I have actually encoded an audio file from ac3 to aac using ffmpeg native aac encoder but the issue is that the file is not playing correctly , more specifically i have played that file in different media player but most of them start from 19 seconds and in vlc it is not even starting till I seek to more than 19 seconds duration.
command i have used is :-
ffmpeg -i source.mkv -map 0:a:0 -c:a aac audio.mp4.
That is the proper way.
Don't know if this will make a difference, try -b:a 400k and -strict experimental.
If you want audio only, convert to m4a or aac.
ffmpeg -i input.mkv -y -c:a aac -b:a 400k -map 0:a:0? -strict experimental output.mp4
Other encoders, may require compiling ffmpeg with use flags:
http://trac.ffmpeg.org/wiki/Encode/AAC
libfdk_aac
libfaac
Related
I am trying to get ffmpeg to work as expected however I am having all kinds of trouble getting it to work.
I need to output a webm and h264 for web play. However, the command I am using, while it used to work a few years ago, does not work at all now.
Both my webm and h264 do not have audio, and neither will play in any browser.
My command for webm is:
ffmpeg -y -i "$KMVAR_File" -c:v libvpx -crf 24 -b:v 1000k -vf scale=720:-2 -c:a libvorbis "$KMVAR_webmPath"
and my command for mp4 is:
ffmpeg -y -i "$KMVAR_File" -c:v libx264 -pix_fmt yuv420p -profile:v baseline -level 3.0 -crf 32 -b:v 1M -minrate 1M -maxrate 1M -bufsize 2M -vf scale=720:-2 -c:a aac -strict experimental -movflags +faststart "$KMVAR_mp4Path"
When playing with multiple audio, downmixing or extracting, there's no "one size fit all" solution with ffmpeg.
Look at https://trac.ffmpeg.org/wiki/AudioChannelManipulation as it provides multiple possible solution to your problem.
(I usually go with the pan filter : not the easiest to use, but more powerful than the map_channel approach)
I use following encoding option.
ffmpeg -i input.wmv -movflags faststart -c:v libx264 -profile:v baseline -acodec aac -ac 2 -ar 48000 -strict -2 /root/output.mp4;
But sometimes last 5~30 seconds audio is erased.
Total video duration is about 3 mins.
what do you think the problem is?
Is this related to computer performance? I use quadcore, 4G ram.
I do use NginxRTMP to generate HLS.
Problem is that HLS has not sound (since it needs AAC)
I try to trancode my RTMP sound to AAC sound with FFMPEG
exec_push ffmpeg -i rtmp://localhost/src/$name -vcodec libx264 -threads 0 -vprofile baseline -preset ultrafast -s 800x600 -acodec libfaac -ar 44100 -f flv rtmp://localhost/hls/$name
Problem: it takes 25% of CPU per stream.
Any idea on how to optimize that ?
I'm looking for a way to add an audio watermark, on specific time, to a video file (with existing audio) . something like: ffmpeg -i mainAVfile.mov -i audioWM.wav -filter_complex "[0:a][1:a] amix=inputs=2:enable='between(t,9,10)' [aud]; [0:v][aud]" -c:v libx264 -vf "scale=1280:720:sws_dither=ed:flags=lanczos, setdar=16:9" -c:a libfdk_aac -ac 2 -ab 96k -ar 48000 -af "aformat=channel_layouts=stereo, aresample=async=1000" -threads 0 -y output.mp4
The above command gives me this error Timeline ('enable' option) not supported with filter 'amix'. amerge didn't work as well. I kind of get lost with filter_complex syntax, specifically with the following conditions
On the main AV file, both audio and video tracks are filtered
Watermark should be between the 9th and 10th second (I already
generated a 1 second, 10k tone file)
The watermark need to survive the proceeding audio transcode
Use
ffmpeg -i mainAVfile.mov -i audioWM.wav
-filter_complex
"[0:a]aformat=channel_layouts=stereo,aresample=async=1000[main];
[1:a]atrim=0:1,adelay=9000|9000[wm];[main][wm]amix=inputs=2"
-vf "scale=1280:720:sws_dither=ed:flags=lanczos,setdar=16:9" -c:v libx264
-c:a libfdk_aac -ac 2 -ar 48000 -b:a 96k
-threads 0 -y output.mp4
It's preferable to perform all filtering in a single filtergraph. But I've kept the video filter as-is.
The specs for the video format are the following:
Aspect Ratio: 1:1
H.264 video compression, high profile, square pixels, fixed frame rate, progressive scan
.mp4 container with leading mov atom, no edit lists
Audio: Stereo AAC audio compression, 128kbps +
Reading through posts and ffmpeg documentation I came up with the following (yeah, I run it on a Windows PC):
ffmpeg.exe -r 30 -i input.webm -vf scale=iw*sar:ih -c:v libx264 -preset slow -profile:v high -c:a aac -strict experimental -ar 44100 -aspect 1:1 output.mp4
But when the video is played within the app that asks for this specification, it only displays black moving pixels, all broken, but you an hear the audio.
I don't really know what else to change on the command, and I have no idea in regards to the ...with leading mov atom specification.
Thanks.
EDIT:
I've tried #Mulvya's answer:
ffmpeg.exe -i input.webm -vf scale=iw*sar:ih,setsar=1 -c:v libx264 -preset slow -profile:v high -pix_fmt yuv420p -r 30 -c:a aac -strict experimental -ar 44100 -ac 2 -b:a 128k -movflags +faststart output.mp4
But the effect is the same once given to the app:
This is the information that ffmpeg spews about the input.webm file:
Use
ffmpeg.exe -i input.webm -vf scale=iw*sar:ih,setsar=1 -c:v libx264 -preset slow -profile:v high -pix_fmt yuv420p -r 30 -c:a aac -strict experimental -ar 44100 -ac 2 -b:a 128k -movflags +faststart output.mp4
Depending on how strict the app is, you may need to check the precise framerate. Use -r 30000/1001 for 29.97. The -movflags +faststart moves the moov atom to the front of the file.
Based on info I found elsewhere, this seems to be what Instagram requires:
ffmpeg.exe -i input.webm -vf scale=640:640,setsar=1 -c:v libx264 -preset slow -profile:v main -level 3.1 -pix_fmt yuv420p -r 30000/1001 -c:a aac -strict experimental -ar 44100 -ac 1 -b:a 64k -t 15 -movflags +faststart output.mp4