Android RTMP直播(续)

作者: 342f294a05c1 | 来源:发表于2017-06-01 14:06 被阅读529次

    软硬件环境

    • ubuntu 16.04
    • Android Studio 2.1.3
    • OTT BOx with android 5.1.1
    • nginx 1.11.3
    • nginx-rtmp-module
    • VLC

    前言

    之前的一篇博文http://www.xugaoxiang.com/blog/index.php/archives/66/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.

    参考文献

    1 https://developer.apple.com/streaming/

    2 https://github.com/arut/nginx-rtmp-module

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        本文标题:Android RTMP直播(续)

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