I'm using ffmpeg to decode and encode signal. It works perfectly and I added filters. For example, I'm using such a command :
ffmpeg -re -i /home/dr_click/live.wav -af "anequalizer=c0 f=200 w=100 g=-5 t=0|c1 f=200 w=100 g=-5 t=0, anequalizer=c0 f=1000 w=100 g=3 t=0|c1 f=1000 w=100 g=3 t=0" -acodec pcm_s16be -ar 44100 -ac 2 -f rtp rtp://127.0.0.1:1234
I'm streaming my file, adding 2 filters with 200 Hz and 1000 Hz as central frequency and 100 Hz width and it works.
With such a filter, I know my gain will be -5db at 200Hz. But what is the gain for frequencies at 250 Hz ? Still -5db ? -4.5db ? -3db ? And same question at 350Hz or any other frequency.
What I'm looking for and didn't found is the way to get the frequency response of such a filter for a bandwith from 20Hz to 20kHz. In other words, what I'd like to know for any frequency is : gain = f (frequency) with a given ffmpeg filter
Thank you for your help,
Dr_Click
i'm working on a quite similar issue. Mine is to replace the system wide 15 band graphical LADSPA equalizer (mbeq_1197, controlled by JACK Rack) with an ffmpeg filter. As it is AFAIK impossible to adjust ffmpeg filter parameters during runtime, I have to rely on my already generated JACK EQ settings and need to transfer them to the ffmpeg EQ. Alas, I could not find any two "comparable" EQs: ffmpeg only offers a 18 band "superequalizer". My previous EQ has 15 bands, so I decided to do some interpolations and compare the frequency responses of the old and the new EQ.
Now to answer your question: I'm not an audio engineer, and I'm sure there are more professional ways. But what I found out for now is my current workflow:
Generate some white noise. In Linux you can e.g. use sox oder Audacity. In Audacity do Generate -> Built-in -> Noise... => White noise (1 min should be enough)
Save the file as WAV.
Apply your filter to this WAV: ffmpeg -i whitenoise.wav -af "<your filter>" whitenoise_filtered.wav
Load the filtered file into Audacity and do Analyze -> Plot Spectrum...
The output will be a little scattered because the white noise is not perfect, but this should be negligible.
Good luck!
Flittermice
Related
I have:
Video file of X length
Audio of Y length
I am trying to achieve an output video that has the following qualities:
The volume level of the added audio should be adjustable
The audio should loop till the end of the video
It should not break even if the input video does not have any audio
I should be able to mute the audio of the source video if needed.
All of the above, in the fastest possible way.
I'm not well versed with FFMPEG, maybe some experts could help.
since you are using a library i assume that you know how to run pure FFmpeg commands
based on your third condition we will divide the solution to two part :
It should not break even if the input video does not have any audio
in order to cover this condition, you can check if there is any audio stream in your video file before running any FFmpeg command with below code:
private boolean isVideoContainAudioStream(String videoPath) {
MediaMetadataRetriever retriever = new MediaMetadataRetriever();
retriever.setDataSource(videoPath);
String hasAudioStream = retriever.extractMetadata(MediaMetadataRetriever.METADATA_KEY_HAS_AUDIO);
if (hasAudioStream != null && hasAudioStream.equals("yes"))
return true;
else
return false;
}
1. Part One :
so if the result of above function is equal to true, your video file contain audio stream so you can run below command :
ffmpeg -i video.mp4 -filter_complex "amovie=/path/to/audio/file/audio.mp3:loop=0,asetpts=N/SR/TB,volume=2.0[audio];[0:a]volume=0.5[sa];[sa][audio]amix[fa]" -map 0:v -map [fa] -vcodec libx264 -preset ultrafast -shortest fout.mp4
in above command we take audio file at a specific path with amovie filter
loop=0, Loop audio infinitely
asetpts=N/SR/TB, Generate timestamps by counting samples
volume=2.0, multiply audio volume by 2.0
video's audio stream is accessible with [0:a] filter pad so we take it and set the volume to half of the input's volume and name it [sa] obviously if you want to mute the audio of the source video you change that part to :
[0:a]volume=0.0[sa]
after that we will mix two audio streams using amix filter and name it [fa], so far we have everything we wanted, and we just want to merge audio and video streams
-vcodec libx264, we are using x264 video encoding because it has lots of configs to gain better performance and speed
-shortest, since we loop audio infinitely, we tell the ffmpeg to continue creating frames until the shortest stream ends (video stream is the short one for sure)
-preset ultrafast, preset is one of the x264 options, ultrafast will give you more encoding speed at the cost of more size in output file, usually using veryfast value for this flag is a good combination of speed and size
2. Part Two :
if the isVideoContainAudioStream function return false (which means your input video is muted) you can run below command:
ffmpeg -i mute_video.mp4 -filter_complex "amovie=/path/to/audio/file/audio.mp3:loop=0,asetpts=N/SR/TB,volume=2.0[audio]" -map 0:v -map [audio] -vcodec libx264 -preset ultrafast -crf 18 -shortest m_fout.mp4
in above command we use another x264 options called CRF
Constant Rate Factor (CRF)
Use this rate control mode if you want to keep the best quality and care less about the file size. This is the recommended rate control mode for most uses.
