I'm calling ffmpeg from a program I'm writing in order to record audio from an audio interface. The audio interface has six channels and what I'd like to do is only record from the first two audio channels, discarding the rest. I can't work out how to do this or if it is even possible from the documentation.
The command I'm using is as follows:
ffmpeg -f alsa -acodec pcm_s32le -ac 6 -ar 44100 -i hw:CARD=K6,DEV=0 output.wav
Is this something that is possible? If so, how?
Use
ffmpeg -f alsa -acodec pcm_s32le -ac 6 -ar 44100 -i hw:CARD=K6,DEV=0
-af "pan=2c|c0=c0|c1=c1" output.wav
The first argument to the pan filter is the number of output channels. Then come the individual channel mixes. Here it is first out channel is first in channel, and a similar assignment for the second.
We want to convert 320kbps mp3 file to 128kbps mp3 so currently we are using below ffmpeg command but its not working.
ffmpeg -i input.mp3 -codec:a libmp3lame -qscale:a 5 output.mp3
Result:-the output bitrate same as input mp3.
And we are following the FFmpeg Encoding guideline for that here is the link :- https://trac.ffmpeg.org/wiki/Encode/MP3
so please suggest any solution.
I tried your shown command (tested on Windows / commandline) :
ffmpeg -i input.mp3 -codec:a libmp3lame -qscale:a 5 output.mp3
Result : It works for me. However the -qscale:a 5 makes FFmpeg decide on an average bitrate for you. With one (320k) MP3 file I got it giving a close convert of 134kbps. This is expected since :
lame option Average kbit/s Bitrate range kbit/s ffmpeg option
-V 5 130 120-150 -q:a 5
Solution :
Instead of making the internal mp3 frames hold different bitrates (that vary to acommodate the "current" perceived audio, eg: think "silent" parts using smaller rate of bits/bytes compared to "busy" audio parts), so just set a constant bitrate of 128kbps as you need.
I would just set it to constant 128kbps manually and explicitly with :
ffmpeg -i input.mp3 -codec:a libmp3lame -b:a 128k output.mp3
I use this shellscript in order to not visit this stackoverflow-page over and over again :)
#!/bin/bash
[[ ! -n $1 ]] && {
echo "Usage: mp3convert <input.mp3> <output.mp3> <bitrate:56/96/128/256> <channels> <samplerate>"
exit 0
}
set -x # print next command
ffmpeg -i "$1" -codec:a libmp3lame -b:a "$3"k -ac "$4" -ar $5 "$2"
Make sure your version of FFmpeg has libmp3lame enabled. The selected answer didn't work for me, but this did:
ffmpeg -v debug -i "input.mp3" -c:a libmp3lame \
-b:a 128k -ac 2 -ar 44100 -vn "output.mp3"
-ac 2 - output has 2 (stereo) audio channels
-ar 44100 - sample rate of 44100Hz, which is ideal for high quality music.
Although, in 2022 I wouldn't recommend converting to 128kbps since storage space is much more cheap and abundant nowadays.
I think -b:a 192k strikes the best balance between compression and quality for most people (unless you're an audiophile with $1000 headphones, and even then you'd be better off using FLAC anyways).
I have a webcam and a separate mic. I want to record what is happening.
It almost works, however the audio seems to play quickly and parts missing while playing over the video.
This is the command I am currently using to get it partially working
ffmpeg -thread_queue_size 1024 -f alsa -ac 1 -i plughw:1,0 -f video4linux2 -thread_queue_size 1024 -re -s 1280x720 -i /dev/video0 -r 25 -f avi -q:a 2 -acodec libmp3lame -ab 96k out.mp4
I have tried other arguments, but unsure if it has to do with the formats I am using or incorrect parameter settings.
Also, the next part would be how to stream it. Everytime I try going through rtp it complains about multiple streams. I tried doing html as well, but didn't like the format. html html://localhost:50000/live_feed or rts rts://localhost:5000
edit:
I am running this on a rpi 3.
