I'm looking for a way to convert wav(16bit, 48kHz, LPCM) into an mxf file with timecode.
Since ffmpeg supports mxf, I'm trying, but I don't know the command.
ffmpeg -i ./input.wav [hh:mm:ss.ff, name1] [hh:mm:ss.ff, name2]... ./output.mxf
I'm expecting the above command, but does anyone know?
MXF is a pain
The default MXF muxer requires video.
The -timecode option with MXF requires video.
The mxf_opatom muxer allows just audio, but only mono with 48000 MHz sample rate, so each channel will need to be in its own MXF file.
Workaround 1: Pipe
ffmpeg -i input.wav -ar 48000 -c:a pcm_s16le -timecode 01:02:03:04 -f nut - | ffmpeg -i - -c:a pcm_s16le -f mxf_opatom output.mxf
I'm assuming your audio is mono (you didn't say what it is). If your input is multichannel then output each channel into its own file.
Use 01:02:03:04 for non-drop timecode, and 01:02:03.04 or 01:02:03;04 for drop.
Workaround 2: Dummy/blank video
Just ignore the video.
Non-drop timecode:
ffmpeg -f lavfi -i color=r=25 -i input.wav -timecode 01:02:03:04 -c:a copy -shortest output.mxf
Drop timecode:
ffmpeg -f lavfi -i color=r=30000/1001 -i input.wav -timecode 01:02:03.04 -c:a copy -shortest output.mxf
Related
We need to detect the 'silence'(s) in the audio channel of a video stream. We have been able to receive a UDP video stream and extract audio from it using the command:
ffmpeg -y -i udp://127.0.0.1:23000 -ab 3000k -ar 44100 -ac 1 test.wav
The audio file was saved only to verify whether audio has been extracted correctly or not.
To detect 'silence'(s) in the audio, we are using the silencedetect filter. We referred to some examples and it seems to work for audio files:
ffmpeg -i audio/file/path -af silencedetect=noise=-50dB:d=0.25 -f null -
We are unable to detect silence(s) in the audio from a video stream. This is the command we came up with:
ffmpeg -y -i udp://127.0.0.1:23000 -ab 3000k -ar 44100 -ac 1 -af silencedetect=noise=-50dB:d=0.25 -f null -
What is it that we are doing wrong? Any help would be appreciated.
Thanks!
I am using ffmpeg to extract the audio from a video. Below code downlaods the audio from a video file. I'm not sure how efficient this program is but I do know that it downloaods it in 48KHZ.
How do I use this program to extract audio from a video in 8Khz because the file is getting too big.
ffmpeg -i video_link -vn output.wav
Use -ar option to change frequency rate
ffmpeg -i video_link -vn -ar 8000 output.wav
If you want to try different formats of audio check the available formats in ffmpeg using ffmpeg -formats and available codecs using ffmpeg -codecs
Here's an example to extract to mp3 file
ffmpeg -i video_link -vn -ar 8000 -f mp3 output.mp3
Edit: as #llogan pointed out, -f option is not needed, ffmpeg automatically mux mp3 file.
ffmpeg -i video_link -vn -ar 8000 output.mp3
I'm trying to calculate the audio + visual difference between a harshly compressed video file and one that hasn't been.
I'm using pipes because ultimately I wish this to take src from a camera stream.
I've managed to get the video results that I'm looking for, but I'm struggling with the audio.
I've added a line to invert the phase of the compressed audio, so that when they add up in the blend they should almost cancel each other out, but that doesn't happen.
ffmpeg -i input.avi -f avi -c:v libxvid -qscale:v 30 -c:a wmav1 - | \
ffmpeg -i - -f avi -af "aeval='-val(0)':c=same" - | \
ffmpeg -i input.avi -i - -filter_complex "blend=all_mode=difference" -c:v libx264 -crf 18 -f avi - | \
ffplay -
I can still hear all the audio, when what I should be hearing are solely compression artifacts. thx
To preface, I'm not sure your method would identify audio compression 'artifacts'
Your command doesn't perform any audio comparison, it only inverts a single channel. Also, the audio and video are compressed twice and the codecs the last ffmpeg command receives are the default AVI codecs of mpeg4 and mp3.
