame = None
for chunk in chunks:
    audioChunk = audioData[int(chunk[0]*samplesPerFrame):int(chunk[1]*samplesPerFrame)]

    sFile = TEMP_FOLDER+"/tempStart.wav"
    eFile = TEMP_FOLDER+"/tempEnd.wav"
    wavfile.write(sFile,SAMPLE_RATE,audioChunk)
    with WavReader(sFile) as reader:
        with WavWriter(eFile, reader.channels, reader.samplerate) as writer:
            tsm = phasevocoder(reader.channels, speed=NEW_SPEED[int(chunk[2])])
            tsm.run(reader, writer)
    _, alteredAudioData = wavfile.read(eFile)
    leng = alteredAudioData.shape[0]
    endPointer = outputPointer+leng
    outputAudioData = np.concatenate((outputAudioData,alteredAudioData/maxAudioVolume))

    #outputAudioData[outputPointer:endPointer] = alteredAudioData/maxAudioVolume

    # smooth out transitiion's audio by quickly fading in/out

    if leng < AUDIO_FADE_ENVELOPE_SIZE:
        outputAudioData[outputPointer:endPointer] = 0 # audio is less than 0.01 sec, let's just remove it.
    else:
        premask = np.arange(AUDIO_FADE_ENVELOPE_SIZE)/AUDIO_FADE_ENVELOPE_SIZE
        mask = np.repeat(premask[:, np.newaxis],2,axis=1) # make the fade-envelope mask stereo
        outputAudioData[outputPointer:outputPointer+AUDIO_FADE_ENVELOPE_SIZE] *= mask
        outputAudioData[endPointer-AUDIO_FADE_ENVELOPE_SIZE:endPointer] *= 1-mask

    startOutputFrame = int(math.ceil(outputPointer/samplesPerFrame))
    endOutputFrame = int(math.ceil(endPointer/samplesPerFrame))
    for outputFrame in range(startOutputFrame, endOutputFrame):
        inputFrame = int(chunk[0]+NEW_SPEED[int(chunk[2])]*(outputFrame-startOutputFrame))
        didItWork = copyFrame(inputFrame,outputFrame)
        if didItWork:
            lastExistingFrame = inputFrame
        else:
            copyFrame(lastExistingFrame,outputFrame)

    outputPointer = endPointer

wavfile.write(TEMP_FOLDER+"/audioNew.wav",SAMPLE_RATE,outputAudioData)

'''
outputFrame = math.ceil(outputPointer/samplesPerFrame)
for endGap in range(outputFrame,audioFrameCount):
    copyFrame(int(audioSampleCount/samplesPerFrame)-1,endGap)
'''

command = "ffmpeg -framerate "+str(frameRate)+" -i "+TEMP_FOLDER+"/newFrame%06d.jpg -i "+TEMP_FOLDER+"/audioNew.wav -strict -2 "+OUTPUT_FILE
subprocess.call(command, shell=True)

deletePath(TEMP_FOLDER)

Answer 2

Expanding on Franck's answer, there are better alternatives to Jumpcutter:

https://github.com/carykh/jumpcutter/issues/180

Edit: Unfortunately this issue has been deleted on Github, but you can see it on from Web Archive: https://web.archive.org/web/20201203163529/https://github.com/carykh/jumpcutter/issues/180

Use Auto Editor:

https://github.com/WyattBlue/auto-editor

Answer 3

Project on Github

I created my own script, that works fine for me:
https://github.com/DarkTrick/python-video-silence-cutter

Logic for cutting silences with ffmpeg

1. Find all silence parts with ffmpeg:
   ffmpeg -i ./example.mp4 -af silencedetect=n=-35dB:d=1 -f null -

2. [This will output to stdout (apparently file storing is not possible). ]

3. prepare two files: one with a video filter, one with an audio filter. These files will tell ffmpeg which parts of the video to use (not to cut)

   • Video filter file content (here video_filter.txt):
     select='between(t,0,2.94075)+between(t,5.6015,5.60152)', setpts=N/FRAME_RATE/TB
   • Audio filter file content (here audio_filter.txt):
     aselect='between(t,0,2.94075)+between(t,5.6015,5.60152)', asetpts=N/SR/TB
   • [You can see their content is almost identical]

4. Run ffmpeg to create the new video:
   ffmpeg -i ./example.mp4 -filter_script:v ./video_filter.txt -filter_script:a ./audio_filter.txt ./example_cut.mp4

Answer 4

The solution I use right now scans the list-input-file and creates small files with "no-silence" snippets from this information. All files are written into the file temp in correct order. When this is done, the parts will be merged and the temporary files will be cleaned. This is not very fast. A total of about 1000 cuts in a 90 Minute clip takes about 5 hours on an i7 processor, 10GB RAM.

I guess a solution with ffmpeg filters would be much faster. But it was too complicated for the moment. Another solution could be this code-example. Both would probably be more efficient because they don't create the tmp-files (with headers, file write, etc).

Anyway, here's the working code in it's current state:

#!/bin/bash
echo "">temp

# File or Directory selection menu with dialog
function fileDialog {
    read -p "$2: " fileselection
    echo "You've selected $fileselection"
}


WORKPATH=$(pwd)

echo "Arbeitsverzeichnis: $WORKPATH"

fileDialog "$WORKPATH" "Select original Video-File"
INFILE=$fileselection
ff=$(ffmpeg -i "$INFILE" 2>&1)
d="${ff#*Duration: }"
LENGTH="${d%%,*}"
echo "Arbeitsverzeichnis: $WORKPATH"
echo "Videodatei: $INFILE"
echo "Gesamtvideolänge $LENGTH"

fileDialog "$WORKPATH" "Select Label-List-File (exported from Audacity)"
LABELFILE=$fileselection
NUMLINES=$(awk 'END { print NR }' "$LABELFILE")
clear
echo "Arbeitsverzeichnis: $WORKPATH"
echo "Videodatei: $INFILE"
echo "Gesamtvideolänge $LENGTH"
echo "Sequence-Label-Datei: $LABELFILE"
echo $NUMLINES lines/sequences found in \'$LABELFILE\'
echo ""
echo "Start processing..."
awk '{d=sprintf("echo Processing Part "NR"/'"$NUMLINES"' named "$3" from "$1"s to "$2"s..."); system(d); s=sprintf("ffmpeg -hide_banner -loglevel panic -accurate_seek -y -i '"$INFILE"' -strict -2 -ss "$1" -to "$2" tmp-"NR".mp4>/dev/null"); system(s); p=sprintf("echo \"$(cat temp)\nfile '"$WORKPATH"'/tmp-"NR".mp4\">temp"); system(p);}' "$LABELFILE"

echo "Merging all "$NUMLINES" parts."
ffmpeg -f concat -i temp -c copy "short-$INFILE.mp4"

echo "Cleaning up..."
rm tmp-*.mp4
rm temp
rm "$INFILE"

Two more small things that should be done: currently this will convert your video to mp4. This is not really necessary and probably only requires loads of CPU time.