ffmpeg lavfi distorted audio - audio

How to play audio at a normal rate?
First 2s play fine, then audio becomes distorted, it seems to speed up.
ffmpeg -f image2 -loop 1 -framerate 25 -i example.jpg -lavfi "amovie=sample.aac:loop=0" -f flv rtmp://192.168.99.100:1935/live/mystream2
Repeating audio with loop:0 returns `[flv # 0000000000708220] Non-monotonous DTS in output stream 0:0;
audio: http://cdn.online-convert.com/example-file/audio/example.aac
Thank you.

The amovie filter does not alter timestamps to make them continuous, upon loop. A setpts filter is needed.
ffmpeg -f image2 -loop 1 -framerate 25 -i example.jpg -f lavfi -i "amovie=sample.aac:loop=0,asetpts=N/SR/TB" -f flv rtmp://192.168.99.100:1935/live/mystream2

Related

How can I stream an image to YouTube using FFmpeg?

I have an image that changes every few seconds. How can I stream it to YouTube using FFmpeg?
ffmpeg -f image2 -loop 1 -i input.jpg -re -f lavfi -i anullsrc -vf format=yuv420p -c:v libx264 -b:v 2000k -maxrate 2000k -bufsize 4000k -g 50 -c:a aac -f flv rtmp://output
-f image2 is needed to manually select the image demuxer. Otherwise, depending on the input format, it may choose a different demuxer that does not allow arbitrary replacing of input.jpg.
Replace input.jpg atomically (such as with mv but not cp) or else it may fail.
YouTube requires audio. The anullsrc filter will generate silent audio.
See FFmpeg Wiki: YouTube Streaming.
(Optional) Use the slowest -preset that provides 25 fps (or whatever frame rate you set using the -framerate image demuxer input option).

How do I convert wav into an mxf file with timecode?

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

How to delay audio after a specific position with ffmpeg?

I have a 10 seconds a.mp4 with two streams: Stream #0 is a video stream and Stream #1 is a audio stream.
Now, I want to delay the audio stream by 4 seconds after the time position 00:03. It is to say, in the output file, I want that: 00:00-00:03 is the original audio, 00:03-00:07 has no sound, 00:07-00:14 is the original 00:03-00:10 audio.
I've tried this:
ffmpeg -i a.mp4 -t 00:00:03 -i a.map4 -itsoffset 4 -ss 00:00:03 -i a.mp4 -map 0:v -map 1:a -map 2:a -codec copy output.mp4
But it seems that there are two audio streams in the output.mp4 and only one of them can be played once. Then I tried amix filter:
ffmpeg -i a.mp4 -t 00:00:03 -i a.mp4 -itsoffset 4 -ss 00:00:03 -i a.mp4 -filter_complex "[1:a][2:a] amix=inputs=2" -map 0:v output.mp4
But it also doesn't work. I'm new to ffmpeg so I have no idea what should I do now? Any idea for me? Very much thanks!
Use the asetpts filter to change timestamps, and aresample to (optionally) insert silence in that gap.
ffmpeg -i a.mp4 -af "asetpts='if(lt(T\,3),PTS,PTS+4/TB)',aresample=async=1" -c:v copy output.mp4
Test without aresample to see if your player is tolerant of large gaps in the audio stream.

