nginx -rtmp多码率，动态码率二级m3u8适应

最新推荐文章于 2024-01-30 15:20:33 发布

haoxiangnia111

最新推荐文章于 2024-01-30 15:20:33 发布

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分类专栏： rtmp

本文链接：https://blog.csdn.net/haoxiangnia111/article/details/115357481

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视频看这里

此处是youtube的播放链接，需要科学上网。喜欢我的分享，记得订阅我的频道，打开旁边的小铃铛，谢谢支持。

前言
之前的一篇博文 https://xugaoxiang.com/2020/01/19/build-video-server-using-nginx-rtmp/已经简单的介绍了如何利用nginx、nginx-rtmp-module和ffmpeg实现基于rtmp协议的直播。今天这篇继续直播这个话题,聊聊hls的应用。

HLS
HLS(Http Live Streaming)是由Apple公司定义的用于实时流传输的协议，HLS基于HTTP协议实现，传输内容包括两部分，一是M3U8描述文件，二是TS媒体文件。

m3u8文件
#EXTM3U
#EXT-X-VERSION:3
#EXT-X-MEDIA-SEQUENCE:6119
#EXT-X-TARGETDURATION:14
#EXTINF:10.625,
6119.ts
#EXTINF:13.667,
6120.ts
#EXTINF:10.000,
6121.ts
如上,m3u8文件是一个描述文件,必须以#EXTM3U开头,之后是切片TS文件的序列.对于直播来讲,m3u8文件需要进行实时的更新,只保留若干个TS切片序列,防止本地存储撑爆硬盘.

多码率支持
针对应用网络多变及不稳定的情况，多数直播都会提供多码率支持，播放器会根据用户当前的网络状况，自动切换到对应的码率上，大大提升用户体验。在服务器端，为了提供多码率的支持，就需要多级m3u8文件。在主m3u8文件不再有TS序列，而是二级m3u8文件,如下所示

#EXTM3U
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1280000
low.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=2560000
mid.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=7680000
hi.m3u8
nginx-rtmp对HLS的支持
nginx-rtmp-module本身对rtmp和hls都有很好的支持,只需要在nginx.conf配置下就ok了

#user nobody;
worker_processes auto;

rtmp_auto_push on;

error_log logs/error.log;
error_log logs/error.log notice;
error_log logs/error.log info;

#pid logs/nginx.pid;

events {
worker_connections 1024;
}

rtmp {

server {

    listen 1935;

    chunk_size 4000;

    # TV mode: one publisher, many subscribers
    #application mytv {

        # enable live streaming
        #live on;

        # record first 1K of stream
        #record all;
        #record_path /tmp/av;
        #record_max_size 1K;

        # append current timestamp to each flv
        #record_unique on;

        # publish only from localhost
        #allow publish 127.0.0.1;
        #deny publish all;

        #allow play all;
    #}

    # Transcoding (ffmpeg needed)
    #application big {
    #    live on;

        # On every pusblished stream run this command (ffmpeg)
        # with substitutions: $app/${app}, $name/${name} for application & stream name.
        #
        # This ffmpeg call receives stream from this application &
        # reduces the resolution down to 32x32. The stream is the published to
        # 'small' application (see below) under the same name.
        #
        # ffmpeg can do anything with the stream like video/audio
        # transcoding, resizing, altering container/codec params etc
        #
        # Multiple exec lines can be specified.

    #    exec ffmpeg -re -i rtmp://localhost:1935/$app/$name -vcodec flv -acodec copy -s 32x32
                    #-f flv rtmp://localhost:1935/small/${name};
    #}

    #application small {
    #    live on;
    #    # Video with reduced resolution comes here from ffmpeg
    #}

    #application webcam {
    #    live on;

        # Stream from local webcam
    #    exec_static ffmpeg -f video4linux2 -i /dev/video0 -c:v libx264 -an
                           #-f flv rtmp://localhost:1935/webcam/mystream;
    #}

application mypush {

live on;

        # Every stream published here
        # is automatically pushed to
        # these two machines
        #push rtmp1.example.com;
        #push rtmp2.example.com:1934;

}

application mypull {

live on;

        # Pull all streams from remote machine
        # and play locally
        #pull rtmp://rtmp3.example.com pageUrl=www.example.com/index.html;

}

application mystaticpull {

live on;

        # Static pull is started at nginx start
        #pull rtmp://rtmp4.example.com pageUrl=www.example.com/index.html name=mystream static;

}

    # video on demand

application vod {

play /opt/www/vod;

}

application vod2 {

play /var/mp4s;

}

    # Many publishers, many subscribers
    # no checks, no recording
    #application videochat {

     #   live on;

        # The following notifications receive all
        # the session variables as well as
        # particular call arguments in HTTP POST
        # request