Use
ffmpeg -i input.avi -f matroska -c:v libxvid -qscale:v 30 -c:a wmav1 - |\
ffmpeg -i input.avi -i - -filter_complex "[0][1]blend=all_mode=difference;[1]aselect=gt(n\,0),asetpts=PTS-STARTPTS[1a];[0][1a]amerge,aeval=val(0)-val(1):c=mono" -c:v rawvideo -c:a pcm_s16le -f matroska - |\
ffplay -
I assume your audio is mono. If your audio has N channels, your aeval will need N expressions where the Mth expression is val(M-1)-val(N+M-1)
I also trim out the first encoded audio frame in order to mitigate encoder delay that Paul mentioned, and it seems to work here.
There might be some delay introduced with encoded audio samples. Also your command is incorrect.
I have some video files that I need to re-encode due to compatibility issues. They are currently mkv files with h.264 video and ac3-a52 audio. I want to keep the h.264 video, convert the container to m4v and create two audio tracks, one with the original ac3-a52 and one copied from that but in aac stereo.
I assume there has to be some sort of audio stream mapping command but I don't know how to map and re-encode at the same time. What command should I enter into ffmpeg to achieve this?
Also, what is the difference between ac3 and ac3-a52? Will an apple TV still be able to pass through ac3-a52 or does that have to be converted to ac3?
this works for me:
ffmpeg -y -i Source.mkv -map 0:v -c:v copy -map 0:a -c:a copy -map 0:a -strict -2 -c:a aac out.mkv
-y – A global option to overwrite the output file if it already exists.
-map 0:v – Designate the video stream(s) from the first input as a source for the output file.
-c:v copy – Stream copy the video. This just muxes the input to the output. No re-encoding occurs.
-map 0:a – Designate the audio stream(s) from the first input as a source for the output file.
-c:a copy – Stream copy the audio. This just muxes the input to the output. No re-encoding occurs.
-strict -2 -c:a aac – Use the native FFmpeg AAC audio encoder. -strict -2 is required as a way that you acknowledge that the encoder is designated as experimental. It is not a great encoder, but it is not too bad at higher bitrates.
According to wikipedia, there is no difference between AC3 and ATSC A/52: the 1st one is the name of the codec, the 2nd is the name of the standard specifying the AC3 codec. Maybe someone have more knowledge about it?
I'm doing the same as the OP, but with an m4v container. I'm using the MacPorts "nonfree" variant of ffmpeg so that I can use libfaac, which gives better audio quality than the built-in AAC encoder and also had the same issue as #dkam. The command line I ended using is like this:
ffmpeg -i input.m4v -map 0:v -c:v copy -map 0:a -c:a:0 copy -map 0:a -c:a:1 libfaac output.m4v
(The videos are for playback on an iPad, which doesn't seem to be able to handle ac3.)
This command will take a video with 1 audio stream, and downmix to stereo and convert the audio stream and add it as a 2nd audio stream. It will be in AAC 384k.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:1 aac -b:a 384k -ac 2 OUTPUT.mkv
Explanation of the command
ffmpeg -i INPUT.mkv The application and input file
-strict -2 Enable downmixing support
-map 0 Tell ffmpeg read all Video, Audio, and Subtitle streams for the following arguments
-c copy Copy everything
-map 0:a:0 Tell ffmpeg to read the first audio stream for the following arguments
-c:a:1 aac Output the audio to a 2nd audio channel (0 = first channel) in aac format. Important! You must change the output channel to a higher number if there are multiple audio streams to prevent overwriting them.