Concat mp4 videos and merge their audios to the final output

I have several videos and photos and need to merge them with the cross-dissolve effect. The algorithm is next:
Create videos from images and add silent audio to them (so they will also have a sound stream):
ffmpeg -y -f lavfi -i anullsrc -loop 1 -i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/ea5c93fd-d946-4742-b8f7-ea9ae4d43441.jpg -c:v libx264 -t 10 -pix_fmt yuv420p -vf scale=750:1280 /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/ea5c93fd-d946-4742-b8f7-ea9ae4d43441.mp4
Combine all the videos and audios into one using this command:
ffmpeg
-i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/temp_68d437c0-f5e2-4651-b07e-91533480b6ef.mp4
-i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/temp_48f3c111-610d-40c7-ac71-6ce2fbb16184.mp4
-i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/temp_1593b5d8-7e16-417d-9372-2267581cd504.mp4
-i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/temp_1ac7f6be-1b12-4e31-b904-1491cc9b9494.mp4
-i /tmp/media/import-2020-Aug-19-Wednesday-05-40-34/temp_ea5c93fd-d946-4742-b8f7-ea9ae4d43441.mp4
-filter_complex
"[0:v]trim=start=0:end=8.032,setpts=PTS-STARTPTS[clip0];
[1:v]trim=start=2:end=13.047,setpts=PTS-STARTPTS[clip1];
[2:v]trim=start=2:end=13.558,setpts=PTS-STARTPTS[clip2];
[3:v]trim=start=2:end=13.186,setpts=PTS-STARTPTS[clip3];
[4:v]trim=start=2,setpts=PTS-STARTPTS[clip4];
[0:v]trim=start=9.032:end=10.032,setpts=PTS-STARTPTS[out0];
[1:v]trim=start=14.047:end=15.047,setpts=PTS-STARTPTS[out1];
[2:v]trim=start=14.558:end=15.558,setpts=PTS-STARTPTS[out2];
[3:v]trim=start=14.186:end=15.186,setpts=PTS-STARTPTS[out3];
[1:v]trim=start=0:end=2,setpts=PTS-STARTPTS[in1];
[2:v]trim=start=0:end=2,setpts=PTS-STARTPTS[in2];
[3:v]trim=start=0:end=2,setpts=PTS-STARTPTS[in3];
[4:v]trim=start=0:end=2,setpts=PTS-STARTPTS[in4];
[in1]format=pix_fmts=yuva420p,fade=t=in:st=0:d=2:alpha=1[fadein1];
[in2]format=pix_fmts=yuva420p,fade=t=in:st=0:d=2:alpha=1[fadein2];
[in3]format=pix_fmts=yuva420p,fade=t=in:st=0:d=2:alpha=1[fadein3];
[in4]format=pix_fmts=yuva420p,fade=t=in:st=0:d=2:alpha=1[fadein4];
[out0]format=pix_fmts=yuva420p,fade=t=out:st=0:d=2:alpha=1[fadeout0];
[out1]format=pix_fmts=yuva420p,fade=t=out:st=0:d=2:alpha=1[fadeout1];
[out2]format=pix_fmts=yuva420p,fade=t=out:st=0:d=2:alpha=1[fadeout2];
[out3]format=pix_fmts=yuva420p,fade=t=out:st=0:d=2:alpha=1[fadeout3];
[fadein1]fifo[fadein1fifo];
[fadein2]fifo[fadein2fifo];
[fadein3]fifo[fadein3fifo];
[fadein4]fifo[fadein4fifo];
[fadeout0]fifo[fadeout0fifo];
[fadeout1]fifo[fadeout1fifo];
[fadeout2]fifo[fadeout2fifo];
[fadeout3]fifo[fadeout3fifo];
[fadeout0fifo][fadein1fifo]overlay[crossfade0];
[fadeout1fifo][fadein2fifo]overlay[crossfade1];
[fadeout2fifo][fadein3fifo]overlay[crossfade2];
[fadeout3fifo][fadein4fifo]overlay[crossfade3];
[clip0][crossfade0][clip1][crossfade1][clip2][crossfade2][clip3][crossfade3][clip4]concat=n=9[output];
[0:a][1:a]acrossfade=d=10:c1=tri:c2=tri[A1];
[A1][2:a]acrossfade=d=10:c1=tri:c2=tri[A2];
[A2][3:a]acrossfade=d=10:c1=tri:c2=tri[A3];
[A3][4:a]acrossfade=d=10:c1=tri:c2=tri[audio] "
-vsync 0 -map "[output]" -map "[audio]" /tmp/media/final/some_filename_d0d2aab0-792a-4540-b2d3-e64abe98bf5c.mp4
And all works pretty well, but if I have, for example:
picture
video
video
picture
Then the sound from the second video is mapping to the first picture and sound from the third video to second video. And the third video actually goes without sound.
It seems like it's happening because the silent sound of the first picture is pretty short. An I right?
If so, how can I increase its duration?
I would much appreciate any help with this!
Assuming 5 inputs of 10 seconds each, all with audio streams*, with ffmpeg 4.3 or newer, use the xfade and acrossfade filters.
ffmpeg
-i in1.mp4
-i in2.mp4
-i in3.mp4
-i in4.mp4
-i in5.mp4
-filter_complex
" [0][1]xfade=transition=fade:duration=2:offset=8[V01];
[V01][2]xfade=transition=fade:duration=2:offset=16[V02];
[V02][3]xfade=transition=fade:duration=2:offset=24[V03];
[V03][4]xfade=transition=fade:duration=2:offset=32[video];
[0:a][1:a]acrossfade=d=2:c1=tri:c2=tri[A01];
[A01][2:a]acrossfade=d=2:c1=tri:c2=tri[A02];
[A02][3:a]acrossfade=d=2:c1=tri:c2=tri[A03];
[A03][4:a]acrossfade=d=2:c1=tri:c2=tri[audio]"
-vsync 0 -map "[video]" -map "[audio]" out.mp4
*if there's no existing audio stream, add one using the command in step 1.
If the existing audio stream of a file isn't 10 seconds long, use these filters on it before acrossfade.
[input]aresample=async=1:first_pts=0,apad,atrim=0:10[filtered]
and then use this filtered stream as input.

ffmpeg to calculate audio/visual difference between compressed and non-compressed video

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.

Resources