ffmpeg怎样将局域网的onvif转成hls

[sycru]

我已经在debian中搭好了ffmpeg,ng,nginx-rtmp-module,也能用ffmpeg将mp4成功送到rtmp。
ffmpeg -re -i /tmp/02.mp4 -vcodec copy -acodec copy -f flv rtmp://127.0.0.1/hls/mystream

但我怎样才能用ffmpeg将rtsp://192.168.0.3:554/转成hls直播呢？
ffmpeg -i "rtsp://192.168.0.3:554/onvif1" -f flv rtmp://127.0.0.1/hls/mystream
总是失败

 

[sycru]

ffmpeg -i "rtsp://192.168.0.3:554/onvif" -f mp4 -vcodec copy -r 25 -s -b:v 2048000 -an -f flv "rtmp://127.0.0.1/hls/mystream"
出错
[rtsp @ 0x564204386b80] UDP timeout, retrying with TCP
ctrl+z退出错误显示

[rtsp @ 0x564204386b80] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://192.168.0.3:554/onvif1':
Metadata:
title : H.264 Video, RtspServer_0.0.0.2
Duration: N/A, start: 0.000000, bitrate: 64 kb/s
Stream #0:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
File '2048000' already exists. Overwrite ? [y/N] y
Output #1, flv, to 'rtmp://127.0.0.1/hls/mystream':
Output file #1 does not contain any stream
Exiting normally, received signal 2.

 

[sycru]

ffmpeg -i "rtsp://192.168.0.3:554/onvif1" -f flv rtmp://127.0.0.1/hls/mystream

[rtsp @ 0x55669067b9c0] UDP timeout, retrying with TCP

[rtsp @ 0x55669067b9c0] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://192.168.0.3:554/onvif1':
Metadata:
title : H.264 Video, RtspServer_0.0.0.2
Duration: N/A, start: 0.000000, bitrate: 64 kb/s
Stream #0:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
[rtmp @ 0x5566906cf800] Cannot open connection tcp://127.0.0.1:1935
rtmp://127.0.0.1/hls/mystream: Immediate exit requested
Exiting normally, received signal 2.

 

[sycru]

ffmpeg -i "rtsp://192.168.0.3:554/onvif2" -f flv "rtmp://127.0.0.1:1935/hls/playlist"
[rtsp @ 0x561151cd2be0] UDP timeout, retrying with TCP
ctl+z退出没有其它错误

 

[sycru]

ffmpeg -i /02.mp4 -strict -2 -c:v libx264 -c:a aac -f hls /usr/local/nginx/html/hls/playlist.m3u8
这个能播放
http://192.168.48.11/hls/playlist.m3u8

 

[Admin]

指定视频编解码器试一下：

ffmpeg -f lavfi -i anullsrc -rtsp_transport udp -i rtsp://192.168.1.3:554/onvif1 -tune zerolatency -vcodec libx264 -t 12:00:00 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp://a.rtmp.youtube.com/live2/stream name

 

[sycru]

指定视频编解码器试一下：

ffmpeg -f lavfi -i anullsrc -rtsp_transport udp -i rtsp://192.168.1.3:554/onvif1 -tune zerolatency -vcodec libx264 -t 12:00:00 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp://a.rtmp.youtube.com/live2/stream name

ffmpeg -f lavfi -i anullsrc -rtsp_transport udp -i rtsp://192.168.0.3:554/onvif1 -tune zerolatency -vcodec libx264 -t 12:00:00 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp://127.0.0.1:1935/hls/playlist

能产生文件，但我用vlc播放不了
http://192.168.48.11/hls/playlist.m3u8
rtmp://192.168.48.11/hls/playlist.m3u8
rtmp://192.168.48.11:1935/hls/playlist.m3u8

都是黑屏幕播不了

 

[sycru]

指定视频编解码器试一下：

ffmpeg -f lavfi -i anullsrc -rtsp_transport udp -i rtsp://192.168.1.3:554/onvif1 -tune zerolatency -vcodec libx264 -t 12:00:00 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp://a.rtmp.youtube.com/live2/stream name

[rtsp @ 0x555becd719a0] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Guessed Channel Layout for Input Stream #1.1 : mono
Input #1, rtsp, from 'rtsp://192.168.0.3:554/onvif1':
Metadata:
title : H.264 Video, RtspServer_0.0.0.2
Duration: N/A, start: 0.000000, bitrate: 64 kb/s
Stream #1:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
Stream #1:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Codec AVOption tune (Tune the encoding to a specific scenario) specified for output file #0 (rtmp://127.0.0.1:1935/hls/play2) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.