        # Make HTTP request & use HTTP retcode
        # to decide whether to allow publishing
        # from this connection or not
     #   on_publish http://localhost:8080/publish;

        # Same with playing
     #   on_play http://localhost:8080/play;

        # Publish/play end (repeats on disconnect)
     #   on_done http://localhost:8080/done;

        # All above mentioned notifications receive
        # standard connect() arguments as well as
        # play/publish ones. If any arguments are sent
        # with GET-style syntax to play & publish
        # these are also included.
        # Example URL:
        #   rtmp://localhost/myapp/mystream?a=b&c=d

        # record 10 video keyframes (no audio) every 2 minutes
      #  record keyframes;
      #  record_path /tmp/vc;
      #  record_max_frames 10;
      #  record_interval 2m;

        # Async notify about an flv recorded
      #  on_record_done http://localhost:8080/record_done;

    #}


    # HLS

    # For HLS to work please create a directory in tmpfs (/tmp/hls here)
    # for the fragments. The directory contents is served via HTTP (see
    # http{} section in config)
    #
    # Incoming stream must be in H264/AAC. For iPhones use baseline H264
    # profile (see ffmpeg example).
    # This example creates RTMP stream from movie ready for HLS:
    #
    # ffmpeg -loglevel verbose -re -i movie.avi  -vcodec libx264
    #    -vprofile baseline -acodec libmp3lame -ar 44100 -ac 1
    #    -f flv rtmp://localhost:1935/hls/movie
    #
    # If you need to transcode live stream use 'exec' feature.
    #
    application hls {
        live on;
        hls on;
        hls_path /opt/www/live;
    }

    # MPEG-DASH is similar to HLS

    #application dash {
    #    live on;
    #    dash on;
    #    dash_path /tmp/dash;
    #}
}

}

HTTP can be used for accessing RTMP stats

http {

server {

    listen      8081;

    location / {
        root /opt/www/;
    }

    # This URL provides RTMP statistics in XML
    location /stat {
        rtmp_stat all;

        # Use this stylesheet to view XML as web page
        # in browser
        rtmp_stat_stylesheet stat.xsl;
    }

    location /stat.xsl {
        # XML stylesheet to view RTMP stats.
        # Copy stat.xsl wherever you want
        # and put the full directory path here
        root /home/djstava/Workshop/Web/nginx-rtmp-module/;
    }

    location /hls {
        # Serve HLS fragments
        types {
            application/vnd.apple.mpegurl m3u8;
            video/mp2t ts;
        }

        root /opt/www/;
        add_header Cache-Control no-cache;
    }

    #location /dash {
        # Serve DASH fragments
    #    root /tmp;
    #    add_header Cache-Control no-cache;
    #}
}

}

在rtmp标签下,指定hls application的根路径/opt/www/live,所有的TS切片文件都存放在这里

ffmpeg推流
推送本地文件
ffmpeg -re -i /opt/www/vod/dhxy1.mp4 -vcodec copy -acodec copy -f flv -y rtmp://192.168.1.88/hls/livestream1
推送成功后,你可以通过如下2个url播放对应的模拟实时流,请确保nginx服务已启动。

rtmp://192.168.1.88/hls/livestream1
http://192.168.1.88:8081/live/livestream1.m3u8
另外 http://192.168.1.88:8081/stat 页面可以显示当前服务的一些信息，如接入的客户端数量、音频、视频的信息等等，见下图

nginx_stat

推送UDP组播数据
ffmpeg -i udp://@224.0.0.2:9000 -vcodec libx264 -acodec aac -strict -2 -f flv -s 1280x720 -q 10 -ac 1 -ar 44100 rtmp://192.168.1.88/hls/livestream
nginx_udp

在以UDP数据为输入源时，ffmpeg会报如下图中的错误信息

nginx_udp_error

这时只需要重新修改下ffmpeg的推流命令就可以，如下

ffmpeg -i ‘udp://@224.0.0.2:9000?fifo_size=2000000&overrun_nonfatal=1’ -vcodec libx264 -acodec aac -strict -2 -f flv -s 1280x720 -q 10 -ac 1 -ar 44100 rtmp://192.168.1.88/hls/livestream
fifo_size的单位是字节,自己酌情增减。

服务端多码率支持
nginx.conf
#user nobody;
worker_processes auto;

rtmp_auto_push on;

error_log logs/error.log;
error_log logs/error.log notice;
error_log logs/error.log info;

#pid logs/nginx.pid;

events {
worker_connections 1024;
}

rtmp {

server {

    listen 1935;

    chunk_size 4000;

    # TV mode: one publisher, many subscribers
    #application mytv {

        # enable live streaming
        #live on;

        # record first 1K of stream
        #record all;
        #record_path /tmp/av;
        #record_max_size 1K;

        # append current timestamp to each flv
        #record_unique on;

        # publish only from localhost
        #allow publish 127.0.0.1;
        #deny publish all;

        #allow play all;
    #}

    # Transcoding (ffmpeg needed)
    #application big {
    #    live on;

        # On every pusblished stream run this command (ffmpeg)
        # with substitutions: $app/${app}, $name/${name} for application & stream name.
        #
        # This ffmpeg call receives stream from this application &
        # reduces the resolution down to 32x32. The stream is the published to
        # 'small' application (see below) under the same name.
        #
        # ffmpeg can do anything with the stream like video/audio
        # transcoding, resizing, altering container/codec params etc
        #
        # Multiple exec lines can be specified.