-b:a 384k 384k bitrate (I don't know what's good for aac stereo but this is really high since it's for 5.1 aac)
-ac 2 Downmix to stereo
OUTPUT.mkv Output file
More examples
A video with two audio streams. Creating a third audio stream by encoding the first.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv
Again a video with two audio streams, but you want to encode the second one
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:1 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv
Can I overlay/downmix two audio mp3 files into one mp3 output file using ffmpeg?
stereo + stereo → stereo
Normal downmix
Use the amix filter:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex amix=inputs=2:duration=longest output.mp3
Or the amerge filter:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex amerge=inputs=2 -ac 2 output.mp3
Downmix each input into specific output channel
Use the amerge and pan filters:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex "amerge=inputs=2,pan=stereo|c0<c0+c1|c1<c2+c3" output.mp3
mono + mono → stereo
Use the join filter:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex join=inputs=2:channel_layout=stereo output.mp3
Or amerge:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex amerge=inputs=2 output.mp3
mono + mono → mono
Use the amix filter:
ffmpeg -i input0.mp3 -i input1.mp3 -filter_complex amix=inputs=2:duration=longest output.mp3
More info and examples
See FFmpeg Wiki: Audio Channels
Check this out:
ffmpeg -y -i ad_sound/whistle.mp3 -i ad_sound/4s.wav -filter_complex "[0:0][1:0] amix=inputs=2:duration=longest" -c:a libmp3lame ad_sound/outputnow.mp3
I think it will help.
The amix filter helps to mix multiple audio inputs into a single output.
If you run the following command:
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
This command will mix 3 input audio streams (I used two mp3 files, in the example below) into a single output with the same duration as the first input and a dropout transition time of 3 seconds.
The amix filter accepts the following parameters:
inputs:
The number of inputs. If unspecified, it defaults to 2.
duration:
How to determine the end-of-stream.
longest:
The duration of the longest input. (default)
shortest:
The duration of the shortest input.
first:
The duration of the first input.
dropout_transition:
The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
For example, I ran the following command in Ubuntu:
FFMPEG version: 3.2.1-1
UBUNTU 16.04.1
ffmpeg -i background.mp3 -i bSound.mp3 -filter_complex amix=inputs=2:duration=first:dropout_transition=0 -codec:a libmp3lame -q:a 0 OUTPUT.mp3
-codec:a libmp3lame -q:a 0 was used to set a variable bit rate. Remember that, you need to install the libmp3lame library, if is necessary. But, it will work even without the -codec:a libmp3lame -q:a 0 part.
Reference: https://ffmpeg.org/ffmpeg-filters.html#amix
For merging two audio files with different volumes and different duration following command will work:
ffmpeg -y -i audio1.mp3 -i audio2.mp3 -filter_complex "[0:0]volume=0.09[a];[1:0]volume=1.8[b];[a][b]amix=inputs=2:duration=longest" -c:a libmp3lame output.mp3
Here duration can be change to longest or to shortest, you can also change the volume levels according to your need.
If you're looking to add background music to some voice use the following command as in the gaps the music will become loud automatically:
ffmpeg -i bgmusic.mp3 -i audio.mp3 -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress=threshold=0.003:ratio=20[bg]; [bg][mix]amerge[final]" -map [final] final.mp3
In this threshold is something whose value will decide how much loud the audio should be, the less the threshold more the audio will be. Ratio gives how much the other audio should be compressed, the more the ratio the more the compression is.
If they are different length, you can use apad to add a silent sound to the shortest one
With Bash
set 'amovie=a.mp3 [gg]; amovie=b.mp3 [hh]; [gg][hh] amerge'
ffmpeg -f lavfi -i "$1" -q 0 c.mp3
Example
You can use the following command arguments:
// Command is here
let commandValue = "-y -i \(recordedAudioPath) -i \(backgroundAudio) -filter_complex [\(0):a][\(1):a]amerge=inputs=\(2)[a] -map [a] -ac \(2) -shortest -preset ultrafast \(outputPath)"
MobileFFmpeg.execute(commandValue)