I used FFmpeg to generate a test clip with color bars and a tone. I also made a special filter to dump out the raw audio data to check it. I was surprised to find that there is significant noise riding on the audio tone after it has gone through the AAC codec. Is this expected? Is there a way to prevent it?
To make the test file I used:
ffmpeg -f lavfi -i "smptehdbars=duration=600:size=1280x720:rate=59.94" -qscale:v 1 -pix_fmt yuv420p smpte_r59_720.mp4
then
ffmpeg -i smpte_r59_720.mp4 -f lavfi -i "sine=frequency=1000:sample_rate=48000:duration=600" -qscale:v 1 -vcodec copy -c:a aac -b:a 192k -shortest -map 0:0 -map 1:0 smpte_r59_720T.mp4
and then
ffmpeg -i smpte_r59_720T.mp4 -y -map 0 -acodec aac -vcodec libx264 -crf 23 -bsf:v h264_mp4toannexb smpte_r59_720T.ts
(Trying to do this all in one step kept failing.)
Other variations on this have varying degrees of noise, sometimes above nominal amplitude and sometimes below.
After finding this problem I pulled a third party test tone .WAV file with 44.1KHz sample rate from the web and checked it. The raw file is clean, but the encoded file I made has noise.
Noise on TS SMPTE bars
Clean audio from MP4
My goal is to have a script that takes an audio file and increases its volume by 50%.
I currently use the following AutoHotKey snippet to encode a file to MP3:
run_string := "bash -c ""\""c:\Program Files\VideoLAN\VLC\vlc.exe\"" -I dummy \""" . file_path . "\"" --sout='#transcode{acodec=mp3,vcodec=dummy}:standard{access=file,mux=raw,dst=\""" . file_path . ".mp3\""}' vlc://quit"""
How can I modify this line to not only encode to mp3, but also increase the volume of the file by 50%? I tried setting --volume 150 but it just made the file play, while I don't want to play, I want to have it saved with that volume.
If you have suggestions for other Windows-compatible tools to modify audio that can do this, (along with instructions on how to do this) I'll be happy to hear about them.
I suggest you to use ffmpeg. it is very powerful, cross platform 32 or 64 bit, audio and video converter. Can be downloaded from Zeranoe FFmpeg - Builds
Below sample commands work for audio extracting from video, or audio converter with volume increasing or decreasing support.
Extract audio from video to MP3, or convert audio to MP3 (sample InputFilePath_VideoOrAudio = "e:\video.mp4" or "e:\audio.m4a")
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec libmp3lame -ab 192k -ar 48000 -sn -dn -vn "E:\out.mp3"
Extract audio from video to MP3 and increase volume 150% while extracting add -af "volume=1.5" parameter.
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec libmp3lame -ab 192k -ar 48000 -sn -dn -vn -af "volume=1.5" "E:\out.mp3"
List of audio converter parameters (mp3,ogg,ac3,wma,flac,wav,aiff,m4a....). to change volume level while converting to audio add -af "volume=VolumeValue" parameter.
VolumeValue=0.5 decrease volume %50
VolumeValue=1.5 increase volume %150
VolumeValue=2.0 increase volume %200 and so on.
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec libmp3lame -ab 192k -ar 48000 -sn -dn -vn -af "E:\out.mp3"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec ac3 -ab 192k -ar 48000 -sn -dn -vn "E:\out.ac3"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -f ogg -acodec libvorbis -ab 192k -ar 48000 -sn -dn -vn "E:\out.ogg"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec wmav2 -ab 192k -ar 48000 -sn -dn -vn "E:\out.wma"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec flac -sn -dn -vn "E:\out.flac"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -sn -dn -vn "E:\out.wav"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -f aiff -sn -dn -vn "E:\out.aiff"
e:\ffmpeg\ffmpeg.exe -y -i "InputFilePath_VideoOrAudio" -acodec aac -ab 192k -ar 48000 -sn -dn -vn "E:\out.m4a"
Note 1: some codecs can be experimental in such case you should use -strict experimental or -strict -2 parameters.