    #    exec ffmpeg -re -i rtmp://localhost:1935/$app/$name -vcodec flv -acodec copy -s 32x32
                    #-f flv rtmp://localhost:1935/small/${name};
    #}

    #application small {
    #    live on;
    #    # Video with reduced resolution comes here from ffmpeg
    #}

    #application webcam {
    #    live on;

        # Stream from local webcam
    #    exec_static ffmpeg -f video4linux2 -i /dev/video0 -c:v libx264 -an
                           #-f flv rtmp://localhost:1935/webcam/mystream;
    #}

application mypush {

live on;

        # Every stream published here
        # is automatically pushed to
        # these two machines
        #push rtmp1.example.com;
        #push rtmp2.example.com:1934;

}

application mypull {

live on;

        # Pull all streams from remote machine
        # and play locally
        #pull rtmp://rtmp3.example.com pageUrl=www.example.com/index.html;

}

application mystaticpull {

live on;

        # Static pull is started at nginx start
        #pull rtmp://rtmp4.example.com pageUrl=www.example.com/index.html name=mystream static;

}

    # video on demand

application vod {

play /opt/www/vod;

}

application vod2 {

play /var/mp4s;

}

    # Many publishers, many subscribers
    # no checks, no recording
    #application videochat {

     #   live on;
     #   on_publish http://localhost:8080/publish;

        # Same with playing
     #   on_play http://localhost:8080/play;

        # Publish/play end (repeats on disconnect)
     #   on_done http://localhost:8080/done;

        # All above mentioned notifications receive
        # standard connect() arguments as well as
        # play/publish ones. If any arguments are sent
        # with GET-style syntax to play & publish
        # these are also included.
        # Example URL:
        #   rtmp://localhost/myapp/mystream?a=b&c=d

        # record 10 video keyframes (no audio) every 2 minutes
      #  record keyframes;
      #  record_path /tmp/vc;
      #  record_max_frames 10;
      #  record_interval 2m;

        # Async notify about an flv recorded
      #  on_record_done http://localhost:8080/record_done;

    #}


    # HLS
    application hls {
        live on;
        hls on;
        hls_path /opt/www/live;
        hls_nested on;

        hls_variant _low BANDWIDTH=800000;
        hls_variant _mid BANDWIDTH=1200000;
        hls_variant _hi  BANDWIDTH=2000000;
    }

    # MPEG-DASH is similar to HLS

    #application dash {
    #    live on;
    #    dash on;
    #    dash_path /tmp/dash;
    #}
}

}

HTTP can be used for accessing RTMP stats

http {

server {

    listen      8081;

    location / {
        root /opt/www/;
    }

    # This URL provides RTMP statistics in XML
    location /stat {
        rtmp_stat all;

        # Use this stylesheet to view XML as web page
        # in browser
        rtmp_stat_stylesheet stat.xsl;
    }

    location /stat.xsl {
        # XML stylesheet to view RTMP stats.
        # Copy stat.xsl wherever you want
        # and put the full directory path here
        root /home/djstava/Workshop/Web/nginx-rtmp-module/;
    }

    location /hls {
        # Serve HLS fragments
        types {
            application/vnd.apple.mpegurl m3u8;
            video/mp2t ts;
        }

        root /opt/www/;
        add_header Cache-Control no-cache;
    }

    #location /dash {
        # Serve DASH fragments
    #    root /tmp;
    #    add_header Cache-Control no-cache;
    #}
}

}
主要看看application hls的内容

application hls {
live on;
hls on;
hls_path /opt/www/live;
hls_nested on;

hls_variant _low BANDWIDTH=800000;
hls_variant _mid BANDWIDTH=1200000;
hls_variant _hi  BANDWIDTH=2000000;