The range of the CRF scale is 0–51, where 0 is lossless, 23 is the default, and 51 is worst quality possible. A lower value generally leads to higher quality, and a subjectively sane range is 17–28. Consider 17 or 18 to be visually lossless or nearly so; it should look the same or nearly the same as the input but it isn't technically lossless.
The range is exponential, so increasing the CRF value +6 results in roughly half the bitrate / file size, while -6 leads to roughly twice the bitrate.
Choose the highest CRF value that still provides an acceptable quality. If the output looks good, then try a higher value. If it looks bad, choose a lower value.
thats it, there is lots of option for x264 encoder, you can check all available options at this link:
H.264 Video Encoding Guide
This is my first time here on stack overflow asking question.
I am stuck and really struggling with this. I am trying to make some of my MXF video files to be EBU r128 standard for its audio.
This means that it has to be -23 and not higher than 0.5.
My current process
Watch_folder > Encoding to MXF > Output_folder
I need to makesure when its comes to output folder, those MXF files are EBU R128 Loudness compliant.
What I have done so Far:
FFMPEG:
ffmpeg -i input.mxf -af loudnorm=I=-23:LRA=7:tp=-2:print_format=json -f null -
got the result:
Input Integrated: -15.1 LUFS
Input True Peak: +0.0 dBTP
Input LRA: 17.1 LU
Input Threshold: -26.2 LUFS
Output Integrated: -17.1 LUFS
Output True Peak: -1.5 dBTP
Output LRA: 5.3 LU
Output Threshold: -27.6 LUFS
Normalization Type: Dynamic
Target Offset: +1.1 LU
then i did
ffmpeg -i input.mxf -af loudnorm=I=-23:LRA=7:tp=-2:measured_I=-15.1:measured_LRA=17.1:measured_tp=0:measured_thresh=-27.6:offset=1.1 -ar 48k -y output.mxf
However, when i put it through the software Eff, it says that its not EBU compliant.
*EDIT:
This also reduces the quality. for example; my 6 Gb becomes 250 MB and you can tell the quality downgraded
ffmpeg-normalize
I did the following
ffmpeg-normalize input.mxf -c:a pcm_s32le -ar 48000 -o output.mxf
but this gives me errors.
if i do it without the output file type, i get a mkv which will not work for me. i need it to be mxf.
OK, a few issues here.
Firstly, if your file is measured at -26.2 LUFS, you'd need to add 3.2 dB to get it to -23. But you can't do that, because your true peak is too high (you'd be over full scale). You'll need to compress (dynamic audio compression, not file/rate compression) the audio or use at least a limiter to achieve this.
A good R128 audio track should be mixed properly rather than just run through a normaliser, otherwise you risk it either failing the standard or unwanted audio effects.
If you don't have access to audio editing software or someone who can do this for you, then FFMPEG does include an audio limiter, which will give you enough headroom to raise the level to -23 LUFS.
You can do that with something like this:
-filter_complex alimiter=level_in=1:level_out=1:limit=1.5:attack=7:release=100:level=disabled
However, tuning a limiter well depends on what the video file is of (music, speech, etc) and it is something that's worth taking some time over. Alter the attack and release values until you get the result you want.
Secondly, the reason that FFMPEG has produced a smaller file of lower quality is because you didn't specify anything in the video section. FFMPEG's default action with video is (usually) to encode to h264, so whatever your codec here is (I am assuming DNxHD from the fact that you're using an MXF wrapper) needs to be specified. FFMPEG will copy the video stream though and leave it alone if you include the option -c:v copy (which means copy video codec, basically).
Post your results once you have tried these...!
I have a bunch of mkv files, with FLAC as the audio codec and FFV1 as the video one.
The files were created using an EasyCap aquisition dongle from a VCR analog source. Specifically, I used VLC's "open acquisition device" prompt and selected PAL. Then, I converted the files (audio PCM, video raw YUV) to (FLAC, FFV1) using
ffmpeg.exe -i input.avi -acodec flac -vcodec ffv1 -level 3 -threads 4 -coder 1 -context 1 -g 1 -slices 24 -slicecrc 1 output.mkv
Now, the files are progressively out of sync. It may be due to the fact that while (maybe) the video has a constant framerate, the FLAC track has variable framerate. So, is there a way to sync the track to audio, or something alike? Can FFmpeg do this? Thanks
EDIT
On Mulvya hint, I plotted the difference in sync at various times; the first column shows the seconds elapsed, the second shows the difference - in secs. The plot seems to behave linearly, with 0.0078 as a constant slope. NOTE: measurements taken by hands, by means of a chronometer
EDIT 2
Playing around with VirtualDub, I found that changing the framerate to 25 fps from the original 24.889 (Video->Frame rate...->Change frame rate to) and using the track converted to wav definitely does work. Two problems, though: VirtualDub crashes when importing the original FFV1-FLAC mkv file, so I had to convert the video to H264 to try it out; more, I find it difficult to use an external encoder to save VirtualDub output.
