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
Related
I create a webrtc peer connection with my server(only stun)
Using pion webrtc for the server
I write the received RTP packets as VP8 and opus streams, as described here, to two pipes (the writers; created with os.Pipe() in golang)
The read ends of these two pipes are received by ffmpeg as inputs (via exec.Command.ExtraFiles) for transcoding using libx264 and aac into a single stream. The command:
ffmpeg -re -i pipe:3 -re -r pipe:4 -c:a aac -af aresample=48000 -c:v libx264 -x264-params keyint=48:min-keyint=24 -profile:v main -preset ultrafast -tune zerolatency -crf 20 -fflags genpts -avoid_negative_ts make_zero -vsync vfr -map 0:0,0:0 -map 1:0,0:0 -f matroska -strict -2 pipe:5
The above command outputs to a pipe(:5) the read end of which is being taken as input by the following:
ffmpeg -hide_banner -y -re -i pipe:3 -sn -vf scale=-1:'min(ih,360)' -c:v libx264 -pix_fmt yuv420p -ca aac -b:a 128k -b:v 1400k -maxrate 1498k -bufsize 2100k -hls_time 1 -hls_playlist_type event -hls_base_url /workdir/streamID/360p -hls_segment_filename /workdir/streamID/360p/360_%%03d.ts -f hls /workdir/streamID/360p.m3u8
This works fine as long as there are no movements of my webcam. The moment that happens the video speed suddenly increases for a split second and audio delay gets introduced. This delay keeps increasing each time I shake my webcam.
The first command in point 4 above - if written to a file separately will be absolutely fine, in terms of a/v sync, even with vigorous camera shaking. The weird audio delay is only when transcoding for hls output irrespective of whether I'm actually viewing it live or playing it back later.
This is my first time working with ffmpeg/hls/webrtc - would be really helpful if I could be pointed in the correct direction at least to be able to debug this or even know why this happens. Any and all help is greatly appreciated
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 tried converting mp4 video to HLS for online streaming which I have successfully done using FFmpeg.
Command:
ffmpeg -i /var/www/html/file_conversion/heli.mp4 -map 0 -profile:v baseline -level 3.0 -s 640x360 -c:v libx264 -b:v 500k -c:a libfdk_aac -b:a 320k -hls_list_size 0 -start_number 0 -hls_init_time 0 -hls_time 2 -f hls /var/www/html/file_conversion/hlstest2/heli.m3u8
But now I am trying to convert the same video with multiple bitrates for dynamic adaptive streaming.
Any idea how can I achieve this?
I was having the same doubt, and found this article:
https://dev.to/nodir_dev/transcode-video-source-to-hls-playlist-format-for-video-on-demand-vod-streaming-3h99
After making transcoding files, just create a *.m3u8 file, with the following content:
#EXTM3U
#EXT-X-STREAM-INF:BANDWIDTH=375000,RESOLUTION=640x360
360_out.m3u8
#EXT-X-STREAM-INF:BANDWIDTH=750000,RESOLUTION=854x480
480_out.m3u8
#EXT-X-STREAM-INF:BANDWIDTH=2000000,RESOLUTION=1280x720
720_out.m3u8
#EXT-X-STREAM-INF:BANDWIDTH=3500000,RESOLUTION=1920x1080
1080_out.m3u8
now just send this file to the streams.
My HLS folder:
image_folder
content_of_master_file
I made the transcoding two resolutions: 1280x720 and 640x480
- Author of the article: Nodirbek Sharipov
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 have setup a HLS live stream on AWS. Significant part of the video plays as expected but in some portions audio is lost while the video continues to play. Please share you thoughts/insights if you have faced such an issue.
I transcode video to h264, baseline3.1 and audio to aac stereo 2 channels before segmenting to 10s files. Following commands are being used:
Transcoding:
ffmpeg -y -i ${file_path} -b:v ${vBitrate} -minrate ${vBitrate} -maxrate ${vBitrate} -c:v libx264 -b:a ${aBitrate} -c:a libfaac -ac 2 -r 25 -s ${resolution} -profile:v baseline -level 3.1 ${output_dir}/${file_name}.ts
Ex: vBitrate: 1m, aBirate: 128k, resolution: 960x540
Splittting:
ffmpeg -i ${file_path} -c:v copy -c:a copy -flags -global_header -map 0 -f segment -segment_time ${segment} -segment_list ${output_dir}/playlist.m3u8 -segment_format mpegts tmp0/${file_name}_%02d.ts
Ex: segment: 10
Setup: segments are being served from Nginx and playlist is managed by a Python/Flask app.