}
这里设定当带宽分别为800k、1200k、2000k的时候，终端都播放相对应的m3u8索引文件

ffmpeg推流
这里需要利用ffmpeg推送3路不同的流，对应上面提到的低、中、高

ffmpeg -re -i ~/Videos/xjcy.mp4 -vcodec copy -acodec copy -b:v 800k -b:a 32k -f flv rtmp://10.10.10.59/hls/livestream_low
ffmpeg -re -i ~/Videos/xjcy.mp4 -vcodec copy -acodec copy -b:v 1200k -b:a 64k -f flv rtmp://10.10.10.59/hls/livestream_mid

ffmpeg -re -i ~/Videos/xjcy.mp4 -vcodec copy -acodec copy -b:v 2000k -b:a 128k -f flv rtmp://10.10.10.59/hls/livestream_hi
推送开始后，hls的root目录下就会生成相应的文件内容，如下图所示

nginx_ffmpeg_variant

此时livestream.m3u8文件内容为

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=800000
livestream_low/index.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1200000
livestream_mid/index.m3u8
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=2000000
livestream_hi/index.m3u8
相应的，播放链接为 http://10.10.10.59:8081/live/livestream.m3u8，播放器需要做的就是根据自身的网络状况，切换到其它的索引文件。

直播节目的录制
直播进行的同时一般都会有本地录制的需求，nginx-rtmp-module提供了这个功能，接下来实践一下。还是看nginx.conf配置文件

application hls {
live on;
hls on;
hls_path /opt/www/live;
hls_nested on;

hls_variant _low BANDWIDTH=800000;
hls_variant _mid BANDWIDTH=1200000;
hls_variant _hi  BANDWIDTH=2000000;

recorder all {
    record all;
    record_suffix -%Y-%m-%d-%H_%M_%S.flv;
    record_max_size 200000K;
    record_path /opt/www/record;
}

}
record all录制所有内容，也可以只录音频或者视频。

推流后/opt/www/record路径下就会自动生成带对应时间戳的flv文件，用vlc测试播放OK

nginx_ffmpeg_record

时移电视
要想实现时移电视(这里指的是服务器端)的话，首先需要在服务器上保留足够的切片文件，比如说你提供1小时的时移，就意味着要有1小时的切片文件，而且索引文件中包含前1小时的切片序列

application hls {
live on;
hls on;
hls_path /opt/www/live;
hls_continuous on;
hls_sync 100ms;
hls_nested on;
hls_playlist_length 5m;
hls_fragment 10s;

hls_variant _low BANDWIDTH=800000;
hls_variant _mid BANDWIDTH=1200000;
hls_variant _hi  BANDWIDTH=2000000;

#exec /home/djstava/Workshop/Web/nginx-1.11.3/build/test.sh;
#exec_kill_signal term;

#recorder all {
#    record all;
#    record_suffix -%Y-%m-%d-%H_%M_%S.flv;
#    record_max_size 6200000K;
#    record_path /opt/www/record;
#}

}

hls_fragment指的是切片文件的长度，这里是10秒，hls_playlist_length指的是索引文件的长度，我这里设的是5分钟。推流开始后，你到切片生成的目录，会发现*.m3u8文件包含了30个ts序列。所以，在上面这种情况下，就只能进行5分钟的时移，当播放进度到达当前直播点时则继续回到直播状态

执行外部shell脚本
比如有个脚本test.sh，内容如下

#!/bin/bash

on_die ()
{
# kill all children
pkill -KILL -P $$
}

trap ‘on_die’ TERM
ffmpeg -re -i /home/djstava/Videos/ygdx.mp4 -vcodec copy -acodec copy -f flv rtmp://10.10.10.48/hls/ygdx &
wait
我这里把它放在hls application中执行,则nginx.conf应如下

application hls {
live on;
hls on;
hls_path /opt/www/live;
hls_continuous on;
hls_sync 100ms;
hls_nested on;
hls_playlist_length 5m;
hls_fragment 10s;

hls_variant _low BANDWIDTH=800000;
hls_variant _mid BANDWIDTH=1200000;
hls_variant _hi  BANDWIDTH=2000000;

exec /home/djstava/Workshop/Web/nginx-1.11.3/build/test.sh;
exec_kill_signal term;

#recorder all {
#    record all;
#    record_suffix -%Y-%m-%d-%H_%M_%S.flv;
#    record_max_size 6200000K;
#    record_path /opt/www/record;
#}

}
当hls服务正常启动时(如上面写过的ffmpeg推流动作),外部脚本test.sh也被执行了。脚本中捕捉了退出的中断信号，也就说，如果ffmpeg推流动作中断了，那么test.sh脚本也就不再执行了

制作RAMDISK
为了提高HLS的读写效率,可以把切片和索引文件操作放在内存中进行.

mount -t tmpfs -o size=512m tmpfs /opt/www/live

https://xugaoxiang.com/2020/01/19/nginx-rtmp-hls/

参考文献
https://developer.apple.com/streaming/
https://github.com/arut/nginx-rtmp-module
https://github.com/arut/nginx-rtmp-module/wiki/Directives