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 ?
Related
I have problem with live http flv stream. I'm using the following command and it works great except for the audio. Audio has delay that increasing over time. I can fix the this by removing refresh rate option on the output (-r 30) but then stream latency goes higher for about half second.
ffmpeg -f v4l2 -threads 0 -video_size 672X420 -i /dev/video1 -f alsa -thread_queue_size 512 -i hw:1,0 -c:a aac -ar 44100 -b:a 128k -c:v libx264 -s 672x420 -r 30 -g 60 -preset superfast -tune zerolatency -strict -2 -f flv rtmp://localhost/live/primary
Can someone explain why this happening and what possible fixes are?
On the client I'm using Chrome with flv.js library.
On the server Node-Media-Server.
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 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
I am trying to forward HLS stream to server via RTSP. On the server, it will be transcoded to the multiple bitrate streams so a wide range of users can watch stream.
To achieve this, I am using ffmpeg.
Should I stream to my server in best quality because my server will do transcoding to lower bitrate streams?
I am using this command:
ffmpeg -i http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8 -c:v libx264 -preset veryfast -maxrate 2000k -bufsize 2000k -g 60 -c:a aac -b:a 96k -ac 2 -ar 44100 -f rtsp -muxdelay 0.1 rtsp://username:password#95.138.139.123:1935/live/myStream
But my stream is pixelated on some moments like on the picture:
How can I fix that?
Is there some better method to achieve this?
Here is log file from ffmpeg.
Log file
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