How can I change the Tone frequency.
This Example only pitches it by keeping the old tone frequency and only decrease the length of File.
For Example, I have a constant 100 Herz tone (as mp3) and I want it to change 90 Herz
ffmpeg -i 100h.mp3 -af atempo=100/90 90h.mp3
This Example doesn't work for me, it sounds the same
inputfile Mp3
outputfile Mp3
finally, by combining the asetrate and resample from Gyan, with atempo, the following works and preserves also the audio length
for example: use 0.9 for 90% of the frequenz
ffmpeg -i test.mp3 -af asetrate=44100*0.9,aresample=44100,atempo=1/0.9 output.mp3
Basic method is
ffmpeg -i 100h.mp3 -af asetrate=44100*0.9,aresample=44100 90h.mp3
where 44100 should be replaced with the input sample rate.
Related
I am processing my video(640 X 1280 dimensions). I want to divide my video horizontally into 2 separate videos(each video will now be 640 X 640 in dimensions),then combine them horizontally (video dimension will be now 1280 X 640)in a single video. I did the research on the internet and my issue was solved and not solved at the same time
I made a batch file and add these commands in it:-
ffmpeg -i input.mp4 -filter_complex "[0]crop=iw:ih/2:0:0[top];[0]crop=iw:ih/2:0:oh[bottom]" -map "[top]" top.mp4 -map "[bottom]" bottom.mp4
ffmpeg -i top.mp4 -i bottom.mp4 -filter_complex hstack output.mp4
Yes,my task got solved but many other issues also came out of it:-
1.) My output video has NO audio in it. No idea why there is no audio in the end results
2.) My main video file (on which I am doing all this) is 258 MB in size. But the result was only 38 MB in size. No idea what is happening? And even worse,I closely looked at the video,results were pretty same (only animation were not as smooth in output file as compared to input file)
3.) It is taking too much time(I know that computing takes some time but maybe there may be some way/sacrifice to make the process much quicker)
Thanks in advance for helping me
Combine your two commands
ffmpeg -i input.mp4 -filter_complex "[0]crop=iw:ih/2:0:0[top];[0]crop=iw:ih/2:0:oh[bottom];[top][bottom]hstack" -preset fast -c:a copy output.mp4
If you need it to encode faster then use a faster -preset as shown in FFmpeg Wiki: H.264.
x264 is a better encoder than your phone so it is not surprising that the file size is smaller.
Or use your player to do it
No need to wait for encoding. Just have your player do everything upon playback. This does not output a file, but only plays the re-arranged video. Example using mpv:
mpv --lavfi-complex="[vid1]split[v0][v1];[v0]crop=iw:ih/2:0:0[c0];[v1]crop=iw:ih/2:0:oh[c1];[c0][c1]hstack[vo]" input.mp4
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
I have two videos of different lengths Video one: 12 minutes Video two: 6 minutes I want to take audio of video one I want to take image of video two And put them together Output video length = 6 minutes. Use ffmpeg one command please help me - thanks ( watch image )____
Use
ffmpeg -i 12m.mp4 -i 6m.mp4 -vf setpts=(PTS-STARTPTS)/1.1 -af atempo=1.1 -map 1:v -map 0:a -shortest new.mp4
The setpts filter alters the video frame timestamps to 1/1.1 of their present value. FFmpeg will drop frames in the cadence needed to preserve source framerate.
The atempo filter speeds up the audio to 1.1 times the original speed.
-map 1:v -map 0:a tells ffmpeg to include the video stream from the 2nd input (6m.mp4) and the audio from the first input.
-shortest tells ffmpeg to conclude conversion when the shorter (of the audio and video) stream ends.
I have found that MP3's encoded with variable bit rate cause the currentTime property to be reported incorrectly, especially when scrubbing. That has wreaked havok on my app and has been a nightmare to debug.
I believe I need to convert all my MP3's to constant bitrate. Can FFMPEG (or something else) help me do that efficiently?
Props to Terrill Thompson for attempting to pin this down*
I also had issues with HTML5 being inaccurate for large mp3s. Since quality was not a big issue for my audio, I converted to constant bit rate of 8kbps, sample rate 8k, mono and it solved my issues.
You can convert to a contant bit rate for a few files using Audacity (export > save to mp3 > constant bit rate).
Or, using FFMPEG:
ffmpeg -i input.wav -codec:a libmp3lame -b:a 8k output.mp3
If you also want to reduce to mono and a 8k sample rate:
ffmpeg -i input.wav -codec:a libmp3lame -b:a 8k -ac 1 -ar 8000 output.mp3
Using the second compressed an hour of audio to under 5MB.
Something else is going on. currentTime should not be influenced by the fact that you are using variable-bit rate MP3s.
Perhaps the context sampleRate is not the same as the sample rate as the MP3s? That will mess up timing of the audio samples because WebAudio will resample the MP3s to the context sample rate.
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.