So, could I avoid using VirtualDub, and simply use ffmpeg for it? Here's the exported vdscript:
VirtualDub.audio.SetSource("E:\\4_track2.wav", "");
VirtualDub.audio.SetMode(0);
VirtualDub.audio.SetInterleave(1,500,1,0,0);
VirtualDub.audio.SetClipMode(1,1);
VirtualDub.audio.SetEditMode(1);
VirtualDub.audio.SetConversion(0,0,0,0,0);
VirtualDub.audio.SetVolume();
VirtualDub.audio.SetCompression();
VirtualDub.audio.EnableFilterGraph(0);
VirtualDub.video.SetInputFormat(0);
VirtualDub.video.SetOutputFormat(7);
VirtualDub.video.SetMode(3);
VirtualDub.video.SetSmartRendering(0);
VirtualDub.video.SetPreserveEmptyFrames(0);
VirtualDub.video.SetFrameRate2(25,1,1);
VirtualDub.video.SetIVTC(0, 0, 0, 0);
VirtualDub.video.SetCompression();
VirtualDub.video.filters.Clear();
VirtualDub.audio.filters.Clear();
The first line imports the wav-converted audio track.
Can I set an equivalent pipe in ffmpeg (possibly, using FLAC - not wav)? SetFrameRate2 is maybe the key, here.
I am trying to clean up a video that was recorded in 2003 in low-light conditions on what was possibly a cameraphone. The video has been cleaned up somewhat (cropped, logos removed and stabilized), but it remains quite jerky, due in large part to its low frame rate. What are some tricks that might clean up the video in this regard? I feel that I am asking for something a bit like tweening in flash animations, but for pixels, whereby additional frames are generated using nearby frames of the video. Does such a trick exist? Is there another way to approach this problem?
To reproduce the video processing so far, take the following steps:
# get video
wget http://www.anwarweb.net/saddamdown.wmv
# crop
ffmpeg -i saddamdown.wmv -filter:v "crop=292:221:14:10" -c:a copy saddamdown_crop.wmv
# remove logo 1
ffmpeg -i saddamdown_crop.wmv -vf delogo=x=17:y=77:w=8:h=54 -c:a copy saddamdown_crop_delogo_1.wmv
# remove logo 2
ffmpeg -i saddamdown_crop_delogo_1.wmv -vf delogo=x=190:y=174:w=54:h=8 -c:a copy saddamdown_crop_delogo_1_delogo_2.wmv
# stabilize
ffmpeg -i saddamdown_crop_delogo_1_delogo_2.wmv -vf deshake saddamdown_crop_delogo_1_delogo_2_deshake.wmv
Note: The video is of the Saddam Hussein execution.
You could try with slowmoVideo: https://github.com/slowmoVideo/slowmoVideo
It's an open source software to create smooth slow motion effects from pixel motion analysis (Windows, Linux, OSX with wine or crossover. Read and write with ffmpeg).
First calculate the slow down ratio: for example if the original video is 18fps and the desired output is 24fps, set the speed of slowmo to 75% (18/24=0.75).
The result depends a lot on the video content, obviously the more fixed are the shots the better.
Anyway you can tweak what they call "Optical Flow", that is the analysis part of the process.
Good luck ;)
I am trying to make an audio file be exactly x second.
So far i tried using the atempo filter by doing the following calculation
Audio length / desired length = atempo.
But this is not accurate, and I am having to tweak the tempo manually to get it to an exact fit.
Are there any other solutions to get this work ? Or am I doing this incorrectly?
My original file is a wav file, and my output in an mp3
Here is a sample command
ffmpeg -i input.wav -codec:a libmp3lame -filter:a "atempo=0.9992323" -b:a 320K output.mp3
UPDATE:
I was able to correctly calculate the tempo by changing the way I am receiving the audio length.
I am now calculating the current audio length using the actual file size and the sample rate.
Audio Length = file size / (sample rate * 2)
Sample rate is something like 16000 Hz. You can get that by using ffprob or ffmpeg.
You are calculating the tempo incorrectly.
Audio length / desired length = atempo
should be:
desired length / Audio length = atempo
This answer was posted as an edit to the question ffmpeg, stretch audio to x seconds by the OP Max Doumit under CC BY-SA 3.0.