Note 2: -ab parameter means audio bit rate. Some devices can not play audio file that bit rate greater than -ab 192k. Use -ab 128k or -ab 192k with -ar 44100 parameters to produce audio file that can be playable most of the mobile devices. -ac 2 parameter means stereo -ac 1 means mono.
to convert specific part of the input file use -ss 00:00:00 and -t parameters. -ss means Start From -t means duration. Important: parameter -ss should placed before the -i parameter, otherwise ffmpeg seeks to -ss position slowly.
Samples: assume that input file duration is 00:20:00 (20 minutes)
using only -ss 00:05:00 means convert input file starting from 5th minute to end of the input file. Duration of the output file will be 15 minutes.
using -ss 00:05:00 with -t 120 or -t 00:02:00 means convert 120 seconds, starting from 5th minute. Duration of the output file will be 120 seconds.
e:\ffmpeg\ffmpeg.exe -y -ss 00:05:00 -i "InputFilePath_VideoOrAudio" -t 120 -acodec libmp3lame -ab 192k -ar 48000 -sn -dn -vn -af "E:\out.mp3"
Note: -y means in advance YES to ffmpeg's yes/no questions such as output file already exist, over write? with -y parameters ffmpeg over writes the output file if it is already exist without asking the user.
-sn disables subtitle, -vn disable video, -dn disable data streams for output file.
If you just want a CLI tool then you could use ffmpeg:
ffmpeg.exe -i test.mp3 -af volume=1.5 loud.mp3
^ ^ ^
input new volume level output name
If you'd like to be able to do it programmatically, looking at your profile I deduced that python should not be a problem :)
So you can use the nice pydub module together with ffmpeg (or avconv which it also supports) for your task.
E.g:
from pydub import AudioSegment
AudioSegment.converter = r"C:\PATH_TO_FFMPEG_DIR\bin\ffmpeg.exe"
sound = AudioSegment.from_mp3("test.mp3") # <- the input file
new = sound.export("loud.mp3", format="mp3", parameters=["-vol", "384"]) # 384 <-> 150% volume
new.flush()
new.close()
The reason for 384 is that the ffmpeg doc states that
-vol volume change audio volume (256=normal)
So 256*1.5 = 384
Tested this on my windows 7 machine just now...
Hope this helps.
The "--volume" option in VLC doesn't actually change the volume of the output video as you would think it would. What you want to do is add the compressor filter and then set the "compressor-makup-gain". Set it to a value from 1-24 depending on how loud you want the video to be. So your command would be something like this:
run_string := "bash -c ""\""c:\Program Files\VideoLAN\VLC\vlc.exe\"" -I dummy \""" . file_path . "\"" --sout='#transcode{acodec=mp3,vcodec=dummy,afilter=compressor}:standard{access=file,mux=raw,dst=\""" . file_path . ".mp3 --compressor-makeup-gain=20\""}' vlc://quit"""
By the way, for anyone who is trying to figure out how to use VLC to increase the volume of the audio in a video file, here's how you can do that:
"C:\Program Files (x86)\VideoLAN\VLC\vlc.exe" yoursourcefile.mp4 :sout=#transcode{acodec=mp3,ab=128,channels=2,samplerate=44100,afilter=compressor}:file{dst=outputfilename.mp4} :sout-all :sout-keep --compressor-makeup-gain=20
Replace "yoursourcefile.mp4" and "outputfilename.mp4" with your own file names. In my experience, VLC crashed about half the time I ran this command, so you may need to try it more than once if it crashes on you.
Run this on a dir to increasing all files volume on that dir, one by one (or else it would eat up all CPU)
FOR %f IN (*) DO (start /wait "" "C:\Program Files
(x86)\VideoLAN\VLC\vlc.exe" %f
:sout=#transcode{acodec=mp3,afilter=compressor}:file{dst=Boost%f}
:sout-all :sout-keep --play-and-exit --compressor-makeup-gain=10)
I believe mp3gain has a command line option for this. You could run this as a separate pass over the generated file:
http://mp3gain.sourceforge.net/