I have a .mov file (codec = motion jpeg) that has an audio stream that includes small pulses at every second.
When I convert this file to mp4 using ffmpeg I notice that all my pulses are now off by one frame.
I simply used "ffmpeg -i source_file.mov target_file.mp4"
Here is an image of the comparison between the audio signals:
A1 is the original audio (.mov) and A2 is the mp4 output audio of ffmpeg.
As you can see the pulses are one frame late compared to the original.
I know that the h264 codec is lossy but one frame offset seems like a big loss if you ask me.
Is there any option I could use with ffmpeg to have a better audio stream ?
Here is the input file: https://www.dropbox.com/s/6y5g7lo5dvu0ub1/BBB_09_tree_trunk_009_ANIM_001.mov?dl=0
Here is the output file:
https://www.dropbox.com/s/10zuzwn0qs8l853/BBB_09_tree_trunk_009_ANIM_001.mp4?dl=0
If you copy the audio over, you shouldn't get the shift.
ffmpeg -i source_file.mov -c:a copy target_file.mp4
I've been working on this issue for my own needs and my file format has to be mp4. I'm working from mxf files. I've tried several options and found this to give the most accurate result (I've removed specifics for simplicity):
ffmpeg -ss 00:00:00.021 -i "input.mxf" -itsoffset -0.044 -i "input.mxf" -c:v libx264 -c:a aac -map 0:a -map 1:v "output.mp4"
Starting the first file at 21ms and mapping it as the audio, then shifting the video back 44ms gave gave me the most accurate sync (within several samples). I don't know why 22ms wasn't as accurate (when that's what the primer sample issue seems to equate to) and I found nothing that allowed me to work more granular, in samples. A filter with a PTS offset had no affect. Perhaps it works differently with different file formats. It's also worth noting that the same command without the -itsoffest gave the same sync result with one difference; the video stream duration was 1 frame and 1ms off the audio and container durations. With the -itsoffest, the durations were only 1ms different. You can use 22ms to achieve an accurate duration, but check your sync, it might be out that slightest bit more.
Also worth noting that I stumbled across some developer commentary on the -itsoffset tag which clarified that it doesn't work on audio, it works on video. It seems like the answer above is suggesting to map the offest against the audio, which apparently is not how the function is built to work. https://trac.ffmpeg.org/ticket/1349
try mpeg2 audio: -acodec mp2 it worked for me
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 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'm using ffmpeg to build a short hunk of video from a machine-generated png. This is working, but the video now needs to have a soundtrack (an [audio] field) for some of the other things I'm doing with it. I don't actually want any sound in the video, so: is there a way to get ffmpeg to simply set up an empty soundtrack property in the video, perhaps as part of the call that creates the video? I guess I could make an n-second long silent mp3 and bash it in, but is there a simpler / more direct way? Thanks!
Thanks to #Alvaro for the links; one of these worked after a bit of massaging. It does seem to be a two-step process: First make the soundtrack-less video and then do:
ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -channel_layout 2.1
-i /dev/zero -i in.mp4 -vcodec copy -acodec libfaac -shortest out.mp4
The silence comes from /dev/zero and -shortest makes the process stop at the end of the video. Argument order is significant here; -shortest needs to be down near the output file spec.
This assumes that your ffmpeg installation has libfaac installed, which it might not. But, otherwise, this seems to be working.
I guess you need to create a media file properly with audio and video stream. As far as i know, there is not a direct way.
If you know your video duration, first create the dummy audio and after when you create the video try to join the audio part.
In superuser, you can find more info link1 link2
I know that there are a million ways to download a video from youtube and then convert it to audio or do further processing on it. But recently I was surprised to see an app called YoutubeToMp3 on mac actually showing "Skipping X mb of video" and supposedly only downloading the audio from the video, without the need to use bandwith to download the entire video and then convert it. I was wondering if this is actually correct and possible at all because I cant find any way to do that. Do you have any ideas ?
EDIT:
After some tests here is some additional information on the topic. The video which I tried to get the audio from is just a sample mp4 file from the internet:
http://download.wavetlan.com/SVV/Media/HTTP/MP4/ConvertedFiles/MediaCoder/MediaCoder_test6_1m9s_XVID_VBR_306kbps_320x240_25fps_MPEG1Layer3_CBR_320kbps_Stereo_44100Hz.mp4
I tried
ffmpeg -i "input" out.mp3
ffmpeg -i "input" -vn out.mp3
ffmpeg -i “input” -vn -ac 2 -ar 44100 -ab 320k -f mp3 output.mp3
ffmpeg -i “input” -vn -acodec copy output.mp3
Unfortunately non of these commands seems to be using less bandwith. They all download the entire video. Now that you have the video can you confirm if there is actually a command that downloads only the audio stream from it and lowers the bandwith usage? Thanks!
After a lot of research I found out that this is not possible and developed an alternative approach:
Download the mp4 header
Parse the header and get the locations of the audio bytes
Download the audio bytes with http range requests and offsets
Assemble the audio bytes and wrap them in a simple ADTS container to produce a playing m4a file
That way only bandwidth for the audio bytes is used. If you find a better approach of doing it please let me know.
For a sample Android APP and implementation check out:
https://github.com/feribg/audiogetter/blob/master/audiogetter/src/main/java/com/github/feribg/audiogetter/tasks/download/VideoTask.java
FFmpeg is capable of accepting an URL as input. If the URL is seekable, then FFmpeg could theoretically skip all the video frames, and thus it would need to download only the data for the audio stream.
Try using
ffmpeg -i http://myvideo.avi out.mp3
and see if it takes less bandwidth.